| 00:00.35 | Crashsys | Do TDM400's echo very bad? (Compared to an X100?) |
| 00:00.42 | Crashsys | Or is anything going to pots going to echo? |
| 00:00.47 | Crashsys | somewhat |
| 00:03.49 | rollergrrl | In the console, is there a way to view which group a zaptel channel is in? |
| 00:04.09 | Crashsys | zap show channel <#> |
| 00:04.40 | Crashsys | ie, zap show channel 2 |
| 00:05.02 | rollergrrl | Guess I'm blind |
| 00:05.07 | rollergrrl | but I can't see which zap group it's in |
| 00:06.14 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:06.15 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:07.41 | X-Rob_ | rollergrrl, uh. Doesn't look like it |
| 00:07.58 | [TK]D-Fender | Crashsys : Any card touching analog (effectively ALL) is susceptable. Zaptel EC has improved a lot but can still be hit/miss |
| 00:08.18 | [TK]D-Fender | Crashsys : Hardware EC is preferable wherever possible |
| 00:09.03 | Crashsys | fender: Yeah, but i'll have to see what the budget dictates... |
| 00:09.12 | Crashsys | KB1 the best general EC? |
| 00:09.15 | Crashsys | software wise |
| 00:10.06 | [TK]D-Fender | Crashsys : MG2 I think in most cases (the default with Zaptel these days IIRC) |
| 00:10.17 | [TK]D-Fender | Crashsys : Are you having current problems? |
| 00:10.20 | Crashsys | Thank ya... |
| 00:11.00 | Crashsys | Yeah, but it has more to do with the shitstreams... |
| 00:11.08 | Crashsys | I mean crapstreams... |
| 00:11.54 | Crashsys | And i'm sure the X100's aint helping it any... so while i'm talking them into getting the polycom's like I originally suggested, i'm going to talk them into a TDM400 too... |
| 00:12.25 | Crashsys | Prolly end up ebay'ing the grandstreams... |
| 00:12.29 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 00:12.29 | [TK]D-Fender | Crashsys : GXP's? |
| 00:12.39 | Crashsys | Unless someone wants to buy 6 BT-200's... used for 1-week... |
| 00:13.00 | Crashsys | It uses the same boot/firmwares as the GXP's... |
| 00:13.23 | [TK]D-Fender | Crashsys 1.0.1.3 firmware for GXP's had serious echo issues even sip-sip |
| 00:13.56 | Crashsys | They get bad echo when they initiate a call from the GS's... otherwise it's fine... |
| 00:15.00 | justinu|laptop | GS are mics are too hot then |
| 00:17.18 | Crashsys | Yeah, but I dont think they can be adjusted... and setting the gain down in zapata makes everything sound like your talking down a hallway... |
| 00:18.54 | grexk | How can I handle priority jumping in RemoveQueueMember? |
| 00:19.51 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:23.43 | grexk | n+101 is not working? |
| 00:24.10 | justinu|laptop | Crashsys: setting your txgain lower didn't help? |
| 00:27.51 | Crashsys | justin: Marginally... |
| 00:28.05 | Crashsys | But by the time the echo fades out, you can barely hear anyone... |
| 00:28.37 | justinu|laptop | ack |
| 00:29.25 | Crashsys | Sounds like someone talking down a long hallway |
| 00:29.30 | Crashsys | You can hear them, but it's faint... |
| 00:41.39 | *** join/#asterisk scoona (n=aaaazz@pool-71-245-225-219.bstnma.fios.verizon.net) |
| 00:43.03 | De_Mon | does anyone have a simple openSER config for doing SIP/UDP <-> SIP/TCP |
| 00:44.41 | Nivex | is anyone else having problems registering to northamerica.sipphone.com lately? |
| 00:46.24 | justinu|laptop | crashsys: what increments are you adjusting gains? |
| 00:46.31 | Crashsys | by .5 |
| 00:46.36 | justinu|laptop | ah, that's pretty fine |
| 00:46.53 | Crashsys | well, I rough it in with whole increments, then tune with .5... |
| 00:47.17 | justinu|laptop | just making sure you didn't think they were db or something |
| 00:47.40 | Crashsys | well, I did think I was adjusting DB or gain... but I still used the same method... |
| 00:47.51 | Crashsys | Ie -2.5 was -2.5db of gain |
| 00:48.38 | justinu|laptop | negative |
| 00:49.16 | justinu|laptop | it's something arbitrary like +/-100% of the capability of the audio driver on the analog cards |
| 00:49.36 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 00:50.04 | Crashsys | Someone needs to make a multi-chan ZTMonitor... |
| 00:50.18 | Crashsys | or a way to adjust gains per channel... |
| 00:50.30 | Crashsys | for those poor saps using analog like me |
| 00:50.35 | justinu|laptop | i thought you could adjust gains per channel |
| 00:50.48 | Crashsys | If you can, i'm all ears... |
| 00:50.53 | Crashsys | I've only seen a globalized gain setting |
| 00:50.56 | Ryushin | So how do I record a company by name directory? |
| 00:51.27 | justinu|laptop | i'm not an expert, but I thought you could put a gain statement for each channel def in zapata? |
| 00:51.43 | Crashsys | ... |
| 00:52.07 | Crashsys | well, I can try :) |
| 00:52.20 | justinu|laptop | i remember doing it for an E&M trunk that was interfaced to a legacy PBX |
| 00:52.34 | justinu|laptop | for some reason we had to adjust half the channels to a different gain |
| 00:52.55 | Crashsys | god i'm fried... would I list the gain before or after the channel statement that I want effected? |
| 00:53.35 | *** join/#asterisk micky (n=micky@zandrox.org) |
| 00:54.09 | puzzled | Crashsys: iirc before |
| 00:54.26 | justinu|laptop | and rememer, reloading chan_zap.so will not read in the new gain parms |
| 00:54.29 | justinu|laptop | you must restart ast |
| 00:54.44 | Crashsys | Yeah... |
| 00:54.53 | Crashsys | wish I had verizon's MW test # |
| 00:54.57 | Crashsys | make this easier |
| 00:55.48 | justinu|laptop | oh yeah, because half the channels were TIE trunks, and the other half was to PSTN |
| 00:55.50 | justinu|laptop | funky stuff |
| 01:04.03 | *** join/#asterisk QMario (n=QMario@unaffiliated/QMario) |
| 01:06.56 | Crashsys | anyone know how I get ztmonitor to go into quantitative mode? |
| 01:11.05 | micky | Hello, i'm trying to convert a PCM (wav) stream resulted from an asterisk channel to an MP3 compatible with flash (32Kbit/s and 22.025 Khz) using lame and stream it in ices. |
| 01:11.23 | micky | <PROTECTED> |
| 01:22.24 | Nivex | micky: chipmunk effect?... is the encoder expecting stereo and you're feeding it mono? |
| 01:27.45 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 01:28.28 | *** join/#asterisk JSabines (i=JSabines@201.153.99.192) |
| 01:29.49 | *** join/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com) |
| 01:30.03 | blebleble | my asterisk box just died, and when i restarted it and login, and try to do a sip show registry i get "No such command 'sip' (type 'help' for help)" |
| 01:31.13 | *** join/#asterisk NetNut404 (n=netnut40@adsl-66-159-224-232.dslextreme.com) |
| 01:31.55 | micky | Nivex stream plays fine in winamp but not in flash... and lame is set to (/usr/local/bin/lame -r -s 8 --cbr -b 24 -x -m m - - | /usr/local/bin/ezstream -c $1) this way it works on anything but flash wich requires the -r setting to be 5 11 22.025 22.5 or 44 .... |
| 01:31.58 | De_Mon | yuck |
| 01:32.11 | micky | Nivex oups the -s setting |
| 01:32.29 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 01:32.50 | micky | Nivex and if i set it to -s (flash compatible-> 5 11 22.025 44) the stream gets the chipmunk effect... in ANY player. |
| 01:33.14 | Nivex | *shrug* I know squat about flash, so I was just taking a shot in the dark |
| 01:33.24 | NetNut404 | has anyone found a way to add the amr codec to asterisk? |
| 01:33.31 | mog | ? |
| 01:38.19 | *** join/#asterisk avar (n=avar@wikipedia/Aevar-Arnfjord-Bjarmason) |
| 01:38.22 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.76.Dial1.SanJose1.Level3.net) |
| 01:38.39 | Ryushin | Are there any ideas why when I try to dial long distance, I get a message saying I must first dial a 1. |
| 01:38.56 | blebleble | Ryshin: did you dial a 1 |
| 01:39.35 | Ryushin | Yea. |
| 01:39.45 | *** join/#asterisk Werdna (i=Andrew@silentflame/member/Werdna) |
| 01:40.20 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 01:40.39 | Ryushin | I have a line that looks like this: exten => _1NXXNXXXXXX,1,Dial(ZAP/1/${EXTEN}) |
| 01:40.58 | Ryushin | This worked fine for a PRI, but this is going out a analog on a different asterisk system. |
| 01:42.46 | NetNut404 | how do I add an amr codec to asterisk ? |
| 01:42.57 | DrukenHME | Ryushin: try putting a second one infront of it |
| 01:43.59 | Ryushin | Yea, I just did that, and it gave me a fast busy. |
| 01:44.00 | Ryushin | Odd. |
| 01:44.34 | *** part/#asterisk avar (n=avar@wikipedia/Aevar-Arnfjord-Bjarmason) |
| 01:47.50 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-2.cybersurf.com) |
| 01:51.19 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:58.06 | NetNut404 | I read at http://www.voip-info.org/wiki/view/Asterisk+H324M that If I get the AMR codec in asterisk that my phone should then be able to talk to "dumb clients" .. how would I do that? |
| 02:01.06 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:04.16 | Ryushin | How do I put a pause in the outbound dial. I think it's dial to fast and it's dropping the 1. |
| 02:04.56 | grexk | How do I remove queues from CLI? |
| 02:04.58 | *** join/#asterisk JSabines (n=alancast@201.153.99.192) |
| 02:05.27 | justinu|laptop | Ryushin: "w" |
| 02:05.37 | Ryushin | Cool, thanks. |
| 02:07.54 | Ryushin | That was it. It just needed a .5 second pause. Thanks justinu|laptop. |
| 02:08.09 | justinu|laptop | np |
| 02:11.46 | *** join/#asterisk Druken (n=jdumais@72.58.232.61) |
| 02:13.47 | hacked`` | guys, i was wondering if anyone can spare a couple minutes and help me get my trunk set up |
| 02:13.58 | NetNut404 | I have a sip client on a cell phone, but it uses the AMR codec and H.263 . How to make this talk to other sip clients through asterisk? (because the codec) |
| 02:14.17 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 02:23.45 | *** join/#asterisk [Outcast] (n=outcast@222-154-72-242.jetstream.xtra.co.nz) |
| 02:24.47 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 02:26.43 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 02:42.33 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 02:46.34 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 02:48.43 | *** join/#asterisk xxi (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 02:51.15 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 02:53.22 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 02:53.49 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 02:53.49 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 02:56.00 | *** join/#asterisk matt_ (n=Matt@82-33-68-44.cable.ubr01.trow.blueyonder.co.uk) |
| 02:56.56 | matt_ | hello, can somebody help me, i have a spa3000 box hooked up to asterisk and dialout works fine but incomming calls dont seem to work |
| 02:57.25 | matt_ | when i dial the pstn number the spa3000 box picks up and play what sounds like a congestion tone down the line |
| 02:57.47 | matt_ | my guess is that the spa3000 dosn't know where to send the call |
| 02:58.11 | matt_ | i have tried setting a default dialplan for the pstn line but it didn't seem to work |
| 02:59.57 | *** join/#asterisk freeepbxxnoobbb (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com) |
| 03:00.53 | freeepbxxnoobbb | Can someone tell me the price range on if my company were to buy pri lines in a colo |
| 03:01.45 | freeepbxxnoobbb | like a t1 pri |
| 03:04.55 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 03:05.11 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 03:05.36 | tengulre | HI,all |
| 03:05.42 | `Tingles` | 15.00/line + some phone pacakge ontop for minutes.. |
| 03:05.45 | `Tingles` | approx |
| 03:06.06 | `Tingles` | that would be for DID lines... |
| 03:12.21 | *** join/#asterisk ojai (n=foo@ca-ventura-cuda1-c1a-233.vnnyca.adelphia.net) |
| 03:14.37 | hacked`` | guys |
| 03:14.39 | hacked`` | when i do, sip show registry, and i see: 514.teliphone.ca:5060 BYOD0000154 23 Registered |
| 03:14.41 | hacked`` | is that a good thing ? |
| 03:15.14 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:15.15 | Druken | yes |
| 03:15.17 | freeepbxxnoobbb | would that be unlimited local calling after that |
| 03:15.50 | Strom_C | freeepbxxnoobbb: it depends which state you're in |
| 03:16.10 | freeepbxxnoobbb | ca |
| 03:16.13 | hacked`` | ya but how come i still get all circuits busy then |
| 03:16.15 | freeepbxxnoobbb | California |
| 03:16.28 | Strom_C | freeepbxxnoobbb: in california, for example, i would expect to pay between $500 and $700 per month plus usage for a fully populated PRI |
| 03:16.52 | Strom_C | oh? where in california? |
| 03:17.24 | freeepbxxnoobbb | orange county\ |
| 03:17.30 | Strom_C | I'm in los angeles :) |
| 03:17.40 | Druken | hacked``: registering means they know where you are.... for incoming calls... |
| 03:17.42 | Druken | not outgoing.... |
| 03:17.58 | freeepbxxnoobbb | yeah if we were to get lines in la how much would it cost |
| 03:18.17 | Strom_C | it's state-level regulation, not county or city-level |
| 03:18.26 | Strom_C | the CPUC is a state agency :) |
| 03:19.03 | freeepbxxnoobbb | Oh so if we were to get a t1 with 24 lines and what would it cost for unlimited local calling |
| 03:19.07 | Strom_C | how much call volume are you anticipating handling? |
| 03:19.50 | freeepbxxnoobbb | All day, So we were seeing if it would be more cost effective to terminate our own lines |
| 03:19.59 | Strom_C | what's "All day" |
| 03:20.16 | Strom_C | how many minutes per month do your traffic engineering estimates show you as using? |
| 03:21.06 | ojai | I'm trying to figure out what hardware I need to get Asterisk going at home. I've got Vonage and only one phone line. I figure the X100P for FXO but not sure what for FXS. The TDM10B? |
| 03:21.11 | freeepbxxnoobbb | about 20000000 |
| 03:21.33 | Strom_C | freeepbxxnoobbb: TWENTY MILLION MINUTES?! |
| 03:21.35 | hacked`` | druken, what are you talking about, i didnt set up incoming yet, only outgoing |
| 03:21.55 | freeepbxxnoobbb | I meant about a million |
| 03:22.00 | freeepbxxnoobbb | sorry typo |
| 03:22.07 | freeepbxxnoobbb | did my math wrong |
| 03:22.11 | Strom_C | freeepbxxnoobbb: um, ok... |
| 03:22.27 | Strom_C | so basically you really have no clue how much traffic you're going to be generating |
| 03:22.37 | freeepbxxnoobbb | no not really |
| 03:22.38 | ojai | so is the hardware investment really ~$160US? |
| 03:22.46 | Strom_C | ojai: just get a TDM22B |
| 03:22.53 | Strom_C | ojai: two FXS, two FXO |
| 03:23.09 | JT | Strom_C: a million minutes is about 100% utilisation on 23 channels |
| 03:23.13 | JT | 24/7 |
| 03:23.20 | freeepbxxnoobbb | yeah |
| 03:23.34 | freeepbxxnoobbb | but actually 4 t1 |
| 03:23.47 | freeepbxxnoobbb | not 24 hours a day |
| 03:23.48 | Strom_C | freeepbxxnoobbb: what kind of business are you running? |
| 03:24.05 | Crashsys | 1-900-hot-asterisk |
| 03:24.13 | Strom_C | ojai: you generally dont want to do somethng as horrendous as using vonage through an fxo port |
| 03:24.13 | freeepbxxnoobbb | psssssssssh |
| 03:24.22 | ojai | Strom_C: the TDM22B is both the FXO and the FXS? |
| 03:24.28 | Strom_C | ojai: yes |
| 03:24.35 | Strom_C | thats what I already said |
| 03:24.50 | ojai | nice -- how much does that run do you know? |
| 03:25.10 | Strom_C | file: when are we going to get sushi |
| 03:25.19 | freeepbxxnoobbb | would the telco be charging us per minute if we call local |
| 03:25.19 | Crashsys | Sashimi! |
| 03:25.21 | Crashsys | good stuff |
| 03:25.29 | Qwell | Strom_C: come to Boston. Be here in like...8 minutes |
| 03:25.51 | Strom_C | freeepbxxnoobbb: no telco is going to sell you an unlimited plan with that kind of usage |
| 03:26.11 | Strom_C | freeepbxxnoobbb: you can. however, get rather cheap per-minute rates from many of the major interexchange carriers |
| 03:26.12 | JT | it's lunchtime in the us? |
| 03:26.26 | Strom_C | Qwell: well let me just walk right on over ther |
| 03:26.29 | Strom_C | s/ther/there/ |
| 03:26.32 | Crashsys | but minutes in allotments... |
| 03:26.32 | Qwell | Strom_C: sure |
| 03:26.40 | Crashsys | and hope you dont go over... |
| 03:26.42 | freeepbxxnoobbb | we were trying to get cheaper then 1/2 a penny |
| 03:27.05 | Strom_C | i don't think you're going to get less than half a cent at only one million minutes per month |
| 03:27.11 | file | Strom_C: yeah just listen for me singing... follow the sound |
| 03:27.28 | JT | Strom_C: what time is it over there? |
| 03:27.31 | freeepbxxnoobbb | even if we terminate our own lines |
| 03:27.37 | Strom_C | it's half past eight in los angeles |
| 03:27.46 | Strom_C | freeepbxxnoobbb: what do you mean "terminate our own lines" |
| 03:27.54 | JT | am or pm, it's important in this case :P |
| 03:27.59 | Strom_C | pm |
| 03:28.01 | JT | ah ok |
| 03:28.07 | freeepbxxnoobbb | we wanted to setup our own gateways |
| 03:28.27 | Crashsys | free: You mean like set up your own exchange in the CO? |
| 03:28.51 | Strom_C | yeah - set up your own full-blown SS7-enabled interconnect? |
| 03:29.28 | freeepbxxnoobbb | something like that |
| 03:29.57 | Strom_C | the phrase "biting off way more than you can chew" comes to mind |
| 03:30.09 | freeepbxxnoobbb | you gotta aim big right |
| 03:30.25 | freeepbxxnoobbb | dont know where to start tho\ |
| 03:30.28 | Strom_C | freeepbxxnoobbb: you haven't even done traffic engineering estimates yet |
| 03:30.30 | Strom_C | do that first |
| 03:30.42 | Crashsys | If your gonna be your own Exchange, sell yourself the minutes...? |
| 03:30.45 | Crashsys | heh |
| 03:30.51 | Qwell | Strom_C: What? People won't just give him money, without a business? |
| 03:30.58 | Qwell | Are you telling me that the Simpsons lied to me? |
| 03:31.03 | freeepbxxnoobbb | lol |
| 03:31.05 | file | Qwell: never! |
| 03:31.39 | freeepbxxnoobbb | coz we cant find any provider with the gateway we need |
| 03:31.55 | freeepbxxnoobbb | so we figure to terminate them ourselves |
| 03:32.23 | Strom_C | freeepbxxnoobbb: what the hell kind of business are you running? are you trying to be the next Vonage or something? |
| 03:33.19 | freeepbxxnoobbb | no we just wanted to sell our faxing service |
| 03:33.36 | Qwell | faxing service? |
| 03:34.03 | freeepbxxnoobbb | but no one has a t38 or t37 gateways |
| 03:34.19 | file | VoIP with Vonage! |
| 03:34.27 | Crashsys | Faxing service... as in FaxSpam? |
| 03:34.34 | Crashsys | like "Eat at joe's, he has crabs"? |
| 03:34.36 | Qwell | or...like...efax? |
| 03:34.50 | Strom_C | oh please god, not efax again |
| 03:34.55 | Crashsys | lol |
| 03:35.22 | Qwell | I hereby coin the term eTelephony |
| 03:35.25 | freeepbxxnoobbb | not spam just the service |
| 03:35.26 | Strom_C | that thing blows rabid donkeys for food stamps in Pyongyang back alleys |
| 03:35.33 | file | Qwell: don't make me throw a pillow at you |
| 03:35.47 | Strom_C | freeepbxxnoobbb: so who is your intended client? |
| 03:35.54 | Qwell | normally, this is where I'd say "nope, can't" |
| 03:35.56 | Qwell | but...yeah |
| 03:35.59 | Strom_C | s/client/customer/ |
| 03:36.10 | file | I have a pillow I don't care about either |
| 03:36.12 | freeepbxxnoobbb | businesses that use email to fax |
| 03:36.15 | Qwell | eeps |
| 03:36.58 | Strom_C | freeepbxxnoobbb: at this point, my best advice to you is: |
| 03:37.01 | Strom_C | ~hafc |
| 03:37.03 | jbot | i guess hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 03:37.14 | Qwell | Strom_C: I knew you'd use that |
| 03:37.25 | Strom_C | yeah |
| 03:37.29 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:37.30 | Strom_C | it's like rtfm, only better |
| 03:37.50 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 03:38.49 | *** join/#asterisk Werdna (i=Andrew@silentflame/member/Werdna) |
| 03:38.57 | freeepbxxnoobbb | so it would not be cost effective? |
| 03:39.21 | freeepbxxnoobbb | 1 company was offering 1/2 a penny per fax sent. |
| 03:39.54 | Qwell | well, how long does it take to transmit a single page? |
| 03:40.06 | file | depends on negotiated speed |
| 03:40.14 | Qwell | exactly, so |
| 03:40.14 | Strom_C | "per fax sent" is a baaaad way to measure it |
| 03:40.31 | freeepbxxnoobbb | 30 seconds to 2 1/2 minutes |
| 03:40.34 | Strom_C | because they may intend on sending 300-page contracts by fax |
| 03:40.50 | file | Qwell: FAST |
| 03:40.50 | Qwell | freeepbxxnoobbb: So, 2.5 minutes, .5c/page |
| 03:40.51 | arcanine | correct me f im wrong, dialogic vfx's are also supported by asterisk |
| 03:40.53 | Qwell | uhh... |
| 03:41.00 | Qwell | .5 / 2.5 = ? |
| 03:41.07 | freeepbxxnoobbb | i know |
| 03:41.12 | freeepbxxnoobbb | how do they do it |
| 03:41.12 | Qwell | That's your required cost per minute |
| 03:42.13 | Qwell | plus, obviously, you need some infrastructure in place to handle it |
| 03:43.18 | freeepbxxnoobbb | I mean how can they hand out those rates ? I tried doing one and was not cost effective so i thought if we terminate it ourselves it would be cheaper |
| 03:43.50 | file | Qwell: can YOU mambo? |
| 03:44.02 | Qwell | mmhmm |
| 03:46.48 | arcanine | does sum1 used a dialogic vfx on asterisk? |
| 03:48.57 | Qwell | Is it bad that I know this? |
| 03:49.49 | X-Rob | Qwell, I'm being a bastard here, but at 300 baud you can get 1200bps. |
| 03:50.03 | X-Rob | baud != bps |
| 03:50.21 | Qwell | huh? |
| 03:50.29 | *** join/#asterisk bmg505 (n=leon@c1-54-16.rndf.isadsl.co.za) |
| 03:50.34 | Strom_C | PEDANT ALERT |
| 03:50.39 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 03:50.43 | Strom_C | *sirens* |
| 03:50.50 | Strom_C | *klaxons* |
| 03:50.58 | X-Rob | D'oh. |
| 03:51.01 | X-Rob | That wouldbe me. |
| 03:51.48 | matt_ | X-Rob: how can you get 1200bps at 300 baud ? |
| 03:51.58 | X-Rob | http://www.totse.com/en/technology/telecommunications/bitsbaud.html |
| 03:52.07 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 03:52.22 | matt_ | baud is the number of signal changes perl second |
| 03:52.25 | matt_ | *per |
| 03:52.34 | Qwell | X-Rob: Who said 300bps? |
| 03:52.45 | X-Rob | * Qwell spams freeepbxxnoobbb 5,000 faxes at 300 baud |
| 03:52.52 | matt_ | you said 300 baud |
| 03:53.11 | file | eep |
| 03:53.15 | file | you saw nothing! |
| 03:53.21 | file | Qwell: you are NOT hearing me sing |
| 03:53.22 | matt_ | so you only have 300 signal changes so how can that be 1200bps with only 2 states |
| 03:53.23 | file | honest. |
| 03:53.24 | matt_ | a 0 and a 1 |
| 03:53.26 | matt_ | or on and off ? |
| 03:53.32 | X-Rob | My problem is that I've just spent the last hour fixing XSS and potential SQL injection issues |
| 03:53.40 | X-Rob | matt_, quick answer: FSK. Long answer, read the URL I pasted |
| 03:54.05 | matt_ | 300 bps |
| 03:54.07 | matt_ | ----------------------- = 30 characters per second |
| 03:54.19 | matt_ | theres like 8 bits in each character |
| 03:54.25 | X-Rob | depends on your parity, stop bits and data bits. |
| 03:54.30 | matt_ | where did you get 1200 from ? |
| 03:54.38 | matt_ | yea 8 max |
| 03:54.42 | matt_ | as then so 7 |
| 03:54.43 | X-Rob | 7N1 is more characters |
| 03:55.12 | X-Rob | A V22 modem (1200bps) runs at 300 baud. |
| 03:55.22 | X-Rob | or is that V24? |
| 03:55.24 | X-Rob | God, it's been years. |
| 03:55.31 | matt_ | 1200 / 8 = 150 |
| 03:55.47 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 03:56.33 | JT | just to reinterate, yes, baud != bps :P |
| 03:56.47 | Qwell | my question is, who ever said bps? |
| 03:56.51 | X-Rob | No one. |
| 03:56.55 | X-Rob | It's all my fault. |
| 03:57.00 | hacked`` | guys |
| 03:57.02 | JT | that's a good question |
| 03:57.03 | hacked`` | whats the path to SIP.conf |
| 03:57.10 | matt_ | <X-Rob> Qwell, I'm being a bastard here, but at 300 baud you can get 1200bps. |
| 03:57.17 | X-Rob | Oh |
| 03:57.20 | X-Rob | actually, I was right |
| 03:57.24 | X-Rob | I did say it |
| 03:57.26 | X-Rob | totally in context |
| 03:57.28 | matt_ | lol |
| 03:57.36 | Qwell | I'm...confused |
| 03:57.45 | X-Rob | I was implying that '300 baud isn't as slow as you think it is' |
| 03:57.53 | Qwell | it's slow enough |
| 03:58.29 | Qwell | yes |
| 03:58.32 | X-Rob | I'm going to shut up now. |
| 03:58.36 | Qwell | X-Rob: k |
| 04:00.03 | *** join/#asterisk pdt (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
| 04:00.58 | Hymie | uniden uip200 users, unite!! |
| 04:01.34 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 04:01.36 | znoG | anyone using IAXmodem? |
| 04:01.46 | pdt | ya |
| 04:01.51 | znoG | latest version? |
| 04:01.56 | pdt | not sure |
| 04:01.59 | *** join/#asterisk dayannn (n=dayannnn@200-233-253-248.xd-dynamic.ctbcnetsuper.com.br) |
| 04:02.06 | znoG | (0.1.14) |
| 04:02.09 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 04:02.17 | dayannn | anyone can help me with asterisk addons install? |
| 04:02.25 | znoG | i upgraded from 0.1.8 i think to 0.1.14, same config, and now when I ring iaxmodem, it rings endlessly |
| 04:02.28 | znoG | hylafax never picks up |
| 04:02.30 | pdt | 0.1.9.2 |
| 04:02.47 | pdt | 0.1.9-2 i mean (rpm version) |
| 04:02.59 | pdt | why upgrade? |
| 04:04.46 | znoG | because there's a few fixes to it that were done recently which got me interested in upgrading |
| 04:06.15 | dayannn | anyone can help me with mysql and asterisk integration? |
| 04:07.10 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 04:10.06 | znoG | pdt: found the prob... faxgetty not running :) |
| 04:10.38 | pdt | doh! |
| 04:10.40 | pdt | ya that'll do it |
| 04:12.10 | znoG | what hardware do you use to receive faxes, pdt? |
| 04:12.27 | pdt | Sangoma A200D and Sangoma A101/A102 |
| 04:12.37 | znoG | ah.. thats probably why you get good quality with it |
| 04:12.45 | znoG | i get a looooooot of bad pixel counts from hylafax |
| 04:13.12 | pdt | I was having huge problems on analog faxing until I started using IAXModem |
| 04:13.15 | pdt | since then it has been good |
| 04:13.22 | pdt | the PRI stuff seems to work pretty solid as well |
| 04:13.22 | znoG | yeah |
| 04:13.30 | znoG | if i send plain text faxes to it, it *mostly* works |
| 04:13.37 | znoG | but i just sent a fax with some graphics in it |
| 04:13.47 | znoG | and it fails right away with bad pixel counts all over the shop |
| 04:13.47 | pdt | what card you using |
| 04:13.55 | znoG | a TDM400 |
| 04:14.07 | dayannn | how to install asterisk-addons? |
| 04:14.13 | dayannn | have a lot of versions |
| 04:14.22 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 04:14.27 | dayannn | my asterisk is 1.2.11 |
| 04:15.09 | znoG | dayannn: www.voip-info.org |
| 04:16.37 | dayannn | znoG i try but i dont found a information to my |
| 04:17.26 | pdt | znoG: you can send me a test fax if you want to see if it's just you or a "normal" problem |
| 04:17.40 | *** join/#asterisk LoneShadow (n=duh@59.92.141.160) |
| 04:20.16 | znoG | pdt: thanks, i appreciate the offer. I haven't got outgoing faxing setup just yet, only incoming. |
| 04:20.27 | znoG | I am sending from a laptop though, not sure if it would be different from a fax machine. |
| 04:20.43 | pdt | might be, are you sending through the pbx or from a real land line? |
| 04:20.46 | *** join/#asterisk ATGeek (n=atg@pdpc/supporter/student/ATravelingGeek) |
| 04:22.37 | znoG | real land line |
| 04:22.48 | znoG | it looks like after retraining many times over, the fax starts to go through with 0 bad lines |
| 04:23.46 | pdt | it's just like a dog... maybe you should feed the pbx a treat |
| 04:24.08 | ojai | Strom_C: sorry -- just getting back -- sorry for the stupid question but what's wrong w/ using vonage though an fxo port? |
| 04:24.40 | znoG | pdt: :) |
| 04:27.59 | *** join/#asterisk Werdna_ (i=Andrew@silentflame/member/Werdna) |
| 04:30.45 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 04:31.12 | dayannn | znog i dont found the especific information |
| 04:31.33 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 04:31.33 | dayannn | anyone have xperience with asterisk-addons and debian 3.1 r2? |
| 04:33.56 | *** join/#asterisk EvilDeshi (n=rattfink@pool-71-117-41-52.mdsnwi.dsl-w.verizon.net) |
| 04:36.11 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 04:37.03 | EvilDeshi | is there an asterisk command i can use to play a audio stream back to a caller? |
| 04:38.36 | EvilDeshi | nm |
| 04:54.46 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 04:56.35 | stephane_ | jour |
| 04:58.16 | rory|zomgzorz | hi stephane_ |
| 04:58.18 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 04:59.24 | *** join/#asterisk ken___ (n=ken@125.212.103.40) |
| 04:59.27 | ken___ | yo |
| 04:59.35 | ken___ | i got this weird problem going on -- |
| 05:00.44 | ken___ | i'm compiling asterisk and zaptel just fine, i have a 400p interface card in this system. wctdm and zaptel are loading fine, i can see /proc/zaptel and /dev/zap just fine however, ztcfg and asterisk both seg fault when they try to load |
| 05:01.07 | ken___ | ztcfg says "line 232: Cannot get number of tones for channel 1 |
| 05:01.08 | ken___ | line 232: Cannot init tones for channel 1 |
| 05:01.16 | ken___ | and then a bunch more errors |
| 05:01.21 | ken___ | anyone know what the problem might be ? |
| 05:02.05 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:02.05 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:04.16 | *** join/#asterisk dalekurt (n=dalekurt@port0096-afo-adsl.cwjamaica.com) |
| 05:06.50 | Strom_C | ojai: are you still here? |
| 05:16.40 | smackus | i am looking for the documentation for setting up the trounk group stuff in zapata.conf. trying to figure out what each digit of then spanmap line means "spanmap => 1,1,0" can someone help me find the documentation. |
| 05:16.50 | Strom_C | yeah |
| 05:17.00 | ojai | Strom_C: yes |
| 05:17.08 | Strom_C | ojai: ok |
| 05:17.55 | Strom_C | the reason you dont want to do vonage over fxo is that there's just too much A/D conversion going on, and with the crappy codec vonage uses, it's not worth the trouble |
| 05:17.56 | Strom_C | you generally want to try and keep it digital from end to end |
| 05:20.01 | *** join/#asterisk LoneShadow (n=duh@59.92.136.210) |
| 05:20.41 | *** join/#asterisk SomethingISODD (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 05:21.46 | ojai | Strom_C: damn |
| 05:22.53 | Strom_C | this is the same reason having analog entrance facilities from the telco is also generally a last-resort |
| 05:22.53 | ojai | we've been getting all kinds of crazy telemarketing calls lately and I've been wanting to play with Asterisk anyway so I was hoping now might be a good time. |
| 05:23.45 | ojai | so it's pretty terrible in general eh? and not worth the hardware investment? |
| 05:23.47 | *** join/#asterisk Sakimustafa (n=yusuf@202.133.14.226) |
| 05:24.13 | Strom_C | well, if done right, it can sound almost like digital |
| 05:24.20 | Strom_C | but it's an art |
| 05:24.40 | sx-wks | morning |
| 05:24.45 | Strom_C | evening |
| 05:24.55 | Sakimustafa | morning |
| 05:25.01 | Strom_C | afternoon |
| 05:25.02 | ojai | hrm, yeah, sounds like something I don't have the time/expertise for |
| 05:25.20 | Strom_C | ojai: if its just for home you should be OK |
| 05:25.21 | Sakimustafa | I am new In this channel |
| 05:25.47 | Sakimustafa | Not expert of VOIP |
| 05:25.54 | ojai | yeah, it's just my wife and me. Asterisk might be a little overkill but I thought it'd be fun to start playing with |
| 05:25.55 | Sakimustafa | but crazy abt it |
| 05:25.57 | LoneShadow | voip-info.org :P |
| 05:26.19 | Strom_C | ojai: for a toy system, you'll be fine |
| 05:26.28 | SomethingISODD | hey any asterisk/php people around tonight i am having a issue with a script and could really use some help. http://pastebin.ca/169766 |
| 05:26.40 | Sakimustafa | I am from Bangladesh |
| 05:26.42 | ojai | nice! |
| 05:27.33 | ojai | earlier, you recommended the TDM22B but would the TDM11B also work for my case since I have just the one Vonage line coming in and just one analog phone line in the house? |
| 05:27.49 | Strom_C | the tdm11b only has one fxo port |
| 05:27.58 | Strom_C | so you wont be able to have vonage + the telco line |
| 05:28.07 | Strom_C | you'll have to choose one or the other |
| 05:28.10 | ojai | ah |
| 05:28.59 | ojai | sorry, I'm just learning this. I thought the telco line plugged into the fxs port |
| 05:29.07 | Strom_C | nope |
| 05:29.12 | Strom_C | telephone sets plug into the fxs port |
| 05:29.46 | ojai | sorry -- telephone sets? |
| 05:29.49 | Sakimustafa | In my office we are using Quintum tenor for voip |
| 05:30.03 | Sakimustafa | Also using remote billing soft |
| 05:30.19 | Sakimustafa | But i want to change it with Linux |
| 05:30.20 | sx-wks | Strom_C: vonage does sip IIRC ? |
| 05:30.47 | Sakimustafa | And Asterisk |
| 05:30.56 | Strom_C | ojai: telephone set is an actual physical instrument - they come in touch tone, cordless, rotary, wall, desk, etc models |
| 05:31.00 | Sakimustafa | Is it possible |
| 05:31.19 | Strom_C | sx-wks: yeah, but they dont give you their credentials |
| 05:31.30 | sx-wks | Strom_C: idiots / assholes |
| 05:31.31 | ojai | Strom_C: but it's not just a regular phone though, eh? |
| 05:31.46 | Strom_C | ojai: yeah, a telephone set is "a regular phone" |
| 05:31.52 | ojai | ah |
| 05:32.13 | Strom_C | http://stromcarlson.com/misc/P1010028.JPG there's one, for example |
| 05:32.21 | ojai | but I'd still need the fxs part of it though right to convert it back out of digital? |
| 05:32.24 | Sakimustafa | So from where can I start |
| 05:32.49 | Strom_C | ojai: what do you mean |
| 05:33.26 | sx-wks | Strom_C: nice old 1940s phone :D |
| 05:33.31 | Strom_C | thanks :) |
| 05:33.38 | Strom_C | western electric 302 |
| 05:33.54 | *** join/#asterisk Werdna (i=Andrew@silentflame/member/Werdna) |
| 05:34.09 | sx-wks | Strom_C: be careful, that bakelite is fragile |
| 05:34.49 | ojai | just from what I'd read, I thought (which I'm sure is wrong) that you get analog from the phone company (in my case Vonage) but Asterisk needs to convert it to digital w/ fxo. but an analog phone needs it converted back out of digital which is where the fxs card is needed? |
| 05:34.49 | Strom_C | sx-wks: well i dont go banging it around like a hammer, but the handset has yet to chip or crack |
| 05:34.55 | ojai | again, I'm sure that's completely wrong :) |
| 05:35.20 | Strom_C | ojai: i suppose that's one way to explain it |
| 05:35.59 | ojai | I'm pretty handy on the linux front but not so handy with telephony (yet) |
| 05:37.18 | sx-wks | ojai: it's pretty easy, took me all of 1 hour to install the digium card and having it working with asterisk |
| 05:37.42 | Strom_C | ojai: you may want to read this document i have on my site called "Telephony 101" |
| 05:38.06 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 05:38.07 | Strom_C | http://stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 05:41.47 | smackus | when dialing into a pri on my system, i get this error: |
| 05:41.48 | smackus | Sep 13 23:41:05 WARNING[24435]: chan_zap.c:8386 pri_dchannel: Ring requested on unconfigured channel 0/3 span 3 |
| 05:41.58 | smackus | but I am positive i have configured it... maybe just not correctly. |
| 05:42.20 | smackus | trying to do a trunk group of two pris. dchannel is on the first span of the t1. |
| 05:43.32 | smackus | here is my config |
| 05:43.32 | smackus | http://pastebin.ca/169781 |
| 05:43.44 | smackus | can anyone see what would be causing this issue for me? |
| 05:43.58 | ojai | Strom_C: thanks -- I'll check that out. and thanks for the help tonight -- I really appreciate it |
| 05:44.23 | Strom_C | ojai: any time :) |
| 05:44.27 | ojai | I just didn't expect the hardware to cost so much :) |
| 05:44.53 | Strom_C | good hardware does tend to be a little pricey |
| 05:45.23 | ojai | very true. thanks again. gotta hit the sack |
| 05:45.52 | Strom_C | smackus: how many spans do you have? four? |
| 05:46.32 | Strom_C | show me your zaptel.conf as well |
| 05:47.36 | smackus | http://pastebin.ca/169789 |
| 05:47.41 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 05:47.53 | smackus | the first two are single t1s. the last two are a trunk group of 2 |
| 05:48.07 | dalekurt | Hey Guys, what codec does IAXTEL use? |
| 05:48.10 | dalekurt | ulaw??? |
| 05:48.18 | Strom_C | smackus: are all four from the telco? |
| 05:48.22 | smackus | yes |
| 05:48.33 | Strom_C | and all served off the same switch, i'd assume? |
| 05:48.44 | sx-wks | Strom_C: however it may not work in the other way around... aka "expensive hardware may not be good hardware |
| 05:48.44 | smackus | yes |
| 05:49.11 | *** join/#asterisk aadilismail (n=aaaaaaaa@202.166.161.18) |
| 05:49.26 | Strom_C | smackus: ok...first thing i'd do is modify your zaptel.conf so you have specifically defined primary, secondary, tertiary, and quaternary timing sources |
| 05:49.50 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 05:51.19 | smackus | Strom_C: like so? http://pastebin.ca/169805 |
| 05:51.29 | Strom_C | yes |
| 05:51.40 | Strom_C | next: |
| 05:51.55 | JT | smackus: |
| 05:51.58 | Strom_C | if spans three and four are sharing the same d-channel, your spans should look like this |
| 05:51.59 | sx-wks | Strom_C: I'm getting an interesting "Xml error: not well-formed (invalid token)-58" on the asterisk page on the wiki |
| 05:52.04 | JT | smackus: there's a page on voip-info re the span config |
| 05:52.11 | JT | it tells you exactly how that line works |
| 05:52.11 | *** join/#asterisk ChrisDE4 (n=ChrisDE@tmo-034-62.customers.d1-online.com) |
| 05:52.19 | Strom_C | yes: http://www.voip-info.org/wiki/view/NFAS |
| 05:52.23 | smackus | JT, thanks, followed it... making mistakes, looking for help |
| 05:52.34 | aadilismail | hi guys ......... |
| 05:53.29 | aadilismail | i m new to asterisk .... can anybody guide... . zaptel version.... libpri .. addons ... sounds..... all are different... or some has to contribute with eachother? |
| 05:53.30 | Strom_C | smackus: trunkgroup => 1,72 |
| 05:53.33 | Strom_C | er |
| 05:53.35 | Strom_C | smackus: trunkgroup => 1,72,72 |
| 05:53.55 | smackus | ah... did not understand that. I thought that was only if there was another backup channel |
| 05:54.03 | Strom_C | i /think/ |
| 05:54.18 | dalekurt | Need a quick help here guys... |
| 05:54.20 | Strom_C | im only going off my experience and what this document says; i have never actually configured NFAS |
| 05:54.42 | dalekurt | Trying to dial out on a IAXTEL connection and getting 'Unable to negotiate codec' and I'm using ULAW |
| 05:54.53 | smackus | still the same |
| 05:54.55 | Strom_C | dalekurt: how about trying "allow=all" and seeing what it negotiates at |
| 05:55.03 | smackus | get the pri_dchannel: error |
| 05:55.07 | aadilismail | guys help |
| 05:55.13 | aadilismail | . zaptel version.... libpri .. addons ... sounds..... all are different... or some has to contribute with eachother? |
| 05:55.13 | dalekurt | Strom_C: Kewl |
| 05:55.28 | Strom_C | aadilismail: just use the latest available version of each one |
| 05:55.43 | aadilismail | i m new to asterisk |
| 05:55.50 | *** join/#asterisk beu (i=beu@freenode/developer/gentoo.developer.beu) |
| 05:56.15 | aadilismail | Storm_c plz guide .... . zaptel version.... libpri .. addons ... sounds..... all are different... or some has to contribute with eachother? or which one is the latest |
| 05:56.25 | Strom_C | aadilismail: I gave you the answer already |
| 05:56.37 | smackus | lol... figured it out. |
| 05:56.40 | Strom_C | aadilismail: no one likes it when you repeat the same question three times in a minute |
| 05:56.43 | smackus | logical spans start with 0 not 1 |
| 05:56.43 | Strom_C | smackus: oh, what was it? |
| 05:56.46 | smackus | thanks all |
| 05:56.47 | Strom_C | ah |
| 05:57.13 | ChrisDE4 | aadilismail: go to ftp.digium.com there you see all available versions and can trigger out which is the newest |
| 05:57.25 | smackus | trunkgroup=> 1,72 |
| 05:57.25 | smackus | spanmap => 3,1,0 |
| 05:57.25 | smackus | spanmap => 4,1,1 |
| 05:57.31 | aadilismail | thanks guys |
| 05:57.36 | Strom_C | smackus: ah, ok |
| 05:58.00 | ChrisDE4 | now, is there anyone who succeeded in sending a fax with txfax? |
| 05:58.11 | dalekurt | Strom_C: I don't get that message, thanks. |
| 05:58.24 | dalekurt | Strom_C: but I get status is 'CHANUNAVAIL' |
| 05:58.36 | ChrisDE4 | I only could send half of a fax :-) |
| 05:58.39 | Strom_C | dalekurt: wwhat? |
| 05:58.51 | dalekurt | Strom_C: I take it that it's not registered.. |
| 05:59.04 | dalekurt | well I made a call to a number on IAXTEL and now I get status is 'CHANUNAVAIL' |
| 05:59.20 | Strom_C | what does your console output say? |
| 05:59.20 | ChrisDE4 | that a normal state for a carrier |
| 05:59.59 | dalekurt | Strom_C: Everyone is busy/congested at this time (1:0/0/1) |
| 06:00.03 | dalekurt | Strom_C: Auto fallthrough, channel 'SIP/2201-081ae790' status is 'CHANUNAVAIL' |
| 06:00.35 | Strom_C | dalekurt: what does "iax2 show peer (whatever iaxtel is called)" return? |
| 06:00.38 | ChrisDE4 | this probably means that your carrier doesn't have a route to this destination? |
| 06:02.26 | dalekurt | Strom_C: It gave me a full read out of that peer, the IP it's connected to and port the codecs, that status... |
| 06:03.01 | Strom_C | what number are you trying to dial via iaxtel |
| 06:03.01 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 06:03.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:03.28 | dalekurt | Strom_C: 1 700 900 0000 |
| 06:03.36 | Strom_C | what is that supposed to go to? |
| 06:03.46 | dalekurt | Strom_C: A recording... |
| 06:04.08 | Strom_C | what about 1-700-428-6000 |
| 06:04.42 | dalekurt | Strom_C: Let me try that |
| 06:05.18 | dalekurt | Strom_C: Everyone is busy/congested at this time (1:0/0/1) |
| 06:05.21 | dalekurt | Strom_C: Auto fallthrough, channel 'SIP/2201-081b13b0' status is 'CHANUNAVAIL' |
| 06:05.47 | dalekurt | Strom_C: Called dalekurt@iaxtel-gw/17004286000@iaxtel |
| 06:05.49 | Strom_C | dalekurt: do me a favor and set your verbosity up to three and then pastebin the full console output |
| 06:06.05 | dalekurt | it's at three now... where is the bin |
| 06:06.10 | Strom_C | pastebin.ca |
| 06:07.41 | dalekurt | Strom_C: http://pastebin.ca/169818 |
| 06:08.09 | tengulre | hi,all! which linux are you using for asterisk? debian, Fedora, ...? |
| 06:08.33 | dalekurt | tengulre: Debian |
| 06:08.33 | ChrisDE4 | dalekurt: your carrier doesnt route this number |
| 06:08.45 | Strom_C | ChrisDE4: nonsense |
| 06:08.49 | Strom_C | that's digium's iaxtel number |
| 06:09.00 | Strom_C | my guess is that iaxtel is on the fritz |
| 06:09.02 | Strom_C | as usual |
| 06:09.10 | dalekurt | ok... I have a FWD. |
| 06:09.26 | tengulre | dalekurt: thanks for answer! |
| 06:09.38 | tengulre | I will using debian too. |
| 06:09.41 | dalekurt | are there other free terminations out there. |
| 06:09.55 | Strom_C | tengulre: the general answer is "use your favorite linux distro" |
| 06:09.57 | ChrisDE4 | strom_c: ok... but this is a message from iaxtel... so iaxtel is broken |
| 06:10.02 | dalekurt | Well Debian is a little trickier, Fedora is easy to learn |
| 06:10.08 | Strom_C | ChrisDE4: that's what I just said |
| 06:10.19 | Strom_C | <Strom_C> my guess is that iaxtel is on the fritz |
| 06:10.19 | Strom_C | <Strom_C> as usual |
| 06:10.38 | ChrisDE4 | yes i know what you said :-) |
| 06:10.51 | dalekurt | I have FWD and IAXTEL, can you recommend any more... :D |
| 06:11.02 | ChrisDE4 | so iaxtel (currently) doesnt route this number :-) |
| 06:11.19 | Strom_C | ChrisDE4: no, iaxtel should route the number, but iaxtel is probably massively broken agai |
| 06:11.23 | Strom_C | s/agai/again/ |
| 06:11.58 | ChrisDE4 | right :-) |
| 06:12.38 | Strom_C | make sure you phrase it correctly; merely stating that they don't route the number implies that it's a translation issue, not a a more fundamental problem |
| 06:13.51 | dalekurt | anyone uses BroadVOice? |
| 06:14.08 | Strom_C | i use broadvoice |
| 06:14.15 | dalekurt | service good? |
| 06:14.22 | Strom_C | im not completely pleased with them, but they're better than most |
| 06:14.37 | dalekurt | ANy other recommendations? |
| 06:14.44 | Strom_C | depends on your neds |
| 06:14.49 | Strom_C | s/neds/needs/ |
| 06:14.53 | ChrisDE4 | terrasip.com |
| 06:15.23 | dalekurt | Needs like points of termination? |
| 06:16.22 | Strom_C | needs like traffic, concurrent calls, DIDs, DNIS, network capacity issues, etc etc etc etc etc |
| 06:16.45 | dalekurt | Hey Strom_C, and ChrisDE4, I got this weird problem earlier... I have a modem (speedtouch) and firewall (monowall) and I got my Natting working to a point... |
| 06:17.25 | dalekurt | Strom_C remember when we were talking about this some time ago, I got the Asterisk to finally work with my firewall and modem. |
| 06:17.36 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 06:17.47 | sx-wks | dalekurt: good |
| 06:17.52 | dalekurt | Well I can get client to register and make calls but noone can hear |
| 06:18.06 | Strom_C | dalekurt: SIP? |
| 06:18.20 | dalekurt | So I would call another extension from outside of the network to another that is in the network and we can't hear each other. |
| 06:18.33 | ChrisDE4 | dalekurt: switch off uPnP at your router |
| 06:19.15 | ChrisDE4 | uPnP is a bug... not a feature :-) |
| 06:19.36 | dalekurt | ChrisDE4: I think I have that turned off on the modem... let me check |
| 06:20.08 | dalekurt | ChrisDE4: yep ont he modem "UPnP: Disabled" |
| 06:21.04 | dalekurt | ChrisDE4: On my inbound routes I have the following |
| 06:21.25 | dalekurt | ChrisDE4: TCP/UDP 5060 aaa.bbb.ccc.ddd 5060 |
| 06:22.11 | dalekurt | ChrisDE4: UDP 2727 aaa.bbb.ccc.ddd 2727 |
| 06:22.21 | dalekurt | ChrisDE4: UDP 2727 aaa.bbb.ccc.ddd 2727 |
| 06:22.29 | dalekurt | ChrisDE4: UDP 4569 aaa.bbb.ccc.ddd 4569 |
| 06:22.39 | dalekurt | ChrisDE4: UDP 10000 - 20000 aaa.bbb.ccc.ddd 10000 - 20000 |
| 06:22.41 | Strom_C | dalekurt: for the love of god, please use pastebin |
| 06:22.51 | dalekurt | sorry for the crap! |
| 06:23.01 | dalekurt | Pastbin it is... |
| 06:26.27 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:26.47 | dalekurt | http://pastebin.ca/169836 |
| 06:26.55 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:28.24 | dalekurt | Any ideas? |
| 06:28.58 | Strom_C | yeah |
| 06:29.02 | Strom_C | dont use sip through a firewall |
| 06:29.09 | tengulre | I got many characters in /var/log/messages, 'FXO PCI Master abort'? what's eror? |
| 06:29.52 | dalekurt | Any other ideas... Strom_C kinda unavoidable... but is that the real cause? |
| 06:30.07 | Strom_C | well, if you want the easy solution :) |
| 06:30.16 | dalekurt | what the hard one |
| 06:30.49 | Strom_C | you're doing the hard one |
| 06:31.05 | dalekurt | the hard one with a solution... |
| 06:31.13 | dalekurt | I just want it to work... properly |
| 06:35.28 | *** join/#asterisk Sir_Diddymus (n=doe@pd95b08e3.dip0.t-ipconnect.de) |
| 06:37.26 | tengulre | any ideas? |
| 06:42.08 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 06:43.00 | aadilismail | take back |
| 06:45.34 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 06:50.24 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 06:53.14 | *** join/#asterisk eject_ck (n=eject@62.64.75.98) |
| 06:56.32 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 06:57.03 | *** join/#asterisk anthonyl (n=rachel@c-67-167-214-149.hsd1.il.comcast.net) |
| 07:05.02 | *** join/#asterisk shtoom (n=godson@202.149.38.38) |
| 07:08.27 | *** join/#asterisk jeffjohnson (n=Jeffjohn@unaffiliated/jeffjohnson) |
| 07:08.29 | jeffjohnson | howdy |
| 07:08.52 | Strom_C | ydwoh |
| 07:09.03 | *** join/#asterisk MACscr (n=MACScr@adsl-75-23-105-65.dsl.peoril.sbcglobal.net) |
| 07:09.42 | jeffjohnson | i have a problem astmanproxy, i try to loging with Asterisk TAPI Line to astman proxy but I always get login failed. Ive configured astmanproxy to use simple password authentication. Astmanproxy successfull connects to asterisk manager... the error message from astmanproxy is: |
| 07:09.44 | jeffjohnson | "Sep 13 20:06:26: asterisk@127.0.0.1 got: Response: Error |
| 07:09.44 | jeffjohnson | Sep 13 20:06:26: asterisk@127.0.0.1 got: Message: No variable specified |
| 07:10.39 | Strom_C | I have no experience with astmanproxy |
| 07:12.45 | *** join/#asterisk af_ (n=af@ip-173-63.sn1.eutelia.it) |
| 07:13.42 | jeffjohnson | its really strange ./ |
| 07:15.56 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:17.27 | *** join/#asterisk Op3r (i=Op3r@dial.dyn.151.107.surfsterisp.com) |
| 07:18.44 | uwe | hello, asterisk is not recording incomming calls ... i know this sounds too odd ... but record_out and in are set to always in sip_additional.conf, and the only thing i did lately is moving old recordings from /var/spool/asterisk/monitor to a subdirectory. any ideas what it could be?! yes, i am using amp, but im trying to figure out what the problem is without changing from amp, im editing files by hand |
| 07:20.22 | smackus | divert.fwd.1.enabled="0" is in the phone.cfg file on the ftp server. it works like I want it to, but I have to do it to 100 phones. Is there some way to do this from within the sip.cfg file? so I dont have to touch every last phone.cfg file? |
| 07:23.04 | *** join/#asterisk alphaque (n=alphaque@60.51.217.61) |
| 07:26.42 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.21.81) |
| 07:26.55 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 07:27.37 | tengulre11 | Any ideas? |
| 07:28.08 | tengulre11 | many 'FXO PCI Master abort' in /var/log/messages |
| 07:29.05 | sx-wks | getting "Power alarm on module 1, resetting!" |
| 07:29.20 | sx-wks | only twice though |
| 07:30.23 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 07:30.38 | *** join/#asterisk RoyK (n=roy@ti211210a080-0574.bb.online.no) |
| 07:31.14 | tengulre11 | sx-wks, are you answer me?? |
| 07:31.21 | sx-wks | no |
| 07:31.24 | tengulre11 | :( |
| 07:31.50 | tengulre11 | any ideas? |
| 07:32.01 | sx-wks | no. never had that |
| 07:32.03 | dalekurt | Guys I'm out |
| 07:32.18 | sx-wks | tengulre11: try to switch PCI ports |
| 07:32.45 | *** join/#asterisk pbx1 (n=pbx1@58.69.92.20) |
| 07:32.55 | tengulre11 | sx-wks: how to switch ports? in BIOS? |
| 07:34.12 | JT | remove card from port |
| 07:34.13 | sx-wks | no, like yank the card from where it sits and put it on another port |
| 07:34.19 | JT | re=insert into different one |
| 07:34.51 | E-bola | Is there anything like phone switches? |
| 07:34.55 | E-bola | like i know u can buy pci cards |
| 07:34.59 | sx-wks | if that doesn't do it, change for a different brand / model of mother board |
| 07:35.00 | E-bola | but what if u need alot of phones connected? |
| 07:35.11 | JT | E-bola: analogue phones i assume |
| 07:35.18 | E-bola | jt: ye |
| 07:35.18 | JT | E-bola: T1/E1 card + channel bank |
| 07:35.30 | sx-wks | JT: or an asterbank |
| 07:36.05 | sx-wks | depends on the number of phones I guess |
| 07:36.05 | E-bola | whats an asterbank? |
| 07:36.05 | JT | or a TDM2400P or whatever it is |
| 07:36.06 | kmilitzer | Hello everyone. I have a strange phenomen on one of my PSTN-Gateways. I make a call like that UAC -> SER -> Asterisk A -> Asterisk B (PSTN GW) all in SIP and have only one way audio coming from the PSTN to SIP. All Firewall-rules etc. seem to be OK. Strange thing is, that it works for a time and then suddenly stops. A restart of asterisk always seems to help. Any ideas? |
| 07:36.14 | sx-wks | E-bola: a USB device that allows you to connect 32 phones to it |
| 07:36.18 | JT | he means astribank |
| 07:36.21 | JT | and stuff that |
| 07:36.22 | JT | it's usb |
| 07:36.24 | sx-wks | JT: yeah that |
| 07:36.24 | E-bola | sx-wks: ohh interesting |
| 07:36.31 | JT | go T1/E1 + channel bank |
| 07:36.33 | E-bola | usb... .is that enterprise suited? |
| 07:36.36 | sx-wks | JT: doesn't mean it doesn't work :D |
| 07:36.39 | JT | it's still one of the best ways |
| 07:36.40 | JT | E-bola: no |
| 07:36.59 | sx-wks | JT: enterprise suited doesn't really mean anything |
| 07:37.18 | JT | ok, you play with your usb, i'll stick to telco tdm interconnect |
| 07:37.20 | sx-wks | it works or it doesn't... the rest is marketting bullshit |
| 07:37.25 | E-bola | nonesence |
| 07:37.30 | E-bola | it can work in varrying degrees |
| 07:37.43 | E-bola | like an smb or a home user might have lower demands than an enterprise |
| 07:38.11 | JT | E-bola: a digitam T100P or TE110P, etc + a CAC or Adtran channel bank is probably the best solution |
| 07:38.22 | JT | you could use a Zhone channel bank if you don't need callerid |
| 07:38.26 | JT | they're fairly cheap |
| 07:38.37 | E-bola | Hmm |
| 07:38.43 | E-bola | does isdn-30 mean anything to you guys? |
| 07:38.46 | JT | yes |
| 07:38.47 | JT | E1 |
| 07:38.48 | E-bola | thats what would be the outgoing line |
| 07:38.53 | JT | that's an E1 |
| 07:38.57 | E-bola | really? |
| 07:38.59 | E-bola | lol never knew |
| 07:39.00 | JT | yes |
| 07:39.03 | E-bola | cool |
| 07:39.08 | JT | isdn-30 is a nice telco name for it |
| 07:39.20 | JT | 2.048Mbit/s channelised PRI CCS E1 :P |
| 07:39.27 | E-bola | im a bit clueless, how is an E1 or isdn 30 terminated at the client? |
| 07:39.30 | E-bola | is it 1 cable or? |
| 07:39.40 | drray | rj48 cable |
| 07:40.13 | JT | 32 X 64kbits, 30 for voice, 1 for D channel signalling, one for framing synch |
| 07:40.18 | JT | RJ-48C to be precise :) |
| 07:40.35 | JT | it looks like an RJ-45 cat5 network cable |
| 07:40.37 | sx-wks | drray: still wondering wtf is the difference between RJ45 and RJ48 |
| 07:40.43 | E-bola | Hmm so far we've had a strictly SIP based setup so i've been doing ok gettinbg by with my IT knowledge |
| 07:40.46 | JT | but the wiring is different for a crossover |
| 07:40.51 | E-bola | but all this low-tech telephone stuff is maybe a bit too hard |
| 07:40.52 | JT | sx-wks: crossover wiring is different |
| 07:40.57 | JT | and it's just a naming standard |
| 07:41.02 | kmilitzer | So nobody got any idea why I suddenly get one way audio. Problem seems to exists since asterisk 1.2.11 |
| 07:41.08 | sx-wks | JT: it's still RJ45 plugs, right ? |
| 07:41.19 | JT | E-bola: lol, low tech, maybe you mean "low level" |
| 07:41.22 | E-bola | kmilitzer: firewall/nat problem maybe? |
| 07:41.29 | E-bola | jt: means the same for me |
| 07:41.39 | Strom_C | E-bola: you should read "Telephony 101" |
| 07:41.41 | JT | sx-wks: yeah, they're technically 8P8C plugs, the RJ they are depends on the wiring and application |
| 07:41.49 | kmilitzer | E-bola: As I said, FW seems to be OK, it just starts after a while. No NAT involved ... |
| 07:41.50 | drray | sx-wks - how it is wired |
| 07:41.55 | Strom_C | E-bola: http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 07:41.57 | JT | E-bola: well it's not really low tech |
| 07:42.01 | sx-wks | ah ok... just yet another way to confuse people :D |
| 07:42.03 | E-bola | Strom_C: well the question is if i should try to learn all the telephone stuff or just give up and pay somebody for it |
| 07:42.04 | drray | it's just rj45 wired another way |
| 07:42.16 | drray | because we need yet another propietary cable |
| 07:42.22 | Strom_C | E-bola: try, and then give up. It's really not that difficult. |
| 07:42.30 | Strom_C | drray: technically, T1 predates ethernet |
| 07:42.30 | JT | which you can make in less than 5min, drray |
| 07:42.30 | drray | don't even try |
| 07:42.41 | JT | drray: ? |
| 07:42.41 | E-bola | Im just uncertain that 1) I will be able to master it to a satisfyiung degree in a reasonable amount of time. and 2) that the end product will be good enough |
| 07:42.50 | drray | JT - I've spent a good 45 minutes not making a cable |
| 07:42.50 | sx-wks | drray: right |
| 07:42.52 | drray | :) |
| 07:43.05 | JT | drray: if you can't make a network cable, you can't make a T1 cable |
| 07:43.13 | drray | JT - also true |
| 07:43.17 | JT | if you can, then you should be fine, given adequate instructions |
| 07:43.26 | drray | black box |
| 07:43.31 | JT | eh |
| 07:43.39 | sx-wks | can't those things auto-MDIX ? |
| 07:43.45 | JT | it's easy to crimp cables once you get the hand of it |
| 07:43.51 | JT | sx-wks: T1? no |
| 07:43.56 | E-bola | its easy to make a network cable |
| 07:44.04 | JT | s/hand/hang/ |
| 07:44.18 | Strom_C | http://www.stromcarlson.com/wiring/t1_crossover.png |
| 07:44.25 | Strom_C | http://www.stromcarlson.com/wiring/T1-loopback.png |
| 07:44.30 | Strom_C | easy stuff |
| 07:44.36 | sx-wks | JT: that's where ethernet is widely superior to T1 :D |
| 07:44.55 | JT | sx-wks: spare me please, only some of the latest gear does auto MDI-X |
| 07:45.02 | JT | mainly due to gigabit ethernet |
| 07:45.09 | JT | and its use of all 4 pairs |
| 07:45.16 | JT | negating the need for crossovers |
| 07:45.17 | Strom_C | yeah, i clearly remember a time when auto mdix was nonexistent |
| 07:45.29 | JT | Strom_C: most 10/100 gear doesn't support it |
| 07:45.34 | JT | except for the newest stuff |
| 07:45.35 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 07:45.35 | Strom_C | yup |
| 07:46.00 | sx-wks | newest as in "in the last 4 years" or so |
| 07:46.05 | sx-wks | which is ages |
| 07:46.05 | JT | anyway, T1 should not get auto MDI-X, it is not designed for retards to be playing with |
| 07:46.12 | sx-wks | lol |
| 07:46.15 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 07:46.23 | drray | hey! |
| 07:46.51 | *** join/#asterisk postel (n=jp@Wikimedia/Postel) |
| 07:47.01 | _Vile | uhm |
| 07:47.13 | _Vile | T-1 is a lower layer |
| 07:47.25 | sx-wks | JT: there enters the "we are better than you are" arguments of telco people :D |
| 07:47.44 | drray | Telco has to work |
| 07:47.45 | sx-wks | _Vile: t1 is just like ethernet |
| 07:47.58 | sx-wks | drray: haha... |
| 07:48.00 | Strom_C | sx-wks: yeah, but telco stuff tends to work very reliably once configured properly...as opposed to IT stuff which is usually just "good enough" |
| 07:48.08 | JT | sx-wks: hey, if you want to have tall poppy syndrome, that's fine, but maybe the reason some telco people might hold that view is because they are |
| 07:48.26 | _Vile | http://en.wikipedia.org/wiki/MDIX |
| 07:48.29 | JT | sx-wks: your attitude on everything seems to be "good enough" |
| 07:48.37 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:48.42 | _Vile | t-1 is the layer ethernet rides on |
| 07:48.43 | sx-wks | Strom_C: tell that to Bloody France telecom and their crappy copper phone lines |
| 07:48.47 | _Vile | lower layer |
| 07:48.53 | JT | Layer 1 |
| 07:48.55 | JT | well |
| 07:48.57 | drray | the crappy copper phone lines that work when the power is out |
| 07:48.58 | sx-wks | _Vile: no it doesn't |
| 07:48.59 | JT | ethernet has layer 2 too |
| 07:49.12 | _Vile | ethernet doesn't have to ride on t-1 |
| 07:49.25 | _Vile | hdsl, but it's a lower layer |
| 07:49.28 | sx-wks | drray: it doesn't always work even when the dang power's on |
| 07:49.32 | Strom_C | sx-wks: sorry, my only experience has been with Bell System telephone companies, who do a damned fine job :) |
| 07:49.37 | _Vile | mdix has nothing to do w/ t-1 |
| 07:49.49 | sx-wks | Strom_C: heh |
| 07:50.02 | Strom_C | [insert standard france joke here] |
| 07:50.33 | sx-wks | [insert standard bush stole the election joke here] |
| 07:50.45 | _Vile | </exit telco guy> |
| 07:50.52 | Strom_C | hey, i despise bush too |
| 07:51.47 | kimron | need a tip: got the message: socket_read: Rejected connect attempt from 192.168.1.101, who was trying to reach '500@' ? |
| 07:51.50 | JT | and i'm in a neutral third party, so screw you guys ;) |
| 07:52.03 | JT | for varying levels of neutral |
| 07:54.08 | _Vile | Strom, bell does do a good job, I'm a CLEC and I hate working with them because their service sucks |
| 07:54.14 | _Vile | but their work processes *rock* |
| 07:54.17 | *** join/#asterisk sajith (n=sajith@61.12.17.162) |
| 07:54.32 | _Vile | doesn't matter which ilec you go to |
| 07:54.40 | Strom_C | _Vile: yeah...it helps to have a friend on the inside |
| 07:54.41 | _Vile | same process, documented, *exactly* what to do |
| 07:54.44 | _Vile | everything is tagged |
| 07:54.48 | _Vile | documented |
| 07:55.06 | Strom_C | yup. I've got shitloads of old Bell System Practices, and the thoroughness amazes me |
| 07:55.10 | _Vile | perfect design on every bit |
| 07:55.20 | _Vile | amazes me too |
| 07:55.27 | JT | i've got to ask |
| 07:55.30 | _Vile | we're friends with all of the ilec locals |
| 07:55.39 | JT | would you use usb for telephony interconnect, Strom_C? :) |
| 07:55.39 | _Vile | it's the corp crap we hate :) |
| 07:55.54 | Strom_C | JT: no :) |
| 07:56.01 | _Vile | but I get amazed every day when I walk into the co |
| 07:56.03 | JT | no surprise :) |
| 07:56.07 | Strom_C | as much as I like USB...no. |
| 07:58.24 | Strom_C | hey, no one has made any GTE jokes yet |
| 07:58.25 | _Vile | most people don't realize exactly how much work goes into giving them a dial tone.. haha, love that thought too |
| 07:58.41 | JT | heh yeah |
| 07:58.59 | JT | or how many millions their local switch is worth, let alone the rest of the infrastructure |
| 07:59.00 | Strom_C | I've got a GTE practice manual...one of the practices inside is for the use of a ladder guard |
| 07:59.13 | Strom_C | and the ladder guard says "KEEP OFF GENERAL SYSTEM" |
| 07:59.26 | Strom_C | my first reaction? "Truer words were never spoken" |
| 07:59.40 | JT | bus bars? |
| 07:59.45 | Strom_C | huh? |
| 07:59.58 | JT | not getting the ladder guards reference |
| 08:00.16 | Strom_C | ladder guard is a thing you put on a ladder to prevent people from climbing up it |
| 08:00.26 | Strom_C | it's basically a big board you put over the rungs |
| 08:00.27 | _Vile | haha |
| 08:00.34 | JT | ah yeah |
| 08:00.44 | _Vile | nice |
| 08:00.53 | JT | so what's the keep off general system thing? |
| 08:01.10 | Strom_C | the ladder guard says "KEEP OFF" and then under it is the General Telephone System logo |
| 08:01.14 | _Vile | keep off<br> |
| 08:01.22 | Strom_C | but it reads as if it says "KEEP OFF GENERAL SYSTEM" |
| 08:01.33 | JT | ah ok |
| 08:03.30 | Strom_C | http://www.stromcarlson.com/misc/P1010065.JPG |
| 08:05.21 | *** join/#asterisk grexk (n=grexk@124.107.72.45) |
| 08:05.23 | *** join/#asterisk LoneShadow (n=duh@59.92.147.70) |
| 08:07.32 | _Vile | http://www.montagar.com/~patj/phone-switches.htm |
| 08:08.15 | _Vile | we park next to a dms |
| 08:08.34 | eject_ck | hi all, I have question: I have SIpura 3000 and conenct it to my PBX on FXO and to my Asterisk server on Ethernet port - I tune calling from PBX to SIP accounts via dialplan. Now I try connect from another SIP account to my interal PBX's phone. How make it right ? |
| 08:09.18 | Strom_C | vile: ive seen that |
| 08:09.35 | *** join/#asterisk acehunky (n=enterux@202.149.38.38) |
| 08:10.18 | _Vile | big switch |
| 08:11.06 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:11.20 | _Vile | these days though, something like a 5e taking up 15-20 racks can be done in less than 1 |
| 08:11.56 | _Vile | less cross connect crap etc |
| 08:12.20 | *** join/#asterisk daredundi (n=shekhar@202.149.38.38) |
| 08:12.37 | Strom_C | yeah |
| 08:12.46 | Strom_C | I was amazed at how small a DMS-250 is |
| 08:13.07 | _Vile | 10x12 can fit it |
| 08:13.15 | JT | heh |
| 08:13.16 | Strom_C | the actual switch is nothing; the whole class 4 office was just optical carrier stuff |
| 08:13.33 | JT | we have only one phone company here that does landline calls that uses north american switches |
| 08:13.47 | JT | they use DMS-100s |
| 08:14.02 | Strom_C | brown and green :) |
| 08:14.08 | _Vile | half a block? |
| 08:14.11 | _Vile | :) |
| 08:14.14 | daredundi | hi |
| 08:14.17 | JT | everything else by a much bigger majority is Ericsson AXE or Alcatel System 12 |
| 08:14.21 | JT | for landlines |
| 08:14.33 | JT | obviously there's Nokia and all sorts of other stuff for mobiles |
| 08:15.26 | _Vile | axe is in qwest territory too |
| 08:15.39 | _Vile | mainly dms's and 5e's though |
| 08:21.34 | *** join/#asterisk sxpert-work (n=sxpert@raph.imag.fr) |
| 08:22.01 | sxpert-work | yop |
| 08:23.50 | ChrisDE4 | is anyone familar to txfax? |
| 08:24.45 | kmilitzer | Once again my one-way-audi problem. As I already said, all works fine for a time and then suddenly I have no audio from the calleing party to the called party. No NAT involved, all FW rules seem to be OK and so on. The strange thing is, that this only happens from SIP->PSTN calls and not the other way round, as incoming calls all are OK. I have the suspecion, that this started with asterisk 1.2.10 ... any ideas? |
| 08:24.46 | *** join/#asterisk angelcry (n=DarkStar@151.53.217.204) |
| 08:24.51 | angelcry | hello peoples :) |
| 08:25.41 | ChrisDE4 | kmilitzer: switch off uPnP at your router :-) |
| 08:26.01 | sxpert-work | kmilitzer: if your one-way-audi causes problems, try a two-way-bmw instead :D |
| 08:26.25 | kmilitzer | You're all very funny this morning ;) |
| 08:26.39 | kmilitzer | .. I wish my customers would take it that easy too :( |
| 08:26.59 | nettie | Hey guys, when asterisk keeps complaining about VAD enabled on cliente: myvoipcarrierIP. I spoken with them but they keep saying VAD is not enabled, just the "comform noise is". Anyone can tell me if VAD activation flag is placed on the sip header or somewhere else just to show them they're wrong and continue to investigate the issue? thanx |
| 08:27.35 | ChrisDE4 | no, is not on the sip header |
| 08:28.20 | nettie | ChrisDE4 do you know what else I could check ? |
| 08:28.31 | Strom_C | nettie: usually, comfort noise goes hand in hand with VAD |
| 08:28.46 | Strom_C | because if you're not doing VAD, there's absolutely no point to having comfort noise turned on |
| 08:28.52 | nettie | ahhh |
| 08:28.54 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 08:28.55 | nettie | exaclty |
| 08:28.59 | nettie | that's what I thought |
| 08:29.03 | nettie | as well |
| 08:29.07 | JT | exactly what i was about to type out |
| 08:29.11 | Strom_C | ive dealt with this too |
| 08:29.17 | JT | after i had finished this piece of chocolate cake |
| 08:29.23 | Strom_C | and, in my opinion, 99% of these carriers are complete yo-yos |
| 08:29.33 | nettie | eheh |
| 08:29.55 | nettie | I realyl would like to know which one you're saving :) |
| 08:31.04 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 08:32.48 | sxpert-work | the only issue I have with my SIP carrier is those bloody "Cirpack keepalive packet" packets |
| 08:33.57 | sxpert-work | and some wierd shit with INVITE, but that may be asterisk fscking up |
| 08:36.35 | *** join/#asterisk faberk64 (n=faberk@213.199.15.249) |
| 08:36.40 | L|NUX | i have installed asterisk on vps but when i try to run it |
| 08:36.45 | L|NUX | it will not start |
| 08:36.51 | L|NUX | as daemon |
| 08:37.03 | JT | errors you receive |
| 08:37.07 | JT | they are what? |
| 08:37.08 | L|NUX | wait |
| 08:37.20 | Strom_C | JT: that sounds like "yoda debugs asterisk" |
| 08:37.40 | JT | Strom_C: multiple short lines |
| 08:37.44 | JT | Strom_C: make for good questions |
| 08:37.51 | Strom_C | balls |
| 08:37.53 | JT | Strom_C: maybe incorrect |
| 08:37.53 | Strom_C | incorporated? |
| 08:38.13 | L|NUX | http://pastebin.ca/169962 |
| 08:38.33 | L|NUX | but when i try this command asterisk -vvvvvvgc |
| 08:38.37 | L|NUX | it will start |
| 08:38.41 | L|NUX | but not as daemon |
| 08:38.47 | Strom_C | what user are you |
| 08:38.50 | L|NUX | root |
| 08:38.56 | L|NUX | i am using VPS |
| 08:39.03 | L|NUX | Virtual Private Server |
| 08:39.05 | Strom_C | aka "I like headaches" |
| 08:39.12 | L|NUX | [root@farrukh asterisk]# id |
| 08:39.12 | L|NUX | uid=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wheel) |
| 08:39.21 | nettie | Strom_C I checked again, Asterisk complains about Comfort Noise being enabled |
| 08:39.31 | Strom_C | nettie: your carrier is on crack. |
| 08:39.52 | nettie | Strom_C well they actually said they have it enabled |
| 08:39.56 | L|NUX | Strom_C : can you help me with this issue |
| 08:40.08 | Strom_C | L|NUX: I have no experience running asterisk on VPS |
| 08:40.16 | L|NUX | awwww |
| 08:40.17 | Strom_C | nettie: have them turn it off |
| 08:40.32 | L|NUX | JT : you arround ? |
| 08:40.34 | *** join/#asterisk erikf (n=forsen@pat-tdc.opera.com) |
| 08:40.38 | JT | possibly around |
| 08:40.45 | nettie | Strom_C they said VAD is disabled, does asterisk generates the same error for CNG and VAD or there's one specific for VAD? |
| 08:40.50 | L|NUX | JT : any idea about my problem |
| 08:41.45 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 08:42.01 | Strom_C | there's a separate error for VAD |
| 08:42.14 | Strom_C | but why the hell do they have CNG if they have VAD disabled? |
| 08:42.38 | JT | L|NUX: is that all the errors you get when you try to start the daemon? |
| 08:42.57 | JT | are you using the safe_asterisk script? |
| 08:43.05 | Sir_Diddymus | L|NUX: did you check access rights? That was my initial problem. Started it from cmdline, run as user root. Started it via init-script, different user... don't know, just a hint... |
| 08:43.29 | nettie | Strom_C I have no idea.. or they push it into the whole call |
| 08:43.35 | L|NUX | JT : well when i type this command service asterisk start |
| 08:43.38 | L|NUX | it will say ok |
| 08:43.40 | L|NUX | no error |
| 08:43.53 | Strom_C | "service asterisk start"? |
| 08:43.58 | Strom_C | what linux are you running? |
| 08:44.02 | L|NUX | CentOS |
| 08:44.05 | *** join/#asterisk RoyK (n=roy@ti211210a080-0574.bb.online.no) |
| 08:44.07 | JT | sounds redhatish |
| 08:44.09 | JT | close enough |
| 08:44.38 | JT | L|NUX: i will assume that service asterisk start runs safe_asterisk (you may check with the init script) |
| 08:45.21 | JT | i'll take a guess that safe_asterisk may not work on a virtual server due to it starting up a console on a tty, and there are usually no /dev/ttyXX on a virtual server |
| 08:45.38 | JT | although it could be something else entirely |
| 08:45.41 | L|NUX | hummm |
| 08:45.52 | L|NUX | then how can i start asterisk on vps |
| 08:45.52 | L|NUX | ? |
| 08:46.09 | JT | modify safe_asterisk to not open a console on a tty |
| 08:46.23 | benjk | <PROTECTED> |
| 08:46.35 | L|NUX | humm |
| 08:46.35 | L|NUX | ok |
| 08:46.43 | JT | although check that service asterisk start actually spawns asterisk first |
| 08:46.51 | JT | and that i'm not completely off the mark |
| 08:47.03 | L|NUX | JT : which lines i have to edit ? |
| 08:47.17 | JT | arrgh i don't know off hand |
| 08:47.25 | L|NUX | awwww |
| 08:47.39 | JT | are you a linux sysadmin? |
| 08:47.45 | L|NUX | yeah |
| 08:47.51 | L|NUX | but i am trying to see |
| 08:47.51 | L|NUX | wait |
| 08:48.02 | JT | then you should be able to work it out, and not need to be spoon fed |
| 08:48.25 | L|NUX | ya |
| 08:49.14 | benjk | whats that virtual consoler business for anyway? is it to mimic some sort of heart beat type mechanism? |
| 08:50.16 | JT | benjk: it just opens up an asterisk cli on a tty? |
| 08:50.21 | sxpert-work | a virtual consoler ? it's a IVR application that consoles you when your (ex) gf sent you off |
| 08:50.30 | JT | i don't think it actually does anything other than act as a user interface |
| 08:50.31 | Strom_C | hahahaha |
| 08:50.47 | benjk | I mistyped, I meant console without 'r' |
| 08:51.04 | sxpert-work | benjk: I figured... it's just that it was way too easy :D |
| 08:51.10 | benjk | :) |
| 08:51.50 | benjk | JT: I realise what it does, but I was asking what purpose it is doing for |
| 08:52.24 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 08:53.56 | benjk | because if it uses the console only in order to advertise the fact that it is still alive, then this would seem to be a rather inefficient and ill designed thing |
| 08:54.34 | JT | benjk: i think it's so admins can play with it |
| 08:54.37 | JT | but i didn't design it |
| 08:54.39 | JT | so i dunno |
| 08:55.00 | benjk | so the virtual console has nothing to do with the restart mechanism at all then |
| 08:56.29 | benjk | in which case all L|NUX has to do is remove the -c from the /usr/sbin/asterisk invocation |
| 09:00.17 | JT | benjk: i would think L|NUX would just need to change CONSOLE=yes # Whether or not you want a console |
| 09:00.27 | JT | to CONSOLE=no |
| 09:00.57 | benjk | fair enough, I presume that will do the same thing, remove the -c from the command |
| 09:01.08 | RoyK | morning, morons |
| 09:01.15 | benjk | morning troll |
| 09:01.19 | RoyK | :) |
| 09:01.22 | benjk | :) |
| 09:01.43 | L|NUX | JT : not working |
| 09:02.45 | *** join/#asterisk Kapsel (i=linknet@irc.thinkgeek.dk) |
| 09:03.13 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:03.36 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 09:03.37 | key2 | hey |
| 09:04.12 | RoyK | hi, ken___ |
| 09:04.12 | RoyK | eh |
| 09:04.16 | RoyK | hi, key2 |
| 09:04.58 | ken___ | RoyK: uh .. .? |
| 09:05.10 | RoyK | ken___: sorry. tab completion fsckup |
| 09:05.44 | JT | L|NUX: ok |
| 09:08.25 | L|NUX | RoyK : howdy |
| 09:09.00 | *** join/#asterisk bXi (i=bluepunk@irssi.co.uk) |
| 09:09.01 | RoyK | hi |
| 09:09.24 | L|NUX | how are you doing |
| 09:09.24 | *** join/#asterisk Kuto (n=kuto@210.213.243.9) |
| 09:09.28 | L|NUX | long time no see |
| 09:12.37 | tengulre11 | I got many 'FXO PCI Master abort' in /var/log/messages. why?? |
| 09:17.30 | tengulre11 | anybody active?? |
| 09:28.22 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 09:32.11 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 09:36.07 | faberk64 | hi to all |
| 09:37.13 | *** join/#asterisk frenzy (n=frenzy@196.46.104.224) |
| 09:37.41 | faberk64 | quick question: I've got problems with incoming Zap calls, sometimes the number is incomplete, so * do not pass the call |
| 09:37.45 | frenzy | can anyone suggest a good tool for creating correct IVR formats I have my IVR in .wav at moment |
| 09:38.00 | faberk64 | I think is * that answer too fast |
| 09:38.11 | faberk64 | How can I fix this problem? |
| 09:38.41 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 09:38.58 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:41.27 | Newbie___ | hi all, where do i go for asterisk@home help ? |
| 09:42.21 | phearless | Newbie___: read the fuc*ing topic |
| 09:44.26 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 09:44.34 | Newbie___ | phearless: i do not have the luxury of a big screen, so the fuc*ing topic is all cramp and all i can see is "Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.12.1, Zaptel 1.2.9.1 released!" |
| 09:45.06 | phearless | the luxury of a big screen ? |
| 09:45.19 | yatesy | get the luxury of a decent IRC client then :P |
| 09:45.30 | phearless | Newbie___ VERSION mIRC v6.12 Khaled Mardam-Bey |
| 09:45.41 | yatesy | haha |
| 09:46.19 | phearless | So mirc do not support the feature : "see the topic of the chan" |
| 09:46.27 | phearless | this is sad. |
| 09:46.34 | Newbie___ | ya is sad |
| 09:46.43 | JT | it does, maybe he has it restricted to a 50pixel high window |
| 09:47.30 | JT | in any case, he'd be seeing the bottom of the topic, not the top, so it makes little sense to me |
| 09:47.31 | sxpert-work | that mIRC thing still exists ??? the horror |
| 09:49.40 | Strom_C | maybe it's windows 3.1 on a 640x480 screen |
| 09:50.34 | e-ddie | i run windows 1.1 on 320x200 |
| 09:50.36 | e-ddie | it's great |
| 09:50.45 | Strom_C | hahahaha |
| 09:56.46 | *** join/#asterisk grexk (n=grexk@124.107.72.45) |
| 10:04.06 | *** part/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 10:06.26 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 10:08.06 | *** part/#asterisk grexk (n=grexk@124.107.72.45) |
| 10:11.29 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 10:11.30 | Bert- | hello there |
| 10:11.52 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 10:11.56 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 10:12.16 | kink0 | hi, anyway to reload g729 licences without stoping the service ? |
| 10:12.54 | kink0 | i have added more licences and I don't like to stop asterisk. |
| 10:13.19 | kink0 | of course reload codec_g729a.so claims is not reloadable module |
| 10:14.07 | RoyK | unload/load |
| 10:14.09 | RoyK | restart now :) |
| 10:14.39 | Bert- | can someone have a look at that pleasE? |
| 10:14.40 | Bert- | http://pastebin.ca/170036 |
| 10:14.51 | Bert- | I got 401 |
| 10:14.54 | Bert- | dunno why |
| 10:17.40 | kaldemar | that's the way registering goes if you require users to authenticate. the server responds with 401 and the nonce (and other info) in the WWW-Authenticate header, and the client should then send a new REGISTER with the authentication info. |
| 10:18.46 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 10:33.14 | Bert- | kaldemar why ? |
| 10:33.26 | Bert- | my phone send a REGISTER and got a unauthorized |
| 10:33.34 | Bert- | why should I send REGISTER AGAIN? |
| 10:33.38 | Bert- | (sorry for caps) |
| 10:34.19 | Bert- | as I MUST register users with dynip |
| 10:34.26 | Bert- | and then |
| 10:34.38 | Bert- | why does it work sometimes then doesn't work anymore ? |
| 10:34.45 | Bert- | without modifiying anything ? |
| 10:34.58 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 10:40.40 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 10:40.57 | _deg_ | how to compile just one module for asterisk? |
| 10:41.10 | _deg_ | i need to add res_odbc |
| 10:47.36 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:54.44 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 10:54.47 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 10:55.02 | Winkie | hey guys, any of you use CDRs for billing and know how to properly track transfers? Becuase I can't seem to find out what transfers are associated with what inbound calls |
| 10:58.27 | *** part/#asterisk frenzy (n=frenzy@196.46.104.224) |
| 11:01.53 | stephane_ | re |
| 11:03.51 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 11:07.07 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:12.31 | *** join/#asterisk IvyUK (n=mark@194.201.148.215) |
| 11:12.32 | *** join/#asterisk TimothyP (n=timothy@212.228-136-217.adsl-static.isp.belgacom.be) |
| 11:12.52 | nfi|ermes | hi, anyone can help me to find documentation about the voice's volume in asterisk ? |
| 11:13.08 | TimothyP | Hi, I'm looking at an example sip.conf file, a number of [XXXX] elements are defined in it, but only on of them has the regexten=1001 parameter, what does it mean and why would only that one have it? |
| 11:13.39 | nfi|ermes | i listen too high voice of the other part |
| 11:15.53 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 11:17.45 | IvyUK | could someone be kind enough to send/pastbin me a zaptel.conf with t/e1's and analogue channels configured? |
| 11:18.13 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:25.31 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 11:25.51 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 11:25.51 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:26.01 | backblue | morning |
| 11:26.27 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 11:27.29 | Ahrimanes | anyone here ever talk to cicero networks about their cellphone voip clients? |
| 11:30.07 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:30.37 | *** join/#asterisk VonGodric (n=VonGodri@tuli.elion.ee) |
| 11:30.40 | VonGodric | hello |
| 11:31.02 | VonGodric | anyone here who can give me a hand and helpt to solve a small problem? |
| 11:32.25 | *** join/#asterisk michael-i (n=michael@141.41.38.58) |
| 11:32.31 | VonGodric | anyone? |
| 11:32.31 | *** join/#asterisk tropobot (n=shekhar@59.184.17.39) |
| 11:33.01 | Makenshi | VonGodric, if you provide a description of your problem people are more likely to respond with an answer |
| 11:33.16 | VonGodric | okay |
| 11:33.21 | VonGodric | I have this small task |
| 11:33.27 | VonGodric | to connect asterisk server |
| 11:33.29 | VonGodric | as a client |
| 11:33.35 | VonGodric | to another SIP based account |
| 11:34.18 | VonGodric | I tryed to skim around |
| 11:34.18 | VonGodric | but don't have clear idea how to do it |
| 11:34.18 | VonGodric | that is somewhere on the internet |
| 11:34.56 | toerkeium | have your ITSP gave you information about how to connect ? |
| 11:35.07 | VonGodric | nope |
| 11:35.16 | VonGodric | I only have account information |
| 11:35.23 | VonGodric | the way how simple clients connect |
| 11:36.34 | toerkeium | you wan't to connect * to a sip account.. provided by who ? |
| 11:36.43 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:37.23 | VonGodric | don't think that you know it |
| 11:37.29 | VonGodric | it's estonian telekom company |
| 11:37.31 | VonGodric | elion |
| 11:38.19 | toerkeium | well.. I am new at all this, but then you have to think about that sip account as a trunk, at first, I guess |
| 11:38.39 | VonGodric | me very new tooo |
| 11:38.51 | VonGodric | trying to learn it all |
| 11:38.52 | toerkeium | what's the information you have about the account ? username, password, ip address? |
| 11:39.03 | VonGodric | and proxy |
| 11:40.10 | toerkeium | try adding in sip.conf username:password@ip.address/extension |
| 11:40.25 | toerkeium | and then try sip show registry to see the state of the registration process |
| 11:40.35 | toerkeium | does your server have any configuration at all ? |
| 11:41.04 | VonGodric | where exactly do you think I should put it? |
| 11:41.05 | toerkeium | sorry, the line in sip.conf is register => username:password@ip.address/extension |
| 11:41.17 | VonGodric | [sip_proxy-out] ` |
| 11:41.18 | toerkeium | at the end of the sip.conf line |
| 11:41.18 | VonGodric | ? |
| 11:42.25 | toerkeium | that line makes asterisk try to register to a sip trunk as any other sip device |
| 11:43.07 | toerkeium | so, as it works by itself, don't include it in any other configuration block, just put it at the end of the sip.conf file and reload your * configuration |
| 11:43.57 | VonGodric | okay going to try |
| 11:43.58 | tropobot | hello there - im tring to get dundi working and reeeeeeely need help |
| 11:43.58 | VonGodric | :P |
| 11:44.25 | tropobot | i have set up two computers in our LAN with Asterisk |
| 11:44.41 | tropobot | Here are the configuration files http://pastebin.ca/170113 |
| 11:44.44 | toerkeium | VonGodric: read the "Asterisk TFOT" book published as PDF, it's a good book and will give you a nice picture of what asterisk is and how it works |
| 11:45.33 | tropobot | The problem is that DUNDILOOKUP() returns an empty string |
| 11:46.18 | tropobot | can anyone please throw me a line |
| 11:46.29 | tropobot | :) |
| 11:46.45 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 11:47.03 | VonGodric | toerkeium: it's a great book. I have it open right now ;) |
| 11:48.06 | toerkeium | VonGodric: read it, I could made lot of things after reading the book, and then go to the asterisk help sites, and come back with a better idea here .. then people will answer :) |
| 11:49.06 | toerkeium | I think that basic questions are not covered here, because it would take lot of time to explain what books or help web sites are already doing |
| 11:49.17 | VonGodric | I am reading it ;) |
| 11:49.44 | *** join/#asterisk shtoom (n=godson@59.184.17.39) |
| 11:56.06 | *** join/#asterisk fulgas (n=fulgas@80.172.227.30) |
| 11:58.52 | tropobot | hello - anyone there? |
| 12:00.18 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:01.28 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 12:02.51 | inspired | ouch |
| 12:03.20 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 12:03.53 | tropobot | anyone there who can help with a dundi problem? i have set up two computers in our LAN with Asterisk. The problem is that DUNDILOOKUP() returns an empty string. Here are the configuration files http://pastebin.ca/170113 - can anyone please throw me a line pleeeeeeez? |
| 12:05.17 | *** join/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 12:08.17 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 12:09.21 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:09.22 | tropobot | VonGodric - you can put the register statement in the 'general' context |
| 12:10.20 | tropobot | then you can build a sip block to communicate with the other peer |
| 12:10.20 | file | [TK]D-Fender: moo |
| 12:11.12 | tropobot | add a context in extensions.conf (or maybe you'd like to add in your default context or whatever) to handle calls coming from the peer |
| 12:11.31 | Winkie | hey [TK]D-Fender thanks for the help yesterday |
| 12:11.43 | Winkie | I still need to find out how to track transfers with CDR |
| 12:11.50 | Winkie | because there seems to be absolutely no facility for anything like it |
| 12:11.51 | Winkie | which is :( |
| 12:11.54 | [TK]D-Fender | file: *arrff* |
| 12:11.57 | [TK]D-Fender | Winkie: np |
| 12:12.52 | tropobot | you can place calls through the trunk by exten => <your extension>,1,Dial(SIP/<name of block used to use the account>/<the extension>) |
| 12:13.00 | file | [TK]D-Fender: wasabi? |
| 12:13.29 | tropobot | can anyone pleeeeez help me a bit with my dundi problem pleeeeez? |
| 12:13.46 | file | I'm trying to go through my music collection and find pieces to make my manager go "wtf?" |
| 12:13.58 | Winkie | tropobot: was that in response to me? :) |
| 12:14.23 | tropobot | no - :) - it was for VonGodric |
| 12:14.26 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 12:14.31 | Winkie | ok good, cause i was totally confused for a second :d |
| 12:14.40 | Winkie | none of you bill using CDR? |
| 12:14.40 | tropobot | im sorry |
| 12:15.22 | Winkie | haha don't apologise |
| 12:15.42 | *** join/#asterisk Szolke (n=Szolke@22-36.adsl.etel.hu) |
| 12:15.47 | tropobot | :) |
| 12:16.21 | Szolke | Hi everybody |
| 12:16.27 | tropobot | hi |
| 12:19.54 | VonGodric | where can I find info on connecting asterisk as a client to another sip server? |
| 12:20.47 | [TK]D-Fender | VonGodric: http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers |
| 12:21.47 | VonGodric | I have no idea what the other server is. it's not asterisk for as far as I know. |
| 12:24.03 | [TK]D-Fender | VonGodric: Same deal. SIP is SIP. You register. You set up a pair for user/peer entires or just a [firend] if you can. |
| 12:25.02 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 12:26.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:26.42 | Szolke | Ihave a problem vit asterisk. We use the v42243. Ican't connect to the running asterisk, with the asterisk -vvvvvrg. The /var/run/asterisk is open. |
| 12:28.20 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:28.22 | [TK]D-Fender | Szolke: To connect to a running * you need to do "asterisk -r" |
| 12:28.46 | [TK]D-Fender | Szolke: What you wrote is the way to run * direct from Linux CLI without becoming a daemon. |
| 12:29.05 | [TK]D-Fender | ERRR... might be mistaken... |
| 12:29.18 | [TK]D-Fender | not sure on r & g together... I think I need more coffee. |
| 12:30.23 | Winkie | incidentally does anyone know of a cheap e1 failover switch? |
| 12:31.40 | nfi|ermes | hi, anyone can help me to find documentation about the voice's volume in asterisk ? |
| 12:31.46 | nfi|ermes | [TK]D-Fender |
| 12:31.53 | nfi|ermes | i listen too high voice of the other part |
| 12:32.10 | *** join/#asterisk NebulousNL_ (n=a@office.telecom.tno.nl) |
| 12:32.12 | *** part/#asterisk NebulousNL_ (n=a@office.telecom.tno.nl) |
| 12:32.58 | Szolke | We use a script to start asterisk when the server start. If type ps aux|grep asterisk: i saw asterisk is running. |
| 12:33.05 | Szolke | -r is not working |
| 12:34.13 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 12:34.48 | Szolke | the file: asterisk.ctl is not exist in the /var/run/asterisk |
| 12:35.01 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 12:36.09 | Bert- | does someone use SIP Phone on a remote network with asterisk please ? |
| 12:36.20 | Bert- | thomson 2030 particulary |
| 12:39.11 | Bert- | always 401 unauthorized |
| 12:39.11 | Bert- | the phone is behind a nat |
| 12:39.11 | Bert- | asterisk too |
| 12:39.11 | Bert- | (not the same) |
| 12:39.11 | Bert- | phone send register, asterisk says trying, then 401 not authorized. Someonetold me it is normal, I have to send another register request |
| 12:39.30 | *** join/#asterisk BertZ (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 12:39.30 | BertZ | disconnected :( |
| 12:39.31 | *** join/#asterisk LoneShadow (n=duh@59.92.154.70) |
| 12:40.16 | Bert- | Well I don't understand why no one is able to explain me clearly what is wrong |
| 12:40.33 | Szolke | BertZ: do you see a outbound proxyoption? |
| 12:40.48 | Bert- | register proxy ? |
| 12:40.53 | Szolke | in the phone' setup |
| 12:41.27 | Szolke | i had this problem with a AT320 too |
| 12:41.38 | Bert- | I have service domain, Registar server address and Proxy server address |
| 12:42.13 | Szolke | did you set the register port to 5060? |
| 12:43.26 | Bert- | yep |
| 12:45.25 | Szolke | can you send me a screenshot from the sip settings? to my mail address |
| 12:45.56 | IvyUK | Winkie: what are you trying to achieve with the failover switch |
| 12:46.42 | Winkie | EvilDeshi: we have two asterisk servers and several E1s, only two at the current site |
| 12:46.52 | Winkie | we want to simply be able to switch these E1s over to another machine remotely |
| 12:47.16 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:47.18 | Winkie | the products obviously exist but they're designed for situations when you have 12+ E1s, and we don't :) |
| 12:47.19 | IvyUK | are they used for inbound ? |
| 12:47.28 | Winkie | they are |
| 12:47.33 | Winkie | in and out |
| 12:47.43 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 12:50.16 | IvyUK | its got mailshotted this, this morning.. it would definatly work for outbound http://www.patapsco.co.uk/applications/isdn_conversion_sharing_and_simulation/share_pris/share%20PRIs_LibD.htm not sure how it would handle inbound |
| 12:53.44 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 12:54.00 | Assid | man.. i wish this freecall/sipdiscount accepted callerid/ani |
| 12:54.10 | Winkie | looks quite nice, we don't need complete failover, a physical disconnect and reconnect is fine anyway, we'll be expecting alarms from our provider |
| 12:54.19 | Winkie | the problem is the physical connections, did you get any pricing info on that IvyUK? |
| 12:54.43 | [TK]D-Fender | Hey I'm working on a Debian install and trying to compile an * add-on that requires curl-config. I did "apt-get install libcurl" and that did at least one package. Is there another devel or similar package someone can point me to? |
| 12:54.45 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 12:55.01 | [TK]D-Fender | - /bin/sh: line 1: curl-config: command not found |
| 12:55.12 | Winkie | [TK]D-Fender: i assume you've run `apt-cache search curl`, it's probably libcurl-dev or similar |
| 12:55.34 | Assid | [TK]D-Fender: libcurl-dev i think |
| 12:55.36 | [TK]D-Fender | Winkie: No... I'm only barely functional on debian |
| 12:55.44 | IvyUK | Winkie: nah no pricing but its only 4 port shouldnt be that much |
| 12:55.49 | Winkie | hat's a shame, debian is beautiful |
| 12:56.03 | Assid | yeah it is |
| 12:56.12 | Winkie | IvyUK: they're all usually insanely priced due to the closed nature of old-telecoms :) |
| 12:56.23 | [TK]D-Fender | DAMN.... |
| 12:56.33 | [TK]D-Fender | I installed it an yet another new error... |
| 12:56.42 | [TK]D-Fender | make: *** No rule to make target `apps_env'. Stop. |
| 12:56.44 | [TK]D-Fender | UGH |
| 12:56.46 | E-bola | hey guys i have an anoying problem |
| 12:56.50 | Assid | make? |
| 12:56.54 | Winkie | [TK]D-Fender: you modified the apps makefile? |
| 12:57.01 | E-bola | When ppl call in we have waiting music by passing the phone we dial the tm parameter |
| 12:57.01 | Winkie | E-bola: given your name it seems quite serious :o |
| 12:57.04 | E-bola | thats working fine |
| 12:57.07 | Assid | what you making? |
| 12:57.08 | *** part/#asterisk michael-i (n=michael@141.41.38.58) |
| 12:57.12 | *** join/#asterisk Druken (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:57.13 | E-bola | But if nobody picks up another phones starts ringing |
| 12:57.24 | [TK]D-Fender | Winkie: This is a seperate install script for ValetParking that works great on RH setups I've tested... |
| 12:57.24 | E-bola | but that restarts the waiting music, is there any way to have the same song continue playing? |
| 12:57.54 | Winkie | [TK]D-Fender: ah, well that's quite interesting, there's obviously a problem with the makefile, are you just running `make` or specifically targeting apps_env? |
| 12:57.56 | [TK]D-Fender | E-bola: You'd have to use an external program that latches onto a streaming source. |
| 12:57.58 | Druken | E-bola: use the same moh context? |
| 12:58.11 | E-bola | Drunken: what do u mean? |
| 12:58.11 | Winkie | as Druken said the same moh context worked for me |
| 12:58.18 | E-bola | i only got 1 context |
| 12:58.34 | Winkie | E-bola: what player do you use? |
| 12:58.39 | Druken | well, my moh stops when it's not being played for anyone else... |
| 12:59.08 | Druken | so i can listen to it, stop, and come back to it 30 min later and it carries on from the point i left it |
| 12:59.13 | E-bola | well in musiconhold.conf i have a [native] clause |
| 12:59.23 | E-bola | mdoe is files |
| 12:59.25 | E-bola | mode even |
| 12:59.40 | E-bola | is that what u mean by player? |
| 12:59.45 | E-bola | its been a while since i set it up hehe |
| 12:59.47 | Druken | that is probably why it don't work the same way... |
| 12:59.59 | Druken | i use the external mpg123 |
| 13:00.08 | IvyUK | I have 124 e1 channels configured in zaptel.conf and it works nicely. I have added a TDM400p to the server and added fxoks=125-128 but when i load zaptel it complains "ZT_CHANCONFIG failed on channel 125: No such device or address (6)" before it has loaded wctdm... have i missed something? my zaptel.conf is @ http://pastebin.ca/170185 |
| 13:00.13 | E-bola | how did u configure that drunken? |
| 13:00.21 | E-bola | im pretty sure i use mpg123 too |
| 13:00.24 | E-bola | i remember compiling it |
| 13:00.46 | [TK]D-Fender | E-bola: What mode is your MoH using? |
| 13:00.50 | E-bola | drunken: how do u activate music on hold? |
| 13:01.02 | E-bola | tkd-fender: files |
| 13:01.11 | [TK]D-Fender | IvyUK: Did you do "modprobe wctdm" ? |
| 13:01.15 | Druken | IvyUK: if you have 124 E1 channels, why are you fucking around with a TDM card? and check your lspci for the tdm card... make sure it's being loaded |
| 13:01.32 | [TK]D-Fender | E-bola: That tells * to NOT use MPG123 and use Native MoH instead. |
| 13:01.51 | E-bola | well is this related to palyer at all? |
| 13:01.56 | E-bola | i mean i think its about my extension plan |
| 13:01.57 | [TK]D-Fender | IvyUK: Also not sure if you should recompile Zaptel as well... |
| 13:02.09 | E-bola | player even |
| 13:02.10 | tzanger | woot |
| 13:02.17 | tzanger | out of 35 boards, 3 have the LEDs on the right way |
| 13:02.19 | tzanger | hahaha |
| 13:02.28 | [TK]D-Fender | E-bola: What I said is what's happening. "mode=files" mean Native MoH not mpg123 |
| 13:02.38 | IvyUK | [TK]D-Fender: done a modprobe wctdm |
| 13:02.42 | [TK]D-Fender | tzanger: Almost 10%! |
| 13:02.46 | Winkie | IvyUK: does your ztcfg -vv return all the channels you're expecting? |
| 13:02.46 | E-bola | [TK]D-Fender: ok, but my question is would using mpg123 solve any problem? |
| 13:02.48 | Druken | tzanger: leds for what ? |
| 13:02.54 | [TK]D-Fender | E-bola: Yes |
| 13:03.05 | E-bola | i mean its not a quwstion of not being able to play stuff, its about not restarting songs when u transfer a call |
| 13:03.12 | IvyUK | Drunken: need some analogue handsets and lspci lists it no problem |
| 13:03.27 | Winkie | IvyUK: does your ztcfg -vv return all the channels you're expecting? |
| 13:03.31 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:03.36 | [TK]D-Fender | IvyUK: You should have gotten ATAs |
| 13:03.42 | Druken | as [TK]D-Fender mentioned, did you modprobe the tdm module? |
| 13:03.51 | puzzled | hi |
| 13:03.51 | E-bola | [TK]D-Fender: maybe im clueless but why woudl it help to switch to using mpg123? |
| 13:03.52 | [TK]D-Fender | IvyUK: pastebin "cat /proc/interrupts" for use please |
| 13:03.53 | [TK]D-Fender | ~pb |
| 13:03.55 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 13:03.57 | Druken | [TK]D-Fender: i concur... |
| 13:04.18 | IvyUK | Winkie: yes ztcfg lists all channels |
| 13:04.19 | [TK]D-Fender | E-bola: mpg123 keeps a continuous stream going and doesn't restart. Native does. |
| 13:04.34 | E-bola | [TK]D-Fender: lol well if thats true i guess my solution is quite simple :) |
| 13:04.35 | [TK]D-Fender | us* |
| 13:04.38 | IvyUK | and yes i have modproce wctdm |
| 13:04.56 | [TK]D-Fender | IvyUK: pastebin please.... |
| 13:05.58 | IvyUK | [TK]D-Fender: which parts do you want ? |
| 13:06.03 | IvyUK | doh sorry |
| 13:06.51 | Winkie | interrupt problems shouldn't stop a span detection though |
| 13:07.04 | Winkie | IvyUK: can you pastebin the contents of the files in /proc/zaptel and your ztcfg -vv output? |
| 13:07.08 | [TK]D-Fender | Assid: I downloaded my app in the wrong folder. stupid mistake! All works :) |
| 13:07.32 | IvyUK | paste bin coming up |
| 13:07.52 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 13:08.03 | *** join/#asterisk gaspiz (n=gaspiz@86.35.34.63) |
| 13:09.09 | [TK]D-Fender | IvyUK: Use pastebin.ca |
| 13:09.12 | IvyUK | http://pastebin.ca/170190 |
| 13:09.20 | gaspiz | where can I read about new features in 1.2.12? I would like to know the new things since 1.2.1 (currently used on my devices) |
| 13:09.39 | IvyUK | ignore the fact that it only goes to 126 i just took two out it did read 128 |
| 13:09.41 | stoffell | gaspiz: Changelog ? I find it the most interesting file to read.. |
| 13:09.44 | [TK]D-Fender | IvyUK: Ok, please PB your zaptel & zapata |
| 13:10.23 | IvyUK | zaptel.conf is http://pastebin.ca/170185 |
| 13:10.30 | gaspiz | stoffell: is it available on web somewhere? |
| 13:10.45 | IvyUK | i thought zapata didnt get used until asterisk was loaded |
| 13:10.58 | stoffell | gaspiz: it's included in the 1.2.12 tgz download.. |
| 13:10.59 | gaspiz | stoffell: found it :) |
| 13:11.02 | stoffell | :) |
| 13:11.36 | [TK]D-Fender | IvyUK: Yeah thats looking pretty normal :/ |
| 13:12.26 | [TK]D-Fender | IvyUK: Wait... ztcfg reacts OK, and its just * that panics? |
| 13:13.23 | tropobot | anyone there who can help with a dundi problem? i have set up two computers in our LAN with Asterisk. The problem is that DUNDILOOKUP() returns an empty string. Here are the configuration files http://pastebin.ca/170113 - can anyone please throw me a line pleeeeeeez? |
| 13:14.40 | IvyUK | [TK]D-Fender: its zaptel starting ... i have added the zaptel startup to http://pastebin.ca/170199 |
| 13:15.54 | *** join/#asterisk shodan (n=shodan@ip187.99-113-216.pppoe4.joliette.intermonde.net) |
| 13:16.06 | [TK]D-Fender | IvyUK: Somewhat silly idea : You sure you have those FXS modules on the right ports on your card? |
| 13:16.13 | IvyUK | [TK]D-Fender: I havent actually tried to use them yet in *... is there a chance they are working anyway? so i should go and try use them? |
| 13:16.35 | IvyUK | i have 4 port card with 4 modules installed |
| 13:16.41 | shodan | anyone knows if using *67 costs something with bell canada if you don't have a special package ? |
| 13:16.43 | [TK]D-Fender | IvyUK: I think if ztcfg fails, * will freak out even worse. |
| 13:17.07 | [TK]D-Fender | IvyUK: Yet you are only configuring 2? |
| 13:17.36 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 13:18.37 | IvyUK | i was doing 4 i was just testing to see if there was something else... i am doing 4 again |
| 13:19.00 | *** join/#asterisk modcuts (n=phono@ppwood.gotadsl.co.uk) |
| 13:19.06 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:19.45 | mut | yay |
| 13:19.52 | mut | finally get my sbc badge |
| 13:19.59 | mut | er at& |
| 13:19.59 | mut | t |
| 13:20.45 | IvyUK | it look like it thinks channels 125-128 are on the TE412P hence why its loading before the wctdm module |
| 13:20.57 | Druken | mut: going to put it beside your community service badge ? |
| 13:21.34 | mut | next to my verizon badge |
| 13:22.27 | Druken | oh.... |
| 13:22.40 | [TK]D-Fender | Who is biggest registrar for domain names again? internic.com? I've got a domain to get set up.... |
| 13:22.50 | [TK]D-Fender | names are escaping me... |
| 13:23.16 | Druken | godaddy :) hahahahahaha |
| 13:23.18 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:23.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:23.46 | [TK]D-Fender | Druken: NOT them.... |
| 13:23.47 | IvyUK | i stay away from the big ones normally damn expensive |
| 13:23.51 | mut | mew |
| 13:24.04 | [TK]D-Fender | IvyUK: I wanted a bigger player who'll let me web-admin my DNS, etc. |
| 13:24.06 | *** join/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 13:24.32 | IvyUK | freeparking.com |
| 13:24.36 | Druken | lots of web based dns admin sites out there... |
| 13:24.49 | mut | i like godaddy myself |
| 13:25.13 | trevarthan | how would I go about recording channels on the fly? For example, in a call center environment where some conversations need to be monitored. I'm using trixbox currently, but I have experience with raw asterisk so I'm not limited to trixbox tools. Any ideas? |
| 13:26.11 | mut | use the manager to type monitor channel x |
| 13:26.17 | [TK]D-Fender | trevarthan: look up "features.conf" on the WIKI, and "show application dial" in * CLI |
| 13:26.37 | bXi | will bristuff work with the latest asterisk ? |
| 13:26.44 | *** join/#asterisk pablus (n=nn@test.conama.cl) |
| 13:27.02 | IvyUK | [TK]D-Fender: thanks for the help going to eat see if that will fix the card while im away |
| 13:27.05 | pablus | morning |
| 13:29.36 | Druken | why bother only monitoring some channels? do them all... |
| 13:29.37 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 13:29.43 | *** join/#asterisk Magicianx (n=magician@116-22.dr.cgocable.ca) |
| 13:29.48 | mut | yea! |
| 13:29.50 | mut | in raw format! |
| 13:30.11 | mut | be like... a bajillion gb |
| 13:30.42 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 13:33.20 | Assid | okay back |
| 13:33.22 | Assid | somewhat |
| 13:36.15 | trelane` | has anyone ever had voicemail randomly delete messages? |
| 13:39.28 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 13:42.58 | Bert- | I see Retransmitting #1 #2 #3 #4 ... |
| 13:43.07 | Bert- | is it because there is no answer on the other side ? |
| 13:48.15 | trelane` | Bert-, where are you seeing that, and what are you trying to do |
| 13:48.56 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:50.06 | *** join/#asterisk gardo (n=gardo@124.104.32.152) |
| 13:50.30 | shodan | where do websites like 411.ca get their information for reverse lookup ? (is there a way to get that for use with asterisk ? I'd love to do a reverse lookup on the cid and embed a google map of the caller in my emails from the voicemail) |
| 13:50.33 | gardo | hi! |
| 13:50.57 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 13:51.18 | gardo | can anyone point me to a good reading on how to setup a sip trunk? |
| 13:51.21 | trelane` | shodan, should you achieve this thing you embark upon you shall truly be enlightened. |
| 13:51.35 | Bert- | trelane In the cli |
| 13:51.54 | trelane` | gardo, don't know if they cover it, but voip-info.org is usually a good start |
| 13:51.54 | trelane` | Bert-, and what are you doing when you see it? |
| 13:51.55 | Bert- | I'm trying to register a thomson 2030 behind nat in Morocco |
| 13:52.02 | Bert- | to my Asterisk, behind another nat in France |
| 13:52.05 | gardo | i have defined a sip trunk in my sip.conf but dont know how to put it in extensions.conf |
| 13:52.14 | Bert- | i'm juste debugging |
| 13:52.21 | Bert- | the fone send register |
| 13:52.24 | gardo | it seems my sip trunk is being ignored |
| 13:52.26 | Bert- | * answer options |
| 13:52.46 | Bert- | I 'm wondering if it is a network issue or not |
| 13:53.34 | shodan | ?? uh ? |
| 13:53.42 | trelane` | gardo, who's the provider, and what's the connection option? |
| 13:53.56 | trelane` | shodan, it's going to be bloody difficult. |
| 13:54.01 | gardo | sip.voipcheap.com |
| 13:54.23 | *** join/#asterisk lirakis (n=tbright@h-68-165-94-219.nycmny83.covad.net) |
| 13:54.23 | gardo | trelane`, ordinary sip is the connection |
| 13:54.56 | lirakis | i have an extension that seems to be ringing itself at early hours of the morning.. usually right around 1am |
| 13:55.00 | trelane` | gardo, it'll look something like exten => <dial pattern>,1,Dial(SIP/sip.voipcheap.com/${EXTEN}) |
| 13:55.20 | lirakis | 2006-09-14 00:57:00 SIP/415... 415 "cust 415" <415> ResetCDR w t ANSWERED 00:21 5009 |
| 13:55.25 | lirakis | that is the cdr record |
| 13:55.35 | trelane` | <PROTECTED> |
| 13:55.46 | shodan | damn , it felt "almost there" seeing this http://411ca.whitepages.com/1234/map_provider?full_address=&city=Crabtree&state=QC&zip=J0K%201B0&country=CA&lat=&long= |
| 13:55.50 | trelane` | (as some sip phones have an alarmclock like feature) |
| 13:56.08 | lirakis | its an adapter.. i believe a grandstream handytone 386 |
| 13:56.08 | gardo | does it suppose to look like this: exten => <dial pattern>,1,Dial(SIP/username:password@sip.voipcheap.com/${EXTEN}) ? |
| 13:56.55 | Bert- | hmm |
| 13:57.06 | Bert- | Asterisk is sending OPTIONS all the time |
| 13:57.22 | Bert- | to a certain host |
| 13:57.24 | trelane` | gardo, you shouldn't need username/password if it's specified in sip.conf (which anything wtih a register line should be |
| 13:57.24 | Bert- | without receiving any paquet from the host |
| 13:57.36 | Bert- | is it possible and why ? |
| 13:57.40 | lirakis | .. one thing i noticed is that is saying the destination is "t" does any one know what that is? |
| 13:57.52 | trelane` | lirakis, I'd check the adapter's config and make sure it doesn't have such a thing... 't' is timeout in a context |
| 13:58.15 | trelane` | ie exten => t,1,somestuff (this is what gets executed if the caller doesn't enter anything) |
| 13:58.31 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 13:58.40 | trelane` | Bert-, are you qualify=yes on any host in sip.conf? |
| 13:58.55 | gardo | trelane, i only have 1 register line in my sip.conf |
| 13:59.06 | trelane` | gardo, is it for voipcheap.com? |
| 13:59.11 | gardo | yes |
| 13:59.33 | lirakis | hmm.. i also have another record that goes to dst "s" |
| 13:59.54 | gardo | trelane`, register => username:password@sip.voipcheap.com:5060 |
| 14:00.14 | trelane` | gardo then use the example I provided and lose the username:password@ |
| 14:00.18 | lirakis | those are the only two.. that show up after 12pm .. and he is complaining that his phone has rung on its own early in the morning |
| 14:00.37 | trelane` | lirakis, s is Start in a context. So when a call enters that context s,1 is the first application to be executed |
| 14:00.40 | gardo | trelane` thanks! let me try that one |
| 14:00.47 | lirakis | .. i am trying to figure out why its ringing.. based on the cdr.. i dont know.. it seems to be calling it self .. but i really dont understand |
| 14:00.50 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:01.11 | lirakis | because src is from the ext. itself "415" |
| 14:01.21 | trelane` | lirakis, again, some phones and ata's have an alarmclock function, they ring themselves at a certain time. Otherwise check your asterisk script (the same can be done via a wakeupcall type app) |
| 14:01.58 | trelane` | lirakis, I'd do the following: 1. check your dialplan and make sure that there is nothing dialing that phone at 1 in the morning. 2. blow off the config on the ATA and reprogram it |
| 14:02.12 | Bert- | <trelane`> Bert-, are you qualify=yes on any host in sip.conf? <--- yes |
| 14:02.25 | trelane` | Bert-, that's what those sip packets are (most likely) |
| 14:02.34 | trelane` | brb now that my car's cooled I'm putting in some antifreeze :) |
| 14:02.35 | Bert- | what mean this option ? |
| 14:02.44 | trelane` | Bert-, it makes sure hte remote end is alive |
| 14:02.49 | Bert- | ok |
| 14:03.13 | lirakis | hmm.. |
| 14:03.22 | lirakis | okay thanks for your help trelane` |
| 14:07.27 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 14:07.38 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 14:09.06 | trelane` | back |
| 14:09.56 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 14:11.15 | backblue | anyone with AOC working? |
| 14:15.00 | lirakis | what is app-calltrace-perform? |
| 14:15.07 | lirakis | i mean.. i know it is a context |
| 14:15.11 | lirakis | but what does it do exactly? |
| 14:15.30 | lirakis | because that is the context that this "self dialing" ext. is executing |
| 14:15.39 | *** join/#asterisk n3c8 (n=broken@pix.office.vaioni.com) |
| 14:18.08 | trelane` | lirakis, err didn't you write the dialplan? |
| 14:19.11 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 14:20.57 | [TK]D-Fender | trelane : No, thats a FreePBX context.... |
| 14:21.15 | [TK]D-Fender | lirakis: Please read the channel topic. |
| 14:21.49 | Druken | ~amp |
| 14:21.52 | jbot | [amp] NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 14:28.41 | [TK]D-Fender | ~freepbx |
| 14:28.43 | jbot | well, freepbx is the Microsoft BOB of PBXes and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 14:28.53 | Bert- | hahaha :) |
| 14:29.00 | Bert- | BOB ? |
| 14:29.06 | Bert- | what is this acronym ? |
| 14:29.15 | Druken | big and over bloated |
| 14:29.17 | Druken | :) |
| 14:29.19 | lirakis | [TK]D-Fender: right.. go to freepbx |
| 14:33.28 | E-bola | Can anybody remember who it is who's writtent he snapanumber.com program? |
| 14:33.34 | E-bola | he was hangin out int his channel |
| 14:36.20 | trelane` | E-bola, remind me to buy you a new keyboard for christmas, good grief! :) |
| 14:36.59 | trelane` | Bert-, it stands for a product that's so bad, the only reason that it ever saw the light of day is that the project manager of Microsoft Bob was at the time banging the CEO of Microsoft (so her projects don't get canceled) |
| 14:37.11 | E-bola | ya its a bit crappy |
| 14:38.12 | E-bola | Im a bit confused about groups in asterisk |
| 14:38.13 | trelane` | a bit? |
| 14:38.24 | trelane` | that's like saying that the Tower of Piza is a bit bent |
| 14:38.42 | E-bola | If i wanna group together a group of phones in asterisk how is that best achived? |
| 14:39.02 | E-bola | like when u dial an extension all those groups ring and whoever picks up first gets the call |
| 14:39.16 | E-bola | i coudl do it with a long dial command, but isnt there a way to dial a group? |
| 14:39.21 | Druken | ahh, but see.. the tower isn't bent.. it just sank on one side... |
| 14:39.47 | trelane` | E-bola, long dial string |
| 14:39.47 | Druken | :P |
| 14:39.53 | trelane` | Druken, bastard. :( |
| 14:40.06 | E-bola | trelane? |
| 14:40.18 | Druken | trelane: you could always use a queue... |
| 14:40.19 | trelane` | E-bola, I use a long dial string, what I'm trying to do is get them all to answer (ie group intercom |
| 14:40.37 | Druken | meetme without a password :) |
| 14:40.41 | E-bola | isnt there a better way? |
| 14:40.45 | trelane` | Druken, got an example? |
| 14:40.54 | Druken | uhmm.... no :) |
| 14:41.00 | trelane` | I've been looking for the code snippit |
| 14:41.10 | Druken | try the wiki ? |
| 14:41.31 | trelane` | Druken, I did :) |
| 14:42.06 | Druken | although, it sounds like your trying to do something diffrent? |
| 14:42.13 | *** join/#asterisk slykens (n=slykens@mail.verimedservices.com) |
| 14:42.22 | Druken | do you want asterisk to ring all the phones and when they answer they are in the confrence? |
| 14:42.32 | Druken | sorta force a confrence setup ? |
| 14:42.59 | trelane` | Druken, I have snoms so I want to add the sipheader answer_after=0 |
| 14:43.02 | trelane` | group intercom basically |
| 14:43.13 | trelane` | I know I'm going to need a meet-me |
| 14:43.35 | trelane` | and a dial with some specified SipAddHeaders |
| 14:43.36 | Druken | sounds like you'd need to do it with call files, and an agi or something... |
| 14:44.09 | slykens | hi all. i know i am in for some pain but has anyone managed to have * behind a pix and a UA behind a generic NAT device and have it work? |
| 14:44.41 | tzanger | hmm |
| 14:44.52 | tzanger | is there a way to see the hint status of a given SIP peer from the CLI? |
| 14:44.57 | tzanger | sip show hints or something similar? |
| 14:45.10 | Druken | have an extension that goes directly into the meetme and use call files to ring each phone to that extension |
| 14:45.14 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:45.25 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 14:46.00 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 14:46.39 | flujan | hi guys... Does the canreinvite feature apply to the iax clients config files? I am having problems with transfer. The calling point always can transfer, but the caller cannot. Even If I use the tT option in the dial command. |
| 14:47.45 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 14:48.03 | *** join/#asterisk xnon (i=xnon@200.82.222.123) |
| 14:48.08 | xnon | hello friends |
| 14:48.38 | xnon | i wanna record and transfer calls and i set in the extension this exten => 55,1,Dial(SIP/55,20,wWtT) it is ok? |
| 14:49.24 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:49.56 | flujan | flujan, I have this problem with queues too. When I receive a call from a queue, I cannot transfer it to another point. |
| 14:50.55 | flujan | oops... the calling point can transfer... not the called one |
| 14:51.18 | xnon | anybody know how is posible record and transfer calls in a extension especific? |
| 14:51.31 | xnon | (SIP/55,20,wWtT) it is ok? |
| 14:52.40 | *** join/#asterisk cvaldess (n=hello@209.Red-83-53-44.dynamicIP.rima-tde.net) |
| 14:52.43 | cvaldess | Hi all |
| 14:53.00 | jbroome | note to self: if having problems compiling asterisk or zaptel, check the website |
| 14:53.14 | jbroome | i completly missed the .1 release for zaptel. der |
| 14:53.35 | *** join/#asterisk tropobot (n=tropobot@202.149.32.17) |
| 14:53.36 | cvaldess | have a problem with sip all call to my client are reporting X-Asterisk-HangupCause: Normal Clearing |
| 14:53.46 | tropobot | can anyone find me? |
| 14:53.49 | puzzled | jbroome: it was not announced |
| 14:54.23 | tropobot | i have some problem with dundi .. can any one help me with that ? www.dundi.com seems to be re-directed to thevoice.digium.com |
| 14:54.33 | cvaldess | even if hangupCause is 34, my client get Normal Clearing |
| 14:54.37 | jbroome | puzzled: which is why I need to check the site. :) Google saved me though. |
| 14:54.53 | puzzled | good |
| 14:54.56 | cvaldess | any one have same problem? |
| 14:55.25 | jbroome | cvaldess: i get the dundi page. |
| 14:55.51 | cvaldess | jbroome> ??? not using dundi here |
| 14:56.21 | jbroome | cvaldess: sorry, that should have gone to tropobot |
| 14:56.46 | cvaldess | ;) |
| 14:58.24 | cvaldess | have a problem with sip all call to my client are reporting X-Asterisk-HangupCause: Normal Clearing |
| 14:58.28 | cvaldess | even if hangupCause is 34, my client get Normal Clearing |
| 14:58.37 | cvaldess | any one have same problem? |
| 14:59.13 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 14:59.35 | *** join/#asterisk tropobot (n=shekhar@202.149.32.17) |
| 14:59.39 | *** part/#asterisk tropobot (n=shekhar@202.149.32.17) |
| 14:59.40 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 15:00.58 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 15:02.02 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:02.24 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 15:04.37 | *** join/#asterisk LoneShadow (n=duh@59.92.188.112) |
| 15:05.10 | *** part/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 15:05.20 | flujan | oops... the calling point can transfer... not the called one |
| 15:05.28 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 15:05.55 | bXi | does one of you have a clue on what goes wrong here http://pastebin.ca/170331 |
| 15:06.24 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.80.246) |
| 15:06.47 | bXi | anything that i should check into ? |
| 15:07.31 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:12.34 | cvaldess | have a problem with sip all call to my client are reporting X-Asterisk-HangupCause: Normal Clearing |
| 15:12.36 | cvaldess | even if hangupCause is 34, my client get Normal Clearing |
| 15:12.40 | cvaldess | any one have same problem? |
| 15:13.10 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:13.14 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:14.01 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:14.09 | *** join/#asterisk Modcuts (n=phono@ppwood.gotadsl.co.uk) |
| 15:14.58 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 15:15.00 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 15:16.34 | _alex_mx_ | bXi, are you using mISDN? |
| 15:16.48 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 15:18.16 | *** join/#asterisk Magicianx (n=magician@116-22.dr.cgocable.ca) |
| 15:21.03 | *** join/#asterisk thieumS (n=darkmind@87.240.67.41) |
| 15:21.25 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 15:22.31 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 15:23.20 | thieumS | Hi, i've got a rtp problem between * 1.2.12.1 and a snom sip phone (behind NAT), when incoming calls (from *, g729) arrive on the snom, i've got only voice in one way (snom -> asterisk), can somebody help please ? |
| 15:24.16 | Bert- | one way audio |
| 15:24.22 | thieumS | yep |
| 15:24.24 | Bert- | you should route some udp ports |
| 15:24.31 | Bert- | on your nat firewall |
| 15:24.35 | Bert- | in INPUT |
| 15:24.41 | puzzled | thieumS: are you using the sip jitterbuffer patch? |
| 15:24.51 | thieumS | nope puzzled |
| 15:25.10 | puzzled | thieumS: ok guess it's a firewall issue then |
| 15:25.15 | thieumS | when i look at ethereal traces, I don't see any RTP from * to snom |
| 15:25.47 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:26.04 | thieumS | that's why it's probably not a firewall issue |
| 15:26.08 | Bert- | is media routed or not ? |
| 15:26.15 | *** join/#asterisk Modcuts (n=phono@ppwood.gotadsl.co.uk) |
| 15:26.20 | Bert- | from where do you take trace ? |
| 15:26.20 | thieumS | nope Bert- |
| 15:26.26 | Bert- | hmm |
| 15:26.28 | thieumS | from * |
| 15:26.47 | Bert- | if média is not routed, then RTP traffic should come from caller ID not asterisk |
| 15:26.56 | thieumS | my * has a public IP, snom phone is behind NAT and registred |
| 15:27.22 | *** join/#asterisk ToyMan (n=stuq@cpe-24-161-95-214.hvc.res.rr.com) |
| 15:27.36 | thieumS | mmm interesting |
| 15:27.37 | jbroome | Can i do a make install with an older version of * running, then quit/restart when the new version is ready? |
| 15:27.56 | Bert- | try to route media |
| 15:28.03 | thieumS | how |
| 15:28.09 | Bert- | set canreinvite=no |
| 15:28.13 | thieumS | ok |
| 15:28.16 | thieumS | thx |
| 15:28.33 | Bert- | reinvite=no |
| 15:28.33 | Bert- | canreinvite=no |
| 15:28.38 | Bert- | for each phone |
| 15:31.02 | Bert- | GALOP TELECOM ? |
| 15:31.03 | *** join/#asterisk qzxcd (n=qzxcd@paranoia.coldev.org) |
| 15:31.05 | qzxcd | hi ppl |
| 15:31.10 | Bert- | ranma :) |
| 15:31.49 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 15:32.36 | toerkeium | guys, what hardware would be the minimal for a asterisk box wich will manage only 3 pure voip trunks with 10 extensions? |
| 15:32.43 | thieumS | thank you very much guys |
| 15:33.00 | Bert- | thieumS c ok ? |
| 15:33.13 | thieumS | non mais je suis sûr que c'est ça |
| 15:33.39 | Bert- | ok |
| 15:33.44 | thieumS | j'ai personne pour tester maintenant ;) |
| 15:34.31 | Bert- | then with media routed you can take traces |
| 15:34.31 | Bert- | but for sure RTP traffic is bloked on one side |
| 15:34.31 | *** join/#asterisk adorah (n=admin@87.68.149.143.cable.012.net.il) |
| 15:34.32 | Bert- | you can set RTP range in rtp.conf |
| 15:34.35 | thieumS | yep my rtp range is correctly set |
| 15:35.06 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 15:35.31 | thieumS | but you're right, i had a canreinvite=yes, and I see in the sdp RTP source IP is different from my * (pretty normal) |
| 15:35.32 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:36.04 | Bert- | toerkeium: depends of which codecs used, number of simultaneous calls, transcoding or not, media routed or not ... |
| 15:36.47 | *** join/#asterisk [Outcast] (n=bill@219-89-206-239.adsl.xtra.co.nz) |
| 15:36.54 | Bert- | actually I use PIII 500 192RAM 40Go HD |
| 15:37.05 | Bert- | have 2 SIP TRUNK and 15 extensions |
| 15:37.08 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 15:37.10 | jbroome | nice |
| 15:37.11 | Bert- | using g729 with transcoding |
| 15:37.20 | Bert- | and media routed |
| 15:37.31 | Bert- | 3-4 simultaneous calls |
| 15:37.42 | jbroome | yay older hardware! |
| 15:37.48 | Bert- | but this box is not dedicaced |
| 15:38.05 | Bert- | there are some others services on it (httpd, samba) |
| 15:38.09 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:38.18 | Bert- | and it is far from fully loaded |
| 15:38.32 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 15:39.11 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:41.29 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 15:41.58 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 15:42.22 | phearless | djbdns works on FC5 ? |
| 15:42.47 | fourcheeze | define "works" |
| 15:43.03 | *** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it) |
| 15:44.02 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 15:44.24 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 15:44.53 | Assid | okay gotta find a dedi box in .sg or .hk |
| 15:46.36 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.2) |
| 15:47.18 | E-bola | whats the english term when u have somebody calling while u are already in a call, and u get a small beep or a blinking light? |
| 15:47.34 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 15:47.39 | *** join/#asterisk mtoups (i=toups@HOWARD.INDYPGH.ORG) |
| 15:47.42 | *** join/#asterisk saftsack (n=oliver@p54A7D56D.dip.t-dialin.net) |
| 15:47.56 | wunderkin | E-bola, call waiting |
| 15:49.13 | flujan | hi guys... i am using the queue app with the T and t options... But when I try to transfer a call, it just hangs up the call. |
| 15:49.18 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:49.19 | flujan | The transfer is not working at all. |
| 15:49.50 | flujan | Just my caller parts can transfer the called extension cannot transfer. Even if I use the tT option. |
| 15:51.30 | smackus | when doing the queue command in extensions.conf, or in the queue settings in the queues.conf, where do i set how long each agent rings for before moving on to the next agent? |
| 15:53.13 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 15:54.21 | af_ | to adjust volumes on zap channels: do I need the patched ztmonitor (that displays numbers?) |
| 15:54.24 | rpm | was the expected release of asterisk 1.4 released for von? |
| 15:54.58 | cvaldess | have a problem with sip all call to my client are reporting X-Asterisk-HangupCause: Normal Clearing |
| 15:55.02 | cvaldess | even if hangupCause is 34, my client get Normal Clearing |
| 15:55.06 | cvaldess | any one have same problem? |
| 15:55.42 | jtexter3 | What's the policies for submitting EXTREMELY minor patches, where it doesn't make sense to have others test? i.e. adding #ifdef __cplusplus and corresponding #endif to header files? I'm getting ready to fax over the disclaimer, so once I have that done, can I just open a bug and submit a patch? |
| 15:59.13 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 15:59.16 | TripleFFFF | how can [agi_dnid] => unknown ?????????? |
| 16:00.02 | TripleFFFF | i get this |
| 16:00.05 | TripleFFFF | is that even possible ? |
| 16:00.17 | SplasPood | Would everyone suggest I get the TE110P for a single E1 deployment, or is there another card you'd suggest? |
| 16:00.39 | TripleFFFF | sangoma |
| 16:00.39 | TripleFFFF | ;) |
| 16:01.04 | tzanger | I've had no issues with digium or sangoma's stuff in standard t1 or pri environments |
| 16:01.12 | tzanger | echo cancellation is iffy with both, unfortunately |
| 16:01.31 | TripleFFFF | k |
| 16:01.32 | tzanger | working on issues with sangoma's octasic chip at the moment, can't wait for the digium RMA to return so I can try digium's octasic |
| 16:01.38 | TripleFFFF | can you tell me that tzanger ? |
| 16:01.41 | TripleFFFF | <PROTECTED> |
| 16:01.44 | tzanger | can you tell you what |
| 16:01.44 | TripleFFFF | on agi |
| 16:01.51 | tzanger | oh man I dunno |
| 16:01.53 | tzanger | I am not an AGI expert |
| 16:01.56 | tzanger | by ANY measure |
| 16:02.01 | TripleFFFF | s,1,agi(/maintenance/routing.php,${MACRO_EXTEN}); |
| 16:02.04 | TripleFFFF | hmm |
| 16:02.05 | tzanger | but I do have a question of my own... |
| 16:02.05 | TripleFFFF | ok |
| 16:02.13 | tzanger | can you see extension hint status from the CLI? |
| 16:02.14 | TripleFFFF | i added the arg |
| 16:02.16 | TripleFFFF | instaf |
| 16:02.21 | TripleFFFF | hint ? |
| 16:02.27 | TripleFFFF | no hints on my side |
| 16:02.27 | tzanger | TripleFFFF: I've had issues of various types trying to get AGI variables to work |
| 16:02.29 | TripleFFFF | dont ue |
| 16:02.31 | tzanger | TripleFFFF: no this is unrelated |
| 16:02.37 | TripleFFFF | oh |
| 16:02.43 | TripleFFFF | sho hints |
| 16:02.46 | TripleFFFF | show hits yes |
| 16:02.51 | TripleFFFF | darn .. hints |
| 16:02.56 | tzanger | oh nice |
| 16:03.18 | TripleFFFF | i use exten => 1001,hint,SIP/MYACOUNT |
| 16:03.26 | TripleFFFF | then exten => 1001,blah |
| 16:03.44 | tzanger | yeah I see show hints now... exactly what I was looking for |
| 16:04.08 | TripleFFFF | <PROTECTED> |
| 16:04.08 | TripleFFFF | ;) |
| 16:04.08 | tzanger | gotta go |
| 16:04.08 | TripleFFFF | yeah |
| 16:04.10 | TripleFFFF | if a cli parse was working right we could use the info |
| 16:04.16 | TripleFFFF | without having an acutaly EYE on it |
| 16:04.53 | SplasPood | TripleFFFF: What sangoma would you suggest, for comparison purposes |
| 16:04.59 | TripleFFFF | no idea |
| 16:05.30 | SplasPood | TripleFFFF: heh |
| 16:05.45 | SplasPood | The TE110P doesn't appear to have any echo cancelation onboard ? |
| 16:06.44 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 16:08.14 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 16:10.22 | aydiosmio | anyone have experience building .dic files for sphinx? |
| 16:12.16 | *** join/#asterisk deb_user (n=none@70-59-108-105.albq.qwest.net) |
| 16:12.31 | deb_user | anybody recommend a good softphone for ubuntu? |
| 16:12.53 | deb_user | I've tried twinkle, linphone, iaxcomm, ekiga...and openwengo |
| 16:13.07 | deb_user | none of them are really spectacular |
| 16:14.31 | coppice | but only openwengo sounds like a porn site :-) |
| 16:16.07 | deb_user | has anybody used sflphone? |
| 16:18.26 | coppice | everyone want to write their own soft phone. few want to collaborate to produce one really good one, and few seem capable of doing anything more than a gui |
| 16:19.30 | deb_user | coppice: as far as I'm concerned, the biggest limitation in * right now is the lack of quality softphones |
| 16:20.00 | deb_user | * is really advanced, and the hardware interfaces are really high quality too |
| 16:20.07 | deb_user | its the endpoints that are lagging behind |
| 16:20.53 | coppice | scratch a little deeper and you will find * has just as many issues as the phones |
| 16:22.11 | deb_user | coppice: at least the community is well organized |
| 16:25.00 | [TK]D-Fender | SplasPood: Clearly not. Didn't read the specs or wonder why there was a whole series of cards that DO offer it and the TE110P was NOT on the list? |
| 16:25.33 | [TK]D-Fender | deb_user: That isn't a limitation of *, thats a limitation of softphones for the protocol of your choice. |
| 16:25.34 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-6-123.slkc.qwest.net) |
| 16:25.45 | SplasPood | [TK]D-Fender: why do you think I asked for confirmation. |
| 16:25.45 | [TK]D-Fender | deb_user: Besides softphones SUCK. |
| 16:25.59 | mut | no always |
| 16:26.06 | SplasPood | [TK]D-Fender: Would you suggest its worth the extra money? |
| 16:26.10 | [TK]D-Fender | SplasPood: Sorry, your wording was a little wierd. No it does not have EC. |
| 16:26.11 | mut | using asterisk to create a 'teamspeak' server for 50 people |
| 16:26.17 | mut | is better than using teamspeak |
| 16:26.34 | deb_user | fender: that was my point exactly |
| 16:26.38 | [TK]D-Fender | SplasPood: So far there is only 1 card I recommend for T1 anyways, and you probably already know the answer. |
| 16:26.49 | mut | you could use a headset on a real phone on an ata, or a speakerphone on a phone via ata |
| 16:26.52 | SplasPood | Heh, actually I don't |
| 16:26.53 | mut | or you could use softphones |
| 16:26.55 | SplasPood | but I'd love to hear it.. |
| 16:27.32 | [TK]D-Fender | deb_user: And by "suck" I mean, who wants to use a PC as a phone anyways? What * needs is better scalability, SIP-B (coming), better queues (desperately... high value feature). |
| 16:28.10 | deb_user | fender: I use my pc as a phone |
| 16:28.17 | *** join/#asterisk Exstatica (i=exstatic@redline.mednor.net) |
| 16:28.20 | deb_user | when I'm working... |
| 16:28.25 | Exstatica | anyone use a cisco 7912 with asterisk? |
| 16:29.09 | mut | . |
| 16:29.35 | [TK]D-Fender | deb_user: Having a stupid piece of software pop up or worse start ringing and make you scramble to be able to answer in time is disruptive. Get a desk phone. |
| 16:30.00 | deb_user | fender: then I would need another fxs card |
| 16:30.17 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 16:30.20 | *** join/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 16:31.00 | mut | many keyboard anymore have shortcut keys |
| 16:31.07 | mut | all ya gotta do is program it to popup your phone |
| 16:31.28 | [TK]D-Fender | deb_user: Never use PCI for FXS unless you have to. ATA's or hardphones. thats it. |
| 16:31.46 | trevarthan | hey guys, When I install my digium T1 card into my machine my voicemail stops working. In the CLI I see it play the voicemail audio, but no sound comes out of my sip phones for the voicemail (ringing and sip-sip works fine). |
| 16:31.47 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 16:31.49 | deb_user | fender: what's an ata? |
| 16:31.51 | *** join/#asterisk jaike (i=jaike@210.5.117.158) |
| 16:32.09 | trevarthan | the digium card is configured (channels show up in `zap show channels`), but not plugged into a PRI. |
| 16:32.09 | [TK]D-Fender | deb_user: OMG... |
| 16:32.21 | trevarthan | When I remove the digium card, voicemail works again. Any ideas? |
| 16:32.30 | [TK]D-Fender | deb_user: http://www.voipsupply.com/product_info.php?products_id=713 |
| 16:33.08 | [TK]D-Fender | deb_user: Lets you use a normal phone as a SIP phone. |
| 16:33.26 | [TK]D-Fender | deb_user: (or other protocol depending on model) |
| 16:33.28 | *** join/#asterisk JCux (i=JCux@200.84.205.238) |
| 16:33.44 | deb_user | yeah...i see that |
| 16:33.46 | deb_user | pretty cheap too |
| 16:33.47 | JCux | hi men |
| 16:33.59 | [TK]D-Fender | deb_user: < $70 for 2 phones. |
| 16:34.03 | JCux | i need a please |
| 16:34.11 | [TK]D-Fender | deb_user: A lot cheaper and functional per-port than PCI. |
| 16:34.27 | deb_user | fender: I've get a three handset cordless |
| 16:34.37 | deb_user | so I could use one plug for three phones |
| 16:34.54 | JCux | what connect 2 clients softphone(xlite) with asterisk@home |
| 16:34.56 | JCux | ? |
| 16:34.58 | [TK]D-Fender | very unfortuante, because they share a single line. |
| 16:35.15 | [TK]D-Fender | JCux: Please read the channel topic.... |
| 16:36.31 | JCux | D-Fender, im sorry, you are know a channel to know about |
| 16:37.07 | [TK]D-Fender | JCux: Topic Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.12.1, Zaptel 1.2.9.1 released! (September 12, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx -=- http://pastebin.ca/ for showing others large amounts of text |
| 16:37.46 | [TK]D-Fender | JCux: Go to #freepbx for support on A@H |
| 16:38.41 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 16:39.07 | JCux | thanks D-Fender, excuse me |
| 16:39.15 | *** part/#asterisk JCux (i=JCux@200.84.205.238) |
| 16:39.57 | markit | hi, can someone tell me how can I quickly get the sounds.txt from the svn of asterisk? I need the one that will soon become in astersik 1.4 |
| 16:40.33 | awannabe | its fairly easy to setup a branch office with a couple FXO ports for backup if the main site/PRI is down, right? |
| 16:42.31 | *** part/#asterisk jaike (i=jaike@210.5.117.158) |
| 16:42.44 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 16:43.00 | inv_arp[work] | does asterisk store outbound calls anywhere by default? or do i set that up? |
| 16:43.17 | SwK[Work] | you mean like CDRs? |
| 16:43.30 | inv_arp[work] | yea numbers |
| 16:43.43 | inv_arp[work] | i dont need to indepth just who I called |
| 16:44.08 | [TK]D-Fender | awannabe: Can be. |
| 16:44.12 | TripleFFFF | [agi_dnid] => unknown |
| 16:44.17 | TripleFFFF | ok i figure that that bug alone cost me arount 5k |
| 16:44.20 | trevarthan | hey guys, When I install my digium T1 card into my machine my voicemail stops working. In the CLI I see it play the voicemail audio, but no sound comes out of my sip phones for the voicemail (ringing and sip-sip works fine). the digium card is configured (channels show up in `zap show channels`), but not plugged into a PRI. When I remove the digium card, voicemail works again. Any ideas? |
| 16:44.22 | TripleFFFF | been routing calls to wrong providers since its empty... |
| 16:44.41 | TripleFFFF | and you know waht .. i think its ALWAYS EMPTY.. when its a call from an IAX trunk |
| 16:44.45 | TripleFFFF | nice one |
| 16:45.09 | TripleFFFF | i guess tomorow will be 1.2.12.2 |
| 16:45.51 | inv_arp[work] | SwK[Work]: so i guess i need to enable somethin for CDR |
| 16:45.52 | sevard | Question guys, I had a PRI card and was getting a lot of echo on local calls, so I went ahead and traded in my digium PRI card for one with a hardware echo canceller onboard and I'm _still_ getting echo on local calls, I called digium yesterday and they told me to recompile the zaptel drivers without 'disable echo can with 2100hz is detected' so I did that, but still echo. |
| 16:46.05 | sevard | They're all in a meeting right now at digium, anyone have any idea what is up with this? |
| 16:46.38 | awannabe | [TK]D-Fender, it just dialplan setup right? if the PRI cant make the call then push out the FXO im assumming |
| 16:47.00 | SwK[Work] | inv_arp[work]: check out voip-info.org for all the options on CDRs... theres CVS, mysql, postgresql etc options |
| 16:47.36 | inv_arp[work] | alright... ill use sqlite |
| 16:47.40 | [TK]D-Fender | awannabe: Correct. |
| 16:47.46 | Juggie | TripleFFFF, hasnt dnid allways been empty w/ iax? |
| 16:48.14 | inv_arp[work] | stupid ez to set up |
| 16:48.29 | TripleFFFF | no idea |
| 16:48.37 | TripleFFFF | still sucks |
| 16:48.39 | Juggie | i'm pretty sure it is |
| 16:48.43 | Juggie | because its a Zap variable |
| 16:48.48 | TripleFFFF | that the prob with asterisk |
| 16:48.52 | Juggie | you should be using exten |
| 16:49.04 | TripleFFFF | i need to actual y TEST every function to see if broken or not on EVERY release |
| 16:49.14 | Juggie | but it hasnt changed |
| 16:49.17 | TripleFFFF | try that.. make asterisk realtime via mysql |
| 16:49.26 | Juggie | i know for sure on like 1.2.9 it was the same dnid is empty on iax |
| 16:49.27 | TripleFFFF | then try a mysql command from the mysql dialp[lanb |
| 16:49.34 | TripleFFFF | as in |
| 16:49.45 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:49.47 | *** join/#asterisk angom (n=angom@red-corp-200.79.129.196.telnor.net) |
| 16:49.58 | TripleFFFF | MYSQL(Connect connid blah blah lbah) |
| 16:50.02 | TripleFFFF | from a mysql table |
| 16:50.05 | TripleFFFF | wont work |
| 16:50.08 | Juggie | yes |
| 16:50.14 | TripleFFFF | mysql module app need to be called static not from realtime |
| 16:50.27 | Juggie | because mysql realtime has NOTHING to do with mysql via dialplan |
| 16:50.32 | awannabe | [TK]D-Fender, cool, thanks for the info! |
| 16:50.52 | Juggie | they are two totally seperate things |
| 16:51.18 | TripleFFFF | yes |
| 16:51.23 | TripleFFFF | but one doest work in other |
| 16:51.35 | Juggie | what? |
| 16:52.08 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 16:52.37 | Juggie | i have no idea what you mean. |
| 16:53.23 | trevarthan | hey guys, When I install my digium T1 card into my machine my voicemail stops working. In the CLI I see it play the voicemail audio, but no sound comes out of my sip phones for the voicemail (ringing and sip-sip works fine). the digium card is configured (channels show up in `zap show channels`), but not plugged into a PRI. When I remove the digium card, voicemail works again. Any ideas? |
| 16:53.53 | trevarthan | (I swear I'll shut up now. I'm just not sure anyone has noticed my post) |
| 16:54.36 | Juggie | never heard of this one before, do you have libpri installed? |
| 16:57.14 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 16:58.09 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 16:58.33 | jbalcomb | Juggie: I've had the same problem. If you just unload the module for the card it will work fine. |
| 16:58.53 | jbalcomb | trevarthan: see above |
| 16:59.57 | jbalcomb | though I assume the agreement here is that it should work with the card loaded |
| 17:00.13 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:00.15 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:00.20 | Juggie | jbalcomb, first i've ever heard of this one, is this problem with a specific version of *? |
| 17:00.51 | jbalcomb | Juggie: I am not sure. I'm on 1.2.5 and it happens like that for me. |
| 17:01.51 | Juggie | install libpri, make clean on zaptel and recompile that, then make clean * and recompile that. |
| 17:01.56 | Juggie | let me know what happens. |
| 17:02.03 | Juggie | (in that order) |
| 17:03.22 | trevarthan | jbalcomb: any solutions besides unloading the module? |
| 17:03.47 | trevarthan | jbalcomb: does it work when the card is actually plugged into a PRI? |
| 17:04.05 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
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| 17:04.15 | *** join/#asterisk sbma44 (n=tomlee@dsl092-174-002.wdc2.dsl.speakeasy.net) |
| 17:05.37 | CunningPike | sevard: Is your echo always far end echo, and always on PSTN calls? |
| 17:05.51 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 17:06.02 | *** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 17:06.34 | jbalcomb | trevarthan: i do not know of any other solution. it does work fine plugged into the PRI. |
| 17:06.43 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 17:07.09 | jbalcomb | trevarthan: My best guess would be something to do with timing and maybe running the zaptel dummy driver would help. |
| 17:07.34 | pollohawk | Does the ChanIsAvail function check to see if the channels are available in the order you specify them as arguments? |
| 17:08.39 | trevarthan | jbalcomb: ok. As long as it works plugged in I'm happy. The pri comes in a few days. I can always remove the card until then. |
| 17:08.49 | trevarthan | thank you! |
| 17:09.40 | *** join/#asterisk vosque (i=hpkdjg24@69.50.222.162) |
| 17:12.45 | sbma44 | hi folks. I have a problem that I haven't been able to make any headway on. I've got a server and a SIP trunk. I can connect to the server using xlite (from outside its local network) and everything seems to work fine. However, I can't make outbound or inbound calls from/to the trunk. When I make an inbound call, I can see the dialplan executing in the asterisk console, but there's no audio -- the line keeps ringing until the v |
| 17:12.52 | sbma44 | can anyone suggest to me how I might go about debugging this? |
| 17:13.14 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 17:13.33 | Juggie | is * behind a firewall? |
| 17:14.46 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 17:15.49 | trevarthan | sbma44: does it look like this? |
| 17:15.50 | trevarthan | xlite -> NAT -> internet -> NAT -> asterisk |
| 17:16.06 | sbma44 | xlite -> NAT, yes |
| 17:16.21 | sbma44 | asterisk is in a colo facility, behind a firewall that I'm assured doesn't involve NAT |
| 17:16.26 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 17:16.35 | sbma44 | but does do address translation from a nonroutable IP pool to a public IP |
| 17:16.39 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 17:16.57 | trevarthan | sbma44: Sounds like nat to me. |
| 17:17.18 | trevarthan | sbma44: You need to set the 'externip' and 'localnet' options in sip.conf. |
| 17:17.23 | sbma44 | I'm afraid I'm not qualified to judge, but our sysadmin was pretty insistent on this point |
| 17:18.08 | sbma44 | trevarthan: I have. externip=69.25.xxx.xxx |
| 17:18.14 | Juggie | sbma44, the linux machine is assigned a internal ip? |
| 17:18.17 | sbma44 | localnet=10.0.0.0/255.0.0.0 |
| 17:18.27 | trevarthan | sbma44: also, make sure xlite has a working STUN server configured. I use stun.fwdnet.net |
| 17:18.44 | sbma44 | traverthan: even though xlite is working? |
| 17:19.15 | trevarthan | sbma44: If your sip client (xlite) is behind a firewall, use STUN. It's a requirement, in my experience. |
| 17:19.16 | sbma44 | juggie: yes, it's got an internal IP |
| 17:19.27 | Juggie | sbma44, sounds like nat to me then :) |
| 17:19.49 | sbma44 | ok. well, xlite isn't the problem. it works fine for accessing internal extensions. I just used it because someone suggested I needed to get some client-side SIP logs |
| 17:19.51 | trevarthan | sbma44: I've got "xlite -> NAT -> internet -> NAT -> asterisk" working here, so I'm not just spouting air. |
| 17:20.09 | Juggie | be sure to set nat=yes in your sip.conf general section, and also externip |
| 17:20.18 | Juggie | trevarthan, stun is not required to make that work. |
| 17:20.26 | trevarthan | sbma44: oh yeah, use nat=yes too. |
| 17:20.54 | trevarthan | Juggie: I don't believe you. :) But I'll test it out tonight. |
| 17:21.10 | Juggie | trevarthan, the wonders of symetric nat my friend. |
| 17:21.34 | sbma44 | and canreinvite=no on the trunk registration? |
| 17:21.43 | sbma44 | or does it not matter? |
| 17:21.45 | *** join/#asterisk rokerr (n=rowan@38.99.162.187) |
| 17:22.03 | trevarthan | juggie: doesn't that depend on whether or not you're using a symetric NAT firewall? I usually use full cone types. |
| 17:22.04 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 17:22.08 | trevarthan | sbma44: I don't think it matters. |
| 17:22.18 | Juggie | set it =no |
| 17:23.04 | sbma44 | okay, did all that |
| 17:23.06 | sbma44 | same behavior |
| 17:23.17 | Juggie | is sound going in either direction? |
| 17:23.18 | sbma44 | I see dialplan execution (verbosity 3), but it just rings & rings |
| 17:23.38 | sbma44 | no, neither. just ringing until the sip vendor's voicemail picks up |
| 17:23.38 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 17:23.42 | trevarthan | sbma44: so maybe your extensions are messed up? |
| 17:24.01 | sbma44 | when I was in here before, some folks looked at my sip log and said that an ACK was failing to make it through to the vendor |
| 17:24.15 | trevarthan | sbma44: I can't really help with the extensions. Just the connectivity and such. |
| 17:24.25 | sbma44 | I don't think they are. I'm just sending it to the asterisk demo extension that ships w/ it. |
| 17:24.54 | trevarthan | sbma44: Can you record a voicemail message and hear the audio when you listen to it? |
| 17:24.54 | Juggie | sbma44, has your * box sucefessuly registered with the provider? |
| 17:25.34 | sbma44 | if I send inbound calls to an extension confirmed to work w/ xlite it does the same |
| 17:25.55 | sbma44 | lemme try recording a vm |
| 17:25.57 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 17:26.06 | Juggie | i'm confused on what you haev working and not working here |
| 17:26.10 | Juggie | so lets start from the beginning |
| 17:26.13 | sbma44 | ok |
| 17:26.24 | Juggie | you are registerd w/ an external sip provider correct? |
| 17:26.48 | Juggie | you are going to haev to answer quickly or i will get bored |
| 17:26.53 | sbma44 | sip show peers? |
| 17:27.00 | sbma44 | Name/username Host Dyn Nat ACL Port Status |
| 17:27.00 | sbma44 | test/test (Unspecified) D N 0 Unmonitored |
| 17:27.00 | sbma44 | sip.broadvoice.com/202517 147.135.20.128 N 5060 Unmonitored |
| 17:27.04 | sbma44 | or does that not indicate if it's registering correctly? |
| 17:27.13 | Juggie | thats for incomming calls |
| 17:27.25 | sbma44 | sorry, wrong one |
| 17:27.33 | sbma44 | sip show registry indicates that yes, I am registered |
| 17:27.45 | Juggie | which reminds me |
| 17:27.46 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 17:27.49 | Juggie | set qualify=yes |
| 17:27.52 | Juggie | for your sip peers |
| 17:27.59 | sbma44 | ok, one sec |
| 17:28.00 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 17:28.44 | sbma44 | alright, qualify=yes, hard * restart, still shows registered |
| 17:28.55 | Juggie | now do sip show peers |
| 17:29.13 | sbma44 | status = OK (94 ms) |
| 17:29.21 | Juggie | ok, good. |
| 17:29.40 | [TK]D-Fender | sbma44: Pastebin your sip.conf masking only the passwords please. |
| 17:29.42 | [TK]D-Fender | ~pb |
| 17:29.43 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 17:29.48 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 17:29.51 | Juggie | ok, so this sip from broadvoice i assume provides pstn access |
| 17:29.57 | Juggie | can you make any outbound calls? |
| 17:29.59 | tRSS | has anyone successfully run asterisk on FC5? |
| 17:30.04 | sbma44 | no, can't make outbound either |
| 17:30.12 | sbma44 | one moment and I'll have a pastebin url for you |
| 17:30.16 | Juggie | k |
| 17:30.29 | Juggie | i suspect you just dont know whatr your doing and you have it incorrectally configured :) |
| 17:30.51 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:31.25 | tRSS | i am having trouble compiling zaptel on FC5? I have checked the voip-info wiki about zaptel/FC5 and digium site, but I can't get it to run. help would be appreciated. |
| 17:32.07 | Juggie | pastebin your compile error (including the entire compile log) |
| 17:32.09 | Juggie | ~pb |
| 17:32.11 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 17:32.15 | [TK]D-Fender | tRSS: Yeah theres a bunch of bug I wasn't able to overcome reasonably quickly so I ditched FC5 for CentOS |
| 17:32.21 | sbma44 | juggie: that's certainly possible, but I'm using a stock config provided by broadvoice, and confirmed it to work on my * box at home (running off a simple dsl connection) |
| 17:32.47 | tRSS | sure, brb after pastebin'ing |
| 17:32.52 | Juggie | sbma44, post your sip.conf and extensions.conf |
| 17:32.59 | Juggie | so we can see what you are doing |
| 17:33.02 | sbma44 | sip.conf: http://pastebin.ca/170549 |
| 17:33.02 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 17:33.20 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:33.22 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:33.47 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 17:33.48 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 17:34.33 | sbma44 | extensions.conf: http://pastebin.ca/170552 |
| 17:34.44 | sbma44 | (almost entirely stock, as you can see) |
| 17:35.19 | [TK]D-Fender | sbma44: I suggest moving your register statement blow all of your settings in [general] and add "canreinvite=no" into [general]. In so doing please remove EVERYTHING that is commented out. |
| 17:36.29 | sbma44 | d-fender: will do |
| 17:36.30 | tRSS | here you go guys: http://pastebin.ca/170553 |
| 17:36.32 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 17:37.11 | Juggie | i would suggest getting outbound calling working first. |
| 17:37.21 | Juggie | remove all the uneeded crap from your extensions.conf |
| 17:37.30 | Juggie | sip.conf is ok to leave the commented stuff as its good for a reference |
| 17:37.40 | Juggie | but extensions.conf delete all the example shit |
| 17:38.06 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 17:38.23 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 17:39.02 | Juggie | first things first |
| 17:39.20 | Juggie | your xten cannot dial out because he has not been allowed to do so |
| 17:39.42 | Juggie | xlite is confined to [test] in which there is only one extension |
| 17:39.56 | sbma44 | actually, I was testing outbound calling using a .call file |
| 17:39.57 | sbma44 | not xlite |
| 17:40.18 | sbma44 | and sending the call into the sip.broadvoice.com context |
| 17:40.48 | Juggie | the broadvoice context still doesnt supply outbound dialing |
| 17:41.55 | tRSS | Juggle: did you get a chance to look at the pastebin url I pasted above? I see you are pretty busy. |
| 17:42.13 | sbma44 | juggie: okay. what do I need to do to make that work? I could've sworn I had an almost identical config on my other machine, where broadvoice outbound dialing is working great. |
| 17:42.35 | sbma44 | extension change, or a change in permissioning in the sip.conf entry for bv? |
| 17:42.48 | Juggie | why aer you showing me the output from a make clean |
| 17:42.57 | Juggie | show me the output from 'make' |
| 17:43.18 | Juggie | but, it looks like you dont have kernel-devel installed |
| 17:43.24 | tRSS | Juggle: You do not appear to have the sources for the 2.6.15-1.2054_FC5smp kernel installed. |
| 17:43.31 | Juggie | well. |
| 17:43.36 | Juggie | thats pretty obvious no? |
| 17:43.46 | tRSS | i have installed kernel-devel (using rpms and yum) |
| 17:43.58 | Juggie | but not kernel-smp-devel |
| 17:44.24 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:44.33 | tRSS | ooh , I have missed kernel-smp-devel. |
| 17:45.05 | tRSS | Juggle: doing that now. thanks for pointing that out |
| 17:45.33 | Juggie | sbma44, look here, http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice speficially at the extensions.conf example. |
| 17:49.11 | Juggie | also, what you pass in as 9999 has to exist in your dialplan in the context=broadvoice (but it doesnt) |
| 17:49.45 | Juggie | i suggest you look @ that link |
| 17:50.17 | Juggie | another good doc is here, http://edvina.net/broadvoice/ just ignore the part about requiring a patch. |
| 17:50.29 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:52.11 | sbma44 | juggie: thanks. making the changes now |
| 17:53.46 | harryvv | I guess most people here are listening in on von |
| 17:55.26 | SplasPood | someone is telling me polycom china told them that there are no model 501s available, only 500s |
| 17:56.11 | [TK]D-Fender | SplasPood: Do not accept the 300/500/600 over the 301/501/601 |
| 17:56.22 | SplasPood | [TK]D-Fender: I know, hence the .. wtf?!? |
| 17:56.41 | SplasPood | I need to buy some phones in china |
| 17:56.44 | SplasPood | shanghai, specificallhy |
| 17:56.49 | [TK]D-Fender | SplasPood: You want the extra memory and sidecar support (though it looks like the sidecar may become SIP independent now) |
| 17:57.10 | SplasPood | they claimed that the 601 was avail |
| 17:57.11 | SplasPood | but not 501 |
| 17:57.17 | SplasPood | due to some.. China Telecom not approving |
| 17:57.21 | SplasPood | or some other BS line |
| 17:57.37 | [TK]D-Fender | I'm witing for the IP 650 to hit distribution.... |
| 17:57.46 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 17:58.43 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 18:01.30 | [TK]D-Fender | I love how so many of the pictures Polycom has in their literature has people "talking" on the phone but clearly the screen is on *IDLE*. dumb |
| 18:01.45 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:03.49 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:09.03 | momelod | hey does anyone have asterisk-stat they can send me.. i cant seem to connect to the webpage |
| 18:09.15 | *** part/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 18:10.51 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:11.41 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 18:12.11 | _deg_ | is there someone that could help me on Asterisk Realtime Static + ODBC + Postgres 7.4 ? |
| 18:12.26 | _deg_ | Asterisk 1.2 btw |
| 18:17.49 | sbma44 | thanks for the help everybody, but I'm afraid I'm still stuck at the same spot I started. dialplan executes, but there's no sound coming from my asterisk machine. |
| 18:18.31 | sbma44 | if I hit an extension that consists of SayAlpha(abc) with a softphone registered to the server, it works fine. if I send incoming broadvoice traffic there, I can see each letter getting spoken in the console, but only get ringing on the phone. |
| 18:19.06 | [TK]D-Fender | sbma44: New sip.conf pastebin please and double check your externIP. |
| 18:19.48 | sbma44 | fender: ok. I still have comments in sip.conf -- I forget if you asked me to take them out or not. preference? |
| 18:20.10 | *** join/#asterisk Adam06 (n=Adam@ip67-95-13-58.z13-95-67.customer.algx.net) |
| 18:20.18 | Adam06 | Hello everyone... anyone around? |
| 18:22.49 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 18:24.30 | wunderkin | ok, i have a question, i have a poly430 and a bt101, when i call into a queue with no members and no moh, the bt101 will stay connected to the queue fine, but on the poly it disconnects after 60 seconds, even though there is rtp, the poly sends a cancel request.. why? |
| 18:24.58 | wunderkin | oooh.. maybe because the call isnt answered? |
| 18:25.07 | [TK]D-Fender | sbma44: Remove all comments permanently |
| 18:25.37 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:26.58 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-a0f562d5f12e43f2) |
| 18:28.33 | sbma44 | [TK]D-Fender: here's my current sip.conf |
| 18:28.34 | sbma44 | http://pastebin.ca/170636 |
| 18:29.44 | *** join/#asterisk TimothyP (n=timothy@83-217-93-182.adsl.realdsl.be) |
| 18:30.12 | syzygyBSD | crap |
| 18:30.32 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.68) |
| 18:30.33 | TimothyP | Hi, I have added an extention [1001] and username=timothy now what should I use to log in ?? because if I use 1001 asterisk says to many retries and if I specify the name it says registration from timothy@..... failed for <ipadress> |
| 18:30.48 | syzygyBSD | I just checked and found the IRQ for a clients ZAP card on one server is shared by IDE and USB |
| 18:31.07 | syzygyBSD | thats bad right? |
| 18:31.38 | [TK]D-Fender | sbma44: Much better... now move your register to below "nat=yes". Next : what ports do yuo have forwarded to your * box? (precisely) |
| 18:31.50 | [TK]D-Fender | syzygyBSD: Very bad combo |
| 18:32.11 | syzygyBSD | ya... I forgot how to update that too, off to search the forums |
| 18:32.52 | [TK]D-Fender | TimothyP: Ditch the username field, and register as 1001 |
| 18:32.59 | TimothyP | ok I'll do that |
| 18:33.16 | TimothyP | also, if I configured just 2 extentions that should be enough to test if my server is working right? |
| 18:33.19 | sbma44 | [TK]D-Fender: will move the register momentarily. As for ports: 5060 UDP/TCP is forwarded. 10000-10100 UDP are forwarded as well, and that range is defined w/ rtpstart/rtpend in rtp.conf |
| 18:33.36 | [TK]D-Fender | TimothyP: Thats a good test as you can test audio more with 2 extens. |
| 18:33.48 | momelod | hey does anyone have asterisk-stat they can send me.. i cant seem to connect to the webpage |
| 18:33.53 | TimothyP | then I have one more question which I could not find in the manual |
| 18:34.05 | [TK]D-Fender | sbma44: Aid for 10000-20000 UDP for RTP, and 5060-5080 UDP for SIP (just safety range) |
| 18:34.43 | [TK]D-Fender | TimothyP: There is no "manual" ;) But ok.... TFOT is a good "guide" mind you./ |
| 18:34.46 | sbma44 | [TK]D-Fender: if possible, I'd prefer to avoid opening more, since I have to go through our NOC guys for that and it could take a while. |
| 18:34.47 | TimothyP | my asterisk server is in the network 10.0.0.0/8 in that network there's a router whicch uses NAT , it createas a lan with 192.168.0.0/24 as a network for my computers, my computers have softphones , can they still use the asterisk server even though they are in another network ? |
| 18:34.56 | sbma44 | I can provide sip debug confirming that those ports are in use, though, if that would be helpful |
| 18:35.01 | TimothyP | [TK]D-Fender, I found a Asterisk handbook |
| 18:35.02 | TimothyP | in PDF |
| 18:35.06 | sbma44 | if abs. necessary I will try to get those opened |
| 18:35.24 | [TK]D-Fender | TimothyP: Yes you can have * span subnets, etc. |
| 18:35.35 | [TK]D-Fender | TimothyP: ... |
| 18:35.37 | [TK]D-Fender | ~book |
| 18:35.46 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:35.46 | TimothyP | anything I need to keep in mind for that? |
| 18:35.58 | TimothyP | there are actually a few good books |
| 18:36.09 | sbma44 | does sip require more than 5060 for its control channel? |
| 18:36.17 | TimothyP | The on from O'Reilly and then the one from those with an orange and black cover |
| 18:36.21 | TimothyP | and then there's the PDF from the site |
| 18:36.24 | [TK]D-Fender | TimothyP: Make sure to set "nat=yes", externip & localnet under [general] in sip.conf |
| 18:37.33 | TimothyP | hang on |
| 18:37.39 | TimothyP | just to make sure I'll do something first |
| 18:37.43 | TimothyP | don't run away :p |
| 18:37.46 | TimothyP | ..please :p |
| 18:38.23 | Adam06 | herm... anyone with alot of knowledge of the Asterisk PBX here? |
| 18:38.33 | jbroome | not in #asterisk, no. |
| 18:38.49 | [TK]D-Fender | "nobody here but us chickens" |
| 18:39.04 | Adam06 | of most of the #php and #programming channels I've been to noone knows anything, just trying to find people who do. :/ |
| 18:39.15 | Adam06 | IE: me for example. |
| 18:39.32 | Adam06 | I know little of Asterisk...... looking for someone who maybe knows some stuff |
| 18:40.22 | aydiosmio | well |
| 18:40.25 | aydiosmio | you're in the right place. |
| 18:40.35 | Adam06 | good ^.^ |
| 18:40.52 | Adam06 | I've been sitting on Asterisk's Wiki for about hte last two hours. |
| 18:42.16 | momelod | is this url working for anyone: http://www.areski.net/ |
| 18:42.22 | momelod | i wanna dl asterisk-stat |
| 18:42.50 | syzygyBSD | how is it possible to find out what Zap card I have in a machine? |
| 18:43.05 | [TK]D-Fender | momelod: http://www.areski.net/asterisk-stat-v2/asterisk-stat-v2_0_1.tar.gz |
| 18:43.09 | momelod | syzygyBSD: use dmesg |
| 18:43.36 | syzygyBSD | hmm.. makes sense |
| 18:43.40 | momelod | [TK]D-Fender: yeah.. for whatever reason i cant connect to that page.. not from work or from home |
| 18:43.59 | aydiosmio | I think Adam06 has short-circuited |
| 18:44.20 | aydiosmio | momelod: still? |
| 18:44.23 | momelod | [TK]D-Fender: could u possibly dl it and then dcc it to me? |
| 18:44.26 | momelod | yeah.. |
| 18:44.26 | aydiosmio | didn't we put this to bed yesterday? |
| 18:44.30 | aydiosmio | hahahaha |
| 18:44.32 | momelod | lol, nope |
| 18:44.35 | momelod | its weird tho |
| 18:44.44 | momelod | i tried from w/ the same resaul |
| 18:44.44 | momelod | y |
| 18:44.48 | aydiosmio | momelod: use nyud |
| 18:44.53 | momelod | nyud? |
| 18:44.54 | *** join/#asterisk jacobp (n=jacobp@dsl-syl-66-242-57-40.ambtel.net) |
| 18:45.05 | momelod | is that a proxy of some kind? |
| 18:45.20 | aydiosmio | of some kind |
| 18:45.24 | aydiosmio | try this |
| 18:45.37 | aydiosmio | http://www.areski.net.nyud.net:8080/asterisk-stat-v2/asterisk-stat-v2_0_1.tar.gz |
| 18:46.13 | aydiosmio | that should download the file through the nyud chache server |
| 18:46.31 | jacobp | when i set dtmf payload to 0 on my Snom sip phones I can use dtmf on external lines, but can't for voicemail internal. if I set it to anything else voicemail works but not dtmf on external lines. Anybody seen this? |
| 18:46.38 | momelod | aydiosmio yes that works |
| 18:46.39 | momelod | thank you |
| 18:46.45 | aydiosmio | np |
| 18:46.56 | aydiosmio | glad it worked |
| 18:46.59 | TimothyP | OH MY GOD , is there any program that is more annoying to work with than DIA ? |
| 18:47.00 | aydiosmio | it's a handy tool |
| 18:47.03 | momelod | not as glad as i am :D |
| 18:47.07 | aydiosmio | OH MY GOD YES THERE IS |
| 18:47.41 | TimothyP | I've been trying for 5 minuts now to connect to objects |
| 18:47.44 | TimothyP | keeps jumping next to it |
| 18:48.04 | aydiosmio | maybe you should avoid askign rhetorical questions |
| 18:48.06 | TimothyP | after such a long time one would expect it to be quite stable |
| 18:48.09 | aydiosmio | that might solve your problem |
| 18:49.05 | aydiosmio | dia as in the gnome drawing program? |
| 18:49.37 | TimothyP | yes |
| 18:49.38 | TimothyP | omg I'm going nuts |
| 18:49.49 | TimothyP | all I want to do is draw a simple diagram to show you guys, if only I had visio :s |
| 18:50.06 | aydiosmio | what's it do? would gimp or inkscape be a better alternative? |
| 18:50.08 | TimothyP | dia is probably the worst piece of software I've ever had to work with, and I just love ubuntu |
| 18:50.17 | aydiosmio | ooooh |
| 18:50.19 | TimothyP | it's a diagram modeling tool |
| 18:50.25 | TimothyP | like Visio |
| 18:50.29 | TimothyP | and kivio |
| 18:50.33 | aydiosmio | right |
| 18:50.39 | TimothyP | Visio is the one application linux doesn't have a good alternative for |
| 18:50.40 | aydiosmio | gnome.org describes it as a drawing tool |
| 18:50.46 | aydiosmio | which is wildly ambiguous |
| 18:51.00 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 18:51.11 | rokerr | It was good for 1998... but doesn't seem to have made a lot of progress since :) |
| 18:51.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:51.12 | jacobp | anyone familiar with Snom sip phones? |
| 18:51.58 | syzygyBSD | lol, telemarketers are funny |
| 18:52.37 | aydiosmio | TimothyP: try Tgif, tkined, and xfig |
| 18:52.48 | TimothyP | ok let me apt get |
| 18:52.59 | aydiosmio | tkined looks the best for network diagramming |
| 18:53.26 | benjk | How about the drawing tool in OpenOffice? |
| 18:53.29 | aydiosmio | reviews of linux diagramming softwares |
| 18:53.30 | aydiosmio | http://www.linuxdevcenter.com/pub/a/linux/2001/02/15/LinuxAdmin.html |
| 18:53.46 | aydiosmio | benjk: not any better than gimp. |
| 18:53.51 | benjk | ok |
| 18:54.11 | TimothyP | The gimp rules for foto editing and stuff but it has a different purpose than Dia or Inkscape |
| 18:54.15 | benjk | even the one for the presenations? the powerpoint counterpart? |
| 18:54.18 | TimothyP | using the right tool for the right job is very difficult sometimes |
| 18:54.34 | aydiosmio | I'm using inkscape in windows |
| 18:54.38 | aydiosmio | it works really well with my tablet |
| 18:55.02 | TimothyP | sweet :) |
| 18:55.16 | TimothyP | repo somewhere for tkined? |
| 18:55.34 | aydiosmio | next page |
| 18:56.59 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 18:58.03 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 19:01.10 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:01.19 | TimothyP | tkined looks really cool but of course it's a deps-disaster |
| 19:01.35 | TimothyP | configure: error: tclConfig.sh not found - use the --with-tcl-config option :p |
| 19:01.46 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 19:03.20 | sbma44 | ok. got a ticket in to have additional ports opened up. but I also noticed this showing up in my console: WARNING[7040]: file.c:587 ast_readaudio_callback: Failed to write frame |
| 19:03.45 | sbma44 | which seems like it could account for the lack of audio. I'm wondering if this thread could provide a plausible explanation for what's going on: http://www.voipuser.org/forum_topic_3921.html |
| 19:04.14 | sbma44 | I've asked the box to be DMZed for traffic coming from the broadvoice proxy that I'm using. Hopefully that will resolve the issue. |
| 19:05.01 | [TK]D-Fender | sbma44: Some NAT devices REALLY don't play nicely, formost is the Cisco PIX. |
| 19:05.49 | *** join/#asterisk PBXtech (n=dburr@70.89.247.188) |
| 19:05.58 | sbma44 | hm. well, if I end up having to leave this datacenter for this application, does anyone have suggestions for asterisk-friendly hosts who can provide quick turnaround? |
| 19:06.03 | PBXtech | anyone know the covad password to unlock a cisco phone? |
| 19:06.27 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 19:06.57 | aydiosmio | cisco phone or cisco ata? |
| 19:07.31 | PBXtech | 7960 |
| 19:08.03 | aydiosmio | and covad provided you with this phone? |
| 19:08.09 | [TK]D-Fender | sbma44: Aside from the obvious NAT issues, what other difficulties are you having? |
| 19:08.22 | syzygyBSD | so if i were using chanspy and someone else way using ZapScan should I be able to hear what they are saying? |
| 19:08.25 | PBXtech | bough and paid for |
| 19:08.39 | PBXtech | want to put it on asterisk |
| 19:08.57 | aydiosmio | heh |
| 19:09.10 | aydiosmio | Vonage will unlock your ATA if you buy the device |
| 19:09.24 | aydiosmio | I can't find anythign on Covad |
| 19:09.39 | sbma44 | [TK]D-Fender: none, really. I work for a political consulting company that usually works in the web realm. We're doing a *-based advocacy app and have all the pieces running fine in a proof-of-conceptish way off of my own hobbyist * box. Now we're trying to get the thing ready for production. |
| 19:09.41 | syzygyBSD | lol.. it makes so much more sense why i hear all the background noise |
| 19:09.54 | sbma44 | and have a hell of a time getting * working within our server farm, as you can see |
| 19:10.46 | *** join/#asterisk NebulousNL_ (n=a@office.telecom.tno.nl) |
| 19:10.52 | [TK]D-Fender | sbma44: So BV may work fine. Another decent (perhaps better) option might be http://connect.voicepulse.com |
| 19:12.13 | sbma44 | well, we're not trying to use BV for production. this was originally going to just be a dev environment. now it's turned into a white whale/SIP proof of concept. |
| 19:12.28 | sbma44 | actually, we use voicepulse for our office PBX. But they won't work for this; the rates they charge to establish simultaneous channels are too high |
| 19:12.32 | stephane_ | soir |
| 19:12.34 | sbma44 | we're shooting for ~100 simultaneous |
| 19:13.14 | sbma44 | right now we're looking at sixtel/vitelity, since they do prepaid and have a pricing model that's appropriate to scaling |
| 19:13.21 | sbma44 | but have also been talking to the bandwidth.com people |
| 19:13.31 | sbma44 | don't think we can deliver the traffic necessary to justify that though |
| 19:15.04 | sbma44 | anyone have any experience w/ vitelity? |
| 19:15.05 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.68) |
| 19:15.30 | [TK]D-Fender | stephane_: jour :) |
| 19:19.39 | *** part/#asterisk jacobp (n=jacobp@dsl-syl-66-242-57-40.ambtel.net) |
| 19:23.48 | aydiosmio | PBXtech: http://www.velocityreviews.com/forums/t34172-ot-cisco-ip-phone-7960-password-recovery.html |
| 19:24.04 | aydiosmio | does this help? |
| 19:26.52 | *** join/#asterisk jmsjms (n=jms@82.71.217.13) |
| 19:27.57 | jmsjms | Hello, could somebody please help? I've spent ages on what I'm sure is a very simple problem |
| 19:28.37 | [TK]D-Fender | jmsjms: Just ask |
| 19:28.45 | PBXtech | thx aydiosmio |
| 19:28.48 | jmsjms | I want to dial a number (via my iax2 provider) from my call menu, then continue in the call menu after the dialed number hangs up. Right now, when the dialed number hangs up, so does the caller |
| 19:29.07 | *** join/#asterisk clive- (n=pirch@dsl-145-56-115.telkomadsl.co.za) |
| 19:29.18 | aydiosmio | np |
| 19:29.22 | jmsjms | the line in my extensions.conf is: exten => s,1,Dial(IAX2/james1&${GRADWELL}/07711111111,10,A(tt-monkeysintro)m|g) |
| 19:29.47 | jmsjms | I've also tried exten => s,1,Dial(IAX2/james1&${GRADWELL}/07711111111,10,A(tt-monkeysintro)mg) |
| 19:29.50 | [TK]D-Fender | jmsjms: Look at "show application dial" there is a parameter that allows you to continue processing after the other side ends the call. |
| 19:30.04 | jmsjms | [TK]D-Fender: It seems to be 'g' |
| 19:30.21 | [TK]D-Fender | jmsjms: Indeed |
| 19:30.32 | jmsjms | I've tried g as the only parameter too (taking out the A and the m) |
| 19:32.13 | [TK]D-Fender | jmsjms: Please pastebin the entire context and CLI output of a call placed against it where you had the remote side hang up. |
| 19:32.26 | jmsjms | ok |
| 19:32.29 | *** join/#asterisk TimothyP (n=timothy@83-217-93-182.adsl.realdsl.be) |
| 19:32.45 | TimothyP | Hi, I'm back, ended up using kivio, best I could do. This is my network layout (more or less) http://blogs.homelinux.org/networklayout.jpg |
| 19:32.56 | *** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it) |
| 19:32.58 | jmsjms | [call-taxi] |
| 19:33.01 | TimothyP | as you can see my phones are in different net , can they still comunitcate with Asterisk ? and what should I keep in mind? |
| 19:33.04 | jmsjms | exten => s,1,Dial(IAX2/james1&${GRADWELL}/07765536961,10,A(tt-monkeysintro)m|g) |
| 19:33.04 | jmsjms | exten => s,2,Festival(If you would like to comment on our service\, please press 1,any) |
| 19:33.10 | jmsjms | exten => 1,1,Festival(Pretend voicemail,any) |
| 19:33.10 | jmsjms | exten => 2,1,Dial(IAX2/james1,10,A(tt-monkeysintro)m|g) |
| 19:33.10 | jmsjms | exten t,1,Goto(s,2) |
| 19:33.56 | jmsjms | the CLI output is: |
| 19:33.59 | jmsjms | <PROTECTED> |
| 19:33.59 | jmsjms | <PROTECTED> |
| 19:33.59 | jmsjms | <PROTECTED> |
| 19:34.01 | jmsjms | <PROTECTED> |
| 19:34.14 | aydiosmio | ACK |
| 19:34.23 | jmsjms | <PROTECTED> |
| 19:34.23 | jmsjms | <PROTECTED> |
| 19:34.23 | jmsjms | <PROTECTED> |
| 19:34.23 | jmsjms | <PROTECTED> |
| 19:34.23 | jmsjms | <PROTECTED> |
| 19:34.25 | jmsjms | <PROTECTED> |
| 19:34.29 | [TK]D-Fender | TimothyP: You are asking for trouble in that setup.... too many NAT's and subnets |
| 19:34.38 | [TK]D-Fender | jmsjms: PASTEBIN |
| 19:34.48 | jmsjms | sorry, what is pastebin? |
| 19:34.49 | aydiosmio | ~pb lol |
| 19:34.55 | [TK]D-Fender | ~pb |
| 19:34.56 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 19:34.58 | [TK]D-Fender | jmsjms: Please read the channel topic |
| 19:35.14 | TimothyP | [TK]D-Fender, I know but it's what I have to live with :s |
| 19:35.23 | [TK]D-Fender | TimothyP: HAVE to? |
| 19:35.51 | TimothyP | yes that's the network I get |
| 19:35.54 | TimothyP | not my choise |
| 19:36.05 | trelane` | can anyone think of a reason why voicemail would randomly disappear? |
| 19:36.11 | [TK]D-Fender | TimothyP: NO change possible? Put your * on the 192.168.0.X subnet |
| 19:36.19 | aydiosmio | freepbx is not a pbx |
| 19:36.23 | aydiosmio | your bit is an idiot |
| 19:36.28 | aydiosmio | your bot is an idiot |
| 19:37.20 | aydiosmio | a hater |
| 19:37.20 | aydiosmio | your bot drank the haterade |
| 19:37.40 | aydiosmio | alright |
| 19:37.41 | aydiosmio | I'm done |
| 19:37.52 | TimothyP | really :p |
| 19:38.11 | smackus | i am using chan is available successfully when another extension dials it. but when i have a a queued call, it does not call that part of the script. I understand that the queue is calling the SIP/ device and that is where i have this all set up. What am I doing wrong to make it so that a queued call will not call chanisavailable? http://pastebin.ca/170735 |
| 19:38.12 | [TK]D-Fender | aydiosmio: I loved that little ditty of yours... really... especially the end. |
| 19:38.32 | aydiosmio | yeah, it wasn't as good as I'd hoped it would be |
| 19:38.37 | aydiosmio | they're usually funnier |
| 19:38.43 | rokerr | What's a reasonable piece of hardware to interface a soho asterisk server with normal telephone lines? |
| 19:39.19 | aydiosmio | rokerr: how many lines? |
| 19:39.53 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 19:40.10 | rokerr | just one incoming and maybe 3 or 4 handsets |
| 19:40.27 | rokerr | serious emphasis on the _home_ in soho :) |
| 19:40.32 | aydiosmio | IP handsets or analog handsets? |
| 19:40.47 | aydiosmio | all the analogs on one line? |
| 19:40.55 | rokerr | i'm willing to consider whichever will work better |
| 19:41.04 | rokerr | unfortunately the house isn't wired with ethernet |
| 19:41.06 | [TK]D-Fender | rokerr: Clarify : How many LINES? |
| 19:41.22 | rokerr | 1 |
| 19:41.24 | TimothyP | anyway is it at all possible ? |
| 19:41.24 | [TK]D-Fender | rokerr: As in coming from the telco. |
| 19:41.43 | smackus | do i have to use local channels to make that work . |
| 19:41.44 | [TK]D-Fender | rokerr: only 1? |
| 19:41.47 | aydiosmio | heh |
| 19:41.52 | TimothyP | the thing is I live in a place where there are different networks, and I happen to be in the network behind the dlink |
| 19:42.19 | rokerr | [TK]D-Fender: mmhmm I was hoping to have just one. |
| 19:42.25 | [TK]D-Fender | TimothyP: Why can't you move * onto the non WAN port on it? |
| 19:42.51 | TimothyP | because it's used by phones in the 172.17.0.0/16 network as well |
| 19:43.17 | [TK]D-Fender | rokerr: Ok, take your pick of the X100P, TDM400P, A200, or SPA-3102, I would suggest the SPA personally. |
| 19:43.34 | aydiosmio | rokerr: you can get a proper TDM400P with one FXO and one FXS |
| 19:43.49 | aydiosmio | but you can get an X100 FXO card for $20 |
| 19:43.53 | [TK]D-Fender | TimothyP: Get another NIC then. |
| 19:44.09 | TimothyP | I'm not allowed to put my computer in the 172..... network |
| 19:44.14 | rokerr | [TK]D-Fender, aydiosmio: coo' |
| 19:44.19 | [TK]D-Fender | Avoid all PCI cards for FXS purposes |
| 19:44.21 | TimothyP | I don't make the rules |
| 19:44.43 | [TK]D-Fender | TimothyP: And their rules don't work. Good luck |
| 19:44.48 | jbroome | hahah |
| 19:45.08 | rokerr | aydiosmio: well the price for the tdmp400p isn't too bad.. |
| 19:45.09 | TimothyP | besides what's the point of asterisk if it can only be used in a single subnet ?what if I wanted to allow someone from a remote location to use a softphone and connnect to my server in order to call a normal landline using local fees |
| 19:45.25 | *** join/#asterisk kamileon (n=kamileon@c-71-207-212-67.hsd1.al.comcast.net) |
| 19:45.25 | [TK]D-Fender | TimothyP: You have *2* NAT sides and * can only forge 1 return IP. They change or you're screwed |
| 19:45.27 | aydiosmio | rokerr: wiht those two modules it's $225 |
| 19:45.58 | rokerr | hmm |
| 19:46.07 | jmsjms | right... sorry about the mess before. I've simplifies my extensions.conf and written about my problem at http://hillj.co.uk/asterisk-problem.txt |
| 19:46.08 | TimothyP | doesn't the router (dlink) take care of the required translation? |
| 19:46.20 | *** join/#asterisk daysmen3 (n=primus@host86-143-5-93.range86-143.btcentralplus.com) |
| 19:46.37 | rokerr | aydiosmio: would using something like that potentially scale up to more lines with a beefier card? |
| 19:46.54 | jbroome | i just saw the diagram. holy crap |
| 19:47.03 | jmsjms | the problem is basically that when I call out, I want the original incoming call to stay connected to the * server when the remote end drops out. The documentation for Dial says that the g option should do this, but it isn't working for me |
| 19:47.08 | [TK]D-Fender | rokerr: Just get an SPA-3102. $95 for 1 FXO and 1 FXS |
| 19:47.19 | TimothyP | and does that also mean I can't connect to my asterisk from the internet? |
| 19:47.28 | aydiosmio | rokerr: the TDM400P can have up to 4 lines and you can get away with having 2 TDM400P in a PC |
| 19:47.44 | [TK]D-Fender | TimothyP: No. * has to forge the SIP return header and can only do 1 IP. |
| 19:47.47 | jbroome | TimothyP: you're going to have to poke a hole in the cisco to get sip through to * |
| 19:48.01 | TimothyP | but that would work |
| 19:48.07 | aydiosmio | [TK]D-Fender is right though, an external ATA would be cheapest instead of using PCI cards |
| 19:48.08 | TimothyP | so to the internet side it would work |
| 19:48.36 | rokerr | [TK]D-Fender: I'd need to use all sip-phones inside the network right? |
| 19:48.43 | jbroome | is your ubuntu machine doing firewalling? |
| 19:49.02 | TimothyP | it does masquerading |
| 19:49.07 | jbroome | i don't understand why you don't hang a switch off the cisco and hook all that stuff up to one subnet |
| 19:49.19 | [TK]D-Fender | rokerr: No, the SPA is a SIP device that will let you use a normal phone as a SIP phone. |
| 19:49.27 | *** join/#asterisk lose_the_grim (n=streppa@65.48.44.34) |
| 19:49.29 | aydiosmio | rokerr: no, the ATA would connect to your pc and then you hook the analog lines up to the device |
| 19:49.38 | [TK]D-Fender | rokerr: as well as taking in your home line. |
| 19:49.46 | lose_the_grim | Do Cisco IP phones support asterisk? |
| 19:49.55 | TimothyP | well let's say I do that (although I'm not allowed) , will in that case I be able to connect from the internet to my asterisk (I'm outside the country for example) using a sip phone and make asterisk redirect my call to a normal landline number for local free |
| 19:49.57 | lose_the_grim | I keep hearing about the SmartNET license |
| 19:49.58 | TimothyP | free* fee |
| 19:49.59 | jbroome | lose_the_grim: yes |
| 19:50.00 | aydiosmio | just like the TDM400P only you connect it to your PC via ethernet instead of PCI |
| 19:50.15 | jbroome | TimothyP: is this at work or your house? |
| 19:50.33 | lose_the_grim | jbroome: Using what protocal? I've got several vendors trying to sell me SmartNET contracts, and it sounds fishy |
| 19:50.48 | TimothyP | where I live I pay no internet, no phones, no electricity etc.... all I need to do is make sure the network works and I can rent the place for free |
| 19:50.49 | TimothyP | :) |
| 19:51.45 | TimothyP | it's in belgium, there's an office there with sip phones, in the 172.... network, my network is the 192. but let's say I can get my pc's in the 172 . Then I go to england and there I'm connected to the internet using ADSL I have a softphone with me and want to make a phoncall to someone in belgium so I connect to the asterisk and it calls out for me using it's phone line to the local number |
| 19:51.45 | jbroome | that is some bad juju. you may be paying rent RSK. |
| 19:52.14 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:52.18 | TimothyP | jbroome, I get lots of other things to , the point is I don't pay any costs, I get car, phone, internet, electricity, etc.... all for free :) |
| 19:52.27 | TimothyP | + large fee every month :) |
| 19:52.41 | TimothyP | and given my age that's a very good chance to save up some money for later |
| 19:53.34 | TimothyP | so Is what I'm suggestiing possible (ignoring the softphones in my own network for the moment :p) |
| 19:54.33 | rokerr | [TK]D-Fender, aydiosmio: ahh ok .. got it now |
| 20:00.30 | aydiosmio | jbroome: you the guy I made that mysql AGI for? |
| 20:01.14 | jbroome | aydiosmio: nope |
| 20:01.28 | aydiosmio | oh it was jbalcomb |
| 20:01.47 | aydiosmio | I've never been in a channel where everyone's nicks were so hard to distinguish |
| 20:06.07 | sevard | Does ztmonitor check dB levels? |
| 20:06.37 | aydiosmio | the volume level depends on the destination device really |
| 20:07.05 | *** part/#asterisk beu (i=beu@freenode/developer/gentoo.developer.beu) |
| 20:07.23 | sevard | i'm asking what it masures in |
| 20:07.24 | aydiosmio | otherwise volume is totally relative |
| 20:08.10 | aydiosmio | gain in dB |
| 20:08.16 | lose_the_grim | Hmmm.... So Cisco Phones will work with asterisk? But hey don't come with SIP firmware, so does SCCP work well? |
| 20:09.00 | SwK[Work] | just get the sip firmware and be done |
| 20:09.20 | aydiosmio | manual says voltage in dBm |
| 20:09.41 | sevard | aydiosmio: so you're saying the Rx or Tx is mostly dependant on the destination device |
| 20:09.41 | lose_the_grim | Okay, and getting the SIP formware requires a SmartNET contract? |
| 20:09.42 | aydiosmio | er wattage |
| 20:10.10 | aydiosmio | yeah, every terminating has it's own amplification qualities |
| 20:10.23 | aydiosmio | if it's powered, loss if it's unpowered |
| 20:10.34 | aydiosmio | so the destination volume is always goign to be relative |
| 20:10.57 | aydiosmio | gain input gain on one phone will be different on other phones |
| 20:11.09 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 20:11.26 | aydiosmio | er I should say, for one input gain, the volume will be different on each phone |
| 20:11.53 | *** part/#asterisk clive- (n=pirch@dsl-145-56-115.telkomadsl.co.za) |
| 20:13.15 | aydiosmio | when using ztmonitor you need to adjust your gain settings based on what you hear at the handset |
| 20:13.43 | Modcuts | Is there anyway of causing a sip client to full into unvailible via a command in the cli? |
| 20:14.28 | aydiosmio | oh man I am all like, talked out |
| 20:14.47 | sevard | like a teenage girl |
| 20:14.53 | aydiosmio | like totally |
| 20:15.39 | [TK]D-Fender | [mute] |
| 20:21.19 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 20:22.57 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 20:23.23 | *** join/#asterisk QMario (n=QMario@unaffiliated/QMario) |
| 20:23.40 | *** join/#asterisk lanceomni (n=jd@c-68-46-236-227.hsd1.fl.comcast.net) |
| 20:27.24 | *** join/#asterisk cvaldess (n=hello@209.Red-83-53-44.dynamicIP.rima-tde.net) |
| 20:27.38 | cvaldess | Hi |
| 20:27.42 | sevard | HI! |
| 20:28.07 | cvaldess | my *-1.2.12 reporting all sip call hangup cause 16 |
| 20:28.15 | cvaldess | even those terminate 34 |
| 20:28.27 | cvaldess | any idea how to solve this |
| 20:34.03 | ghenry | can anyone help with BT UK Caller ID? ANyone care to share their zapata.conf? |
| 20:35.35 | bhrobinson | got a question on inbound calls from third party sip connections |
| 20:35.46 | aydiosmio | awesome |
| 20:35.53 | aydiosmio | maybe you could share it with us sometime |
| 20:36.00 | aydiosmio | we can work you in tuesday |
| 20:36.25 | sevard | best response yet. |
| 20:36.34 | bhrobinson | I have a provider trying to use one of my channels using only ip authentication. when they connect, I get the dialog posted at http://pastebin.ca/170827 |
| 20:36.47 | bhrobinson | it shows connected, but I get no conversation... |
| 20:36.49 | *** join/#asterisk liamo (n=liam@86.43.74.94) |
| 20:36.54 | bhrobinson | and thanks... feel the love :) |
| 20:38.15 | bhrobinson | can you not peer traffic like that? |
| 20:40.59 | aydiosmio | <PROTECTED> |
| 20:41.03 | aydiosmio | oops |
| 20:41.30 | *** join/#asterisk QMario_ (n=QMario@unaffiliated/QMario) |
| 20:42.59 | *** join/#asterisk [hC] (n=root@donkey.voxter.ca) |
| 20:43.00 | aydiosmio | my question is |
| 20:43.09 | aydiosmio | WHO MISSPELLED "TOO" |
| 20:43.13 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 20:43.16 | aydiosmio | Got SIP response 400 "Content To Short" |
| 20:43.18 | jhiver | hey guys |
| 20:43.38 | jhiver | i was wondering if there was a way i could do ARA but like /without/ using a database |
| 20:43.46 | aydiosmio | bhrobinson: is 10.25.2.226 your provider? |
| 20:43.50 | bhrobinson | lol... that is a horrid leadtek device... just happeend to get in there while we tried to lod the others |
| 20:43.50 | [hC] | Im having people experiencing jitter and echo on an installed system.. their internet connection is fine, but im testing their lan cabling, what is acceptable jitter on a lan? |
| 20:44.01 | [hC] | 5ms should be fine, i would presume. |
| 20:44.05 | aydiosmio | bhrobinson: there's stuff on google about this bro |
| 20:44.11 | jhiver | I kind of have a custom auth mechanism and i was wondering if it'd be possible to plug it into asterisk somehow... |
| 20:44.15 | aydiosmio | firmware upgrade fixes it |
| 20:44.20 | bhrobinson | aydiosmio, no... that is a ata unit here for fax... |
| 20:44.36 | Juggie | [hC], you know damn well 5ms is fine |
| 20:44.43 | Juggie | <400ms round trip is fine. |
| 20:44.50 | Juggie | its packet loss thats a problem |
| 20:45.34 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:45.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:45.36 | luke-jr_work | erm |
| 20:45.40 | luke-jr_work | 400ms is horrible |
| 20:45.48 | aydiosmio | yeah that's pretty bad |
| 20:45.54 | aydiosmio | < 100ms is the standard |
| 20:46.32 | syzygyBSD | how many pairs does a t1 use? |
| 20:46.41 | aydiosmio | 2 |
| 20:46.44 | wunderkin | 42 |
| 20:46.48 | [hC] | I just cant figure out why these people are having problems |
| 20:46.49 | luke-jr_work | 3729924 |
| 20:46.50 | wunderkin | heh |
| 20:46.53 | aydiosmio | wunderkin: that's not the answer to everything |
| 20:46.59 | [hC] | their DSL connection is rock solid at like 16ms |
| 20:46.59 | luke-jr_work | 65536 |
| 20:47.02 | [hC] | no loss |
| 20:47.12 | [hC] | yet they constantly have jittered audio, dropped calls, and echo |
| 20:47.13 | aydiosmio | hs |
| 20:47.15 | [hC] | they are going straight out a PRI |
| 20:47.22 | [hC] | after one iax hop to me |
| 20:47.27 | [hC] | so i figure it must be in their lan somewhere. |
| 20:47.28 | aydiosmio | [hC]: this is an * box? |
| 20:47.35 | [hC] | yeah * all the way thru |
| 20:47.46 | [hC] | client does iax to me using *, i connect them to pri using * |
| 20:47.51 | [hC] | we are using g729 to connect the two. |
| 20:47.58 | cvaldess | my *-1.2.12 reporting all sip call hangup cause 16 |
| 20:47.59 | aydiosmio | check the system resources, free RAM, CPU and make sure there are no interrupt conflicts with the ethernet device |
| 20:48.00 | *** join/#asterisk RMooe (n=HiTMaN@80-235-135-138.cable.ubr07.newt.blueyonder.co.uk) |
| 20:48.01 | cvaldess | even those terminate 34 |
| 20:48.04 | cvaldess | any idea how to solve this |
| 20:48.08 | RMooe | hello |
| 20:48.14 | [hC] | is g729 extremely succeptible to jitter and echo? |
| 20:48.17 | aydiosmio | bhrobinson: I don't see the problem then |
| 20:48.29 | aydiosmio | [hC]: not particularly |
| 20:48.38 | [hC] | plenty of free cpu, memory, |
| 20:48.49 | aydiosmio | and the problems might not be related |
| 20:48.52 | [hC] | they have a zap interface which is sharing an interrupt with the usb controller, but they arent even using the zap device. |
| 20:48.58 | [hC] | its all iax2 out of here. |
| 20:49.01 | cvaldess | need help with SIP HangupCause report |
| 20:49.14 | syzygyBSD | so is there any reason I couldnt' use cat5 as two t1s? |
| 20:49.32 | aydiosmio | [hC]: change out the ethernet cards if traffic in ethereal looks normal |
| 20:49.39 | RMooe | i could really use a hand here , i have been trying, and reading google, FAQ's and all over for the answer , after 7hours of this.. i don't know what to say, i've tried reinstalling asterisk with no luck - my question is.. how can i register to Asterisk with no username or password.. everytime i am using insecure=very and host as the static IP.. still AUTH FAILED - PLEASE help me? |
| 20:49.57 | [hC] | aydiosmio: i just may.. i just finished a 12 minute ethereal capture to see if i can find something weird. |
| 20:50.52 | aydiosmio | and use a phone connected directly to the box if possibel to do your testing |
| 20:51.01 | aydiosmio | to avoid the rest of th network |
| 20:51.46 | aydiosmio | syzygyBSD: you can, not the best idea there is though |
| 20:51.51 | aydiosmio | you may get crosstalk |
| 20:52.09 | aydiosmio | if you see a lot of frame errors, that'll be the reason |
| 20:52.27 | mtoups | Hi, I am having a problem with Monitor() wav recordings clipping, similar to this bugreport: http://bugs.digium.com/view.php?id=5823&nbn=13 |
| 20:52.33 | syzygyBSD | well, i actually have 50 pair running from my dmark, just wondering if I can run 4 wires instead of 8 |
| 20:52.36 | aydiosmio | probability is low |
| 20:52.46 | RMooe | please help? |
| 20:52.55 | mtoups | but it looks like that bug was closed without being fixed (as "unable to reproduce") |
| 20:53.30 | aydiosmio | RMooe: what are you registering? |
| 20:53.36 | syzygyBSD | RMooe: pastbin your sip.conf |
| 20:53.49 | aydiosmio | syzygyBSD: T1 is 2 pair, 4 wires |
| 20:53.59 | aydiosmio | so 2 T1s on a cat-5 |
| 20:54.13 | Strom_C | you'd have to be bonkers to run two T1s on a single cat5 cable |
| 20:54.15 | RMooe | willdo syzygyBSD thanks |
| 20:54.22 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:54.29 | syzygyBSD | ya.. I was just wondering if I could do 24 t1s on a 50 pair |
| 20:54.33 | RMooe | just the [] part i'm using? because otherwise its as default.. my outgoing providers aren't in there yet due to a freesh install |
| 20:54.35 | Strom_C | theoretically yes |
| 20:54.46 | aydiosmio | 16:51 < aydiosmio> you may get crosstalk |
| 20:54.48 | aydiosmio | 16:52 < aydiosmio> if you see a lot of frame errors, that'll be the reason |
| 20:54.48 | syzygyBSD | lol, good answer Strom_C |
| 20:54.54 | Strom_C | but at that point you'd be better off gettng a DS3 and demuxing it at the far end |
| 20:55.04 | Strom_C | since a DS3 is 28 DS1s |
| 20:55.32 | aydiosmio | mmm fiber optic interconnect |
| 20:55.48 | syzygyBSD | well, if we are selling individual T1s then we have to get them up to our data center from the dmark, they have to come in as a t1 |
| 20:55.56 | aydiosmio | 24 T1 eh? think you could do TDm over gigabit? |
| 20:56.08 | aydiosmio | that'd be fun |
| 20:56.10 | Strom_C | syzygyBSD: what? put a channel bank on either end and run a single DS3 |
| 20:56.55 | Strom_C | that's what they bloody invented multiplexing for :) |
| 20:57.13 | Adam06 | What would cause a Polycom 301 to constantly not find the boot server? |
| 20:57.24 | Strom_C | misconfiguration |
| 20:57.25 | aydiosmio | Adam06: non-existant host server |
| 20:57.42 | Adam06 | Host server exists, I can ping it no problem. |
| 20:58.00 | Adam06 | Strom > I've checked the configuration, and it doesn't seem to be wrong |
| 20:58.10 | syzygyBSD | Strom_C: why convert the signal twice? |
| 20:58.50 | Strom_C | syzygyBSD: becuse it's easier and more efficient to run a single DS3 than it is to run a whole ton of copper T1s |
| 20:59.11 | lanceomni | is the final version of Fedora Core 5 able to properly compile the Zaptel drivers? |
| 20:59.17 | Juggie | yes |
| 20:59.19 | RMooe | lanceomni i was able |
| 20:59.20 | Strom_C | also, assuming you use the right fiber, it will then be easy to upgrade to OC3 if demand increases |
| 20:59.26 | lanceomni | thanks |
| 20:59.31 | RMooe | np |
| 21:00.03 | syzygyBSD | Strom_C: lol, ya, but the copper is already there |
| 21:00.09 | RMooe | syzygyBSD its 4 lines, pastebin.com won't 'send' the paste ??? it just keeps loading, jesus.. |
| 21:00.15 | RMooe | truely annoying ;) |
| 21:00.26 | syzygyBSD | try pastebin.ca |
| 21:00.32 | syzygyBSD | ~pastebin |
| 21:00.34 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 21:00.55 | Strom_C | syzygyBSD: well, i suppose you could do that. 50 pair cable, 25 T1s...how long is the cable? |
| 21:01.10 | syzygyBSD | Strom_C: no more then 200 feet |
| 21:01.22 | syzygyBSD | just from the dmark to our rack |
| 21:01.35 | Strom_C | btw, it's spelled "demarc" |
| 21:01.47 | Strom_C | </pedantic> |
| 21:01.47 | syzygyBSD | fine grammer nazi |
| 21:01.52 | syzygyBSD | demarc |
| 21:01.55 | syzygyBSD | sorry |
| 21:02.14 | Strom_C | I suppose you could get away with that |
| 21:02.23 | Strom_C | but that's a hell of a lot of wiring |
| 21:02.29 | RMooe | syzygyBSD : http://88.208.209.44/sip.txt |
| 21:02.46 | RMooe | should this not allow any username/pass to register from this IP? |
| 21:04.14 | syzygyBSD | ya, I am not looking forward to the wiring, but we won't fill it up right away, and I don't know what we will use them all for, we want to leave it as open as possible but get as much of the wiring done |
| 21:04.49 | Strom_C | where do all these T1s come from? telco? |
| 21:06.04 | syzygyBSD | nowhere yet, but probably telco, maybe have one go to the bosses house, my brothers |
| 21:06.29 | Strom_C | if you're getting all those T1s from the telco, they're probably going to deliver them via DS3 anyway and demux them at the prem |
| 21:06.43 | syzygyBSD | ya, the problem is we don't know yet... |
| 21:06.56 | Strom_C | well, why dont you find out first /before/ you invest time and money into wiring |
| 21:07.03 | syzygyBSD | we will start off with one and grow... |
| 21:07.15 | Strom_C | start off with a single T1? |
| 21:07.22 | syzygyBSD | well that is what I am doing right now, the planning |
| 21:07.33 | syzygyBSD | that is why I am asking the questions |
| 21:08.06 | syzygyBSD | Strom_C: I think so, but we already have 50 pair run from our rack to the demarc |
| 21:08.31 | syzygyBSD | i am just figuring out the best way to wire that so we have as many options in the future as possible |
| 21:08.32 | Strom_C | right, but i think a better angle is to find out "how can we accomplish [goal]" rather than "how can we kludge [existing hackjob] to accomplish [goal]" |
| 21:08.54 | syzygyBSD | Strom_C: true, but the problem is we dont' have a goal yet |
| 21:08.58 | aydiosmio | lorf |
| 21:09.08 | Strom_C | ok.... |
| 21:09.17 | Strom_C | what's your short term goal |
| 21:09.24 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 21:09.30 | syzygyBSD | RMooe: can you pastebin your entire sip.conf |
| 21:10.41 | *** part/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:11.21 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 21:11.31 | syzygyBSD | well, reall short term is to get this wired... lol, ok, no help there, but we want to get a few t1s from the demarc to the rack. These will most likely be T1s. We will be starting with one though |
| 21:11.48 | Strom_C | but for what purpose |
| 21:11.51 | syzygyBSD | we may sell them to various people or buisinesses or run them to our houses |
| 21:11.54 | Strom_C | what are you trying to accomplish |
| 21:12.35 | syzygyBSD | One T1 will be for phones, hopefully a node on the SS7 network |
| 21:13.03 | syzygyBSD | I don't know much about that yet, one thing at a time |
| 21:13.13 | RMooe | syzygyBSD i could, but its just the default stuff.. i did 'make samples' and add this on the end |
| 21:13.22 | syzygyBSD | RMooe: please do |
| 21:13.25 | RMooe | ok |
| 21:14.04 | syzygyBSD | also if you could include your extensions.conf |
| 21:14.17 | RMooe | extensions.conf is purposely blank |
| 21:14.20 | bhrobinson | aydiosmio, I just caught up on all you said... local phones work fine... it is only the relay that is not working |
| 21:14.26 | RMooe | i don't have any outgoings configured syzygyBSD |
| 21:14.39 | syzygyBSD | that doesn't matter, I need to see your contexts |
| 21:14.43 | RMooe | hold on i'll get all of it |
| 21:14.55 | RMooe | well the only problem is i can't register as any user/password |
| 21:14.57 | RMooe | and i need to |
| 21:15.04 | *** join/#asterisk file2 (n=IrcNet@138.sub-70-201-85.myvzw.com) |
| 21:15.04 | *** mode/#asterisk [+o file2] by ChanServ |
| 21:15.18 | syzygyBSD | yes, that is why I am asking for the files I am |
| 21:15.51 | RMooe | ok hold on |
| 21:17.58 | RMooe | http://195.72.131.100/sip.conf |
| 21:18.09 | RMooe | extensions.conf is truely blank, my one context will be 'default' |
| 21:18.21 | syzygyBSD | will be or is? |
| 21:18.25 | syzygyBSD | you need that defined |
| 21:18.36 | RMooe | is default |
| 21:18.42 | aydiosmio | bhrobinson: I didn't see any other errors in the past you posted, so I don't know what your issue is |
| 21:19.27 | lose_the_grim | What phone/panel would you guys recommend for a receptionist? |
| 21:19.52 | syzygyBSD | RMooe: change the [register] to [test] |
| 21:19.57 | RMooe | ok |
| 21:20.02 | syzygyBSD | and type=peer to type=friend |
| 21:20.25 | RMooe | yes |
| 21:20.49 | syzygyBSD | now where is the call you are testing originating from /terminating? |
| 21:21.00 | cvaldess | need help with SIP HangupCause report |
| 21:21.15 | RMooe | well it will be orginating from here .. because its a test |
| 21:21.19 | RMooe | terminating? UK ? |
| 21:21.37 | syzygyBSD | what are you connecting to? |
| 21:21.41 | RMooe | i will connect an outsource provider |
| 21:21.44 | syzygyBSD | the sip device at the other end |
| 21:21.46 | RMooe | i know how to do that.. |
| 21:21.48 | RMooe | but |
| 21:22.13 | RMooe | i thought iknew how to do this too.. but it obviously isn't like it seems, i thought that [register] block would allow any incoming connection with any user/psw to register to Asterisk |
| 21:22.15 | RMooe | this is what i need |
| 21:22.46 | syzygyBSD | right... one thing at a time |
| 21:22.49 | lose_the_grim | Anyone have any luck with the Polycom Soundpoint IP Attendant? |
| 21:23.14 | syzygyBSD | you say an incomming connection, from what? |
| 21:23.23 | RMooe | from a SIP phone |
| 21:23.35 | bhrobinson | aydiosmio, thanks...I will keep trying then |
| 21:23.56 | syzygyBSD | ok, |
| 21:24.15 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:24.19 | syzygyBSD | after you made the changes I asked for have you reloaded? |
| 21:24.21 | RMooe | but it must register by IP .. or with any user/password |
| 21:24.22 | aydiosmio | bhrobinson: run asterisk with verbosity and debugging at 10 |
| 21:24.24 | RMooe | yes i have done |
| 21:24.37 | RMooe | 'sip reload' & 'reload' |
| 21:24.40 | syzygyBSD | RMooe: do you have any exten => entries in extensions.conf |
| 21:24.43 | RMooe | nope |
| 21:24.47 | syzygyBSD | add one |
| 21:24.48 | RMooe | must i put this? |
| 21:24.49 | RMooe | ok |
| 21:25.03 | syzygyBSD | exten => 123,1,Noop(connected) |
| 21:25.26 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 21:25.33 | harryvv | Anyone here know of a company selling 604 DIds ? I talked to a CLEC that btw is a buyer/compeditor of telus and thay do but only using there own ata gear. Some reason a asterisk box would only talk to there nortel switch for 15 min and die. Like to use them but need to find somone else. |
| 21:25.54 | syzygyBSD | do you want a username/password on your sip phone? (recommended strongly) |
| 21:25.58 | RMooe | i have added the extensions |
| 21:26.00 | RMooe | no i don't |
| 21:26.04 | RMooe | this is the point :) |
| 21:26.09 | syzygyBSD | right, ok |
| 21:26.18 | aydiosmio | harryvv: telus? are you talkign about the area code 604? |
| 21:26.21 | syzygyBSD | reload again |
| 21:26.23 | aydiosmio | in the US? |
| 21:26.24 | RMooe | have done |
| 21:26.28 | syzygyBSD | sip debug |
| 21:26.33 | RMooe | yes |
| 21:26.38 | syzygyBSD | dial 123 from the phone |
| 21:26.48 | RMooe | sec |
| 21:27.17 | C6Vette | what would be causing this when I do a stop now.... overrun!!! (at least 126351.116 ms long), stdout:Broken pipe, Yuck! Error in buffer handling... |
| 21:27.32 | RMooe | ok |
| 21:27.38 | RMooe | wait, i can't register |
| 21:27.39 | RMooe | on the phone |
| 21:27.40 | harryvv | aydiosmio yes |
| 21:27.46 | syzygyBSD | anything come through on sip debug |
| 21:28.07 | harryvv | I want to use and resell navigata services but there switches dont seem to be compatible with asterisk |
| 21:28.13 | RMooe | yes |
| 21:28.19 | syzygyBSD | pastebin all of it |
| 21:28.27 | RMooe | sec |
| 21:28.49 | syzygyBSD | If I was looking for a punchdown block, what is another name for it |
| 21:29.02 | aydiosmio | oh 604 is in canada |
| 21:29.29 | harryvv | that is the name |
| 21:29.39 | harryvv | what are u trying to do? |
| 21:29.49 | syzygyBSD | I was hoping newegg would have one |
| 21:29.59 | syzygyBSD | connect 50 pair |
| 21:30.02 | aydiosmio | syzygyBSD: 110-block |
| 21:30.03 | Adam06 | does VSFTP offer a tftp plugin? |
| 21:30.09 | Adam06 | or ability to server tftp? |
| 21:30.13 | Adam06 | serve* |
| 21:30.16 | aydiosmio | no |
| 21:30.50 | aydiosmio | Adam06: there are plenty of fre tftp clients available |
| 21:30.50 | RMooe | http://88.208.209.44/dump.txt |
| 21:31.12 | aydiosmio | e.g. http://perso.orange.fr/philippe.jounin/tftpd32.html |
| 21:31.28 | RMooe | i am not registered to the server syzygyBSD |
| 21:31.31 | Adam06 | looking for Linux :p |
| 21:31.34 | syzygyBSD | RMooe: add extension s to default |
| 21:31.52 | syzygyBSD | exten => s,1,Noop(blah blah) |
| 21:32.31 | RMooe | ok |
| 21:32.44 | syzygyBSD | then try it again, and update that dump.txt |
| 21:33.58 | aydiosmio | Adam06: isn't vsftp a windows application? |
| 21:34.10 | syzygyBSD | aydiosmio: no |
| 21:34.16 | RMooe | ok i did that |
| 21:34.54 | aydiosmio | oh, I was thinking of somethign else |
| 21:35.44 | sevard | What the hell is ztmonitor actually measuring in |
| 21:35.56 | sevard | aydiosmio: this isn't dB |
| 21:36.03 | syzygyBSD | Hz? |
| 21:36.18 | [hC] | percentage of total gain? its like a vu meter or something. |
| 21:36.20 | sevard | any way to know for sure? the manpage isn't telling me shit. |
| 21:36.30 | [hC] | oh |
| 21:36.31 | [hC] | its hz. |
| 21:36.36 | [hC] | -vv will show you hz |
| 21:36.36 | sevard | are you sure? |
| 21:36.41 | RMooe | this shouldn't be: Registration from '<sip:aaa@195.72.131.100:5060>' failed for '80.235.135.138' - Username/auth name mismatch |
| 21:36.41 | [hC] | absolutely |
| 21:37.04 | syzygyBSD | RMooe: it looks like you are behind a nat router |
| 21:37.12 | sevard | I dial a 1004 test number and I get 16575 (hz?!) |
| 21:37.13 | aydiosmio | sevard: dBm |
| 21:37.20 | RMooe | but even when i'm not |
| 21:37.24 | RMooe | i still cannot register to the service |
| 21:37.25 | syzygyBSD | add 'nat=1' to your sip.conf |
| 21:37.27 | sevard | aydiosmio says dBm [hC] says hz |
| 21:37.29 | RMooe | i have tried from other services |
| 21:37.33 | RMooe | servers |
| 21:37.52 | syzygyBSD | RMooe: ok, one thing at a time, i can't debug those services right now |
| 21:37.56 | *** join/#asterisk Waverly360 (n=mirc@adsl-154-162-92.bna.bellsouth.net) |
| 21:38.04 | cvaldess | need help with SIP HangupCause report, all calls reporting 16 |
| 21:38.09 | syzygyBSD | that and you have a username on your phone right now |
| 21:38.18 | RMooe | yes i know |
| 21:38.20 | syzygyBSD | so you really should add one to sip.conf |
| 21:38.21 | sevard | so... |
| 21:38.22 | RMooe | but i should be able to use any username |
| 21:38.27 | sevard | is it dB or Hz |
| 21:38.32 | RMooe | if it change it to aaa or bbb or zzzz |
| 21:38.49 | aydiosmio | Hz wouldn't matter for volume/gain |
| 21:38.54 | *** join/#asterisk Nand0 (n=Nando@unaffiliated/nand0) |
| 21:39.23 | jmsjms | Hello, could somebody please help? I've spent ages on what I'm sure is a very simple problem |
| 21:39.27 | jmsjms | I want to dial a number (via my iax2 provider) from my call menu, then continue in the call menu after the dialed number hangs up. Right now, when the dialed number hangs up, so does the caller |
| 21:39.51 | Waverly360 | Hey guys, I just had a problem with a pbx and a ring-all queue. There are 5 phones in the queue, and out of the 5, only 2 looked like they were ringing while logged into the CLI. I'm not actually on-site, so I can't tell if those two were actually ringing, but no one answered. |
| 21:40.01 | jmsjms | the line in my extensions.conf is: exten => s,1,Dial(IAX2/james1,30,ghH) |
| 21:40.04 | Waverly360 | So I restarted asterisk, and all 5 started ringing again. |
| 21:40.17 | Waverly360 | looking in the log file I found this: app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 21:40.21 | Waverly360 | Any clues what's going on here? |
| 21:40.23 | jmsjms | but for some reason, the 'g' parameter doesn't seem to do what it says on the tin |
| 21:41.01 | RMooe | still with the nat=yes syzygyBSD, its not authing with any user/pass |
| 21:41.01 | aydiosmio | there's a reference to a VU meter, VU meters display volume levels in dB |
| 21:41.07 | RMooe | do i need to add 'insecure=very' ? or something |
| 21:41.11 | syzygyBSD | no |
| 21:41.13 | RMooe | i did try all of this |
| 21:41.21 | syzygyBSD | insecure=very has to do with invites |
| 21:41.25 | RMooe | ok |
| 21:41.34 | RMooe | then how can i auth by IP and not by user/password |
| 21:43.12 | aydiosmio | sevard: http://packages.debian.org/stable/net/tftpd |
| 21:43.19 | aydiosmio | sevard: http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.html |
| 21:44.09 | rajiv|work | anyone see a problem where ast thinks you have a vm message but it cannot play it ?Sep 14 17:41:27 WARNING[26600]: file.c:512 ast_openstream_full: File /var/spool/asterisk/voicemail/default/207/INBOX/msg0000 does not exist in any format |
| 21:44.36 | *** join/#asterisk zeppelin_ (n=zeppelin@201.35.139.252) |
| 21:44.57 | mtoups | speaking of dB, I am having a problem with Monitor() wav recordings clipping, similar to this bugreport: http://bugs.digium.com/view.php?id=5823&nbn=13 but the bugreport seems to have been closed as "unable to reproduce" |
| 21:45.08 | mtoups | anybody else seen this or know more? |
| 21:45.20 | aydiosmio | sevard: from what I'm seeing the meter output of ztmonitor is a VU meter and 0db is at the center, where the level should be under a test signal |
| 21:45.41 | aydiosmio | e.g. http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment |
| 21:46.09 | RMooe | syzygyBSD any suggestions |
| 21:46.13 | sevard | aydiosmio: do a ztmonitor <chan> -v -v -v |
| 21:46.44 | aydiosmio | I don't have any zap channels:) |
| 21:46.57 | syzygyBSD | nope |
| 21:47.01 | RMooe | ah |
| 21:47.04 | RMooe | stuck ? lol |
| 21:47.26 | *** join/#asterisk rsd (n=chaos@200.181.133.130) |
| 21:47.59 | syzygyBSD | well, you phone is trying to auth by username, since that doesn't match the nothing asterisk has... |
| 21:48.18 | syzygyBSD | authentication information has to match |
| 21:48.19 | RMooe | but even when i auth as nothing |
| 21:48.39 | syzygyBSD | RMooe: why can't you just use a usename/pass? |
| 21:48.45 | RMooe | because |
| 21:49.04 | RMooe | a switch i would like to use doesn't have that option |
| 21:49.16 | syzygyBSD | huh? |
| 21:49.32 | sx-wks | hmmm... using postgresql transactions to assure of atomicity of things happening in the asterisk is FUN :D |
| 21:49.56 | sx-wks | such as conferences serial numbers & stuff :D |
| 21:50.10 | RMooe | i have a switch that cannot use a username and password |
| 21:50.24 | syzygyBSD | a switch? |
| 21:50.44 | RMooe | yes |
| 21:50.58 | syzygyBSD | how does that have anything to do with SIP? |
| 21:51.08 | RMooe | because it connects to my Asterisk |
| 21:51.10 | RMooe | or is supposed to |
| 21:51.14 | RMooe | but it doesn't support auth |
| 21:51.15 | RMooe | only via IP |
| 21:51.22 | RMooe | so i need to do ? imagine: Allow (IP) |
| 21:51.24 | RMooe | to connect |
| 21:51.28 | RMooe | lol.. you understand? |
| 21:51.48 | syzygyBSD | I have never heard of a switch that connects to asterisk (anyone here correct me if i am wrong) |
| 21:51.59 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-64-159.cybersurf.com) |
| 21:52.05 | RMooe | lol you don't sell VoIP? |
| 21:52.31 | syzygyBSD | not like that |
| 21:52.40 | RMooe | ok. |
| 21:52.44 | RMooe | well thats why |
| 21:52.53 | syzygyBSD | what is the name of the product? |
| 21:52.53 | RMooe | but its besides the point, point is.. i can't make this work |
| 21:52.57 | RMooe | been trying about 8hours |
| 21:53.12 | RMooe | which product |
| 21:53.12 | *** join/#asterisk CrashSys (i=kumba@loki.crashsys.com) |
| 21:53.12 | syzygyBSD | this 'switch' |
| 21:53.33 | CrashSys | If I change the echo_Can in zconfig.h and recompile zaptel, do I need to recompile asterisk or just reload zaptel? |
| 21:53.39 | RMooe | still chan_sip.c:11084 handle_request_register: Registration from '<sip:@195.72.131.100>' failed for '88.208.209.44' - Username/auth name mismatch |
| 21:53.51 | RMooe | 'nextone' |
| 21:54.02 | intralanman | RMooe: that's an SBC isn't it? |
| 21:54.24 | RMooe | SBC? |
| 21:54.30 | intralanman | session border controller |
| 21:54.33 | CrashSys | SBC? Small Block Chevy? :) |
| 21:54.38 | intralanman | or that |
| 21:54.39 | intralanman | lol |
| 21:54.42 | Strom_C | Southwestern Bell Corporation? |
| 21:55.04 | syzygyBSD | intralanman: looks like it |
| 21:55.16 | intralanman | i was looking at them a while back |
| 21:55.27 | syzygyBSD | "IntelliConnectâ„¢ session switches provide a robust and highly programmable route engine" .... |
| 21:55.28 | syzygyBSD | hah |
| 21:55.37 | RMooe | lol |
| 21:55.40 | RMooe | ok? so i can't change it |
| 21:55.50 | aydiosmio | SBC aka telco conglomerate C |
| 21:55.51 | RMooe | i don't wanna sit here picking it apart man, i really need to fix this problem |
| 21:55.54 | RMooe | its a real shit |
| 21:55.56 | CrashSys | So, do I need to recompile asterisk if I change the echo_can in zaptel? or just reload? :) |
| 21:56.14 | aydiosmio | CrashSys: somthing that can easily be tested. |
| 21:56.29 | intralanman | RMooe: it doesn't have different levels of access? |
| 21:56.43 | intralanman | you should be able to set it to peer or access or sub or something |
| 21:56.46 | RMooe | it doesn't have |
| 21:57.08 | RMooe | look, somebody help me get it fixed i'll provide you some hours to cuba? |
| 21:57.15 | intralanman | RMooe: why do you need it? |
| 21:57.19 | aydiosmio | to cuba? |
| 21:57.20 | RMooe | because its very important |
| 21:57.25 | intralanman | using it for media proxy or what? |
| 21:57.27 | CrashSys | Hours to cuba? |
| 21:57.29 | CrashSys | ... |
| 21:57.33 | RMooe | yes hours to cuba |
| 21:57.34 | CrashSys | Like miami vice style? |
| 21:57.37 | RMooe | no.. |
| 21:57.40 | aydiosmio | that's like giving someon free dead batteries |
| 21:57.43 | CrashSys | ohh... |
| 21:57.43 | RMooe | cuba is a costly country? |
| 21:57.51 | intralanman | for whom? |
| 21:57.58 | RMooe | ok, i don't - |
| 21:57.58 | CrashSys | Cubans I guess... |
| 21:58.01 | RMooe | lets not discuss this |
| 21:58.04 | RMooe | i am asking for help |
| 21:58.06 | aydiosmio | yes |
| 21:58.07 | intralanman | cuba is a worthless country |
| 21:58.11 | aydiosmio | let's move to ACTION! |
| 21:58.17 | aydiosmio | intralanman: not true |
| 21:58.20 | intralanman | ok |
| 21:58.28 | intralanman | worthless to me |
| 21:58.29 | aydiosmio | many of our baseball players are cub... oh yeah, pretty useless I guess |
| 21:58.39 | intralanman | rofl |
| 21:58.46 | syzygyBSD | RMooe: so let me see if I see through this lack of information, you have lots of phones behind a nextone and don't want to add a sip entry for each |
| 21:58.57 | sevard | Can I blow a tone down a PRI without initiating or terminating a call? |
| 21:59.01 | RMooe | no, not true |
| 21:59.03 | RMooe | the bottom line is |
| 21:59.07 | RMooe | i have a client using nextone switch |
| 21:59.07 | sevard | Does anyone know that? |
| 21:59.10 | RMooe | he wants to connect to asterisk |
| 21:59.11 | Strom_C | sevard: no |
| 21:59.12 | RMooe | won't change his switch |
| 21:59.15 | RMooe | being a pain in the ass |
| 21:59.18 | RMooe | i tried enough hours already |
| 21:59.25 | sevard | Strom_C: I didn't think it was possible, my telco guy is being an asshole |
| 21:59.28 | RMooe | so now i need to either a.) pay somebody to fix it |
| 21:59.33 | Strom_C | sevard: explain the problem |
| 21:59.34 | RMooe | well thats it |
| 21:59.38 | CrashSys | wow... mark2/aggressive did the trick :) |
| 21:59.55 | sevard | Strom_C: He wants me to blow a 1000hz tone down a PRI channel without having to call in from the PSTN or from a normal phone |
| 21:59.57 | CrashSys | these crappy x101p's get a stay of execution till I find some ebay hardware :) |
| 22:00.30 | syzygyBSD | I haven't done anything with nextone, but it is hard to believe any sip device wouldn't have the ability for a usename/password |
| 22:00.35 | intralanman | RMooe: so he's connecting his POS nextone to your asterisk box? |
| 22:00.36 | CrashSys | BTW... if your shitstream BT-200's or GXP's echo really bad, get the beta firmware... fixes 95% of the echo :) |
| 22:00.46 | Strom_C | sevard: why? |
| 22:00.52 | RMooe | yeah i guess |
| 22:00.58 | RMooe | syzygyBSD seriously |
| 22:01.16 | syzygyBSD | seriously |
| 22:01.19 | intralanman | RMooe: make him a peer and trust his IP |
| 22:01.21 | sevard | Strom_C: He doesn't want to come across the PSTN because he says it'll be testing multiple systems intstead of just one |
| 22:01.35 | Strom_C | sevard: but WHAT are you trying to troubleshoot? |
| 22:01.42 | intralanman | syzygyBSD: i'll agree that some devices don't |
| 22:01.44 | sevard | Storm_C: dB levels |
| 22:01.53 | CrashSys | I guess you would need to make a 1K audio file in native asterisk format, and just use play? |
| 22:02.03 | intralanman | like nextone and sansay for SBC's won't auth with another device |
| 22:02.11 | CrashSys | make a call from one system to the next? |
| 22:02.29 | CrashSys | use .call's... |
| 22:05.39 | *** join/#asterisk dasenjo_ (n=dasenjo@63.245.86.130) |
| 22:06.20 | CrashSys | quit |
| 22:06.22 | CrashSys | doh |
| 22:08.41 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 22:09.38 | syzygyBSD | intralanman: do you know how to allow them to connect to asterisk |
| 22:10.09 | intralanman | they can... they just have to know the ip of the asterisk box and let their traffic go to that ip |
| 22:10.20 | intralanman | the asterisk box just sees them as a peer |
| 22:10.23 | syzygyBSD | how does asterisk trust them though... |
| 22:10.31 | intralanman | or friend as the case may be |
| 22:10.34 | intralanman | host= |
| 22:10.39 | RMooe | i did host= |
| 22:11.17 | syzygyBSD | RMooe: the tests you gave me were from the nextone? |
| 22:11.27 | RMooe | they were from the Asterisk box |
| 22:11.33 | RMooe | i am connecting via SIP phone |
| 22:11.34 | RMooe | for now |
| 22:11.37 | RMooe | to make things easier |
| 22:11.53 | intralanman | i have a sansay set up |
| 22:11.54 | syzygyBSD | well a sip phone wont' act the same way, so that doesn't make it easier |
| 22:12.01 | intralanman | which is the same as a nextone only cheaper ;) |
| 22:12.05 | RMooe | ok, well i am now trying to connect from an actual switch |
| 22:12.10 | syzygyBSD | maybe intralanman can help since he has done it before |
| 22:12.17 | RMooe | intralanman please help? |
| 22:12.53 | intralanman | if i understand correctly, i'd be happy to |
| 22:13.08 | intralanman | i just wanna make sure i'm totally sure about what you want to do |
| 22:13.37 | intralanman | your asterisk box is to route calls for someone behind the nextone? |
| 22:20.41 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 22:24.23 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 22:26.34 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.109.250) |
| 22:26.44 | Greek-Boy | who is the cheapest wholesale provider? |
| 22:32.19 | syzygyBSD | can anyone recommend a device for multiple T1 termination/origin? |
| 22:32.45 | cvaldess | any one can helpme with SIP problem??? |
| 22:32.57 | bkruse | ........ |
| 22:33.05 | bkruse | oh please give me a hint! |
| 22:33.08 | syzygyBSD | and the problem is... |
| 22:33.15 | cvaldess | my *-1.2.12 reporting all sip call hangup cause 16 |
| 22:33.23 | cvaldess | even those terminate 34 |
| 22:34.08 | cvaldess | printing $HANGUPCAUSE on exten h we can see all causes ok |
| 22:34.45 | cvaldess | but at sip end only get report of hangupcause 16 |
| 22:35.33 | cvaldess | any idea???? |
| 22:36.37 | eKo1 | cvaldess: What? |
| 22:37.20 | cvaldess | my SIP only report to the client side hangupcause=16 |
| 22:38.08 | Greek-Boy | can anyone recommend a cheap IAX provider? |
| 22:38.12 | eKo1 | That does not compute. Syntax error. |
| 22:38.23 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6E601.versanet.de) |
| 22:41.24 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 22:49.01 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:50.00 | syzygyBSD | what do people hwere use for data t1 termination |
| 22:50.44 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 22:53.34 | X-Rob | a cable |
| 22:53.44 | syzygyBSD | ha ha ha |
| 23:00.24 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:03.46 | *** join/#asterisk mercestes (n=merceste@216.54.143.242) |
| 23:05.43 | *** join/#asterisk grexk (n=grexk@124.107.72.45) |
| 23:18.57 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 23:19.54 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:24.07 | Strom_C | If a SIP user is set to host=dynamic but sends a call without having registered, what happens? is the call rejected? |
| 23:24.27 | [hC] | As far as i am aware, yes. |
| 23:25.19 | Strom_C | alright, thanks |
| 23:32.09 | De_Mon | WARNING[3730]: chan_sip.c:4029 set_destination: Can't find address for host '3149' |
| 23:32.21 | De_Mon | I can make calls, but can't receive them |
| 23:32.33 | De_Mon | sip show registry lists the correct address for the extension too. |
| 23:33.13 | Strom_C | when was the last time it registered? |
| 23:34.12 | De_Mon | no idea |
| 23:34.23 | De_Mon | the phone rings, just can't hear any talking |
| 23:34.38 | De_Mon | where would I check the last time it registered? |
| 23:37.24 | De_Mon | looks like about 2hrs ago according to the console |
| 23:38.57 | Strom_C | weird. |
| 23:39.02 | Strom_C | NAT? |
| 23:39.32 | De_Mon | yeah |
| 23:40.32 | De_Mon | it's a SPA941 |
| 23:48.44 | Strom_C | how much NAT? is the SPA behind NAT and the asterisk box on a public IP? |
| 23:54.58 | X-Rob | De_Mon, you have a 'host=3149' line. That's not correct. |
| 23:55.17 | X-Rob | it's host=dynamic or host=ip.address or host=valid.dns.name |
| 23:57.02 | *** join/#asterisk dalekurt (n=dalekurt@port0096-afo-adsl.cwjamaica.com) |
