| 00:00.06 | hmmhesays | my snom 190 likes to randomly lose its password |
| 00:01.27 | GTX | heh hmmhesays |
| 00:01.33 | GTX | Is there anyway to hard reset them |
| 00:01.39 | GTX | Like holding down a few buttons on start? |
| 00:02.03 | riddlebox | what do you guys think about the GrandStream GXP-2000? |
| 00:02.28 | Zodiacal | i can't belive no one knows where asterisk stores changed ext. passwords.. |
| 00:02.39 | *** join/#asterisk rene1 (n=rene1@189.140.234.203) |
| 00:03.14 | rene1 | hello |
| 00:04.21 | hmmhesays | what are you asking? |
| 00:04.25 | knathraak | okay, it appears I was using the A(x) option in the Dial() app correctly. it's just that it immediately starts playing the announcement rather than waiting for the callee to answer, so by the time the callee does answer, they don't hear anything, and the calls are bridged |
| 00:05.33 | knathraak | anyone know how to tell Dial() to wait until the callee has answered to play back the announcement? |
| 00:06.06 | knathraak | peeked at the source for app_dial.c, but I don't get any clue there. |
| 00:11.20 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 00:13.13 | AndyCap | anyone know who really makes this or what's its name is http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=270054669203 ? |
| 00:16.02 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 00:16.36 | rene1 | looks like a pa 1688 based phone to me |
| 00:18.18 | AndyCap | rene1: sounds like a start. thanks. :) |
| 00:18.39 | rene1 | sure |
| 00:18.51 | rene1 | it is dirt cheap tho 20 bucks?? |
| 00:19.15 | rene1 | i never saw them under 45 bucks |
| 00:19.18 | wasim | yeh |
| 00:19.34 | AndyCap | rene1: you mean http://www.voip-info.org/wiki/view/PA168 right? |
| 00:19.40 | rene1 | yes |
| 00:19.58 | rene1 | wasim is the expert in those matters |
| 00:20.34 | AndyCap | rene1: heh, I guess the price will rise as the auction end nears. |
| 00:21.01 | rene1 | i think i once saw 20 bucks SIP hardphones but i cant recall the brand |
| 00:23.13 | *** join/#asterisk coppice (n=chatzill@213.143.17.210.dyn.pacific.net.hk) |
| 00:24.13 | AndyCap | I think we have a winner: http://www.iplink.cn/newsite/en/products/JR168-200d.htm |
| 00:25.50 | rene1 | cool :) |
| 00:26.14 | rene1 | not all ther phones look toyish |
| 00:26.29 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 00:26.45 | Dovid | anyone here have clients on bell south ? |
| 00:26.53 | AndyCap | no, -820 looked much better |
| 00:26.58 | rene1 | the 210 is nice too |
| 00:27.38 | AndyCap | hmm, what's the difference. 210 has iax, and 820 not? |
| 00:27.49 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 00:28.17 | AndyCap | G729/GSM/iLBC Heh, wonder if iplink pays their royalties |
| 00:28.36 | rene1 | the 820 seems to have iax |
| 00:28.56 | rene1 | gsm is royalty free... g729 was like USD 1 for product |
| 00:29.12 | rene1 | i guess ilbc does not need to pay patents |
| 00:29.17 | pipipi | anybody have any other suggestions for a free voip-phone other than xten-lite? xten-lite's giving me some issues. |
| 00:29.28 | rene1 | sjphone |
| 00:29.39 | pipipi | thanks |
| 00:29.47 | hads | Actually, I believe later versions of gsm are covered by patents. |
| 00:29.49 | pipipi | any other suggestions for free ones, that will work on linux? |
| 00:30.04 | AndyCap | rene1: yeah, I was thinking of G729. and I guess there's a large initial/minimum payment |
| 00:30.17 | AndyCap | and .cn isn't all that big on IP anyway |
| 00:30.21 | rene1 | heh |
| 00:30.25 | rene1 | no they arent |
| 00:30.51 | AndyCap | pipipi: kphone, ekiga maybe, cornfed, idefisk (iax) |
| 00:31.05 | AndyCap | hmm, no, not idefisk, but some other iaxclient based phone |
| 00:31.19 | rene1 | mmm the yate project has a phone |
| 00:31.21 | rene1 | looks nice |
| 00:31.46 | rene1 | for lin & win |
| 00:32.48 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:32.54 | AndyCap | softphones with numberpads in the GUI are such a turnoff. :) |
| 00:33.18 | AndyCap | unless they're made for palm/windows ce/symbian et.al |
| 00:33.37 | pipipi | ok thanks |
| 00:33.42 | pipipi | thats enough phones :) |
| 00:33.54 | pipipi | brb, reboot |
| 00:33.55 | pipipi | thanks guys |
| 00:33.56 | *** part/#asterisk knathraak (n=knath@pool-71-252-87-31.washdc.east.verizon.net) |
| 00:34.05 | AndyCap | hehe, and the snom 360 softphone. :P |
| 00:35.15 | rene1 | heh |
| 00:36.18 | *** part/#asterisk rene1 (n=rene1@189.140.234.203) |
| 00:40.59 | wasim | that 810 is not bad looking at all ... |
| 00:48.42 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:51.31 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:52.35 | *** join/#asterisk Marshall16 (n=Marshall@cpe-24-95-78-61.columbus.res.rr.com) |
| 00:57.32 | *** join/#asterisk sumasuma (n=sumaoksu@61.14.86.23) |
| 01:01.09 | *** join/#asterisk beehive (n=michael@pool-71-246-224-231.washdc.fios.verizon.net) |
| 01:02.31 | beehive | hello all. opinions regarding the VoiceOne company (the sip termination bunch... not the gui) would be appreciated. |
| 01:04.48 | Zodiacal | any ideas why i have to reset my polycom's after i reload asterisk in order for the polycoms to see the hints of cisco phones? |
| 01:08.15 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 01:08.29 | hoobastooba | is it possible to make an agent from agents.conf be a cell phone number? |
| 01:09.28 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:10.59 | nick125 | anyone here have any experience with viatalk? |
| 01:11.28 | nick125 | hoobastooba: you could make it a SIP channel SIP:host/number I think, or something like that |
| 01:13.48 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 01:15.29 | hoobastooba | better yet, is there a way to add an agent from agents.conf to a queue permanently associated to the same extension so they do not have to log on and off? |
| 01:25.41 | *** join/#asterisk pipipi (n=suedoh@adsl-68-126-112-221.dsl.irvnca.pacbell.net) |
| 01:26.12 | pipipi | hi, anybody know why when i plugin my speakers/microphone and i use asterisk, the person on the other end hears a Echo? |
| 01:27.28 | pipipi | its weird, im not sure if its the headset quality that is bad, or if its Asterisk causing the Echo |
| 01:27.38 | pipipi | though when i turn the XTen volume all the way down it becomes ok |
| 01:29.25 | carrar | feedback |
| 01:29.33 | carrar | use headphones |
| 01:29.57 | carrar | a nice USB headset will do you nicely |
| 01:30.12 | Dovid | if i want a phone to re-register with asterisk once a minute is it a setting in a phone or in asterisk ? |
| 01:30.22 | pipipi | heh |
| 01:30.23 | carrar | phone |
| 01:30.30 | pipipi | how about fxotune, will that help? |
| 01:30.36 | pipipi | i've been seeing some information on this here |
| 01:31.55 | carrar | not as much as using the USB headset |
| 01:36.58 | Dovid | does anyone know how to set the registry settings in SNOM so it registers once a minutee ? |
| 01:37.04 | Dovid | can this be it ? |
| 01:37.06 | Dovid | SIP Session Timer (s): |
| 01:38.57 | *** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au) |
| 01:40.05 | Penggu | hi all. if there are say, 20 extensions, each with their own extension number, can a 'person' be dialled (without dumping the phone extension #s) by tracking them through a 'login' wherever they may be, by the same number? |
| 01:40.23 | pipipi | carrar: why is a USB headset better than a normal headset? |
| 01:40.37 | Penggu | or perhaps they're known to be at a specific desk most of the time, so a 'default' extension their number would be mapped to |
| 01:40.39 | Penggu | or something like that |
| 01:41.11 | Penggu | basically having a layer or user extenion #s on top of, and independant of phone extension #s. |
| 01:41.12 | beehive | VoiceOne... opinions anyone? |
| 01:41.17 | Penggu | layer of = layer of |
| 01:41.21 | Penggu | argh |
| 01:43.58 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 01:44.13 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:47.23 | Dovid | will there be any load on asterisk if a pone re-registers every minute ? |
| 01:50.29 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 01:53.47 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 02:00.32 | *** join/#asterisk msmullin (n=msmullin@63-230-0-101.slkc.qwest.net) |
| 02:00.56 | msmullin | hello |
| 02:03.42 | msmullin | i would like to use Asterisk to set up a small home-office pbx for 2 people w/ handsets for each. can anyone recommend some cheap hardware? |
| 02:04.20 | gerphimum | how cheap is cheap |
| 02:04.24 | hoobastooba | use softphone if you want cheap |
| 02:04.32 | Dovid | lol |
| 02:04.45 | Dovid | i luv when clients want the cheapest and then complain about hardware |
| 02:04.48 | msmullin | i'm willing to pay whatever it takes, but i would like the cheapest :) it will be a hobby system for me at first |
| 02:05.08 | hoobastooba | cuz cheap hardphone will cause you to have to spend an extra $22.95 for a box of bullets so you can shoot 19 other people then yourself |
| 02:05.22 | hoobastooba | especially if it is hobby... use softphone. |
| 02:05.25 | hoobastooba | i like xlite. |
| 02:05.31 | gerphimum | msmullin >> do you need to build a new system or do you have old parts floating around |
| 02:05.31 | hoobastooba | quality phone for what I do. |
| 02:05.57 | Dovid | lol |
| 02:06.06 | Dovid | use xlite for testing |
| 02:06.08 | msmullin | hoobastooba: thanks ... looking it up now |
| 02:06.17 | msmullin | gerphimum: i have a linux box i can use, but no PBX equipment |
| 02:06.27 | gerphimum | what kind of phone service do you have |
| 02:06.27 | hoobastooba | you will save yourself many hours of heartache, that a cheap phone can cause |
| 02:08.06 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:08.23 | msmullin | is it cheaper to get FXO or FXS |
| 02:08.40 | gerphimum | fxo is needed for outside lines, fxs is for internal phones |
| 02:09.12 | gerphimum | so, if you have 2 phone lines, youll need 2 fxo ports, and if you have 2 internal analog phones, youll need 2 fxs ports |
| 02:09.46 | hoobastooba | for hobby use you can use an sip provider until you decide if you want to make the investment for pci devices |
| 02:09.48 | msmullin | ah thanks for clarifying |
| 02:09.52 | msmullin | so is it feasible/cheap to use $9 analog phones from walmart with a PBX? |
| 02:09.58 | hoobastooba | yep |
| 02:10.08 | gerphimum | well, yeah, but the fxo/fxs card is pretty expensive |
| 02:10.13 | gerphimum | $250 for a decked out one |
| 02:10.14 | hoobastooba | but you will spend quite a bit for hardware to support it |
| 02:10.45 | hoobastooba | get an sip provider for incoming service and use softphone for testing. |
| 02:10.59 | hoobastooba | once you get the hang of it, figure out if you want to do fxs or something else. |
| 02:11.20 | heison | ~seen JerJer |
| 02:11.39 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 28d 9h 38m 35s ago, saying: 'apt-file search lspci'. |
| 02:11.54 | msmullin | i was hoping i could get away with a linux box, couple voice modems, and $9 wal-mart phones |
| 02:11.59 | msmullin | lol |
| 02:12.01 | hmmhesays | no |
| 02:12.08 | gerphimum | modems wont work :| |
| 02:12.18 | hmmhesays | you can get away with a linux box a couple cheap ata's and $9 dollar walmart phones |
| 02:12.35 | hmmhesays | and a couple hundred bucks for someone to write a dialplan for you |
| 02:15.13 | msmullin | gerphimum: i have a single outside line. regular dial tone service |
| 02:16.00 | gerphimum | http://www.digium.com/en/products/hardware/tdm400p.php |
| 02:16.11 | msmullin | so if i use soft phones, and i have already set up a linux box on my home network... what equipment would i need to get that outside line set up with asterisk if not a voice modem? a fxo card? |
| 02:16.23 | gerphimum | for the win, if you want to convert your home phone line into a decked out pwnage line |
| 02:17.07 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 02:18.23 | msmullin | ah i see |
| 02:18.38 | msmullin | this has 4 fxs/fxo modules |
| 02:18.46 | msmullin | if i use softphones won't i only need 1? |
| 02:18.54 | gerphimum | its a wildcard |
| 02:19.03 | gerphimum | you can put up to 4 daughger modules |
| 02:19.09 | *** join/#asterisk lule (i=lule@host76.201-252-119.telecom.net.ar) |
| 02:19.17 | gerphimum | so if you have a softphone (and thats it), youll just need 1 fxo daughter card |
| 02:19.36 | *** part/#asterisk lule (i=lule@host76.201-252-119.telecom.net.ar) |
| 02:19.49 | msmullin | so i will need to purchase this wildcard + 1 fxo daughter card |
| 02:19.54 | gerphimum | roger. |
| 02:19.59 | msmullin | can you link me to the daughter card? |
| 02:20.31 | gerphimum | you should google tdm400p and look for resellers |
| 02:20.38 | gerphimum | they sell them as a package most of the time |
| 02:20.44 | gerphimum | i dont personally know any good resellers |
| 02:21.39 | msmullin | ok cool |
| 02:21.47 | msmullin | so what are those 4 ports i see on the card in the picture for? |
| 02:22.05 | gerphimum | theyre for the daugheter modules |
| 02:22.17 | gerphimum | you can put either an fxo or an fxs card in those places |
| 02:22.18 | msmullin | ah so i wouldn't plug in any phone lines or hardphones |
| 02:22.20 | gerphimum | you can mid and match |
| 02:22.28 | gerphimum | mix* |
| 02:22.51 | gerphimum | you plug in hardphones to fxs ports, and phone lines into fxo ports |
| 02:23.28 | msmullin | i'm wondering if theres a single PCI card that includes the functionality i get from a wildcard + 1 fxo daughter board |
| 02:23.53 | gerphimum | there is, but from what i hear its not that good... |
| 02:23.59 | gerphimum | i think its the digium x100 |
| 02:24.09 | gerphimum | youd have to ebay it though, they dont make them anymore |
| 02:24.15 | gerphimum | they have clones, though... |
| 02:26.09 | msmullin | wow those are pretty cheap on eBay too |
| 02:26.15 | msmullin | what sort of problems have you heard of? |
| 02:26.28 | gerphimum | just that theyre all around crappy :| |
| 02:26.31 | gerphimum | i have no experience with them |
| 02:26.39 | gerphimum | youd have to talk to someone who actually has :- |
| 02:26.42 | gerphimum | :-/* |
| 02:27.00 | msmullin | ok thanks this helps a lot |
| 02:27.36 | msmullin | i'm envisioning a small mini-itx linux box with this digium x100 clone PCI card installed |
| 02:27.43 | msmullin | for under $300 |
| 02:27.57 | hads | Stop imagining. X100 cards are bad. |
| 02:28.17 | gerphimum | ah, im not crazy. *whew* |
| 02:28.21 | msmullin | lol! |
| 02:28.28 | hoobastooba | so wildcard...you will be much happier |
| 02:28.46 | hoobastooba | then if you ever want to do fxs and fxo you will have the ability. |
| 02:28.57 | hoobastooba | *do wildcard |
| 02:29.05 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:29.07 | hads | Digium TDM400/Sangoma A200/Linksys SPA3102 |
| 02:29.22 | hoobastooba | i vote sangoma |
| 02:29.29 | msmullin | it looks like google has the wildcards cheaper than the digium direct pricing |
| 02:29.33 | msmullin | they must be used |
| 02:29.46 | hoobastooba | not always |
| 02:29.54 | hoobastooba | digium is usually higher priced. |
| 02:30.14 | hads | Digium sell a retail box, the resellers sell OEM packed versions. |
| 02:30.34 | hoobastooba | good explanation... I did not realize that myself |
| 02:30.52 | msmullin | ahi see |
| 02:31.22 | msmullin | hads: can you link me to your recommendation there? i'm having trouble finding anything on Google |
| 02:31.33 | *** join/#asterisk Crumbles (i=Crumbles@24-178-73-251.dhcp.gwnt.ga.charter.com) |
| 02:31.52 | Crumbles | someone please help me before I throw my asterisk server off of the top floor of my house |
| 02:31.52 | hads | link to what? |
| 02:31.56 | msmullin | Digium TDM400/Sangoma A200/Linksys SPA3102 |
| 02:32.06 | *** join/#asterisk Laggy_McGee (n=jchadwic@pool-71-245-122-155.cmdnnj.fios.verizon.net) |
| 02:32.11 | hoobastooba | those are three different items |
| 02:32.14 | *** part/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 02:32.15 | msmullin | ohh ok |
| 02:32.17 | hads | msmullin: Those are some choices for a FXO device, there are more. |
| 02:32.26 | hoobastooba | Sangoma A200 |
| 02:32.26 | gerphimum | is the linksys any good ? |
| 02:32.43 | hoobastooba | http://www.voipsupply.com/index.php?cPath=99_359 |
| 02:33.26 | hoobastooba | i use sangoma products and have been happier than with digium. dunno about linksys |
| 02:33.46 | Crumbles | I'll paypal someone $50 if they can help me with this seriously annoying problem that I can't figure out |
| 02:34.06 | hoobastooba | the linksys is a standalone device, not pci |
| 02:34.14 | hoobastooba | check out the link i sent. |
| 02:34.21 | msmullin | looking now |
| 02:34.22 | hads | Crumbles: Ask a question, someone may answer. |
| 02:35.08 | heison | anyone from Nufone here? |
| 02:35.22 | Crumbles | ok, I have EVERYTHING working with my asterisk server and Sunrocket except INBOUND calls. For some reason those aren't working. When I run asterisk with -rvvv nothing shows up on the console. However, if I go into sip debug mode, a bunch of crap shows up when I make an inbound call. I can't figure out why it's not working though. |
| 02:35.24 | justinu|laptop | they don't come 'round here no more |
| 02:35.55 | hads | Crumbles: "bunch of crap" doesn't help. |
| 02:35.59 | Crumbles | lol |
| 02:36.00 | Crumbles | yea |
| 02:36.00 | Crumbles | well |
| 02:36.05 | Crumbles | I have the TXT file if you wnat to look at it |
| 02:36.10 | Crumbles | I can't really paste it in this channel |
| 02:36.12 | EyeCue | Crumbles, tcpdump on your inbound port and see if you see the request coming in from the outsite |
| 02:36.15 | hoobastooba | pastebin.ca |
| 02:36.17 | hads | ~pb |
| 02:36.18 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:36.46 | Crumbles | ok, hang on |
| 02:36.48 | Crumbles | let me look at that |
| 02:37.34 | justinu|laptop | off topic question: filezilla seems to have major speed issues with sftp over ssh2, and apparently it's a known bug. question is, what's a good alternative? |
| 02:37.48 | EyeCue | sftp is always slow :| |
| 02:38.00 | EyeCue | but it would be nice if it turns out to be a bug :) |
| 02:38.01 | Crumbles | ok, there is no "asterisk" channel selection for that paste website |
| 02:38.14 | EyeCue | Crumbles, just paste url here |
| 02:38.17 | Crumbles | roger |
| 02:38.31 | justinu|laptop | the sftp preferences panel in filezilla mentions the bug |
| 02:38.39 | Crumbles | http://channels.debian.net/paste/4466 |
| 02:39.03 | EyeCue | ooer, does it now |
| 02:39.14 | EyeCue | only inc ertain os's ? |
| 02:39.33 | EyeCue | ah, it does too |
| 02:39.34 | justinu|laptop | on windows |
| 02:39.47 | EyeCue | oh, its the ssh2 lib in putty |
| 02:39.48 | justinu|laptop | not sure about others, but apparently it uses putty's sftp/ssh2 code which has issues |
| 02:40.26 | justinu|laptop | i couldn't get it to work with ssh1, not sure if I need to modify something on the openssh side for that to work |
| 02:40.50 | EyeCue | i just use tls/ssl, im happy with that |
| 02:41.11 | justinu|laptop | ftp over ssl? |
| 02:41.26 | EyeCue | well, ssl encrypted ftp server |
| 02:41.33 | EyeCue | with tls for the init challenge shit |
| 02:41.35 | justinu|laptop | yeah... i gotta look into that |
| 02:41.39 | EyeCue | pure-ftpd |
| 02:41.46 | EyeCue | im pretty sure nc does it too |
| 02:41.47 | EyeCue | unsure though |
| 02:41.53 | justinu|laptop | cool, thanks |
| 02:42.11 | EyeCue | ive used pure exclusively for a while, nice virtual user config, with sql backend support |
| 02:42.30 | justinu|laptop | i'm gonna emerge it and check it out |
| 02:44.37 | *** join/#asterisk ShadwDrgn (n=arcane@adsl-33-167-175.asm.bellsouth.net) |
| 02:44.55 | ShadwDrgn | i bet it's because he mis-spelled "canreinvite" in his sip.conf and put "careinvite" |
| 02:45.07 | Crumbles | *sigh* |
| 02:45.10 | Crumbles | I fixed that! |
| 02:45.11 | ShadwDrgn | LOL :D |
| 02:45.28 | EyeCue | laf |
| 02:45.31 | msmullin | does Asterisk have a web control panel? |
| 02:46.27 | hoobastooba | msmullin: depends on what you want to do |
| 02:46.32 | hoobastooba | astguiclient does exist. |
| 02:46.53 | hoobastooba | but by the time you figure out how to install it, you will find that it is easier to just modify the configs when you need changes |
| 02:46.58 | Crumbles | hads, any idea? |
| 02:49.53 | Crumbles | also, if you guys want to see my threads with all my .CONF files, and stuff I have tried, you can look here: http://forums.digium.com/viewtopic.php?t=11125 |
| 02:52.34 | msmullin | it would be cool if someone wrote a Plesk or CPanel module for Asterisk management :) |
| 02:53.06 | hoobastooba | that is what astguiclient is... but installing is just as much work as configuring asterisk to begin with |
| 02:55.27 | *** join/#asterisk alerios (n=alerios@190.24.100.110) |
| 02:55.55 | *** join/#asterisk sumasuma (n=suma@61.14.86.23) |
| 02:59.31 | msmullin | i found the asterisk o'reilly book so i'm reading that now |
| 02:59.40 | msmullin | learning what modules and daughterboards are :) |
| 03:00.22 | msmullin | have any of you sold asterisk pbx installation services or hardware before? |
| 03:02.04 | *** part/#asterisk icyfire0573 (n=icyfire@u1016342.ul.warwick.net) |
| 03:02.13 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 03:04.40 | tengulre | does the asterisk as a h323 gatekeeper? |
| 03:05.01 | Crumbles | so, does anyone want to help me troubleshoot why my inbound calls aren't working? pweese? |
| 03:06.18 | ShadwDrgn | did you spell 127.0.0.1 right? |
| 03:06.39 | tengulre | anybody ? |
| 03:08.19 | tengulre | anybody alive? |
| 03:09.26 | hmmhesays | Crumbles: i'm a cheap 75.00 |
| 03:09.33 | hmmhesays | <PROTECTED> |
| 03:09.36 | Crumbles | deal if you can fix this |
| 03:09.42 | hmmhesays | k |
| 03:09.49 | hmmhesays | what do you have for a set up |
| 03:09.50 | Crumbles | it's 90% done |
| 03:09.54 | Crumbles | only one thing isn't working |
| 03:10.06 | hmmhesays | explain to me your problem |
| 03:10.08 | Crumbles | ok |
| 03:10.13 | hmmhesays | you can even pm me if you like |
| 03:10.21 | hmmhesays | people that pay can have that privilege |
| 03:17.02 | sumasuma | what is the best gui for conference management ? |
| 03:33.52 | hoobastooba | anyone here using hylafax? |
| 03:34.47 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 03:34.48 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:40.02 | msmullin | oh there's a new version of Asterisk GUI that looks much better 1.4 |
| 03:41.41 | *** part/#asterisk Crumbles (i=Crumbles@24-178-73-251.dhcp.gwnt.ga.charter.com) |
| 03:44.00 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 03:44.07 | *** join/#asterisk Abydos313 (i=abydos31@ppp-71-134-22-163.dsl.irvnca.pacbell.net) |
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| 03:54.23 | *** join/#asterisk cryptnix (n=andrew@68-188-226-83.dhcp.bycy.mi.charter.com) |
| 03:56.33 | *** join/#asterisk awannabe (n=brad@ip24-251-135-202.ph.ph.cox.net) |
| 03:57.19 | awannabe | hey all, i have two PRIs on seperate trunk groups, wahts the easiest way to have * pick between the two? basically try one, then the 2nd one if first one is busy. its the best way? |
| 03:58.24 | Juggie | if the line is busy? or if all the channels are busy |
| 03:58.45 | awannabe | well i just want to make sure outgoing calls use both pris basically |
| 03:59.13 | awannabe | we did two seperate trunk groups for redudnacny, so i want to have * try to use both, but if one is down it will just use the other |
| 04:00.04 | Juggie | * does that by default. |
| 04:00.35 | awannabe | how does it kow two use both trunk groups? |
| 04:01.05 | Juggie | configure your trunks in /etc/zaptel.conf and then in /etc/asterisk/zapata.conf create a channel group containing the channels from both trunks. |
| 04:01.17 | awannabe | oh wait, im a iddiot |
| 04:01.46 | awannabe | being sick will do that to you, duh the channel group itself knows what channels are in it, ok ill be quiet again...lol |
| 04:01.51 | Juggie | then you can do whatever kind of dial you want, round robin, least used, start from the bottom, start from the top, etc. |
| 04:01.55 | *** join/#asterisk nortex (n=nortex@66.76.52.50) |
| 04:01.56 | Juggie | look at the docs for chan_zap |
| 04:02.03 | awannabe | yeah, im in that now |
| 04:02.34 | awannabe | i guess sequential is the best probally? just so you dont have channels all over the place in use |
| 04:04.35 | msmullin | so if i have a single POTS line i can either buy a Digium TDM400P wildcard + 1 FXO and use softphones or buy a Digium TDM400P wildcard + 1 FXO + 2 FXS + 2 hardphones that's all I need to get started with asterisk (assuming i have the linux box and other home networking equipment)? |
| 04:05.06 | awannabe | msmullin, pretty much, but why use analog phones? |
| 04:05.15 | gerphimum | theyre cheap |
| 04:05.16 | pipipi | msmullin: u can start using Asterisk with 1 Machine for Asterisk, a Switch, and a Single Workstation Machine :) |
| 04:05.30 | awannabe | using a analog telephone adapter will be cheaper then FXS ports |
| 04:05.42 | pipipi | actually you dont even need the switch, you can use Crossover cables between Asterisk Machine -> Workstation |
| 04:05.45 | msmullin | pipipi: how will i get incoming calls? you mean i can use softphones to call between computers on the network? |
| 04:05.47 | pipipi | but then you can only call yourself :P |
| 04:05.53 | pipipi | right |
| 04:05.56 | msmullin | awannabe: analog phones? what's the alternative? |
| 04:06.00 | pipipi | heh, i was just playing around... nevermind me then |
| 04:06.06 | hads | Yeah, a SPA3102 would give you one FXO and one FXS |
| 04:06.07 | awannabe | voip phones, heh |
| 04:06.17 | msmullin | how much are voip phones |
| 04:06.17 | hads | But IP phones are much more fun. |
| 04:06.22 | hads | 42 |
| 04:06.29 | awannabe | agrred |
| 04:06.43 | awannabe | i dont even know what the cheapest phone is anymore |
| 04:06.56 | msmullin | ya i'm just interested in the cheapest hardphones |
| 04:06.56 | awannabe | the snoms rock, specially for mass provising, uber pimp |
| 04:07.04 | msmullin | i had planned on $9 wal-mart phones |
| 04:07.14 | msmullin | but if i get 2 FXS modules, that means i need digital hardphones right? |
| 04:07.22 | *** join/#asterisk KuJaX (n=one@c-67-164-204-41.hsd1.ut.comcast.net) |
| 04:07.31 | gerphimum | no, analog hardphones |
| 04:07.38 | awannabe | yeah analog phones |
| 04:07.42 | gerphimum | digital phones plug into your computer network |
| 04:07.46 | gerphimum | via ethernet |
| 04:07.47 | awannabe | just el cheapo target specials |
| 04:07.53 | msmullin | ah that sounds better |
| 04:08.07 | awannabe | i just got coordless phones from walmart for 10 bucks, lol |
| 04:08.08 | msmullin | so i can get the wildcard + 1 FXO for the POTS line-in and then hook up some VoIP phones to the network ? |
| 04:08.17 | gerphimum | yep |
| 04:08.23 | msmullin | that is what i'd like to do |
| 04:08.27 | awannabe | yeah, FXO are for POTS/normal telephones lines |
| 04:08.28 | msmullin | any VoIP phone recommendatons? |
| 04:08.36 | awannabe | i love the snoms |
| 04:08.59 | hads | msmullin: For price and decent quality the Linksys SPA range is quite decent. |
| 04:09.00 | msmullin | awannabe: can you link mE? |
| 04:09.10 | hmmhesays | quintum series is ok |
| 04:09.13 | hmmhesays | spendy though |
| 04:09.15 | hads | msmullin: Google! |
| 04:09.24 | awannabe | http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-41917430016.htm |
| 04:09.47 | awannabe | i use voipsupply.com for all my gear, but unless your buying more then a few there pricing is kind of crap, heh |
| 04:10.25 | msmullin | i wonder if there are voip phones for less than $30 |
| 04:10.35 | awannabe | nope |
| 04:10.49 | awannabe | you get what you pay for with voip phones for the most part, something for 30 would be not even useable |
| 04:11.40 | msmullin | in that case, the cost of the VoIP hard phones would make me want to go back to the 2 FXS modules and just use analog phones |
| 04:11.57 | msmullin | ...maybe |
| 04:12.29 | msmullin | i wonder if Skype would work as a softphone |
| 04:12.33 | awannabe | you cant really do anything with analog phones |
| 04:12.44 | awannabe | skype works with skype only |
| 04:12.54 | hads | Skype = silly |
| 04:13.05 | hmmhesays | you can use uplinke to forward a skype call to asterisk |
| 04:13.09 | hmmhesays | kind of clunky but it works |
| 04:13.22 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 04:14.10 | awannabe | the polycom 301s are decent |
| 04:14.15 | awannabe | forgot how much they are |
| 04:15.28 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:15.28 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:16.09 | *** join/#asterisk TheCops (n=henri@got.securebinary.com) |
| 04:16.11 | TheCops | hi |
| 04:17.52 | TheCops | I have something weird in my log file, on idle, asterisk generate this log: "Stopping retransmission on '3148069417b9f57e008f3c3a08ee7aed@66.158.143.66' of Request 102" and Auto destroying call '39516f44-410b5f22-665d0a8f@72.10.145.3', someone can explain what that mean and what is the problem? |
| 04:19.26 | msmullin | would one of these work? |
| 04:19.27 | msmullin | http://cgi.ebay.com/CISCO-CP-7960G-CH1-7960-IP-PHONE-Cisco-7960_W0QQitemZ120052409798QQihZ002QQcategoryZ61840QQrdZ1QQcmdZViewItem |
| 04:19.31 | msmullin | they're going cheap on eBay |
| 04:19.56 | awannabe | ewww, dont get ciscos!! |
| 04:20.20 | msmullin | but any VoIP phone will work, right? |
| 04:20.38 | awannabe | that talks SIP, well SIP I would suggest |
| 04:20.57 | msmullin | so VoIP + SIP |
| 04:20.58 | msmullin | ok |
| 04:21.12 | msmullin | thanks guys |
| 04:21.16 | msmullin | gtg ttyl |
| 04:21.32 | awannabe | a buck for that, that thing is broke and stolen i bet! lol |
| 04:32.34 | file | [TK]D-Fender: so I was like omg |
| 04:33.06 | nick125 | awannabe: probably... |
| 04:33.16 | awannabe | yeah, i think so to |
| 04:33.39 | nick125 | considering that phone usually goes for...what? $200? $300? |
| 04:34.00 | awannabe | last time i check like 275 refurbed |
| 04:34.06 | nick125 | aah |
| 04:34.08 | awannabe | that phone also had the SCCP license |
| 04:34.18 | nick125 | $565 new |
| 04:34.20 | awannabe | so then technially you have to buy smartnet to get the SIP firmware on it, heh |
| 04:38.17 | awannabe | i dont like the ciscos, they have no features, its just a phone really, heh |
| 04:38.46 | gerphimum | they look sweet |
| 04:38.49 | Sedorox | where can you buy smartnet for a phone? |
| 04:39.08 | Sedorox | I got a 7912g.. and I wanna get it working in general.. but I don't have the files for the tftp server |
| 04:39.19 | awannabe | authorized cisco reseller, but many wont sell you it unless u buy from them |
| 04:39.27 | Sedorox | ah |
| 04:39.29 | awannabe | since they dont really make alot on the smartnet, heh |
| 04:39.33 | Sedorox | yea |
| 04:39.50 | Sedorox | I mean I have a cisco router that I could connect it to.. have it working with asterisk for the fxs vic... |
| 04:39.57 | Sedorox | but it won't go past "configuring IP" |
| 04:40.27 | awannabe | you got SIP firmware on it? |
| 04:40.35 | Sedorox | I'm not sure whats on it right now |
| 04:40.51 | awannabe | if u got to settings and crap it will tell u |
| 04:41.08 | awannabe | well, it will say like SIP server or something, or Callmanger server, i forget noe |
| 04:41.15 | awannabe | i havent been in one for like a year, heh |
| 04:41.19 | Sedorox | I can get to the firmware versions.. |
| 04:41.36 | awannabe | i think it says SIP or SCCP on there |
| 04:41.56 | Sedorox | call manager 1 |
| 04:41.56 | Sedorox | hmm |
| 04:42.24 | Sedorox | :/ |
| 04:42.31 | awannabe | yeah, you can get the SIP firmware on various sites |
| 04:42.36 | awannabe | if you google u should be able to get it |
| 04:42.42 | awannabe | i dont ahve it anymore, or Id give it to you, heh |
| 04:42.47 | awannabe | oh wait |
| 04:42.55 | awannabe | 7912 is SCCP only i think |
| 04:43.19 | Sedorox | no.. I'm seeing links to the sip firmware on voip-info |
| 04:43.22 | Sedorox | but you need cco login |
| 04:43.22 | Sedorox | :/ |
| 04:43.43 | ShadwDrgn | lol |
| 04:43.45 | awannabe | for the 7912? |
| 04:43.47 | ShadwDrgn | this is great |
| 04:43.53 | Sedorox | I find it funny you can freely download the 40/60 firmware.. but not the 12 |
| 04:43.58 | ShadwDrgn | now we can't get my sipphone to work externally |
| 04:44.01 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 04:44.15 | ShadwDrgn | and i have like practically every port ever forwarded here |
| 04:44.16 | Sedorox | haychr15@email.pct.edu, woerus14@email.pct.edu, herchr56@email.pct.edu, ziechr96@email.pct.edu, webmic27@email.pct.edu, youeri66@email.pct.edu |
| 04:44.19 | Sedorox | errr shit |
| 04:44.24 | Sedorox | firmware and other software can be found at: |
| 04:44.24 | Sedorox | http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml |
| 04:47.47 | awannabe | hope no one spams those poor email addresses, lol |
| 04:48.17 | Sedorox | yea.. I do too |
| 04:48.23 | Sedorox | sometimes I hate putty |
| 04:48.55 | awannabe | poop to putty, all about ssh from linux prompt, heh |
| 04:49.16 | Sedorox | lol |
| 04:49.25 | Sedorox | well my linux box is playing a video right now |
| 04:50.27 | awannabe | ahh |
| 04:50.45 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
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| 05:08.57 | ShadwDrgn | why would i be able to hear the voicemail but not another extension when dialed directly? |
| 05:09.01 | ShadwDrgn | my phone is internet and his is lan |
| 05:09.06 | ShadwDrgn | but they're both registered |
| 05:09.20 | ShadwDrgn | and when i call to his asterisk box the voicemail system works perfect but not phone 2 phone |
| 05:09.21 | ShadwDrgn | :~( |
| 05:10.01 | aadilismail | hi |
| 05:28.51 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 05:39.53 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.12.181.revip2.asianet.co.th) |
| 05:40.16 | HaMYaI | hi , is there a way to see a list of active meetme room #? |
| 05:41.58 | JunK-Y | meetme list <confno> |
| 05:59.53 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
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| 06:07.13 | hmmhesays | ok I need to get the latest version of x-plane |
| 06:07.43 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 06:12.22 | *** join/#asterisk sandra78 (n=sandrita@200.106.40.1) |
| 06:13.04 | sandra78 | hi does anyone has used the grandstream gxw4108? |
| 06:13.14 | sandra78 | with asterisk? |
| 06:14.10 | HaMYaI | JunK-Y: yeah, that's what I'm doing |
| 06:15.09 | HaMYaI | JunK-Y: by reading configuration file and do a "asterisk -rx "meetme list <roomno>" but this isn't practical |
| 06:15.30 | *** join/#asterisk voxtop (n=voxtop@210.9.120.138) |
| 06:15.40 | HaMYaI | JunK-Y: isn't there a direct command to list it out? |
| 06:15.49 | *** part/#asterisk voxtop (n=voxtop@210.9.120.138) |
| 06:15.53 | *** join/#asterisk voxtop (n=voxtop@210.9.120.138) |
| 06:17.38 | gmustafa | ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ |
| 06:22.47 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
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| 06:38.02 | russellb | http://www.voip-news.com/feature/voip-top-50-111306/ |
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| 07:41.01 | Greyh0und | Anyone know how to get dtmf to work with a Siemens Gigaset SL1 and a PAP2T? |
| 07:44.25 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 07:47.33 | uwe | hello, ive installed asterisk from source on a debian sid system, the strange thing is that the ownership of some of /var/run subdirectories changes to asterisk ... any idea why this happens ? |
| 07:47.48 | uwe | sorry, on a debian etch system, not sid |
| 07:48.06 | *** join/#asterisk oej (n=oej@213.115.215.55) |
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| 08:16.17 | voxtop | are the kernel headers needed to compile asterisk |
| 08:16.19 | voxtop | ? |
| 08:18.20 | Xen^ | i believe no |
| 08:18.25 | Xen^ | but for zaptel you need them :) |
| 08:18.27 | hads | Zaptel, yes. Asterisk, no. |
| 08:18.48 | Xen^ | hads : i won :p |
| 08:18.55 | hads | :) |
| 08:19.02 | voxtop | thanks |
| 08:19.26 | Xen^ | Welcome |
| 08:19.27 | hads | Xen^: Ah but were you using both hands? ;) |
| 08:19.45 | Xen^ | lol i am typing from single hand ;) |
| 08:19.50 | Xen^ | right hand only |
| 08:19.58 | hads | Damn, me too :) |
| 08:20.17 | Xen^ | ok |
| 08:20.34 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
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| 08:23.28 | voxtop | do you need your kernel source to compile zaptel, or just the headers? |
| 08:24.01 | Xen^ | well you need kernel source as well as headers |
| 08:24.02 | Xen^ | :) |
| 08:24.16 | Xen^ | i believe headers will be installed if you install glibc using rpm |
| 08:24.24 | voxtop | Xen^: wouldn't the source come with the headers? |
| 08:25.12 | hads | Just the headers will do. |
| 08:25.26 | Xen^ | yeah it will come with header but when you compile any software it will look into /usr/include /usr/local/include |
| 08:25.29 | *** join/#asterisk jserve (n=mail@vpngw.de.callforcenetwork.eu) |
| 08:25.33 | jserve | Good morning |
| 08:26.07 | Xen^ | good evening |
| 08:26.07 | Xen^ | :) |
| 08:28.30 | *** join/#asterisk DerPraktikant (n=Tgu@p54B4D6B5.dip.t-dialin.net) |
| 08:28.35 | DerPraktikant | hi |
| 08:29.50 | DerPraktikant | i got a problem with the compiling of asterisk " cp: cannot stat `.cleancount': No such file or directory |
| 08:29.50 | DerPraktikant | make: *** [cleantest] Error 1" somebody knows what i am doing wrong? |
| 08:30.56 | Xen^ | what is Os |
| 08:30.57 | DerPraktikant | somebody there? |
| 08:31.03 | DerPraktikant | suse 10.1 |
| 08:31.53 | Xen^ | hummm |
| 08:32.03 | Supaplex | alsaplayer plays this 8kbit 16bit mono wav atleast twice as fast as it should. (maybe it's going stereo?). what else can I use to test sounds on the builtin sound card? |
| 08:33.09 | DerPraktikant | i compiled , like in the manual said , the lipri and zaptel |
| 08:33.23 | DerPraktikant | no error |
| 08:33.38 | Xen^ | from where you have downloaded asterisk source ? |
| 08:33.39 | DerPraktikant | but then by compiling the asterisk comes this error |
| 08:33.52 | DerPraktikant | Digium |
| 08:34.43 | Xen^ | if you are trying to compile zaptel on SUSE 10.1 it cant work |
| 08:34.53 | Xen^ | Novell decided that SUSE users shouldn't build their own kernel mods |
| 08:35.09 | Xen^ | you need to fix the OS first back to normal before you can build kernel mods on SUSE 10.1 |
| 08:37.24 | *** join/#asterisk DerPraktikant (n=Tgu@p54B4D6B5.dip.t-dialin.net) |
| 08:37.40 | DerPraktikant | back ~ sorry i got strg+s and the porgram goes off |
| 08:37.55 | L|NUX | DerPraktikant : if you are trying to compile zaptel on SUSE 10.1 it cant work |
| 08:38.05 | L|NUX | DerPraktikant : Novell decided that SUSE users shouldn't build their own kernel mods |
| 08:38.18 | L|NUX | DerPraktikant : you need to fix the OS first back to normal before you can build kernel mods on SUSE 10.1 |
| 08:38.36 | L|NUX | DerPraktikant : there was some article on an opensuse or novell forum that explains what to do to make it build kernel mods again |
| 08:39.32 | DerPraktikant | thank u |
| 08:39.39 | DerPraktikant | i will get another kernel |
| 08:40.15 | Supaplex | any suggestions besides alsaplayer to play 8khz 16bit mono wavs? |
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| 08:41.35 | L|NUX | DerPraktikant : i suggest you to use another distro |
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| 08:47.03 | tengulre | hi,all |
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| 08:59.01 | DerPraktikant | no way , my advisor dont want to change ^^ |
| 08:59.13 | DerPraktikant | u know chefs and so on -.- |
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| 09:08.46 | voxtop | under "Just A Quick Note" at http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+Debian it says to run "make oldconfig; make dep". do they mean from /usr/src/linux/ or from /usr/src/zaptel-VERSION/ ? |
| 09:09.47 | voxtop | n/m =P |
| 09:15.04 | tengulre | anybody can help me buy a asterisk PSTN card from USA? |
| 09:15.51 | hads | Why? What do you need help with? |
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| 09:34.11 | voxtop | after i add "opermode=AUSTRALIA" to /etc/zaptel.conf and reboot, is there any way to tell whether this setting was actually accepted/is being used? |
| 09:34.42 | tzafrir | voxtop, you don't really have to reboot |
| 09:34.47 | frenzy | hi... I've configured a tdm 2400 with channels 17-24.. incoming works very smooth however when I send calls to ZAP/18 it goes to ZAP/17 why is that happening? |
| 09:34.51 | tzafrir | voxtop, you use kernel 2.6, right? |
| 09:35.19 | tzafrir | and you have the wrong file, btw. BRB |
| 09:35.28 | tzafrir | (opermode is a kernel parameter) |
| 09:36.18 | voxtop | tzafrir: 2.6.16 |
| 09:37.14 | voxtop | tzafrir: wrong file? several docs/tuts i've read say to add this to /etc/modprobe.d/zaptel : install wcfxo /sbin/modprobe --ignore-install wcfxo opermode=AUSTRALIA && /sbin/ztcfg |
| 09:38.04 | FuriousGeorge | anyone else find themselves competing with 6 line phones |
| 09:38.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:38.25 | voxtop | FuriousGeorge: shwa? |
| 09:38.50 | FuriousGeorge | voxtop: wha? |
| 09:39.32 | voxtop | FuriousGeorge: how can a [6-line] phone compete with a person? |
| 09:41.11 | frenzy | could someone please help... sending calls to ZAP/18 going to ZAP/17 |
| 09:42.38 | FuriousGeorge | voxtop: no i meant people ask me for a price on a "phone system" for their startup small businesses but i feel like im always talking them out of just getting a 6 line analog "pseudo-pbx" for 400 bucks |
| 09:42.43 | FTexcom | frenzy outbound routes? |
| 09:43.59 | frenzy | yap |
| 09:45.00 | voxtop | FuriousGeorge: ah, gotcha |
| 09:45.12 | FTexcom | frenzy what are you outbound routes? |
| 09:46.30 | frenzy | i'm currently testing.. so basically I've disabled ZAP/17 and only using the ZAP/18 trunk |
| 09:46.51 | tzafrir | voxtop, wrong file indeed. |
| 09:46.56 | FTexcom | frenzy can you please show me what your CLI says? |
| 09:47.02 | tzafrir | voxtop, in modprobe.conf, set: |
| 09:47.26 | tzafrir | wcfxo does not support the opermode AUSTRALIA |
| 09:47.36 | tzafrir | only FCC and one other |
| 09:47.44 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:47.49 | backblue | morning |
| 09:47.50 | tzafrir | (the X100P's chip is very limited) |
| 09:48.03 | tzafrir | voxtop, and to set it, simply use: |
| 09:48.23 | tzafrir | options wctdm opermode=AUSTRALIA |
| 09:48.37 | tzafrir | much cleaner, instead of the "install" monsters people use |
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| 09:49.06 | frenzy | could this have something do with groups? |
| 09:49.37 | FTexcom | frenzy yes, it could be...check your zapata-auto.conf |
| 09:49.56 | Inverted | anyone have any idea why I am getting Failed to execute '/var/lib/asterisk/agi-bin/test.agi': No such file or directory |
| 09:49.56 | Inverted | -rwxr-x--- 1 asterisk asterisk 30661 Nov 14 03:16 /var/lib/asterisk/agi-bin/test.agi |
| 09:50.46 | backblue | Inverted: chmod o+a /var/lib/asterisk/agi-bin/test.agi |
| 09:50.49 | backblue | ups |
| 09:50.51 | voxtop | tzafrir: /etc/modprobe.conf doesn't exist on my box |
| 09:50.57 | backblue | <PROTECTED> |
| 09:51.06 | tzafrir | voxtop, which distro do you use? |
| 09:51.14 | voxtop | tzafrir: Debian 3.1 |
| 09:51.31 | tzafrir | voxtop, even better: use a seaprate /etc/modprobe.d/zaptel |
| 09:51.43 | voxtop | tzafrir: yeah, that's what i'm editing atm |
| 09:52.05 | hads | Or use /etc/modprobe.d/00-zaptel then it won't get overwritten. |
| 09:52.25 | tzafrir | And get the init.d script from the recent zaptel debs. It actually works |
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| 09:52.29 | Inverted | backblue: it just failed and I changed it to -rwxr-x--x 1 asterisk asterisk 30661 Nov 14 03:16 test.agi |
| 09:52.32 | voxtop | tzafrir: my current /etc/modprobe.d/zaptel : http://rafb.net/paste/results/MJdf0218.html |
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| 09:52.59 | voxtop | tzafrir: i've just compiled the latest stable zaptel source (1.2.11) |
| 09:53.35 | backblue | Inverted: what was the error? do o+r too |
| 09:55.04 | Inverted | backblue: Failed to execute '/var/lib/asterisk/agi-bin/test.agi': No such file or directory |
| 09:55.04 | Inverted | -rwxrwxrwx 1 asterisk asterisk 30661 Nov 14 03:16 test.agi |
| 09:55.40 | backblue | Inverted: cat /var/lib/asterisk/agi-bin/test.agi | head -1 |
| 09:56.14 | voxtop | tzafrir: i just read through modprobe.conf's manpage. so i should just add "options wctdm opermode=AUSTRALIA" to the top of /etc/modprobe.d/zaptel ? |
| 09:56.34 | tzafrir | yes. |
| 09:56.42 | Inverted | backblue: #!/usr/bin/perl |
| 09:56.51 | tzafrir | And do yourself a favour to remove the "install" line of wctdm |
| 09:57.43 | voxtop | tzafrir: why should i remove the "install" line of wctdm? |
| 09:58.02 | tzafrir | voxtop, try http://bugs.digium.com/view.php?id=7613 instead |
| 09:58.55 | voxtop | tzafrir: thanks, reading |
| 09:58.58 | tzafrir | because it is uselessly confusing: of the module loaded successfully but zaptel.conf had an error, reporting this as an error from modprobe causes strange results |
| 09:59.43 | voxtop | tzafrir: after i've reinserted wctdm, how can i confirm that opermode=AUSTRALIA is set? |
| 10:00.01 | tzafrir | what kernel do you have? |
| 10:00.05 | Inverted | backblue: i just found the prob |
| 10:00.07 | Inverted | agi-bin # /var/lib/asterisk/agi-bin/webchartmd.agi |
| 10:00.07 | Inverted | -bash: /var/lib/asterisk/agi-bin/webchartmd.agi: /usr/bin/perl^M: bad interpreter: No such file or directory |
| 10:00.37 | tzafrir | Inverted, don't use dos text files |
| 10:00.50 | tzafrir | Inverted, what editor do you use? |
| 10:01.06 | voxtop | tzafrir: 2.6.16-2-686-smp |
| 10:01.24 | Inverted | tzafrir: i accidentally overwrote a file today and spent the whole day sifting thru a dd backup of the partition recovering it |
| 10:01.28 | tzafrir | voxtop, cat /sys/modules/wctdm/parameters/opermode |
| 10:01.40 | tzafrir | also check /sys/modules/wctdm/version |
| 10:02.32 | voxtop | tzafrir: awesome possum, thanks. do i need to do anything about/with _opermode? |
| 10:02.47 | tzafrir | voxtop, no. |
| 10:02.53 | voxtop | k |
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| 10:03.27 | voxtop | tzafrir: so if i remove the "install" line for wctdm from /etc/modprobe.d/zaptel , wctdm will still be loaded into the kernel? |
| 10:04.01 | tzafrir | sure. When you run 'modprobe wctdm', or at boot, by hotplug/udev |
| 10:04.09 | voxtop | thanks again |
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| 10:05.22 | mattfletcher | has anyone used ubuntu's apt-get flavour of asterisk with a unistim (nortel) device? if so, what tricks does one need to pull? |
| 10:06.54 | voxtop | tzafrir: after adding "options wctdm opermode=AUSTRALIA" to the top of /etc/modprobe.d/zaptel , do i need to remove and insert wctdm to get the opermode= effect? i tried just 'sudo modprobe wctdm' and checked /sys/..../opermode but it's still set to FCC |
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| 10:10.26 | agarguest | hi |
| 10:10.36 | agarguest | what's up? who's up? |
| 10:11.50 | voxtop | tzafrir: n/m |
| 10:12.02 | tzafrir | busy |
| 10:14.12 | agarguest | i'm looking for bulk termination in canada any suggestions? |
| 10:14.18 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
| 10:16.51 | dlynes_laptop | agarguest: you could try www.five9snetwork.com; they've got servers at peer one in toronto and vancouver |
| 10:17.22 | dlynes_laptop | agarguest: their standard a-z wholesale route pricing is on their website |
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| 10:24.49 | voxtop | if a kernel module is loaded and in use, is it possible to reload it with different options? IE: reload wctdm with a different "opermode=". or is a reboot required? |
| 10:27.52 | tzafrir | voxtop, you need to unload that specific module, as this specific parameter is used at init time |
| 10:28.01 | voxtop | thanks tzafrir |
| 10:28.19 | tzafrir | but generally you can also write parameters through /sys/module/XXX/parameteters |
| 10:29.51 | voxtop | i added "options wctdm opermode=AUSTRALIA" and rebooted. however, asterisk fails to start: http://rafb.net/paste/results/mOWpx017.html |
| 10:30.11 | voxtop | i'm not sure how to debug this. any suggestions? |
| 10:33.53 | tzafrir | is there /dev/zap/1 ? |
| 10:33.55 | *** join/#asterisk saftsack (n=oliver@p54A7FDD9.dip.t-dialin.net) |
| 10:35.08 | voxtop | tzafrir: yep: http://rafb.net/paste/results/2LXtfO81.html |
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| 10:50.27 | voxtop | tzafrir: any idea why opermode=AUSTRALIA would cause this problem? |
| 10:51.53 | tzafrir | what exactly do you have on /etc/modprobe.d/zaptel ? |
| 10:52.07 | tzafrir | hmm... no. The module was loaded... |
| 10:52.21 | tzafrir | Just make sure that the zaptel init.d script runs ztcfg |
| 10:52.32 | saftsack | what telephones have the better voice quality? the polycoms or the ciscos? |
| 10:53.24 | voxtop | tzafrir: when you said that debian's zaptel-source package has a better init-script for zaptel, were you referring to the stable, testing or unstable version? |
| 10:53.41 | tzafrir | testing/unstable |
| 10:53.53 | tzafrir | The stable package is rather broken |
| 10:54.09 | tzafrir | Won't even build properly |
| 10:54.23 | tzafrir | (zaptel-source, I mean) |
| 10:54.43 | voxtop | tzafrir: this is what's in /etc/modprobe.d/zaptel : http://rafb.net/paste/results/b7U3Iy53.html |
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| 10:55.51 | tzafrir | voxtop, what do you see in /proc/zaptel/* ? |
| 10:56.39 | voxtop | tzafrir: i noticed that there was a line with "wctdm" in /etc/modules . i commented it out, rebooted, and asterisk successfully started. |
| 10:56.44 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 10:57.30 | tzafrir | voxtop, strange. /etc/modules is a list of modules to load very early on |
| 10:58.04 | voxtop | tzafrir: when i saw "wctdm" in /etc/modules , i figured i should comment it out in case it didn't know about opermode=AUSTRALIA yet |
| 10:58.10 | tzafrir | Having wctdm explicitly there is generally a good idea |
| 10:58.23 | voxtop | hrm |
| 10:58.27 | Chris-NB | hi |
| 10:58.43 | Chris-NB | someone using e164.arpa and e164.org for enum lookups? |
| 10:59.14 | Chris-NB | is there a risk of high delays because of these lookups? |
| 11:00.23 | backblue | Chris-NB: you should use your own e164, with cache. |
| 11:00.32 | backblue | your own dns with e164 i mean |
| 11:00.48 | Chris-NB | backblue, how would I do that? |
| 11:00.58 | backblue | setup a dns server |
| 11:01.07 | Chris-NB | ok, so only for caching |
| 11:01.11 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:01.25 | backblue | y |
| 11:01.32 | Chris-NB | using e164.arpa and e164.org, but cache on (a lokal) dns server |
| 11:01.34 | voxtop | tzafrir: well, it looks like setting opermode=AUSTRALIA has helped, as calls on my FXOs are clearer now |
| 11:04.13 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss) |
| 11:04.15 | Madkiss | hi all |
| 11:04.37 | Madkiss | in extensions.conf, can I configure two events with the same priority? |
| 11:07.51 | voxtop | what is zttest for? |
| 11:10.17 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 11:10.35 | saftsack | does anybody know something about the voice quality of the handytone ata's? |
| 11:11.07 | voxtop | saftsack: you mean Grandstream's ATAs? |
| 11:11.13 | monsted | seems quite good on the 488 one i have |
| 11:11.26 | saftsack | yes i mean the grandstream ata |
| 11:11.33 | monsted | the FXO port is apparently close to getting to work, though |
| 11:11.37 | monsted | err |
| 11:11.44 | monsted | close to impossible to get to work |
| 11:11.54 | voxtop | saftsack: i find the audio quality on my 286, 486, and 496 quite good |
| 11:12.11 | saftsack | monsted, why that? |
| 11:12.19 | saftsack | voxtop, do you have an comparable device? |
| 11:12.43 | voxtop | saftsack: however, i find them to be really annoying with registrations. after a while (hours or days), my Grandstream ATAs can't make or receive calls. i have to reboot them |
| 11:12.50 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 11:13.07 | voxtop | saftsack: i've >10 sip ATAs and IP phones |
| 11:13.11 | voxtop | oops |
| 11:13.12 | monsted | saftsack: dunno, it doesn't seem to speak SIP properly and doesn't dial as it's supposed to |
| 11:13.16 | voxtop | saftsack: i've tested >10 sip ATAs and IP phones |
| 11:13.38 | saftsack | voxtop, who paid those tons of atas and phones? |
| 11:13.43 | monsted | voxtop: have you tested AudioCodes boxes? |
| 11:13.51 | voxtop | saftsack: i'd go with Linksys' ATAs rather than Grandstream's |
| 11:13.58 | voxtop | saftsack: the company i work for |
| 11:14.01 | voxtop | monsted: nope |
| 11:14.17 | saftsack | voxtop, because of the voice quality? |
| 11:14.26 | saftsack | voxtop, what was the best sounding ip phone? |
| 11:15.20 | mosty | the snom320 is the clearest sounding phone/ata i've used |
| 11:15.38 | monsted | the ciscos have very nice sound quality |
| 11:15.49 | voxtop | saftsack: i'd go with Linksys ATAs rather than Grandstreams because the Grandstreams have registration problems (in my experience), fugly GUIs, have to be rebooted to use even the smallest settings change, etc |
| 11:16.03 | saftsack | voxtop, did you test a polycom or a cisco phone? |
| 11:16.45 | voxtop | saftsack: i tested a Polycom IP301, but tossed it aside cuz it takes way too long to boot up, has very few features, and is expensive |
| 11:17.08 | saftsack | whats about the voice quality of the polycom? |
| 11:17.19 | voxtop | saftsack: i can't remember what the audio quality of the Polycom IP301 was like, but Snoms are A+ |
| 11:18.02 | saftsack | i heard that the polycoms audio quality is quite better than the quality of the snoms |
| 11:18.30 | saftsack | voxtop, did you test a cisco phone too? |
| 11:18.49 | voxtop | saftsack: nope, no Cisco phones. too expensive and restrictive |
| 11:19.03 | saftsack | what do you mean with restrictive? |
| 11:19.25 | monsted | licensing is a pain |
| 11:19.34 | voxtop | monsted: exactly |
| 11:19.46 | saftsack | licensing of what? its a telephone, or? ^^ |
| 11:19.50 | voxtop | saftsack: you don't just buy a Cisco IP phone. you have to buy a software license for it too |
| 11:19.51 | monsted | but that's how cisco hardware is sold |
| 11:20.12 | monsted | you buy a phone and a license for either Skinny or SIP firmware |
| 11:20.35 | monsted | and without a CCO account and a valid service contract, you can't download firmware for it |
| 11:20.54 | voxtop | saftsack: i've read and heard fantastic things about Cisco's IP phones |
| 11:20.59 | voxtop | saftsack: i just haven't used one before |
| 11:21.15 | saftsack | hmm so what is the best phone to buy? a snom 320? |
| 11:21.38 | monsted | if you're a cisco shop and have all those things in place already, cisco phones are great - if it'd be your only cisco items, it's probably too much of a hassle |
| 11:21.50 | saftsack | the snom telephones are crap in my opinion. not because of the sound quality or the software but the quality of the phone itself is crap imho |
| 11:22.30 | voxtop | saftsack: if you think that about Snom IP phones, don't even consider Grandstream products |
| 11:22.56 | voxtop | saftsack: i have no idea why you think the quality of Snom phones is crap though |
| 11:22.59 | saftsack | voxtop, humm i had another experience. my grandstream here has a heavy handset and a good quality |
| 11:23.33 | monsted | i wonder if these nortel phones can be convinced to work with anything other than the nortel pbx'es |
| 11:23.41 | saftsack | but the snom phone has an extremely easy handset and the button which indicates a hangup was broken after 2 weeks |
| 11:23.42 | RoyK | does anyone know if there is native psql support somewhere for 1.2? |
| 11:25.23 | voxtop | saftsack: have you tried only 1 snom? |
| 11:25.48 | saftsack | yes its a telephone from elmeg but its a snom just labeled |
| 11:25.53 | voxtop | saftsack: i have a Snom 300, 320 and 360 and they all perform great. i also feel the build quality is superior to Grandstream GXP-2000s |
| 11:26.02 | voxtop | saftsack: elmeg? |
| 11:26.10 | saftsack | but they all have a low weight handset, right? |
| 11:26.39 | saftsack | http://www.idealo.de/preisvergleich/OffersOfProduct/348949_-ip-290-elmeg.html |
| 11:26.40 | saftsack | this one |
| 11:26.54 | voxtop | saftsack: i'd say the Snom and GXP-2000 handsets weigh the same |
| 11:27.02 | saftsack | i have a budgetel here |
| 11:27.12 | saftsack | the handset from the budgetel is quite nice |
| 11:27.19 | saftsack | budgetone i mean |
| 11:27.39 | voxtop | saftsack: that elmeg thing is strange; i've never seen or heard of it before |
| 11:27.44 | *** join/#asterisk mega (i=mega@gateway/tor/x-55434ba3f3a97c64) |
| 11:27.50 | saftsack | yes but it is a snom 320 |
| 11:28.10 | saftsack | http://www.funkwerk-ec.com/portal/galerien/pics/elmeg_products/IP290/elmeg_IP290_Product_132_RGB_1mp-midRes.jpg |
| 11:28.12 | mosty | royk: there is for cdr, but i think realtime requires odbc for postgresql support |
| 11:29.12 | *** join/#asterisk CMike (i=daemon@c-614071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 11:32.39 | Madkiss | I have two local clients connected to my asterisk, 2000 and 3000. When calling 3000 from the client with 2000, that should just work, shouldn't it? |
| 11:33.18 | mosty | madkiss: depends on the dialplan, and where those sip peers start in the dialplan |
| 11:33.20 | InfraRed | depending if you fixed your extensions.conf |
| 11:34.09 | Madkiss | exten => 2000,1,Dial(SIP/2000,20,tTr) |
| 11:34.18 | Madkiss | Is in my Dial-Plan for [2000] |
| 11:35.27 | mosty | madkiss: what context do calls from 3000 start in? |
| 11:35.43 | Madkiss | uhm, that information is in sip.conf, isn't it? |
| 11:36.12 | mosty | yes |
| 11:37.22 | Madkiss | they all start in default |
| 11:38.44 | mosty | ok, well step through the dialplan and see how the calls move around, and see if there is a path to dial'ing the extension you're after |
| 11:39.19 | *** join/#asterisk roxy_ (n=roxy_@machine219.bibu2.u-cergy.fr) |
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| 11:39.53 | *** part/#asterisk bhrobinson (n=brobinso@northtx-cust-rtr.telwestonline.com) |
| 11:40.19 | roxy_ | If I get a TDM400P with 3FXS and 1 FXO, does that mean that caller will be able to call outside just one at a time ? |
| 11:40.53 | roxy_ | Or that all 3 can use the FXO line but the quality will not be as good ? |
| 11:41.46 | mosty | roxy: one analogue line means one call at a time |
| 11:42.15 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 11:42.21 | FreezeS | hey guys |
| 11:42.34 | FreezeS | is there a way to increase the sensitivity of a PRA card ? |
| 11:42.57 | FreezeS | mine says: SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 11:43.47 | Madkiss | well, I think that worked. |
| 11:43.48 | Madkiss | thanks. |
| 11:46.03 | RoyK | mosty: ok |
| 11:46.06 | RoyK | mosty: odbc is broken in 1.2, it seems |
| 11:46.17 | roxy_ | mosty: thanks |
| 11:46.20 | mosty | royk: odbc works for me |
| 11:46.32 | RoyK | http://bugs.digium.com/view.php?id=8359 |
| 11:49.13 | mosty | ahh, well i'm not using 1.2.13 yet |
| 11:52.13 | litage | does zap echo cancellation have any effect on sip calls? |
| 11:52.29 | FreezeS | I've got a problem with desynchronisations. They happen once per second (almost) |
| 11:52.42 | mosty | litage: maybe if you have a zap - sip bridged call |
| 11:53.12 | *** join/#asterisk Jenna (i=CherryRe@gateway/tor/x-809ef818a3a8df10) |
| 11:59.45 | Jenna | hi ! anyone know if trixbox is video conferencing capable or needs addon to make it go ? |
| 12:06.32 | CMike | I'm playing with a couple of Grandstream videophones.. hm .. |
| 12:06.43 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 12:06.54 | CMike | I guess I need h264... or is there a firmware for the phone that supports h.263 ? |
| 12:08.01 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:08.29 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-245.range86-136.btcentralplus.com) |
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| 12:09.16 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:09.16 | agarguest | hi |
| 12:09.58 | agarguest | CMike: what do you think of the grandstream videophones? |
| 12:10.12 | CMike | just unpacked them basiclly |
| 12:10.21 | agarguest | how is the look, quality? |
| 12:10.31 | agarguest | any parts make of cardboard :) |
| 12:10.41 | CMike | they look nice though.. I've one vizufon.. and that I don't like |
| 12:10.55 | agarguest | cool |
| 12:12.16 | FTexcom | I like the spa941 |
| 12:12.21 | FTexcom | is the one I use |
| 12:12.38 | agarguest | i need some more poe phones |
| 12:12.57 | agarguest | i like the spa942 but i have never touched one |
| 12:12.59 | FTexcom | thomson has a couple of poe phones |
| 12:13.37 | agarguest | i have some grandsteam gxp2000 and bt101 phones |
| 12:13.44 | agarguest | they work great but feel cheap |
| 12:14.05 | monsted | agarguest: ciscos and some poe injectors? |
| 12:14.17 | FTexcom | I care most about realibilty than "looks" |
| 12:14.23 | heison | ~seen JerJer |
| 12:14.45 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 28d 19h 41m 41s ago, saying: 'apt-file search lspci'. |
| 12:14.45 | agarguest | i have a 24 port poe switch at home |
| 12:14.45 | *** join/#asterisk Crumbles (i=Crumbles@24-178-73-251.dhcp.gwnt.ga.charter.com) |
| 12:14.46 | agarguest | ciscos are too much $$$$ for home use |
| 12:14.50 | agarguest | i love the bt101 only wish they had a poe model |
| 12:15.03 | monsted | agarguest: well, i got a good deal on my 7960s :) |
| 12:15.29 | agarguest | the gxp2000 is too big and buttons are clunky and display logic software isn't very nice eventhough its a nice big lcd |
| 12:15.38 | monsted | "You don't need those test phones any more, right? Yoink!" |
| 12:15.48 | agarguest | how much did you pay? |
| 12:16.13 | monsted | 0 :) |
| 12:16.33 | agarguest | so you are the guy that stole all of my clients phones? |
| 12:16.35 | agarguest | :) |
| 12:17.53 | agarguest | if anyone can tell me how useful is the display on the spa942, it would be helpful. |
| 12:19.24 | monsted | "Cisco Unified IP Phone 7971G-GE, 7961G-GE, 7941G-GE: A suite of voice over IP phones delivers the latest technology and advancements in Gigabit Ethernet VoIP." |
| 12:19.33 | monsted | wait, when do we need GigE for VoIP? :) |
| 12:19.45 | agarguest | the mother-in-law |
| 12:20.04 | agarguest | hey , that was funny |
| 12:20.12 | monsted | ah, right |
| 12:20.20 | *** join/#asterisk xnon_ (i=xnon@200.8.30.3) |
| 12:21.35 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 12:21.42 | monsted | i should get some toys from them if we buy the 35000 phones we're planning.. |
| 12:22.47 | agarguest | The Grandstream Budgetone IP Phone is an award-winning next generation IP network telephone based on industry open standards. Built upon innovative technology, Grandstream IP Phone features market leading superb sound quality and rich functionalities at ultra-affordable price. |
| 12:23.10 | agarguest | $39.00 |
| 12:23.30 | agarguest | 35,000 phones? |
| 12:24.39 | agarguest | i gotta reboot bye bye all |
| 12:26.27 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
| 12:31.16 | cpm | hrmm, are those grandstream desksets any good? |
| 12:31.19 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 12:31.28 | AsteriskAlbania | does asterisk suport FAX ? |
| 12:31.42 | AsteriskAlbania | I dont see T38 codec with show codecs ? |
| 12:32.01 | monsted | any idea what to do about a cisco 7960 that attempts to register to asterisk, gets a "401 Unauthorized" but never retries a proper authenticated register? |
| 12:32.23 | cpm | Results 1 - 50 of about 2,110,000 for Asterisk Fax. ( |
| 12:32.48 | FTexcom | cpm use iaxmodem, it's the only reliable thing I have managed to found |
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| 12:52.22 | cpm | FTexcom, thanks for the clue |
| 12:57.13 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 12:57.56 | hieunm_vips | hi all, Could I ask you some question ? |
| 12:58.22 | hieunm_vips | Is there a built-in support for FoIP (T38) in Asterisk ? |
| 12:59.01 | hieunm_vips | If not, how could I use FoIP with Asterisk ? |
| 12:59.03 | *** join/#asterisk xnon_ (i=xnon@200.8.30.3) |
| 12:59.26 | *** join/#asterisk HumpBack (n=sdcdcssc@gentoo/developer/HumpBack) |
| 12:59.30 | HumpBack | Hello |
| 13:00.10 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
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| 13:03.32 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 13:03.47 | puzzled | morning |
| 13:03.49 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 13:04.09 | sivana | if I'm connecting one * box to another * using a TE100P, what signal do I want, fxoks or e&m? |
| 13:04.48 | hieunm_vips | sivana, I think you should you fxsks |
| 13:06.48 | puzzled | isn't that one of the wct drivers? TE100P is for a single PRI right? |
| 13:06.52 | sivana | yea |
| 13:06.56 | sivana | I think it's e&m I want |
| 13:07.10 | puzzled | afaik that's for analog |
| 13:07.12 | sivana | to create a T1 between the two |
| 13:07.18 | mattfletcher | When an fax is received via my TDM400P, I get the error "channel.c:2326 set_format: Unable to find a codec translation path from unknown to unknown". What can I do to fix this? |
| 13:07.22 | *** join/#asterisk bunjiboys (n=bunjiboy@dingbot.nordit.dk) |
| 13:09.45 | *** join/#asterisk t_shravan (n=shravan@59.93.118.105) |
| 13:10.09 | t_shravan | does any one know how do i work with asterisk and dialogic |
| 13:10.21 | mattfletcher | Googling for that particular error gave 0 results, and smaller parts of it were unhelpful. |
| 13:12.16 | *** join/#asterisk synthetiq (n=syntheti@vonmail.vonworldwide.com) |
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| 13:14.09 | HumpBack | I get strange messages when loading the wctdm driver on a TDM400P Card: http://pastebin.com/824079 |
| 13:14.36 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 13:16.04 | bunjiboys | Ive got a problem i cant seem to figure out, and i can see from various Google and list searches that i havent been the only one having this, but havent been able to find a solution yet. Ive got 3 * boxes, one as the primary "hub" on which the two others registers. Box A (HUB) and Box B are located in the same LAN, and Box C is in a remote location (No NAT). When calls gets sent from Box A (hub) to Box C (remote) i sometimes get a problem with the |
| 13:17.41 | mattfletcher | Right, I'm getting there with my fax codec problem. I found this forum post http://forums.digium.com/viewtopic.php?t=3776 which suggests I need to turn codecs on that I do not already have on. It suggests ulaw and alaw. Where would I turn these on for a TDM400P-based system? |
| 13:19.09 | JT | t_shravan: you require a proprietry driver that will be released some time soon apparently for asterisk business edition |
| 13:19.13 | JT | which costs money |
| 13:21.52 | *** join/#asterisk jm|work (n=jamiem@dilbert.jamiem.com) |
| 13:23.03 | *** join/#asterisk knathraak (n=knath@pool-71-252-87-31.washdc.east.verizon.net) |
| 13:23.44 | knathraak | Hello--I'm having a heck of a time getting the Dial() app to work the way I need it to--maybe someone could help |
| 13:24.11 | knathraak | I'm trying to use the A(x) option to play a message to the callee before being bridged |
| 13:24.45 | knathraak | i'm using to zap fxo channels, one incoming, and one outgoing |
| 13:25.49 | knathraak | running asterisk with verbose output (asterisk -d -vvvv -f) indicates that it thinks the number dialed as answered, when it hasn't |
| 13:26.26 | knathraak | consequently it goes ahead and plays the announcement, and when the callee does answer he hears nothing and the calls are bridged |
| 13:26.55 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 13:27.29 | knathraak | i've tried callprogress=yes in zapata.conf, but in this case it detects ringing but never detects the answer |
| 13:27.44 | *** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro) |
| 13:27.57 | knathraak | any ideas, or a better/different way to accomplish the same thign? |
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| 13:34.12 | monsted | bunjiboys: your questions make more sense if they don't get cut off in the midle ;) |
| 13:34.30 | monsted | banjoboys: "i sometimes get a problem with the" |
| 13:35.01 | bunjiboys | argh.. irssi usually cuts it out itself :) |
| 13:36.27 | JT | not really good practice to send run on lines anyway |
| 13:36.57 | monsted | i know the banjoboy - he's a n00b anyway ;) |
| 13:36.58 | JT | it's like good punctuation and paragraphing, eventually peeople stop reading and paying attention if it's all a massive hard to follow paragraph |
| 13:37.20 | bunjiboys | ... i sometimes get a problem with the one-way audio (incoming caller can hear my voice, but i cant hear them). When this happens the console spews out tons of "Received mini frame before first full frame". Ive tried disabling / enabling jitterbuffer and trunking in all possible combinations. Wondered of someone might have the "nugget" of info which can resolve this problem :) |
| 13:37.37 | JT | monsted: who is this "banjoboy"? |
| 13:37.41 | bunjiboys | JT: me |
| 13:37.54 | JT | your nick is bunjiboys |
| 13:38.01 | monsted | close enough :) |
| 13:38.57 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 13:38.58 | *** join/#asterisk temujen (n=temujen@63.78.3.195) |
| 13:39.01 | froguz | Hi |
| 13:39.07 | temujen | Howdy |
| 13:39.35 | temujen | Is it too early to ask questions? |
| 13:39.52 | knathraak | it's always early somewhere :-) |
| 13:39.52 | froguz | how do i do now to compile Asterisk 1.4 on a i586? the Makefile has changed since 1.2.X |
| 13:40.07 | temujen | lol |
| 13:40.14 | InfraRed | froguz: with great difficulty |
| 13:40.17 | *** join/#asterisk Sasch (n=Sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 13:40.23 | Sasch | hi all |
| 13:40.29 | Crescendo | Take off every zig. |
| 13:40.49 | Sasch | can help me with sip wireless phone ?? |
| 13:40.51 | temujen | Howdy |
| 13:41.11 | froguz | InfraRed, i hope you're just joking |
| 13:42.00 | temujen | Has anyone deployed a server with more than one digium card? (Trying to identify what the outer limits are of the hardware, software ...) |
| 13:42.17 | bunjiboys | froguz: you know that from 1.4.x autoconf is used so you have to ./configure it now, and not just make it? |
| 13:42.43 | Sasch | <temujen> I use a tdm400P with ubuntu server 6.10 |
| 13:43.22 | froguz | it seems Makefile now gets the arch type from the system (export PROC), but uname -r shows an i686 CPU, while i'm working on a Via C3 |
| 13:43.57 | froguz | bunjiboys, the problem is that ./configure is recognizing my CPU as i686 |
| 13:44.18 | bunjiboys | ah ok, just werent sure if you just tried to make it as previous versions without a configure |
| 13:46.01 | froguz | i'm guessing i'll have to move to a 858 kernel |
| 13:46.15 | knathraak | temujen: i've got two tdm400s in mine |
| 13:46.42 | froguz | is there another way to fix it? can i force asterisk to install as i586? |
| 13:46.58 | *** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br) |
| 13:48.03 | Crescendo | Is there any known issues with SpeeX not working 80% of the time on transcoded calls from other codecs, and not working at all when calling out using SpeeX to dial out? |
| 13:48.03 | froguz | './configure PROC=i586' ? |
| 13:49.05 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:49.20 | monsted | froguz: isn't a C3 a 686-class CPU? |
| 13:49.22 | dlynes_laptop | froguz: You're using the default kernel for your distribution of linux, and it was compiled for i686, so that it would work on most systems out in the field |
| 13:49.31 | dlynes_laptop | monsted: no |
| 13:49.45 | dlynes_laptop | monsted: it has its own optimizations in the kernel |
| 13:50.14 | dlynes_laptop | monsted: it's a low power cpu generally for embedded applications |
| 13:50.20 | monsted | i know |
| 13:50.25 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:50.36 | monsted | but i believe it implements the feature set of a 686 |
| 13:50.58 | dlynes_laptop | Yeah...same instruction set as a 586 |
| 13:51.22 | monsted | hence, a 686 kernel should work fine |
| 13:51.23 | dlynes_laptop | so almost the full set of a 686 |
| 13:51.37 | dlynes_laptop | yes...686 is designed to work on most chipsets |
| 13:51.51 | dlynes_laptop | I think he's wanting one that'll work optimally on his via though |
| 13:51.56 | monsted | but do as froguz said |
| 13:52.05 | froguz | monsted, kernel work fine, the problem is the 'illegal instruction' each time i try to run asterisk |
| 13:52.05 | monsted | (if you really want to force it to something else) |
| 13:52.08 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 13:52.24 | monsted | err, sorry, got you confused :) |
| 13:52.42 | monsted | but yes, there's an option to set the architecture |
| 13:52.44 | *** join/#asterisk KriS83 (n=kmarcrof@212.202.141.92) |
| 13:53.02 | KriS83 | Hi |
| 13:53.02 | froguz | monsted, wich is...? |
| 13:53.34 | monsted | -march=cpu on gcc is one |
| 13:53.43 | KriS83 | I have a short question. Will Asterisk 1.4.X support multiple sip registrations for a single sip account? |
| 13:54.16 | froguz | i used to just change the PROC parameter in Makefile |
| 13:54.19 | monsted | but if your kernel doesn't match your CPU, you'd be better off fixing that, then compiling asterisk as usual |
| 13:54.21 | KriS83 | e.g. 2 Softphones register in on one sip account? Both ring when the sip extension is called? |
| 13:55.04 | jm|work | KriS83: why not give them both separate extensions? |
| 13:55.20 | jm|work | soft1 <6001> and soft2 <6002> |
| 13:55.25 | *** join/#asterisk SoftIce (n=newbie@vc-196-207-45-253.3g.vodacom.co.za) |
| 13:55.25 | *** join/#asterisk drkludge (n=drkludge@ip68-231-40-97.ph.ph.cox.net) |
| 13:55.38 | SoftIce | hi, I need gettinge errors about comfort noise and advised to turn it off |
| 13:55.42 | SoftIce | how can i do that? |
| 13:55.43 | jm|work | _6XXX,1,Dial(SIP/6001&SIP/6002) |
| 13:56.03 | dlynes_laptop | SoftIce: it would depend on what phone/softphone you're using...it's not an asterisk setting; it's a phone setting |
| 13:56.12 | dlynes_laptop | SoftIce: they're not errors either; they're warnings |
| 13:56.17 | SoftIce | ahh |
| 13:56.20 | KriS83 | jm|work, cos I'd like to be able to login with multiple softphones from different places and still be available at one single extension |
| 13:56.26 | SoftIce | so its not asterisk, its the phones people are using? |
| 13:56.31 | dlynes_laptop | SoftIce: correct |
| 13:56.32 | SoftIce | what happens if they using normall handset? |
| 13:56.42 | KriS83 | e.g. HomePhone rings and I'm at work, so home and work rings |
| 13:56.42 | dlynes_laptop | SoftIce: because asterisk doesn't support it |
| 13:56.42 | froguz | i think i better return to the old good 1.2.X for these machines |
| 13:56.43 | SoftIce | just ordinary telephones and im getting that error |
| 13:57.00 | dlynes_laptop | SoftIce: you mean an analog handset? |
| 13:57.03 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 13:57.05 | SoftIce | dlynes_laptop: I see, but I get it nearlly every second |
| 13:57.14 | SoftIce | as I have high volume coming through |
| 13:57.17 | KriS83 | jm|work, to complicated as I have way to many locations |
| 13:57.22 | SoftIce | dlynes_laptop: yes, analog handset |
| 13:57.31 | dlynes_laptop | SoftIce: then it's a setting in your ATAs (sipura 2000/2002/pap2-na/cisco ata-186/...) |
| 13:57.36 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 13:57.39 | jm|work | why didn't you dedicate you "home number" to a group? :/ |
| 13:57.48 | jm|work | aside from that I don't know your answer |
| 13:57.54 | grEvenX | how does asterisk work with db connections? |
| 13:57.58 | grEvenX | does it use several connections, or just one? |
| 13:58.02 | KriS83 | jm|work, k |
| 13:58.04 | grEvenX | thorugh ODBC and realtime |
| 13:58.35 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:58.49 | KriS83 | jm|work, I guess it would work with SER infront of asterisk.. but I though I read it somewhere it was going to be included in asterisk at some point |
| 13:58.55 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:59.02 | KriS83 | just wanted to know if this point was now :) |
| 13:59.57 | Sasch | i have a problem with sip wireless telephone |
| 14:00.20 | Sasch | if i call 10 (wireless telephone) to my grandstream asterisk send me |
| 14:00.38 | Sasch | Nov 14 14:59:45 NOTICE[3848]: chan_sip.c:2007 auto_congest: Auto-congesting SIP/10-081a4490 |
| 14:01.01 | Sasch | but if i call 13 my grandstream with my wireless telephone all work |
| 14:01.03 | Sasch | why ???? |
| 14:01.14 | drkludge | Do * IVR scripts allow type-ahead or bargein type scripting i.e. the person can enter their response before the voice prompt finishes? |
| 14:01.22 | Sasch | excusme for my english ... I'm italian and I don't speak very well :-P |
| 14:02.06 | shellshark | do you have qualify=yes on peer 10? |
| 14:02.06 | monsted | KriS83: have each of your phones on a seperate extension, make on extension that dials them all and have the homephone number dial the group extension |
| 14:02.32 | KriS83 | monsted, but I need to have it dynamicly. |
| 14:02.35 | Sasch | <shellshark> I look |
| 14:02.39 | froguz | Sasch did you set different sip accounts for your WiFiPhone and Grandstream? |
| 14:02.42 | KriS83 | the above was just an example |
| 14:02.48 | shellshark | Sasch: if you do, try sip show peers in the CLI |
| 14:02.54 | shellshark | Sasch: and see what 10's status is |
| 14:02.55 | Sasch | qualify=1000 |
| 14:02.57 | monsted | KriS83: you don't know the number of phones in the group ahead of time? |
| 14:02.58 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 14:03.04 | KriS83 | I don't want to keep adding softphones to my config cos my colleges add new phones |
| 14:03.11 | KriS83 | No I don't |
| 14:03.14 | shellshark | Sasch: sip show peer 10 |
| 14:03.21 | shellshark | Sasch: see what it's status is |
| 14:03.46 | mattfletcher | Is there anyone here who has incoming faxes via zap working? If so could they share what codecs faxing is using and how that is set up? |
| 14:03.50 | shellshark | Sasch: it is possible that 10 is not registering correctly, while it is still able to make outbound calls |
| 14:04.11 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 14:04.19 | Sasch | ok i restart m,y telephone |
| 14:04.36 | Sasch | <PROTECTED> |
| 14:04.36 | Sasch | <PROTECTED> |
| 14:04.50 | shellshark | ok, now: sip show peer 10 |
| 14:04.54 | Sasch | i call 10 |
| 14:04.57 | shellshark | what's the status? |
| 14:05.04 | Sasch | <PROTECTED> |
| 14:05.05 | Sasch | Nov 14 15:04:10 NOTICE[3922]: chan_sip.c:2007 auto_congest: Auto-congesting SIP/10-081a4760 |
| 14:05.05 | Sasch | <PROTECTED> |
| 14:05.05 | Sasch | <PROTECTED> |
| 14:05.05 | Sasch | <PROTECTED> |
| 14:05.05 | Sasch | <PROTECTED> |
| 14:05.14 | Sasch | pastebin don't work :-P |
| 14:05.22 | shellshark | pastebin.ca does ;) |
| 14:05.34 | Sasch | i try pastebin.com don't work |
| 14:05.36 | shellshark | what does sip show peer 10 say? |
| 14:06.12 | mattfletcher | Does no-one here have incoming faxes working? |
| 14:06.29 | dlynes_laptop | mattfletcher: lots of people dooooo |
| 14:06.30 | Sasch | <mattfletcher> I have |
| 14:06.57 | Sasch | peer 10 is a WIP6000 Samsung |
| 14:07.01 | *** join/#asterisk W9SH (n=Steve@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 14:07.02 | dlynes_laptop | mattfletcher: they're generally using hylafax in combination with iaxmodem though, not rxfax/txfax |
| 14:07.09 | SoftIce | dlynes_laptop: why would I get an error 500 server internal error, ----- circuit-busy |
| 14:07.14 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 14:07.14 | mattfletcher | any chance someone could tell me how i tell it which codecs to use? i am getting an error |
| 14:07.23 | SoftIce | meaning the carrier i'm using is busy? |
| 14:07.29 | coppice | most people actually use rxfax/txfax |
| 14:07.31 | SoftIce | 1 of my routes? |
| 14:07.38 | dlynes_laptop | SoftIce: because the idiots that you're handing your callss off onto are idiots |
| 14:07.48 | Sasch | <shellshark> what do you think about this problem ?? |
| 14:07.53 | dlynes_laptop | coppice: really? |
| 14:08.02 | coppice | yes. |
| 14:08.03 | dlynes_laptop | coppice: I've never gotten those to work past one page |
| 14:08.05 | SoftIce | dlynes_laptop: maybe they can't just handle the number of calls |
| 14:08.13 | SoftIce | I get nearlly a call a second |
| 14:08.20 | SoftIce | so I guess they can't handle the traffic? |
| 14:08.20 | shellshark | Sasch: i'm not going to help you any further until you do what i ask ;) |
| 14:08.22 | coppice | i can send 100 page faxes all day |
| 14:09.06 | dlynes_laptop | coppice: your latest version includes rxfax/txfax again? |
| 14:09.16 | coppice | iaxmodem is just a wrapper around spandsp, so hylafax can talk to it |
| 14:09.24 | dlynes_laptop | coppice: the last version I downloaded didn't have them...only the previous versioon |
| 14:09.26 | Sasch | idon't undertsand ... google don't translete well your word -.-' |
| 14:09.31 | mattfletcher | The error is http://pastebin.ca/248276 |
| 14:09.51 | stephane_ | re |
| 14:10.00 | froguz | Sasch wich word? |
| 14:10.21 | Sasch | what does sip show peer 10 say? |
| 14:10.40 | *** part/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 14:11.01 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 14:11.16 | mattfletcher | Any ideas, clever faxing people :)? |
| 14:11.23 | supjigatr | Anyone have a Zhone Zplex? I have trouble getting the T1 to come up. They indicate OUS. |
| 14:11.39 | ambriento | Sasch, on the CLI>, you have to type: sip show peer 10 |
| 14:11.49 | Sasch | ok |
| 14:11.51 | Sasch | one moment |
| 14:12.20 | pif | does hylafax include a good print-to-fax solution for windows clients? |
| 14:12.20 | froguz | Sasch, do a 'sip show peer 10' and paste it into http://pastebin.ca |
| 14:12.28 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
| 14:12.49 | Sasch | http://pastebin.ca/248278 |
| 14:12.58 | dlynes_laptop | pif: yes |
| 14:13.09 | dlynes_laptop | pif: it makes a great fax print server |
| 14:13.15 | dlynes_laptop | pif: just a bit of a pia to set up |
| 14:13.29 | pif | dlynes_laptop : with no need to embed the fax number in the document? |
| 14:13.52 | shellshark | Sasch: Status : UNKNOWN |
| 14:13.58 | dlynes_laptop | pif: you can tell it to autogenerate your cover pages for you |
| 14:14.07 | shellshark | Sasch: your wireless phone is not registering properly, probably due to a codec mismatch |
| 14:14.24 | pif | dlynes_laptop : at what stage do you enter the fax number? |
| 14:14.26 | shellshark | Sasch: you only are allowing alaw, try allowing ulaw also |
| 14:14.29 | ambriento | shellshark, a codec has nothing to do with registration |
| 14:14.35 | dlynes_laptop | pif: during the print |
| 14:14.39 | pif | cool |
| 14:14.45 | shellshark | ambriento: i've had codec mismatch deny registration plenty of times |
| 14:14.57 | dlynes_laptop | pif: but all that might have changed...I haven't used hylafax in about 6 or 7 years, either |
| 14:14.58 | Sasch | ok |
| 14:15.02 | froguz | Sasch, i think you should re-read the network configuration of your WiFi phone |
| 14:15.05 | pif | thanks for the info |
| 14:15.11 | dlynes_laptop | pif: they might have made the process a little better since then |
| 14:15.19 | *** join/#asterisk ToyMan (n=stuq@74-32-30-166.dsl1.mdl.ny.frontiernet.net) |
| 14:15.53 | ambriento | shellshark, really? I've never seen that before. |
| 14:15.58 | mattfletcher | I guess it can't be done then |
| 14:16.12 | Sasch | <froguz> why ?? |
| 14:17.08 | shellshark | ambriento: yeah, and i'm not talking about codec negotiation when a call is placed, i'm talking about when a device tries to just register |
| 14:17.43 | Sasch | now CLI return Status : OK (113 ms) |
| 14:17.52 | Sasch | but if I call it don't call |
| 14:18.03 | froguz | Sasch, because maybe you didn't setup well the SIP Proxy address |
| 14:18.14 | ambriento | shellshark, that's odd to me |
| 14:18.26 | Sasch | <froguz> in my telephone ?? |
| 14:18.36 | froguz | or the mask, or the gateway if you're behind a router |
| 14:18.41 | temujen | In my reading, the O'Reilly book doesn't recommend more than one digium card per asterisk server, is this still a limitation? |
| 14:18.49 | froguz | sure, in your phone |
| 14:19.03 | Sasch | proxy in my telephone is correct |
| 14:19.16 | Sasch | when i start it asterisk found it |
| 14:20.07 | froguz | temujen, if you're using a good server that can deal well with IRQs, i think you shouldn't worry about it |
| 14:20.48 | Sasch | CtRiX ci sei????????????? |
| 14:20.49 | froguz | Sasch do you read/write spanish? |
| 14:21.03 | temujen | My dialers are currently running 16 T1's; if I bought a powerful enough box can I build a box to support that with asterisk? |
| 14:21.16 | Sasch | <froguz> I'm italian .... |
| 14:22.26 | ambriento | temujen, you should take a look at sangoma's card. They have a new one with 8 E1/T1 ports |
| 14:22.39 | froguz | Sasch, italian is a romantic languaje as the spanish is too, so maybe we can understand each other better in a PV |
| 14:23.08 | ambriento | froguz, are you spanish? |
| 14:23.32 | froguz | ambriento, chileno ;) |
| 14:24.41 | Sasch | <froguz> ok but i use Virc and i don't know in wich mode is make a PV |
| 14:24.41 | *** join/#asterisk _fenlander (n=fenlande@82.152.81.57) |
| 14:24.42 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 14:25.39 | froguz | temujen, if you're going to try such number of T1's maybe you should try cards with built-in DSP processors or T1 gateways |
| 14:25.43 | dlynes_laptop | temujen: there's even one or two guys in here running four digium quad t1 cards in one machine |
| 14:26.03 | ambriento | que bueno froguz |
| 14:26.04 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:26.34 | froguz | oops! be carefull ambriento, ppl get weird here when we use other than english |
| 14:26.58 | dlynes_laptop | froguz: shenme yi ci |
| 14:27.19 | froguz | hahahahaha... and that's why |
| 14:27.41 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:27.45 | dlynes_laptop | :) |
| 14:27.47 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 14:28.00 | dlynes_laptop | froguz: you don't speak Mandarin? |
| 14:28.04 | dlynes_laptop | froguz: I'm shocked |
| 14:28.37 | ambriento | ni hao |
| 14:28.39 | Sasch | CtRiX ci sei????????????? |
| 14:29.06 | ambriento | Sasch, CtRix isn't italian. I know that |
| 14:29.22 | dlynes_laptop | ambriento: he's in Italy |
| 14:29.28 | coppice | CtRiX is italian, actually |
| 14:29.49 | Sasch | CtRiX I know ..... dai dimmi che ci sei che finalmente parlo italiano :-P |
| 14:30.07 | *** join/#asterisk zx6r (n=appak@ottawa-hs-206-191-39-130.d-ip.magma.ca) |
| 14:30.21 | SplasPood | If I have an asterisk box with multiple clients on it, distinguished by contexts in the dialplan, and I had it setup such that calls from one client to another remained within the asterisk box rather than bouncing out to a peer and back in, how would I split the CDR such that each client (accountcode) gets its own record (one inbound, one outbound) |
| 14:30.23 | ambriento | coppice, only if it's a short for some words |
| 14:30.26 | knathraak | hi, all. got a problem with my digium analog cards |
| 14:30.33 | mattfletcher | Do the codecs listed in sip.conf affect all other channel types? If not, where do I tell a zap channel which codecs it can use? |
| 14:30.47 | coppice | kui dei yung yi dai lei man, use pu tong hua. ho man faan! |
| 14:30.47 | froguz | dlynes_laptop, i have a micronet gateway that tell me 'yi ci' everytime i make a call, hehehehe... is that 'thank you' or something? |
| 14:30.52 | dlynes_laptop | mattfletcher: most zap channels only support ulaw/alaw |
| 14:31.05 | dlynes_laptop | froguz: shenme yi ci == what is it? |
| 14:31.09 | knathraak | when I use Dial() to dial a pstn number over an fxo port it assumes the line is answered rather than waiting for actual answer |
| 14:31.29 | mattfletcher | my error makes even less sense then: "Unable to find a codec translation path from unknown to unknown" |
| 14:31.37 | knathraak | any suggestion? |
| 14:32.10 | *** join/#asterisk Sasch (n=Sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 14:32.31 | knathraak | for what it's worth asterisk is behind our company's avaya pbx, could the avaya be confusing asterisk/zaptel? |
| 14:32.35 | dlynes_laptop | mattfletcher: what's the source channel and what's the target channel? |
| 14:33.22 | dlynes_laptop | knathraak: as soon as you pick up the fxo port, it assumes it's answered...there's no reliable answer signalling on analog ports afaik |
| 14:33.39 | dlynes_laptop | knathraak: if you want that, you'll need to be using isdn or pri |
| 14:33.56 | *** join/#asterisk alerios (n=alerios@190.24.100.110) |
| 14:34.27 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:34.28 | knathraak | so no way to go to voicemail if no answer? |
| 14:34.39 | mercestes | knathraak: HuH? |
| 14:34.45 | *** join/#asterisk nategs500 (n=nronchet@193.82.139.15) |
| 14:34.46 | ambriento | brb |
| 14:34.46 | monsted | knathraak: should be easy enough |
| 14:34.47 | alerios | tzafrir: Hi. what's the workarround to let asterisk group edit /etc/zaptel.conf ? |
| 14:34.53 | mercestes | knathraak: of course you can go to VM if no answer. |
| 14:35.16 | monsted | knathraak: ah, the avaya probably picks up the call - you might need to do the VM there instead |
| 14:35.25 | mercestes | knathraak: Goto(s-${DIALSTATUS}) s-NOANSWER,1,Voicemail(${EXTEN}) |
| 14:35.29 | tzafrir | alerios, not really sure. One option is that root edits it and updates destar |
| 14:35.30 | knathraak | sorry, responding to dlynes_laptop's observatio that there's no way to detect answer on fxo |
| 14:36.20 | tzafrir | This is why I wanted to have seperate objects for hardware lines: root updates the information about the availble hardware lines |
| 14:36.50 | mercestes | alerios: Zaptel is part of group dialout. If you add asterisk to group dialout then the user asterisk can edit zaptel.conf with your group privs. |
| 14:37.02 | nategs500 | I have a device which sends the TRYING on a different port to the INVITE |
| 14:37.08 | nategs500 | when asterisk send the CANCEl it sends to the port that the TRYING came from is this right? |
| 14:37.35 | nategs500 | They seem to think that it should send the CANCEL on the same port as the INVITE |
| 14:38.01 | knathraak | monsted, the avaya isn't picking up the call, it's going to the external pstn number as it should |
| 14:38.04 | alerios | mercestes: zaptel.conf seems to be owned only by root by default (on debian systems) |
| 14:38.21 | alerios | tzafrir: you mean the zapscan script? |
| 14:38.27 | knathraak | monsted: it just rings and rings, and asterisk lets it because it assumes it has already picked up |
| 14:39.05 | tzafrir | alerios, genzaptelconf. zapscan is something a bit different |
| 14:39.39 | alerios | tzafrir: yes, that. |
| 14:40.19 | alerios | tzafrir: but right now destar edits zaptel.conf. could we use a symbolic link? |
| 14:40.49 | tzafrir | alerios, well, this ignores the problem. |
| 14:41.00 | tzafrir | Do you want to allow non-root to write to zaptel.conf? |
| 14:41.08 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 14:41.46 | tzafrir | The thing is that asterisk is perfectly capable of running ztcfg . but it is currently normally run by riit |
| 14:43.26 | tzafrir | A third option is a SUID layer that will do sanity tests (that is: generate zaptel.conf or whatever) |
| 14:43.44 | tzafrir | I don't like it either |
| 14:44.33 | alerios | I don't think I could integrate destar and genzaptelconf before etch :S |
| 14:48.06 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 14:49.01 | mattfletcher | dlynes_laptop: sorry, got distracted |
| 14:49.25 | mattfletcher | dlynes_laptop: the call is coming is on zap/3, and I am trying to process it with rxfax |
| 14:50.08 | tzafrir | alerios, can you derfine an interface to "add lines"? "add spans" |
| 14:50.13 | tzafrir | or whatever? |
| 14:51.10 | *** join/#asterisk jmacz (n=jmacz@190.24.100.110) |
| 14:52.00 | alerios | tzafrir: can't we just make processes on the asterisk group being capable of writing zaptel.conf, or does that go against FHS? |
| 14:52.40 | *** join/#asterisk mega (i=mega@gateway/tor/x-9614e3faf064689c) |
| 14:52.47 | tzafrir | currently it is root. This is zaptel, not asterisk. zaptel.conf is zaptel's configurtation, not asterisk's |
| 14:54.25 | alerios | I see |
| 14:55.23 | *** part/#asterisk mega (i=mega@gateway/tor/x-9614e3faf064689c) |
| 14:56.46 | alerios | <tzafrir> alerios, can you derfine an interface to "add lines"? "add spans" -> could you explain this further? |
| 14:57.21 | tzafrir | if I want to write a script (e.g: genzaptelconf, or a wrapper to it) that will add the lines |
| 14:57.34 | tzafrir | it will run as root |
| 14:57.46 | alerios | yes |
| 15:00.25 | alerios | you may be right, maybe I still got time to make genzaptelconf create destar's zaptel definitions |
| 15:00.26 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 15:00.46 | mattfletcher | Is there anyone who can help me with this fax problem? It's really getting me down now. It is so close, but just will not work] |
| 15:01.52 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 15:02.02 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 15:04.24 | *** join/#asterisk whisperer (i=xsilver@193.165.254.32) |
| 15:04.44 | pif | in a multi-bri setup, if one bri is disabled in zaptel.conf, will the telco try to send any call through it ? or will the call be routed only on the active spans? |
| 15:05.29 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:07.07 | whisperer | Hi there, have U ever seen this "Unable to build sip pvt data for MWI" ??? I got a lot of this in log before * gone (got frozen). What does it mean ?? I tried forums but unsuccesfuly. |
| 15:08.27 | *** join/#asterisk my007ms (n=my007ms@217.139.224.194) |
| 15:09.54 | my007ms | good day all |
| 15:10.04 | my007ms | pleas i need help in fax with asterisk |
| 15:10.20 | my007ms | when i try send fax to my asterisk box i see in log file |
| 15:10.21 | my007ms | Unknown RTP codec 109 received |
| 15:10.28 | my007ms | it's come over and over |
| 15:10.45 | my007ms | and fax machine say connection error |
| 15:11.00 | my007ms | is the problem |
| 15:14.24 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 15:14.31 | b11d | morning lads |
| 15:14.39 | *** join/#asterisk X-Rob_ (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 15:16.21 | pif | why would a sip host be UNREACHABLE when ping'able and listening on the sip port? |
| 15:16.45 | b11d | protocol mismatch? |
| 15:17.10 | b11d | run sip debug on that peer and take a look at the details |
| 15:19.48 | my007ms | what Unknown RTP codec 109 received mean in asterisk log |
| 15:20.57 | whisperer | 'Unable to build sip pvt data for MWI' what does it mean ?? |
| 15:24.19 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 15:27.22 | *** part/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 15:28.44 | *** join/#asterisk Mportnoy (n=test@201.199.68.150) |
| 15:29.15 | *** join/#asterisk pepehammer (i=pepe@adsl-074-228-211-182.sip.asm.bellsouth.net) |
| 15:29.16 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:29.56 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:29.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:30.02 | b11d | geezuz |
| 15:30.18 | pepehammer | does someone know where to locate the asterisk source code in trixbox? |
| 15:31.29 | zx6r | save me jebus! |
| 15:33.53 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 15:34.32 | pepehammer | nobody? |
| 15:34.43 | *** join/#asterisk jaike (i=jaike@210.5.119.11) |
| 15:35.35 | *** part/#asterisk knathraak (n=knath@pool-71-252-87-31.washdc.east.verizon.net) |
| 15:35.47 | mattfletcher | Who/where can I go to for help with a problem receiving a fax over Zap? The call is sent to the fax extension, and rxfax kicks in (the first few screeches are heard) then after about 30 seconds, i get an error. This error in fact: http://pastebin.ca/248276 |
| 15:36.39 | my007ms | pleas i need help in fax with asterisk |
| 15:36.45 | my007ms | when i try send fax to my asterisk box i see in log file Unknown RTP codec 109 received |
| 15:38.41 | *** join/#asterisk tao2 (n=tao@69.63.74.254) |
| 15:39.06 | Peaceful | How do you debug when asterisk crashes? I've got 1.2.10 on a production system that has started crashing at least once per day, randomly. |
| 15:39.18 | ManxPower | my007ms: Turn off T.38 in the SIP device that handles fax. |
| 15:39.58 | jaike | Peaceful: 1.2.10 was really buggy |
| 15:40.05 | my007ms | let me check if that if it work i will name u the king |
| 15:40.07 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:40.34 | tao2 | hi is there a module that i can install that can much numbers to names from a database .I dont really know how to do much with database and would like to know if there is a program that i could install to do it. if the names could come up instead of the numebers on the CLI |
| 15:40.35 | jaike | upgraded to 1.2.12.1 and never had a crash ever since |
| 15:41.00 | Peaceful | jaike: so you had crashes with 1.2.10 as well? |
| 15:41.45 | jaike | yup. a lot |
| 15:41.49 | jaike | 1.2.9.1 was better |
| 15:42.54 | mercestes | I rolled back to 1.2.9 myself, but I'm onto 1.2.13 right now. |
| 15:43.13 | Peaceful | mercestes: had any problems with 1.2.13? |
| 15:43.43 | Peaceful | the weird thing is that I've been running 1.2.10 since it came out, and I had maybe one crash ever before last week. |
| 15:43.47 | mercestes | Peaceful: Not yet. I haven't cranked it into full production yet but it appears pretty stable.. |
| 15:43.51 | Peaceful | Then one or two crashes every day |
| 15:44.10 | mercestes | Peaceful: It's not that it doesn't work now that surprises me, it's that it worked before that has me confused. |
| 15:44.45 | tao2 | do you know of any problems running HooDaHek Version: 1.2.3 |
| 15:44.51 | Peaceful | ya, me too, especially since the usage patterns in my office haven't changed |
| 15:45.17 | mercestes | Peaceful: so if you agree then either upgrade or downgrade since you have reached the expected result..:) |
| 15:45.27 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 15:45.39 | Peaceful | well, if I upgrade to 1.2.13, should I go ahead and upgrade zaptel and libpri as well? |
| 15:45.56 | mercestes | Peaceful: Yes, and asterisk-addons if you use it. |
| 15:47.08 | boddy | hii All I am hearing a sound that coming from background like echo when I am making sip call with using g729 |
| 15:47.10 | jaike | Peaceful: mixmonitor was causing most of my crashes before 1.2.12.1 |
| 15:47.29 | zx6r | 1.2.13 looks ok up until now. rolling into production on saturday, should be interesting :P |
| 15:47.35 | Peaceful | jaike: hmm, I just started using mixmonitor two days ago, but the crashes started a week and a half ago |
| 15:47.36 | boddy | is there any mechanism to prevent echo |
| 15:47.48 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 15:47.53 | jaike | peaceful: then you should upgrade to 1.2.12.1 |
| 15:47.58 | Peaceful | I'll just upgrade everything tonight, and hope... |
| 15:48.05 | Peaceful | k, thanks guys |
| 15:49.05 | boddy | have you any idea about myy question |
| 15:49.17 | zx6r | any of you guys using cisco 7940's with the latest sccp flash? |
| 15:49.39 | ManxPower | zx6r: Why would we want to? |
| 15:49.49 | *** join/#asterisk evisu (i=hIRC@bzq-88-153-168-214.red.bezeqint.net) |
| 15:49.55 | zx6r | ManxPower, for a change of pace :P |
| 15:51.44 | RoyK | http://www.msfirefox.com/about-microsoft-corporation.html |
| 15:51.51 | hello2007 | does asterisk do 3 way calling or it must be a feature on the phone????? |
| 15:52.23 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:53.19 | b11d | thats awesome royk |
| 15:53.36 | b11d | thats not real eh |
| 15:53.49 | boddy | ManxPower:Have you any idea about my question ? |
| 15:54.27 | b11d | . |
| 15:55.10 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 15:55.41 | ManxPower | boddy: what is your question? |
| 15:55.49 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 15:56.23 | ManxPower | hello2007: on Zap it is Asterisk for SIP it's the phone. |
| 15:58.21 | hello2007 | on sip it must be the phone?if my sip phone doen t no have 3 way calling ?i canot do it through asterisk ? |
| 15:59.20 | ManxPower | hello2007: There is the DTMF transfer hack, of course. See features.conf and "show application dial". The good solution is to stop being such a cheap ass and buy phones that support transfers. Hmm, I actually can't think of a phone that does not support any form of transfer. |
| 15:59.43 | *** join/#asterisk punkgode (n=punkgode@rev-200-40-119-222.netgate.com.uy) |
| 16:00.30 | boddy | ManxPower:I am hearing a sound that coming from background like echo when I am making sip call with using g729 |
| 16:00.30 | mattfletcher | Where can I find/buy support for rxfax? |
| 16:00.34 | boddy | is there any mechanism to prevent echo |
| 16:00.55 | ManxPower | boddy: yes, depending on the actual technologies involved. |
| 16:01.03 | hello2007 | it s not for me , some asked this question and i was investigating , i don t care if the phone sing and dance |
| 16:01.11 | my007ms | ManxPower, there is not option with that name " T.38" |
| 16:01.28 | ManxPower | my007ms: what device are you using? |
| 16:02.10 | boddy | ManxPower:this technologies on asterisk side or client side ? |
| 16:02.24 | *** part/#asterisk pepehammer (i=pepe@adsl-074-228-211-182.sip.asm.bellsouth.net) |
| 16:02.29 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 16:02.32 | ManxPower | boddy: Standard echo must be removed by the device that is doing the PSTN/VOIP interface |
| 16:02.55 | my007ms | ManxPower, http://www.sipura.com/ |
| 16:04.12 | punkgode | hello to all, I'm trying to implement callprogress detection on my asterisk box for analog lines, and I wonder if anyone knows the difference between the tonezones defined in dsp.c, indications.conf and zonedata.c (for zaptel module) |
| 16:04.15 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 16:04.20 | ManxPower | my007ms: You need to edit the Asterisk Makefile and uncomment the option about deprecated G726, then rebuild Asterisk |
| 16:04.20 | tao2 | Is there a called id module for * that i can use to show names that match to numbers |
| 16:04.25 | punkgode | i can't understand how they work toghether |
| 16:04.49 | *** join/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
| 16:05.55 | tmccrary | Can anyone recommend an FXO gateway that doesn't suck? I don't mean a digium card (have them, they're great, but I need a standalone gateway). I have an audiocodes and while it works.... it is "staticy" and generally pretty terrible. |
| 16:06.02 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 16:06.17 | punkgode | I've already patched dsp.c and zonedata.c and modified indications.conf (for uruguay), but I don't know where in the process of callwaiting those files fit |
| 16:06.21 | ManxPower | tmccrary: as far as I can tell they all suck. |
| 16:06.24 | mattfletcher | Are there any more options I can set in zapata.conf regarding faxes? So far I have set detectfax = both, rxgain and txgain = 9 |
| 16:06.51 | ManxPower | mattfletcher: why are you using faxdetection? That is only for use when you want to try to use the same phone number for fax and voice. |
| 16:07.04 | mattfletcher | correct |
| 16:07.11 | tmccrary | ManxPower: Doh, that really sucks. :) |
| 16:07.16 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:07.16 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:07.53 | *** join/#asterisk HumpBack (n=sdcdcssc@gentoo/developer/HumpBack) |
| 16:07.57 | HumpBack | Hello all |
| 16:08.00 | punkgode | mattfletcher, you need to be sure that your RJ-11 is properly wired |
| 16:08.04 | mattfletcher | That's exactly what I want to do, and it seems to work. The call is intercepted, and diverted to the "fax" extension in extensions.conf. Then it SEEMS as though rxfax is ignoring the fax |
| 16:08.07 | ManxPower | tmccrary: give the SIPura box a try. It's not easy to get working well, is prone to echo, but is cheap and not the end of the world if you wasted your money. |
| 16:08.24 | mattfletcher | punkgode: if i can hear voice does that mean fax should work? |
| 16:08.27 | punkgode | mattfletcher, ok, that's not your problem then |
| 16:08.36 | mroth_imm | Hello all. Is it possible to record outbound analog calls using an X100P (FXO only)? |
| 16:08.36 | ManxPower | mattfletcher: once the call gets to RXFax Asterisk's fax detection is no longer in play |
| 16:08.38 | HumpBack | I just sent a mail to the asterisk-users list. I would apreciate any help to see if i'm having hardware problems |
| 16:08.59 | ManxPower | mroth_imm: yes, just like with any other type of card. |
| 16:09.04 | mroth_imm | Details at: http://lists.digium.com/pipermail/asterisk-users/2006-November/172276.html |
| 16:09.39 | punkgode | mattfletcher, fax detection won't work with inverted wiring |
| 16:09.54 | mattfletcher | i cannot find any sites or people that have seen this problem. The error I get is http://pastebin.ca/248276. |
| 16:09.57 | ManxPower | mroth_imm: my statement stands |
| 16:09.57 | punkgode | sorry for my english, and slow replys :P |
| 16:10.13 | mroth_imm | ManxPower: okay, thank you for the confirmation |
| 16:10.32 | ManxPower | <PROTECTED> |
| 16:10.45 | mattfletcher | punkgode: what do you mean by fax detection? the fact that it is a fax is detected, as it enters the right extension. is there further detection within rxfax perhps? |
| 16:10.53 | mattfletcher | manxpower: how could i tell |
| 16:10.56 | punkgode | mattfletcher, fax tone detection |
| 16:10.57 | ManxPower | rxfax does not detection |
| 16:11.02 | *** part/#asterisk jaike (i=jaike@210.5.119.11) |
| 16:11.03 | ManxPower | mattfletcher: you would set it. |
| 16:11.10 | mattfletcher | where? |
| 16:11.18 | ManxPower | FaxOverVoiceOverIP does not work well. |
| 16:11.20 | punkgode | mattfletcher, that's why I correct the statement saing it was not your problem |
| 16:11.32 | ManxPower | how are the calls arriving at Asterisk |
| 16:11.41 | mattfletcher | not trying it over ip, it is coming in on a tdm400p |
| 16:11.48 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:11.57 | mattfletcher | all i'm doing yet is storing the tiff on the server |
| 16:12.48 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 16:12.51 | spaghetty | hi all |
| 16:12.54 | b11d | high |
| 16:13.05 | ManxPower | mattfletcher: then I have no idea what the problem is. |
| 16:13.13 | spaghetty | i need to know if i can use a sip request to an asterisk server |
| 16:13.16 | ManxPower | here is my WORKING rxfax line: |
| 16:13.20 | mattfletcher | i cannot see where to define codecs for a zap channel. the only definitions are within sip.conf |
| 16:13.27 | spaghetty | to generate a call |
| 16:13.28 | ManxPower | exten => fax,7,RxFax(/tmp/fax-${UNIQUEID}.tiff) |
| 16:13.38 | ManxPower | mattfletcher: you don't set codecs on zap |
| 16:13.41 | mattfletcher | right |
| 16:13.51 | mattfletcher | so why on earth is the error about codecs then? |
| 16:13.54 | mattfletcher | it makes no sense |
| 16:13.54 | spaghetty | between two diferente user |
| 16:14.11 | ManxPower | mattfletcher: I have never seen that error with RXFax in my 5 years of working with Asterisk |
| 16:14.20 | mattfletcher | http://pastebin.ca/248276 - unable to find a codec translation path |
| 16:14.25 | ManxPower | spaghetty: all SIP calls do that. |
| 16:14.43 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 16:14.47 | mattfletcher | there are more copied of that wording in online copies of asterisk's source code than people who have actually seen it as an error! |
| 16:15.02 | spaghetty | no ManxPower the real caller sould dead there |
| 16:15.12 | ShadwDrgn | wouldn't there be tons of NAT problems using wifi phones on public APs? |
| 16:15.18 | ManxPower | spaghetty: I do not understand? |
| 16:15.20 | spaghetty | and the 2 call enerated sould be connected |
| 16:15.24 | hoobastooba | I am getting choppy calls and seeing deadlock errors suddenly. http://pastebin.ca/248375 |
| 16:15.32 | hoobastooba | can anyone help me get past this? |
| 16:15.39 | ManxPower | ShadwDrgn: not really, since there are no NAT problems with SIP if you set things up right. |
| 16:15.39 | hoobastooba | has been working fine all week, then just started |
| 16:15.59 | ShadwDrgn | ManxPower: wouldn't you have to set up the router to forward ports to the phone? |
| 16:16.06 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 16:16.07 | ManxPower | ShadwDrgn: No. |
| 16:16.12 | ShadwDrgn | ok i'll ask further in a moment |
| 16:16.14 | ShadwDrgn | brb |
| 16:16.18 | ShadwDrgn | thx for help in advance |
| 16:16.22 | ManxPower | ShadwDrgn: you only have to do that if the Asterisk SERVER is behind NAT |
| 16:17.04 | mattfletcher | Well thank you for trying to help me ManxPower |
| 16:17.07 | ShadwDrgn | which in this case unfortunately it is |
| 16:17.24 | ShadwDrgn | however i'm trying to convince it's owner to make linux his router thus solving that problem. |
| 16:17.28 | mattfletcher | I swapped in your rxfax line and get exactly the same message |
| 16:17.30 | ManxPower | ShadwDrgn: Yes, but the Asterisk SERVER is not going to be roaming between NAT routers and APs |
| 16:17.32 | spaghetty | I need to do a 3way call |
| 16:17.43 | hoobastooba | anyone? |
| 16:18.02 | ShadwDrgn | so if i take the server out from behind the NAT it should solve all the stupid sipphone problems i'm having with connecting calls but not hearing voice right? |
| 16:18.05 | ManxPower | set localnet and externip in sip.conf, set the correct port forewarding for SIP and RTP Audio and you are done |
| 16:18.18 | spaghetty | or something like that |
| 16:18.18 | ShadwDrgn | i've done all of that |
| 16:18.22 | ManxPower | ShadwDrgn: only if you don't have the NAT router for Asterisk set up correctlu |
| 16:18.38 | ShadwDrgn | i've forwarded 8000-10000 in the router to asterisk udp |
| 16:18.39 | ManxPower | spaghetty: What device are you needing to do a 3-wy call from? |
| 16:18.42 | ShadwDrgn | and that's in my rtp.conf |
| 16:18.45 | ShadwDrgn | and 5060 also |
| 16:18.55 | spaghetty | ManxPower: all on sip channels |
| 16:18.56 | ShadwDrgn | however when i register with my softphone at another location i get rings but no voice |
| 16:19.07 | ManxPower | ShadwDrgn: that should work. what is your localnet and externip set tp |
| 16:19.11 | spaghetty | the server should do everithing |
| 16:19.11 | ShadwDrgn | the call connects but no one can hear |
| 16:19.18 | ShadwDrgn | hold on i'll show you |
| 16:19.21 | ManxPower | spaghetty: I already told you. |
| 16:19.36 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 16:19.36 | spaghetty | when ? |
| 16:19.43 | ShadwDrgn | externip = 24.178.73.251 |
| 16:19.43 | ShadwDrgn | localnet = 192.168.0.119/255.255.255.0 |
| 16:19.56 | ShadwDrgn | that's correct right? |
| 16:20.18 | spaghetty | I'm able to do this now only with manger protocol |
| 16:20.30 | *** part/#asterisk punkgode (n=punkgode@rev-200-40-119-222.netgate.com.uy) |
| 16:21.20 | ShadwDrgn | ?? is that not right? |
| 16:21.26 | ManxPower | ShadwDrgn: your loalnet is invalid |
| 16:21.32 | ShadwDrgn | ah |
| 16:21.41 | ShadwDrgn | what should it be? |
| 16:21.47 | ShadwDrgn | /24? |
| 16:21.51 | ManxPower | localnet = 192.168.0.0/255.255.255.0 |
| 16:22.05 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 16:22.10 | ShadwDrgn | alright |
| 16:22.32 | ManxPower | spaghetty: what part of "see 'show application dial' and see features.conf" do you not understand? |
| 16:22.41 | ShadwDrgn | so now with that and a reload i shouldn't have to forward ports on routers the sipphones are behind, right? |
| 16:22.51 | ShadwDrgn | i just set NAT=yes and host=dynamic |
| 16:22.55 | hoobastooba | asterisk is running wild, using 4GB of memory. |
| 16:22.59 | ShadwDrgn | any other settings i need to make them work? |
| 16:23.10 | ManxPower | ShadwDrgn: should never have to port forward on the NAT router that does NAT for the SIP client. |
| 16:23.11 | ShadwDrgn | i also did canreinvite=no but i'm not sure if that's correct. |
| 16:23.27 | tao2 | has any one here used hoodahek ? |
| 16:24.07 | ShadwDrgn | ManxPower: I will test this as soon as my friend gets back. when we make extension<->extension calls it messes up. |
| 16:24.29 | ShadwDrgn | however calls going to voicemail or out the trunk work fine |
| 16:24.31 | ShadwDrgn | it's strange. |
| 16:24.32 | spaghetty | ManxPower: simply I've not read it ... |
| 16:24.35 | ManxPower | ShadwDrgn: It is pretty easy to mess up sip.conf to have an incoming call not match the [whatever] entry in sip.conf |
| 16:24.52 | ShadwDrgn | well the proper extension rings, but when you pick up no sound is heard |
| 16:25.16 | dasenjo | Hi, I'm using Dial with tTm options .. the defualt moh sounds on transfer, but I want to play the transfer/pbx-tranfer.gsm file at start too. Is it possible? |
| 16:25.18 | ManxPower | ShadwDrgn: put the output of |
| 16:25.22 | ShadwDrgn | and i know my extensions.conf is right because someone here set it up for me |
| 16:25.33 | ManxPower | "sip show peers" and the CLI output of a call on pastebin.ca |
| 16:25.50 | ShadwDrgn | when my friend gets back to help me test i will do that |
| 16:27.35 | ShadwDrgn | thanks again for your help |
| 16:27.42 | ShadwDrgn | i'm new to asterisk, but i like to think i'm learning quicklty |
| 16:27.47 | ShadwDrgn | quickly too |
| 16:29.11 | ShadwDrgn | ManxPower: is there perhaps a really good wifi phone you would suggest i get? |
| 16:29.20 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:30.26 | *** join/#asterisk MasterYoda (i=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 16:30.34 | *** part/#asterisk MasterYoda (i=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 16:30.47 | L|NUX | mog : hey |
| 16:31.04 | L|NUX | mog : arround ? |
| 16:31.54 | ShadwDrgn | anyone know what the best (or a really good quality) hardware wifi sipphone is? |
| 16:32.03 | mroth_imm | ManxPower: I can't find any example configurations of recording an outbound call originating at an analog phone connected via an X100P...could you point me in the right direction? |
| 16:32.15 | *** join/#asterisk Alexander_I (i=mnichols@nat/digium/x-1554283512dd4777) |
| 16:32.25 | Alexander_I | does chan_sip.c keep a dns cache in 1.2? |
| 16:32.34 | *** part/#asterisk Abydos313 (i=abydos31@ppp-71-134-22-163.dsl.irvnca.pacbell.net) |
| 16:33.33 | *** join/#asterisk MasterYoda (i=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 16:33.46 | MasterYoda | oops, signed in twice, same question |
| 16:35.03 | tmccrary | ShadwDrgn: I have a few Hitachi IP-5000 phones and they are pretty reliable. |
| 16:35.10 | Corydon-w | MasterYoda: what was the question? |
| 16:35.23 | MasterYoda | does chan_sip.c keep a dns cache |
| 16:35.28 | tzanger | tmccrary: what's talk/standby time like? provisioning? ringers? loudness? |
| 16:35.38 | Corydon-w | MasterYoda: not a dns cache, no |
| 16:35.38 | mercestes | This is probably a total n00b question by now but prog is good for listening to .gsm files on a windows machine?? |
| 16:35.41 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:35.49 | Corydon-w | MasterYoda: but it does resolve hostnames just once |
| 16:36.01 | MasterYoda | Corydon-w: can I tell it to resolve again? |
| 16:36.20 | ShadwDrgn | tmccrary: thanks. |
| 16:36.25 | Corydon-w | MasterYoda: only by removing the peer, doing a reload, then reinstituting the peer and reloading again |
| 16:36.37 | iulius | mercestes: windows media player will play them; you may need to change the file extension from .wav to .gsm though |
| 16:36.44 | MasterYoda | Corydon-w: it is a register => line |
| 16:36.47 | Corydon-w | MasterYoda: basically you have to get the structure to go away |
| 16:37.06 | tao2 | is there any one out there that could help with the intall of hooDaHek |
| 16:37.07 | MasterYoda | Corydon-w: it appears it is not just chan sip, I can't ping that address from this machine |
| 16:37.08 | Corydon-w | MasterYoda: in that case, no, you'd need to unload chan_sip.so |
| 16:37.12 | mercestes | iulius: Hmm..windows media player didn't even pick them up...let me dink with it a little more. |
| 16:37.14 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 16:37.37 | Corydon-w | MasterYoda: or you could remove the register, do a reload, etc. |
| 16:37.59 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 16:38.01 | MasterYoda | Corydon-w: it does not seem to be an issue with chan_sip, looks like a system wide thing |
| 16:38.06 | mercestes | Media player is not happy with Gsm. |
| 16:38.10 | Corydon-w | MasterYoda: ah |
| 16:38.22 | Corydon-w | mercestes: is the length of the file odd or even? |
| 16:38.28 | MasterYoda | Corydon-w: thanks anyway |
| 16:38.30 | mercestes | even. |
| 16:38.37 | Corydon-w | mercestes: dunno then |
| 16:38.48 | mercestes | wait...both. |
| 16:38.50 | mercestes | lol. |
| 16:38.58 | mercestes | there is a bunch of them..but Media Player is happy with none of them. |
| 16:38.58 | *** part/#asterisk MasterYoda (i=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 16:39.02 | Corydon-w | mercestes: it should play even-length files fine |
| 16:39.42 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 16:39.53 | mercestes | Any other suggestions to play a gsm file? Audacity doesn't like them either..:/ |
| 16:40.05 | Corydon-w | mercestes: .gsm or .WAV ? |
| 16:40.15 | mercestes | .gsm. |
| 16:40.35 | *** join/#asterisk SwK (n=Silik0nJ@br0.asteriasgi.com) |
| 16:40.37 | Corydon-w | mercestes: oh, yeah, media player doesn't like those. Try recording in wav49 instead |
| 16:40.41 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 16:41.07 | mercestes | Oh...no way to play gsm?? |
| 16:41.20 | Corydon-w | mercestes: MS did something weird with their gsm format, so it's different than the standard format |
| 16:41.32 | mercestes | oh... |
| 16:41.33 | _fenlander | use quicktime |
| 16:41.35 | mercestes | stupid MS. |
| 16:41.51 | mercestes | Audacity in Linux work with gsm? |
| 16:42.06 | Corydon-w | Just use 'play' in Linux |
| 16:42.30 | mercestes | okies...thanks |
| 16:44.44 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:47.06 | *** join/#asterisk cryptnix (n=andrew@68-188-226-83.dhcp.bycy.mi.charter.com) |
| 16:47.59 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 16:48.49 | *** join/#asterisk SoftIce (n=newbie@vc-196-207-45-253.3g.vodacom.co.za) |
| 16:49.06 | SoftIce | hi, where in the trixbox admin panel can I configure a phone or find all phones on the network with their mac address, etc? |
| 16:49.35 | SoftIce | neer mind wrong channel |
| 16:49.38 | froguz | what if i don't want asterisk to send an e-mail with the voicemail? just wanted to rescue voicemail from the phone. |
| 16:49.39 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 16:50.23 | brookshire | froguz: don't add an email address |
| 16:50.34 | brookshire | to voicemail.conf |
| 16:50.52 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 16:50.54 | froguz | there isn't a more elegant way? |
| 16:51.23 | ShadwDrgn | hey nevermind about the extension to extension calls not work |
| 16:51.29 | ShadwDrgn | working... it looks like fixing the localnet fixed it |
| 16:51.32 | ShadwDrgn | thanks again ManxPower |
| 16:51.36 | brookshire | how is that not elegant? |
| 16:51.54 | ShadwDrgn | only one more problem to work out and everything will be exactly how i want it |
| 16:51.55 | ShadwDrgn | lol |
| 16:53.06 | ManxPower | ShadwDrgn: contrary to what people say SIP+NAT is NOT hard to get working as long as you follow a few rules. |
| 16:53.24 | ShadwDrgn | yeah |
| 16:53.36 | ShadwDrgn | i can't believe i made such an amatuer mistake with the localnet though |
| 16:53.46 | froguz | brookshire, i think 'sendemail=' option would be better, so asterisk doesn't even has to try to send e-mails |
| 16:53.47 | ShadwDrgn | duh @ 0 needing to be at the end |
| 16:53.52 | ShadwDrgn | but now my problem is |
| 16:53.54 | ManxPower | ShadwDrgn: Apparently you are not a neworking person. |
| 16:53.57 | ShadwDrgn | i am |
| 16:54.01 | ShadwDrgn | that's why i feel stupid |
| 16:54.06 | ShadwDrgn | i KNOW that i just totally didn't see it |
| 16:54.54 | ShadwDrgn | anyway my problem now is that i have two phone numbers with sunrocket |
| 16:54.59 | ShadwDrgn | i have one ringing extension 2000 |
| 16:55.02 | ShadwDrgn | that works great |
| 16:55.10 | ShadwDrgn | we'll call that 6781111111 |
| 16:55.31 | ShadwDrgn | the line looks like this: exten => 6781111111,1,Goto(2000,1) |
| 16:55.41 | ShadwDrgn | but when i try to make the second number ring 2001 |
| 16:55.43 | ShadwDrgn | with |
| 16:55.54 | ShadwDrgn | exten => 6782222222,1,Goto(2001,1) |
| 16:56.01 | ShadwDrgn | it doesn't work... :( it rings 2000 |
| 16:56.31 | ManxPower | don't flood the channel, use pastebin.ca |
| 16:56.32 | *** part/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 16:56.35 | ShadwDrgn | sorry |
| 16:56.43 | ManxPower | watching the CLI will help you |
| 17:01.34 | *** join/#asterisk designdream (n=felipe@rrcs-71-40-49-30.sw.biz.rr.com) |
| 17:01.36 | ShadwDrgn | ManxPower |
| 17:01.37 | ShadwDrgn | http://pastebin.ca/248398 |
| 17:01.52 | SoftIce | hi, i've configured my phone with its ip, netmask, timezone, gateway, etc etc. now it comes to a quetion that says Account: and on the far right it says 123 |
| 17:01.58 | SoftIce | I cant get past that option on the phone |
| 17:02.11 | b11d | nice |
| 17:02.28 | ShadwDrgn | it should be ringing 2001 |
| 17:02.44 | luke-jr_work | is it possible to lookup an email from a voicemail #? |
| 17:02.47 | luke-jr_work | in the dialplay |
| 17:02.47 | designdream | I am currently having a problem with transfering phones via polycom 501s.. when the user transfers to my phone.. all presence plugins still show that person on the phone after they hang up |
| 17:02.49 | luke-jr_work | plan* |
| 17:03.35 | ManxPower | ShadwDrgn: you are calling from a local sip phone? |
| 17:03.41 | ShadwDrgn | no i'm calling from a cellphone |
| 17:03.53 | ManxPower | ShadwDrgn: so you are calling a SIP provider? |
| 17:04.03 | ShadwDrgn | yes |
| 17:04.16 | hoobastooba | anyone else here having trouble with memory allocation and swap space with asterisk? i have 4 gb in one machine and it seems like it just is not enough. |
| 17:04.17 | ManxPower | sounds to me that the incoming call from the SIP provider is sending the call to the wrong DID |
| 17:04.17 | ShadwDrgn | i'm calling my trunk |
| 17:04.20 | ShadwDrgn | but there'r two numbers |
| 17:04.29 | ManxPower | ShadwDrgn: don't use the word "trunk". |
| 17:04.32 | ShadwDrgn | sorry |
| 17:04.36 | ManxPower | Asterisk has no such thing as a SIP trunk |
| 17:04.49 | ShadwDrgn | yeah i wondered why people used that term |
| 17:04.55 | ManxPower | ShadwDrgn: pastebin your register => lines without the passwords |
| 17:05.05 | ShadwDrgn | sure |
| 17:05.46 | BrokenNoze | Hi people. anyone know if a specific call can be removed from a queue via the Manager API? |
| 17:05.51 | Crescendo | Is there any known issues with SpeeX not working 80% of the time on transcoded calls from other codecs, and not working at all when calling out using SpeeX to dial out? |
| 17:06.46 | ShadwDrgn | ManxPower http://pastebin.ca/248403 |
| 17:07.11 | ManxPower | ShadwDrgn: you have a problem with your provider, not Asterisk |
| 17:07.17 | ShadwDrgn | :( |
| 17:07.19 | ShadwDrgn | but in sip show peers |
| 17:07.31 | ShadwDrgn | wait do i need a [] in sip.conf for both register lines? |
| 17:07.34 | ManxPower | ShadwDrgn: sip show peers shows remote devices registered to Asterisk |
| 17:07.47 | ShadwDrgn | oh |
| 17:08.09 | mroth_imm | Is it possible to record a call dialed from an analog phone connected to the "Phone In" port of an X100P? |
| 17:08.36 | BrokenNoze | anyone? |
| 17:08.56 | pif | my asterisk is behind is behind a nat router, what should I absolutely put in sip.conf 's general section? |
| 17:08.58 | ManxPower | mroth_imm: no. the two ports on the X100P are hardwired togather. |
| 17:09.06 | pif | externhost ? |
| 17:09.12 | luke-jr_work | :\ |
| 17:09.27 | ManxPower | pif: localnet and externip or externhost, forward the ports on your router |
| 17:09.41 | mroth_imm | ManxPower: Thanks! I'll just substitute a Snom in my mock up then : ) |
| 17:09.44 | pif | ok |
| 17:11.57 | dioedu | hello all |
| 17:12.29 | dioedu | realtime static configuration support all configuration files ? |
| 17:12.50 | dioedu | for example zapata.conf ? |
| 17:13.44 | dioedu | if i put in extconfig.conf the line zapata.conf => mysql,asterisk,zapata_config |
| 17:14.11 | dioedu | asterisk read the configuration from mysql table ? |
| 17:17.10 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 17:20.42 | *** join/#asterisk networ (n=lamer@wirphone2.it-help.cz) |
| 17:21.08 | networ | hello guys |
| 17:21.21 | *** join/#asterisk dexteruk (n=charper@adsl-68-90-35-43.dsl.hstntx.swbell.net) |
| 17:21.56 | networ | i would ask you for a little problem... i cannot start asterisk as service... |
| 17:22.38 | *** join/#asterisk dexteruk (n=charper@adsl-68-90-35-43.dsl.hstntx.swbell.net) |
| 17:22.51 | *** part/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 17:23.30 | ManxPower | networ: "asterisk -cvvv" that will allow you to see the error that is preventing Asterisk from starting |
| 17:24.10 | networ | ManxPower i can see any problem... |
| 17:24.37 | networ | sometimes wrote me segmentation fault |
| 17:25.01 | networ | i reinstalled |
| 17:25.05 | networ | but same |
| 17:25.10 | SoftIce | well its most probabbly a user error |
| 17:25.16 | SoftIce | and starting it as a service is trying to use another user |
| 17:25.27 | networ | i trying as root |
| 17:25.27 | SoftIce | asterisk -cvvv is starting as root |
| 17:25.31 | ManxPower | SoftIce: not by default |
| 17:25.42 | *** join/#asterisk arkonadev (n=julian@65.106.115.194.ptr.us.xo.net) |
| 17:25.43 | dexteruk | networ: what does it say in the /var/log/asterisk/full |
| 17:25.49 | networ | w8 |
| 17:26.04 | arkonadev | hey is there a Berkeley DB viewer that would allow me to see the contents of a berkeley db file? |
| 17:26.14 | pif | strangely a remote asterisk does not "see" my behind-net asterisk |
| 17:26.22 | pif | s/net/nat/ |
| 17:26.31 | networ | dexteruk i have only debug |
| 17:26.32 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-245.range86-136.btcentralplus.com) |
| 17:26.55 | pif | even though the router forwards the correct ports |
| 17:27.23 | pif | the remote asterisk sees iax but not sip |
| 17:28.18 | luke-jr_work | is it possible to lookup an email from a voicemail # in the dialplan? |
| 17:28.44 | ManxPower | luke-jr_work: not unless you write an AGI |
| 17:29.08 | *** part/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 17:29.33 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 17:30.51 | *** join/#asterisk Alric (n=nbowyer@avantacom.com) |
| 17:31.15 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-63-46.cybersurf.com) |
| 17:32.38 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:33.50 | luke-jr_work | ManxPower, :( |
| 17:34.25 | *** part/#asterisk arkonadev (n=julian@65.106.115.194.ptr.us.xo.net) |
| 17:39.15 | ShadwDrgn | ManxPower |
| 17:39.55 | ShadwDrgn | in my sip debug it shows the incoming calls as: "To: <sip:6782222222@sunrocket.com>;tag=as37ee1a12" |
| 17:40.32 | ShadwDrgn | how do i match that in my extensions.conf? |
| 17:41.01 | ManxPower | ShadwDrgn: exten => 6782222222 should do it. |
| 17:41.14 | ShadwDrgn | it doesn't though :( |
| 17:41.27 | *** join/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 17:49.08 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-8b774fbc391b325d) |
| 17:49.44 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 17:52.09 | HumpBack | Any one here ever tried calling the Digium Support Lines? |
| 17:52.20 | HumpBack | I'm trying to call via IAX tell with no luck |
| 17:54.42 | Alric | Anyone had significant "deal-breaker" kind of problems with the 2.0 Polycom firmware? |
| 17:55.09 | ManxPower | Alric: no, but I don't use 2.x because of all the reported problems with it. |
| 17:55.31 | Alric | ManxPower: What version have you settled on? |
| 17:55.38 | ManxPower | 1.6.x |
| 17:55.53 | ShadwDrgn | ManxPower: if i pasteit my sip debug do you think you would be able to figure out why my extension line isn't catching the incoming call? |
| 17:56.03 | ManxPower | ShadwDrgn: no., |
| 17:57.46 | *** join/#asterisk clive- (n=pirch@dsl-242-165-63.telkomadsl.co.za) |
| 17:57.54 | Alric | exten => _.,1,NoOp(Hit ${EXTEN}) ? :) |
| 18:00.46 | *** join/#asterisk saftsack (n=saftsack@pD9E0509C.dip.t-dialin.net) |
| 18:00.51 | luke-jr_work | ManxPower, can an AGI do a lookup via Asterisk or something? or do I need to parse it myself as well? |
| 18:01.45 | tao2 | ok i have spent hours and hours trying to makes names to match numbers for incoming caller id .... please can someone help me |
| 18:01.52 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 18:01.54 | Stephnie | hi |
| 18:02.30 | *** join/#asterisk kph100 (n=kph100@206-248-157-22.dsl.teksavvy.com) |
| 18:02.30 | Stephnie | any website to buy VOIP DIDs??? and VoIP TOLL FREE Numbers?? Other than voxbone |
| 18:02.44 | luke-jr_work | lots |
| 18:02.55 | clive- | stephie plenty, but is voxbone bad? |
| 18:02.56 | Stephnie | like ? |
| 18:03.11 | Nugget | asterlink.com!! or connect.voicepulse.com!! |
| 18:03.12 | tao2 | I am shore it is a easy ish database table entry but i just dont have the know how |
| 18:03.13 | luke-jr_work | http://www.teliax.com/?a=25846674 |
| 18:03.38 | clive- | well asterlink dont return emails, so I wouldnt recommend them |
| 18:03.42 | Stephnie | not voxbone is gud..but in some countries thery are not good.. |
| 18:03.43 | ManxPower | luke-jr_work: Asterisk has no support for doing what you want. You'll have to write it all yourself. |
| 18:03.50 | ManxPower | WHY would you need to get the info? |
| 18:03.52 | luke-jr_work | ManxPower, hmm... sed might work |
| 18:03.56 | luke-jr_work | ManxPower, for mailing a fax |
| 18:04.25 | ManxPower | *nod* parse voicemail.conf yourself. |
| 18:04.45 | Stephnie | clive-: have you tried asterlink ? |
| 18:04.54 | kph100 | I have created cdr table in mysql. However, no records are created in the table. do i need to add something in the dialplan? |
| 18:05.26 | clive- | stephie, I trie dto try them, but their customer service so far is not good |
| 18:05.26 | clive- | maybe nufone is a good option |
| 18:05.33 | Stephnie | alright |
| 18:05.36 | Stephnie | let me check nufone as well |
| 18:06.06 | zx6r | can you migrate voicemail.conf to a db table? would probably make parsing it a lot easier |
| 18:09.08 | Stephnie | couldnt find anythign related to DID at both websites.....voicepulse and asterlink |
| 18:11.42 | *** join/#asterisk bluregard (n=matt@c-67-163-72-68.hsd1.il.comcast.net) |
| 18:11.50 | bluregard | hi all |
| 18:12.08 | *** join/#asterisk _santiago_ (n=santiago@debian/developer/santiago) |
| 18:13.00 | bluregard | can anyone tell me if the global channel variables section of the API docs is the best reference for variables available in the dialplan? |
| 18:13.31 | ManxPower | bluregard: no, README.variables is |
| 18:14.00 | bluregard | Didn't see that one. Thank you |
| 18:14.16 | ManxPower | <PROTECTED> |
| 18:14.22 | *** join/#asterisk ucfMethod (n=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 18:14.48 | bluregard | yeah, it looks like the same doc actually |
| 18:15.07 | ucfMethod | can anyone recommend a good VOIP to PSTN SIP trunk provider? I know about the list on voip-info, I am asking for a human opinion of some of the better ones people have actually used. |
| 18:16.45 | ManxPower | They all suck. |
| 18:16.52 | ManxPower | Teliax seems to suck less than most |
| 18:16.53 | ucfMethod | really |
| 18:17.19 | ucfMethod | I am moving away from PRIs because they are just damn expensive, and we only have like 20 employees who actually make calls |
| 18:17.34 | bluregard | ManxPower: they're both the same, thanks though. |
| 18:17.42 | ucfMethod | of which probably only spend like 50 minutes a day on the phones combined |
| 18:18.19 | ManxPower | bluregard: Also various applications set variables. See the application docs for that info |
| 18:21.14 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-245.range86-136.btcentralplus.com) |
| 18:22.56 | bluregard | very good |
| 18:24.15 | *** join/#asterisk ohad (n=ohad@19-231-13-72.cosmoweb.net) |
| 18:25.06 | ohad | . |
| 18:26.54 | networ | guys - can anybody explain me why asterisk end with segmentation fault (full log - http://pastebin.com/824292), please? |
| 18:28.38 | ManxPower | networ: what verison of ASterisk? |
| 18:29.11 | ManxPower | networ: that pastebin shows no indication of a segfault |
| 18:29.16 | *** join/#asterisk fadey (n=dima@87.218.18.106) |
| 18:29.27 | networ | ManxPower its 1.2.12.1 |
| 18:29.42 | networ | ManxPower yes in log is not wrote |
| 18:30.16 | ManxPower | have you tried 1.2.13 it has several ciritcal bug fixes. Also see README.backtrace to generate a backtrace from the core file |
| 18:30.28 | networ | i will try |
| 18:30.46 | networ | thanks |
| 18:31.47 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-235-141.red.bezeqint.net) |
| 18:35.06 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:36.49 | tmccrary | God I hate audiocodes and their worthless hardware..... |
| 18:37.25 | ManxPower | tmccrary: then why do you use them? |
| 18:39.59 | monsted | tmccrary: hmm? what's wrong with it? |
| 18:40.13 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 18:40.52 | monsted | somebody in our company likes them - they bought 19 8-port FXO ones and a truckload of 2-port FXS ones to support fax on a nortel setup :) |
| 18:42.00 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:42.07 | ManxPower | tmccrary: welcome to hell. |
| 18:44.20 | tao2 | what toll free number can i dail in the US ? |
| 18:46.03 | hmmhesays | 1800-spank-me |
| 18:46.17 | hmmhesays | tmccrary: they're ok |
| 18:46.24 | hmmhesays | i prefer quintum gateways over audiocodes |
| 18:47.13 | hmmhesays | http://www.youtube.com/comment_servlet?all_comments&v=WeWq6rWzChw&fromurl=/watch%3Fv%3DWeWq6rWzChw%26eurl%3D |
| 18:47.18 | hmmhesays | sony rocks |
| 18:47.21 | *** join/#asterisk graphyx (n=graphyx@205.208.245.105) |
| 18:47.32 | *** join/#asterisk fuzzybunny (n=fuzzybun@host-64-179-8-41.buf.choiceone.net) |
| 18:47.32 | *** part/#asterisk graphyx (n=graphyx@205.208.245.105) |
| 18:47.40 | *** join/#asterisk TmBerg (n=TmBerg@pdpc/supporter/basic/TmBerg) |
| 18:47.50 | fuzzybunny | hello everyone, I was wondering will asterisk run on a gumstix (for home use) |
| 18:48.04 | hmmhesays | google would answer that for you |
| 18:48.05 | ManxPower | tao2: 800-784-2433 |
| 18:48.33 | fuzzybunny | hmmhesays asterisk says PIII 500 mhz |
| 18:48.41 | tao2 | thanks |
| 18:48.43 | [hC] | fuzzybunny: there has been a lot of work done on gumstix and asterisk. |
| 18:48.46 | [hC] | fuzzybunny: so yes, it works. |
| 18:48.46 | monsted | tao2: just grab a support number from any american isp or something like that |
| 18:49.03 | [hC] | fuzzybunny: kristian kielhofner did a bunch of work on it |
| 18:49.43 | tao2 | one more thing . if i am * box is behind a firewall do i have to fwd port 5060 only ? |
| 18:51.20 | networ | i make coredump ... are here anybody who will explain me please (core dump - http://pastebin.com/824312) |
| 18:51.56 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 18:52.34 | *** join/#asterisk mhart (n=chatzill@cowboy.intuit.com) |
| 18:52.53 | ManxPower | tao2: no. see the wiki |
| 18:53.02 | saftsack | tao2, no rtp 10000 - 20000 too |
| 18:53.16 | saftsack | and set the localnet and externip variable |
| 18:53.30 | ManxPower | UDP not TCP of course |
| 18:53.40 | saftsack | yes this is true |
| 18:53.46 | saftsack | i mean rtp -> udp |
| 18:54.12 | mhart | I'm having trouble getting asterisk to wait for answer on an outgoing/.call file via Gizmo. Call works fine, it just doesn't wait for answer before running my AGI script. channel status is always 6 (line is up). Anyone have any advice? |
| 18:54.47 | ManxPower | mhart: analog ports are considered answered as soon as dialing is finished. |
| 18:55.09 | mhart | thx |
| 18:55.26 | *** join/#asterisk fuzzybunny (n=fuzzybun@host-64-179-8-41.buf.choiceone.net) |
| 18:55.36 | fuzzybunny | sorry [hC] I'm back |
| 18:55.43 | hmmhesays | that shouldn't be so |
| 18:55.49 | hmmhesays | that is some broken answer supervision |
| 18:56.00 | froguz | mhart, maybe answeronpolarityswitch=yes on your zaptelćonf may help |
| 18:56.04 | ManxPower | hmmhesays: uh, analog ports don't have answer supervision in Asterisk |
| 18:56.14 | froguz | *zapata.conf |
| 18:56.45 | mhart | frog - software only - no zaptel board. Would that still work? |
| 18:56.45 | ManxPower | froguz: that MIGHT work if the device does answer supervision and they don't usually do that |
| 18:56.45 | froguz | nop |
| 18:56.45 | fuzzybunny | [hC] do you see any issues with being 100mhz slower than min-spec for home use? |
| 18:56.47 | hmmhesays | ManxPower: that sucks |
| 18:56.52 | hmmhesays | good thing I don't use them then |
| 18:58.29 | froguz | mhart, if you don't have any zaptel board and you're using just SIP, then Answer() should work |
| 18:59.35 | *** part/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 18:59.54 | mhart | frog - Not sure what you mean... I don't need to answer, exactly. The .call file dials out and then passes the channel control to my AGI script. The wait for answer parameter in the call file is ignored... or as Manx said, it's recognized as answered as soon as the call goes out. |
| 19:00.29 | mhart | If it was an incoming call, then I could pass to the script and Answer()... if that's what you mean? |
| 19:00.48 | froguz | yep |
| 19:00.55 | mhart | thanks |
| 19:00.58 | fadey | Hello, everyone. Could someone advise me a tool for asterisk stress-testing? |
| 19:02.06 | Corydon-w | A rubber band? |
| 19:02.21 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 19:02.22 | backblue | why does asterisk send presence busy, if someone it's not online? |
| 19:02.29 | Nugget | give accounts to a dozen 14 year old girls. |
| 19:03.00 | Corydon-w | Nugget: it's as good as anything else. Maybe better. ;-) |
| 19:03.10 | mhart | thanks all - I'll band-aid it for now until I can go digital |
| 19:06.01 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-f1a56fed98f8d94e) |
| 19:06.52 | *** join/#asterisk gerphimum (n=trekkie@cpe-68-206-83-62.satx.res.rr.com) |
| 19:06.59 | tao2 | where do i set my and set the "localnet and externip variable" my TB is on a private adderess and i am setting up a trunk |
| 19:07.11 | *** part/#asterisk fuzzybunny (n=fuzzybun@host-64-179-8-41.buf.choiceone.net) |
| 19:08.38 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 19:09.53 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-beaebe1d3b6fc70c) |
| 19:11.57 | tao2 | i keep getting all cercites are busy where can i see a logfile too see y |
| 19:12.13 | tao2 | when setting a a voip out call |
| 19:12.26 | tao2 | it could be my fire wall ? |
| 19:12.30 | ManxPower | asterisk does not have the concept of a SIP trunk. |
| 19:12.33 | *** join/#asterisk oej_ (n=oej@apollo.webway.se) |
| 19:12.39 | ManxPower | you set localnet and externip in sip.conf |
| 19:13.41 | fadey | ManxPower: how the "trunk analogs" are called in SIP terms then? |
| 19:14.32 | dexteruk | fadey: Sip temination is what we use to connect a sip channel to the outside world not a trunk |
| 19:15.29 | *** join/#asterisk arkonadev (n=julian@65.106.115.194.ptr.us.xo.net) |
| 19:15.38 | ManxPower | Asterisk does not CARE what the endpoint is, it's just a sip device |
| 19:15.42 | arkonadev | anyone here using the asterisk-java.org jar for the manager API? |
| 19:15.57 | arkonadev | im trying to figure out how to register a custom event class from the music on hold event |
| 19:16.25 | dexteruk | fadey: If you are using AMP then they class all outbound connections as TRUNK's but SIP as a protocol does not have a trunk unlike IAX that support a trunk fuction that allows you to have more concurrent calls with less bandwidth |
| 19:17.36 | tao2 | ok |
| 19:18.09 | fadey | dexteruk: thanks. |
| 19:18.14 | ManxPower | amp uses lots of terms and configurations that are just plain wrong |
| 19:19.13 | fadey | So, in SIP I have channels then? And what's a trunk? Bunch of channels? |
| 19:20.32 | *** join/#asterisk gputer (n=gputer@60.254.14.209) |
| 19:20.35 | gputer | hello all |
| 19:21.11 | b11d | fire away then |
| 19:21.21 | b11d | we just sit here waiting for newbs to ask for direction |
| 19:21.36 | gputer | lol b11d that's well put |
| 19:21.51 | b11d | i wasnt really kidding though.. seriously.. if you have questions, just ask them |
| 19:21.52 | gputer | ok.. if i was to get directly to business.. i am not sure if i am in the right place |
| 19:22.07 | gputer | but i have been reading about asterisk for quite some time.. and now am stuck with a unique issue |
| 19:22.12 | b11d | this is a place for asking for advice about telco hardware and the Asterisk package. |
| 19:22.23 | gputer | i run a small company and we suddenly have 10 offices in india now |
| 19:22.24 | b11d | well lets talk about that unique issue |
| 19:22.28 | b11d | cool |
| 19:22.30 | gputer | and we want to connect all of them on voip |
| 19:22.35 | b11d | good idea |
| 19:22.47 | gputer | one way is to get sip phones and put them on live ip |
| 19:22.57 | gputer | at different offices, and they can call each other. but this is not what we want to do |
| 19:23.02 | b11d | i dont know what you mean by "live" but ok |
| 19:23.03 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 19:23.12 | gputer | live = static (indian speak) |
| 19:23.13 | brookshire | gputer: it really depends on how many people you have at each location |
| 19:23.20 | b11d | oh.. |
| 19:23.21 | gputer | 2 people at each location |
| 19:23.27 | b11d | you'll be fine |
| 19:23.29 | brookshire | generally, you could centralize your entire phone system |
| 19:23.35 | b11d | let him finish |
| 19:23.36 | gputer | but anyway.. from my some what limited theorotical knowledge |
| 19:23.38 | brookshire | and only have one asterisk pbx for the entire organization |
| 19:23.40 | gputer | this is what i would want to do |
| 19:23.48 | b11d | ok this one is yours brookshire |
| 19:23.50 | gputer | put sip boxes from company lke dlink / sipura / linksys |
| 19:23.55 | gputer | at each location (LOL) |
| 19:24.01 | gputer | and put regular analog lines |
| 19:24.12 | gputer | and put a sip server at an isp end which actually acts as a directory server |
| 19:24.21 | brookshire | gputer: if you only have two people at each location, i would definitely put an asterisk box in a colo or something with lots of bandwidth, and centralize the whole infrastructure |
| 19:24.27 | gputer | now i am not sure if i am going in the right direction and what is the rightt way of doing it |
| 19:24.29 | brookshire | much easier to deal with.. |
| 19:24.44 | gputer | ok great.. so i was somewhat on the right manner |
| 19:24.52 | gputer | i mean right direction |
| 19:24.57 | gputer | sorry just a tad bit tired today |
| 19:25.00 | brookshire | now if you have 5 or more people at a location, you might want to put an asterisk box there |
| 19:25.05 | gputer | but have been racking my brains for the lst 5 days on this |
| 19:25.14 | brookshire | that's just my suggestion |
| 19:25.18 | gputer | ok great |
| 19:25.23 | gputer | broookshre you are right in a manner |
| 19:25.39 | gputer | see i dont' want to connect any of this infrastructure to pots or to isdn or any other telco |
| 19:25.53 | gputer | i just want one office to be able to call each other using their internet connection at the lowest possible cost |
| 19:26.36 | brookshire | gputer: while you are doing that, i would probably centralize all of your long distance |
| 19:26.49 | brookshire | :) |
| 19:26.51 | *** join/#asterisk DerekD (n=Derek@216.222.31.184) |
| 19:27.04 | gputer | brookshire.. that is a good idea |
| 19:27.11 | gputer | but there are legal issues in india.. which i dont' want to get into |
| 19:27.15 | gputer | right now.. it is very simple |
| 19:27.19 | gputer | all my senior team sits in delhi |
| 19:27.31 | gputer | and i have small tech / marketing / sales people across the country |
| 19:27.40 | gputer | each of them have a local city number.. which they use with customers |
| 19:27.53 | gputer | but we have a lot of calls going in between offices at this time.. as we are still small and tight on resources |
| 19:28.08 | arkonadev | is there a manager action to request all the queues? |
| 19:28.25 | brookshire | arkonadev: yes |
| 19:28.42 | arkonadev | what's the action called? |
| 19:28.49 | brookshire | but that's the extent of my knowledge there |
| 19:28.50 | brookshire | :) |
| 19:29.04 | arkonadev | lol |
| 19:29.35 | arkonadev | anybody have an idea? |
| 19:30.10 | brookshire | arkonadev: i've mainly just played around with http-manager |
| 19:30.17 | brookshire | there is a demo in that, i believe, for queues |
| 19:30.52 | arkonadev | ok im playing with the asterisk-java.org Manager API and I am not seeing an action to do it |
| 19:31.19 | Xen^ | mog : arround ? |
| 19:32.06 | brookshire | it also matters how you setup queues |
| 19:32.46 | arkonadev | all the queues have been setup through a web interface |
| 19:32.54 | arkonadev | were using a fonality asterisk system |
| 19:33.18 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 19:34.20 | *** join/#asterisk Dr-Linux (n=nono@202.59.73.131) |
| 19:35.19 | mog | hey xnon |
| 19:35.22 | mog | er Xen^ |
| 19:35.31 | mog | i think i know whats wrong in your bt |
| 19:35.31 | xnon | hey |
| 19:35.33 | xnon | hello |
| 19:35.36 | mog | but im not sure how it happened |
| 19:35.38 | Xen^ | hey mog |
| 19:35.40 | mog | can you do bt full |
| 19:35.42 | mog | on that core |
| 19:35.56 | Xen^ | then you need to give me one more dayu |
| 19:35.57 | Xen^ | :) |
| 19:35.58 | Xen^ | heeh |
| 19:36.01 | Xen^ | or wait |
| 19:36.20 | xnon | hey friends anybody know about trixbox system? |
| 19:36.28 | *** join/#asterisk jsmith (n=jsmith@h460507c2.area3.spcsdns.net) |
| 19:36.29 | sevard | ~trixbox |
| 19:36.31 | jbot | extra, extra, read all about it, trixbox is NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 19:36.34 | xnon | yes! |
| 19:37.01 | sevard | yes. |
| 19:37.05 | *** part/#asterisk clive- (n=pirch@dsl-242-165-63.telkomadsl.co.za) |
| 19:37.16 | *** part/#asterisk jsmith (n=jsmith@h460507c2.area3.spcsdns.net) |
| 19:38.03 | zx6r | any of you guys ever use the [applicationmap |
| 19:38.07 | zx6r | ] section of features.conf? |
| 19:39.49 | b11d|bbl | pffts |
| 19:39.53 | b11d|bbl | its easier to ask "who doesnt" |
| 19:40.01 | b11d|bbl | ok that was a lie.. |
| 19:41.00 | gputer | b11d bbl / brook.. can i continue ? |
| 19:41.14 | *** part/#asterisk TmBerg (n=TmBerg@pdpc/supporter/basic/TmBerg) |
| 19:41.24 | zx6r | hahaha |
| 19:42.27 | b11d|bbl | im gone |
| 19:42.28 | b11d|bbl | sorry |
| 19:44.23 | gputer | hmm |
| 19:44.24 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 19:44.28 | gputer | brookshire ? |
| 19:45.00 | *** join/#asterisk marbahlarbs (n=marbahla@74.132.136.242) |
| 19:45.07 | bluregard | is there a variable to grab the calling extension from extensions.conf rather than grabbing the name from sip.conf like ${CALLERID(num)} does? |
| 19:45.28 | Xen^ | mog : strange now its not crashing |
| 19:45.32 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:45.34 | mog | hehe |
| 19:45.35 | Xen^ | mog : but call is not going |
| 19:45.51 | Xen^ | mog : from sip to gtalk its not going |
| 19:45.52 | Xen^ | <PROTECTED> |
| 19:45.56 | Xen^ | [Nov 14 22:46:48] WARNING[21518]: chan_sip.c:1855 retrans_pkt: Maximum retries exceeded on transmission M2IxNDAwMjY4MzE5OTNiNDBiMDkyY2ZmNWE4M2VjMGE. for seqno 2 (Critical Response) |
| 19:45.57 | Xen^ | [Nov 14 22:46:48] WARNING[21518]: chan_sip.c:1872 retrans_pkt: Hanging up call M2IxNDAwMjY4MzE5OTNiNDBiMDkyY2ZmNWE4M2VjMGE. - no reply to our critical packet. |
| 19:47.00 | marbahlarbs | Lame question: what hardware would I need if I want to use two asterisk servers as a transparent bridge between an analog phone and my phone switch over an IP network? |
| 19:47.05 | dlynes_laptop | bluregard: can you reword your question? It's not making much sense...${CALLERID(num)} is grabbing the information from the channel driver, not sip.conf |
| 19:47.22 | Xen^ | mog : any idea ? |
| 19:47.49 | dlynes_laptop | marbahlarbs: phone switch == keysystem/pbx? |
| 19:48.03 | marbahlarbs | dlynes: yes |
| 19:48.12 | dlynes_laptop | marbahlarbs: i.e. you want to make asterisk an overglorified ata? |
| 19:48.48 | dlynes_laptop | marbahlarbs: you need something that'll give you at least one fxs port and at least one fxo port |
| 19:48.48 | bluregard | dlynes_laptop: well in sip.conf I have a user called [vanna] who has extension => 100 in extensions.conf. When I use ${CALLERID(num)} it try's to supply "vanna" rather than "100" like I need it to. |
| 19:49.17 | marbahlarbs | basically we're connecting another building to my phone network and there's fiber but no copper so I need to get the phones hooked up as cheaply as possible. Next year we're going all voip |
| 19:49.21 | bluregard | if I change [vanna] to [100] in sip.conf it works as expected. |
| 19:49.47 | dlynes_laptop | bluregard: Set(CALLERID(num)=100) |
| 19:50.41 | dlynes_laptop | marbahlarbs: ah...so there'll be a lot more than one analog phone then, right? |
| 19:51.00 | marbahlarbs | there will be two phones |
| 19:51.15 | bluregard | dlynes: it says in the docs that ${CALLERID()} is built-in and can't be set only read in the dial plan. |
| 19:51.45 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 19:51.46 | dlynes_laptop | bluregard: your docs are wrong |
| 19:51.47 | bluregard | ooh, nevermind. I misread that. |
| 19:51.50 | marbahlarbs | of course, there's nothing stopping them from one day saying there will be 10 phones |
| 19:51.52 | Zeeek | ah |
| 19:52.03 | marbahlarbs | heh |
| 19:52.19 | dlynes_laptop | marbahlarbs: ok, so you've got several phones in the other building, right? |
| 19:52.21 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 19:52.22 | bluregard | yeah, I was looking at the line below it for the depreciated ${CALLERID}, sorry. |
| 19:52.28 | dlynes_laptop | marbahlarbs: just only two will be getting hooked up initially? |
| 19:52.29 | bluregard | thank you |
| 19:52.35 | dlynes_laptop | bluregard: no problem |
| 19:52.54 | marbahlarbs | dlynes: correct, there is office space for several people, but only two are slated for phones |
| 19:53.24 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 19:53.32 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 19:54.10 | dlynes_laptop | marbahlarbs: I take it you don't anticipate 10 people or more before you go pure voip though, right? |
| 19:54.23 | dlynes_laptop | marbahlarbs: or even much more than two people? |
| 19:54.27 | marbahlarbs | correct |
| 19:54.39 | dlynes_laptop | marbahlarbs: Yeah...I would just get an analog card then |
| 19:54.49 | dlynes_laptop | marbahlarbs: like a digium tdm400p, or a sangoma a200 |
| 19:55.02 | marbahlarbs | awesome, thanks for your help =) |
| 19:55.06 | dlynes_laptop | marbahlarbs: or go with a tdm2400/sangoma a200d |
| 19:55.23 | dlynes_laptop | marbahlarbs: the latter will give you echo cancellation which you'll probably need for a loop that long |
| 19:55.25 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 19:55.29 | wwalker | I'm getting "cause not handled" from a server that's trying to make a call through the other server. the "other" server can both send and receive calls fine thru our SIP provider. I can't figure out why the "other" server is rejecting the call. |
| 19:55.42 | wwalker | http://rafb.net/paste/results/M7NOAU64.html is the sip debug info from both sides |
| 19:55.44 | dlynes_laptop | marbahlarbs: i.e. hardware echo can |
| 19:56.06 | marbahlarbs | dlynes: excellent |
| 19:56.12 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 19:57.10 | dlynes_laptop | marbahlarbs: just make sure you get at least as many fxo ports and at least as many fxs ports as you need phones |
| 19:57.48 | dlynes_laptop | marbahlarbs: if you anticipate more than two phones before you go pure voip, and you want to go with digium, make sure you get the tdm2400p, so that you have growing room |
| 19:58.04 | dlynes_laptop | marbahlarbs: the tdm400p only goes to a maximum of four ports |
| 19:58.28 | dlynes_laptop | marbahlarbs: the sangoma a200/a200d and the digium 2400p all have capacity for up to 24 ports |
| 19:58.30 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:58.31 | marbahlarbs | dlynes: perfect |
| 19:58.51 | Dr-Linux | dlynes_laptop: a few sipura's or channel bank is not good for that setup? |
| 19:59.02 | marbahlarbs | dlynes: would that mean 12 fxo and 12 fxs for a total or 12 lines? |
| 19:59.08 | Dr-Linux | i heard tdm2400p is not very good |
| 19:59.11 | dlynes_laptop | marbahlarbs: correct |
| 19:59.11 | marbahlarbs | dlynes: total of |
| 19:59.27 | marbahlarbs | dlynes: thanks very much for your help |
| 19:59.53 | dlynes_laptop | Dr-Linux: I don't think he's going to be looking at something that would need enough channels to warrant a channel bank...besides he's switching to pure voip soon |
| 20:00.29 | Dr-Linux | i see |
| 20:00.37 | dlynes_laptop | Dr-Linux: the sipuras are also too expensive on a per port basis for the quality they offer compared to digium 2400p's/sangoma a200(d)'s |
| 20:01.04 | dlynes_laptop | Dr-Linux: with the hardware echo cancellers on those cards they're miles better than the sipuras |
| 20:01.27 | ManxPower | pure voiip is a bad idea |
| 20:01.31 | dlynes_laptop | Dr-Linux: the only benefit the sipuras have is g729/723 codec built in |
| 20:02.14 | Dr-Linux | dlynes_laptop: cool, thanks for the nice info |
| 20:02.16 | dlynes_laptop | Dr-Linux: those wall warts take up a minimum of three power ports for the older sipuras because of their poor design, and up to two for the new design |
| 20:02.35 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 20:02.36 | dlynes_laptop | Dr-Linux: if you have vertically stacked power ports though, the new ones only take up one power port |
| 20:02.39 | Dr-Linux | dlynes_laptop: did you try 1.4 beta? |
| 20:02.52 | dlynes_laptop | Dr-Linux: I'm going to be trying it out at home and at the office |
| 20:03.25 | Dr-Linux | hhm.. |
| 20:03.28 | dlynes_laptop | Dr-Linux: the office is kind of up in the air at the moment though, so I don't want to waste my time on that right now |
| 20:03.55 | Dr-Linux | actually i would like to know some extra good things in 1.4beta before installing it |
| 20:04.06 | Dovid | i currently have monitor-format=gsm in queues.conf to record the calls but it records the call's as 2 files one of the caller and one of me, how do i have asterisk record the entire convo in one file |
| 20:04.16 | dlynes_laptop | Dr-Linux: there's a lot of good reasons to use 1.4 over 1.2 |
| 20:04.30 | dlynes_laptop | Dovid: MixMonitor() |
| 20:04.48 | dlynes_laptop | Dovid: Hopefully MixMonitor() will be more stable in 1.4...it's quite unstable in 1.2 |
| 20:04.50 | Dovid | so instead of |
| 20:04.50 | Dovid | monitor-format=gsm |
| 20:04.52 | Dr-Linux | dlynes_laptop: can you please explain a few .. major things? |
| 20:05.03 | Dr-Linux | in the light of production setup |
| 20:05.05 | Dovid | i do minmonitor-format=gsm ? |
| 20:05.18 | Dovid | mixmonitor-format=gsm |
| 20:05.19 | wwalker | http://rafb.net/paste/results/j8VJbx46.html is excerpts from my sip and extensions files. any ideas why it's rejecting the call as show in http://rafb.net/paste/results/M7NOAU64.html sip debug? |
| 20:05.27 | dlynes_laptop | Dovid: oh...sorry...I missed the part where you said it's for queues |
| 20:05.31 | Dovid | lol |
| 20:05.47 | dlynes_laptop | Dovid: MixMonitor is for individual call setup |
| 20:05.53 | dlynes_laptop | Dovid: not for a call center |
| 20:05.55 | Dovid | i know |
| 20:06.02 | Dr-Linux | is their any new things in 1.4beta for queues/agents/applications |
| 20:06.06 | Dovid | so i can just record b4 it goes to the queue |
| 20:06.15 | dlynes_laptop | Dr-Linux: for applications, definitely |
| 20:06.22 | Dovid | thanks |
| 20:06.24 | dlynes_laptop | Dr-Linux: I don't know about hte other stuff though, because i dno't use it |
| 20:06.41 | dlynes_laptop | Dr-Linux: so i haven't been tracking anything for queues/agents |
| 20:06.51 | Dr-Linux | hhm.. |
| 20:07.10 | Dr-Linux | i think i should wait for 1.4 stable |
| 20:07.22 | dlynes_laptop | Dr-Linux: for your production system? |
| 20:07.47 | dlynes_laptop | Dr-Linux: why not set up a 1.4 system for your personal phone in the company? |
| 20:08.01 | dlynes_laptop | Dr-Linux: that way if it crashes at least your boss can't get a hold of you :) |
| 20:08.24 | wwalker | I have debug and verbose set to 30 but don't know how to see why asterisk is rejecting the call. I assume it's a dialplan match failure, but how can _1XXXXXXXXXX not match 15556461234? |
| 20:08.55 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 20:08.57 | bXi | doesnt _ match something as well? |
| 20:09.36 | ManxPower | _ means "the following is a pattern match" |
| 20:09.59 | ManxPower | wwalker: chances are something more specific is matching 5556461234 or you have a context problem |
| 20:10.59 | wwalker | it goes into the context specified by the sip.conf, right? |
| 20:11.27 | dlynes_laptop | wwalker: assuming it's coming from a sip device, yes |
| 20:11.45 | *** join/#asterisk gatuno (n=gatuno@250.Red-81-32-126.dynamicIP.rima-tde.net) |
| 20:12.00 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 20:12.04 | dlynes_laptop | wwalker: if you want help, it's better to post your extensions.conf file in its entirety |
| 20:12.13 | wwalker | dlynes_laptop: it is. |
| 20:12.27 | dlynes_laptop | wwalker: above you said it was excerpts |
| 20:12.59 | tao2 | thanks for the help need to go and sleep |
| 20:13.25 | wwalker | dlynes_laptop: "it is" referred to "coming from a sip device" |
| 20:13.49 | dlynes_laptop | ah |
| 20:13.59 | dlynes_laptop | thought you were talking about your extensions.conf file |
| 20:14.26 | networ | have anyone experince with h323 on asterisk? |
| 20:14.47 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 20:16.01 | djflux | anyone tried using Ekiga (or any SIP client for that matter) with asterisk 1.4 beta? |
| 20:16.42 | djflux | DTMF during a call doesn't seem to work (unless I have something configured incorrectly - which could very well be the case :) ) |
| 20:16.46 | *** part/#asterisk InfraRed (n=bigboss@198.63.211.131) |
| 20:17.42 | *** join/#asterisk saftsack (n=oliver@p54A7C5B7.dip.t-dialin.net) |
| 20:17.51 | saftsack | does someone of you use nokias e60? |
| 20:18.44 | ManxPower | djflux: does it work with 1.2.13? |
| 20:20.13 | djflux | ManxPower: yes |
| 20:21.14 | *** part/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
| 20:21.24 | djflux | I can see the DTMF 2 being sent in a wireshark packet capture but if I press 2 which should go to the "more technical information about Asterisk" it doesn't start playing that part of the demo |
| 20:25.14 | ManxPower | I guess that is why they call 1.4 BETA |
| 20:29.31 | wwalker | dlynes_laptop: ManxPower: I see that it's probably a match in a different context getting called. What do I need to look at to understand why the context specified in the sip friend is not the one being used? |
| 20:30.22 | wwalker | I can post the extensions.conf and the sip.conf but I want to understand the rules for the context used for an incoming SIP call |
| 20:30.45 | ManxPower | the incoming call is not matching the corrrect [deviceentry] |
| 20:31.11 | ManxPower | That's why I always put context=INVALID in sip.conf [general] and then put the correct context= line for each sip.conf entry |
| 20:31.11 | wwalker | what criteria are used to do that match? |
| 20:31.22 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 20:31.28 | ManxPower | wwalker: That is dark magic that nobnody understands |
| 20:31.41 | wwalker | ManxPower: that is what I fear |
| 20:31.47 | ManxPower | but for incoming calls [this] would expect the incoming call to come with a username [this] |
| 20:31.54 | wwalker | "the Windows component" |
| 20:31.54 | ManxPower | ..er.. "this" |
| 20:32.12 | ManxPower | but you should be seeing the errors on the console |
| 20:32.16 | wwalker | <PROTECTED> |
| 20:32.32 | wwalker | no errors save what is in that paste message |
| 20:32.46 | wwalker | (http://rafb.net/paste/results/M7NOAU64.html) |
| 20:33.43 | ManxPower | that looks to me like an incoming call, not an outgoing call |
| 20:35.04 | ManxPower | wwalker: is this a call made from a SIP phone? |
| 20:35.05 | wwalker | top half is the server making the call. (call-test) bottom half is the server (not) accepting the call |
| 20:35.26 | ManxPower | wwalker: no, the bottom half is asteirsk not getting a response from the server |
| 20:35.30 | ManxPower | now WHAT part is not working? |
| 20:35.30 | *** join/#asterisk saftsack (n=oliver@p54A7C5B7.dip.t-dialin.net) |
| 20:35.32 | wwalker | ManxPower: no, call is being made via the Manager on call-test |
| 20:35.49 | ManxPower | wwalker: you are trying to make things complicated aren't you? |
| 20:35.57 | ManxPower | wwalker: I don't even see a Dial line in the output |
| 20:36.39 | hohum | http://cgi.ebay.com/ebaymotors/ws/eBayISAPI.dll?ViewItem&viewitem=&item=300047752494 |
| 20:37.05 | ManxPower | hohum: there is 30 seconds of my life I'll never get back. |
| 20:37.25 | hohum | ManxPower: well you might as well just end it all now, then, eh? |
| 20:38.08 | wwalker | first 3 lines. an external manager connects and attempts the call on call-test. call-test instantly spews that sip debug. as does cuervo (the upstream Asterisk server that then should forward the call on to our VoIP provider (cuervo is making and receiving calls thru them fine) |
| 20:38.40 | ManxPower | wwalker: I don't deal with sip debug since the problem is almost always something else. |
| 20:39.01 | ManxPower | Traditional Dial(SIP/${EXTEN}@sipconfentry) |
| 20:39.14 | hmmhesays | has anyone gotten chan_gtalk to work in asterisk 1.4 yet? |
| 20:39.46 | Juggie | Manx, isnt it Dial(SIP/peer/exten) |
| 20:40.26 | hmmhesays | Juggie: there are various allowed syntax' |
| 20:40.34 | Juggie | ic. |
| 20:41.09 | wwalker | I'm going to delete 95% of the crap in sip.conf and extensions.conf from the previous admin. |
| 20:44.51 | froguz | two weeks ago i put on sip.conf: disallow=all|allow=g729|allow=ulaw and just ulaw in a linksys PAP2, when i tried to make a call it failed because of codecs |
| 20:44.51 | wwalker | in a devicemap entry of type user, is the context the context used locally or the context requested on the peer when a call is placed? |
| 20:45.14 | froguz | shouldn't asterisk select the negotiate the second codec (ulaw)? |
| 20:45.58 | ManxPower | Juggie: that never seems tow ork |
| 20:47.39 | froguz | it just find that there wasn't g729 and give me channunavailable |
| 20:48.39 | ManxPower | froguz: asterisk thinks it can passtru g729 |
| 20:48.47 | ManxPower | that is why I never allow more than 1 codec for a phone |
| 20:50.02 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 20:50.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:50.51 | froguz | ManxPower, and when it find there's no transcoder license just close the channel instead trying to renegociate the codec? |
| 20:50.54 | *** join/#asterisk alexcti (n=chatzill@adsl-074-247-240-219.sip.mia.bellsouth.net) |
| 20:51.16 | *** join/#asterisk kph100 (n=kph100@206-248-157-22.dsl.teksavvy.com) |
| 20:52.11 | froguz | if that were the case, is an interesting improvement to report |
| 20:52.46 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 20:52.50 | iCEBrkr | Homies! |
| 20:53.10 | froguz | can i dynamically select the codec priority for a specific call? |
| 20:53.11 | ManxPower | froguz: correct. |
| 20:53.40 | ManxPower | froguz: you cannot dynamically select codec for INCOMING call. You can for OUTGOING calls (before the Dial) see README.variables |
| 20:54.02 | froguz | cool, that's what i need |
| 20:54.37 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 20:56.01 | kph100 | in my asterisk setup, no cdf records are generated. do i need to enable cdr in a conf file? |
| 21:06.34 | wwalker | is there any way to get asterisk to tell me what device was matched on an incoming call? which context? |
| 21:08.50 | Juggie | wwalker, yes. |
| 21:09.13 | Juggie | set the incomming context on your peer/channel so you know where it came from. |
| 21:10.19 | file | there is also setvar |
| 21:10.43 | Alric | Setvar is yummy. Whoever made that is a genius. |
| 21:10.57 | ManxPower | wwalker: look at README.variables and use Noop to print that info on the console |
| 21:12.21 | wwalker | Juggie: I'm setting it at both ends and it seems to not be using that context, that's why I want asterisk to tell me what the hell it thinks the context is (and what it thinks the device is |
| 21:12.34 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:13.21 | ManxPower | wwalker: Noop(Context is ${CONTEXT}) |
| 21:13.27 | ManxPower | it is not rocket science. |
| 21:14.36 | *** join/#asterisk jm|work (n=jamiem@zen.jamiem.com) |
| 21:15.36 | ManxPower | wwalker: you did my suggestion of context=INVALID in sip.conf [general]? |
| 21:21.37 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 21:21.51 | *** join/#asterisk Peri (n=redanti@xtreme-3-150.dyn.aci.on.ca) |
| 21:25.06 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 21:25.09 | *** join/#asterisk cryptnix- (n=andrew@207.241.137.70) |
| 21:26.05 | hoobastooba | I am using the polycom phones... latest firmware upgrade has left me with some annoying issues. has anyone else ever experienced where after a firmware update the phones act like the keys are stuck, unresponsive or slow? |
| 21:26.20 | ManxPower | hoobastooba: known issue with 2.0.x |
| 21:26.42 | ManxPower | "known issue" = discussed on the mailinglists. |
| 21:27.12 | hoobastooba | comforting... |
| 21:27.44 | ManxPower | downgrade to the latest 1.6.x and don't worry about it./ |
| 21:28.10 | hoobastooba | cool... sp.ld and my sip.cfg files is all, right? |
| 21:28.40 | ManxPower | sip.ld |
| 21:28.43 | ManxPower | but yes |
| 21:29.29 | hoobastooba | fat fingers ;) |
| 21:29.55 | Peri | what's standard proceedure to log into a queue? |
| 21:30.22 | hoobastooba | standard procedure? |
| 21:30.28 | hoobastooba | depends on how you want it to work |
| 21:30.43 | hoobastooba | are you logging in from the phone its self? or some other way ie AGI? |
| 21:31.17 | cryptnix- | I'm having a small problem with people saying that they're having problems calling me ... says the service has been disconnected or its being checked for problems i suspect its related to updating to much or to little w/ my provider ... any ideas? |
| 21:32.01 | hoobastooba | i wish i could help you... i have the same issue with one company and have not resolved it. |
| 21:32.01 | hoobastooba | Peri: ? |
| 21:32.46 | Peri | logging in from the phone it self |
| 21:33.56 | hoobastooba | there are three different ways that I know of... one is AddQueueMember... This is more dynamic. the other two rely on agents.conf and that is AgentLogin and AgentCallbackLogin. AgentCallbackLogin works exactly how i want it to, but causes alot of issues for my regarding stability, so i use addqueuemember. AgentLogin makes the agent stay off hook listening to hold music, and whey they hang up it logs them out. |
| 21:34.09 | hoobastooba | go to the wiki and look up agents |
| 21:34.24 | hoobastooba | many variations... thats why i kind of laughed at standard. |
| 21:34.29 | hoobastooba | very flexable. |
| 21:34.34 | hoobastooba | flexible |
| 21:34.43 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 21:34.52 | Peri | *nods* |
| 21:34.58 | Peri | understood |
| 21:35.06 | Peri | agentcallbacklogin looks to be the direction i'll go in |
| 21:35.27 | hoobastooba | *warning... it has some problems. |
| 21:35.52 | Peri | oh? |
| 21:36.30 | hoobastooba | my queues would periodically either crash or lock up completely causing me to have to restart asterisk... not good in production environment. much happier with addqueuemember. |
| 21:37.02 | hoobastooba | i add Local/<extension number>/n to the queue and it works awesome for me. |
| 21:37.24 | ManxPower | agentcallbacklogin MIGHT be removed in 1.4 and certinally will be in 1.6 |
| 21:38.35 | wwalker | ManxPower: so i would have to add "Noop(Context is ${CONTEXT})" to every context/exten match in the file until I find it? (that's a LOT of places) |
| 21:38.45 | Alric | ManxPower: Because the same thing can be accomplished with addqueuemember? |
| 21:39.16 | Peri | ok |
| 21:39.20 | Peri | good to know |
| 21:39.25 | wwalker | and renumber every dial plan? |
| 21:39.26 | Peri | so i'll have to look up addqueuemember :) |
| 21:40.11 | wwalker | or is there something that gets run before the switch to the context? |
| 21:41.18 | hoobastooba | Alric: addqueumember and agentcallbacklogin are very different. |
| 21:41.36 | hoobastooba | you loose alot of the features of the agents.conf file by using addqueuemember. |
| 21:41.47 | hoobastooba | agentcallbacklogin is very buggy and unstable |
| 21:41.54 | hoobastooba | that is the reason for the removal |
| 21:42.15 | Peri | not a whole lot of documentation to addqueuemember for a noob like me >.< |
| 21:42.29 | hoobastooba | hang on... |
| 21:42.40 | hoobastooba | its actually very simple |
| 21:42.58 | ucfMethod | These polycom 501s dont play nice... anyone know how to force a file system format.. i read voip-info but my 501's dont have Menu | Settings | Advanced? |
| 21:43.11 | hoobastooba | http://www.voip-info.org/wiki-Asterisk+cmd+AddQueuemember |
| 21:43.37 | hoobastooba | <PROTECTED> |
| 21:43.51 | hoobastooba | they play great!!! they are very easy to use in my opinion |
| 21:44.32 | hoobastooba | Peri: all you would do is something like this... exten => 1000,1,AddQueueMember(queuename[|interface][|penalty]): |
| 21:44.33 | *** join/#asterisk mhart (n=chatzill@cowboy.intuit.com) |
| 21:45.38 | hoobastooba | AddQueueMember(helpdesk|Local/6000|1) |
| 21:45.38 | hoobastooba | <PROTECTED> |
| 21:45.58 | hoobastooba | then to remove them it is removequeuemember |
| 21:46.02 | hoobastooba | check that link i sent |
| 21:46.20 | hoobastooba | ucfMethod:? |
| 21:47.13 | ucfMethod | Well, i have a interesting situation. I currently have an asterisk box on 192.168.30.3 which is working just fine and has all 20+ employees using it... I have created another box, .30.11 which has the latest Asterisk version installed, and I want 2 particular phones to using that server instead. I have all the user/passwords created correctly on the new box, but I can't get the phones to use this new FTP server instead of the old one |
| 21:47.56 | hoobastooba | have you set the server settings on the phone to connect to the new server? |
| 21:48.00 | ManxPower | ucfMethod: Just change the Cisco phone config. |
| 21:48.01 | ucfMethod | they just boot up and log into .3 and grab the .cfgs and boot properly. I have tried changing the Server Settings in the menu, but after a reboot it still just grabs the old stuff |
| 21:48.19 | hoobastooba | works fine for me |
| 21:48.40 | kph100 | anyone knows why I get: cdr_odbc: Error in Query -1 ? |
| 21:48.45 | hoobastooba | all i have to do is make sure that the files on the new server are different from the old ones so that it knows to download them |
| 21:48.53 | wwalker | if the server is rejecting the call, then where could I put a Noop that would get run? |
| 21:49.09 | mhart | I've setup Asterisk w/Gizmo - purely software right now - works fine for incoming as well as outgoing. I'm new to this telecom stuff. However, I can't detect an answered line. I understand that this is because the SIP channel I'm using is analog (thanks Manx). Would Digium or Pika or some other hardware give me some kind of call progress/answer detection? Or do I need a different SIP... |
| 21:49.11 | mhart | ...provider or something? |
| 21:49.19 | ucfMethod | I dont see anything in /var/log/messages showing that the phones are logging into the new server |
| 21:49.28 | ManxPower | wwalker: you would not. If the server is rekjecting the call then there is no context |
| 21:50.17 | ManxPower | mhart: no, SIP is not analog. |
| 21:50.17 | ucfMethod | hoobastooba: do you know a keyboard command ( similar to 4, 6, 8, * ) which will hard core wipe these phones? |
| 21:50.17 | ManxPower | you did not tell me that it was purely software. I assumed you were using a gizmo device with an analog port. |
| 21:50.42 | ManxPower | mhart: it is the gateway that has to signal answer or not. the problem is in your PSTN gateway. |
| 21:50.51 | wwalker | so you didn't read the problem description at all before answering. Thanks for wasting my time |
| 21:51.18 | mhart | Manx - Not using a gizmo device - using sip.conf to connect to the sipphone.com gateway |
| 21:51.39 | ManxPower | mhart: then the issue is with sipphone.com. |
| 21:52.13 | mhart | Manx - thanks - so maybe a different SIP provider will get the call progress stuff to Asterisk? |
| 21:52.26 | ShadwDrgn | does the newest version of asterisk not support supervised call transfer? |
| 21:52.32 | ManxPower | wwalker: I answered your question and was too busy to figure out if you were confused about it. |
| 21:53.17 | ManxPower | mhart: I'm sure sipphone.com provides correct answer supervision for PSTN calls. I suspect its an issue with whatever gizmo gateway they are using. Contact them. |
| 21:54.07 | mhart | ManxPower: Okay - although Gizmo support is really bad - they seem to be clueless when it comes to Asterisk. |
| 21:54.31 | hoobastooba | ucfMethod: sorry, not off the top of my head |
| 21:54.39 | *** join/#asterisk zumbush (n=zumbush@c-2cbce253.038-31-73766c10.cust.bredbandsbolaget.se) |
| 21:56.20 | CunningPike | ucfMethod: Does 468* not work? |
| 21:56.29 | wwalker | ManxPower: no, you didn't. |
| 21:56.34 | wwalker | and I wasn't |
| 21:57.33 | ManxPower | I won't recall seeing your answer about my question about context=INVALID |
| 21:59.22 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:00.20 | ShadwDrgn | :( looks like i'm running asterisk 1.0.11.1 |
| 22:00.25 | ShadwDrgn | so i might not have attended transfering |
| 22:00.54 | ManxPower | ShadwDrgn: 1.0.X does not have DTMF transfers. Phone based or Zap based transfer should work just fine |
| 22:01.24 | ShadwDrgn | gentoo doesn't have an unmasked package for 1.2 |
| 22:01.25 | ShadwDrgn | lol |
| 22:01.35 | ShadwDrgn | i would have to upgrade and i don't feel like dealing with that. |
| 22:02.28 | ManxPower | ShadwDrgn: you could buy a phone that supports transfers |
| 22:02.39 | ShadwDrgn | yeah |
| 22:03.20 | ShadwDrgn | know any good wifi phones that support it? :D |
| 22:03.40 | ManxPower | All WiFi phones suck,. |
| 22:03.43 | ShadwDrgn | LOL :D |
| 22:03.53 | ShadwDrgn | there has to be a good one somewhere |
| 22:04.00 | Dr-Linux | what phone we call Wifi phones? :S |
| 22:04.02 | ShadwDrgn | otherwise this whole project was pretty much pointless |
| 22:04.07 | Dr-Linux | ManxPower |
| 22:04.10 | ShadwDrgn | for me |
| 22:04.39 | ManxPower | Dr-Linux: We call them "Not ready for production use" |
| 22:04.41 | ucfMethod | hoobastooba: it appears that its doing what I want, but the phones reboot and just come up with their old configs.. I will keep searching google... Thanks again. |
| 22:05.35 | Dr-Linux | ManxPower: yeah but what's the difference between Wifi and other phones? i mean Wifi is a genaric name or what |
| 22:05.35 | ManxPower | ucfMethod: are their old configs on the old server? |
| 22:05.55 | ManxPower | Dr-Linux: WiFi = 802.11a/b/g |
| 22:06.22 | ucfMethod | ManxPower: Yeah, I guess I could temp shutdown vsftpd on that old server so there is no possible way for them to grab the files, but I think I would see something in /var/log/messages |
| 22:06.33 | Dr-Linux | that's wireless |
| 22:06.37 | ManxPower | ucfMethod: they are prolly getting their bootserver stuff via DHCP |
| 22:07.07 | Dr-Linux | 802.11 stuff are wireless |
| 22:07.09 | ucfMethod | ManxPower: good point, crap, didnt think to look there.. actually its the same machine, the DHCP server and Asterisk... sneaky |
| 22:07.28 | ManxPower | ucfMethod: I recommend leave the config files on the old server and just change them |
| 22:07.40 | ManxPower | when you are ready to really migrate, move all the config files and change the DHCP stuff |
| 22:07.50 | ManxPower | Dr-Linux: correct. |
| 22:08.02 | ucfMethod | ManxPower: good looking out, in dhcp.conf on the old machine was this line "option tftp-server-name "192.168.30.3";" |
| 22:08.12 | ucfMethod | ManxPower: sneaky DHCP |
| 22:08.22 | ucfMethod | THANKS ALL |
| 22:08.34 | ManxPower | ucfMethod: if you change it your existing phones will try to download their configs from the new server as soon as they renew their lease |
| 22:09.20 | ucfMethod | ManxPower: I guess I can turn off DHCP on these 2 special phones, and manually set things. |
| 22:09.31 | Peri | home time |
| 22:09.33 | *** part/#asterisk Peri (n=redanti@xtreme-3-150.dyn.aci.on.ca) |
| 22:13.11 | *** join/#asterisk sjobeck (n=sjobeck@66-224-185-167.atgi.net) |
| 22:14.10 | ShadwDrgn | x-lite supports transfer |
| 22:14.13 | ShadwDrgn | but the button is greyed |
| 22:14.46 | ShadwDrgn | oh i see you have to upgrade for it. lol |
| 22:17.26 | [TK]D-Fender | ShadwDrgn : Also known as "X-Pro" |
| 22:17.46 | [TK]D-Fender | ShadwDrgn : Or "Taunts with Transfer" |
| 22:18.06 | Alric | Ugh. Ports 1-3 work on my TDM card. 4 doesn't pick up :( |
| 22:18.38 | [TK]D-Fender | ShadwDrgn : Add my opinion to that of ManxPower's in the "WiFi phones suck. All of them. HARD. And without protection" |
| 22:20.08 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-155-20-221.red.bezeqint.net) |
| 22:22.57 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 22:24.45 | mercestes | slightly off topic: Is there a way/law that I can force a carrier to provide me CDR's in some form of text parseable format instead of via paper billing or pdf snapshot? |
| 22:26.18 | ManxPower | mercestes: the best way is to switch to a carrier that provides that information |
| 22:26.45 | mercestes | ManxPower: That's the plan...lol. |
| 22:27.01 | *** join/#asterisk andresmujica (n=andresmu@201.244.244.253) |
| 22:27.07 | andresmujica | Hello there |
| 22:27.22 | andresmujica | i need to made a short pause between digits while i´m dialing |
| 22:27.39 | ManxPower | andresmujica: what technology? |
| 22:27.46 | andresmujica | zap |
| 22:28.03 | ManxPower | mercestes: or if you want to be evil, call up your sales rep and ask him/her if they know of a company that does provide that information |
| 22:28.16 | mercestes | That is an idea as well. |
| 22:28.37 | ManxPower | andresmujica: Dial(Zap/g1/5551212wwww1234) would wait 2 seconds on mostports except for PRI |
| 22:29.17 | ManxPower | For Voip or PRI ports see "show application dial" and pay special attention to the D() option |
| 22:29.55 | andresmujica | upps... it´s via a PRI |
| 22:30.09 | andresmujica | well E1 to be most accurate |
| 22:30.22 | ManxPower | Channelized voice E-1 or E-1 PRI? |
| 22:30.33 | andresmujica | e1 pri |
| 22:30.43 | ManxPower | then you'll need the D() option |
| 22:31.05 | andresmujica | ok.. thanks MAnx |
| 22:31.37 | ManxPower | On PRI the number is sentallatonce |
| 22:32.17 | andresmujica | the D parms would be wwww ? |
| 22:32.30 | ManxPower | D() supports w |
| 22:32.46 | andresmujica | ok. |
| 22:33.15 | andresmujica | i´ll be reading about it and testing it |
| 22:36.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:38.05 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 22:38.56 | *** join/#asterisk saftsack (n=saftsack@pD9E0509C.dip.t-dialin.net) |
| 22:45.34 | saftsack | has anyone of you a nokia e60ß |
| 22:47.21 | glitch- | 7110 here |
| 22:47.23 | glitch- | heh |
| 22:47.49 | saftsack | does it support wlan? |
| 22:47.59 | glitch- | yeah |
| 22:48.20 | saftsack | oh i see 7110. its a pots telephone *G* |
| 22:48.44 | glitch- | =) |
| 22:49.42 | *** join/#asterisk docelm0 (n=vircuser@c-69-138-91-104.hsd1.de.comcast.net) |
| 22:52.11 | andresmujica | MAnxpower? |
| 22:52.19 | andresmujica | i´m having trouble.. |
| 22:52.35 | andresmujica | i need to send an 8 digit number to another plant so i can get dial tone |
| 22:52.47 | andresmujica | after i get the dial tone form the other plant |
| 22:53.02 | andresmujica | i´ve got to send a 10 digit number (the actual number i want to call) |
| 22:53.13 | andresmujica | plant=pbx sorry.. |
| 22:54.04 | andresmujica | i´m using the D(www) but how can i specify that i need to wait for the dial tone after the first 8 digits? |
| 22:54.43 | Dovid | you can use send dtmf |
| 22:54.44 | Dovid | http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF |
| 22:54.48 | ucfMethod | ManxPower: worked beautifully.. I changed dhcp.conf and now everything automagically worked. Now its time to port my 1.0 Asterisk extensions.conf to 1.2.13 |
| 22:55.18 | *** join/#asterisk matt_ (n=matt@2001:4bd0:2056:1:220:edff:feb4:7c9d) |
| 22:57.09 | andresmujica | senddtmf for sending the 10 digit number (the actual number i want to dial??) |
| 22:58.17 | Dovid | senddtmf |
| 22:58.27 | Dovid | is when u wana send it at the remote location |
| 22:58.44 | Dovid | u want to dial a number |
| 22:58.53 | andresmujica | hmmm... |
| 22:58.55 | Dovid | get a dial tone and then enter another number correct ? |
| 22:58.58 | andresmujica | ok i´ll try that |
| 22:59.10 | Dovid | use the dial command to try the other plant |
| 22:59.23 | Dovid | and then use wait to give time for the other system to pick up |
| 22:59.28 | Dovid | and then use send dtmf |
| 22:59.32 | *** join/#asterisk vexorg (n=vexorg@70.26.139.4) |
| 22:59.33 | andresmujica | yeap |
| 22:59.42 | andresmujica | and i can do that from the agi script |
| 22:59.48 | Dovid | if u want |
| 22:59.51 | andresmujica | so i don´t need a new dialing mcro |
| 23:00.04 | Dovid | any specific reason u r using ago as opposed to regular dial plan ? |
| 23:00.07 | Dovid | agi* |
| 23:01.17 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 23:01.24 | andresmujica | i´m using an agi for autodial getting the numbers from a database |
| 23:01.36 | Dovid | ah ok |
| 23:01.42 | andresmujica | yeap.. |
| 23:02.18 | Dovid | u should be able to use senddtmf from the agi |
| 23:03.16 | andresmujica | i´ll try that thanks a lot!! |
| 23:04.28 | *** join/#asterisk philippel (n=p_lindhe@c-24-16-243-129.hsd1.mn.comcast.net) |
| 23:04.48 | *** join/#asterisk voxtop (n=voxtop@210.9.120.138) |
| 23:06.16 | philippel | hi - anyone know off hand how to get an audiocodes fxo module to detect and send a SIP BYE to asterisk when the pstn initiated caller hangup? I've tried polarity reversal and current disconnect, neither seem to do it - and the line is a stanard ks line that is doing polarity reversal? |
| 23:09.54 | *** join/#asterisk Eonz (n=Icarus@irc.americatelnet.com.pe) |
| 23:09.55 | vader-- | anyone ever link asterisk with nagios in here? |
| 23:10.14 | vader-- | im looking for a way my nagios server can send something to asterisk so asterisk can call me if there is a problem |
| 23:10.58 | andresmujica | vader: i´ve seen some nagios plugins but don´t really know them a lot. but i know for sure there are some asterisk plugins for nagios.. |
| 23:10.58 | *** join/#asterisk monemran (i=monemran@202.52.210.197) |
| 23:11.59 | vader-- | ya |
| 23:13.55 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:14.12 | voxtop | does zap's echo canceller affect sip calls? |
| 23:14.24 | EmleyMoor | I don't suppose any of you has a comprehensive "charge rates and prefixes" list for BT, do you? |
| 23:14.46 | EmleyMoor | voxtop: Are you using a Zap channel for any part of the call? |
| 23:15.15 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 23:15.28 | voxtop | EmleyMoor: no |
| 23:15.36 | EmleyMoor | Shouldn't do then |
| 23:15.55 | voxtop | EmleyMoor: is non-hardware echo cancellation available for sip calls? |
| 23:16.00 | brodiem | Can * change the codec during an active call? i.e. if a call is negotiated as g729 during the sip transaction, and then is realized that the media is a fax, can it change to ulaw? |
| 23:17.09 | voxtop | brodiem: i think so, but don't know how you'd go about it |
| 23:21.52 | *** join/#asterisk brc_ (n=brc___@pdpc/supporter/basic/brc) |
| 23:22.36 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 23:23.01 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 23:23.19 | Druken | evening everyone |
| 23:23.46 | voxtop | morning Druken |
| 23:23.58 | Druken | top o the mornin to ya |
| 23:24.15 | *** join/#asterisk bbblack (n=bblack@129.174.112.186) |
| 23:25.12 | bbblack | hi all. is anybody familiar with canreinvite and nat strategies in asterisk 1.4beta? |
| 23:25.22 | voxtop | when playing with txgain and rxgain in zapata.conf , is it enough to just run "reload" in the * CLI, or does * need to be restarted? |
| 23:25.36 | Druken | restart |
| 23:25.47 | voxtop | thanks |
| 23:26.01 | voxtop | is it enough to restart *, or does the whole box need to be restarted? |
| 23:26.12 | Druken | just asterisk |
| 23:26.13 | EmleyMoor | If I want to support a pulse dial phone internally but tone dial outwards, do I need to specify "pulsedial=yes" in any particular position in zapata.conf? |
| 23:26.47 | Druken | uhmm.... why in hell would you want a pulse dial phone to begin with ? |
| 23:27.17 | EmleyMoor | Druken: I have an old British Yeoman with a nice bell - but it's rotary |
| 23:27.43 | bbblack | I have two SIP phones, both behind nat and nat=yes in sip.conf for them. One has canreinvite=yes, one has canreinvite=no. When I call from the one with canreinvite=yes to the other, after the destination answers it gets reinvited with the SDP containing the local IP of the other NATed device, resulting in audio problems |
| 23:28.32 | bbblack | it's the same if the source is not NATed, since it will start sending RTP traffic to a local IP |
| 23:28.45 | bbblack | any ideas? |
| 23:28.49 | Druken | EmleyMoor: i would assume you could just specify rotarydial on the one device.... ? |
| 23:29.06 | EmleyMoor | So, just where I set up my FXS port(s)? |
| 23:29.18 | JT | bbblack: set them both to canreinvite=no? |
| 23:29.23 | EmleyMoor | Will try that when I get a chance |
| 23:29.29 | Druken | yeah... set it for the specific port that phone will be on |
| 23:29.33 | voxtop | EmleyMoor: search voip-info.org |
| 23:30.00 | bbblack | JT: yes, but if the first phone really can do reinvites, it will block this feature for other destinations that also can do reinvites |
| 23:30.00 | Druken | nat=yes == canreinvite=no :) |
| 23:30.37 | *** join/#asterisk ginvent (n=FreePBX4@adsl-75-36-45-122.dsl.sndg02.sbcglobal.net) |
| 23:30.39 | bbblack | I can have nat=no and canreinvite=yes for the first, call the other and I'll still get the local IP in reinvite |
| 23:30.54 | EmleyMoor | I will use a softphone where I need to send tones - until such time as I start getting hardware IP phones |
| 23:31.24 | EmleyMoor | (I also have a nice tone dial wall Yeoman but being a wall model I would rather have that in the hall) |
| 23:31.31 | ginvent | Anyone know why I can't dial out sipphone? I can dial 411 and 1800 numbers, and i HAVE minutes... but when I dial out regular number, asterisk gets a congested message and I get "all circuits are busy" |
| 23:32.14 | EmleyMoor | ginvent: Presumably your dialplan is set up correctly? |
| 23:32.38 | bbblack | ginvent: it might be that your provider need the calls to come from a certain username in order to bill you correctly. Try add fromuser=... in the sip.conf for your trunk |
| 23:32.50 | ginvent | hmmm... ok. |
| 23:34.17 | voxtop | when i jump into the * console, i use generally use 'asterisk -rvvv'. is it possible to suppress the "Remote UNIX connection" messages? |
| 23:35.13 | EmleyMoor | Are BT's ring cadences documented anywhere? |
| 23:35.42 | bbblack | voxtop: [grep -R "Remote UNIX connection" *] in the asterisk source directory and take them out |
| 23:36.03 | ginvent | Thanks bbblack!!! YOUR MY HERO!!! fromuser was missing... duh. |
| 23:36.57 | EmleyMoor | I've found a document about a Sipura ATA and BT ring cadences - will they translate well into *? |
| 23:37.07 | bbblack | :) only if I could find a developer interested in canreinvite problems :D |
| 23:37.33 | JT | bbblack: it sounds like it really can't do reinvites :P |
| 23:37.54 | bbblack | JT: yes, but asterisk still does them, and it really shouldn't |
| 23:37.58 | *** join/#asterisk HumpBack (n=sdcdcssc@gentoo/developer/HumpBack) |
| 23:38.02 | HumpBack | Hello all |
| 23:38.14 | HumpBack | I've sent some messages to the users list |
| 23:38.22 | bbblack | because you cannot really do it, at that point you don't even know the true RTP source since no RTP packet hit Asterisk yet from the natted device |
| 23:38.24 | JT | i don't see why you set canreinvite to yes |
| 23:38.26 | HumpBack | about the problem with the fxo ports |
| 23:38.32 | HumpBack | on a tdm400 |
| 23:38.59 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 23:38.59 | bbblack | JT: because one part really can do reinvites, and is needed when it is talking with other that can do it |
| 23:39.26 | bbblack | but when the other side is nated and cannot do them, the first still gets reinvited with a local IP when it actually shouldn't |
| 23:40.43 | bbblack | JT: kevin just told me this morning that one side reinvite is correct, so in theory you can have different paths for the two way audio |
| 23:41.09 | EmleyMoor | I got my TDM400 today but they forgot to ship the cables I ordered :-( |
| 23:41.14 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 23:41.21 | EmleyMoor | Still, it's in and appears to be ready |
| 23:41.43 | SplasPood | If I have an asterisk box and one sip phone in contextA dials a number that routes to another number locally hosted on that box (in contextB), right now I keep the calls local, and end up with a single CDR record as if there was only an inbound call to the number in contextB.. I'd like to have two CDRs, one for the outbound "call" from A to B, and another for the "inbound" call from A to B... Possible? |
| 23:41.52 | SplasPood | (or better yet, did that explanation even make sense) |
| 23:42.03 | JT | EmleyMoor: you need special cables for a TDM400P? |
| 23:42.07 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 23:42.28 | sbingner | hey anybody know what the general recommended line to user ratio is? |
| 23:42.52 | bbblack | SplasPood: check ForkCDR, ResetCDR, etc... |
| 23:43.30 | EmleyMoor | JT: Yes - we don't use the RJ11 at the line end here |
| 23:43.40 | SplasPood | bbblack: Well I did.. ForkCDR seemed to be the way to go, but it seemed one of the two CDR entries would have 0 billsec with a result of NO ANSWER at that point... I'll check again |
| 23:43.44 | bbblack | JT: in my logic, if one side is behind NAT, the other side shouldn't be reinvited, even if it can support reinvites |
| 23:43.45 | EmleyMoor | (modem cables would work, I guess but I keep few of those) |
| 23:44.22 | sbingner | ie. how many users would you generally want behind a 23-channel T1 |
| 23:44.22 | JT | EmleyMoor: ah ok, guess i'm used to crimping my own cables :P |
| 23:44.22 | EmleyMoor | Also, I want to wire some sockets (to provide ring capacitor) and am no good at fitting RJ11s so wanted some spare cables to use |
| 23:44.37 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 23:44.58 | bbblack | anyway, on my way to bugs.digium.com |
| 23:45.02 | JT | sbingner: that really depends on the usage scenario |
| 23:45.19 | sbingner | JT: I know, unfortunately I need an estimate w\ithout a good usage scenario |
| 23:45.41 | JT | sbingner: so you have absolutely no idea what sort of company it is? |
| 23:45.49 | sbingner | it's military deployed |
| 23:46.01 | JT | is it a callcentre |
| 23:46.02 | sbingner | supporting up to 200 users in a deployed environment |
| 23:46.03 | voxtop | i added "options wctdm opermode=AUSTRALIA" to the beginning of /etc/modprobe.d/zaptel . after rebooting, asterisk couldn't start, saying it was unable to use /dev/zap/1 despite it existing. i fixed this by removing "wctdm" from /etc/modules . why would that have fixed the problem? |
| 23:46.03 | JT | is it a factory |
| 23:46.04 | sbingner | no |
| 23:46.06 | JT | is it an office |
| 23:46.11 | JT | narrow it down |
| 23:46.14 | sbingner | it's a bunch of people all over the world |
| 23:46.20 | JT | i see |
| 23:46.51 | sbingner | so there will be a number of sites, and their primary phone service will be thru this solution |
| 23:47.04 | sbingner | probably 30-40 sites and total 200 phones |
| 23:47.26 | JT | voxtop: maybe the option was incorrect, i dunno? |
| 23:48.00 | JT | sbingner: hrm, will there be 200 users a lot of the time? |
| 23:48.18 | sbingner | probably, since they're mostly in the same time zone |
| 23:48.24 | JT | try and think of what they'll be using it for, and how frequent |
| 23:48.27 | sbingner | but some won't be |
| 23:48.41 | JT | you could probably get away with one PRI if usage was moderate |
| 23:49.21 | sbingner | I don't know what they'll be using it for lol... need to call home to the family... need to talk to their commanding officer.... we don't have good definitions |
| 23:49.41 | JT | you'd probably need at least 2 PRIs to be safe |
| 23:49.55 | JT | well defining the requirement accurately is pretty important here :P |
| 23:49.57 | sbingner | that's what I was thinking |
| 23:50.06 | sbingner | I know, it's a pain heh |
| 23:50.17 | sbingner | military is fun ;) |
| 23:51.59 | Druken | count on 10 > 1 for residential use... |
| 23:52.06 | Druken | 200 users == 20 channels |
| 23:53.11 | voxtop | JT: i don't think so, cuz without "wctdm" in /etc/modules and with "opermode=AUSTRALIA" in /etc/modprobe.d/zaptel , asterisk works perfectly |
| 23:54.02 | JT | voxtop: did wctdm previously load ok? |
| 23:56.41 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.238.157.Dial1.SanJose1.Level3.net) |
| 23:58.37 | *** join/#asterisk andresmujica (n=andresmu@201.244.244.253) |
