| 00:00.24 | Keizer | Mine had to be updated to support sip |
| 00:00.30 | JayTee52 | ah |
| 00:01.17 | JayTee52 | ManxPower, I fixed my Grandstream MOH problem. It was a firmware bug with Silence Suppression. It would show as disabled even when it wasn't. |
| 00:03.19 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:03.47 | ManxPower | JayTee52: Welcome to the world of Grandstream. I figure it will be a week or two before you find a bug in your current firmware, etc. |
| 00:04.26 | ManxPower | I bet you could have bought a high end polycom for the amount of time you spent on the GS phone. |
| 00:04.59 | JayTee52 | ManxPower, it's ok. I think I've got my boss convinced to buy the Polycoms when I showed him a demo of the feature sets with Exchange UM and Active Directory. Anything that ties to Windows gets him all excited. |
| 00:07.00 | JayTee52 | but at least this gives us a less expensive option for deploying inexpensive phones to dirty areas like the elephant barn, the lion-baboon holding etc or our Dolphin building which has a high humidity and salt content. |
| 00:10.26 | TJNII | I'd use Grandstreams in the environments where you know there going to get killed and save the polycoms for more hospitible areas. |
| 00:10.35 | TJNII | But that may just be me..... |
| 00:10.39 | JayTee52 | that's my idea |
| 00:10.45 | TJNII | 10-4 |
| 00:11.09 | JayTee52 | the Polycoms will go to the executives and admin staff and the Grandstreams will go to our animal keepers in the biomes. |
| 00:12.12 | TJNII | A friend of mine sent me a link for a mil-spec SIP phone for field deployment, but I lost the link.... |
| 00:12.19 | TJNII | I think it was a Cisco, though. |
| 00:21.28 | *** join/#asterisk saftsack (n=saftsack@p4FC75C46.dip.t-dialin.net) |
| 00:24.46 | asdx | how can i know which codec is being used on a call |
| 00:24.56 | asdx | from the asterisk cli |
| 00:25.07 | jjshoe | show channel |
| 00:25.07 | drmessano | sip show channels |
| 00:25.19 | jjshoe | be sure to pass in the channel you want to know the codec of |
| 00:25.32 | asdx | nice, thanks |
| 00:27.22 | Keizer | Anyone else see this: WARNING[16014]: chan_sip.c:2751 retrans_pkt: Hanging up call 003094c4-06290004-17fa319a-7820d43b@192.168.200.201 - no reply to our critical packet |
| 00:29.10 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-45b7b17bc306537c) |
| 00:30.12 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 00:30.50 | jjshoe | Keizer I've never seen it, so unds like you're not getting a response to a critical sip packet, sip debug ip 192.168.200.201 |
| 00:32.33 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 00:32.56 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:32.56 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:37.35 | Keizer | jjshoe: That command doesn't work in the CLI |
| 00:37.40 | Keizer | I'm looking at the CLI |
| 00:37.48 | Keizer | It shows the two channels for the endpoints |
| 00:37.57 | Keizer | But no audio goes between them |
| 00:38.10 | Keizer | When Asterisk hangs up the CISCO phone doesn't even know it |
| 00:40.03 | JayTee52 | are the endpoints on separate VLANs? |
| 00:42.58 | Keizer | JayTee52: They might be |
| 00:43.14 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 00:43.56 | JayTee52 | that could be a problem, another thing is to check which port each phone is configured for RTP, default if 5040. |
| 00:46.05 | JayTee52 | are either phone displaying any errors? are you getting any console errors? |
| 00:46.51 | JayTee52 | other than that earlier error you posted |
| 00:46.55 | JayTee52 | or warning |
| 00:47.24 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 00:47.53 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 00:50.02 | JayTee52 | Keizer, is the 192.168.200.201 phone the Cisco? |
| 00:55.54 | *** join/#asterisk kyron (n=kyron@modemcable086.140-70-69.static.videotron.ca) |
| 00:56.27 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 00:58.37 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:58.37 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 01:05.03 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 01:05.17 | *** join/#asterisk JerJer[mobile] (n=jj@45.210.254.244) |
| 01:06.03 | lirakis | is there a way to get "CDR(start)}","${CDR(answer)}","${CDR(end)}" from cdr_custom .. into cdr_mysql with * 1.4 |
| 01:07.25 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 01:08.50 | *** part/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 01:09.18 | Jumpie | lol i just got my first telemarketer spam on my test DID |
| 01:09.27 | JerJer[mobile] | lirakis: its already in there |
| 01:10.28 | lirakis | JerJer[mobile]: hrmm |
| 01:10.33 | jjshoe | Juggie I've gotten telemarketer calls within an hour. |
| 01:10.37 | lirakis | okay .. i guess i just need to define the fields? |
| 01:10.37 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 01:11.05 | JerJer[mobile] | here is a fun situation.... Avaya SIP Phone and Asterisk on the same physical network -> everything works totally fine |
| 01:11.38 | lirakis | JerJer[mobile]: do you know of a copy paste table structure for 1.4 ? .. so i dont have to type it all out .. im sure voip info's is way old |
| 01:11.53 | JerJer[mobile] | simply move the phone to a different subnet -> Asterisk responds with a 503 - no other changes whatsoerver, no other log messages |
| 01:11.54 | lirakis | JerJer[mobile]: different networks .. no audio ? |
| 01:11.59 | lirakis | JerJer[mobile]: oh ok |
| 01:12.10 | lirakis | JerJer[mobile]: sorry .. i had a similar problem yesterday |
| 01:12.31 | JerJer[mobile] | lirakis: nope sorry, I do not use cdr_mysql |
| 01:12.40 | *** join/#asterisk sack (n=sack@193.Red-88-24-154.staticIP.rima-tde.net) |
| 01:13.14 | lirakis | JerJer[mobile]: what about sip debug |
| 01:14.00 | JerJer[mobile] | all that i see is asterisk sending back a 503 - no other verbose, notice, error or debug messags |
| 01:14.10 | *** join/#asterisk disgrntld81 (n=asdf@CPE-69-23-215-92.wi.res.rr.com) |
| 01:14.50 | lirakis | <PROTECTED> |
| 01:14.59 | lirakis | <PROTECTED> |
| 01:15.01 | JerJer[mobile] | its on - that's how i know asterisk is sending a 503 |
| 01:15.18 | lirakis | JerJer[mobile]: wasnt sure if you were reading it off the phone |
| 01:15.39 | lirakis | JerJer[mobile]: does the phone register properly |
| 01:15.59 | JerJer[mobile] | when on the same network as asterisk, yes |
| 01:16.29 | JerJer[mobile] | I am at interop now - testing asterisk and openser against many different vendor's equipment |
| 01:17.21 | JerJer[mobile] | the situation in question here is dealing with NAC - network access control - so there is 802.1x authentication happening, but all that is very much lower level than what asterisk should care about |
| 01:17.26 | *** join/#asterisk paci` (n=paci@cpe-075-182-072-065.nc.res.rr.com) |
| 01:17.36 | paci` | exten => 680,1,Dial(100@roflwaffle.com) |
| 01:17.38 | paci` | work/no work? |
| 01:17.45 | JerJer[mobile] | no work |
| 01:17.51 | paci` | How to |
| 01:17.58 | lirakis | paci`: you are missing the channel type |
| 01:18.00 | JerJer[mobile] | add a technology |
| 01:18.07 | paci` | Er |
| 01:18.11 | paci` | Hmm? |
| 01:18.13 | lirakis | Dial(SIP/blah@blalskdf.com) |
| 01:18.17 | JerJer[mobile] | d'oh |
| 01:18.20 | JerJer[mobile] | :) |
| 01:18.36 | paci` | oh |
| 01:19.47 | lirakis | paci`: plus you probably need to define 100@roflwaffle.com as a peer .. not sure though... for out of network (off asterisk ) diaing you often have to use transfer ... but if you define it as peer 100 .. you can just Dial(SIP/100) |
| 01:19.56 | JerJer[mobile] | lirakis: no |
| 01:20.24 | JerJer[mobile] | it will send a sip request to that username and dns name |
| 01:20.27 | lirakis | JerJer[mobile]: mmkay.. i remember doing some openser work that was stateless msging .. asterisk didnt like dial .. so i had to use the transfer command |
| 01:21.25 | JerJer[mobile] | having a peer defined would give you more control, yes |
| 01:21.51 | JerJer[mobile] | having and using a peer that is |
| 01:25.15 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 01:25.40 | drmessano | roflwoffle? Is roflcopter busy? |
| 01:25.56 | JerJer[mobile] | man this is so strange - i don't get why asterisk would even care about the network |
| 01:28.41 | *** join/#asterisk sack (n=sack@193.Red-88-24-154.staticIP.rima-tde.net) |
| 01:30.33 | lmadsen | JerJer[mobile]: zup zup y0 |
| 01:30.44 | lirakis | JerJer[mobile]: i thought that maybe it was netfilter related.. but if so, you wouldnt see the request |
| 01:30.52 | JerJer[mobile] | hangin in Belmont, CA at interop |
| 01:31.44 | JerJer[mobile] | we got openser SIMPLE presence working swimmingly |
| 01:32.00 | JerJer[mobile] | but simple -> xmpp isn't cooperating atm |
| 01:32.08 | *** join/#asterisk machpo (n=mcp@ketone.jawfish.com) |
| 01:37.53 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 01:38.08 | JerJer[mobile] | ugh |
| 01:38.30 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:46.28 | *** join/#asterisk JerJer[mobile] (n=jj@45.210.254.244) |
| 01:46.52 | JerJer[mobile] | d'oh - someone tripped over the power cable |
| 01:47.51 | paci` | how would I send someone back to the main menu |
| 01:49.12 | JerJer[mobile] | gotyo |
| 01:49.16 | JerJer[mobile] | goto* |
| 01:49.30 | paci` | goto what? |
| 01:50.41 | denon | goto(context,s,1) or somethin |
| 01:50.45 | denon | show application goto |
| 01:52.04 | *** join/#asterisk micander (n=Michael_@ip70-181-134-119.sd.sd.cox.net) |
| 01:52.42 | *** join/#asterisk Kyoshi (i=whoa@pool-71-167-117-15.nycmny.fios.verizon.net) |
| 02:00.46 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 02:01.11 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 02:05.45 | *** join/#asterisk bintut (n=bintut@bb116-15-101-181.singnet.com.sg) |
| 02:06.28 | Keizer | Anyone here used OpenSER? |
| 02:06.52 | Keizer | The Asterisk book says it makes for a good SIP proxy |
| 02:07.33 | Keizer | Documentation I have seen requires modifying the Asterisk source code to integrate them. But the version of Asterisk was 1.2. Does anyone know if you still have to modify the source? |
| 02:08.27 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 02:11.47 | husimon | um, what sets the time on the phones with asterisk? |
| 02:11.54 | husimon | shouldn't it be the asterisk server? |
| 02:12.46 | husimon | or is that a dhcp set option |
| 02:12.51 | plik | husimon: depends on the phone, and whe |
| 02:13.03 | husimon | because for some odd reason my phones are all an hour off |
| 02:13.07 | plik | whether you have ntp set up on the asterisk server |
| 02:13.08 | husimon | and the asterisk server is correct |
| 02:13.21 | plik | ntp can be set by dhcp too |
| 02:13.28 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.177.153) |
| 02:13.30 | husimon | the ntp server is correct too |
| 02:13.36 | husimon | rather, the time on it is |
| 02:13.53 | husimon | probably fucking daylight savings timed me |
| 02:13.54 | plik | hmmm maybe there's a manual over-ride for summer time on/off |
| 02:13.59 | husimon | even though i'm in hawaii |
| 02:14.23 | *** join/#asterisk _pepo_ (n=Pepo@190.154.62.145) |
| 02:17.51 | *** part/#asterisk _pepo_ (n=Pepo@190.154.62.145) |
| 02:24.54 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-131-138.dsl.stlsmo.sbcglobal.net) |
| 02:31.52 | SwK | anyone have any contacts over at sipphone? |
| 02:40.35 | *** part/#asterisk paci` (n=paci@cpe-075-182-072-065.nc.res.rr.com) |
| 02:48.23 | *** join/#asterisk wulfy814 (n=askme@static-acs-24-154-122-15.zoominternet.net) |
| 02:48.44 | wulfy814 | evening folks! I think I have a simple one-- |
| 02:48.59 | wulfy814 | I have a viking speakerphone connected via Sangoma A200 FXS port |
| 02:49.39 | wulfy814 | and when I get an incoming call on one of the FXO ports, if callerID can't be read it's defaulting to "200" which is the callerID of the Viking |
| 02:59.32 | SwK | wulfy814, as in wulf consulting? |
| 03:00.43 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 03:05.50 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 03:08.36 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-122-158.PHNX.QWEST.NET) |
| 03:09.11 | asdx | is there a good sip/iax cli client? |
| 03:09.17 | wulfy814 | SwK yep |
| 03:09.28 | asdx | thx |
| 03:10.08 | asdx | ah oops |
| 03:10.40 | wulfy814 | http://www.pastebin.ca/977851 <---is my zapata.conf |
| 03:11.53 | bintut | i'm wondering why when i have a call session through the pots network incoming/outgoing on my Zap/4 channel, i got a very noisy sound or wink.. :( any idea why i'm getting this? |
| 03:20.15 | bintut | anyone has an idea about hearing a noisy sound when dealing with POTS network using the digium's dev kit card? info about my setup is at http://www.privatepaste.com/110mcguapk |
| 03:26.20 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:28.31 | *** join/#asterisk bl4q (n=Bl@dslb-088-066-234-107.pools.arcor-ip.net) |
| 03:28.31 | drmessano | cli client? |
| 03:28.43 | bintut | drmessano: you're asking me? |
| 03:29.41 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.221) |
| 03:30.25 | drmessano | [23:08] <asdx> is there a good sip/iax cli client? |
| 03:30.27 | drmessano | No |
| 03:30.40 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 03:31.38 | asdx | command line interface |
| 03:31.39 | asdx | :( |
| 03:32.01 | drmessano | I know what cli means |
| 03:32.47 | drmessano | I wasn't answering "no" to you, asdx, I was answering "no" to bintut |
| 03:33.10 | drmessano | the idea of a cli client intrigues me |
| 03:35.29 | *** join/#asterisk LuChris (n=inc@bb116-15-24-217.singnet.com.sg) |
| 03:36.07 | LuChris | hi i am new and like to play with asterisk .. where should i go for beginner? |
| 03:37.34 | drmessano | ~book |
| 03:37.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:37.43 | drmessano | Start off by reading the book |
| 03:37.54 | drmessano | It will teach you a lot |
| 03:39.15 | jblack | bah. books are for people that can read. |
| 03:39.39 | jblack | well, except for the ones with the pictures of the pretty, nekkid womenfolk |
| 03:42.14 | *** join/#asterisk LuChris (n=inc@bb116-15-24-217.singnet.com.sg) |
| 03:42.20 | LuChris | sorry got dc |
| 03:42.28 | LuChris | i meant if i want to install it, where should i go to .. |
| 03:48.05 | drmessano | Get the book |
| 03:48.29 | drmessano | It's a free PDF.. and if you like it, you can buy the treekiller copy and give the author some well deserved credit |
| 03:48.43 | drmessano | It will walk you through everything |
| 03:49.10 | drmessano | That's not a brush off.. but the book is VERY good |
| 03:49.58 | LuChris | thanks alot! |
| 03:50.15 | drmessano | Anytime |
| 03:50.30 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 03:50.30 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 03:51.08 | LuChris | drmessano can i pm u? |
| 03:51.17 | LuChris | http://tfot.leifmadsen.com/ is down |
| 03:52.06 | drmessano | no no |
| 03:52.11 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
| 03:52.13 | drmessano | http://downloads.oreilly.com/books/9780596510480.pdf |
| 03:52.22 | drmessano | That's the link for the PDF |
| 03:52.40 | drmessano | It was in that bot reply |
| 03:53.07 | bintut | <drmessano> cli client? |
| 03:53.15 | bintut | <bintut> drmessano: you're asking me? |
| 03:53.49 | drmessano | I responded.. does your scroll not work? |
| 03:54.07 | drmessano | [23:32] <drmessano> I wasn't answering "no" to you, asdx, I was answering "no" to bintut |
| 03:54.29 | LuChris | i downloaded that. thanks.. does not have reader installed, so i thought i can use the html site |
| 03:54.30 | bintut | drmessano: i see.. |
| 03:54.49 | bintut | LuChris: http://www.the-asterisk-book.com/unstable/ |
| 03:55.16 | LuChris | anyway, if i do not have a sip, can i use this? and if in my country, there is no sip provider, how to go about setting up a sip machine? |
| 03:55.38 | drmessano | LuChris: It's in the book |
| 03:55.45 | drmessano | Get it and read |
| 03:56.14 | LuChris | ok .. thanks |
| 03:57.23 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:57.42 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:02.40 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 04:03.30 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 04:08.30 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 04:08.40 | bintut | why is it that running 'sip show peers' gives me the status "UNMONITORED" on the registered x-lite phones? |
| 04:09.29 | *** join/#asterisk Charun (n=Me@eth244.wa.adsl.internode.on.net) |
| 04:09.53 | jblack | try putting "qualified=yes" in your sip.conf |
| 04:10.16 | Charun | Hi, is there a way to check what timing source asterisk is currently using? |
| 04:10.20 | jblack | pardon, qualify=yes |
| 04:11.58 | bintut | jblack: ok |
| 04:14.16 | bintut | jblack: got it working. thanks.. :) |
| 04:14.53 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 04:15.02 | bintut | jblack: any idea about hearing a noisy sound when dealing with POTS network using the digium's dev kit card? info about my setup is at http://www.privatepaste.com/110mcguapk |
| 04:15.42 | bintut | jblack: i hear a noisy sound like electrical scratch or something whenever my pots line is involve.. :( |
| 04:31.00 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582637.dsl.bell.ca) |
| 04:31.17 | *** join/#asterisk kerx (n=pmessri@adsl-69-105-53-95.dsl.irvnca.pacbell.net) |
| 04:31.33 | drmessano | Your paste is invalid |
| 04:31.36 | kerx | hi everyone, new to voip really, just a quick question here. |
| 04:31.53 | kerx | i want to get voip setup for business, on a broad perspective, every user in the office will have a soft phone and an extension |
| 04:32.15 | kerx | i want to signup with broadvoice, but i can't tell if they offer multiple calls to route to different extensions |
| 04:32.27 | kerx | so that we can have multiple phone calls established at the same time through our own device (asterisk) |
| 04:32.41 | kerx | not even sure what the terminology for this is in the POTS world (maybe channels?) |
| 04:32.57 | drmessano | channels is correct |
| 04:33.08 | drmessano | You need to check with BV |
| 04:33.19 | kerx | i see, so if i were to go with either a POTS, or T1(digital), or even a VOIP provider I have to look for a feature with multiple channels? |
| 04:33.32 | jblack | what you do with the call, once you get it, is up to you. The number of simultaneous channels that you get varies from provider to provider, with a tendancy of # of channels following cost. |
| 04:33.45 | kerx | jblack: roger |
| 04:33.59 | kerx | makes sense, but even on broad voice's features list: |
| 04:33.59 | kerx | http://www.broadvoice.com/features.html |
| 04:34.12 | kerx | you can't tell if they offer this feature |
| 04:34.23 | jblack | who says I can't? |
| 04:34.45 | drmessano | Well, I don't see it listed |
| 04:34.50 | kerx | not sure about you, i was implying an average user like myself browsing the internet would have a hard time seeing this on there website because it's not listed |
| 04:34.53 | kerx | right :) |
| 04:35.11 | kerx | well ill ask them tomorrow morning |
| 04:35.14 | kerx | another quick question |
| 04:35.21 | kerx | i have an office with 20 users |
| 04:35.31 | kerx | we have DSL :O |
| 04:35.39 | kerx | what do you guys think? |
| 04:35.51 | drmessano | You're gonna need a bigger pipe |
| 04:36.02 | jblack | generally, flat rate plans are limited to 2 concurrent calls. |
| 04:36.05 | kerx | we have the elite package so we have 3-6Mbps/0.7Kbps down/up |
| 04:36.13 | jblack | Pay as you go, are typically as many as you use. |
| 04:36.23 | drmessano | .7Kbps = eek |
| 04:36.25 | kerx | jblack: roger, ill give them a ring tomorrow |
| 04:36.35 | jblack | if you're looking at 20 users, then you may want to consider a T1 from a local provider in part of your assessment. |
| 04:36.42 | kerx | drmessano: i was wrong, i meant KBps, so you could roughly say 1Mbps for up |
| 04:36.48 | jblack | I take it you're a callcenter |
| 04:36.50 | kerx | T1 = $$$ up the wazoo |
| 04:36.53 | kerx | not at all |
| 04:36.55 | kerx | software company |
| 04:37.00 | kerx | i wouldn't consider having more than 2 phone calls really |
| 04:37.01 | drmessano | 1Mbps just the same |
| 04:37.08 | drmessano | Ah |
| 04:37.09 | kerx | but... |
| 04:37.14 | jblack | yeah, they can run about $600 a month. It depends on how many calls you're pushing. |
| 04:37.14 | drmessano | So no more than 2 concurrent? |
| 04:37.23 | kerx | lets say 2 concurrent calls |
| 04:37.33 | jblack | I doubt you'll only be doing 2 with 20 users. 5 is more like it. |
| 04:37.34 | kerx | how bad of a quality would you estimate? |
| 04:37.43 | drmessano | If you QoS the crap out of it, you could get by |
| 04:37.45 | kerx | jblack: were a software company :) |
| 04:37.54 | kerx | developers hate talking on the phone |
| 04:38.06 | kerx | im trying to take a $550 telco bill |
| 04:38.10 | kerx | and make it $100 tops :) |
| 04:38.21 | jblack | And? I'm a software developer. I spend as much as 1-2 hours a day on the phone for large clients. |
| 04:38.28 | jblack | each. |
| 04:38.29 | kerx | thats true |
| 04:38.40 | jblack | My record is 14 hours |
| 04:38.45 | kerx | well, most of the guys here dont talk to an external client |
| 04:38.51 | kerx | its weird i know |
| 04:38.54 | kerx | we have conference calls |
| 04:38.58 | kerx | that may last 3-4 hours |
| 04:39.00 | jblack | well, you know your business. |
| 04:39.02 | kerx | that will tie up a single connection |
| 04:39.14 | kerx | im mainly worried about 1 connection - 2 connections |
| 04:39.20 | jblack | Ok... how about this. |
| 04:39.28 | kerx | if we experience problems w/ 1 connection already, then ... shit .... |
| 04:39.37 | jblack | IPKall, free unlimited incoming (washington number).. $0.00 a month |
| 04:39.51 | jblack | 1 cahnnel or 50. doesn't matter. nothing. |
| 04:40.19 | jblack | Outgoing, look at pay as you go.. typically 1 to 3 cents per minute, per channel, rounded to six seconds or less. |
| 04:40.43 | jblack | If you really have people that don't like to talk, you could get your phone costs down to... ten bucks a month. |
| 04:40.51 | drmessano | IPKALL rocks |
| 04:41.14 | kerx | i see, nice |
| 04:41.19 | jblack | FOr outgoing, I prefer callwithus, but everyone has their favorite. |
| 04:41.26 | kerx | i was really just considering: http://www.broadvoice.com/rateplans_unlimited_busi.html |
| 04:41.31 | kerx | $35.42/month |
| 04:41.32 | kerx | unlimited |
| 04:41.54 | jblack | do you want our advice, or do you want to give us advice? |
| 04:42.01 | jblack | because I don't remember asking you for suggestions. |
| 04:42.01 | kerx | no, im sorry, i was giving you feedback |
| 04:42.05 | kerx | roger |
| 04:42.07 | kerx | shuts up |
| 04:42.17 | jblack | Ok. Let's do this... |
| 04:42.33 | jblack | How much time per call do your callers make, on average, and calls a day? |
| 04:42.39 | jblack | 20 * x * x * 30 |
| 04:43.04 | kerx | i have the phone bill, one sec |
| 04:43.08 | jblack | perfect |
| 04:43.23 | kerx | 3 lines |
| 04:43.28 | kerx | line 1: |
| 04:43.55 | jblack | Assuming broadvoice is two channels, you'll have to use more than 1400 minutes a month total. |
| 04:43.57 | kerx | - longest unit: 4.49 |
| 04:44.18 | kerx | - shortest unit: :30 |
| 04:44.26 | kerx | line 2: |
| 04:44.44 | kerx | - longest unit: 85:56 |
| 04:44.45 | jblack | It's total minutes that count. ;) |
| 04:44.51 | kerx | isn't that unit? |
| 04:45.02 | jblack | That's your longest call, and your shortest call. ;) |
| 04:45.10 | kerx | yeah, i can't find that in the bill |
| 04:45.24 | kerx | ill give you a total call after |
| 04:45.33 | jblack | Ok, can you figure out an average call length, and total # of calls? |
| 04:45.40 | kerx | heh |
| 04:45.50 | kerx | fawk, ok let me sum up |
| 04:45.56 | kerx | your right, makes more sense |
| 04:46.19 | kerx | conference room is main one, so let me start with that |
| 04:46.26 | jblack | Based on what you said before, keeping costs down, you have to figure out what you're already using. |
| 04:47.08 | jblack | Flat rate services tend to be a field goal maneuver. Less than this, and flat rate is cheaper. More than that, and flat rate is no longer an option at all. |
| 04:47.35 | jblack | pardon, less than this, and pay as you go is cheaper. More than that, and you can't do flat rate at all. |
| 04:47.56 | jblack | If you can fit in between the two, flat rate is often a little cheaper. Rarely cheaper enough to deal with the headache, imho |
| 04:48.10 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 04:48.11 | kerx | conference room total: 11.4 |
| 04:48.20 | kerx | conference room total calls: 13 |
| 04:48.40 | kerx | so 11.4 hours with 13 calls |
| 04:48.57 | jblack | Ok, So, that's 690 minutes. |
| 04:49.05 | kerx | 52 minutes each call |
| 04:50.04 | jblack | with a provider like callwithus, that would have cost you....eight and a half bucks |
| 04:50.13 | jblack | For all 11+ hours. |
| 04:51.03 | kerx | not bad |
| 04:51.05 | jblack | You probably spend more than that on a coffee run |
| 04:51.10 | kerx | shit, thats awesome |
| 04:51.12 | kerx | :) |
| 04:51.12 | jblack | That's what 1.25 cents a minute works out to. |
| 04:51.18 | kerx | well, theres 2 things |
| 04:51.27 | kerx | 1. voip quality w/ our DSL |
| 04:51.36 | kerx | 2. the phone number (can we keep it as we transfer providers) ? |
| 04:52.07 | jblack | 1. If you have enough bandwidth, call performance will be superior to traditional POTS. |
| 04:52.13 | *** part/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 04:52.16 | bintut | drmessano: you're telling me my paste was invalid? |
| 04:52.26 | jblack | 2. That depends upon the provider. Generally speaking, companies that do porting tend to charge a little more per minute. |
| 04:52.34 | kerx | k |
| 04:52.58 | jblack | if you remember, I mentioned ipkall, which is _free_ unlimited incoming, but only offers washinton state numbers. No porting there. |
| 04:53.01 | kerx | to refer back to #1, what type of bandwidth would you say has this type of performance ? |
| 04:53.13 | jblack | Callwithus, which is among the cheaper (but not the cheapest), doesn't offer porting (or e911 for that matter) |
| 04:53.35 | jblack | That depends upon the codec. You should be able to google those. ulaw is 64kbit. gsm is... oh, 8 or 16 or something. |
| 04:54.00 | jblack | maybe it's 24. |
| 04:54.09 | jblack | anyways, are you doing nothing but voip on the dsl line? |
| 04:54.10 | drmessano | bintut: Yes, your paste is invalid |
| 04:54.58 | jblack | cause a dsl line that's also being used for web browsing (split amonst 20 people, no less), isn't likely to give a useable result. |
| 04:54.59 | kerx | jblack: well, i can QoS the line, but it acts as a dedicated server line, where we would probably have 1-2 developers at most ssh'ed in during the 9-5 work hours |
| 04:55.12 | kerx | we have 2 DSL's |
| 04:55.16 | bintut | call performance over the internet is not absolutely reliable.. you don't have to consider only on your side but on the next hops that your call will be passing through until such time it reaches to the target callee |
| 04:55.18 | kerx | 1 for the split 20 web traffic |
| 04:55.23 | kerx | 1 for the *above* mentioned |
| 04:55.25 | bintut | drmessano: maybe it already expired.. |
| 04:55.36 | bintut | drmessano: i will paste again |
| 04:55.51 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 04:56.14 | jblack | I'm all for saving money, but there's a point when one becomes pennywise and pound foolish. |
| 04:56.31 | kerx | what do you guys think about taking one of our POTS lines and getting a converter using asterisk ? |
| 04:56.32 | jblack | By the time you're dickering over "$20 a month" versus "$22 a month".... |
| 04:56.50 | kerx | jblack: really if i can take $550 to $10 |
| 04:56.55 | kerx | it would be worth this time |
| 04:57.04 | kerx | if im taking $550 to $350, i wont consider it at this time |
| 04:57.15 | kerx | but another thing is the convenience and the advanced features of using asterisk |
| 04:57.25 | kerx | much more than just 3 plain old POTS lines |
| 04:57.36 | jblack | Based on what you've told me, if the dsl line is good enough, you're looking at about $25 - $60 a month. |
| 04:57.50 | kerx | yeah, that would be worth the time |
| 04:57.59 | jblack | but don't underestimate how many concurrent lines you need. Someone may need a line to dial 911 |
| 04:58.24 | kerx | obvcourse, the minute u mentioned that i realized most likely, legally as a business you need to have 911 available |
| 04:58.37 | kerx | so i would be considering that while looking for a provider |
| 04:58.44 | kerx | but what do you think about my question above |
| 04:58.53 | kerx | taking one of our POTS lines and converting it into asterisk ? |
| 05:00.52 | bintut | drmessano: kindly check this out => http://www.privatepaste.com/911eZrcYqT |
| 05:02.09 | *** join/#asterisk Nuonec (n=nuonec@124.43.51.233) |
| 05:02.38 | Nuonec | Hi All |
| 05:02.59 | bintut | kerx: i suggest you use g.729a codec over the internet if your itsp supports it |
| 05:04.12 | kerx | thanks, not sure about codecs, not really at this point yet |
| 05:04.15 | bintut | there's a big difference when it comes to bandwidth consumption per call session using g.711u/a codec compared to g.729a codec |
| 05:04.17 | kerx | but i will keep this in mind |
| 05:04.24 | kerx | oh i see |
| 05:04.30 | jblack | I think that's a choice. |
| 05:04.44 | jblack | make yourself a spreadsheet, and do comparisons. |
| 05:04.47 | kerx | is there any kind of sample bandwidth numbers people have done for a phone call ? |
| 05:04.54 | bintut | g.729u/a uses at least 64kbps per call, not yet included the headers itself |
| 05:04.56 | kerx | like a single voip call with g.729a |
| 05:05.01 | Nuonec | I am a newbie and want to experimment with Asterisk. I want to start with a basic hardware which does not cost me much. What are ur suggestions? |
| 05:05.14 | jblack | being bandwidth starved, assuming you're not faxing, look up gsm. |
| 05:05.26 | jblack | It's reasonably good, and saves a _lot_ of bandwidth |
| 05:05.28 | *** join/#asterisk tinapa (n=niwre@67.159.178.21) |
| 05:05.38 | bintut | whereas g.729a codec uses at least 8kbps (to 14kbps i think) only per call session |
| 05:05.50 | kerx | shit |
| 05:06.02 | tinapa | hi there, how can i check why my php agi stalles on connecting to asterisk manager.conf? |
| 05:06.06 | kerx | still pretty high for our DSL upload |
| 05:06.10 | jblack | gsm is 13.2kb/sec. |
| 05:06.12 | kerx | i see the problem there |
| 05:06.17 | kerx | not bad though |
| 05:06.30 | bintut | g.729a license is a one time purchase per device/machine |
| 05:06.39 | bintut | whereas bandwidth is a recurring cost |
| 05:06.50 | jblack | which for a 768 kilobit line works out to up to... probably around 10 calls at once. |
| 05:07.42 | kerx | not bad, so for a single call i should be fine, as long as I do QoS |
| 05:07.52 | kerx | so a sudden file transfer doesn't drop the call |
| 05:07.54 | jblack | I have no idea. |
| 05:08.14 | kerx | but why do u have to purchase a license for g.729a ? |
| 05:08.34 | jblack | I said gsm, not g.729a. And because there's a patent for it |
| 05:08.49 | Nuonec | what is the Asterisk compatible cheapest analog telephony devices that I can use to start with. |
| 05:08.51 | bintut | so, using a g.729a codec will only require you from 80kbps to 140kbps assured bandwidth with 10 concurrent calls |
| 05:09.29 | jblack | Heh. you can't fit 80x10 on a 768 kilobit line. |
| 05:09.33 | bintut | Nuonec: sipura ata? |
| 05:09.34 | jblack | Perhaps 5 |
| 05:10.08 | Nuonec | bintut: ok I will check that. thanks |
| 05:10.13 | jblack | once a link becomes much more than 1/2 loaded, latency starts rocketing up. |
| 05:11.13 | jblack | unless you control both sides of the link |
| 05:11.19 | bintut | Nuonec: yw. :) |
| 05:12.15 | kerx | well, i think you guys have given me more than enough info :) |
| 05:12.21 | kerx | like i said, i expect max 2 connections |
| 05:12.24 | LuChris | how many channels can i support if i am on a dsl 10mbps down and 1mpbs down? |
| 05:12.26 | bintut | anyone here can enlighten me regarding the noise that i am hearing when the pots line is involve in a call session through my zap/4 channel? |
| 05:12.32 | kerx | based on e911 stuff you mentioned jblack |
| 05:12.40 | tinapa | what does this command do? "manager show eventq" |
| 05:12.41 | kerx | i might even consider keeping 1 single POTS and converting |
| 05:12.52 | kerx | allows us to drop 2 phone lines, and keep 1 |
| 05:13.00 | kerx | we can gain the advanced asterisk PBX features |
| 05:13.09 | kerx | we should be good :) |
| 05:13.15 | kerx | now to go off and look for the cost of some of those cards |
| 05:13.47 | jblack | consider ATAs as well. |
| 05:13.57 | bintut | http://www.openvox.com.cn/ :D |
| 05:14.15 | jblack | heh. I might buy food from china, but never tech. :) |
| 05:14.38 | bintut | http://www.snom.com/ |
| 05:15.53 | bintut | the info of my digium dev kit tdm card is at http://www.privatepaste.com/911eZrcYqT |
| 05:20.06 | kerx | cool thanks |
| 05:21.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:37.29 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 05:37.29 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 05:38.08 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:38.08 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 05:45.54 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:47.55 | *** join/#asterisk CCFL_Man2 (i=860c6200@pool-70-105-206-237.scr.east.verizon.net) |
| 05:51.17 | LuChris | what linux package do i need to enable voice mail? i am running on centos. |
| 05:51.34 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:51.34 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 05:53.43 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 05:53.43 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 05:55.09 | *** join/#asterisk watchy (n=watchy@h200.176.255.206.cable.cmdn.cablelynx.com) |
| 05:55.55 | watchy | anyway to convert a frac t1 to 2 frac t1s? |
| 05:57.51 | lmadsen | talk to the phone comp? |
| 05:58.08 | watchy | hrm |
| 05:58.21 | watchy | well thing is, they are bringin in a frac T1 with 12 lines. |
| 05:58.36 | watchy | i just wanna give 6 lines to my PBX and 6 to my Portmaster |
| 06:00.20 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 06:00.20 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 06:06.32 | *** join/#asterisk Sir0x (n=cmayer@p5B084951.dip.t-dialin.net) |
| 06:07.19 | *** join/#asterisk [hC] (n=hardcore@66.119.187.130) |
| 06:08.32 | LuChris | trying to compile asterisk, got into an error. make[1]: *** [ilbc/libilbc.a] Error 2 |
| 06:08.34 | LuChris | any idea? |
| 06:10.15 | watchy | you google it? |
| 06:11.28 | LuChris | yes .. need to diff |
| 06:11.58 | LuChris | dont understand it |
| 06:12.22 | watchy | gimme the url |
| 06:12.48 | LuChris | http://lists.digium.com/pipermail/svn-commits/2004-July/002718.html |
| 06:13.36 | watchy | what version u trying to compile |
| 06:13.52 | LuChris | 1.4.19 |
| 06:14.01 | watchy | hmm |
| 06:14.06 | watchy | version of linux? |
| 06:14.33 | watchy | that url u gave me was from a version of * from like |
| 06:14.34 | LuChris | centos 5.1 |
| 06:14.35 | watchy | 2004 |
| 06:14.49 | watchy | so thats not a useful link |
| 06:15.20 | LuChris | there is a thread on asteriskguru.com but there is a mysql error |
| 06:15.37 | LuChris | http://www.asteriskguru.com/archives/asterisk-cvs-asterisk-codecs-makefile122123-vt53064.html |
| 06:15.41 | watchy | haha how ghetto |
| 06:15.54 | watchy | haha |
| 06:16.03 | watchy | jesus how hards it to keep phpbb running |
| 06:16.15 | Juggie | LuChris, whats the full error |
| 06:16.30 | watchy | yea that would also help |
| 06:16.31 | LuChris | i need to paste like 3 lines.. sorry if i flooded you guys |
| 06:16.35 | watchy | go for it |
| 06:16.38 | LuChris | make[2]: *** No rule to make target `anaFilter.o', needed by `libilbc.a'. Stop. |
| 06:16.38 | LuChris | make[1]: *** [ilbc/libilbc.a] Error 2 |
| 06:16.38 | LuChris | make: *** [codecs] Error 2 |
| 06:16.44 | watchy | i think everyone else is aslee |
| 06:16.44 | watchy | p |
| 06:17.07 | LuChris | hope so |
| 06:17.13 | Juggie | can you please pastebin.ca the entire thing |
| 06:17.41 | Juggie | but to get past this problem simply, just do a make menuconfig and disable ilbc |
| 06:17.50 | Juggie | in the codecs part |
| 06:18.07 | LuChris | hmmm if i disable it, i dont have any codecs? |
| 06:18.13 | Juggie | no |
| 06:18.15 | Juggie | just not ilbc |
| 06:18.19 | Juggie | which you will probally not use |
| 06:18.28 | Juggie | theres a problem with that codec currentally, long story. |
| 06:18.54 | Juggie | licenseing issue, it was removed from asterisk, but now the asterisk make file downloads the source when you run make... it doesnt work as well |
| 06:18.54 | LuChris | ok .. in that case, i will disable it |
| 06:18.56 | Juggie | its a long story |
| 06:19.03 | Juggie | just disable that perticular codec and try again |
| 06:19.48 | LuChris | btw, i ran into some issue regarding voice mail .. odbc and imap can conflicting each other.. how do i resolve this? and which should i use? imap or odbc? |
| 06:20.07 | LuChris | are conflicting* |
| 06:20.07 | Juggie | odbc is more mature |
| 06:20.13 | *** join/#asterisk Kyoshi (i=whoa@pool-71-167-117-15.nycmny.fios.verizon.net) |
| 06:20.28 | LuChris | i do not have any imap install, but it is conflicting for odbc |
| 06:22.04 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 06:22.19 | watchy | is putting voicemal in imap a good idea? |
| 06:23.28 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 06:23.29 | Al_WinKiller | hi guys,, anybody around ? |
| 06:23.44 | Al_WinKiller | or everybody is snoring ? |
| 06:23.48 | Al_WinKiller | :D |
| 06:23.56 | LuChris | auto downloading a 14mb of wav files |
| 06:24.26 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 06:26.04 | Al_WinKiller | I am facing a problem , got asterisk with about 30 clients registered on it,,, and some phone start ringing by its self ( see nothing on cli ), can't figure it out,, have someone face this kind of problem ? |
| 06:26.27 | watchy | what kinda phones |
| 06:26.34 | watchy | you turn on sip debug? |
| 06:26.51 | Al_WinKiller | and that happened mostly in the night , phones are panasonic |
| 06:27.24 | Al_WinKiller | I got some cisco ip phones (7960), they are ok |
| 06:27.33 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:28.27 | Al_WinKiller | yes, on debug I see calls only, no error msgs |
| 06:28.50 | Al_WinKiller | and I use alsy asterisk flash operator panel |
| 06:29.59 | Al_WinKiller | ye,, see also some error msg about noise incomplete on client ( calls go via cisco 5350 ),, but I think it is not the problem |
| 06:32.37 | watchy | no idea. very strange |
| 06:32.46 | Al_WinKiller | any suggestion ? ( for me this problem sounds very stupid,,, but it is still a problem, I have to solve it ) |
| 06:33.09 | Al_WinKiller | yep, agree with you |
| 06:33.18 | watchy | you try new firmware? |
| 06:34.00 | Al_WinKiller | on voice gw ? (I use alvarion with last firmware (SIP version, cuz they come with H323 installed)) |
| 06:35.18 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 06:35.42 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:36.18 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 06:44.28 | *** join/#asterisk mikefut (n=mikefut@lan.integra-media.com) |
| 06:45.43 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 06:45.54 | *** part/#asterisk mikefut (n=mikefut@lan.integra-media.com) |
| 06:51.03 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 06:51.32 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.43) |
| 06:57.14 | DarKnesS_WolF | ~nat |
| 06:57.15 | jbot | methinks nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 06:57.29 | DarKnesS_WolF | ~nat |
| 06:57.30 | jbot | i heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 06:57.34 | DarKnesS_WolF | were is the docs :P |
| 07:00.45 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.43) |
| 07:07.02 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 07:15.44 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:20.46 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 07:22.18 | esaym | howdy |
| 07:25.24 | esaym | I am redirecting an extension with the goto command in extensions.conf. For the step above the goto command I would like the dialing person to here ringing for 2 seconds. I know that can be done but I forgot how..... |
| 07:26.27 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 07:27.44 | esaym | I can use wait,2 for the step before but the dialing person will only hear silence for those 2 seconds |
| 07:33.16 | esaym | is it that hard? or just that easy? |
| 07:39.01 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:39.21 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 07:39.38 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 07:40.36 | esaym | hmm |
| 07:46.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:48.07 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 07:48.25 | esaym | so sleepy |
| 07:52.40 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-31-46.w90-16.abo.wanadoo.fr) |
| 08:05.16 | esaym | ah ha! |
| 08:05.19 | esaym | exten => 111,1,Ringing |
| 08:05.19 | esaym | exten => 111,2,Wait,5 |
| 08:08.51 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:09.55 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:10.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:16.49 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 08:17.40 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 08:24.54 | yang | I am wondering about TAPI, does it require write access in manager.conf ? |
| 08:31.39 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 08:41.32 | *** join/#asterisk hohum (i=dcorbe@apollo.wavelen.net) |
| 08:46.50 | *** join/#asterisk hohum (i=dcorbe@apollo.wavelen.net) |
| 08:48.37 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 08:49.28 | *** join/#asterisk af_ (n=getsmart@88-149-240-26.dynamic.ngi.it) |
| 08:50.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:52.32 | *** join/#asterisk hohum (i=dcorbe@apollo.wavelen.net) |
| 08:52.49 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:06.12 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 09:06.51 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 09:10.13 | *** join/#asterisk CrashHD (n=CrashHD@76-14-102-131.rk.wavecable.com) |
| 09:12.24 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:23.20 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 09:30.41 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 09:36.40 | *** join/#asterisk wordzilla (n=me@d58-106-139-71.sbr4.nsw.optusnet.com.au) |
| 09:40.34 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 09:43.40 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:48.02 | *** join/#asterisk g0ats (n=fuck@c-71-236-3-229.hsd1.ga.comcast.net) |
| 09:48.13 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 09:52.12 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-31-46.w90-16.abo.wanadoo.fr) |
| 09:58.07 | *** join/#asterisk equanimity (n=alex@cpc2-oxfd7-0-0-cust208.oxfd.cable.ntl.com) |
| 09:58.44 | *** join/#asterisk Euge_uk (n=eugeneev@host213-106-226-245.no-dns-yet.ntli.net) |
| 10:00.20 | *** join/#asterisk Euge_uk (n=eugeneev@host213-106-226-245.no-dns-yet.ntli.net) |
| 10:04.01 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279551708.dsl.bell.ca) |
| 10:04.40 | Euge_uk | Hi I am getting an error when making inbound calls into a newly built asterisk server, I have never used IRC before so please forgive me if i am a little slow on this! |
| 10:05.00 | Euge_uk | <PROTECTED> |
| 10:06.36 | Euge_uk | I am using a 4 port Sangoma card in the asterisk server and i have an Aculab card in the remote system acting at the network |
| 10:10.23 | yang | Euge_uk: i think you have a wrong configuration in zapata.conf |
| 10:10.39 | yang | Have you tried configuring zaptel with ztcfg |
| 10:12.29 | yang | http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf |
| 10:12.53 | Euge_uk | switchtype=euroisdn |
| 10:12.54 | Euge_uk | context=REMOTEAGENT |
| 10:12.54 | Euge_uk | group=1 |
| 10:12.54 | Euge_uk | signalling=pri_cpe |
| 10:12.54 | Euge_uk | pridialplan=unknown |
| 10:12.54 | Euge_uk | prilocaldialplan=unknown |
| 10:12.56 | Euge_uk | channel =>1-15,17-31 |
| 10:13.09 | yang | Euge_uk: please use http://openpaste.org |
| 10:13.27 | Euge_uk | sorry :) |
| 10:13.40 | yang | ~ook |
| 10:13.44 | yang | ~book |
| 10:13.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 10:13.48 | yang | ~paste |
| 10:13.49 | jbot | [paste] http://rafb.net/paste/, or see also pb |
| 10:13.56 | yang | ~pb |
| 10:13.57 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 10:15.22 | yang | Euge_uk: well I think that the config looks proper, but I am not an expert, try to reboot the machine, maybe the span was taken, and reboot will release it, happens to me occasionally |
| 10:16.13 | *** join/#asterisk panerai_go (n=neo@88-149-176-89.static.ngi.it) |
| 10:17.28 | panerai_go | Hi, I need to play the queue member id when the call is connected, like "you're now talking with operator number 1234". Has anyone had experience on this |
| 10:19.01 | yang | beside zapata.conf, you also need a proper /etc/zaptel.conf |
| 10:19.13 | yang | and you can test things with ztcfg -vvvv |
| 10:20.16 | Euge_uk | ztcfg returns without errors, it was originally showing an error but i re-compiled the wanpipe drivers and that seemed to resolve it |
| 10:20.59 | s0ck | hi... any hardware recommendations for getting 5 x analogue lines into asterisk (uk) |
| 10:21.59 | panerai_go | sock: nothing expecial is needed unless you want perform softecho |
| 10:22.36 | panerai_go | s0ck: just be sure the motherboard has a free IRQ for the pci card you're going to install |
| 10:24.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:28.07 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:28.47 | yang | Euge_uk: i think you are missing one thing |
| 10:29.17 | yang | Euge_uk: paste your zaptel.conf to pastebin |
| 10:30.13 | yang | Euge_uk: do you have dchan=16 in zaptel.conf ? |
| 10:32.54 | s0ck | panerai_go: cheers, any ideas on a 'channelbank' ? will it run pstn lines? |
| 10:34.55 | s0ck | otherwise, i think it will be a tdm805b i guess i need |
| 10:35.40 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 10:36.51 | *** join/#asterisk felipex (n=felipex@213-140-21-233.fastres.net) |
| 10:36.55 | felipex | hi at all |
| 10:37.48 | felipex | in asterisk 1.4 i have this error |
| 10:37.49 | felipex | <PROTECTED> |
| 10:37.52 | felipex | why ? |
| 10:38.19 | tzafrir | felipex, maybe you're missing a specific module? |
| 10:39.12 | felipex | the Page app in which module is ? |
| 10:43.18 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:43.35 | *** join/#asterisk RoyK (n=roy@ti200720a080-0467.bb.online.no) |
| 10:45.48 | *** join/#asterisk grEvenX (n=even@1mldj72.ip.ssc.net) |
| 10:52.29 | Euge_uk | yang: i am just trying to work out how to use pastebin lol |
| 10:53.26 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 10:53.33 | Euge_uk | yang: ahh i paste into it and then everyone else can see it, i am getting there slowly! |
| 10:53.48 | tinapa | is there anyone using gnudialer? |
| 10:54.49 | Euge_uk | yang: i have pasted it now |
| 10:55.14 | yang | Euge_uk: you need to paste here the url |
| 10:55.42 | Euge_uk | yang:http://openpaste.org/en/6006/ |
| 10:56.09 | *** join/#asterisk morex (n=m@91.84.56.12) |
| 10:56.21 | morex | Hey all |
| 10:56.35 | morex | Got a weird problem with callerIDs when dialling over zap from a local channel |
| 10:56.38 | yang | Euge_uk: looks good to me |
| 10:57.01 | morex | In all our other installations, it's taking the callerID from the local channel |
| 10:57.21 | morex | But for some reason in this one it seems to be taking it from the context in which the local channel was originated |
| 10:57.28 | Euge_uk | yang: it must be on the network site then i guess, i will get another ISDN plugged in this afternoon and see what happens |
| 10:57.35 | Euge_uk | yang: thanks for the help |
| 10:57.42 | morex | Just wondered if anyone else has run into this one? |
| 10:58.11 | yang | Euge_uk: ok, I hope that you fix it |
| 11:00.52 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:08.50 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 11:09.45 | *** join/#asterisk Jck_true (n=lc-LameC@cpe.atm2-0-79281.0x5730d7ea.hrnxx11.customer.tele.dk) |
| 11:09.51 | Jck_true | Howdy... |
| 11:10.34 | Jck_true | Can you get any softwre for you compauter that works like skype but connects to a voip server like asterisk? |
| 11:12.54 | Jck_true | Found it here http://www.voip-info.org/wiki-Open+Source+VOIP+Software |
| 11:13.31 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 11:13.38 | casix | hello |
| 11:13.54 | *** part/#asterisk Jck_true (n=lc-LameC@cpe.atm2-0-79281.0x5730d7ea.hrnxx11.customer.tele.dk) |
| 11:14.33 | casix | I've read that there is a variable called ${DIP_CODEC} that show the codec that is using but it is allways blank. There is another variable for that? |
| 11:14.50 | casix | ${SIP_CODEC} sorry |
| 11:18.55 | *** join/#asterisk yang (n=jan@212.103.134.208) |
| 11:19.23 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:19.42 | yang | I am having a question regarding manager.conf - I can only use my TAPI client if I have write = call,log,verbose,command,agent,user enabled in manager.conf, but with this someone can also fu*k up the dialplan...? |
| 11:21.18 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 11:22.06 | *** join/#asterisk _foxfire_ (n=_foxfire@cica-adm.fe.up.pt) |
| 11:23.00 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:23.00 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 11:27.09 | *** join/#asterisk viperdude (n=viperdud@87-127-248-176.no-dns-yet.enta.net) |
| 11:28.05 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 11:31.46 | casix | I've read that there is a variable called ${SIP_CODEC} that show the codec that is using but it is allways blank. There is another variable for that? |
| 11:32.50 | lmadsen | you might wanna check out the SIP* dialplan functions like SIPHEADER() |
| 11:36.15 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 11:38.30 | casix | thx |
| 11:40.56 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 11:42.58 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 11:43.53 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 11:46.44 | adeel | can i have multiple localnet entries in sip.conf if i'm behind 2 firewalls? |
| 11:48.23 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 11:52.58 | *** join/#asterisk nirz (n=nir@mail2.tikalnetworks.com) |
| 11:59.11 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 12:00.37 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 12:05.31 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 12:06.19 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 12:06.36 | *** join/#asterisk pdugas (n=pdugas@74.95.28.33) |
| 12:06.40 | *** part/#asterisk pdugas (n=pdugas@74.95.28.33) |
| 12:07.13 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:09.16 | *** join/#asterisk NoamRotter (n=noam@mail.browarnik.com) |
| 12:09.38 | *** part/#asterisk morex (n=m@91.84.56.12) |
| 12:12.03 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 12:13.10 | *** join/#asterisk appelza (n=psteyn@dsl-240-144-73.telkomadsl.co.za) |
| 12:13.38 | appelza | asterisk 1.2: where do I set the ammount of seconds to ring before going to voicemail? |
| 12:14.04 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 12:16.22 | *** join/#asterisk saftsack (n=oliver@p4FC77B07.dip.t-dialin.net) |
| 12:17.28 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 12:21.27 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:21.45 | *** join/#asterisk bmg505 (n=leon@196-209-79-189-tbnb-esr-2.dynamic.isadsl.co.za) |
| 12:22.41 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 12:23.08 | *** join/#asterisk ManxPower (n=manxpowe@46.sub-70-220-162.myvzw.com) |
| 12:23.46 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:27.56 | *** join/#asterisk sack (n=sack@140.Red-81-38-32.dynamicIP.rima-tde.net) |
| 12:29.16 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 12:29.45 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 12:36.15 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:37.12 | hi365 | what are the benifits of switchvox? how dows it differ from standard asterisk? |
| 12:38.37 | ManxPower | hi365: We don't know Switchvox |
| 12:39.22 | _foxfire_ | hello, any1 can answer me a question about the "dial" command ? |
| 12:39.28 | ManxPower | ~ask |
| 12:39.29 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:41.17 | _foxfire_ | i am trying to send an fax using txfax, everything work ok, that is if the call is answered, rejected or busy. But if there is noanswer on the other side i am not able to catch the end of the call where i run my cleanup script |
| 12:41.45 | _foxfire_ | is ther any diferent behaviour in the dialplan for an noanswer ? |
| 12:42.58 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-31-46.w90-16.abo.wanadoo.fr) |
| 12:43.09 | _foxfire_ | i was able to get that by adding the "g" option to the dial command, but then as soon as the call ius anwsered it continues and does not wait till the end of the call |
| 12:44.01 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 12:50.26 | [TK]D-Fender | hi365: It doesn't "differ". Its just another GUI. |
| 12:50.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:50.39 | *** join/#asterisk Psychobilly (n=Fuzz@athedsl-417978.home.otenet.gr) |
| 12:50.54 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-177.static.twtelecom.net) |
| 12:51.18 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 12:52.11 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 12:52.42 | Psychobilly | hello, is there a way to get all the messages when i enable sip debugging without having to be connected on the asterisk console? can i 'redirected' them or smth to stdout |
| 12:52.59 | Psychobilly | s/redirected/redirect |
| 12:53.37 | Psychobilly | or send them to a logfile in real time |
| 12:53.57 | mvanbaak | Psychobilly: have a look at logger.conf |
| 12:54.21 | appelza | asterisk 1.2: where do I set the ammount of seconds to ring before going to voicemail? |
| 12:54.24 | Psychobilly | a log will be 2nd chance, what about stdout? |
| 12:54.31 | Psychobilly | is there any way? |
| 12:54.46 | Corydon76-dig | No. |
| 12:56.19 | Corydon76-dig | Set up verbose logging from logger.conf and just log the output |
| 12:56.50 | Corydon76-dig | There is no "stdout" when it comes to the console |
| 12:56.57 | Psychobilly | ok thx |
| 13:00.34 | JayTee52 | appelza, it's a parameter option at the end of the Dial statement. It's well documented |
| 13:00.40 | JayTee52 | ~book |
| 13:00.41 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:02.11 | ManxPower | appelza: You don't. Calls won't automatically go to voicemail |
| 13:02.51 | ManxPower | appelza: Which GUI are you using? |
| 13:04.19 | *** join/#asterisk DirtyD (n=DirtyD@ool-18bddaa0.dyn.optonline.net) |
| 13:04.22 | DirtyD | morning. |
| 13:04.40 | JayTee52 | sorry, yeah. Calls won't go to voicemail automatically, you'd have to add a second statement after the Dial one for that extension. |
| 13:04.54 | DirtyD | I'm getting "carrier lost" with rxfax. Not even sure where to start trouble shooting. |
| 13:05.29 | ManxPower | DirtyD: Do you have rxgain and/or txgain in zapata.conf? |
| 13:05.30 | JayTee52 | the timeout parameter just controls how long the Dial function will run until someone picks up or it runs out and executes the next option |
| 13:05.51 | DirtyD | manx: no. |
| 13:06.04 | DirtyD | manx: I'm using T400P T1 card. |
| 13:06.25 | ManxPower | DirtyD: That is a very old card (first generation) |
| 13:10.32 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:12.27 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:15.05 | *** join/#asterisk jorgeraidel2000 (n=jorgerai@c-66-176-203-45.hsd1.fl.comcast.net) |
| 13:15.24 | jorgeraidel2000 | hola a todos |
| 13:17.10 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:17.16 | JayTee52 | hi george |
| 13:17.18 | jorgeraidel2000 | hola |
| 13:17.21 | jorgeraidel2000 | hi |
| 13:18.51 | jorgeraidel2000 | hey i need :( |
| 13:19.01 | ManxPower | Ask your question |
| 13:19.46 | ManxPower | JayTee52: looks like appelza is one of those people that ask a question then leave |
| 13:19.57 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:20.13 | ManxPower | JayTee52: I bet he's a GUI user. |
| 13:20.17 | JayTee52 | yep, and I wasn't thinkin clear enough to stipulate that it didn't automatically go VM. Glad you caught that one. |
| 13:20.32 | JayTee52 | sucks down more coffee |
| 13:20.49 | JayTee52 | ManxPower, I got that impression. |
| 13:22.23 | JayTee52 | Katty, hi! |
| 13:23.25 | JayTee52 | [TK]D-Fender, good morning! |
| 13:24.40 | [TK]D-Fender | JayTee52: Yeah, I figured I would jsut sit that one out :) |
| 13:26.43 | JayTee52 | I'd started to assume he had a Dial statement and then another as the next priority to route to VM because he said Asterisk 1.2 |
| 13:27.27 | appelza | Thanks |
| 13:28.08 | appelza | Yeah, I'm using FreePBX but the GUI's wait time I set doesn't seem to be affected, so wanted to try do it manually. Not ur problem, but thought there was a way to do it. |
| 13:28.43 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:28.43 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:29.27 | JayTee52 | what works for Asterisk might not work for other "Asterisk like" products. Asking a question in here might get you the right answer for an Asterisk system but then you have no guarantee it'll work on another like FreePBX or trixbox. |
| 13:30.23 | Katty | JayTee52: OHAI |
| 13:31.40 | JayTee52 | Katty, how are your PRI's doing? |
| 13:32.04 | *** join/#asterisk willianmazzardo (n=willianm@189-30-238-14.smace701.dsl.brasiltelecom.net.br) |
| 13:33.38 | [TK]D-Fender | appelza: Yup, go figure out how to configure FreePBX for this. It should be trivial. |
| 13:33.54 | Katty | JayTee52: well. we only have 1. but it's doing just fine, thanks (= |
| 13:36.44 | JayTee52 | Katty, glad to hear it. I'll be moving from using * as a voip bridge off of a Nortel PBX to being the front door to our telco and using 4 PRI spans, 2 for inbound and outbound and 2 to route calls to and from the Nortel PBX till everything's moved off it and onto *. |
| 13:37.17 | Katty | JayTee52: sounds skeery. |
| 13:37.27 | JayTee52 | yeah, I want to drink alot but I don't :-) |
| 13:38.21 | JayTee52 | I've got one TE212P configured and setup with a X-over between the two spans for testing before I change the config and test using the live PRIs. |
| 13:38.26 | appelza | [TK]D-Fender: Orly? It seems trivial, as I simply change the value and click apply..but then it doesn't take effect and there is nothing in the logs. Just dont reply to my questions please. I understand I cannot be helped here, at first I thought it was a shared setting between the two. End of story. |
| 13:38.36 | appelza | die. |
| 13:38.58 | *** part/#asterisk appelza (n=psteyn@dsl-240-144-73.telkomadsl.co.za) |
| 13:39.26 | JayTee52 | damn, wished he stuck around long enough for me to tell him to "get frakked" |
| 13:40.13 | [TK]D-Fender | Wow... and here I was trying to reassure him that the GUI should have taken at least THAT into account and that he need only digg a tiny bit more to customize it. Sheesh. |
| 13:40.16 | JayTee52 | cuz that was just plain rude |
| 13:40.36 | JayTee52 | now I'm kinda pissed that I tried to help in the first place. |
| 13:41.13 | JayTee52 | specially when they start dissin my mentors |
| 13:43.03 | *** join/#asterisk AsteriskRo (n=Asterisk@190.36.185.138) |
| 13:43.08 | [TK]D-Fender | And for nothing. I didn't flame him or FreePBX on that one. He just has serious reading comprehension issues (or a raging inferiority complex) |
| 13:43.24 | Jumpie | morning |
| 13:43.29 | AsteriskRo | hi |
| 13:43.47 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 13:43.50 | AsteriskRo | anybody there to ask about connect Asterisk PBX to SIP provider? |
| 13:43.51 | UnixDog | Freepbx=bloat |
| 13:44.19 | JayTee52 | UnixDog, morning! |
| 13:44.25 | JayTee52 | hi Jumpie |
| 13:44.30 | UnixDog | no need for all the database backend crap that breaks with every other update |
| 13:44.36 | znoG_ | hey, I've suddenly started seeing the following error: chan_sip.c:11081 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete. |
| 13:44.40 | [TK]D-Fender | AsteriskRo: Have you checked with your provider first? Who are they? |
| 13:44.41 | UnixDog | morning |
| 13:44.42 | znoG_ | any ideas where its coming from? |
| 13:45.11 | UnixDog | znoG_: what version of asterisk ? |
| 13:45.15 | AsteriskRo | D-Fender: I'm using VIVOphone, i have all details about server ip, etc... |
| 13:45.36 | znoG_ | UnixDog: 1.2.9.1 |
| 13:45.48 | AsteriskRo | i can login with a xlite softphone, but can't with my pbx |
| 13:46.16 | UnixDog | ok major old ok |
| 13:46.47 | UnixDog | I am in 1.4.19 |
| 13:47.33 | [TK]D-Fender | UnixDog: You'll never guess which major GUI distro was based on that version for many releases ;) |
| 13:47.56 | JayTee52 | bet it starts with a t :-) |
| 13:47.58 | UnixDog | trashbox |
| 13:48.28 | *** join/#asterisk NirS_ (n=Nir@212.150.121.71) |
| 13:48.31 | NirS_ | hey all |
| 13:48.47 | AsteriskRo | hey NirS |
| 13:48.48 | [TK]D-Fender | AsteriskRo: pastebin your peer entry, register statement (both masking only the password), the complete CLI output of a failed call attempt at verbose 10, SIP debug enabled. |
| 13:48.49 | [TK]D-Fender | ~pb |
| 13:48.50 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:48.53 | NirS_ | hey AsteriskRo |
| 13:49.07 | NirS_ | say, anyone got the L parameter on dial working properly? |
| 13:49.12 | AsteriskRo | ok |
| 13:49.13 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-131-138.dsl.stlsmo.sbcglobal.net) |
| 13:49.23 | NirS_ | I have one that works fine in terms of limiting the call |
| 13:49.28 | NirS_ | by the announcements don't happen |
| 13:49.31 | NirS_ | any ideas ? |
| 13:49.35 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 13:49.45 | bsdwarrior | is anyone running asterisk on freebsd7 in production yet ? |
| 13:50.08 | LemensTS | Can you do unix/posix timestamp in asterisk? I need to send it to a variable in my dialplan... |
| 13:50.22 | UnixDog | yes |
| 13:50.28 | *** join/#asterisk voipnet-tech (n=voipnet-@216.195.128.62) |
| 13:50.41 | UnixDog | I have 1.4.18.1 on 200 boxes |
| 13:50.46 | Jumpie | jeez |
| 13:50.49 | UnixDog | all freebsd 7.0 |
| 13:51.09 | UnixDog | works with no issues |
| 13:51.22 | Jumpie | ill be honest, my fios connection connecting to my incoming DID |
| 13:51.25 | Jumpie | isnt a bad quality at all |
| 13:51.29 | Jumpie | of course it sjust one call... |
| 13:51.31 | PepOSX | UnixDog, the port is working fine? |
| 13:51.35 | UnixDog | yes |
| 13:51.37 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:51.38 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 13:51.40 | PepOSX | good |
| 13:51.46 | *** join/#asterisk af_ (n=getsmart@88-149-240-26.dynamic.ngi.it) |
| 13:51.47 | UnixDog | I am about to have the port updated to 1.4.19 |
| 13:52.02 | UnixDog | there is a step you have to take with the port |
| 13:52.11 | UnixDog | thhat is missed right now |
| 13:52.23 | tzafrir | UnixDog, what was the previous version? |
| 13:52.30 | [TK]D-Fender | LemensTS: "core show function STRFTIME" , ${EPOCH} |
| 13:52.31 | glaz | the zaptel driver works fine on 7.0 ? |
| 13:52.33 | PepOSX | UnixDog, good |
| 13:52.36 | bsdwarrior | unixdog, I'm trying to get away from linux since we run bsd, have you had any software issues under bsd with asterisk ? |
| 13:52.49 | UnixDog | zaptel 1.4.6_4 works fine |
| 13:52.58 | UnixDog | but its being updated |
| 13:53.07 | bsdwarrior | unixdog, we use to have crashes frequently (4 years ago) and wierd problems. development was quicker under linux so we switched |
| 13:53.09 | PepOSX | glaz, but with linux better |
| 13:53.11 | AsteriskRo | D-Fender, give me a minut, I've made so many changes.... |
| 13:53.19 | glaz | PepOSX: obviously. heh |
| 13:53.21 | UnixDog | linux is not better |
| 13:53.30 | PepOSX | linux is not better |
| 13:53.32 | bsdwarrior | unixdog, I would agree |
| 13:53.39 | PepOSX | but asterisk is better on linux |
| 13:53.44 | UnixDog | no |
| 13:53.55 | UnixDog | I find asterisk to be alot more stable on bsd |
| 13:53.58 | bsdwarrior | unixdog, how largescale of a system are you running under freebsd 7 ? / |
| 13:53.58 | glaz | Ok let make this clear |
| 13:54.11 | UnixDog | and performance overall is better due to better memory management |
| 13:54.13 | glaz | There is a lot more documentation on how to run/debug/configure asterisk on linux |
| 13:54.16 | glaz | is this better? |
| 13:54.54 | wyoming | I'm a bit peeved at counterpath (eyeBeam) right now. Eyebeam's detected that a new version is available, and has downloaded it, and is offering to install it. But it never actually does, and so you get it constantly offering to install it. I'll have to download the new version manually, and install it, I'm sure, but I thought I'd complain to counterpath, but it's been months, and I can't remember what username I used, or the password, and it SURE is |
| 13:54.54 | wyoming | n't email that info if I can't remember it. Sheesh. Welcome to a typical day in the life of Murphy. |
| 13:55.01 | AsteriskRo | i used Asterisk-GUI with no succes on making the outgoing call, now I changed it directly to the files |
| 13:55.13 | UnixDog | the only big change for asterisk on bsd vs linux is layout . and a few build patches that fix a few things |
| 13:55.20 | UnixDog | you all need to learn your facts |
| 13:55.33 | UnixDog | the os war of what os is best is bullshit |
| 13:55.46 | glaz | humm, OS talking, I prefer BSD to linux |
| 13:55.50 | UnixDog | the fact is this its really what you feel more comfortable on |
| 13:56.17 | UnixDog | but I do find that the fact linux caches all your avail mem is a pain |
| 13:56.17 | glaz | I've been running FreeBSD and openBSD as servers for years. But I had a lot of issues with * on bsd, not on linux tho... |
| 13:56.32 | UnixDog | and most linux distros install way to much crap on the hd |
| 13:56.35 | wyoming | not even my name is right today |
| 13:56.43 | voipnet-tech | hi all... I've built a few systems that are supposed to be identical motherboards. I am using the onboard sound card to do overhead paging. On earlier systems I used chan_oss, and ran into some paging amps that made this sound like crap, so I switched to chan_alsa and the problem was cured. Now I have a system that when set for chan_alsa calling console/dsp returns "Everyone is busy/congested at this time". changing it to chan_oss, it works |
| 13:56.53 | LemensTS | TKD-Fender: thanks the wiki says STRPTIME heh |
| 13:57.05 | UnixDog | a basic install of centos is over 1.5 gigs now days thats as bad as windows |
| 13:57.07 | glaz | FreeBSD can also install a lot of crap during the installation if you don't know what you're doing. |
| 13:57.11 | UnixDog | no |
| 13:57.18 | glaz | yes. |
| 13:57.19 | UnixDog | not true |
| 13:57.22 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 13:57.22 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 13:57.39 | UnixDog | if you do a bas einstall of freebsd its only the os |
| 13:57.46 | glaz | I always strip down my FreeBSD installation |
| 13:57.49 | UnixDog | and the portstree and the base src |
| 13:57.51 | UnixDog | thats it |
| 13:57.53 | codefreeze | There, that's better |
| 13:58.04 | UnixDog | nothing else |
| 13:58.35 | UnixDog | and if you do a minimal install its only 148 megs and thats no ports tree and no /usr/src base src |
| 13:58.54 | glaz | I said, if you dont know what you're doing |
| 13:59.07 | glaz | if you install EVERYTHING it will take more than 2 gigs |
| 13:59.15 | UnixDog | well if you dont take the time to read first its your own fault |
| 13:59.26 | *** join/#asterisk Jocko (i=daemon@c-67-168-165-185.hsd1.wa.comcast.net) |
| 13:59.26 | glaz | you can do the same thing with linux |
| 13:59.37 | glaz | my slackware installations take 300Mg |
| 13:59.49 | UnixDog | I have yet to see a linux distro with a minimal install other then slackware |
| 14:00.14 | UnixDog | slackware is close to bsd in the way they do things |
| 14:00.26 | glaz | I know, ubuntu,rh,fedora,debian,etc.. they all take a lot of space. |
| 14:00.42 | Jocko | Anyone here know of an asterisk app that reads filenames to a caller? |
| 14:00.45 | UnixDog | but the fact about asterisk on bsd is layout |
| 14:00.50 | UnixDog | for the most part |
| 14:00.51 | glaz | it is, the bootup process is similar to bsd also. |
| 14:00.58 | UnixDog | we use usr land |
| 14:01.17 | UnixDog | asterisk on linux is in /varand /usr/lib |
| 14:01.23 | UnixDog | we change this |
| 14:01.43 | glaz | Yeah, and I like it that way. |
| 14:01.55 | glaz | I would *love* to run asterisk on FreeBSD. |
| 14:01.58 | UnixDog | we use /usr/local /u/l/share/asterisk /u/l/l/asterisk |
| 14:02.22 | glaz | /u/l/e/asterisk |
| 14:02.56 | UnixDog | and a few patches to make it build better on bsd . but for the most part asterisk is now nativly supported on bsd |
| 14:03.07 | UnixDog | but I can not say that about asterisk 1.6 |
| 14:03.25 | UnixDog | many changes have happened and we have to do alot more patching. |
| 14:03.46 | hi365 | [TK]D-Fender: (re: switchvox) so its like asterisknow/freepbx/trixbox? is the asterisk the same? |
| 14:03.58 | UnixDog | the biggest issue is we make some major changes to zaptel |
| 14:04.18 | [TK]D-Fender | hi365: Its just a GUI. |
| 14:04.27 | Jumpie | switchvox i thought was just a platform? |
| 14:04.31 | Jumpie | server+diff cards |
| 14:04.38 | hi365 | well its a real rip off then |
| 14:04.43 | UnixDog | we have the digium asterisk gui on bsd also |
| 14:05.02 | UnixDog | we even have sangoma on bsd |
| 14:05.42 | glaz | great, i'll give it a try |
| 14:05.42 | [TK]D-Fender | AsteriskRo: Pastebin what you've done, be it in users.conf or sip.conf. |
| 14:05.57 | glaz | I have to convert a Cisco CM to asterisk tonite, I'll give 7.0 a try. |
| 14:06.21 | AsteriskRo | TKD-fender: Â the SIP provider must be on what file? users.conf or sip.conf ? using ASTERISK-GUI it is saved on users.conf and it doesn't work, i made some changes and put it on sip.conf and the other phone rings but without any audio |
| 14:06.31 | [TK]D-Fender | AsteriskRo: EITHER. |
| 14:06.31 | glaz | AsteriskRo: sip.conf |
| 14:06.50 | [TK]D-Fender | AsteriskRo: users.conf just gets parsed out into the equivalent anyways |
| 14:07.01 | AsteriskRo | ok |
| 14:07.07 | tzafrir | UnixDog, what exactly do you call "minimal install"? |
| 14:07.08 | NirS_ | anyone using call limiting on dial with announcements? |
| 14:07.15 | UnixDog | for those of you using asterisk on bsd there is #asterisk-bsd also |
| 14:07.16 | [TK]D-Fender | AsteriskRo: if you have no audio then you most likely have a NAT issue. Read up : |
| 14:07.17 | NirS_ | I can't get it to work properly for some reason |
| 14:07.19 | [TK]D-Fender | ~sipnat |
| 14:07.19 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:07.21 | [TK]D-Fender | ^^^^^^^^^ |
| 14:07.36 | UnixDog | minimal install is a custom install |
| 14:07.41 | UnixDog | and its easy to do |
| 14:08.05 | UnixDog | and if you join #asterisk-bsd I can walk you threw one |
| 14:08.12 | UnixDog | this is not the channel for it |
| 14:08.14 | tzafrir | UnixDog, Debian, e.g. gives you a pretty minimal one. But for smaller systems people tend to build custom systems with e.g. buildroot |
| 14:08.23 | tzafrir | astlinux is an example of that |
| 14:08.27 | UnixDog | dont need to on bsd |
| 14:08.36 | tzafrir | UnixDog, all distros give you a "custom install" |
| 14:08.47 | UnixDog | a minimal install off a bsd disk is 148 megs |
| 14:09.21 | mort_gib | UnixDog: so both OpenBSD and freeBSd will run Asterisk |
| 14:09.36 | UnixDog | yes |
| 14:09.44 | UnixDog | and netbsd |
| 14:09.58 | mort_gib | The saying was that Zaptel would cause issues... |
| 14:09.59 | [TK]D-Fender | UnixDog: All with Zaptel fully functional? |
| 14:10.02 | tzafrir | I can build a minimal Debian system to be around that size, if I want (using installer preseeding). But this does not include Asterisk |
| 14:10.03 | UnixDog | its in the ports tree |
| 14:10.11 | tzafrir | asterisk + dependencies take some more |
| 14:10.34 | tzafrir | And 148 and such surely does not include a build environment. I assume you build on a separate machine |
| 14:10.36 | UnixDog | freebsd minimal + asterisk+deps = 390 megs when done |
| 14:11.00 | UnixDog | unless you add mysql and radious and all the other addons |
| 14:11.07 | UnixDog | then it can be upto 500 megs |
| 14:11.17 | *** join/#asterisk [gnubie] (n=[gnubie]@cm85.omega113.maxonline.com.sg) |
| 14:11.24 | [gnubie] | waves to all.. |
| 14:11.34 | bsdwarrior | ubuntu is worthless in a datacenter. it ends of lifes versions so quickly |
| 14:12.00 | mort_gib | But is size really that important?? -I'm building a Soekris 5501, that will actually features a 40GB hdd |
| 14:12.24 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 14:12.25 | UnixDog | I use alix boards with microdrives |
| 14:12.43 | UnixDog | and I still try to keep it small for more space for storage |
| 14:12.53 | tzafrir | well, my Asterisk live CD is less than 250MB. And that is really not optimized for size |
| 14:13.00 | UnixDog | and a alix is perfect for a mini pbx |
| 14:13.06 | mort_gib | we have solid state drives coming out |
| 14:13.16 | UnixDog | wich one |
| 14:13.23 | tzafrir | I e.g. use apache (freepbx is included) and never tried to optimize for size - it is well below one CD |
| 14:13.42 | UnixDog | freepbx to me is not a good thing |
| 14:13.52 | UnixDog | but to each his own |
| 14:14.48 | tzafrir | well, give me a better alternative |
| 14:14.54 | mort_gib | Yes, each to his own, or each server/function to it's own |
| 14:15.08 | tzafrir | It's a live CD, needs to be usable by anybody |
| 14:15.40 | Jumpie | ok im embarking on doing fresh asterisk isntall today |
| 14:15.41 | Jumpie | :D |
| 14:15.41 | UnixDog | the digium gui |
| 14:15.55 | UnixDog | there is a paid gui out there called thirdlane |
| 14:16.07 | UnixDog | there is another project gui called askozia |
| 14:16.16 | tzafrir | that happens to not be usable by someone who has a windows system with the default browser installed |
| 14:16.56 | tzafrir | askozia is a complete OS, not just a GUI |
| 14:17.12 | tzafrir | and I haven't heard anybody really using its GUI to manage Asterisk |
| 14:17.13 | UnixDog | you can port the gui elsewhere |
| 14:17.21 | tzafrir | I'm not going to use any non-free software |
| 14:17.24 | UnixDog | its just php pages |
| 14:17.46 | AsteriskRo | TKD-fender there is the peer info and CLI output |
| 14:18.07 | Jumpie | im curious if i have a normal install and i dont have a need for alot of tweaking if freepbx will be suitable, or if i should just stick to the configs |
| 14:18.13 | Jumpie | im sure i know what most would say :D |
| 14:18.19 | UnixDog | thirdlane is not free but it has 200% more then freepbx and does not require a database like mysqland peardb to run |
| 14:18.37 | AsteriskRo | yes, I supose it can be a NAT problem, because the SIP provider server uses a outbound proxy and i don't know how to register in that outbound proxy |
| 14:18.38 | UnixDog | and it has a full users portal for users |
| 14:19.04 | UnixDog | take time and look at the demo on thirdlane.com |
| 14:19.15 | UnixDog | I will even walk you threw it |
| 14:19.23 | UnixDog | and show you the users portal |
| 14:20.00 | [gnubie] | what makes a zap trunk/channel that connects to a pots network noisy? everytime that my zap trunk is involved (inbound or outbound calls) through pots, i hear noise like an electrical circuits or something |
| 14:20.32 | Jocko | Can someone point me to custom dialplan/macro samples? I need to list files in a directory based on caller information? |
| 14:20.39 | UnixDog | call the phone comapny and ask for them to install line filters |
| 14:20.54 | errr | is it possible to use AMI to get the IP of sip devices? |
| 14:21.18 | hi365 | errr: just run a simple script |
| 14:21.39 | hi365 | on 'asterisk -rx :sip show devices"' |
| 14:21.47 | errr | hi365: well right now I use asterisk -rx and parse that info, but I would like to use AMI if possible |
| 14:21.51 | hi365 | (not that that answers your question..) |
| 14:22.21 | casix | quit |
| 14:22.29 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 14:22.46 | teknoprep | how do i change which ringtone is used for polycom phones from the config file ? |
| 14:22.52 | teknoprep | on the tftp boot server ? |
| 14:23.30 | UnixDog | well for one you can add wav files to the cfg file |
| 14:23.59 | teknoprep | i just want all my polycom phones to ring with the ringback ringtone |
| 14:24.24 | AsteriskRo | [TK]D-Fender: that was modifying directly the files (sip.conf and extensions.conf), but i'd like to do it with the ASTERISK-GUI interface |
| 14:24.43 | UnixDog | menu 3 1 4 on the phone |
| 14:24.44 | [TK]D-Fender | AsteriskRo: You need to fix your NAT settings. Go read that guide. |
| 14:24.46 | [TK]D-Fender | ~sipnat |
| 14:24.47 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:24.53 | UnixDog | and choose the line you want to change |
| 14:25.03 | UnixDog | and choose the ring tone |
| 14:25.07 | AsteriskRo | [TK]D-Fender: what guide? |
| 14:25.12 | [TK]D-Fender | AsteriskRo: and * doesn't support G723 : [Apr 9 09:43:08] WARNING[5071]: channel.c:2755 set_format: Unable to find a codec translation path from g723 to ulaw |
| 14:25.32 | [TK]D-Fender | AsteriskRo: Notwithout a TC400 anyways... |
| 14:25.52 | AsteriskRo | [TK]D-Fender: ok |
| 14:25.52 | UnixDog | g729 and g723 are paid codecs |
| 14:26.03 | *** join/#asterisk SomethingISODD (n=wno@S010600a0d1757bfb.cg.shawcable.net) |
| 14:26.12 | SomethingISODD | hello |
| 14:26.28 | teknoprep | UnixDog, i wanted to do that from the config files tho.. i know how to doit from the menu |
| 14:26.30 | *** join/#asterisk chigital (n=chigital@213.131.236.199) |
| 14:26.43 | UnixDog | ok hold on |
| 14:26.52 | UnixDog | let me grab my manual |
| 14:27.01 | SomethingISODD | Can anyone tell me what is the max concurrent calls asterisk can handle |
| 14:27.11 | SomethingISODD | or does it depend on the cpu/ram |
| 14:27.12 | UnixDog | depends on your hardwaare |
| 14:27.13 | NirS_ | SomethingISODD, that number varies |
| 14:27.15 | voipnet-tech | 950,125 |
| 14:27.16 | [TK]D-Fender | SomethingISODD: Depends on about a million things |
| 14:27.33 | NirS_ | SomethingISODD, I have machines running around 160 conurrent channels (signalling + media) |
| 14:27.39 | SomethingISODD | ok i am trying to figure out what hardware i need to handle 1000 calls |
| 14:27.45 | NirS_ | and I have other machines running around 1400 channels (signalling only) |
| 14:27.49 | voipnet-tech | 1.21 Gigawatts |
| 14:27.51 | NirS_ | 1000 calls ? |
| 14:28.07 | [TK]D-Fender | voipnet-tech: load res_fluxcapacitor.so |
| 14:28.09 | SomethingISODD | NirS_ yes thats what the client is requesting it be able to handle |
| 14:28.16 | NirS_ | voipnet-tech, 1.21 gigawatts, how could you have been so careless |
| 14:28.32 | NirS_ | BRB |
| 14:28.51 | [TK]D-Fender | NirS : s'ok, it'll maintain space-time stability as long as it doesn't reach 88 MPH ;) |
| 14:30.00 | AsteriskRo | [TK]D-Fender: yes, it was a codec issue :-$ |
| 14:31.18 | BBHoss | jigawatts |
| 14:32.18 | UnixDog | teknoprep: look at the pm |
| 14:33.13 | NirS_ | SomethingISODD, sounds like an automatic dialer to me, is that correct ? |
| 14:33.26 | teknoprep | UnixDog, ty |
| 14:33.53 | UnixDog | change the word Default to match the done you want |
| 14:33.59 | NoamRotter | ?????? |
| 14:34.00 | UnixDog | done/tone |
| 14:34.25 | teknoprep | <PROTECTED> |
| 14:34.45 | teknoprep | i would change that to say se.rt.1.ringer="12" |
| 14:35.53 | SomethingISODD | NirS no he wants to run a calling card program off of asterisk |
| 14:36.14 | *** join/#asterisk gandhijee (n=root@host-66-202-34-165.spr.choiceone.net) |
| 14:36.29 | teknoprep | i would say put 4 x quadcore opteron's in with 16gb of RAM |
| 14:36.36 | UnixDog | I believe so |
| 14:36.45 | UnixDog | the manual is not saying |
| 14:36.50 | teknoprep | yeah i will find out |
| 14:36.52 | UnixDog | let me look further |
| 14:36.52 | teknoprep | i tryed it out |
| 14:36.53 | gandhijee | hey, can i use the console to connect 2 calls? |
| 14:36.57 | teknoprep | i am rebooting now |
| 14:37.11 | Jumpie | im scared to wipe all this away |
| 14:37.17 | Jumpie | somethin i was semi familiar with, going to the dark side :D |
| 14:37.20 | teknoprep | UnixDog, yeah that didn't work |
| 14:37.22 | Jumpie | putting cent os disc 1 in now :D |
| 14:37.27 | UnixDog | ok |
| 14:37.37 | Jumpie | fender, i was told centos has like, a package install for asterisk |
| 14:37.39 | UnixDog | let me test here and I will let you know |
| 14:37.52 | Jumpie | kinda equiv to sudo aptget -install asterisk |
| 14:37.54 | [TK]D-Fender | Jumpie: DON'T |
| 14:37.55 | Jumpie | in ubuntu |
| 14:38.00 | Jumpie | dont do it that way? |
| 14:38.03 | Jumpie | k |
| 14:38.10 | [TK]D-Fender | Jumpie: Just compile from source like the rest of us. |
| 14:38.13 | Jumpie | grr i wish things were easy lol |
| 14:38.17 | Jumpie | aight lemme get centos on here |
| 14:38.27 | Jumpie | itll have a way to wipe whats existing right? |
| 14:38.41 | Jumpie | says goodbye to trix |
| 14:40.07 | UnixDog | <DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="12" se.rt.1.callWait="6" se.rt.1.mod="1"/> |
| 14:40.15 | UnixDog | there you go teknoprep |
| 14:40.17 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:40.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:40.24 | teknoprep | that worked for you ? |
| 14:40.27 | teknoprep | thats what i tryed |
| 14:40.28 | UnixDog | yes |
| 14:40.31 | teknoprep | didn't work for me |
| 14:40.37 | UnixDog | it worked on my 501 |
| 14:40.41 | teknoprep | yeah i have a 501 also |
| 14:40.45 | DirtyD | will rxfax work on with a T22OP T1/PRI card? |
| 14:40.53 | UnixDog | hmm |
| 14:40.58 | UnixDog | what firmware you running |
| 14:41.14 | teknoprep | latest |
| 14:41.24 | UnixDog | 2.2.2 or 3.0 |
| 14:41.28 | teknoprep | 2.2.2 |
| 14:41.36 | teknoprep | shoudl i get 3.0 ? |
| 14:41.56 | UnixDog | my 501 is on 2.2.0 |
| 14:42.02 | UnixDog | let me update and test |
| 14:42.34 | teknoprep | i can't use 3.0 on a 501 |
| 14:42.37 | teknoprep | no one can |
| 14:43.01 | teknoprep | i didn't think sip 3.0 was out |
| 14:43.11 | teknoprep | i have 4.0 boot rom and 2.2.2 sip |
| 14:44.33 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584618.dsl.bell.ca) |
| 14:45.46 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 14:46.47 | Jumpie | setting up asterisk slightly intoxicated |
| 14:47.17 | AsteriskRo | [TK]D-Fender: but how can i make it work from the basic settings of the ASTERISK-GUI? |
| 14:47.26 | UnixDog | ok |
| 14:47.31 | UnixDog | getting and testing |
| 14:47.38 | UnixDog | brb |
| 14:47.52 | *** join/#asterisk lupino3 (n=lupino3@217-133-45-108.b2b.tiscali.it) |
| 14:48.00 | gandhijee | anyone? |
| 14:48.08 | NirS_ | any one with dial |
| 14:48.23 | NirS_ | dial L option experience ? the bloody thing won't playback announcements |
| 14:48.38 | lupino3 | hello, I have a small problem with Asterisk 1.4.18: I need to issue a "sip reload" once asterisk is up in order to load static realtime sip configuration |
| 14:48.46 | [TK]D-Fender | AsteriskRo: Go ask in the GUI channel. You might NOT have a way in there. |
| 14:49.02 | lupino3 | does anyone know what's the problem? |
| 14:49.08 | [TK]D-Fender | AsteriskRo: AsteriskGUI is not a "complete" solution, but rather a losser framework. |
| 14:49.11 | lupino3 | at startup I see 0 peers |
| 14:49.48 | lupino3 | I issue a "sip reload" |
| 14:49.52 | lupino3 | and magically I see my peers |
| 14:49.55 | lupino3 | can anyone help me? |
| 14:49.56 | lupino3 | tia |
| 14:52.50 | AsteriskRo | [TK]D-Fender: ok, thanks |
| 14:53.09 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 14:53.42 | UnixDog | ok all you asterisk-bsd users join #asterisk-bsd |
| 14:53.46 | UnixDog | we need you |
| 14:54.47 | Jumpie | hey unix |
| 14:54.54 | Jumpie | im installing centos from scratch, what kinda install should i do? |
| 14:54.56 | Jumpie | server/server gui, etc |
| 14:56.10 | *** join/#asterisk quigon (n=matias@nat/redhat/x-4fb6c8296b7f3d2a) |
| 14:56.54 | AsteriskRo | [TK]D-Fender: nobody talks on asterisk-gui channel :( |
| 14:57.13 | [TK]D-Fender | AsteriskRo: Do it manually then. |
| 14:57.31 | AsteriskRo | [TK]D-Fender: can I show you the changes in extensions.conf ? |
| 14:58.09 | *** join/#asterisk nighty^ (n=nighty@p1022-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 14:58.10 | [TK]D-Fender | AsteriskRo: Your problems aren't in extensions.conf |
| 14:58.23 | [TK]D-Fender | AsteriskRo: No audio = sip.conf issue |
| 14:58.29 | Jumpie | fender, i did desktop gnome and server, is that adequate? |
| 14:58.38 | [TK]D-Fender | AsteriskRo: and possibly split across users.conf |
| 14:58.53 | [TK]D-Fender | Jumpie: I like the "EVERYTHING" option personally. |
| 14:58.59 | AsteriskRo | [TK]D-Fender: ok, but besides that i changed extensions.conf too |
| 14:59.19 | [TK]D-Fender | Jumpie: Don't be a schmuch who spends 10 times as long trying to add stuff in afterwards just to get things working. |
| 14:59.30 | Jumpie | fender, but i really dont need all the fluff, and i only burned the first cd |
| 14:59.32 | [TK]D-Fender | AsteriskRo: Whats the current problem? |
| 14:59.34 | Jumpie | i want strictly asterisk, and sshd |
| 14:59.43 | Jumpie | but ya i dont wanna be a schmuk either |
| 14:59.52 | *** join/#asterisk andreambrosio (n=andreamb@bdv75-9-88-165-95-162.fbx.proxad.net) |
| 15:00.00 | [TK]D-Fender | Jumpie: Keep downloading... |
| 15:00.25 | andreambrosio | i need help on a workaround to use instant messaging and presence under asterisk |
| 15:00.26 | Jumpie | :D |
| 15:00.43 | andreambrosio | would anybody walk me through that? |
| 15:01.19 | *** join/#asterisk kyron (n=kyron@modemcable086.140-70-69.static.videotron.ca) |
| 15:01.31 | lupino3 | nevermind, it worked updating to 1.4.19 :) |
| 15:01.40 | lupino3 | thanks, however |
| 15:01.44 | lupino3 | have a nice day |
| 15:02.09 | Jumpie | k i do them all also packages from centos exstras |
| 15:02.11 | Jumpie | lol ill be here a bit |
| 15:02.31 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:03.26 | andreambrosio | great |
| 15:03.39 | andreambrosio | I'm sorry, can I PM you? |
| 15:06.26 | AsteriskRo | [TK]D-Fender: in extensions.conf the GUI adds this line: exten = _99XXXXX,1,Macro(trunkdial,${trunk_1}/${EXTEN:0}), and the call can't be maked, i changed to exten => _99XXXXX,1,Dial(SIP/${EXTEN}@xx.xxx.com |
| 15:08.12 | Jumpie | AsteriskRo i already learned man |
| 15:08.13 | AsteriskRo | trunkdial Macro is basically this: exten => s,1,Dial(${ARG1}) |
| 15:08.19 | Jumpie | you arent gonna get alot of help on qui questions |
| 15:08.37 | Jumpie | everything was always different than most of the books and help in here and people just dont wanna help troubleshoot it |
| 15:08.39 | Jumpie | so go all the way like i am |
| 15:09.11 | plik | hurrah, another one sees the light... congrats Jumpie |
| 15:09.19 | Jumpie | :) |
| 15:09.25 | Jumpie | i was dragged through the depths of hell first |
| 15:09.45 | andreambrosio | what's the purpose of the irc channel, then? |
| 15:10.12 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:10.35 | [TK]D-Fender | AsteriskRo: Indeed if its GUI generated dialplan don't ask here. |
| 15:10.49 | AsteriskRo | ok |
| 15:10.51 | Jumpie | crap it says it needs all 6 cds |
| 15:10.54 | Jumpie | i thought it could just download |
| 15:10.56 | [TK]D-Fender | AsteriskRo: it was working fine before, so if you mess with it you might screw up and it'll stop working. |
| 15:11.12 | [TK]D-Fender | Jumpie: Thinking is best left to trained professionals. |
| 15:11.23 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 15:12.59 | AsteriskRo | <PROTECTED> |
| 15:13.18 | [gnubie] | what makes a zap trunk/channel that connects to a pots network noisy? everytime that my zap trunk is involved (inbound or outbound calls) through pots, i hear noise like an electrical circuits or something.. but if i connect directly the pots line to an analog phone, the call quality is perfect.. any idea? |
| 15:13.18 | Jumpie | fender heh |
| 15:13.21 | Jumpie | i thought thats what you SAID |
| 15:13.22 | Jumpie | my bad |
| 15:13.34 | andreambrosio | anyone, please, does anybody know how to add presence and IM to asterisk? |
| 15:14.19 | [TK]D-Fender | [gnubie]: bad card or module. |
| 15:15.06 | [gnubie] | [TK]D-Fender: how will i test to confirm that i have a broken card or module(s)? i'm using the digium dev kit with 1 fxo and 1 fxs |
| 15:15.12 | [TK]D-Fender | andreambrosio: * doesn't support IM at the moment, presence is already supported through the "hint" priority. |
| 15:15.25 | [TK]D-Fender | [gnubie]: Try another card / module. |
| 15:15.45 | [gnubie] | [TK]D-Fender: that i don't have an extra one.. :( |
| 15:15.48 | JayTee52 | gnubie, UnixDog also suggested a hour ago when you asked the same question that you might need your telco to add line filters. |
| 15:16.04 | [TK]D-Fender | And I hope never to hear the name "Digium Dev Kit" again. Stupid term for a TDM411B. |
| 15:16.21 | [gnubie] | [TK]D-Fender: but definitely not related to configuration only? |
| 15:16.25 | [TK]D-Fender | JayTee52: If a normal phone is fine, the cards screwed |
| 15:16.27 | andreambrosio | i've read it |
| 15:16.38 | [TK]D-Fender | [gnubie]: How are gains on it? |
| 15:16.38 | JayTee52 | ah, hadn't seen that entry |
| 15:16.38 | andreambrosio | however, there seems to be a work around |
| 15:16.53 | [gnubie] | [TK]D-Fender: wait, i'll check |
| 15:17.00 | JayTee52 | yep, if the line is good with a standard analog then the noise is on the card. |
| 15:17.07 | [TK]D-Fender | [gnubie]: If its straight noise, its 99% surely a bad card / module |
| 15:17.25 | LemensTS | <PROTECTED> |
| 15:17.27 | andreambrosio | and for the presence stuff, since I'm under freepbx and not editing conf files, I'm a little lost |
| 15:17.40 | LemensTS | but its not updating my MYSQL table with TOM.... |
| 15:17.49 | [gnubie] | [TK]D-Fender: it's not a straight noise.. sometimes it stops then after sometime it goes back |
| 15:18.03 | [TK]D-Fender | [gnubie]: I'm talking "static-like" |
| 15:18.34 | andreambrosio | the exten => ext#,hint,SIP/peername decalration maust be put into extensions.conf file, but under any specific context?? |
| 15:18.44 | andreambrosio | or I just let it loose anywhere? |
| 15:18.50 | [gnubie] | [TK]D-Fender: where can i find again the gains? is it inside the /etc/fxotune.conf ? |
| 15:19.05 | JayTee52 | zapata.conf |
| 15:19.28 | [TK]D-Fender | andreambrosio: in a context specified by the watcher's "subscribecontext" if present, otherwise in, or linked to their "context" |
| 15:19.53 | [gnubie] | currently, the configuration that i have for rxgain and txgain of my zapata.conf is 0.0 |
| 15:20.06 | [TK]D-Fender | [gnubie]: And those levels sound alright volume-wise? |
| 15:20.17 | andreambrosio | wow... first, how do I do this name stuff, it seems you may better read my messages if your nick is in front of my messages... |
| 15:20.21 | [gnubie] | [TK]D-Fender: volume wise, yes |
| 15:20.28 | [TK]D-Fender | [gnubie]: With "normal" volume set on any phones you sue with them. |
| 15:20.31 | *** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb) |
| 15:20.35 | LemensTS | this is actually the whole thing in case thier was a null value in mysql that was preventing it: -- Executing MYSQL("Zap/1-1", "Query resultid 10 INSERT INTO info SET fullname=Tom, inout=in, timestamp=1207754356, notes=hi, ipaddress=207.14.218.34") |
| 15:20.37 | [TK]D-Fender | [gnubie]: Then you've got a bad card / module. |
| 15:20.40 | andreambrosio | and second, I'm sorry, but I'm kind of a newbie, so I didn't quite get what you meant |
| 15:21.20 | [TK]D-Fender | andreambrosio: those are the 2 parameters in sip.conf that will determin which context in extensions.conf you will need to place your "hint" in. |
| 15:21.37 | [gnubie] | [TK]D-Fender: sorry, i don't understand on what you said for: => With "normal" volume set on any phones you sue with them. |
| 15:22.24 | [TK]D-Fender | [gnubie]: I'm talking about using a phone with * that isn't jacked up to compensate for a low line for instance. One which when set at an average volume sounds "normal" when using that line. |
| 15:22.51 | [TK]D-Fender | [gnubie]: Which you pretty much confirmed to be the case. Call Digium for support, and they will probably RMA the card. |
| 15:22.55 | [gnubie] | [TK]D-Fender: ever since i used this card to my asterisk box, i already have this noise problem.. i just ignored it.. but lately, it gets too noisy already |
| 15:23.46 | andreambrosio | [TK]D-Fender: sorry, I just don't get it... |
| 15:24.36 | [gnubie] | [TK]D-Fender: i'm in asia and i don't have a termination in the US. i also moved out from my home country where i bought this digium dev kit.. |
| 15:24.39 | [TK]D-Fender | andreambrosio: if you put "subscribecontext=myhintsarehere" then when your phone subscribes to a hint, it will look in the [myhintsarehere] context in extensions.conf. |
| 15:25.08 | *** join/#asterisk Claude2008 (n=Claude81@87-194-221-2.bethere.co.uk) |
| 15:25.08 | [TK]D-Fender | andreambrosio: If you DON'T set that value, then it will look in the phone's "context=" entry |
| 15:25.24 | [gnubie] | anyone here from digium? |
| 15:25.32 | [gnubie] | :( |
| 15:25.33 | [TK]D-Fender | [gnubie]: Call them. its free |
| 15:25.34 | andreambrosio | [TK]D-Fender: hum, I'm staaaaarting to get it... |
| 15:26.04 | [gnubie] | [TK]D-Fender: you mean, they have a toll free number? |
| 15:26.15 | andreambrosio | [TK]D-Fender: actually, I'm sorry, I think I was not supposed to deal with this kind of complicated stuff before knowing better how asterisk works... |
| 15:26.25 | [TK]D-Fender | [gnubie]: yes, or you can call them right over IAX2 |
| 15:26.39 | [gnubie] | i'll check their website |
| 15:26.45 | [gnubie] | [TK]D-Fender: thank you.. :) |
| 15:26.57 | [TK]D-Fender | andreambrosio: Indeed. If you don't even know how to set up a SIP peer, you're definitely in trouble. |
| 15:27.33 | andreambrosio | [TK]D-Fender: well, that I do know |
| 15:29.15 | *** join/#asterisk blq (n=Bl@dslb-088-064-141-136.pools.arcor-ip.net) |
| 15:29.15 | andreambrosio | [TK]D-Fender: let's say I want extension 301 and 302 to be detected when they're on line... the line on extension.conf would then be exten => 301,hint,301 right? |
| 15:29.15 | andreambrosio | and same for 302 |
| 15:29.38 | [TK]D-Fender | andreambrosio: "exten => 301,hint,SIP/301" assuming that your sip peer for that extensions in [301] |
| 15:31.20 | andreambrosio | [TK]D-Fender: hum, great... then since I'm dealing with freepbx, I'd have to put these lines, let's say, in extensions_custom.conf, right again? |
| 15:31.53 | [TK]D-Fender | andreambrosio: Ok, FreePBX? You're pretty much screwed. |
| 15:31.58 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:32.11 | [TK]D-Fender | andreambrosio: Go ask in their channel if there is a way to do this. |
| 15:32.14 | [TK]D-Fender | ~freepbx |
| 15:32.14 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:32.29 | andreambrosio | [TK]D-Fender: really??? is it that bad?? |
| 15:32.36 | andreambrosio | [TK]D-Fender: ok, thank you very much |
| 15:32.47 | andreambrosio | [TK]D-Fender: and sorry to take your time... |
| 15:32.48 | [TK]D-Fender | andreambrosio: it controls everything and you may not be able to do things the way you want with it. |
| 15:34.13 | *** join/#asterisk Faithful (n=Faithful@vg102.vodafone.com.au) |
| 15:34.59 | [gnubie] | i can't find the 1800 number of digium.. :( |
| 15:35.10 | [gnubie] | their iaxtel accounts are always busy |
| 15:35.11 | andreambrosio | [TK]D-Fender: thanks again, it was really helpful |
| 15:35.55 | [TK]D-Fender | [gnubie]: its in the sample extensions.conf..... |
| 15:36.14 | [TK]D-Fender | [gnubie]: the one that contacts "misery.digium.com" |
| 15:36.14 | [gnubie] | [TK]D-Fender: let me check.. thanks again.. |
| 15:36.47 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:36.59 | [TK]D-Fender | [gnubie]: 877.DIGIUM1 (toll free) or |
| 15:37.04 | [TK]D-Fender | [gnubie]: Off their site |
| 15:40.26 | [gnubie] | [TK]D-Fender: 877 only if within the US continent |
| 15:40.42 | [TK]D-Fender | [gnubie]: And you are where? |
| 15:40.44 | [gnubie] | i mean, north american continent |
| 15:40.47 | [gnubie] | asia |
| 15:40.59 | [TK]D-Fender | [gnubie]: Then go find the IAX2 dial command |
| 15:41.11 | [gnubie] | ok |
| 15:41.33 | AsteriskRo | any other irc channel to ask basic questions? |
| 15:43.21 | AsteriskRo | i have basic problems with dialplan |
| 15:44.27 | UnixDog | look at the asterisk source dir |
| 15:44.34 | UnixDog | there is a dir called configs |
| 15:44.45 | UnixDog | look at the extensions.conf.sample |
| 15:45.08 | UnixDog | it has alot to give you basics |
| 15:45.29 | *** join/#asterisk esk (i=esk@esk.cs.usu.edu) |
| 15:47.15 | LemensTS | i can insert sql statment directly in phpmyadmin on localhost, but from asterisk dialplan it wont (its on the same box, localhost)...i dont know whats wrong |
| 15:49.31 | *** join/#asterisk MicW (n=michael@dslb-088-074-131-117.pools.arcor-ip.net) |
| 15:50.04 | MicW | hi |
| 15:50.15 | UnixDog | password |
| 15:50.17 | UnixDog | username |
| 15:50.48 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 15:50.55 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 15:51.01 | UnixDog | database connector not configure correctly |
| 15:51.39 | MicW | i have different ring groups implemented in the following way: a number of phones is dialed with a timeout. after the timeout a bigger number of phones is dialed. the problem is that sometimes the phones rings very strange when switching from one group to the next (the phones are snom 320 and 360) |
| 15:51.56 | LemensTS | UnixDog: its connecting to the database for sure. |
| 15:52.12 | LemensTS | if i type the password wrong to test it gives an error is how i know |
| 15:52.38 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 15:55.31 | UnixDog | ok |
| 15:56.24 | rupa | update: voicepulse will do the callerid name database dip *IF* you use the connect0[1-3] rather than the nyc or sfo gateways |
| 16:01.26 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 16:02.26 | *** join/#asterisk BingoPajama (n=chatzill@38.98.105.130) |
| 16:02.58 | *** join/#asterisk kyron (n=kyron@modemcable086.140-70-69.static.videotron.ca) |
| 16:03.12 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:03.14 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:04.21 | [TK]D-Fender | rupa: Thats worthy of a WIKI update. |
| 16:04.40 | rupa | ~wiki |
| 16:04.41 | Qwell | rupa: they don't otherwise? |
| 16:04.43 | Qwell | that's...dumb |
| 16:04.45 | Qwell | ~wikis |
| 16:04.46 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:04.47 | *** part/#asterisk roe_ (n=roe___@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 16:04.52 | rupa | ok, I"ll go update |
| 16:05.22 | rupa | Qwell, i'm in the impression that connect0[1-3] have more features than the official ones |
| 16:08.33 | [gnubie] | [TK]D-Fender: i just e-mailed the support@digium.com |
| 16:08.55 | [gnubie] | [TK]D-Fender: last question, is there any tool i can use to test my digium dev kit card? |
| 16:09.28 | [TK]D-Fender | [gnubie]: pastebin "cat /proc/interrupts" and "dmesg" |
| 16:09.34 | Qwell | are we still selling those as a dev kit? O.o |
| 16:10.28 | rupa | are changes "global" variables in the dialplan visible across all threads or just visible in the thread where the change occurred? |
| 16:11.33 | [TK]D-Fender | rupa: Global = all |
| 16:11.57 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:12.30 | rupa | bummer, ok, so my hack to handle hangups within a call macro isn't gonna work well. The real solution is to handle the 'h' extension within a macro but that looks to be more code than I'm comfortable doing... |
| 16:13.07 | *** join/#asterisk willianmazzardo (n=willianm@189-30-238-14.smace701.dsl.brasiltelecom.net.br) |
| 16:13.36 | [gnubie] | [TK]D-Fender: wait, i'll reboot.. i got a messy dmesg output |
| 16:13.40 | [gnubie] | brb |
| 16:13.42 | *** part/#asterisk [gnubie] (n=[gnubie]@cm85.omega113.maxonline.com.sg) |
| 16:15.20 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 16:15.34 | s0ck | anyone played with chan_mobile? |
| 16:15.39 | s0ck | via bluetooth... |
| 16:16.11 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 16:16.57 | *** part/#asterisk AsteriskRo (n=Asterisk@190.36.185.138) |
| 16:19.21 | *** join/#asterisk spjuden (n=pithen@mail.graphlogic.com) |
| 16:19.33 | *** join/#asterisk [gnubie] (n=[gnubie]@cm85.omega113.maxonline.com.sg) |
| 16:19.38 | [gnubie] | waves |
| 16:20.15 | [gnubie] | [TK]D-Fender: still there? kindly check this out => http://www.privatepaste.com/611Ip202dL |
| 16:22.55 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 16:23.16 | *** join/#asterisk JerJer[mobile] (n=jj@45.210.254.244) |
| 16:23.38 | spjuden | Whats the best way to create a mapping between user extensions and their actual zap channels? Do I just make a 'exten => 112,1,..." line for each one in the dialplan? |
| 16:24.18 | tzafrir | spjuden, yes |
| 16:24.26 | [TK]D-Fender | [gnubie]: Nope, everything looks fine |
| 16:25.17 | [gnubie] | [TK]D-Fender: does it mean that there's no problem with my card? |
| 16:25.22 | JayTee52 | spjuden, I'm doing that now with extensions dedicated to a specific zap channel. I could pastebin my zapata.conf and extensions.conf if you'd like to see it. |
| 16:25.42 | spjuden | tzafrir, seems like there would be a cleaner way...with SIP i can just _XXX, Dial(SIP/${EXTEN}) |
| 16:25.49 | *** join/#asterisk kyron_ (n=kyron@modemcable086.140-70-69.static.videotron.ca) |
| 16:26.59 | tzafrir | spjuden, I often use something like _XXX,1,Dial(Zap/6${EXTEN:1}) |
| 16:27.16 | spjuden | JayTee52, I'm actually doing that now, but thank you |
| 16:27.25 | JayTee52 | yw |
| 16:27.35 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 16:27.41 | tzafrir | err |
| 16:27.52 | tzafrir | something like _6XXX,1,Dial(Zap/${EXTEN:1}) |
| 16:28.15 | Qwell | tzafrir: I think he means fxs |
| 16:28.22 | spjuden | so extension 6001 goes to ZAP/1 |
| 16:28.24 | spjuden | etc |
| 16:28.34 | JayTee52 | what I did was make a context for each zap channel called [channel1out] and so on for each channel and put the pattern matching in there and then do an include in each extension context for that channel. |
| 16:30.12 | [gnubie] | anyway, i have to leave now.. it's late already in my place |
| 16:30.18 | *** join/#asterisk CVirus (n=GoD@82.201.178.99) |
| 16:30.20 | [gnubie] | [TK]D-Fender: thank you once again.. |
| 16:30.24 | [gnubie] | waves to all.. |
| 16:30.27 | *** part/#asterisk [gnubie] (n=[gnubie]@cm85.omega113.maxonline.com.sg) |
| 16:30.28 | tzafrir | JayTee52, this is for dialing to the phones, You describe how to dial from it |
| 16:31.31 | JayTee52 | tzafrir, well in my situation I have each extension's zap channel in zapata.conf point to it's own context and in that context just dial the SIP phone. |
| 16:31.37 | *** join/#asterisk logyati (n=suporte@201.29.54.95) |
| 16:31.44 | logyati | hello! |
| 16:31.52 | UnixDog | yello |
| 16:32.00 | logyati | can * do rtp proxy? |
| 16:32.35 | logyati | media proxy |
| 16:32.57 | spjuden | If I were to add zap users via users.conf, does that just dynamically create the individual mappings in the default context? |
| 16:33.01 | [TK]D-Fender | logyati: thats the default w/o reinvites |
| 16:33.18 | [TK]D-Fender | logyati: thought not an actual "proxy" so much as B2BUA |
| 16:33.31 | logyati | so i just need to setup canreinvite=no? |
| 16:36.32 | Jumpie | ugh my machine takes forever to burn cd rw |
| 16:37.00 | UnixDog | get a newer cdrw/dvdrw |
| 16:37.54 | *** join/#asterisk bminish (n=bminish@89.19.72.24) |
| 16:38.27 | spjuden | Why can't users.conf be available realtime as well? |
| 16:38.35 | Jumpie | its brand new |
| 16:38.37 | Jumpie | just crappy |
| 16:38.44 | Jumpie | taking me like 20 mins per disc on these centos disks |
| 16:39.21 | UnixDog | well thats what you get for burning centos cd's |
| 16:39.38 | Qwell | freebsd CDs burn at least 30x faster |
| 16:40.06 | *** join/#asterisk Arcu (n=ilvantus@67.108.111.143.ptr.us.xo.net) |
| 16:40.23 | UnixDog | I use 1 cd |
| 16:41.08 | UnixDog | and where possiable I use my freebsd usb drive to do installs |
| 16:42.00 | Jumpie | lol |
| 16:42.18 | Jumpie | freebsd was the very first *nix os i messed with |
| 16:42.22 | Jumpie | was in 1999 |
| 16:42.24 | UnixDog | usb thumb drive that is |
| 16:42.34 | Jumpie | then i had to adapt to solaris |
| 16:42.35 | Jumpie | :D |
| 16:42.47 | Qwell | sounds like a step up |
| 16:43.03 | UnixDog | slow laris |
| 16:43.36 | spjuden | heh..thats what we used to call it in college...then we managed to replace the whole slowaris lab with linux |
| 16:45.12 | spjuden | now i think its a windows lab because "computer scientists dont need to know *nix any more" |
| 16:46.08 | Jumpie | i think certain government agencies/applications needed to be on solaris |
| 16:46.12 | Jumpie | alot of homegrown db stuff |
| 16:46.16 | UnixDog | thats because with windows they dont teach you the backside of life |
| 16:46.19 | Jumpie | and didnt wanna spend time/money to convert |
| 16:46.28 | UnixDog | they think the gui os is the key to the world |
| 16:46.30 | Jumpie | bah never wrong with being well rounded |
| 16:46.46 | UnixDog | *nix is the best for servers |
| 16:46.53 | UnixDog | windows for a server is crap |
| 16:46.59 | UnixDog | imho |
| 16:47.15 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:47.34 | spjuden | agreed... im trying to talk my boss into switch out of 2003 and going to a samba pdc before we have to switch to the next windows version |
| 16:47.48 | Jumpie | lol |
| 16:48.13 | spjuden | i dont want to manage a vista pc, never mind a vista server |
| 16:48.31 | UnixDog | lol |
| 16:48.59 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 16:49.33 | *** join/#asterisk xenonex (n=xenonex@82.200.211.72) |
| 16:53.29 | DirtyD | I can call rxfax from my Dialplan. It answers, I hear fax tones. When I try faxing, the fax fails, and I see an error message in /var/log/rxfax.log (lost carrier). When I installed span-dsp, I did so from rpm. I did the ./build.sh install for agx-ast-addons. Does this mean that everything is installed right? Was I suppoed to patch something? I'm using astering 1.4.19 |
| 16:55.00 | DirtyD | What is the purpose of patching asterisk source when it comes to using app_txfax or app_rxfax? |
| 16:55.37 | *** join/#asterisk GBLRA77 (n=GBLRA77@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 16:57.19 | Jumpie | argh it barely needed disc 3 4 5 |
| 17:02.24 | [TK]D-Fender | Jumpie: </bitch></whine></nag></complain> |
| 17:02.42 | DirtyD | do I have to patch asterisk 1.4.19 to get spandsp working? |
| 17:02.53 | DirtyD | err, to get app_rxfax.c working? |
| 17:03.59 | rupa | [TK]D-Fender, advise. I modified app_dial.c to have a new option to not tear down the caller when the callee hangs up during the execution of a macro. I also have a minimal dialplan that shows how to use it. What do I do with this? It is currently against 1.4.x. Do I just keep it as a private patch or something else? |
| 17:03.59 | tzafrir | have you tried building it in the agx-addons? |
| 17:04.08 | DirtyD | tzafrir: yes. |
| 17:04.12 | tzafrir | and? |
| 17:04.50 | DirtyD | it works, I can call the rxfax extension over my T1. I hear fax tones. But when I fax, it always fails with "carrier timeout" error. |
| 17:05.07 | [TK]D-Fender | rupa: "g" doesn't work for you? |
| 17:05.33 | rupa | [TK]D-Fender, no, the caller gets supervised right away |
| 17:05.49 | [TK]D-Fender | rupa: ? |
| 17:06.14 | Jumpie | heh |
| 17:06.19 | rupa | [TK]D-Fender, the caller is answered if the macro hangs up -- no way to prevent that. right after the macro returns, the result is checked and res is assigned to 0 which causes the caller to be answered. |
| 17:06.38 | rupa | [TK]D-Fender, I need to keep res=-1 so the caller isn't answerd |
| 17:07.41 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:08.40 | [TK]D-Fender | rupa: That makes no sense. caller doesnt' get answered unless you do it explicitly, dial bridges, or the channel plays audio. |
| 17:09.41 | rupa | [TK]D-Fender, I agree, but unfortunately, if you hang up during the execution of the macro the caller is answered right away. I can come up with a minimal dialplan that demonstrates it. |
| 17:09.58 | rupa | or if you don't have g, then the caller is hung up on |
| 17:10.02 | [TK]D-Fender | rupa: please do. |
| 17:10.11 | rupa | ok, gimme a bit to get it |
| 17:10.21 | [TK]D-Fender | rupa: pastebin it all up, code & CLI |
| 17:10.27 | rupa | ok |
| 17:10.51 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 17:14.21 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:15.52 | *** join/#asterisk xenonex (n=xenonex@82.200.211.72) |
| 17:17.04 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:18.23 | JerJer[mobile] | has anyone seen cdr_csv like not log calls? |
| 17:18.54 | JerJer[mobile] | i just upgrade a box for a customer and now cdr_csv is totally not writing out any CDRs - but its loaded |
| 17:19.42 | tzafrir | cdr_custom? |
| 17:20.17 | tzafrir | cdr status? |
| 17:20.50 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 17:22.55 | Jumpie | ah i like this build |
| 17:23.08 | JerJer[mobile] | cdr_csv |
| 17:24.10 | JerJer[mobile] | CDR logging: enabled |
| 17:24.10 | JerJer[mobile] | CDR mode: simple |
| 17:24.10 | JerJer[mobile] | CDR output unanswered calls: no |
| 17:25.02 | JerJer[mobile] | is cdr.conf required now ? |
| 17:28.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:32.44 | Jumpie | aight its all ready...lol |
| 17:34.06 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 17:35.38 | *** join/#asterisk drcode (n=user1@85.65.12.156.dynamic.barak-online.net) |
| 17:35.46 | drcode | hi all |
| 17:35.52 | drcode | I need help with QOS on sip |
| 17:37.51 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:38.05 | drcode | any one up |
| 17:42.29 | Jumpie | this is a huge updatelol |
| 17:42.36 | *** join/#asterisk beasty (n=beasty@about/apple/macbook/beasty) |
| 17:42.39 | beasty | morning y'all |
| 17:43.04 | beasty | i'm pondering if it's possible to set the 'incomming number from a user'? |
| 17:46.22 | jjshoe | mornin |
| 17:46.39 | Qwell | jjshoe: you're missing out on the fun of updating astcli |
| 17:46.52 | Qwell | it's already been committed, and had a few enhancements added |
| 17:46.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:47.02 | jjshoe | Qwell awesome, what's the link to it again? |
| 17:47.10 | Qwell | bugs are closed |
| 17:47.14 | Qwell | it's in trunk |
| 17:47.18 | Qwell | (and 1.6) |
| 17:47.21 | Qwell | ((I think)) |
| 17:47.27 | lirakis_work | i hate nextone |
| 17:47.40 | Qwell | jjshoe: if you can get that approved, just open a new bug with changes |
| 17:47.53 | jjshoe | http://svn.digium.com/svn/asterisk/trunk/contrib/scripts/astcli ? |
| 17:47.54 | rupa | [TK]D-Fender, http://www.pastebin.ca/978574 |
| 17:47.57 | Qwell | yeah |
| 17:48.15 | Qwell | added a '-' command for looping, like asterisk -r |
| 17:49.17 | rupa | Qwell, if you wanna look at that paste as well I'd appreciate it. Implementing locate me type functionality and having trouble dealing with hangup from within a dial() macro |
| 17:49.37 | drcode | I have internet -> Linux -> SIP device -> phone , I want to put QOS in my linuz , any idea |
| 17:49.38 | drcode | ? |
| 17:50.17 | rupa | you want QOS on the sip channel or the RTP channel? probably the latter is more noticeable to users |
| 17:50.26 | drcode | yes |
| 17:50.41 | drcode | I want to give sip qos |
| 17:50.48 | drcode | I did somthing but it dosnt work well |
| 17:51.03 | rupa | drcode, what part doesn't work well? |
| 17:51.19 | drcode | I will past what I did |
| 17:51.21 | jjshoe | Qwell I see a bug with that version |
| 17:51.31 | Qwell | what's that? |
| 17:51.39 | jjshoe | Qwell the file handle doesn't get closed for the config until exit |
| 17:51.50 | rupa | drcode, what is the symptoms you are experiencing that requires you to do QOS? |
| 17:52.10 | Qwell | jjshoe: process_credentials? |
| 17:52.18 | *** join/#asterisk qdk_ (n=qdk@195.242.194.42) |
| 17:52.33 | drcode | When I dowload from internet there is bad audio on my phone |
| 17:52.42 | rupa | drcode, that isn't a SIP problem that is RTP |
| 17:52.46 | Qwell | what's that then, just close ($fh)? |
| 17:53.05 | drcode | tc qdisc add dev [My Wan] root handle 1: prio priomap 2 2 2 2 2 2 2 2 1 1 1 1 1 1 1 0 |
| 17:53.20 | rupa | SIP is control channel, RTP is audio channel |
| 17:53.27 | jjshoe | Qwell yes, close($fh) |
| 17:53.33 | jjshoe | after the while |
| 17:53.36 | drcode | tc filter add dev [My Wan] protocol ip parent 1: prio 1 u32 match ip dport 5060 0xffff flowid 1:1 |
| 17:53.38 | Jumpie | hey guys |
| 17:53.38 | Qwell | right |
| 17:53.45 | Qwell | yay, my perl doesn't suck :p |
| 17:53.50 | Jumpie | for a 50 concurrent call or less 'baseline' server |
| 17:53.56 | jjshoe | Qwell it's probably not an issue, but it could cause unexpected behaviour |
| 17:54.00 | Jumpie | whats some good hardware specs im lookin at for a server? |
| 17:54.01 | drcode | tc filter add dev [My Wan] protocol ip parent 1: prio 1 u32 match ip sport 5060 0xffff flowid 1:1 |
| 17:54.09 | drcode | 5060 is rtp or SIP ? |
| 17:54.17 | Jumpie | raid, 2gb ram, how many ghz cpu and what kinda hard disk space? |
| 17:54.17 | jjshoe | drcode usually sip. |
| 17:54.31 | drcode | did I put it corect? |
| 17:54.34 | drcode | the QOS??? |
| 17:54.47 | rupa | drcode, look in your rtp.conf -- you can configure your port range. then do QOS on those ports. rtp is UDP |
| 17:54.58 | rupa | drcode: also, don't past to the channel |
| 17:55.02 | drcode | k |
| 17:55.11 | drcode | rtp.conf where? |
| 17:55.13 | drcode | in tc/ |
| 17:55.14 | drcode | ? |
| 17:55.20 | rupa | /etc/asterisk/rpt.conf |
| 17:55.32 | rupa | err, rtp.conf :) |
| 17:55.53 | drcode | I can put range or ports with tc? |
| 17:56.00 | drcode | or rtp protocol? |
| 17:56.47 | rupa | you define the range for asterisk in /etc/asterisk/rtp.conf. You then configure your QOS implementation to prioritize those ports with protocol UDP. I don't use tc directly, so can't help you there. |
| 17:57.01 | rupa | uses tomato firmware on a linksys wrt54l |
| 17:57.04 | *** join/#asterisk Skarmeth (n=Skarmeth@iris.aspec.com.br) |
| 17:57.46 | drcode | k |
| 17:58.19 | rupa | you can also prioritize SIP which will help call control, but that isn't as critical |
| 17:58.25 | Qwell | jjshoe: fixed |
| 17:58.29 | jjshoe | Qwell cool |
| 17:58.54 | drcode | I see |
| 17:59.05 | drcode | the problem is more on rtp |
| 17:59.17 | rupa | drcode, right |
| 17:59.48 | drcode | k |
| 17:59.51 | drcode | thnx |
| 18:01.32 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:05.45 | jjshoe | Qwell the only thing I pondered in my head... would be building in the tab completion :D |
| 18:06.14 | jjshoe | Qwell I know samy has it in sql++ though, so I may take a look at how it does that. |
| 18:07.31 | *** join/#asterisk s0lid (n=s0lid@210.213.198.61) |
| 18:10.53 | *** join/#asterisk asdx (n=diego@adsl-152-102.click.com.py) |
| 18:10.54 | asdx | hi |
| 18:11.01 | asdx | anyone knows if teliax do sip as well? |
| 18:11.11 | asdx | is teliax a good option? |
| 18:11.41 | *** join/#asterisk equanimity (n=alex@cpc2-oxfd7-0-0-cust208.oxfd.cable.ntl.com) |
| 18:12.22 | [TK]D-Fender | rupa: -- SIP/rlaptop-081ef130 answered Local/105@phones-0a35,2 |
| 18:12.31 | [TK]D-Fender | rupa: I think Screen force answers it |
| 18:12.40 | Qwell | jjshoe: I thought about tab completion too |
| 18:12.53 | Qwell | was thinking maybe use the output of "help", but *shrug* |
| 18:13.55 | rupa | [TK]D-Fender, If a Macro returns from pbx_exec a non-zero response, dial() forces res=0 which then falls through as an answer even though the callee side has hung up. That is the behavior I modified. |
| 18:14.53 | [TK]D-Fender | rupa: Ok, more than I can comment on ATM. Post it to Mantis... |
| 18:15.10 | rupa | [TK]D-Fender, ok, no problem. Mantis? |
| 18:15.29 | rupa | oh, ticket system |
| 18:15.30 | [TK]D-Fender | rupa: the * bugtracker |
| 18:15.32 | Qwell | bugs.digium.com |
| 18:15.54 | Nugget | bugs file |
| 18:16.01 | Qwell | files bug |
| 18:16.15 | jjshoe | Qwell yes, help is where I would get the available tab completeable items |
| 18:16.20 | file | pokes Nugget |
| 18:16.52 | Jumpie | is quad core xeon 2.3 sufficient for a 20 call or less system? |
| 18:17.02 | file | Nugget: don't make me... ummm... get equiraptor to thwap you later! |
| 18:17.10 | Qwell | ~thwap nugget |
| 18:17.10 | jbot | ACTION thwaps nugget on the nose with a 2 by 4 |
| 18:17.11 | _ShrikE | Jumpie: more than |
| 18:17.12 | Qwell | does it now |
| 18:17.26 | Jumpie | thanks |
| 18:17.33 | *** join/#asterisk r0d3nt (i=r0d3nt@foster.stonedcoder.org) |
| 18:17.36 | Jumpie | 2, vs 4 gb ram? |
| 18:17.48 | Nugget | ow! |
| 18:17.50 | Jumpie | 4 probably, minor price diff |
| 18:17.55 | Qwell | Jumpie: for 20 calls? both are overkill |
| 18:18.02 | *** join/#asterisk ManxPower (n=manxpowe@110.sub-75-203-113.myvzw.com) |
| 18:18.03 | Qwell | extreme overkill |
| 18:18.03 | Nugget | You can never have too much ram. |
| 18:18.07 | _ShrikE | Jumpie: like qwell said.. 2gb would be fine |
| 18:18.15 | drcode | I can't tell TC to use ports 10000 - 100050? |
| 18:18.20 | drcode | sport or dport? |
| 18:18.21 | Jumpie | thanks |
| 18:18.32 | Jumpie | ak |
| 18:18.40 | Jumpie | ah, i found some good 1ru on pbxselect.com |
| 18:18.54 | rupa | drcode, you only have control of outbound -- so sport |
| 18:19.01 | Jumpie | wodcrest 5.110 4mb cache, 2gb ram, dual onboard 1000 nic, four 160gb |
| 18:19.03 | drcode | k |
| 18:19.05 | Jumpie | for $1800 |
| 18:19.13 | drcode | I know rtp is from 10000 to 100050 |
| 18:19.15 | ManxPower | I believe max port number for TCP/UDP is 65536 and 100050 is larger than 65536 |
| 18:19.21 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 18:19.31 | drcode | 10050 |
| 18:19.43 | Jumpie | i thought range was 10000 to 2000 |
| 18:19.44 | ManxPower | next time don't typoe it twice |
| 18:19.44 | Jumpie | er 20000 |
| 18:20.21 | drcode | k |
| 18:21.19 | Jumpie | http://www.pbxselect.com/supermicro-highend-asterisk-server-hardware-only-p-557.html |
| 18:21.56 | rupa | the range is what you set in rtp.conf |
| 18:22.04 | drcode | k |
| 18:22.05 | Jumpie | yup |
| 18:22.22 | drcode | but TC in linux can I put range of ports? |
| 18:23.43 | asdx | [Apr 9 13:23:35] WARNING[7356]: chan_sip.c:1785 __sip_xmit: sip_xmit of 0x81d7460 (len 399) to 63.211.239.14:5060 returned -1: Operation not permitted |
| 18:23.53 | asdx | i'm getting that on the asterisk cli all the time |
| 18:23.54 | asdx | why? |
| 18:24.31 | ManxPower | drcode: you might ask on a LINUX channel |
| 18:24.55 | drcode | k |
| 18:24.56 | drcode | thanx |
| 18:25.10 | *** join/#asterisk l2cache (n=l2cache@117.178.101.97.cfl.res.rr.com) |
| 18:25.27 | rupa | drcode, or instead of using tc directly use a firewall script that has QOS support. there are lots. A bit easier to setup. |
| 18:25.42 | l2cache | Does anyone have any experience with tshark or tethereal and capturing RTP payload packets? |
| 18:26.19 | drcode | rupa: in linux ? what? |
| 18:26.22 | plik | hmmmm, it says "SIP Trunking is poised to challenge legacy TDM (Time Division Multiplexing) and fixed line circuits for the establishment and transmission of voice communications." :/ |
| 18:26.45 | plik | at http://www.pbxselect.com/sip-trunking.php?osCsid=c329f0e40b854934259cbb28d4710382 |
| 18:26.52 | rupa | drcode, google. depending on your distro there should even be packages. |
| 18:27.03 | drcode | k |
| 18:27.05 | drcode | thnx |
| 18:27.08 | UnixDog | overkill for a asterisk server |
| 18:27.09 | ManxPower | plik: Much like The Easter Bunny will replace St. Mark in catholic saints. One exists, one does not. |
| 18:27.36 | ManxPower | ~trunk |
| 18:27.37 | jbot | i heard trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 18:27.45 | plik | but how are he poor newbs supposed to learn |
| 18:28.07 | Qwell | pbxselect = voip supply. fun |
| 18:28.12 | ManxPower | That site is like a poster boy for using marketing terms and pretending they are technical terms |
| 18:28.19 | Jumpie | if its so wrong why would trixbox devs put that on there? |
| 18:28.22 | Jumpie | lol doesnt that show ignorance |
| 18:28.28 | ManxPower | plik: learn what? |
| 18:28.31 | Jumpie | i understand the concept of 'sip trunk' though |
| 18:28.52 | ManxPower | Jumpie: exactly how is a "sip trunk" different from any other SIP gateway? |
| 18:29.01 | Jumpie | its not different |
| 18:29.04 | Jumpie | its just how they word it i guess |
| 18:29.13 | jjshoe | Qwell lots of noise in this channel :\ |
| 18:29.21 | ManxPower | "trunk" implies you can only send one call at a time over the connection. |
| 18:29.21 | Jumpie | im not cheerleading the term, just saying i kknew what they meant when i was setting it up in trix earlier |
| 18:29.22 | asdx | i put the register => user:pass@den.teliax.net on my sip.conf (under general) but it's trowing some warnings |
| 18:29.25 | asdx | anyone? |
| 18:29.32 | plik | ManxPower: erm.. anything properly, witch so much wrongness erporting to be from a trustworthy source |
| 18:29.42 | Jumpie | asdx does your box resolve that host |
| 18:29.55 | plik | s/witch/with/ |
| 18:30.01 | [TK]D-Fender | asdx: Show us the warnings. |
| 18:30.01 | asdx | Jumpie: yes |
| 18:30.02 | ManxPower | asdx: too bad you didn't tell us the error message or we might have been able to help. |
| 18:30.15 | asdx | i already did |
| 18:30.18 | jameswf-home | woohoo it it troll time |
| 18:30.18 | asdx | but i will do it again |
| 18:30.20 | asdx | 1min |
| 18:30.25 | asdx | [Apr 9 13:28:08] WARNING[23757]: chan_sip.c:1785 __sip_xmit: sip_xmit of 0x81d7ad8 (len 399) to 63.211.239.14:5060 returned -1: Operation not permitted |
| 18:30.29 | asdx | [Apr 9 13:28:09] WARNING[23757]: chan_sip.c:1785 __sip_xmit: sip_xmit of 0x81d7ad8 (len 399) to 63.211.239.14:5060 returned -1: Operation not permitted |
| 18:30.42 | [TK]D-Fender | asdx: pastebin the complete CLI output including SIP DEBUG. |
| 18:30.49 | asdx | [TK]D-Fender: ok |
| 18:30.59 | ManxPower | that doesn't really have anything to do with "register => user:pass@den.teliax.net on my sip.conf" |
| 18:30.59 | *** join/#asterisk chigital (n=chigital@p4FE5FD2D.dip.t-dialin.net) |
| 18:31.40 | x86 | I've got an issue with echo cancellation only happening on certain channels |
| 18:31.53 | x86 | here's my zapata.conf: http://pastebin.ca/978646 |
| 18:32.15 | x86 | all chanels seem to be OK except channel 37 |
| 18:32.33 | x86 | 25 is perfectly fine |
| 18:32.43 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 18:33.06 | jameswf-home | x86 HWEC os soft |
| 18:33.07 | x86 | when 37 bridges, and I do zap show channel 37, I get "Echo Cancellation: 128 taps, currently OFF |
| 18:33.09 | Jumpie | lol ManxPower |
| 18:33.24 | x86 | jameswf-home: english? |
| 18:33.30 | LemensTS | exten => _1231,n,MYSQL(Query resultid ${connid} INSERT\ INTO\ info\ SET\ fullname=Tom\, inout=in\, timestamp=${EPOCH}\, notes=hello\, ipaddress=207.14.218.34) |
| 18:33.36 | [TK]D-Fender | x86: "or" |
| 18:33.38 | jameswf-home | i it hardware EC |
| 18:33.41 | LemensTS | does that look ok? |
| 18:33.49 | x86 | [TK]D-Fender: ah... hardware |
| 18:33.59 | ManxPower | x86: do you have echocancelwhentdmbridged enabled/? |
| 18:34.09 | Qwell | s/tdm// |
| 18:34.17 | x86 | ManxPower: yep |
| 18:34.26 | x86 | (that's in my pastebin config as you can see) |
| 18:34.26 | ManxPower | x86: It sucks to be you. |
| 18:34.42 | ManxPower | Just remember that EC is always off for channels that are not off hook |
| 18:35.12 | x86 | ManxPower: 13:33 < x86> when 37 bridges, and I do zap show channel 37, I get "Echo Cancellation: 128 taps, currently OFF" |
| 18:35.23 | ManxPower | x86: does it also show off hook? |
| 18:35.26 | jameswf-home | EC only goes on for voice calls heading outside the server... all other times it is off |
| 18:35.26 | x86 | ManxPower: what I meant by "bridges" is "on an active call" |
| 18:35.32 | x86 | ManxPower: yep |
| 18:35.39 | jameswf-home | bridge is zap to zap |
| 18:35.59 | x86 | jameswf-home: bridge is any to any... in this case, zap to iax2 |
| 18:36.03 | ManxPower | I find that is trivial to fix echo problems with T-1 ports. |
| 18:36.41 | x86 | hmm |
| 18:36.43 | jameswf-home | 90% of T1 echo is actualy delay from a sip phone |
| 18:36.44 | ManxPower | x86: you are not echo canceling channels 1-23 |
| 18:36.58 | x86 | ManxPower: not worried about 1-23 at all |
| 18:37.02 | ManxPower | jameswf-home: no. k90% of T-1 echo is the FAR END analog |
| 18:37.03 | x86 | ManxPower: worried about 37 ;) |
| 18:37.11 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 18:37.22 | x86 | jameswf-home: not using SIP phones |
| 18:37.50 | ManxPower | x86: enable ec on 1-24 and see what happens |
| 18:37.51 | x86 | jameswf-home: analog phones connected to a rhino channel bank going T1 into this sangoma a104d-x |
| 18:37.53 | jameswf-home | EC works on the local hybrid with a T1 there is no local hybrid |
| 18:38.11 | ManxPower | jameswf-home: you sure are confused today. |
| 18:38.13 | x86 | jameswf-home: zap to iax2 is certainly hybrid ;) |
| 18:38.28 | ManxPower | x86: where hybrid = 2 wires |
| 18:38.42 | jameswf-home | would have to draw a picture and is a crappy artist |
| 18:38.57 | x86 | 2 wires = analog ;) |
| 18:39.14 | ManxPower | All calls with an analog portion have echo. You cannot perceive this echo if the audio delay is very low. |
| 18:39.57 | x86 | if you're doing analog to analog it's nothing |
| 18:40.08 | ManxPower | When you add VoIP to the mix, delay gets high enough you can hear the echo. |
| 18:40.17 | x86 | right |
| 18:40.23 | x86 | i understand how it works |
| 18:40.33 | ManxPower | Much like shouting in a small room (analog) or a big room (VoIP). Do you really think your voice is not bouncing off the walls in the small room? |
| 18:41.34 | outtolunc | duct tapes my mouth shut and bounces myself off the walls |
| 18:41.45 | ManxPower | x86: look at any odd config options on the channel bank for that channel |
| 18:42.09 | jameswf-home | Also EC is unidirectional. the Cancellation is always towared the box you cancel PSTN -> Asterisk NOT pstn <- Asterisk.. so if you have an EC on your t1 the cancelation is not servicing audio to the chanelbank it is servicing audio from the channelbank |
| 18:42.12 | x86 | I don't understand why channel 25 gets echo cancelled just fine but 37 does not |
| 18:42.48 | jameswf-home | has a huge office with big windows... cant bounce mutch |
| 18:42.49 | outtolunc | have you checked the logs to see if 37 is getting its ec disabled by the other end? |
| 18:43.04 | outtolunc | (or anywhere in the middle) |
| 18:43.18 | x86 | other end is IAX |
| 18:43.23 | ManxPower | a fax tone would disable EC |
| 18:43.59 | outtolunc | on some sites i have to use disable ec enable define (whatever the exact verbage is) |
| 18:44.13 | outtolunc | er enable ec disable <G> |
| 18:46.35 | outtolunc | zconfig.h #define NO_ECHOCAN_DISABLE |
| 18:46.40 | x86 | ManxPower: so if i just take that channel off-hook, EC should get enabled? or does it wait until after the call is bridged somewhere |
| 18:48.09 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-14-252.w90-11.abo.wanadoo.fr) |
| 18:49.44 | x86 | ok |
| 18:49.56 | x86 | got it to show that EC was on during the call |
| 18:50.03 | x86 | Echo Cancellation: 128 taps, currently ON |
| 18:50.20 | x86 | but you can still hear yourself |
| 18:50.31 | x86 | and the other end hears your echo as well |
| 18:51.09 | x86 | rxgain perhaps? |
| 18:51.11 | grandpapadot | Hi all. In my sip.conf, I have disallow=all, then allow=g729,gsm. How do I tell asterisk to always try g729 first before gsm? |
| 18:51.33 | x86 | grandpapadot: asterisk will always try the first codec listed first |
| 18:52.06 | grandpapadot | In this case it's not, it's using gsm where g729 is clearly avaiable if I just define it. This is 1.2.24. Hrmmm... |
| 18:52.11 | x86 | ManxPower: so what do I do if echo cancel is enabled but there is still echo on a bridged call |
| 18:52.26 | x86 | grandpapadot: ah, well 1.2 is no longer supported anyway |
| 18:52.47 | *** join/#asterisk DirtyD (n=DirtyD@ool-18bddaa0.dyn.optonline.net) |
| 18:52.49 | x86 | grandpapadot: try allow=g729 and then another line under that which says allow=gsm |
| 18:52.56 | DirtyD | I'm about to give up on rxfax. |
| 18:53.00 | x86 | grandpapadot: perhaps it's getting confused with both on one line |
| 18:53.03 | grandpapadot | x86: Thanks. |
| 18:53.12 | x86 | DirtyD: i had much better luck with hylafax/iaxmodem |
| 18:53.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:53.41 | outtolunc | you can also just line them up like... allow=g729,ilbc,gsm,ulaw,alaw,speex |
| 18:53.50 | outtolunc | in order of pref |
| 18:54.23 | grandpapadot | Right, but the ordingal preference isn't working ... |
| 18:54.25 | DirtyD | x86: Hylafax requires the use of a hardware fax modem right? |
| 18:54.45 | grandpapadot | Keeps connecting with gsm, and g729 is my preference. |
| 18:55.06 | outtolunc | is the other side g729 capable? |
| 18:55.20 | outtolunc | and do you have the codec and lics installed? |
| 18:55.25 | grandpapadot | Yea, that's what I've been using, but sometimes I run out of licenses and want to fall back to gsm. |
| 18:55.46 | grandpapadot | But when I do allow=g729,gsm - it connects with gsm every time. |
| 18:56.17 | outtolunc | maybe you should get more lics <G> |
| 18:57.49 | [TK]D-Fender | DirtyD: No, go lookup "IAXModem" |
| 18:57.56 | grandpapadot | Well, there's more to it than this, I'm just dumbing it down to the actual issue, i.e., trying to force an ordinal preference to my codecs on outbound calls. |
| 18:58.33 | x86 | DirtyD: no, that's where iaxmodem comes in |
| 19:00.41 | x86 | anyone ever try oslec? |
| 19:02.12 | [TK]D-Fender | x86: Plenty of people. In general its considered better that Zaptel default options, weaker than any of the paid ones. |
| 19:02.19 | DirtyD | x86: IAX? Even if hylafax is installed on the same machine as asterisk? |
| 19:02.28 | x86 | DirtyD: that's how i run it |
| 19:02.33 | [TK]D-Fender | DirtyD: it uses IAX as a virtual interface. |
| 19:02.45 | x86 | DirtyD: asterisk <--> IAX <--> iaxmodem <--> hylafax |
| 19:02.47 | [TK]D-Fender | DirtyD: Just try it already... |
| 19:02.54 | DirtyD | My calls are coming from PSTN over a PRI, will that be a problem? |
| 19:03.06 | x86 | jesus christ dude RTFM |
| 19:03.20 | x86 | gah, sorry, i'm in a pissy mood |
| 19:03.42 | x86 | [TK]D-Fender: well i've got on-board hardware EC... should i bugger with oslec? |
| 19:03.55 | x86 | [TK]D-Fender: i've still got bad echo even with EC turned on |
| 19:04.05 | x86 | [TK]D-Fender: should I mess with rxgain/txgain or something? |
| 19:04.49 | [TK]D-Fender | x86: Wouldn't work anyways. If you've got HWEC it patches in. EC is either hard or soft on that interface as a whole, not port based |
| 19:04.51 | UnixDog | if you dont mess you wont learn |
| 19:04.55 | [TK]D-Fender | (IIUC) |
| 19:04.58 | JT | i don't think oslec will be any better than hwec |
| 19:05.03 | UnixDog | its all about learning the software |
| 19:05.11 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 19:05.13 | [TK]D-Fender | x86: Just ask yourself whats so buggered up with that port. |
| 19:05.15 | UnixDog | back up your dir and start playing with files |
| 19:05.22 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 19:05.24 | [TK]D-Fender | x86: Swap the termination to see if its the loop |
| 19:05.44 | *** join/#asterisk susinths (n=susinths@sos3-1x-dhcp065.studby.uio.no) |
| 19:07.57 | tripps | apparently level 3 is spitting back my calls to them - my provider tells me i don't have the correct north american dialing plan for level 3. however I've got the basic 10 and 11 digit definitions. any ideas? |
| 19:08.12 | x86 | [TK]D-Fender: well, it only echos when going from that channel to IAX |
| 19:08.27 | JT | lol iax |
| 19:08.27 | x86 | [TK]D-Fender: if i bridge the call onto the PSTN (over T1), there is no echo at all |
| 19:08.45 | [TK]D-Fender | x86: And from that channel to SIP? And try that pair swap I suggested |
| 19:09.28 | x86 | [TK]D-Fender: user --> ZAP --> asterisk --> IAX --> asterisk --> ZAP --> Toshiba PBX |
| 19:10.00 | x86 | [TK]D-Fender: and it seems like now it's multiple channels having the issue, not just 37 as originally explained to me |
| 19:10.15 | x86 | [TK]D-Fender: so I doubt it's the loop |
| 19:10.17 | [TK]D-Fender | x86: Get a complete picture and map it out better. |
| 19:10.19 | JT | try sip |
| 19:10.29 | *** join/#asterisk vlt (n=dm@suez.activ-job.com) |
| 19:10.36 | [TK]D-Fender | x86: And attempt to isolate what side the issue traces back to. |
| 19:10.44 | *** join/#asterisk asdx (n=diego@adsl-136-13.click.com.py) |
| 19:10.49 | [TK]D-Fender | x86: You have 3 points of failure there |
| 19:10.52 | x86 | [TK]D-Fender: I'm getting echo on both sides |
| 19:11.01 | [TK]D-Fender | x86: maybe even 4 |
| 19:11.13 | x86 | [TK]D-Fender: when I talk, not only can I hear my own echo, but the other end can hear my echo as well |
| 19:11.52 | UnixDog | if its a x100p its your own fault |
| 19:11.58 | x86 | [TK]D-Fender: that's from a wired desk phone (analog)... if I use this crappy radioshack 900mhz cordless phone, there's no echo at all (on a different zap channel) |
| 19:12.03 | [TK]D-Fender | x86: Get to it... start at source and work your way back through each step including protocol testing. |
| 19:12.09 | x86 | UnixDog: sangoma a104d-x |
| 19:12.20 | UnixDog | ok and what ver of wanpipe |
| 19:12.34 | x86 | UnixDog: 3.2.1 |
| 19:12.52 | UnixDog | back step to 3.2.0 and see if it goes away |
| 19:12.59 | UnixDog | is another step |
| 19:13.01 | x86 | UnixDog: wanpipemon shows me echo cancel is enabled, as does zap show channel 37 in CLI |
| 19:13.10 | vlt | Hello. I have three asterisks connected to each other. A is public, B and C behind NAT but connected via VPN. Problem: B can't register to A anymore while B can. A to B via VPN is ok, and B to A is fine too. I restarted all servers but the problem remains. Any idea what to check first? |
| 19:13.27 | x86 | well I think I'll try first moving the asterisk <--> asterisk trunk to SIP instead of IAX |
| 19:13.51 | asdx | [TK]D-Fender: can you help me? http://pastebin.com/m14cdcc48 |
| 19:13.59 | x86 | then I'll munge around with the rxgain/txgain settings |
| 19:14.01 | asdx | [TK]D-Fender: i'm trying to register with teliax |
| 19:14.02 | vlt | sorry, I mixed up A, B and above ... I'll try again. |
| 19:14.15 | asdx | [Apr 9 14:14:04] NOTICE[23757]: chan_sip.c:7492 sip_reg_timeout: -- Registration for 'adilwali@den.teliax.net' timed out, trying again (Attempt #137) |
| 19:14.38 | asdx | i have core set verbose 6 |
| 19:15.10 | [TK]D-Fender | asdx: I told you to enable SIP DEBUG and show everything. I'm not asking again. |
| 19:15.28 | asdx | [TK]D-Fender: electricity went away here, i missed that part |
| 19:16.15 | [TK]D-Fender | asdx: Never ask for help on a SIP issue with pastebinning your sip.conf and complete CLI debug at verbose 10 & SIP debug enabled. |
| 19:16.27 | asdx | [TK]D-Fender: ok |
| 19:16.57 | tripps | my telco is telling me i need a CDN NPI north american dial plan and "it's coming across as unknown" ? |
| 19:17.31 | ManxPower | tripps: pridialplan=national |
| 19:17.37 | asdx | SIP Debugging enabled |
| 19:17.52 | asdx | [TK]D-Fender: should i pastebin the output of that? |
| 19:18.08 | vlt | A=public, B and C behind NAT. B->A doesn't work, while C->A is fine. B->C via VPN works too. Connection is IAX2. Any idea where to look? |
| 19:18.28 | ManxPower | vlt: standard port forwarding stuff |
| 19:18.40 | [TK]D-Fender | asdx: of the full SIP communication that causes the error |
| 19:18.43 | plik | any uk people here been or going to "Unified Communications / VoIP for Business '08" at Olympia? |
| 19:18.51 | Jumpie | hmm whats the diff between perc 5/i and sas 5/i again? |
| 19:19.09 | Qwell | Jumpie: s/perc/sas/, apparently :) |
| 19:19.15 | Jumpie | lol |
| 19:19.20 | Jumpie | has to do with raid or something |
| 19:19.37 | tripps | ManxPower, using a SIP gateway so I presume that would be in the settings there somewhere? mediant 1000 in my case |
| 19:19.53 | ManxPower | tripps: You said you are using IAX |
| 19:20.10 | [TK]D-Fender | Jumpie: Yes, the PERC is RAID, the other isn't |
| 19:20.48 | tripps | ManxPower, not in this case. this is * --> mediant 1000 -> PRI |
| 19:20.51 | [TK]D-Fender | ManxPower: I think your aim is a bit off.. |
| 19:21.10 | ManxPower | [TK]D-Fender: I see that. |
| 19:21.16 | ManxPower | tripps: I have no idea what you are talking about. |
| 19:21.17 | vlt | ManxPower: Look like it, yes. Hmmm, worked for over a year now ... strange. |
| 19:21.18 | [TK]D-Fender | ManxPower: My turn next! |
| 19:21.23 | Jumpie | oh... |
| 19:21.24 | Jumpie | my bad |
| 19:21.26 | Jumpie | ty |
| 19:21.57 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 19:23.25 | asdx | [TK]D-Fender: http://pastebin.com/m29a0139a |
| 19:24.35 | [TK]D-Fender | asdx: And where are you configs in there? |
| 19:25.03 | x86 | ugh... been so long i forgot how to dial asterisk to asterisk over SIP |
| 19:25.35 | x86 | JT: if I have Dial(IAX2/user:pass@host/exten), how do i convert that to a SIP dial string? |
| 19:25.53 | Jumpie | ugh i was hoping to have a 1ru server, with a 1 T1 card and 4 fxo all for under 2 grand |
| 19:26.07 | x86 | Jumpie: rofl! |
| 19:26.21 | Jumpie | :) |
| 19:26.25 | Jumpie | maybe i can for 3 |
| 19:26.29 | x86 | Jumpie: maybe a used celeron with an x100p |
| 19:26.33 | Jumpie | which is initially what i had planned |
| 19:26.39 | asdx | [TK]D-Fender: http://pastebin.com/m6f8049c9 |
| 19:26.45 | Jumpie | the 4fxo and 1 t1 should be under 600 combined |
| 19:26.52 | Jumpie | lookin on digium's site |
| 19:26.59 | Jumpie | i just was hoping for a one stop shop thats all |
| 19:27.01 | x86 | yeah if you don't want EC sure |
| 19:27.08 | Jumpie | arhg |
| 19:27.09 | Jumpie | :P |
| 19:27.17 | x86 | non-EC T1 cards are cheap |
| 19:27.17 | Jumpie | you know i had no echo on my test setup, but it was only one call |
| 19:27.21 | Jumpie | yea..shit man |
| 19:27.25 | Jumpie | ok so a another grand |
| 19:27.36 | x86 | cheap,good,reliable.... pick two ;) |
| 19:27.37 | Jumpie | do you know of any one vender places that have a server w/ cards? |
| 19:27.45 | Jumpie | lol yeah |
| 19:27.46 | x86 | voipsupply.com |
| 19:28.03 | Jumpie | on pbxselect they had some supermicro ones but no digium cards, they also had 'trix box' servers |
| 19:28.05 | Jumpie | but i dont want that |
| 19:28.08 | Jumpie | k ill look there too |
| 19:29.21 | asdx | [TK]D-Fender: i tried with a free voipjet account before and with a simply register => it got 'registered' |
| 19:29.21 | asdx | [TK]D-Fender: but with teliax i'm getting connection timed out |
| 19:29.27 | [TK]D-Fender | asdx: Make sure your extenal IP hasn't changed... |
| 19:29.49 | [TK]D-Fender | asdx: and I don't see a peer for placing/receiving calls in there. |
| 19:30.01 | asdx | yeah i haven't put that yet |
| 19:30.14 | asdx | i wanted to see if it registered first |
| 19:30.20 | Jumpie | gives fender a grandstream pbx appliance for a gift |
| 19:30.50 | asdx | [TK]D-Fender: external ip, you mean the ip address from the server (VPS) where my asterisk is? |
| 19:30.59 | [TK]D-Fender | asdx: Yes |
| 19:31.17 | [TK]D-Fender | asdx: Actually.. its public & fixed, isn't it? |
| 19:31.23 | *** part/#asterisk gmaruz1 (n=maruzz@213-140-6-122.ip.fastwebnet.it) |
| 19:31.45 | asdx | [TK]D-Fender: yes |
| 19:32.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:32.09 | x86 | how do i dial a SIP extension with user:pass@host? |
| 19:32.10 | [TK]D-Fender | asdx: Ok, check your firewall settings. |
| 19:32.51 | vlt | ManxPower: Yes, you're right. It's the NAT router. I can see the iax packets go out and answers come back on ppp0 but they aren't forwarded to eth0 where Asterisk waits for them. Thank you. |
| 19:33.18 | asdx | shit |
| 19:33.29 | asdx | i see lot of 'DROP' rules on the server |
| 19:33.50 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-89-235.vif.net) |
| 19:34.00 | [TK]D-Fender | asdx: Also double check that you are using the correct host for registration. |
| 19:34.24 | flush | yo |
| 19:34.24 | asdx | ok |
| 19:34.37 | flush | asterisk is cool |
| 19:34.48 | flush | my box is the O.G |
| 19:35.09 | asdx | [TK]D-Fender: should i do a flush of their iptables rules there? |
| 19:35.11 | asdx | clean up |
| 19:35.24 | Jumpie | fender |
| 19:35.29 | Jumpie | what doyou feel about phone bochs appliances |
| 19:35.55 | asdx | wow this stuff continues |
| 19:36.08 | Jumpie | there is one here 80gb hd, 1.6ghz core duo mobile proc , sangoma 1 T1 + echo cancelation for $2400 |
| 19:36.14 | Jumpie | and 1gb ram |
| 19:37.06 | x86 | gah come on... someone MUST know how to dial a SIP peer with user:pass in the dial string |
| 19:37.28 | x86 | Jumpie: that's only the T1 card too, no POTS card |
| 19:37.36 | ManxPower | x86: Never done that, always used sip.conf entry. |
| 19:37.37 | Jumpie | yeah, i can add a pots card too |
| 19:37.45 | Jumpie | that i can get cheap |
| 19:37.48 | Jumpie | er |
| 19:38.28 | [TK]D-Fender | Jumpie: link? |
| 19:38.51 | asdx | [TK]D-Fender: http://pastebin.com/m1a158d42 <-- they have a lot of iptables rules on their server, i think that's what causing the time out problem |
| 19:39.05 | asdx | i'm going to try somewhere else |
| 19:39.11 | Jumpie | www.voipsupply.com/product_info.php?products_id=3804 |
| 19:39.27 | Jumpie | im wary of only 80gb though, and 1.6ghz proc |
| 19:40.09 | Jumpie | ah, the one product 3808 |
| 19:40.19 | Jumpie | is better, another $100 and 160gb and 2gb |
| 19:40.21 | ManxPower | If I pre built a tellabs echo cancel card + cage + power supply + setup, does anyone think someone would buy it? |
| 19:40.32 | Jumpie | im sure |
| 19:40.38 | Jumpie | thast the kinda things im trying to buy for test equip |
| 19:40.41 | Jumpie | what would you charge? |
| 19:41.06 | ManxPower | Jumpie: That would depend on what I paid for the equipment in the first place. |
| 19:41.43 | Jumpie | oh are you talking, building new stuff, oh i thought you meant like gathering some old parts and selling somethin used on ebay |
| 19:41.52 | Jumpie | still, people would do it |
| 19:42.25 | jblack | Has anyone seen problems with callerid with callwithus lately? |
| 19:42.27 | ManxPower | No, I was thinking GETTING the stuff on eBay (new none of this stuff is affordable), putting it togather, testing, config to customer need |
| 19:42.31 | x86 | ok, still have echo with SIP |
| 19:42.42 | x86 | this officially SUCKS ASS |
| 19:42.49 | [TK]D-Fender | Jumpie: Its a while-box system with sotware loaded. No better than you can do yourself, and less easily waranteed than a Dell box. |
| 19:43.06 | Jumpie | hmm ok |
| 19:43.12 | Jumpie | yeah i was on dell's site doing comparisons too |
| 19:43.27 | Jumpie | a 2970 power edge, i got configed up to 1796$ |
| 19:43.35 | asdx | [TK]D-Fender: i cleaned the firewall settings and it registered now |
| 19:43.37 | asdx | [TK]D-Fender: thx :) |
| 19:43.56 | Jumpie | raid 1 80gb, dual core opteron 2212 |
| 19:44.04 | Jumpie | 2gb ram |
| 19:44.12 | Jumpie | so i guess i could ya, get the sangoma cards myself |
| 19:45.16 | x86 | gah fuck it... I'll just do PSTN between the two branch offices since both IAX2 and SIP have echo so bad |
| 19:45.31 | ManxPower | Jumpie: The tellabs stuff is what Sprint and AT&T use for EC, they are hell to setup and wire. |
| 19:46.48 | Jumpie | heh |
| 19:46.54 | Jumpie | vz uses tellabs for their fios OLTS |
| 19:47.03 | Jumpie | honestly our more cmmon platform |
| 19:48.06 | Jumpie | ManxPower do you think i should steer from rhino? |
| 19:48.09 | Jumpie | and go digium/sangoma? |
| 19:48.24 | vlt | /proc/sys/net/ipv4/netfilter/ip_conntrack_udp_timeout* was set to 0 ... now it work again. Thanks |
| 19:48.56 | ManxPower | Jumpie: If I was spending my own money I'd buy a Sangoma, if I was using someone else's money I'd buy either a (recent model) Digium card or a Sangoma card. |
| 19:49.24 | Jumpie | k |
| 19:49.26 | Jumpie | man well shit |
| 19:49.36 | Jumpie | the oly pre built stuff with t1 cards is like rhino |
| 19:49.38 | ManxPower | vlt: someone over there was thinking "DNS doesn't need more than a 10 second timeout!" |
| 19:50.02 | Jumpie | other than that phone bochs whic had a sangoma |
| 19:50.34 | [TK]D-Fender | Jumpie: What are you whining about now? Just by a stupid server ans slap a card in. *sheesh* |
| 19:50.42 | glaz | Anywhere I can find a good how to configure a Digium PRI/T1 digital card? |
| 19:50.44 | Jumpie | fender, i know that man |
| 19:50.50 | ManxPower | Jumpie: Many of the "clone cards" use the original Zapata Open Source design. That design had some problems and Digium moved away from that design two generations ago. I don't think Rhino uses the Zapata design, but I could be wrong. |
| 19:50.56 | Jumpie | im trying to stick with the same stuff and just debating on WHAT server to buy, or if i go a prbuilt |
| 19:51.08 | Jumpie | ah hmm |
| 19:51.18 | Jumpie | fender im probably leaning toward this dell power edge anyway |
| 19:52.06 | ManxPower | Jumpie: I'd buy the server prebuilt -- without telephony cards, etc. Just a plain server with a good warranty nd exchange policy |
| 19:52.12 | Jumpie | ok |
| 19:52.21 | Jumpie | well i have worked with dell servers before and been satisfied |
| 19:52.22 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:52.23 | Jumpie | so ya ill go for that |
| 19:52.27 | ManxPower | If it is not compat with the card you get, exchange it for a different model |
| 19:52.43 | Jumpie | actually it was listed as one of the known and tested good ones on digium's site |
| 19:52.46 | Jumpie | :) |
| 19:53.23 | ManxPower | And no matter what brand of card you get, there's a chance it won't work with your hardware -- Sangoma has historically been more aggressive about fixing such issues. |
| 19:53.34 | Jumpie | ok |
| 19:53.43 | Jumpie | hence, sangoma is tyhpically a bit more pricey too isnt it? |
| 19:53.49 | Jumpie | but in this arena, you definately get wha tyou pay for |
| 19:53.56 | ManxPower | Jumpie: not by any significant percent |
| 19:54.41 | ManxPower | Jumpie: Sangoma is a zillion times better than the original Zapata design. |
| 19:54.46 | Jumpie | it'll be nice when i have the capital to have like, a boxful of these cards |
| 19:54.52 | Jumpie | oh, i beleive it |
| 19:56.46 | vlt | What can cause echo in a SIP/IAX only connection and how to avoid it? [SIP phone]---SIP---[*]---IAX2---[*]---SIP---[SIP phone] |
| 19:56.48 | Jumpie | manx, what would you prefer for asterisk |
| 19:56.49 | Jumpie | amd or intel? |
| 19:57.20 | Jumpie | the only closer runner next to the dells are these supermicro 1ru servres, they use intel woodcrest 5110 and the dells are amd dual core opteron |
| 19:58.27 | *** join/#asterisk gmaruz1 (n=maruzz@213-140-6-122.ip.fastwebnet.it) |
| 20:02.05 | susinths | how much RAM does * need for a couple of concurrent calls? |
| 20:02.32 | mihinomenest | how much do you have? |
| 20:02.40 | susinths | 768MB |
| 20:02.52 | mihinomenest | that'll be fine. |
| 20:03.08 | susinths | in an old Compaq server dual p3 733Mhz |
| 20:03.22 | susinths | really |
| 20:03.35 | mihinomenest | I ran an asterisk server on a Via C3 1.2GHz with 1GB of ram and it handled 10 simultaneous calls no problem. |
| 20:03.51 | mihinomenest | this was with g.711u. |
| 20:03.53 | susinths | impressive :) |
| 20:03.57 | mihinomenest | sip in, sip out. |
| 20:04.20 | susinths | ok |
| 20:04.33 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 20:04.33 | susinths | jeg just wanna setup my server to test * |
| 20:04.48 | susinths | i * |
| 20:05.11 | mihinomenest | I'm actually running asterisk on a freebsd 6.2 system in my basement. it's got two p3 833s and 1gb of ram. |
| 20:05.22 | mihinomenest | the only problem is my craptacular DSL. |
| 20:05.55 | susinths | oh |
| 20:05.57 | susinths | i c |
| 20:06.44 | susinths | actually, i got also 1024MB of ram , 4*256MB, but the server says 768MB |
| 20:06.59 | susinths | i gave up trying to find out |
| 20:07.28 | mihinomenest | huh, how bizzare. |
| 20:07.29 | susinths | but im gonna put the server at my friend ADSL2, with 20Gb/1Gb |
| 20:07.49 | susinths | yeah, it's so annoying, i really need the rest of 256 |
| 20:07.49 | mihinomenest | do you me mbps? |
| 20:07.55 | mihinomenest | s/me/mean |
| 20:07.57 | denon | heh, he means quota |
| 20:08.05 | susinths | what? |
| 20:08.05 | mihinomenest | ouch. |
| 20:08.19 | susinths | sorry? |
| 20:08.23 | mihinomenest | 20Gbps/1Gbps isn't possible for a DSL line rate. |
| 20:08.37 | susinths | it think its 24/1 |
| 20:08.37 | mihinomenest | 20Mbps/1Mbps is a lot more reasonable. |
| 20:08.46 | susinths | sorry |
| 20:08.55 | mihinomenest | no worries. |
| 20:08.56 | susinths | yeah , Mbps |
| 20:08.57 | susinths | hehe |
| 20:09.05 | mihinomenest | I happened to have worked for an ISP in a previous life. |
| 20:09.16 | susinths | i c, hehe |
| 20:09.17 | *** join/#asterisk aleff (n=alef@c9063318.virtua.com.br) |
| 20:09.25 | aleff | hi |
| 20:09.35 | susinths | do u got a lot of knowledge with * |
| 20:09.46 | mihinomenest | not as much as others in the channel. |
| 20:09.51 | susinths | ok |
| 20:09.53 | mihinomenest | I can hum a few bars. |
| 20:09.53 | *** join/#asterisk bcnl (n=nobody@24.86.64.144) |
| 20:10.00 | susinths | ok |
| 20:10.06 | susinths | like AGI |
| 20:10.09 | aleff | how do I select the events I want to receive on the manager interface? I know that I must select it on the login, but I dont know how |
| 20:10.09 | susinths | programming |
| 20:10.18 | aleff | I tried Event: name, but still get all the others |
| 20:10.20 | mihinomenest | not so much. |
| 20:10.23 | susinths | oki |
| 20:10.30 | mihinomenest | I can setup the sip.conf and the dial-plan for it. |
| 20:10.41 | mihinomenest | I had a unix guru that took it from there. |
| 20:10.47 | susinths | im going out for an hour, will be back |
| 20:10.51 | susinths | talk later |
| 20:11.01 | mihinomenest | was the ISPs windows admin. |
| 20:11.35 | susinths | ok |
| 20:11.41 | susinths | bye 4 now |
| 20:18.20 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 20:19.18 | Greek-Boy | what does one do in a situation where a particular user has his voicemail box on the main enterprise asterisk box but happens to be at a branch office on the branch office asterisk box? |
| 20:19.48 | vlt | "NOTICE[2077]: chan_iax2.c:5172 register_verify: No registration for peer 'hylafax' (from 192.168.1.1)" <-- I got this every minute from an iaxmodem client. In my iax.conf it appears as "type=user host=dynamic". What causes this notice? |
| 20:21.47 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 20:21.49 | *** part/#asterisk Porks (n=Porks@201.62.79.12) |
| 20:21.53 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 20:22.15 | bsdwarrior | my default what does asterisk use to play the music on hold? its 1.2.14 |
| 20:23.08 | *** join/#asterisk ta^3 (n=tacvbo@189.146.192.67) |
| 20:24.33 | mihinomenest | bsdwarrior: probably mpg123 |
| 20:25.10 | bsdwarrior | my config is very strange, I'm assuming is just doing directory based. |
| 20:25.19 | bsdwarrior | mode=files is commented out in musiconhold.conf |
| 20:25.42 | mihinomenest | what isn't commented out? |
| 20:27.18 | *** join/#asterisk mandd (n=dache@dsl-135-236.aei.ca) |
| 20:28.09 | mandd | is there a way to force my own called id? |
| 20:28.31 | mandd | when some one dials out I mean |
| 20:30.49 | mihinomenest | you mean "caller id"? |
| 20:31.57 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-122-158.PHNX.QWEST.NET) |
| 20:32.12 | bsdwarrior | mihinomentest ill pastebin |
| 20:33.06 | bsdwarrior | http://pastebin.com/dcff6597 |
| 20:33.54 | *** join/#asterisk niZon (n=niZon@tande.voinetworks.net) |
| 20:33.56 | mandd | mihinomenest yes |
| 20:33.57 | mandd | sorry. |
| 20:34.35 | asdx | [Apr 9 15:33:57] WARNING[25808]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 20:34.42 | asdx | i'm getting that now |
| 20:34.52 | asdx | when trying to make a call |
| 20:35.19 | mihinomenest | mandd: I know you can set that in sip.conf, I suppose it's possible for zap/iax as well. |
| 20:35.37 | mandd | how do I do it in SIP |
| 20:35.49 | mandd | i am in sip wiki right now, what should I look for |
| 20:35.53 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 20:35.57 | mihinomenest | callerid=<string> |
| 20:36.17 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 20:37.27 | exvito | hi all, can anyone point to a european supplier of single port HFC based ISDN cards which are 3.3v PCI compatible ? (got a batch of them but they're 5v only!) |
| 20:37.32 | Jumpie | gar wtf |
| 20:37.40 | Jumpie | trying to untar these files right from digium.org |
| 20:37.43 | Jumpie | and ti says its not in gzip format |
| 20:38.55 | exvito | ...i've been google'ing very hard to no avail. *sigh!* |
| 20:39.48 | mihinomenest | why can't you use 5V cards? |
| 20:41.23 | *** join/#asterisk paci`` (n=paci@cpe-075-182-072-065.nc.res.rr.com) |
| 20:41.27 | paci`` | <PROTECTED> |
| 20:41.28 | paci`` | <PROTECTED> |
| 20:41.28 | paci`` | <PROTECTED> |
| 20:41.28 | paci`` | <PROTECTED> |
| 20:41.43 | exvito | mihinomenest: no such slots on the systems... |
| 20:41.54 | Qwell | linux emulation from ports? |
| 20:41.56 | NirS | Hi all |
| 20:42.01 | NirS | I need a chan_local expert |
| 20:42.05 | NirS | any available ? |
| 20:42.07 | paci`` | Qwell, yeah |
| 20:42.10 | mihinomenest | paci``: why doesn't it work on bsd? |
| 20:42.34 | paci`` | mihinomenest, its a pain in the ass to get app conference working for some reason |
| 20:43.02 | mihinomenest | everything else works? |
| 20:43.11 | paci`` | yea |
| 20:43.14 | paci`` | but we want a conference |
| 20:43.16 | paci`` | XD |
| 20:43.20 | mihinomenest | do you have a ztdummy driver installed? |
| 20:43.23 | *** join/#asterisk Warp4 (n=robert@207.251.204.194) |
| 20:43.26 | Warp4 | hi all |
| 20:43.38 | Warp4 | well got asterisk up and running ... sorta |
| 20:43.44 | Warp4 | just not able to dial out |
| 20:44.00 | Warp4 | (using Trixbox) |
| 20:44.03 | paci`` | mihinomenest, |
| 20:44.07 | Qwell | ~trixbox |
| 20:44.08 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 20:44.08 | paci`` | no, I couldn't get that working |
| 20:44.19 | paci`` | mihinomenest, and thts not needed for app_Conference, only meetme |
| 20:45.05 | mihinomenest | all the documentation I've read says that it is needed, but I've never tried to get it to work. |
| 20:45.38 | asdx | can someone take a look at my config and tell me what is wrong? http://pastebin.com/m75e85fdc |
| 20:45.38 | paci`` | yeah |
| 20:45.44 | asdx | i'm trying to make calls over SIP |
| 20:45.45 | asdx | with teliax |
| 20:45.54 | asdx | i used the same config for IAX2 before |
| 20:45.56 | asdx | and worked fine |
| 20:46.01 | Jumpie | lol |
| 20:46.09 | Jumpie | i cant even install :( |
| 20:46.11 | *** join/#asterisk SomethingISODD (n=wno@S010600a0d1757bfb.cg.shawcable.net) |
| 20:46.12 | Jumpie | says not a gzip file |
| 20:46.31 | SomethingISODD | hello question i have a pri 31 channels. on out bound calling i keep getting channel is inuse |
| 20:46.34 | mihinomenest | is it bz2? |
| 20:47.23 | asdx | anyone? |
| 20:47.28 | NirS | [TrixBox] is a blasfemy to the Asterisk community, the fact that a would be called "Open Source" product utilizes trojan and hidden code methodologies is an insult |
| 20:47.47 | [hC] | SomethingISODD: hey |
| 20:47.54 | SomethingISODD | hello |
| 20:47.56 | [hC] | Qwell: you may want to set this as a topic, its pretty big news |
| 20:48.01 | Qwell | ? |
| 20:48.33 | glaz | where does asterisk-gui store the admin user and password? |
| 20:49.01 | Qwell | [hC]: what news now? |
| 20:49.24 | [hC] | There is a western canada wide outage on Allstream (AT&T Canada's) network. There was a fiber cut in toronto that severed all of western canada off from the PSTN |
| 20:49.30 | Qwell | nice |
| 20:49.36 | [hC] | Calling out will result in channel unavailable |
| 20:49.40 | Qwell | Juggie: WHAT DID YOU DO? |
| 20:49.40 | [hC] | Calling some cell phone will not work |
| 20:49.41 | Jumpie | no its .gz |
| 20:49.45 | [hC] | Calling IN will result in fast busy |
| 20:49.48 | Jumpie | sorry tryin to get irc on my asterisk box |
| 20:49.53 | Jumpie | having some tropble hehe |
| 20:49.54 | [hC] | The outage has been going on now for almost an hour. |
| 20:50.00 | [hC] | And of course, there is no ETA :) |
| 20:50.02 | Qwell | [hC]: IT360 is going on in TO right now, and Juggie is there. :p |
| 20:50.11 | Qwell | Totally his fault |
| 20:50.24 | [hC] | Qwell: Oooooo. So is kristian kielhofner. |
| 20:50.25 | [hC] | I smell trouble |
| 20:50.36 | Qwell | notifies support :p |
| 20:51.00 | SomethingISODD | Does anyone know about outbound calls via pri? |
| 20:51.09 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 20:51.15 | mandd | my sip.cof, http://pastebin.com/m6bc21e20, cant figure out why I cant call a local extension |
| 20:51.19 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 20:51.20 | mandd | all phones are on and connected. |
| 20:51.55 | Qwell | [hC]: yeah, no kidding |
| 20:52.19 | [hC] | SomethingISODD: uh, did you not just see what i said? |
| 20:52.29 | SomethingISODD | ya but that has nothing to do with me :-) |
| 20:52.39 | [hC] | SomethingISODD: oh really? how so? |
| 20:52.41 | SomethingISODD | i am working on a clients server overseas |
| 20:52.56 | [hC] | SomethingISODD: Ah. I saw calgary in your hostname. |
| 20:53.14 | SomethingISODD | :-p |
| 20:53.42 | [hC] | SomethingISODD: check your PRI alarms or settings in zaptel.conf/zapata.conf ? |
| 20:53.46 | asdx | can someone take a look at my config and tell me whats wrong? i'm trying to make outbound calls with Teliax/SIP http://pastebin.com/m75e85fdc |
| 20:54.04 | asdx | i'm getting "service unavailable" |
| 20:54.06 | SomethingISODD | ok thanks |
| 20:54.12 | Qwell | asdx: see above |
| 20:54.16 | Qwell | okay, topicing |
| 20:54.45 | *** part/#asterisk BingoPajama (n=chatzill@38.98.105.130) |
| 20:55.15 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.4.19 (2008/04/02), 1.6.0-beta7.1 (2008/03/29), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -= Widespread PSTN outages in Canada - beware. |
| 20:55.22 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.4.19 (2008/04/02), 1.6.0-beta7.1 (2008/03/29), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Widespread PSTN outages in Canada - beware. |
| 20:55.46 | asdx | Qwell: ? |
| 20:56.10 | Qwell | oh, you said teliax. dunno where they are |
| 20:56.15 | Qwell | read that as Telus :p |
| 20:56.21 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 20:56.27 | *** join/#asterisk blaylock (n=blaylock@snap.helixsystems.com) |
| 20:56.34 | file | smacks Qwell |
| 20:56.39 | Qwell | what? :( |
| 20:56.44 | Qwell | honest mistake! |
| 20:56.47 | asdx | teliax is in the us |
| 20:56.56 | blaylock | does externnotify string have to be a c,c++ file or can it be a bash script? |
| 20:57.05 | Qwell | blaylock: anything that can take args |
| 20:57.23 | blaylock | Qwell, i cant get mine to run the script |
| 20:57.28 | blaylock | dont see any errors either |
| 20:57.29 | Qwell | is it +x? |
| 20:57.29 | niZon | anyone know how to calculate the max number of channels in use at any time using the CDR? |
| 20:57.33 | blaylock | yeah |
| 20:57.44 | Qwell | using full path? |
| 20:57.51 | blaylock | yeah |
| 20:58.02 | blaylock | /etc/asterisk/nameofscript |
| 20:58.03 | mvanbaak | niZon: with some programming logic it should not be too hard |
| 20:58.14 | mvanbaak | blaylock: is /etc mounted noexec ? |
| 20:58.16 | blaylock | i put the script in the asterisk directory |
| 20:58.20 | mvanbaak | (which it should be) |
| 20:58.40 | blaylock | well i can exec the script as root from there |
| 20:58.46 | *** join/#asterisk asdx (n=diego@adsl-136-13.click.com.py) |
| 20:58.50 | blaylock | so i dont see why asterisk wouldnt |
| 20:59.16 | *** topic/#asterisk by file -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.4.19 (2008/04/02), 1.6.0-beta7.1 (2008/03/29), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 20:59.48 | Qwell | eep? |
| 21:00.34 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:00.44 | comprookie2000 | mandd, remove the , from your pastbin link |
| 21:02.22 | Jumpie | hey guys |
| 21:02.25 | comprookie2000 | mandd, looks like your extensions.conf |
| 21:02.27 | Jumpie | what is a recommended irc client for centos |
| 21:02.28 | Jumpie | lol |
| 21:02.39 | jblack | irc2 |
| 21:02.39 | Jumpie | im having some asterisk install problems and dont want to go back and forth to my lappy |
| 21:02.39 | [hC] | file: why remove that? |
| 21:03.11 | blaylock | Qwell, nevermind I had the script write a file and didnt specify a location. when asterisk ran it, the file showed up in /tmp |
| 21:03.37 | blaylock | Qwell, thanks though |
| 21:03.41 | eric2 | 723 codec... from where do I get? |
| 21:04.08 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 21:08.14 | colinm_ | eric2: After you stop at the bank and withdraw a good US$50,000 or so, you call the DSP Group. |
| 21:08.46 | eric2 | serious? it's that pricey? |
| 21:09.02 | Qwell | eric2: you can get a transcoder card from Digium that supports g723 |
| 21:09.09 | eric2 | and here I am thinking it's cheap as 729 |
| 21:09.42 | eric2 | so g723 is out of the question then... :( |
| 21:11.28 | colinm_ | If a PCI card solution works, the transcoder card Qwell's talking about is "only" $1500. |
| 21:11.37 | riddlebox | does anyone know what would cause hints to be slow? it takes like 4-5 seconds before I see it come across in the cli |
| 21:11.46 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 21:11.48 | asdx | hi |
| 21:11.55 | asdx | i can make calls with IAX |
| 21:11.57 | asdx | but not with SIP |
| 21:12.20 | asdx | can someone help? |
| 21:13.09 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 21:13.21 | Daviey | gah.. why can't pickupgroup / callgroup e a string!? |
| 21:13.23 | bitzero | grins |
| 21:13.31 | bitzero | I just got word that I get to do a call center deployment. |
| 21:13.38 | bitzero | My first one. |
| 21:13.44 | Daviey | unlucky |
| 21:14.10 | Daviey | you do realise what a headache large deployments are? :) |
| 21:14.31 | bitzero | My largest deployment thus far has been 30 seats :) |
| 21:14.38 | bitzero | so I'm actually looking forward to the challenge. |
| 21:14.38 | Daviey | and this one will be? |
| 21:14.43 | *** join/#asterisk susinths (n=susinths@sos3-1x-dhcp065.studby.uio.no) |
| 21:14.48 | bitzero | 53 |
| 21:14.51 | bitzero | not huge. |
| 21:15.02 | Daviey | aww, that should be fine.. well done |
| 21:15.13 | Daviey | How are you provisioning? |
| 21:15.17 | bitzero | I'll be doing all softphones + sugarCRM integration |
| 21:15.24 | *** join/#asterisk asdx (n=diego@adsl-136-13.click.com.py) |
| 21:15.27 | bitzero | also a new setup for me. |
| 21:15.46 | asdx | why would IAX2 work just fine for making calls and not SIP? |
| 21:15.47 | bitzero | has done all granstream phones with the occasional softphone thus far. |
| 21:15.48 | asdx | my configuration? |
| 21:15.55 | bitzero | is gay for grandstream GXP phones. |
| 21:15.58 | susinths | is it possible to use * + voipbuster |
| 21:16.15 | Jumpie | hmm |
| 21:16.20 | jblack | what's up with companies these days? |
| 21:16.42 | jblack | Used to be, that you could bribe someone with a large pizza, or slipping a 50 into the shipping box. |
| 21:16.51 | *** join/#asterisk Jumpie (n=jumpie@pool-96-231-155-171.washdc.fios.verizon.net) |
| 21:16.53 | Jumpie | ok |
| 21:17.23 | susinths | is it also possible to make 2 simultaneous calls with 1 voip account? |
| 21:17.32 | jblack | susinths: Typically, yes. |
| 21:17.36 | Qwell | susinths: if your provider lets you |
| 21:17.50 | Kyoshi | anyone have any success with asterisk realtime using ms sql server? |
| 21:17.57 | asdx | can someone help me???????? |
| 21:18.10 | susinths | ok... |
| 21:18.25 | keith4 | asdx: what happens when you try to place a sip call? |
| 21:18.35 | susinths | do u know if voipbuster/ voipstunt lets one do it? |
| 21:18.36 | asdx | i get "service not available" on my softphone |
| 21:18.55 | asdx | keith4: i can make IAX2 calls just fine..... |
| 21:18.56 | jblack | Last year, I slipped a 50 in with a laptop that I needed sony to service. Laptop came back in 3 days, fixed, with 'Customer indicated serious rush' and mailed me a check for the $50. |
| 21:19.26 | keith4 | asdx: this is just a stab in the dark, but have you checked the logs? |
| 21:19.49 | jblack | susinths: I'm not familiar with that provider. Just google for "voipbuster channels" |
| 21:19.58 | Jumpie | hey guys, i get a crazy error when i try to tar, please look at http://jumpie.pastebin.org/28644 |
| 21:20.07 | susinths | ok, i will, thaks |
| 21:20.10 | susinths | thanks* |
| 21:20.19 | jblack | Most providers allow for at least 2 channels, even the "unlimited" ones. Pay as you go ones are typically unlimited channels. |
| 21:20.47 | keith4 | Jumpie: probably corrupt file? |
| 21:20.58 | keith4 | what does "file zaptel-1.4-current.tar.gz" show? |
| 21:21.28 | UnixDog | no its just you |
| 21:21.33 | UnixDog | wrong window |
| 21:21.51 | Jumpie | grr. it was right from digium.org |
| 21:21.53 | Jumpie | lemme check |
| 21:22.05 | Kyoshi | http://jumpie.pastebin.org/28645 |
| 21:22.09 | jblack | is curious about the what unixdog got asked |
| 21:22.22 | asdx | keith4: can you take a look at my config? http://pastebin.com/m73456ba9 |
| 21:22.42 | asdx | keith4: that should work for SIP right? |
| 21:22.46 | jblack | I'd say it's probably a corrupted tarball. |
| 21:22.49 | asdx | keith4: it works for IAX2 |
| 21:22.54 | jblack | Oh.. silly me. Take off the - |
| 21:23.02 | keith4 | shouldn't matter |
| 21:23.05 | jblack | as in "tar xvfz zaptel-1.4-current.tgz" |
| 21:23.07 | Jumpie | [root@ippbx asterisk]# file zaptel-1.4-current.tar.gz |
| 21:23.07 | Jumpie | zaptel-1.4-current.tar.gz: ASCII text, with no line terminators |
| 21:23.08 | keith4 | asdx: check. the. log. |
| 21:23.14 | Jumpie | wtf |
| 21:23.17 | Qwell | you downloaded the html |
| 21:23.31 | Jumpie | but how if the file says its there in directory listing |
| 21:23.40 | jblack | Also, you don't want the dash. (That says read the tarball from stdin) |
| 21:24.06 | jblack | jumpie: You downloaded the page listing the tarball, not the tarball itself, is what I think qwell is trying to say. |
| 21:24.11 | keith4 | Jumpie: file should say something like " gzip compressed data, from Unix" |
| 21:24.28 | Jumpie | my bad..hmm im going off the voip info install guide |
| 21:24.51 | Jumpie | says do wget http://downloads.digium.com/pub/zaptel/zaptel-1.4-current.tar.g |
| 21:24.52 | Jumpie | er gz |
| 21:25.48 | Jumpie | if i made a mistake on a url, then it downloaded the html, i remember, but as far as i can tell the actual .gz file was pulled |
| 21:25.55 | Jumpie | do i have to change some thing first |
| 21:26.14 | Jumpie | i think the location change sincedf this faq? |
| 21:26.28 | keith4 | lol |
| 21:26.34 | keith4 | http://downloads.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz |
| 21:26.38 | keith4 | "link releases/zaptel-1.4.10.tar.gz" |
| 21:26.45 | keith4 | Jumpie: how big is the file you downloaded? |
| 21:26.56 | Qwell | 34 bytes |
| 21:26.57 | keith4 | try this one: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/zaptel/zaptel-1.4.10.tar.gz |
| 21:26.58 | Qwell | fixing :p |
| 21:29.02 | [hC] | that screwed me up too |
| 21:29.09 | [hC] | changing it so that the download links on the site are referrer links |
| 21:29.13 | [hC] | so wget downloads the html |
| 21:29.55 | Jumpie | lol yeap qwell |
| 21:30.03 | Jumpie | thanks well |
| 21:30.07 | Qwell | fixed, should hit the mirrors in a sec |
| 21:30.20 | Jumpie | qwell ya the other ones look ok, all much bigger |
| 21:30.25 | Jumpie | i should have suspected a 34byte file :D |
| 21:30.44 | keith4 | maybe it was a really, really effificient zip algorithm |
| 21:31.01 | Jumpie | haha |
| 21:31.03 | Jumpie | 1000000000000000000:! |
| 21:31.06 | jblack | bzip42 |
| 21:34.29 | Jumpie | fixed qwell |
| 21:34.39 | Jumpie | 1.6m right |
| 21:35.40 | keith4 | yah |
| 21:41.13 | Greek-Boy | damn |
| 21:41.21 | Greek-Boy | struggling to get voicemail to work with imap |
| 21:42.23 | *** join/#asterisk mcfloppy (n=info@88-134-142-77-dynip.superkabel.de) |
| 21:42.24 | mcfloppy | hello |
| 21:43.43 | Katty | don't copy that floppy |
| 21:43.45 | mcfloppy | i have a usb hfc card. with hisax, i see incoming calls in var/log/messages. but how can i use capi with the card? when i load capidrivers, capinfo says: capi is not installed |
| 21:44.16 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 21:45.31 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 21:45.39 | mcfloppy | buerots:/usr/src/linux# capiinfo |
| 21:45.39 | mcfloppy | capi not installed - No such device or address (6) |
| 21:45.48 | asdx | is there a way i could put in the dialplan |
| 21:45.57 | asdx | _X instead of _XXXXXXXXXXXXXXX |
| 21:46.11 | asdx | so if i type a lot of digits, it will match as well |
| 21:48.28 | *** join/#asterisk autojack (n=owen@pdpc/supporter/active/autojack) |
| 21:49.01 | autojack | I'm looking at the following example line from a dialplan: exten => #,3,GotoIf($[${DIALSTATUS} = CONGESTION]?4:5) can anyone explain what the conditions for the 4:5 part are? |
| 21:49.18 | mcfloppy | or can i use the hisax driver with asterisk? |
| 21:50.28 | esk | asdx: you might try _X. |
| 21:50.58 | outtolunc | it means if your dialstatus is equal to congestion, then goto #,4 else #,5 |
| 21:51.22 | Greek-Boy | anyone here doing voicemail imap storage using courier-imap server? |
| 21:51.37 | outtolunc | meaning, do the next dialplan step, or skip over it to the next one (in this case) |
| 21:52.34 | autojack | outtolunc: hmmm. that seems redundant given the next steps: |
| 21:52.50 | autojack | exten => #,4,Playback(vm-nobodyavail) |
| 21:52.50 | autojack | exten => #,5,Goto(s,1) |
| 21:52.51 | outtolunc | which means jack to me <G> |
| 21:52.57 | autojack | oh, well maybe not. |
| 21:53.12 | outtolunc | its correct |
| 21:53.17 | autojack | yeah I get it now. |
| 21:53.26 | outtolunc | if congestion, playback *something* to caller |
| 21:53.26 | autojack | if congestion, play the file, then goto s,1 |
| 21:53.30 | autojack | otherwise just go back |
| 21:53.31 | autojack | yeah |
| 21:53.33 | autojack | OK cool. |
| 21:55.33 | mcfloppy | can i use hisax with asterisk? |
| 21:56.53 | asdx | esk: just _X? |
| 21:57.50 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:57.59 | *** part/#asterisk gmaruz1 (n=maruzz@213-140-6-122.ip.fastwebnet.it) |
| 21:58.21 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 21:58.45 | asdx | esk: doesn't work |
| 21:58.48 | asdx | rejected because extension not found. |
| 22:01.09 | *** join/#asterisk Acxty (n=FAclon@201.220.132.138) |
| 22:01.19 | asdx | anyone? |
| 22:01.52 | lesouvage | A customer asks for anti virus software on the Asterisk server to protect it. Is there any documented and known virus related issue thing with an Asterisk server? |
| 22:02.16 | lesouvage | asdx: If this is about an incoming line the good news is that it reach your Asterisk server. |
| 22:02.23 | JT | asdx: didn't esk say _X. ? |
| 22:02.27 | tzanger | lesouvage: you can get anti-virus software for linux |
| 22:02.27 | JT | not _X |
| 22:02.47 | Acxty | Hi guys. I am trying to develop a telephone interface to interact with some gps units. My Idea is this one. Call a number when It answer ask the person the password, verify it and then tell the options to the client, say press 1, press 2, etc. That commands need to be send by UDP to a different computer. My question is this one, can asterisk handle all this interactivity? |
| 22:02.57 | lesouvage | tzanger: but does it makes sense? |
| 22:03.09 | tzanger | lesouvage: of course not, but do customer requirements often make sense? |
| 22:03.09 | jjshoe | JT I imagine so, since . matches any number of digits. |
| 22:03.37 | jjshoe | Acxty yes. |
| 22:04.06 | lesouvage | tzanger: good question, do you have a name or pointer of the business selling this pointless linux anti virus software? |
| 22:04.49 | Acxty | jjshoe, thanks that is what I needed to know, asterisk can run under and linux system right? |
| 22:04.58 | jjshoe | Acxty yes. |
| 22:06.19 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:07.37 | Acxty | do I need a modem connected to the PC or something like that, better where can I find information explaining all that I need to run asterisk? |
| 22:07.49 | jjshoe | ~book |
| 22:07.49 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 22:08.23 | jjshoe | it's free, get it. |
| 22:08.42 | tzanger | lesouvage: clamav does it, I think that mcafee and another that escapes me also do it |
| 22:08.46 | tzanger | just google for "linux anti virus" |
| 22:08.52 | Acxty | hehe thanks again |
| 22:09.50 | jblack | Yeah. It looks like IPKall is no longer passing through CID on incoming calls. |
| 22:10.32 | *** join/#asterisk anthm (n=anthm@mb40736d0.tmodns.net) |
| 22:10.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:12.06 | [hC] | jblack: since when? Its still working for me as of this morning. |
| 22:12.11 | [hC] | I use my ipkall number every day |
| 22:12.26 | jblack | sip or iax? |
| 22:12.32 | asdx | JT: so it would be _X.,1,Dial(...) ? |
| 22:12.39 | [hC] | jblack: IAX i believe. |
| 22:12.58 | [hC] | Let me check |
| 22:14.06 | [hC] | SIP |
| 22:15.41 | *** join/#asterisk NirS (n=NirS@87.68.3.201.cable.012.net.il) |
| 22:16.31 | JT | asdx: |
| 22:16.34 | JT | ~thebook |
| 22:16.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:17.24 | asdx | i'm reading it |
| 22:17.46 | riddlebox | i dont understand how the company I work for pays ActiveVoice for support and I wait in queue for hours |
| 22:17.51 | asdx | but my boss is bugging me |
| 22:17.54 | asdx | i can't concentrate |
| 22:18.02 | asdx | i will close my IM |
| 22:18.43 | lesouvage | asdx: what internal phone numbers do you have and is it SIP or IAX2. It should be something like _X.,Answer() next line Dial(SIP/<your internal number>,20,t) |
| 22:19.39 | asdx | lesouvage: for now i just want to be able to call normal phones from PSTN, I use SIP. |
| 22:20.11 | lesouvage | asdx: so you are truing to dial out |
| 22:20.27 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:20.27 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 22:21.23 | *** join/#asterisk RoyK (n=roy@ip-79-17-149-91.dialup.ice.no) |
| 22:21.41 | *** join/#asterisk RoyK (n=roy@ip-79-17-149-91.dialup.ice.no) |
| 22:22.03 | asdx | lesouvage: i can make calls already but i want something like this: if a person dials 59521201964 from the softphone i want to have something like _X. on the dialplan and not something like _XXXXXXXXXXX |
| 22:22.22 | asdx | lesouvage: because people will be dialing numbers with more than 11 digits |
| 22:22.34 | jblack | is there a way to debug iax auth? |
| 22:22.56 | asdx | lesouvage: is there a better way to do that? |
| 22:22.58 | paci`` | goes to take a jaggy |
| 22:24.06 | asdx | nevermind, i found the pattern matching section :) |
| 22:25.19 | *** join/#asterisk [T]ank (n=[T]ank@c-71-195-194-193.hsd1.ut.comcast.net) |
| 22:26.13 | Jumpie | hey guys i was trying to install zaptel and got You do not appear to have the sources for the 2.6.18-53.1.14.el5xen kernel installed. |
| 22:26.31 | jblack | Trying to reach s@ WHAT? |
| 22:26.37 | lesouvage | asdx: the idea is that you can make clever number matchings and so limit the permissions to internal, local, or national calls. It can also be used for least cost routing (using the provider that charged the least for the specific destionation) |
| 22:27.00 | Jumpie | guess i goaa do yum -y install kernel-xen-devel ? |
| 22:27.37 | [T]ank | i am seeing an issue where when i get a call that goes into queue and is answered by an agent. This agent transfers the call to another extension, that original agent shows in use until the person he transferred the call to hangs up with the caller. what could cause that |
| 22:27.55 | asdx | lesouvage: i see, i will have to do something like that |
| 22:28.01 | asdx | set permissions and stuff |
| 22:28.59 | glaz | I am trying to configure a T122 card from digium. Everything I try to reload my chan_zap module I get an error. Can someone help? |
| 22:29.25 | glaz | s/Everything/Every time |
| 22:30.07 | lesouvage | glaz: could you copy the error messages on pastebin |
| 22:30.13 | lesouvage | ~pastebin |
| 22:30.14 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:30.17 | glaz | lesouvage: of course, hold on. |
| 22:30.40 | glaz | lesouvage: http://rafb.net/p/heJeE334.html |
| 22:31.22 | *** part/#asterisk RoyK (n=roy@ip-79-17-149-91.dialup.ice.no) |
| 22:32.34 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:33.00 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 22:33.05 | Jumpie | do i need to make, make install or do something over again when i get my fxo card? |
| 22:33.27 | Jumpie | ah, just dont do last step yet |
| 22:33.38 | *** join/#asterisk killfill (n=killfill@200.73.13.54) |
| 22:33.40 | killfill | hey |
| 22:33.51 | killfill | asterisk 1.4. cannot log the queue_log to a database, can it? |
| 22:34.13 | glaz | lesouvage: any idea^ |
| 22:34.42 | asdx | can a single X store more than one number? |
| 22:34.54 | asdx | s/store/match |
| 22:36.29 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-166-37.dsl.tul2ok.sbcglobal.net) |
| 22:38.33 | jjshoe | asdx have you read up on how dial plans work yet? |
| 22:39.21 | *** join/#asterisk nn (n=nn@75.108.238.127) |
| 22:39.36 | jjshoe | asdx if you're not going to listen to people here, at least read this http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns |
| 22:40.04 | asdx | ok |
| 22:40.46 | drmessano | Maybe Trixbox is a better solution for you |
| 22:40.59 | nn | anyone know of a decent cheap outbound SIP provider with unlimited US? |
| 22:41.07 | jjshoe | asdx "ok", as in you blatently don't listen? yuck. that's my last bit of help to you. |
| 22:41.23 | drmessano | He doesn't like to read |
| 22:42.01 | *** join/#asterisk Katty (n=Angela@adsl-68-92-251-223.dsl.stlsmo.swbell.net) |
| 22:42.28 | *** part/#asterisk bkw_ (n=brian@adsl-70-234-166-37.dsl.tul2ok.sbcglobal.net) |
| 22:44.26 | Claude2008 | hello |
| 22:44.28 | Claude2008 | can any one help |
| 22:44.30 | Claude2008 | how can i display Caller ID number on my Phone |
| 22:44.32 | Claude2008 | i keep getting uknown displayed on my Telphone Screen |
| 22:44.38 | Claude2008 | i have subscribed for the caller ID by my telphone supplier |
| 22:44.46 | Claude2008 | so they are transmiting the signal |
| 22:44.58 | asdx | jjshoe: i didn't say that |
| 22:45.01 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 22:45.01 | Claude2008 | but not sure how to set it how on my box to read that info |
| 22:45.20 | Claude2008 | can any one give me direction please |
| 22:46.35 | jjshoe | Claude2008 what kind of incoming line? analog? t1? sip? |
| 22:46.56 | *** part/#asterisk nn (n=nn@75.108.238.127) |
| 22:47.38 | Claude2008 | no i have ISDN |
| 22:47.46 | Claude2008 | i am using Sangoma A500 Card |
| 22:47.54 | Claude2008 | its a digital line |
| 22:49.55 | jjshoe | bri? |
| 22:51.31 | Claude2008 | yes it is BRI |
| 22:54.02 | tripps | anyone seen war dialing from 8778156572? i've got 50 calls to various DIDs from that number in sequence starting at 3 am this morning |
| 22:55.28 | jjshoe | Claude2008 which bri driver? |
| 22:55.39 | glaz | lesouvage: ?? |
| 22:56.34 | jjshoe | tripps did you bother to call it back? |
| 22:56.49 | jjshoe | tripps seems like a fishing numer |
| 22:56.54 | jjshoe | number* |
| 22:57.03 | jjshoe | or maybe not |
| 22:57.06 | jjshoe | it's from franklyn bank |
| 22:57.40 | jjshoe | whatever it is, seems fishy. |
| 22:57.54 | tripps | jjshoe, oh is it? it left a message where there was VM - "We have an important message for you, please hold for an operator!" |
| 22:58.17 | tripps | maybe someone screwed up the time on the script and it was legit? ;) |
| 22:58.30 | tripps | it wouldn't call every DID tho if it was |
| 22:58.53 | tripps | maybe a false incoming DID |
| 22:59.28 | tripps | oh yes definitely a phishing number |
| 22:59.42 | jjshoe | tripps poor fishing, because it doesn't ask for a ss# or anything useful |
| 22:59.45 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 22:59.51 | jjshoe | and it uses a bank name no one has ever heard of |
| 22:59.53 | tripps | i like the text to speech message |
| 23:00.09 | jjshoe | tripps are you in detroit? |
| 23:00.17 | tripps | jjshoe, no houston |
| 23:02.05 | jjshoe | well I just emailed the anti phishing team of franklin bank |
| 23:02.06 | jjshoe | we'll sere |
| 23:02.20 | tripps | jjshoe, awesome thanks! |
| 23:03.21 | Claude2008 | jjdhoe i am using Sangoma Driver |
| 23:03.43 | Claude2008 | WANPIPE |
| 23:04.22 | Claude2008 | thats the card i am using http://www.sangoma.com/products_and_solutions/hardware/digital_telephony_and_data/a500.html |
| 23:06.05 | jjshoe | I've not had the joy of using a bri, so I have no real comment :) |
| 23:06.07 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 23:06.52 | lesouvage | Are there any official figures about the uptime of an Asterisk server on a high end server. What uptime can be expecten on a full blown high end server. |
| 23:07.52 | lesouvage | Claude2008: you should be aware that when you built the Wanpipe driver zaptel is patched so you have to build zaptel after you built wanpipe. |
| 23:08.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:09.23 | *** join/#asterisk galeras (n=galeras@201.245.230.59) |
| 23:13.10 | *** join/#asterisk paulQ (n=PaulQ@72.29.76.254) |
| 23:13.12 | galeras | Sirs, please give me some tip about this: most of my remote iax calls are fine (using g729), however a few remote clients have a bad incoming audio, however from same network sip calls to another providers are fine. |
| 23:13.27 | *** part/#asterisk [T]ank (n=[T]ank@c-71-195-194-193.hsd1.ut.comcast.net) |
| 23:13.46 | paulQ | Semi quick question. Is there anyway to get a list of Actively logged into a certain queue without parsing thru Queues() in the manager API? |
| 23:13.56 | paulQ | QueueStats for some reason does not provide this information |
| 23:14.30 | jjshoe | paulQ from a script? or just in general? |
| 23:14.42 | jjshoe | give us a frame of reference |
| 23:15.39 | paulQ | jjshoe API call to a PerlAGI script. |
| 23:15.47 | paulQ | So have access to manager API and AGI functions |
| 23:15.59 | paulQ | I'm guessing parsing all that is my only method? |
| 23:16.05 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 23:16.07 | lesouvage | galerus: it is probably a codec issue. Try g.721 and see if the problem is still there |
| 23:16.21 | paulQ | Bar recompiling the app_queue to provide a Agents variable |
| 23:16.30 | lesouvage | paulq: do "show queues" on the cli |
| 23:16.58 | paulQ | Right, there is no number of logged in agents per Queue |
| 23:17.14 | paulQ | Just a list of the agents, So I would have to parse out all the information |
| 23:17.34 | paulQ | for each queue and for each agent |
| 23:17.38 | paulQ | decent amount of parsing |
| 23:19.04 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 23:19.29 | paulQ | So best is parse the long output |
| 23:19.32 | paulQ | or modify the source |
| 23:21.31 | lesouvage | paulQ: I don't have any queues available but as far as I know "show queues" shows the queus with the logged in agents with the status. |
| 23:21.48 | paulQ | Right its just not a very "parsable" manner |
| 23:21.57 | paulQ | There is no agent count |
| 23:22.11 | *** join/#asterisk VaNNi (n=VaNNi___@lgb-static-216.70.165.200.mpowercom.net) |
| 23:23.12 | paulQ | <PROTECTED> |
| 23:23.38 | paulQ | Best way I guess is to count up is count the items in q->members |
| 23:23.45 | paulQ | and make a new option.. |
| 23:25.24 | paulQ | Guessing q->members is an array |
| 23:30.13 | Claude2008 | Lesouvage i used an RPM Driver that is supplied bye Sangoma, that is compatiable for Trixbox |
| 23:30.13 | Claude2008 | so dose that meen i have have to recomple the zaptel after installing teh RPM driver |
| 23:31.11 | denon | ~trixbox |
| 23:31.11 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 23:36.07 | *** join/#asterisk pdugas (n=pdugas@74.95.28.33) |
| 23:36.12 | *** part/#asterisk pdugas (n=pdugas@74.95.28.33) |
| 23:45.55 | *** join/#asterisk ZachMen (n=Zach@74-131-58-176.dhcp.insightbb.com) |
| 23:46.07 | ZachMen | what is a good cordless sip phone? |
| 23:47.07 | jjshoe | aastra dect phone <3 |
| 23:47.32 | *** part/#asterisk galeras (n=galeras@201.245.230.59) |
| 23:47.37 | jjshoe | I haven't used the snom dect phone, if it's anything like the snom360... yuck. |
| 23:47.45 | drmessano | Cordless SIP phones are pretty lame |
| 23:47.56 | ZachMen | what about sip cordless dect phone? |
| 23:47.58 | jjshoe | but I would dodge wifi sip phones. |
| 23:48.38 | drmessano | Why not just buy an ATA and your choice of cordless? |
| 23:48.51 | ZachMen | guess i could... |
| 23:49.02 | [hC] | jjshoe: whats wrong with the snom360? |
| 23:49.15 | jjshoe | ZachMen the aastra dect phone has sick range on it, I can get from my building to the parking ramp 150 yards away |
| 23:49.36 | jjshoe | [hC] what isn't? |
| 23:49.55 | [hC] | jjshoe: i recently played with one, and liked it quite a bit. im really curious what it was you didnt like? |
| 23:50.35 | ZachMen | so what is a good ATA? can i still place calls on hold, transfers etc? |
| 23:50.36 | jjshoe | [hC] the firmware upgrade process. the fact they brick with no way to un-brick them. the side car has many flaws, including connecting two callers together, instead of placing the current caller on hold, and answering the next call. etc. |
| 23:50.59 | JT | snoms are ugly, have accoustic echo issues, just bleh really |
| 23:51.02 | drmessano | You can do a lot of things with an ATA |
| 23:51.28 | ZachMen | drehlecom what is a good ata? |
| 23:52.03 | drmessano | I like the linksys ATA's, but some call me a moron |
| 23:52.28 | ZachMen | Handytone 286 ? |
| 23:52.41 | drmessano | EWWW |
| 23:52.48 | ZachMen | k |
| 23:52.49 | drmessano | Stay away from Grandstream |
| 23:52.50 | ZachMen | :-p |
| 23:52.54 | drmessano | FAR away |
| 23:52.58 | ZachMen | i have a gx2000 no problems with it |
| 23:53.17 | jjshoe | Grandstreams are more junky then snom's |
| 23:53.17 | drmessano | BRB, I need to go wash my eyes out with bleach |
| 23:53.35 | ZachMen | LOL.... i was lookin at a cisco phone too.. |
| 23:53.53 | ZachMen | just using a wifi ap to coonect it back the the network |
| 23:54.06 | drmessano | A good Linksys ATA and a cordless works like a champ for me |
| 23:54.16 | drmessano | Wifi handsets suck |
| 23:54.24 | drmessano | Suck power.. Like crazy |
| 23:54.25 | gandhijee | hey, anyone know what tone setting is used for tiawan? |
| 23:54.30 | ZachMen | what is a "good" linksys ata |
| 23:54.32 | drmessano | You're wasting your money |
| 23:54.33 | ZachMen | brb |
| 23:54.42 | drmessano | PAP2T |
| 23:54.56 | jjshoe | dect and be done with it. |
| 23:55.51 | drmessano | I just don't see the point in buying a SIP device glued to a cordless phone technology that may be obsolete in a year |
| 23:56.04 | drmessano | If you get an ATA and a consumer cordless, the flexibility is yours |
| 23:56.34 | *** join/#asterisk quigon (n=matias@nat/redhat/x-e3570440df33e35c) |
| 23:59.41 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 23:59.44 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |