| 00:00.04 | SgtDitt | lol. alot of the same members in this channel, and no one answers in other. thought i would give it a try |
| 00:00.07 | [TK]D-Fender | <PROTECTED> |
| 00:00.10 | [TK]D-Fender | WTF!? |
| 00:00.28 | Yarq | exten => 6001,2,Dial(IAX2/00905377369696@NuFone) |
| 00:01.29 | Yarq | well for ex. I have asterisk running on my office located in Germany |
| 00:01.42 | Yarq | and I am currently in France for vacation |
| 00:02.09 | Yarq | I connect to asterisk which is in germany using SIP |
| 00:02.20 | Yarq | as my phone does not support IAX2 |
| 00:02.47 | Yarq | and I dial 6001 which should connect me to the nufone server and make an outbound call using IAX2 |
| 00:02.53 | Yarq | thats what I'm trying to do |
| 00:03.11 | [TK]D-Fender | Yarq, I shoul not be seeing these 2 things MIXED |
| 00:03.47 | Yarq | Well you mean I cant do that |
| 00:05.20 | Yarq | :( |
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| 00:11.08 | Yarq | http://pastebin.ca/661724 |
| 00:16.32 | Yarq | what am i doing wrong |
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| 00:27.06 | Yarq | pff |
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| 00:28.52 | raidenz | Hi guys |
| 00:29.33 | raidenz | I compiled libpri+zaptel and installed the latest asterisk 1.4.xx and I see on the CLI that chan_zap has been loaded but I don't see any zap commands on the cli. Any ideas? |
| 00:31.40 | raidenz | ? |
| 00:31.58 | _ShrikE | raidenz: Did you compile asterisk after zaptel? |
| 00:32.05 | raidenz | yes |
| 00:32.12 | _ShrikE | thats all I got |
| 00:33.40 | raidenz | I even checked after configuring asterisk ( include/asterisk/autoconfig.h), that both ZAPTEL and PRI has been defined. |
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| 00:34.27 | AJaymn | Anyone use the meetme gui? |
| 00:35.49 | Yarq | No authority found |
| 00:39.25 | Yarq | Its been 3 hours I'm trying to connect to NuFone |
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| 00:52.29 | Yarq | . |
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| 00:54.27 | ShakataGaNai | Question - Every time music on hold starts, it restarts the song. It doesn't keep playing. Possibly related, when I restart *, it says "mpg123: no proccess killed" (* version 1.4.10.1). Any sugestions as to what I've done wrong? |
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| 01:00.28 | joat | ShakataGaNai: that's how it works... it plays from a selection of files on your machine, vice connecting to a stream... see the voip-info site for using streams for moh |
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| 01:01.18 | VJFROMGT | I want to put multiple addresses in host,, how can i do that eg,, i have host=123.151.125.12 and i wan to include 1 more address |
| 01:01.34 | VJFROMGT | can i do like ip1&ip2&ip3? |
| 01:01.41 | VJFROMGT | or is there some other way of doing it? |
| 01:05.04 | ShakataGaNai | joat: yes, but in some cases it just drops the music and restarts every time someone is put on hold |
| 01:05.10 | ShakataGaNai | it doesn't "keep" its place |
| 01:09.05 | Yarq | Grr.. |
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| 01:20.46 | dagrim | Anybody know of any providers that offer TOLL DIDs? |
| 01:27.31 | dagrim | Havent had any luck finding one.. been looking all day |
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| 02:10.41 | zcionn_ | is it possible to see what codec is being used in CLI? |
| 02:11.37 | onats | is there a listing available on the web to see what hardware has already been used and tested for installation with asterisk? |
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| 02:44.40 | ChulJin | Good evening Gentlemen! |
| 02:56.07 | ChulJin | by any chance, is discussion of DUNDi allowed here |
| 02:59.19 | file | allowed? sure |
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| 03:08.17 | [TK]D-Fender | onats, there is a lits on the WIKI, but there are ver few RECOMMENDED bits. What are your needs? |
| 03:08.41 | [TK]D-Fender | file, likely to get anywhere? Not so great odds ;) |
| 03:10.35 | file | [TK]D-Fender: quite |
| 03:12.49 | ChulJin | sorry, froze up for a bit. |
| 03:13.00 | ChulJin | is the dundi-test context gone altogether? |
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| 03:52.21 | ChulJin | so, allowed byt just ineffective? :P |
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| 06:56.01 | _Krieger_ | i tried to make user fixation by IP-address by setting in users.conf 'host = x.x.x.x' instead of 'host = dynamic', but it does not give what expected... |
| 06:57.38 | JerJer | fun |
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| 07:29.11 | crisen | hi, i have a variable in extensions.conf that is 'sip:1234@example.com' |
| 07:29.31 | crisen | what extension.conf goodness do i use to strip certain parts of that variable and get just '1234'? |
| 07:34.37 | crisen | awesome.. got it |
| 07:34.43 | crisen | ${SIP_HEADER(TO):5:11} |
| 07:34.47 | crisen | peace |
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| 08:53.00 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 08:55.23 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 08:55.53 | asterisknerds | <PROTECTED> |
| 08:58.13 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 09:24.36 | *** join/#asterisk eject_ck (n=eject@galling.glancer.volia.net) |
| 09:28.21 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-49-107.w81-251.abo.wanadoo.fr) |
| 09:28.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:39.27 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 09:44.20 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:53.27 | *** join/#asterisk kkn088 (n=kikoun@88-138-6-52.adslgp.cegetel.net) |
| 09:55.24 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 09:55.54 | asterisknerds | <PROTECTED> |
| 09:59.34 | *** join/#asterisk tsuk (n=tsuki@host86-146-44-140.range86-146.btcentralplus.com) |
| 10:01.26 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 10:02.50 | eject_ck | I'm trying build cdr_addon_mysql |
| 10:03.27 | eject_ck | I have mysql.h in /usr/local/incude/mysql/ but it dont want to compile |
| 10:10.19 | scooby2 | probably need a -I/usr/local/include/mysql |
| 10:14.18 | *** join/#asterisk JulHer (n=julio@244.Red-217-125-14.staticIP.rima-tde.net) |
| 10:17.27 | eject_ck | I added |
| 10:17.27 | eject_ck | gmake -I-I/usr/local/include/mysql |
| 10:17.27 | eject_ck | same sesult |
| 10:18.59 | eject_ck | http://eject.name/pastebin/pastebin.php?show=22 |
| 10:19.14 | eject_ck | brr |
| 10:20.14 | eject_ck | I created symlink to it |
| 10:20.19 | eject_ck | same |
| 10:28.06 | mvanbaak | did you run ./configure ? |
| 10:28.21 | mvanbaak | that should find out the correct parameters for the compiler |
| 10:30.39 | eject_ck | mvanbaakm, sure |
| 10:30.40 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 10:31.03 | eject_ck | I changed path to mysql.h in code |
| 10:31.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:32.50 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 10:33.54 | *** join/#asterisk shtoom (n=shtoom@59.181.111.61) |
| 10:41.32 | eject_ck | what minimal set of mudules I need to use SIP ? |
| 10:42.54 | *** join/#asterisk julien_re (n=julien@AStDenis-103-1-5-193.w81-248.abo.wanadoo.fr) |
| 10:43.05 | julien_re | hi there |
| 10:43.13 | julien_re | i've got a question about RealTime |
| 10:43.27 | julien_re | can anybody help? |
| 10:43.37 | julien_re | "realtime load sipusers username 999" returns a result |
| 10:43.49 | julien_re | but I can't dial realtime peers |
| 10:45.07 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 10:46.21 | *** join/#asterisk engrxyz (n=fgsfgfs@82-34-18-23.cable.ubr03.basi.blueyonder.co.uk) |
| 10:55.11 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 10:55.41 | asterisknerds | <PROTECTED> |
| 11:00.47 | *** join/#asterisk fakhir (i=Fakhir@unaffiliated/fakhir) |
| 11:06.03 | eject_ck | is it possible save sip.conf data in sql database? |
| 11:12.04 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:23.36 | *** join/#asterisk MrMister2 (n=mrmister@89.181.55.48) |
| 11:26.26 | julien_re | yes |
| 11:26.30 | julien_re | eject_ck: with realtime |
| 11:26.37 | julien_re | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 11:35.52 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 11:36.33 | eject_ck | tnx |
| 11:42.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:49.13 | *** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net) |
| 12:03.23 | *** join/#asterisk robh71_ (n=robh71@host-65-124-86-25.entouch.net) |
| 12:22.34 | *** join/#asterisk QbY (n=Kelvin@208.36.224.228.ptr.us.xo.net) |
| 12:27.58 | *** join/#asterisk saftsack (n=saftsack@217.224.111.189) |
| 12:38.10 | *** part/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net) |
| 12:53.23 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 13:04.54 | xheliox | There's no plausible reason for SIP phones to become unreachable (using qualify) on a LAN, unless the phone itself was unplugged or the switch dies, or the network card in the server is sputtering, right? I'm not over looking some silly "droppocketsrandomlyeveryx" setting in sip.conf, am I? ;) |
| 13:11.33 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 13:12.31 | Sweeper | hmm. what's the best way to do totally interaction-free agent logins? |
| 13:14.09 | xheliox | Sweeper: Interaction free? What do you mean exactly? |
| 13:16.05 | Sweeper | I found it. I'm going to use the phone microbrowser to let the agents login/logoff of each queue, so I wanted the phone to be able to dial a number, and log the agent in without the user having to do anything else |
| 13:16.43 | xheliox | That works. |
| 13:17.06 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 13:19.27 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 13:20.59 | eject_ck | Is it possible use automatic calls routing in my case PSTN -> SPA3K -> ASTERISK SIP -> SPA3K -> PSTN |
| 13:21.47 | Sweeper | eject_ck: sure |
| 13:21.49 | eject_ck | I have 2 offices and 2 SPA-3102 Linksys inside it |
| 13:22.40 | eject_ck | how can I doit? I have 20 SIP accounts - I plan use 2xx for 1st office and 3xx for 2nd |
| 13:23.02 | eject_ck | Is I understand I must configure my extentions |
| 13:23.16 | Sweeper | well, you'll have to explain a bit better what you want to do |
| 13:23.26 | *** join/#asterisk CodeBanshee (n=chris@194.164.236.240) |
| 13:23.49 | eject_ck | Sweeper, ok |
| 13:23.50 | Sweeper | you saying you want to take calls from office one, send them over sip to office 2, and then hook into the pstn? |
| 13:23.58 | eject_ck | yes |
| 13:25.47 | Sweeper | easy enough. get asterisk up, and set up 2 sip peers, get the 3102's registered, tell asterisk to take all incoming calls from office one and dial them out the office 2 fxo |
| 13:27.27 | eject_ck | exten => 2XX,1,Dial(SIP/office2/1${EXTEN}) |
| 13:27.46 | Sweeper | erm |
| 13:27.55 | eject_ck | and exten => 3XX,1,Dial(SIP/office1/1${EXTEN}) |
| 13:27.57 | Sweeper | do you have more sip phones there or something? |
| 13:28.15 | Sweeper | or a key system? |
| 13:28.20 | eject_ck | office1 and offcie2 is SPA-3102 devices |
| 13:28.31 | eject_ck | yes I have key systems inside both offcies |
| 13:28.35 | Sweeper | ahhhhhh |
| 13:28.41 | Sweeper | makes more sense now :) |
| 13:29.08 | eject_ck | is my way wight ? |
| 13:29.14 | eject_ck | right |
| 13:29.15 | Sweeper | yea, that looks good, except you need exten => _2XX |
| 13:29.28 | eject_ck | why _ |
| 13:29.29 | Sweeper | the underscore tells asterisk you're using a pattern |
| 13:29.36 | eject_ck | ah |
| 13:29.44 | Sweeper | 2XX would literally be extension 2XX |
| 13:29.56 | eject_ck | what with exten => _XXX,1,Dial(SIP/${EXTEN},12,tr) |
| 13:29.57 | Sweeper | since you can have alphanumeric extens |
| 13:30.05 | eject_ck | that I already have |
| 13:30.19 | eject_ck | can I override it |
| 13:30.33 | Sweeper | well, you can put it afterwards |
| 13:30.43 | eject_ck | at the end ? |
| 13:30.59 | Sweeper | so that it only gets to there if _3XX and _2XX don't happen |
| 13:31.11 | Sweeper | better to put it in its own context, and include it |
| 13:31.35 | eject_ck | I do`nt understand contexts at this moment :( |
| 13:31.48 | eject_ck | where I can get more info it TFOP |
| 13:32.08 | eject_ck | I use only [default] |
| 13:32.22 | Sweeper | ~book |
| 13:32.32 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:32.33 | Sweeper | stupid jbot |
| 13:32.50 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-45-146.nyc.res.rr.com) |
| 13:33.11 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 13:34.23 | eject_ck | Sweeper, can I dial PIN for auth in SPA ? |
| 13:34.34 | Sweeper | mmm |
| 13:34.39 | Sweeper | you should probably do that in asterisk |
| 13:35.30 | eject_ck | how ? |
| 13:35.37 | Sweeper | http://www.voip-info.org/wiki-Asterisk+cmd+Read |
| 13:35.42 | Sweeper | http://www.voip-info.org/wiki-Asterisk+cmd+Gotoif |
| 13:35.48 | Yourname` | Hi, can someone help me with the following dialplan please? The POTS number calls is not even ringing, but it gives the tones when dialed: http://pastebin.ca/662200 |
| 13:36.50 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:38.06 | eject_ck | Sweeper, can hel with this issue - for example my SPA have pin 12121 and is registered with name 100 and connected to PSTN |
| 13:38.56 | Sweeper | eject_ck: sorry, got stuff on my plate atm. do some reading, it's good for you :D |
| 13:39.15 | eject_ck | I added exten => _2XXX,1,Dial(SIP/100/${EXTEN}) and want call from my X-lite to 2109 via SPA |
| 13:39.51 | eject_ck | how I can add read into this exten ? |
| 13:40.12 | Sweeper | http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities |
| 13:42.11 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:44.32 | [TK]D-Fender | Yourname`, You're DIALING and extension for it to SEND you a fax? Thats backwards. |
| 13:45.06 | tzanger | [TK]D-Fender: nah, I had to do that about a decade ago with many companies... they had fax gateways and you dialed the extension of the fax you wanted |
| 13:46.25 | Yourname` | [TK]D-Fender: Yeah.. I'm dialing a POTS line to it to send a fax to that line.. |
| 13:48.36 | [TK]D-Fender | Yourname`, Just have * do the dialoutitself. Because thats your PHONE talking to TXFax |
| 13:48.58 | eject_ck | how send DTMF code in exten ? |
| 13:49.07 | Yourname` | [TK]D-Fender: Aaaahhhh, you're RIGHT! Now.. how can I make it dial out itself, and what do I have to do to invoke it? |
| 13:49.31 | [TK]D-Fender | Yourname`, lookup "call files" and "asterisk auto dial out" on the WIKI |
| 13:49.44 | Yourname` | [TK]D-Fender: Thank you :) |
| 13:49.52 | [TK]D-Fender | Yourname`, And "AMI Originate" while you're at it |
| 13:50.17 | Yourname` | [TK]D-Fender: You rock, rockstar! I'll get on it right now before wife drags me out to Mr. Bean. |
| 13:55.49 | [TK]D-Fender | off for a few hours, later all.... |
| 13:56.25 | eject_ck | how I can add "Inputing PIN" and that ${EXTEN} afert answer in my exten ? |
| 13:56.27 | eject_ck | exten => _2XXX,1,Dial(SIP/199,10,rt) |
| 13:57.57 | eject_ck | 1. I need call to SIP/199 and then input pin 121212# and that input ${EXTEN} is it possible ? Read() and Goto() are not usefull I think :( |
| 14:00.29 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 14:02.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 14:07.10 | Sweeper | eject_ck: then you need to read more :P |
| 14:08.54 | eject_ck | Sweeper, ok |
| 14:14.22 | *** join/#asterisk agarguest (n=agargues@CPE001839ee0745-CM00111ade4822.cpe.net.cable.rogers.com) |
| 14:16.03 | *** join/#asterisk implicit (n=implicit@210.16.55.38) |
| 14:17.05 | eject_ck | Sweeper, can't find how :( |
| 14:17.23 | Sweeper | sigh |
| 14:17.30 | eject_ck | ? |
| 14:17.36 | Sweeper | ok |
| 14:17.48 | Sweeper | you need to prompt for pin BEFORE you dial 199 |
| 14:17.55 | Sweeper | so you'll Answer() |
| 14:18.05 | Sweeper | then use Background() to play the message |
| 14:18.12 | Sweeper | then use Read() to get the pin |
| 14:18.18 | eject_ck | I want don`t answer pin |
| 14:18.31 | Sweeper | ...? |
| 14:19.23 | Sweeper | explain |
| 14:19.42 | eject_ck | I want that user dial _2XXX and it will be 1. Dial SIP/199, wait for answer, input PIN 11111 (automaticaly by asterisk ) and than input ${EXTEN} |
| 14:19.44 | *** join/#asterisk implicit_ (n=implicit@210.16.55.38) |
| 14:19.52 | eject_ck | Understand ? |
| 14:20.20 | Sweeper | oh, you want asterisk to send the pin, and then dial. sure |
| 14:20.25 | eject_ck | yes! |
| 14:20.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:20.44 | Sweeper | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SendDTMF |
| 14:20.46 | *** join/#asterisk implicit (n=implicit@210.16.55.38) |
| 14:23.35 | eject_ck | exten => _2XXX,1,Dial(SIP/199) |
| 14:23.35 | eject_ck | exten => _2XXX,n,SendDTMF(55555#${EXTEN}) |
| 14:23.35 | eject_ck | exten => _2XXX,n,Congestion |
| 14:23.55 | Sweeper | why Conjestion? |
| 14:24.01 | Sweeper | *Congestion |
| 14:24.08 | eject_ck | my miss |
| 14:24.20 | eject_ck | is other correct |
| 14:24.32 | eject_ck | 55555# - PIN |
| 14:24.42 | eject_ck | 199 - number of registred SPA |
| 14:24.42 | Sweeper | yea, that's right |
| 14:24.59 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 14:26.23 | eject_ck | I try call 2109 (109 is internal number in key system) but I still asked for bib by pstn |
| 14:26.26 | eject_ck | by SPA |
| 14:26.50 | eject_ck | so _2XXX,n,SendDTMF(55555#${EXTEN}) - not send |
| 14:27.41 | *** join/#asterisk Weezey (n=weezey@206.210.109.233) |
| 14:28.17 | Weezey | zaptel-1.4 trunk's make install gives me: make: *** No rule to make target `install-inlcude', needed by `install-programs'. Stop. |
| 14:29.42 | eject_ck | gmake |
| 14:29.50 | Weezey | k |
| 14:30.08 | Weezey | nope, same thing. |
| 14:32.31 | Weezey | heh, nice |
| 14:32.38 | Weezey | I'm gonna submit a patch. |
| 14:38.50 | eject_ck | how I SENDDTMF inside DIAL ? |
| 14:39.52 | Weezey | Note: The Dial command comes with the 'D' option that is also able to send DTMF to a channel. |
| 14:45.47 | eject_ck | Weezey, how I can use it? |
| 14:46.07 | eject_ck | how I can send 12345 to chanel inside Dial() |
| 14:47.34 | Weezey | D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) |
| 14:47.52 | Weezey | eject_ck: voip-info.org is your friend. |
| 14:48.36 | Weezey | sweet, i reported a bug. I feel fullfilled. |
| 14:50.59 | eject_ck | :) |
| 14:51.05 | eject_ck | bug in Makefile ? |
| 14:51.24 | eject_ck | IS next exten is of? exten => _2XXX,1,Dial(SIP/199,D(55555#${EXTEN})) |
| 14:55.01 | eject_ck | I missed timeout |
| 14:55.14 | eject_ck | exten => _2XXX,1,Dial(SIP/199,5,D(55555#${EXTEN})) |
| 14:56.09 | *** join/#asterisk Yarq (i=Nwm@88.250.219.95) |
| 14:56.11 | Yarq | Hello |
| 14:59.01 | Yarq | http://pastebin.ca/662267 (WHEN using Nufone to place international call I get this) and cant call the number |
| 14:59.36 | Weezey | do they accept international calls? |
| 14:59.40 | Yarq | yes |
| 14:59.48 | Yarq | on their website they have termination to my country |
| 14:59.55 | Weezey | right. |
| 15:00.03 | Yarq | but not so cheap |
| 15:00.17 | Weezey | hmm, I dunno, looks like it's rejecting it on their dial plan, I'd contact them. |
| 15:01.23 | Yarq | Well Strange |
| 15:02.26 | Yarq | I want to place a call to the country 0090 , area 0537 , number 736 96 96 to test (exten => _011N.,1,Dial,IAX2/Nufone2@NuFone/${EXTEN} ) is what I have in my extensions.conf |
| 15:07.23 | Weezey | Turkey cellular? |
| 15:08.06 | Yarq | yes |
| 15:08.14 | Yarq | Nothing works |
| 15:08.21 | Yarq | UK landline does not work also |
| 15:08.47 | Weezey | huh, maybe they're busted, I'd contact them. |
| 15:09.02 | Weezey | maybe you just need international calling enabled on your account? |
| 15:09.10 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 15:09.13 | masus | hi all |
| 15:10.08 | masus | is this possible for MSSQL ----> "exten => test,3,MYSQL(Query r ${connid} SELECT fromuser FROM sipusers WHERE id=${CALLBACKNUM})" |
| 15:10.45 | *** join/#asterisk arava (n=arava@c-69-248-101-151.hsd1.nj.comcast.net) |
| 15:11.59 | masus | please :) |
| 15:13.49 | Yarq | http://pastebin.ca/662279 if you have a config like this how do you dial international ? |
| 15:16.09 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) |
| 15:17.15 | masus | "application cmd mssql" ??? |
| 15:19.35 | masus | ;( |
| 15:20.41 | Yarq | Call rejected by 64.34.214.150: No authority found |
| 15:23.51 | JerJer | add some authority and you will be good |
| 15:25.06 | Yarq | jerjer I cant add an outhority to the remote host as it is my voip provider |
| 15:25.11 | JerJer | Weezey: goog suggestion |
| 15:25.20 | JerJer | *good |
| 15:25.59 | Yarq | I've contacted them they told me that I have a dialplan problem |
| 15:26.09 | Yarq | that the account is active for international calling |
| 15:28.01 | JerJer | they have? |
| 15:29.21 | JerJer | Yarq: try making another call |
| 15:29.36 | Yarq | well US works fine |
| 15:29.48 | JerJer | you might be pleasantly surprised |
| 15:30.29 | Yarq | surprised ? |
| 15:31.10 | JerJer | just try it - grr |
| 15:31.34 | Yarq | :) does not work |
| 15:31.38 | Yarq | Grr. |
| 15:31.54 | JerJer | then you do have a dialplan problem |
| 15:32.07 | JerJer | i saw nothing from you |
| 15:32.20 | Yarq | you saw nothing from me |
| 15:32.49 | Yarq | hold on |
| 15:34.01 | JerJer | i'm only watching for your nufone2 device |
| 15:34.29 | Yarq | reloading |
| 15:34.55 | Yarq | US dialing works |
| 15:35.00 | JerJer | i see a call to a toll free number |
| 15:35.03 | Yarq | yes |
| 15:35.06 | Yarq | US dialing works |
| 15:35.58 | Yarq | what code do I have to use inorder to make a INT. call |
| 15:36.14 | JerJer | 011 |
| 15:36.31 | JerJer | then if there is any local dialing prefix you must drop it |
| 15:36.52 | JerJer | for example - UK they express numbers like this 44 (0) 734 23125 |
| 15:36.57 | JerJer | drop the 0 in that case |
| 15:37.01 | Yarq | _011NXXNXXXXXXX |
| 15:37.11 | JerJer | _011N. is enough |
| 15:37.12 | Yarq | JerJer I know that you have to remove the area codes 0 |
| 15:37.33 | Yarq | Reloading |
| 15:37.49 | JerJer | since like Italy has valid 4 digit telephone numbers (mostly governmental) |
| 15:37.58 | JerJer | 905 |
| 15:38.50 | Yarq | well this time it went through but did not get a call here |
| 15:39.22 | JerJer | i see progress and answered |
| 15:39.36 | JerJer | for the number ending in 6 you just dialed |
| 15:39.47 | JerJer | progress |
| 15:39.55 | Yarq | ok thistime it worked |
| 15:40.39 | JerJer | good - now blog about the fact that you have international service from NuFone :D |
| 15:40.50 | Yarq | where ? |
| 15:40.52 | Yarq | :D |
| 15:41.13 | Yarq | JerJer: A question are international calls blocked by default ? |
| 15:44.09 | Yarq | Thankyou Jeremy :) |
| 15:46.22 | JerJer | at this juncture you have to ask for international access - with our next round of changes there will be a GUI option to apply |
| 15:46.45 | Yarq | Why do you do that for security purposes? |
| 15:46.48 | JerJer | that is new accounts have to ask |
| 15:46.59 | JerJer | because of stupid people |
| 15:47.05 | Yarq | stupid people ? |
| 15:47.25 | JerJer | a very few stupid people made everyone jump through unnecessary hoops |
| 15:47.45 | Yarq | hmm |
| 15:48.34 | JerJer | Yarq: if you haven't already, you really should go read our past blog entries |
| 15:48.43 | Yarq | you mean |
| 15:49.00 | Yarq | the blog on your site or the nufone blog |
| 15:49.06 | JerJer | nufone blog |
| 15:49.07 | Weezey | ... |
| 15:49.34 | JerJer | my personal blog is just a resource for those needing asterisk / openser guidance |
| 15:49.58 | JerJer | and of course to provide backlinks :) |
| 15:50.04 | Yarq | :) |
| 15:50.05 | Yarq | hehe |
| 15:50.23 | JerJer | but don't tell anyone |
| 15:50.24 | Yarq | well Jeremy sorry about the duplicated posts yesterday but It was not my fault |
| 15:50.45 | Yarq | one time Ive got an error response from HTTPD on nufone |
| 15:50.52 | Yarq | and the other time Ive got a connection error |
| 15:51.05 | Yarq | I hope Nufone is not a one man business :P |
| 15:51.13 | Yarq | as I have only seen you around and nobody else |
| 15:51.32 | JerJer | hey is anyone here have chan_h323 compiled and running on their system? if so pastebin an ldd /usr/sbin/asterisk for me plz |
| 15:52.11 | JerJer | go look for yourself nufone.net/about |
| 15:52.37 | Yarq | JerJer |
| 15:52.38 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:52.43 | Weezey | http://en.wikipedia.org/wiki/User:Jerjer |
| 15:52.57 | JerJer | granted lots of people hate me |
| 15:52.58 | Yarq | to be honest I dont believe shit which writes on the internet |
| 15:53.04 | JerJer | but ya know i hate them, so its ok |
| 15:53.28 | Yarq | Everyone can write something on a webpage :P |
| 15:53.32 | Yarq | anyway Thank you alot, |
| 15:53.36 | JerJer | i'm not frenchy |
| 15:54.01 | JerJer | nor am I a high school student that shot up the place |
| 15:54.38 | Yarq | JerJer could you close my trouble tickets please |
| 15:54.39 | JerJer | but i am on Scott Liard's (sic) stupidity list - which oddly enough has serious relevance for some reason |
| 15:55.21 | JerJer | Yarq: i'll inform support - as many many others will tell you, I shouldn't do support |
| 15:55.49 | Yarq | but you do |
| 15:56.00 | Yarq | as you replied to my posts |
| 15:56.00 | JerJer | i just yell at people |
| 15:56.08 | Yarq | yeah i know that |
| 15:56.22 | Yarq | most CTO's suck because of that |
| 15:56.30 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 15:56.32 | JerJer | thank you |
| 15:56.33 | Yarq | to be honest I hate them |
| 15:56.43 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 15:56.47 | JerJer | i feel the love |
| 15:56.56 | Yarq | http://www.youtube.com/watch?v=EJJL5dxgVaM |
| 15:57.00 | Yarq | thats funny |
| 15:57.18 | JerJer | YA leme flip open my iPhone - NOT |
| 15:57.25 | JerJer | er it doesn't flip :D |
| 15:58.14 | JerJer | i have no sound currently, so watching videos is kinda useless |
| 15:58.47 | Yarq | hmm |
| 15:59.06 | Yarq | SetCIDNum does not work Grr. |
| 15:59.21 | JerJer | use ${CALLERID} |
| 15:59.31 | JerJer | well the function - |
| 15:59.32 | *** join/#asterisk saftsack (n=saftsack@217.224.111.189) |
| 15:59.54 | JerJer | Set(CALLERID(number)=911) |
| 16:00.33 | Yarq | huh |
| 16:00.44 | Yarq | +777999777 is calling :S |
| 16:00.52 | Yarq | but I have Set(CALLERID(number)=2025551212) |
| 16:01.41 | JerJer | yeah, depending on the carrier we use international caller*id could be totally fscked |
| 16:01.53 | JerJer | i hate it, but nothing i can do - currently |
| 16:02.04 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:02.27 | Yarq | :( |
| 16:02.54 | JerJer | submit a trouble ticket |
| 16:03.08 | Yarq | telling what |
| 16:03.09 | JerJer | just be detailed - time of call, number dialed, caller*id value set etc |
| 16:03.37 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:10.36 | Yarq | well strange first it was showing +777999777 now itsprivate |
| 16:13.39 | JerJer | yes, it varies depending on exactly which carrier we send the call to |
| 16:15.59 | Yarq | wow what a poor quality :D |
| 16:16.13 | Yarq | need to change the codec I think |
| 16:16.53 | Yarq | do you have carriers in every country or do you use global carriers ? |
| 16:18.20 | JerJer | we have private ip and/or tdm connections to all of our carriers |
| 16:18.33 | JerJer | and only ever use ulaw in our transport |
| 16:30.24 | *** join/#asterisk saftsack (n=saftsack@217.224.111.189) |
| 16:31.38 | masus | is this possible for MSSQL ----> "exten => test,3,MYSQL(Query r ${connid} SELECT fromuser FROM sipusers WHERE id=${CALLBACKNUM})" |
| 16:31.59 | masus | MYSQL --> MSSQL |
| 16:33.12 | *** join/#asterisk implicit (n=implicit@vc241224.vpn.uci.edu) |
| 16:35.32 | masus | ;( |
| 16:35.36 | masus | no one any idea |
| 16:35.51 | masus | only yes or no , i'll try to findout |
| 16:40.37 | *** join/#asterisk ethemc (i=masus@88.248.73.6) |
| 16:42.09 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 16:43.14 | *** join/#asterisk Ebola (n=Ebola@host86-138-7-229.range86-138.btcentralplus.com) |
| 16:47.39 | Yarq | why ulaw ? |
| 16:51.05 | *** join/#asterisk Zeppelin_ (n=zeppelin@201-35-77-13.paemt700.dsl.brasiltelecom.net.br) |
| 16:52.13 | Yarq | well this time when called nothing showed up :S |
| 16:52.24 | Yarq | very strange |
| 16:55.17 | *** join/#asterisk implicit (n=implicit@210.16.55.38) |
| 16:56.21 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 17:05.54 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:06.23 | _Krieger_ | i tried to make user fixation by IP-address by setting in users.conf 'host = x.x.x.x' instead of 'host = dynamic', but it does not give what expected... plz give any advice |
| 17:09.18 | *** join/#asterisk saftsack (n=saftsack@p57A7749B.dip.t-dialin.net) |
| 17:13.57 | *** join/#asterisk mihinomenest (i=4Vzg@66.255.220.17) |
| 17:16.07 | Yarq | are there any critical fixes since 1.4.9 |
| 17:16.45 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 17:17.44 | *** join/#asterisk Peaceful (n=peaceful@70.98.162.62) |
| 17:17.52 | Peaceful | Is there no asterisk-sounds for 1.4? |
| 17:18.51 | *** join/#asterisk Super_Cat_Frog (n=bob@82-40-170-180.cable.ubr02.blac.blueyonder.co.uk) |
| 17:20.13 | _Krieger_ | Peaceful: there is. After 'make install' do 'make sounds' as i remember |
| 17:20.27 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:20.52 | _Krieger_ | it will download sounds |
| 17:21.15 | Peaceful | _Krieger_, ok, I'll try that. |
| 17:22.32 | Peaceful | _Krieger_, it appears that only installs the core sounds, and it already did that with 'make install' |
| 17:22.48 | Peaceful | I'm talking about all the extra sounds that came in asterisk-sounds for 1.2 |
| 17:23.39 | Peaceful | After poking around a bunch, I found what looks like some extra sounds in: http://downloads.digium.com/pub/telephony/sounds/ |
| 17:23.43 | *** join/#asterisk booray (n=booray@adsl-75-3-250-156.dsl.irvnca.sbcglobal.net) |
| 17:23.54 | Peaceful | but they're in about a half-dozen different formats |
| 17:24.07 | Peaceful | Do we just download those now, instead of installing asterisk-sounds? |
| 17:25.02 | Sweeper | Peaceful: surely there's a package? |
| 17:26.19 | Peaceful | Sweeper, that's what I'm asking. The files at the location above are at least tarred up, but in the past there was an installer that put them in the right place. |
| 17:26.54 | _Krieger_ | aaa, there`re also points in 'make menuselect' - you can select what sounds to install, and sounds are in different formats. Maybe you need that |
| 17:27.10 | *** join/#asterisk ManxPower (n=manxpowe@015-852-105.area5.spcsdns.net) |
| 17:27.17 | Peaceful | make menuselect? what's that? |
| 17:27.41 | ManxPower | Peaceful: that is the 1.4 way of building asterisk |
| 17:28.11 | Peaceful | A ton has changed since 1.2! |
| 17:28.12 | *** join/#asterisk Chris-NB (n=chris@smtp13.tech.t-mobile.at) |
| 17:28.48 | Peaceful | Is there an upgrade guide around here somewhere? I'm following my 1.2 notes for doing a new 1.4 installation, and I keep going astray |
| 17:29.10 | ManxPower | Peaceful: you mean like the UPGRADE.txt that is included in 1.4 source code? |
| 17:29.23 | Peaceful | ManxPower, lol, that would probably work |
| 17:30.49 | Peaceful | Ok, I'll give UPGRADE.txt a nice, thorough reading. Thanks for the help, guys! |
| 17:32.46 | Yarq | http://pastebin.ca/662392 any idea why I cant dial an internal extension |
| 17:33.07 | _Krieger_ | please explain, can field 'host' in users.conf be used to accept registration only from defined ip-address? |
| 17:34.42 | Yarq | btw. can you disable ZAP on the machine that is only voip gateway? |
| 17:37.31 | *** join/#asterisk evisu (i=hIRC@bzq-88-155-184-243.red.bezeqint.net) |
| 17:40.56 | Yarq | I want to unload unused modules can i stop the zapchan or not |
| 17:42.45 | *** join/#asterisk tomcontr3 (n=tomcontr@145-79-246-201.adsl.terra.cl) |
| 17:43.01 | tomcontr3 | hi, I have a litle question about digium hardware |
| 17:43.19 | tomcontr3 | I would like to buy this card: http://www.digium.com/en/products/hardware/b410p.php |
| 17:43.45 | tomcontr3 | but I not sure about the specs that talk about the Voltage (3,3 or 5,0) |
| 17:44.03 | tomcontr3 | how do I know if my Mother board supports that? |
| 17:46.02 | Super_Cat_Frog | hi - i cant make internal calls - i get 404's, when i try to call either voicemail (3000) or a sipgate number. Here's my config: http://www.tvfrog.co.uk/asterisk/ |
| 17:51.02 | ManxPower | tomcontr3: they only way to know for sure is check the specs for your motherboard |
| 17:51.18 | ManxPower | Super_Cat_Frog: What device is trying to make the call, and what context= setting is for that device? |
| 17:51.20 | _Krieger_ | tomcontr3: read MB manual |
| 17:52.07 | Super_Cat_Frog | its user 'bob', context default - its a seiments gigaset s450ip |
| 17:52.07 | evisu | guys, i'm trying to pass a variable to the dialplan using manager originate -- but the variable is just blank when i try and use it... do i need to use SetVar in some form before i actually use the variable? |
| 17:52.36 | ManxPower | Super_Cat_Frog: do you have an exten => 3000 in the [default] section of extensions.conf? |
| 17:53.17 | Super_Cat_Frog | yes, [default] does include for all the other sections, which includes voicemail |
| 17:53.19 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 17:54.21 | ManxPower | for one thing you are missing the [global] (or is it [globals] section of extensions.conf |
| 17:54.38 | ManxPower | also on MY computer there are no line breaks in the file. |
| 17:54.57 | ManxPower | which means, you prolly created the file in dos/windows and then uploaded it to Asterisk |
| 17:54.58 | Super_Cat_Frog | manx - you need to view the source |
| 17:55.09 | Super_Cat_Frog | it was created in nano on a freebsd box |
| 17:56.38 | ManxPower | well put in a [globals] section right after [general] |
| 17:57.26 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:57.42 | Super_Cat_Frog | ManxPower: ive just looked up the globals section of extensions.conf - and i cant see anything that applies to me, i'm not using any global variables |
| 17:57.47 | ManxPower | BTW, extensions 3001 and 3002 will never work thgat way. |
| 17:58.00 | ManxPower | Super_Cat_Frog: if it does not exist asterisk con get confused and ignore the first real context. |
| 17:59.00 | ManxPower | every extension MUST have at least a priority 1 or it will be ignored. |
| 17:59.07 | Super_Cat_Frog | ManxPower: ah, so i just need an empty [globals] ? |
| 17:59.44 | ManxPower | yup |
| 18:01.25 | ManxPower | I normally only put a few things in [globals] Usually SITE=sitename and PSTN=Zap/g1 |
| 18:01.40 | ManxPower | SITE is used in error reporting macros I wrote |
| 18:01.43 | VJFROMGT | how do i disco a call from cli? |
| 18:02.06 | VJFROMGT | i mean,, how do i disco a call using cli? |
| 18:02.13 | ManxPower | VJFROMGT: you need a disco ball first, then lots of cocane and a large floor and good sound system |
| 18:02.23 | VJFROMGT | lol |
| 18:02.33 | VJFROMGT | how do i terminate a call using cli?> |
| 18:02.44 | ManxPower | VJFROMGT: "show applications like hangup" |
| 18:02.56 | ManxPower | eer...er... sorry. |
| 18:02.58 | ManxPower | that is wrong. |
| 18:03.05 | ManxPower | "help softhangup" |
| 18:03.14 | VJFROMGT | help softhangup |
| 18:03.31 | ethemc | applcation cmd mysql --> mssql |
| 18:03.34 | ethemc | is this possible ? |
| 18:04.39 | ethemc | "exten => MsSql(Connect connid localhost asterisk asterisk asterisk)" |
| 18:06.50 | ethemc | ;( ;( |
| 18:09.31 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:11.05 | *** join/#asterisk bkw_ (n=brian@pool-71-246-222-63.washdc.fios.verizon.net) |
| 18:11.54 | bkw_ | I thought asterisk was a pbx :P |
| 18:12.06 | bkw_ | now its a full Application Platform too? |
| 18:12.29 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:14.04 | ManxPower | Asterisk was never really a PBX. It is a PBX Toolkit that lets you build a PBX. |
| 18:19.07 | *** join/#asterisk robh71_ (n=robh71@host-65-124-86-25.entouch.net) |
| 18:22.24 | Super_Cat_Frog | ManxPower: ive added an empty globals section, but that didn't help - any other ideas? |
| 18:22.35 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 18:23.03 | *** join/#asterisk extr3m (n=js@213.134.125.3) |
| 18:24.26 | extr3m | anyone here happen to know what "span" to use on an isdn-bri card (Digium T1/E1 card) |
| 18:28.08 | *** part/#asterisk ethemc (i=masus@88.248.73.6) |
| 18:28.57 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 18:29.25 | bkw_ | ManxPower, its the same with FreeSWITCH its a tool kit |
| 18:32.36 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.188.154) |
| 18:34.32 | *** join/#asterisk robh71_ (n=robh71@host-65-124-86-25.entouch.net) |
| 18:35.03 | extr3m | how can i see what zaptel cards i currently (according to the system?) have installed? |
| 18:35.10 | Sweeper | except freeswitch has tools to make tools to make pbx's ;) |
| 18:35.17 | Sweeper | asterisk is quite a bit more refined atm |
| 18:35.37 | Sweeper | luckily adhearsion 0.8.0 comes with freeswitch support :D |
| 18:36.00 | Weezey | extr3m: ztcfg -vvvv |
| 18:36.01 | rudholm | extr3m: ztcfg -vv |
| 18:36.22 | Sweeper | extr3m: ztcfg -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 18:36.27 | Weezey | heh |
| 18:36.53 | rudholm | ztcfg -v-with-a-bar-on-top |
| 18:36.54 | extr3m | ehm thats the thing... i dont have any channels configured since it complains about my span |
| 18:37.13 | extr3m | and /proc/zaptel/ doesnt contain any info |
| 18:37.15 | rudholm | extr3m: oh, if you want to see the card, try lspci |
| 18:37.38 | Sweeper | extr3m: if proc doesn't have anything, sounds like a kernel module might not be loaded? |
| 18:37.40 | extr3m | .. gentoo doesnt have lspci :S |
| 18:38.04 | Super_Cat_Frog | extr3m: it does, you just have to install it |
| 18:38.15 | extr3m | oh.. how do i isntall it then? |
| 18:38.18 | *** join/#asterisk logicwrath (n=some@c-68-41-24-98.hsd1.mi.comcast.net) |
| 18:38.38 | extr3m | or should i ask that in #gentoo perhaps? |
| 18:38.45 | Super_Cat_Frog | extr3m: 1 second |
| 18:38.59 | Super_Cat_Frog | emerge pciutils |
| 18:39.00 | rudholm | emerge pciutils |
| 18:39.02 | rudholm | hehe |
| 18:39.06 | Super_Cat_Frog | woo! i win |
| 18:39.19 | rudholm | hey, I was in another window! |
| 18:39.20 | rudholm | no fair! |
| 18:39.21 | rudholm | :) |
| 18:39.34 | extr3m | thanx, emerging now |
| 18:39.44 | Super_Cat_Frog | my nearest gentoo box is 200 miles away, so we're even |
| 18:40.03 | extr3m | uhm... |
| 18:40.12 | extr3m | and if the card isnt visible with lspci..? :S |
| 18:40.17 | rudholm | yeah, that extra couple ms of latency evened the handicap :-p |
| 18:40.17 | Weezey | I need more boxes spread over the globe |
| 18:40.27 | extr3m | my boss will kill me if i broke that several thousand skr pci card.. :S |
| 18:40.32 | Yarq | Sweeper: why 62x "v"'s? |
| 18:40.47 | Weezey | extr3m: dammaged in shipping |
| 18:40.59 | Sweeper | Yarq: to make sure zaptel knows you're serioues |
| 18:41.14 | extr3m | ima try to move it to another pcislot... |
| 18:42.23 | rudholm | I'm finally going to move from 1.2 to 1.4, is there some kind of extensions.conf lint program I can use or even just a list of syntax changes so I know if I need to change anything in my extensions.conf? |
| 18:45.09 | Weezey | rudholm: if you use SVN 1.4 it'll tell you when each thing loads what needs to be changed. |
| 18:45.28 | Weezey | not sure about release. |
| 18:47.27 | rudholm | ok, is it pretty clear about the syntax changes? I'd like to minimize downtime. |
| 18:47.47 | rudholm | ideally, I'd like to test my config files before I upgrade |
| 18:48.04 | rudholm | or at least see a list of syntactical constructs that have been deprecated/changed |
| 18:49.47 | extr3m | hmm.. is there any kernel features that need to be set for a Digium TE110P card to be "visible" to the system? |
| 18:49.55 | extr3m | or bios features perhaps? |
| 18:50.05 | rudholm | sure, you have to load the driver module |
| 18:50.20 | rudholm | shouldn't need to do anything with bios settings |
| 18:50.22 | extr3m | ive loaded zaptel ... its visible in lsmod |
| 18:51.20 | extr3m | cause the "diode" isnt even blinking telling me theres no cable in the card |
| 18:51.27 | extr3m | as it did prior to me moving it to this new server |
| 18:51.28 | extr3m | :s |
| 18:51.38 | rudholm | I use analog cards, and there's a driver besides zaptel that's needed |
| 18:52.08 | rudholm | I have wctdm24xxp, zaptel, and crc_ccitt |
| 18:52.29 | extr3m | wcte11xp and crc_ccitt both are loaded on my sys |
| 18:52.30 | rudholm | crc_ccitt is part of the kernel distribution |
| 18:52.34 | rudholm | ok |
| 18:53.48 | extr3m | any other tips/suggestions? |
| 18:54.20 | rudholm | I don't have experience with the T1 cards so that's about all I can think of |
| 18:54.41 | rudholm | maybe someone who has used that card knows of the common oversights |
| 19:02.41 | bkw_ | rudholm, what card do you have? |
| 19:02.52 | bkw_ | if its a T1 card its not wctdm24xxp |
| 19:03.45 | rudholm | yeah, that's why I said "I use analog cards..." |
| 19:03.55 | bkw_ | what distro you on? |
| 19:03.56 | rudholm | about 10 lines up |
| 19:04.16 | bkw_ | extr3m, is who I ment to direct that to :P |
| 19:04.26 | bkw_ | brb |
| 19:04.43 | bkw_ | Sweeper, just an FYI anything over say 4 v's is pointless |
| 19:05.05 | rudholm | I think sweeper was just being silly with all those v's |
| 19:05.29 | rudholm | I don't think it was meant seriously |
| 19:06.06 | Sweeper | :P |
| 19:12.52 | *** part/#asterisk pnlarsson (n=pnlarsso@c83-248-12-187.bredband.comhem.se) |
| 19:23.46 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 19:26.25 | extr3m | i think i got it to work now |
| 19:26.32 | extr3m | swapped it from one pci-slot to another |
| 19:26.35 | extr3m | a few times |
| 19:26.47 | extr3m | ...got a nasty "Red alarm" on it tho with zttool |
| 19:26.54 | extr3m | wonder what thats all aboot |
| 19:29.15 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 19:30.21 | mace | ftp.digium.com not accepting ftp connections for anyone else? |
| 19:40.00 | *** join/#asterisk implicit (n=implicit@210.16.55.38) |
| 19:41.25 | lisandropm | heelo |
| 19:41.39 | lisandropm | I just compiled asterisk with libpri |
| 19:41.47 | lisandropm | then I do a asterisk -rvvvvvvv |
| 19:41.54 | lisandropm | and reload chan_zap.so |
| 19:42.07 | lisandropm | but it says that the module cannot be found |
| 19:42.12 | lisandropm | did I miss something? |
| 19:48.31 | nDuff | lisandropm: is the module actually there in /usr/lib/asterisk/modules (or local equivalent)? |
| 19:48.43 | lisandropm | checking |
| 19:49.28 | lisandropm | nDuff: no, so I think I missed some config variable |
| 19:49.30 | nDuff | lisandropm: do you have the zaptel package installed? (libpri alone isn't enough) |
| 19:49.53 | lisandropm | nDuff: zaptel was the first package to be compiled |
| 19:50.17 | lisandropm | i do also modprobed ds1x1f (the module for my E1 card) |
| 19:50.42 | [TK]D-Fender | lisandropm, in order : libpri, zaptel, THEN asterisk |
| 19:51.00 | [TK]D-Fender | lisandropm, What card do you have? |
| 19:51.09 | lisandropm | [TK]D-Fender: I did it in this order: zaptel, libpri, asterisk |
| 19:51.19 | [TK]D-Fender | lisandropm, then yuo did it wrong. |
| 19:51.25 | lisandropm | [TK]D-Fender: I have a Telecom Design group's ds1x1f |
| 19:51.43 | lisandropm | [TK]D-Fender: ok, so in the book is worng |
| 19:51.47 | lisandropm | *wrong |
| 19:52.09 | lisandropm | I mean, "*, the future of technology" |
| 19:52.49 | lisandropm | [TK]D-Fender: zaptel needs libpri compiled before compiling it?? |
| 19:53.55 | [TK]D-Fender | lisandropm, Yes. they are seperate and Zaptel DEPENDS on it. |
| 19:54.02 | *** join/#asterisk VoIPMasta (n=dopeshow@148.240.6.48) |
| 19:54.04 | VoIPMasta | Hi there |
| 19:54.11 | [TK]D-Fender | lisandropm, And in the book the libpri chapter clearly comes before the zaptel comiling one. |
| 19:54.49 | [TK]D-Fender | lisandropm, Anyways, jsut redo them all. |
| 19:55.03 | lisandropm | [TK]D-Fender: actually, it compiles zaptel first |
| 19:55.09 | lisandropm | anyway, I'll re do them :-) |
| 19:55.29 | VoIPMasta | I'm having the infamous "cause 50 no authority found" error, I've looked everywhere and can't find what's wrong... however maybe I'm already to messed up in my confs (as I've been modifying them all day long), would anyone please take a look to see if at a first glance you can spot the problem? |
| 20:00.29 | VoIPMasta | http://www.pastebin.ca/662489 |
| 20:02.05 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 20:05.21 | *** join/#asterisk ZefK (n=Zefk@wsc-fo.b.astral.ro) |
| 20:06.37 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 20:06.38 | [TK]D-Fender | VoIPMasta, context=entrantes .... |
| 20:06.38 | [TK]D-Fender | [entrante] |
| 20:07.08 | x86 | when setting up zapata.conf, can I use something like channel => 25-35,37-39,41 ? |
| 20:08.04 | x86 | aka, non-sequential |
| 20:11.02 | [TK]D-Fender | x86, yes |
| 20:11.25 | x86 | even though zaptel.conf is setup for 25-48? |
| 20:11.37 | x86 | it's an FXO channel bank and I only want to use certain ports on it |
| 20:12.02 | [TK]D-Fender | x86, Sure. |
| 20:12.11 | x86 | sexcellent |
| 20:12.27 | x86 | was hoping so, otherwise when people try to dial out, it would have given them a dead line ;) |
| 20:14.35 | lisandropm | [TK]D-Fender: great, now I have the chan_zap module |
| 20:14.45 | lisandropm | let's see if I can make work this card now |
| 20:15.26 | VoIPMasta | [TK]D-Fender, I'm still getting "no authority found" |
| 20:15.31 | [TK]D-Fender | lisandropm, I had real trouble finding reference to anyone successfully using it, and the links to the MFG's site are dead. |
| 20:16.21 | lisandropm | [TK]D-Fender: yes, I had the same problem :-(. |
| 20:16.37 | Sweeper | zaptel only depends on libpri if you want to use a pri :P |
| 20:16.38 | [TK]D-Fender | VoIPMasta, exten => _1000,1,Dial(IAX2/username) <- where the heck is [username]? And as a peer you want to dial through you didn't specify the exten OR context. |
| 20:16.41 | lisandropm | the worst part is that I have a Siemens Hicom 300 connected in the other side |
| 20:16.44 | VoIPMasta | I get a "Rejected connect attempt from [server2] who was trying to reach s@ |
| 20:16.53 | VoIPMasta | The main issue is why it's not assigning a context... |
| 20:16.57 | Sweeper | you can compile zaptel sans libpri just fine~ |
| 20:17.07 | lisandropm | Sweeper: i have a pri :-) |
| 20:17.11 | [TK]D-Fender | Sweeper, And you only need Depends when you start leaking BS again! ;) |
| 20:17.19 | x86 | haha |
| 20:17.25 | [TK]D-Fender | VoIPMasta, Do that in your dial. |
| 20:17.35 | Sweeper | [TK]D-Fender: I guess you'd know all about that~ |
| 20:17.43 | VoIPMasta | [TK]D-Fender, I can't modify the Dial on Server 2 |
| 20:17.51 | lisandropm | and I do not knw if the Hicom is corrctly configured (i'm unable to touch it) |
| 20:23.14 | lisandropm | ok, here goes nother question (if it is not correct to do this questions here, please tell me): do I have to set /etc/asterisk/zapata.conf correctly in order to get a non-yellow alarm in zttool? |
| 20:27.47 | lisandropm | anyone? |
| 20:28.54 | [TK]D-Fender | lisandropm, For PRI, hell yeah |
| 20:29.39 | ZefK | Hi. Does anyone tested Cisco 7940G ? (voice quality on handset, headset, speaker, full-duplex, voicemail indications). Thx |
| 20:30.12 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 20:31.00 | Sweeper | ZefK: buy polycom :v |
| 20:32.17 | ZefK | Sweeper: I already bought 2 models from: Aastra, Policom, Avaya and Linksys. |
| 20:32.47 | Sweeper | ZefK: then don't bother with the cisco :D |
| 20:33.04 | Sweeper | they're clunkier and not as easily provisioned as polycoms, and cost more |
| 20:34.24 | ZefK | Sweeper: On polycoms you need an amp when you want to use headsets. This is not very nice. |
| 20:35.03 | Sweeper | an amp? o.O |
| 20:35.13 | Sweeper | wtf kind of headset are you using? |
| 20:36.16 | ZefK | HW251N SupraPlus Wideband Monaural Headset |
| 20:36.23 | Sweeper | huh |
| 20:36.38 | lisandropm | http://pastebin.ca/662513 <-- Any ideas why asterisk is ignoring some parameters? |
| 20:36.49 | Sweeper | I've just got some generic pos, I can hear things fine... |
| 20:36.59 | lisandropm | of course, any guide for pri configuration will be most welcomed :-D |
| 20:37.04 | Sweeper | you know you can turn up the headset volume, right? >.> |
| 20:37.12 | [TK]D-Fender | lisandropm, How about showing us your CONFIG, and not jsut the erros it spews out |
| 20:37.31 | lisandropm | [TK]D-Fender: yes! I just didn't wanted to bother! |
| 20:37.39 | [TK]D-Fender | lisandropm, BRILLIANT |
| 20:38.09 | ZefK | Sweeper: I know that Polycoms are the best, but I want to find out who's on second place . :) |
| 20:38.39 | [TK]D-Fender | ~phones |
| 20:38.42 | jbot | it has been said that phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 20:38.45 | [TK]D-Fender | ^^^^^^ |
| 20:39.24 | *** join/#asterisk bkw_ (n=brian@pool-71-246-222-63.washdc.fios.verizon.net) |
| 20:39.38 | lisandropm | http://pastebin.ca/662517 <-- my config [TK]D-Fender |
| 20:40.00 | [TK]D-Fender | lisandropm, looks fine |
| 20:40.34 | VoIPMasta | [TK]D-Fender, isn't there a way to assign a context to incoming calls that don't have a context? |
| 20:40.51 | [TK]D-Fender | lisandropm, And for everyones sake permanently remove ALL commented out lines. |
| 20:41.12 | lisandropm | [TK]D-Fender: gret, I'll do it |
| 20:41.16 | [TK]D-Fender | VoIPMasta, Should be according to the "context=" set. |
| 20:41.45 | lisandropm | so I guess I will have to look for the problem in the hicom or in the card |
| 20:41.56 | VoIPMasta | [TK]D-Fender, however as you can see, I'm setting a context in the iax.conf user definition, however it still says that the call is trying to reach "s@" |
| 20:42.40 | [TK]D-Fender | VoIPMasta, well ADD the context to your dial. |
| 20:42.48 | *** join/#asterisk guillote_GNU (n=guillote@host35.200-117-217.telecom.net.ar) |
| 20:42.51 | [TK]D-Fender | VoIPMasta, AND the exten. |
| 20:42.56 | VoIPMasta | [TK]D-Fender, I can't modify server2 |
| 20:43.16 | [TK]D-Fender | VoIPMasta, Fix. Your. DIAL. |
| 20:43.31 | [TK]D-Fender | VoIPMasta, And if you can't, TFB. |
| 20:45.35 | VoIPMasta | [TK]D-Fender, The problem with server 2 is that it uses just Dial(IAX2/${EXTEN}) as 99.9% of the IAX peers connecting to that server are directly registered to it and are regular clients, not asterisk servers. |
| 20:45.35 | [TK]D-Fender | VoIPMasta, Too bad. |
| 20:45.55 | [TK]D-Fender | VoIPMasta, Don't go to a mechanic, ask him to fix your car, then deny him access to it. |
| 20:45.57 | VoIPMasta | [TK]D-Fender, If I modify the AGI to send IAX2/${EXTEN}@context, most IAX2 softphones will have issues |
| 20:46.36 | [TK]D-Fender | VoIPMasta, You're going to have to stop thinking that 1 way will work for enerything and split stuff up. |
| 20:47.01 | VoIPMasta | [TK]D-Fender, So what I'm trying to do is to fool SERVER 2 into "thinking" that SERVER 1 is just a regular softphone (peer) registered to it |
| 20:47.37 | lisandropm | http://pastebin.ca/662526 <-- without commented lines, just in case |
| 20:48.12 | VoIPMasta | [TK]D-Fender, Ok, do you think that it will work if I switch to SIP instead of IAX2? |
| 20:48.52 | ZefK | Another question: I need to connect 30 ip hard phones to my * box. There is any method to encrypt traffic between * and the ip phones ? |
| 20:49.05 | [TK]D-Fender | VoIPMasta, definately better as you don't pass contexts with SIP |
| 20:49.26 | [TK]D-Fender | ZefK, No, * does not support encryption yet |
| 20:50.38 | ZefK | [TK]D-Fender: SO, the password will be in clear text? The phones are world wide . |
| 20:51.50 | [TK]D-Fender | ZefK, you know there are about 5+ protocols invovled with the word "traffic" and you'd be advised to be SPECIFIC about which one you're talking about. |
| 20:52.37 | [TK]D-Fender | ZefK, SIP encrypts the password, but RTP is not encrypted. Then there is the thought for PROVISIONING, which would depend on the phone. |
| 20:52.42 | ZefK | [TK]D-Fender: I just need the best method to protect the server. |
| 20:52.57 | [TK]D-Fender | ZefK, You are being dangerously vague. |
| 20:53.06 | VoIPMasta | [TK]D-Fender, Ok, I'm switching to SIP, however I'm getting an error msg saying that it can't authenticate [callerid]@server2, I'm registering to server2 from server1 using SIP so shouldn't the call go through? |
| 20:53.40 | [TK]D-Fender | VoIPMasta, Show me configs, your descriptions suck. |
| 20:54.06 | [TK]D-Fender | Registering has NOTHING to do with authing calls ot setting contexts, or ANYTHING |
| 20:54.33 | lisandropm | well, thanks everyone and specially [TK]D-Fender for the help. I will try to search the Hicom's config now :-( |
| 20:54.52 | [TK]D-Fender | lisandropm, When did you get this card? |
| 20:54.59 | lisandropm | last year |
| 20:55.01 | lisandropm | in december |
| 20:55.11 | lisandropm | tc-dg's web site was still up |
| 20:55.25 | [TK]D-Fender | lisandropm, Oh well.... past RMA time. Thats what you get for going 15th tier.... |
| 20:56.00 | lisandropm | 15th tier? I'm sorry, I'm argentinian, and I do not know that expressio |
| 20:56.03 | lisandropm | *expression |
| 20:56.23 | [TK]D-Fender | lisandropm, "Cheap shit" <-------- :) |
| 20:56.29 | lisandropm | :-D |
| 20:56.35 | VoIPMasta | [TK]D-Fender, http://www.pastebin.ca/662529 |
| 20:56.41 | [TK]D-Fender | ~ygwypf |
| 20:56.42 | jbot | i guess ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 20:57.31 | [TK]D-Fender | VoIPMasta, Again sadly lacking, I can't see what you are DIALING. |
| 20:57.56 | VoIPMasta | The error msg is: chan_sip.c:10515 handle_request_invite: Failed to authenticate user "My caller ID" mysipusername@server2.mydomain.com |
| 20:58.03 | [TK]D-Fender | VoIPMasta, And you aren't listing everything on both sides. |
| 20:58.15 | VoIPMasta | I'm dialing SIP/username (username is numeric) |
| 20:58.23 | [TK]D-Fender | Ok, times up, gotta jet, back later. |
| 20:58.34 | VoIPMasta | On one side everything is RealTime that's why there aren't any sip or iax confs |
| 20:58.48 | VoIPMasta | ok, thanks for trying |
| 20:59.00 | [TK]D-Fender | VoIPMasta, Next time DUMP THE TABLE. |
| 20:59.15 | [TK]D-Fender | VoIPMasta, No excuse for not showing its CONTENTS |
| 20:59.44 | VoIPMasta | I was just trying to make a quick post |
| 21:15.31 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 21:15.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:19.29 | Super_Cat_Frog | ive just setup asterisk with the digium web frontend, but i cant make any calls from my sip phone (including voicemail, which was automatically setup by the web gui), incoming calls work perfectly - any ideas? |
| 21:20.32 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 21:22.24 | Super_Cat_Frog | is there anywhere i can see what asterisk is trying to do - the standard logfile isn't outputting anything, my phone is giving me a 404 |
| 21:23.48 | *** part/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:26.01 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 21:26.58 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:32.47 | *** join/#asterisk zpertee (n=chatzill@cpe-24-166-81-113.neo.res.rr.com) |
| 21:43.14 | [TK]D-Fender | Super_Cat_Frog, turn on SIP debug in * cli |
| 21:47.30 | VoIPMasta | [TK]D-Fender, just for reference, I fixed it, the problem was just the "insecure" statement that was missing |
| 21:59.04 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 22:00.27 | *** join/#asterisk zpertee (n=chatzill@cpe-24-166-81-113.neo.res.rr.com) |
| 22:16.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:22.25 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:23.34 | Yourname` | Hi. Is there an easy way to configure about 100+ Aastra phones to connect to asterisk? |
| 22:24.38 | rudholm | most phones support config via automatic downloading of config files |
| 22:25.26 | Yourname` | rudholm: So, I'm guessing all the config files need to be made individually, and then let the phones download them from a TFTP server? |
| 22:25.37 | rudholm | yes |
| 22:25.59 | rudholm | sometimes the phone can download more than one config, so you can put the global settings in one file, and the per-phone settings in individualized files |
| 22:26.13 | rudholm | usually it's something like <MAC_ADDRESS>.cfg or .xml or something |
| 22:26.56 | Yourname` | rudholm: Gotcha! Sounds easy. Now, in the CLI.. when a new phone registered, it said "saved useragent Aastra" somethingsomething. Now the scrollback is cleared and I want to see what one of them had as a useragent, is there a way in the CLI for me to find out each phones useragent? |
| 22:27.21 | Yourname` | rudholm: That way it'll be easier for me to check out the aastra's model number, etc |
| 22:27.41 | rudholm | maybe "show sip peers" on the CLI |
| 22:27.57 | rudholm | er sip show peers |
| 22:27.58 | rudholm | rather |
| 22:28.41 | rudholm | ah, ok, do "sip show peer <peer_name>" |
| 22:28.58 | rudholm | and you'll see a bunch of variables for that peer, including the Useragent |
| 22:29.34 | *** join/#asterisk KDan (n=KDan@zux221-072-201.adsl.green.ch) |
| 22:30.14 | *** join/#asterisk CVirus (n=GoD@82.201.222.102) |
| 22:36.09 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:36.14 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:38.32 | Yourname` | rudholm: That worked, thanks! |
| 22:38.41 | rudholm | np |
| 22:38.48 | Yourname` | Trying to figure out some dialplan issue now, lol |
| 22:39.05 | rudholm | I'm in the middle of upgrading from 1.2.x to 1.4.x |
| 22:39.08 | rudholm | so far, so good |
| 22:39.25 | rudholm | but it's kind of scary --always afraid I overlooked something |
| 22:48.11 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.210) |
| 22:50.44 | Yourname` | lol |
| 22:53.34 | Peaceful | What do you do when you have a Digium TE210P, libpri, zaptel, asterisk installed, drivers loaded, but /proc/zaptel is empty?? |
| 22:54.11 | [TK]D-Fender | Peaceful, recompile libpri, zaptel, initialize zapte, then compile * again |
| 22:54.35 | Peaceful | What do you mean by "initialize zapte[l]"? |
| 22:56.50 | Peaceful | do you mean, set up /etc/zaptel.conf and run ztcfg? |
| 22:57.42 | [TK]D-Fender | Peaceful, after modprobing your card, yes |
| 22:58.26 | Peaceful | I assume, I'm 'rmmod'ing beforehand. |
| 22:58.27 | [TK]D-Fender | Peaceful, usually no need |
| 23:00.20 | *** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net) |
| 23:01.44 | *** join/#asterisk Steven_elvisda (n=Elvisda@202.47.107.60) |
| 23:01.53 | Peaceful | ztcfg still gives me: ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 23:02.25 | Peaceful | probably because there's still nothing in /proc/zaptel, even after recompiling, modprobe'ing, and ztcfg'ing |
| 23:02.56 | Peaceful | dmesg tells me: Zapata Telephony Interface Registered on major 196 |
| 23:03.01 | Peaceful | Zaptel Version: 1.4.4 |
| 23:03.05 | Peaceful | Zaptel Echo Canceller: MG2 |
| 23:04.26 | Peaceful | Googling "ZT_SPANCONFIG failed on span 1: No such device or address (6)" mostly just gets me posts about "your card must be broken", but I just moved this (working) card from an old machine to a new one. |
| 23:05.20 | Yourname` | Hmm, how can I set the number of rings before an extension should go to voicemail? |
| 23:05.43 | Peaceful | Another guy (from googling) said he had to poweroff, not reboot. But I've tried rebooting and powering off/on with no effect either. |
| 23:05.44 | [TK]D-Fender | Peaceful, pastebin "cat /proc/interrupts" and "dmesg" |
| 23:05.55 | [TK]D-Fender | ~pb |
| 23:05.56 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 23:06.40 | Yourname` | ~voicemail |
| 23:06.45 | [TK]D-Fender | And digium folk around? |
| 23:06.48 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 23:06.50 | [TK]D-Fender | any* |
| 23:06.56 | [TK]D-Fender | bkruse_home, TIMING! |
| 23:07.13 | [TK]D-Fender | bkruse_home, make: *** No rule to make target `install-inlcude', needed by `install-programs'. Stop. |
| 23:07.14 | Peaceful | dmesg is here: http://rafb.net/p/GxFI1r93.html |
| 23:07.33 | [TK]D-Fender | bkruse_home, typo in Zaptel 1.4.5 's makefile <------- |
| 23:08.00 | Peaceful | /proc/interrupts is here: http://rafb.net/p/a6fhYd15.html |
| 23:09.02 | Peaceful | I hit that typo today, too, when I thought 1.4.5 was the most current by looking in svn /tags/. Then I noticed 1.4.4 was listed as current on the web site |
| 23:09.09 | [TK]D-Fender | Peaceful, modprobe your card & zaptel again |
| 23:09.23 | [TK]D-Fender | Peaceful, 1.4.5. is current on their site |
| 23:09.43 | Peaceful | [TK]D-Fender, not here: http://asterisk.org/downloads |
| 23:09.55 | Peaceful | top right box, 1.4.4 is listed |
| 23:09.55 | tzafrir_laptop | [TK]D-Fender, already fixed in svn |
| 23:10.08 | [TK]D-Fender | Curios |
| 23:10.18 | Peaceful | I assumed that was because 1.4.5 was b0rked |
| 23:10.51 | Peaceful | ...though it still installed when I fixed the typo. |
| 23:11.03 | Peaceful | Could downgrading to 1.4.4 have something to do with my problem? |
| 23:11.28 | tzafrir_laptop | Peaceful, do you see the card in lspci? |
| 23:11.36 | Peaceful | tzafrir_laptop, nope |
| 23:11.53 | tzafrir_laptop | hmmm.... are you sure? |
| 23:12.02 | tzafrir_laptop | could you pastebin the output of lspci? |
| 23:12.12 | tzafrir_laptop | what card is it? |
| 23:12.16 | Peaceful | absolutely, lspci is: http://rafb.net/p/LpmoLK67.html |
| 23:12.34 | Peaceful | card is "Digium TE210P Dual T1/E1 PCI Interface" |
| 23:12.59 | Peaceful | driver is wct2xxp, which is just an alias for wct4xxp |
| 23:13.34 | tzafrir_laptop | first get the card to appear in lspci, then worry about drivers and such |
| 23:13.41 | Peaceful | One port has a red light, the other port is lights off |
| 23:13.56 | Peaceful | ok, I tried switching it to the only other available slot, no effect |
| 23:14.04 | tzafrir_laptop | it's not connected well, malfunctioning, or whatever |
| 23:14.24 | Peaceful | you think it's not fully seated? |
| 23:14.30 | tzafrir_laptop | maybe it's of the wrong type? PCI vs. PCIexpress? |
| 23:15.09 | Peaceful | I don't believe those slots are physically compatible -- anyway, it's in a PCI-X slot, I believe. |
| 23:15.13 | [TK]D-Fender | Peaceful, TE210P = 3.3V only. Maybe got it in the wrong slot, or the wrong board entirely. |
| 23:15.34 | Peaceful | Hmm, I'll grab the MoBo specs... |
| 23:16.02 | Peaceful | Supermicro X7DVL-E Dual Xeon Motherboard, 2 x GB LAN |
| 23:16.36 | [TK]D-Fender | Peaceful, I'd try rebooting it... |
| 23:16.42 | Peaceful | It's in a riser in a 2U case |
| 23:17.09 | Peaceful | I've rebooted twice, and powered off/on once |
| 23:17.36 | *** join/#asterisk tomcontr3 (n=tomcontr@145-79-246-201.adsl.terra.cl) |
| 23:17.37 | [TK]D-Fender | Peaceful, And just a warning that your e1000 might be rather unappreciated by your card.... |
| 23:17.48 | Peaceful | ? |
| 23:17.53 | tomcontr3 | hi, does anyone knows a good billing software for asterisk? |
| 23:18.08 | Yourname` | Hmm, maxsilence isn't putting the call to voicemail after 6 seconds.. |
| 23:18.18 | Peaceful | PCI-X * 2x 64-bit 133/100MHz PCI-X (3.3V) slots -- according to http://www.supermicro.com/products/motherboard/Xeon1333/5000V/X7DVL-E.cfm |
| 23:18.18 | Peaceful | "PCI-X |
| 23:18.18 | Peaceful | <PROTECTED> |
| 23:18.40 | Peaceful | er, that was weird. Stupid copy/paste almost flooded. |
| 23:19.25 | Peaceful | [TK]D-Fender, are you saying the onboard e1000 may be fighting with the card? |
| 23:20.04 | Qwell | e1000 is notoriously bad at interrupts, or something |
| 23:20.05 | [TK]D-Fender | Peaceful, A seperate issue, but yes. |
| 23:20.20 | Peaceful | My current production server with an identical card also has an e1000 - had no problems with it yet...that I know of. |
| 23:20.28 | Qwell | I don't know if that's changed recently or not |
| 23:20.30 | [TK]D-Fender | Qwell[], "cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory" <- * Addons error. I got with 1.4.2 |
| 23:20.32 | Qwell | I think it might've |
| 23:20.36 | Peaceful | identical card, meaning te120p |
| 23:20.48 | [TK]D-Fender | [root@localhost .libs]# ls -l |
| 23:20.49 | Peaceful | er, 210p |
| 23:20.58 | Qwell | [TK]D-Fender: it's unmaintained |
| 23:21.02 | [TK]D-Fender | -rwxr-xr-x 1 root root 1820314 Aug 18 19:23 libchan_h323.1.0.1 |
| 23:21.11 | [TK]D-Fender | Qwell[], meaning? |
| 23:21.16 | Qwell | meaning...use the in tree one |
| 23:21.55 | [TK]D-Fender | Qwell[], AKA Addons 1.4.2 = busted and not going to get fixed on your FTP/HTTP? |
| 23:22.23 | Qwell | it may eventually... I don't think anybody has reported a bug yet though |
| 23:23.05 | [TK]D-Fender | Qwell[], I hate the though of SVN-ing this stuff... |
| 23:23.15 | Qwell | addons rarely gets a release |
| 23:25.33 | Peaceful | No effect after rebooting (again). ztcfg still says "ZT_SPANCONFIG failed on span 1: No such device or address (6)" and there's still nothing in /proc/zaptel, and the card doesn't show up in lspci |
| 23:25.41 | Peaceful | boo |
| 23:29.03 | *** join/#asterisk robh71_ (n=robh71@host-65-124-86-25.entouch.net) |
| 23:31.11 | Peaceful | The riser card also has a RAID card on it, which seems to be working fine, since that's what I'm booting my system off of, but I'll go up and turn the machine off and wiggle the riser card anyway to make absolutely sure that its seated all the way. Any other suggestions for things to try? |
| 23:31.54 | tzafrir_laptop | Peaceful, again, first get the card to show on lspci |
| 23:32.18 | tzafrir_laptop | Without that, no use to get a driver to probe it from the PCI bus |
| 23:35.00 | rudholm | wow, g722 sounds really good :) |
| 23:35.56 | tomcontr3 | does any know which is the default root password for mysql in trixbox? |
| 23:36.20 | *** join/#asterisk elvisthedj (n=kris@72.24.181.169) |
| 23:36.44 | [TK]D-Fender | tomcontr3, Wrong channel <------ |
| 23:37.37 | tomcontr3 | im just asking just in case that some one knows, because no one answers in the other channel |
| 23:38.19 | elvisthedj | Hi all. Just wondering if anybody knows of any lcdproc clients for asterisk? I have a headless server and it sure would be nice :) |
| 23:39.04 | tsuk | tomcontr3: http://www.google.com/search?q=trixbox+mysql+default+password |
| 23:39.33 | Peaceful | tzafrir, hmmm, the bios says PCI-X slots 5 and 6 (which I assume are the top riser slots that' I've tried) are set to PCI-X 133MHz. |
| 23:39.55 | Peaceful | As digium says: "The TE410P is a 32-bit 33MHz card keyed for 3.3 volt operation." I'll try changing the BIOS setting to "PCI 33MHz" and see if that fixes it. |
| 23:40.03 | Peaceful | Sound logical? :) |
| 23:44.29 | Peaceful | aaaaaaaaaaaaaand no effect |
| 23:44.34 | elvisthedj | would there be a better channel to ask about an asterisk lcdproc client? (don't say #lcdproc :p ) |
| 23:45.00 | Peaceful | dang. It was set to "autodetect" the frequency anyway. I hoped that it was autodetecting wrong and forcing 33MHz would fix it. |
| 23:46.44 | Peaceful | elvisthedj, assuming that's an independent project, I'd go scope out their homepage and see if they have a regular channel or mailing lists etc. |
| 23:49.01 | elvisthedj | Peaceful: Thanks. At least now I know I'm not using an invisible font :) I just thought surely this had been done before. In fact, I see it here http://www.venturevoip.com/c3.php |
| 23:49.02 | *** join/#asterisk QbY (n=Kelvin@208.36.224.228.ptr.us.xo.net) |
| 23:50.07 | elvisthedj | Peaceful: not sure if they wrote their own client for that project or if they just borrowed an open source project (as they did with asterisk, astlinux, lcdproc, etc.) |
| 23:50.20 | elvisthedj | Peaceful: anyway.. i'll keep looking :) |
| 23:50.48 | Peaceful | I would bet there's only 4 or 5 people here anyway, and we all have probably never had lcd-based hardware to pipe asterisk info to anyway |
| 23:50.58 | Peaceful | good luck |
| 23:51.43 | QbY | I have a rather unique problem. We have endpoints connected to OpenSER, and Asterisk serving as an announcement/voicemail server. Would like to let Asterisk play certain announcements when certain conditions exist, for example a timeout on a call. So the call originates at the endpoint and goes to our softswitch, that advises the endpoint the audio is on port 12345 .. when the call fails and its forked to asteirsk for the announcement aster |
| 23:51.52 | QbY | (damn, didn't realize it was book) |
| 23:51.53 | *** join/#asterisk MdeP (n=mdep@51-90-22-190.adsl.tie.cl) |
| 23:52.09 | elvisthedj | maybe I should start buying lcd's for * junkies :) |
| 23:52.28 | tsuk | no i haven't either, but i'd love to |
| 23:52.33 | Peaceful | hehe |
| 23:54.48 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:55.34 | lesouvage | . |
| 23:55.38 | *** join/#asterisk kkn088 (n=kikoun@84.4.72.80) |
| 23:58.05 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 23:58.14 | Peaceful | Anyone else have a suggestion for getting a TE210P to show up in lspci? I've tried rebooting, powering off/on, making sure the card and riser are both seated firmly, changing slots in the rise, and manually setting the riser slots to PCI-33MHz. |
| 23:58.34 | Peaceful | And so far, no dice. One LED is solid red. The other LED is off. |
| 23:58.45 | Peaceful | Nothing shows up in lspci |
| 23:59.56 | tsuk | Peaceful: no expert, but you could try different slots. i read nasty things about irq sharing |