| 00:01.11 | *** join/#asterisk voltagex (n=voltagex@124-254-89-30-dsl.ispone.net.au) |
| 00:04.25 | ManxPower | jaycampbell: Polycom Soundpoint IP |
| 00:08.32 | *** join/#asterisk mivck (n=mv@190.83.23.93) |
| 00:10.44 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:10.44 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:14.41 | eryc | No ethernet interface found for seeding global EID You will have to set it manually. |
| 00:15.19 | *** part/#asterisk Vahram (n=VX@82.198.7.227.satgate.net) |
| 00:15.37 | eryc | How do i set it manually? |
| 00:15.41 | voltagex | why are extension 8 and 9 are not working with my dialplan? http://pastebin.ca/415992 |
| 00:17.20 | Corydon-w | eryc: see configs/dundi.conf.sample. You will want to set entityid=... |
| 00:17.52 | eryc | thanks! |
| 00:21.55 | ping2921 | anyone knows how i can play g.711 files on xp? |
| 00:25.09 | *** join/#asterisk frized17 (n=w@210.1.133.104) |
| 00:27.28 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-66-89.dsl.peoril.sbcglobal.net) |
| 00:27.49 | ManxPower | voltagex: do you see anything on the CLI? Are you using SIP or ZAP? |
| 00:28.12 | voltagex | ManxPower: hang on, making changes |
| 00:28.16 | ManxPower | voltagex: Simple. You don't have a priority 1 for either 8 or 9 |
| 00:28.25 | MACscr | when i do: sip show peers in the CLI, it shows my sip trunk as a monitored status of Unmonitored. How can i set it up so that it is monitored? |
| 00:28.40 | ManxPower | You can't have a (n)ext priority without having a (1)st priority |
| 00:29.11 | ManxPower | MACscr: you don't need it monitored. The qualify= option will make it monitored. |
| 00:29.58 | *** join/#asterisk anthonyc (n=x@adsl-69-111-224-67.dsl.frsn02.pacbell.net) |
| 00:30.01 | MACscr | ManxPower : i just wanted to have it "monitored" so that i could keep track of ping rates, etc |
| 00:31.18 | ManxPower | MACscr: that does not use ping. It measure the response time of the device when asterisk sends it a SIP OPTIONS packet. If the device is busy doing something it may ignore the OPTIONS packet and then asterisk shows that device offline. |
| 00:31.24 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 00:33.00 | frized17 | i just want to ask can asterisk work without the connectiong to pstn? |
| 00:33.09 | anthonyc | yes |
| 00:33.51 | *** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177814943.dsl.bell.ca) |
| 00:33.53 | frized17 | it means can all the user can talk using soft phone connecting in one asterisk server |
| 00:34.32 | fab5freddy | I just installed Trixbox 2.0, but when I use user: maint pass: password it says invalid login for the Admin gui? I'm lost! |
| 00:34.42 | MACscr | ManxPower : thanks, qualify worked perfectly |
| 00:34.46 | Strom_C | ~trixbox |
| 00:34.50 | jbot | trixbox is, like, junk - avoid. It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/, or known as 'sh1tbox', STAY AWAY! |
| 00:35.25 | MACscr | lol |
| 00:35.46 | MACscr | so funny how #asterisk hates gui's so much |
| 00:36.06 | fab5freddy | jbot: since trixbox already did the installation for me, can you help me configure from the console? |
| 00:36.11 | MACscr | heck, i bet a lot of the "community" wasnt even happy about astersiknow |
| 00:36.36 | ManxPower | MACscr: If the guis followed the config methods we are all familiar with it would not be a problem. |
| 00:36.38 | MACscr | fab5freddy : rofl, thats a bot |
| 00:36.51 | Strom_C | i think this proves my point :) |
| 00:37.01 | MACscr | fab5freddy : join the #freepbx channel as suggested |
| 00:37.14 | ManxPower | Most of us can't even start to help with the bastardized, overly complex, and downright TWISTED way the GUIS conifgure asterisk |
| 00:37.16 | MACscr | Strom_C : lol, true |
| 00:37.22 | ManxPower | it is basically like learning a totally different PBX. |
| 00:37.44 | Strom_C | i should start a gui based distro and call it nubbox |
| 00:38.03 | ManxPower | Strom_C I didn't know you were a machoist. |
| 00:38.13 | Strom_C | dogballs? |
| 00:38.20 | JT | or an ASM frontend to write your dialplans with - surely that will be popular |
| 00:38.30 | JT | pattern matchin with the efficiency of MOV |
| 00:38.52 | Strom_C | no no - make it possible to write dialplans by sheer telepathy alone |
| 00:39.03 | Strom_C | surely that will make it easier to tell the PBX "Do what I mean, not what I say" |
| 00:39.14 | JT | fuzzy logic AI pbx |
| 00:39.22 | Strom_C | haha |
| 00:39.29 | Strom_C | HAL900 |
| 00:39.31 | JT | you can make it learn |
| 00:39.33 | Strom_C | er |
| 00:39.34 | Strom_C | 9000 |
| 00:39.35 | JT | punish it and reward it |
| 00:39.48 | Strom_C | "I'm sorry, Dave, but your call did not go through." |
| 00:40.08 | JT | heh |
| 00:40.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 00:40.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:40.43 | fab5freddy | are you folks IT managers? |
| 00:41.10 | Strom_C | fab5freddy: I've heard the argument a hundred times that the secretary needs to be able to administer the phone system |
| 00:41.41 | JT | and the janitor needs to check the pri debug output |
| 00:42.15 | MACscr | lol |
| 00:42.46 | fab5freddy | just curious if you folks has experience with VMWare ESX Enterprise with VMotion |
| 00:43.10 | ManxPower | If the secretary needs to admin the phone system, then go buy a Nortel BCM. It has a nice GUI interface. No, the secratary won't be able to admin it without knowing telecom and PBXs, but at least you will stop pestering us. |
| 00:44.23 | ManxPower | One of my clients went out and bought a couple of BCMs for a few offices that I would not do an asterisk install cheap enough for them to use Asterisk. They went back to paying me to install asterisk at their offices. |
| 00:44.50 | Strom_C | as a consultant, when the client says "We need [someone] to handle moves, adds, and changes in addition to some other work they're already doing for us" and I hear "We're essentially too cheap to care about our phones" |
| 00:44.59 | JT | ManxPower: all the secretary or any non-tech user needs to be able to do on the PBX is change names of stations and recordings |
| 00:45.02 | JT | nothing else |
| 00:45.04 | JT | etc |
| 00:45.18 | Strom_C | because, inevitably, the secretary will inadvertently fuck up the whole system |
| 00:45.33 | Sweeper | but then you get paid for consultation |
| 00:45.35 | d00gster | where is the configuration does the incoming route config get stored? |
| 00:45.37 | Sweeper | to fix it :D |
| 00:45.38 | JT | voicemail settings, station name settings |
| 00:45.47 | ManxPower | The IT manager accidently erased all the Cisco access lists on the company firewall a few weeks ago. That was a festive day. |
| 00:45.48 | ping2921 | I have a bunch of audio files encoded in g.711 that would like to hear on winxp. anyones where to find the player? |
| 00:46.07 | ManxPower | d00gster: Asterisk does not really have the concept of an "incoming route" |
| 00:46.19 | Strom_C | Sweeper: I do not like getting calls where the client says "OH MY GOD WE FUCKED OUR PHONES YOU NEED TO COME OUT HERE RIGHT NOW" |
| 00:46.39 | ManxPower | It has Zap ports, for PSTN and IAX and SIP users for incoming calls |
| 00:46.39 | d00gster | so how does it know how to deal with incoming calls? |
| 00:46.47 | Sweeper | Strom_C: surely you charge convenience fees, right? |
| 00:46.52 | Sweeper | and overtime |
| 00:46.54 | Sweeper | etc etc :D |
| 00:47.03 | JT | point is it's still inconvenient |
| 00:47.08 | Strom_C | Sweeper: yes, I definitely do...but a happy client is ALWAYS better than a panicky client |
| 00:47.10 | ManxPower | d00gster: give me an example an incoming call on some protocol or port and I'll tell you. |
| 00:47.31 | d00gster | sip call through my provider |
| 00:47.46 | Sweeper | Strom_C: probably. but obstinate cheapasses learn quickly :p |
| 00:48.18 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 00:48.36 | ManxPower | d00gster: the call comes into asterisk and matches an entry in sip.conf. Usually the provider will send you the destination number. Then asterisk will check the [context] in extensions.conf (as specified by the context= line in that sip.conf entry) and try to find a matching exten => line for the destination number |
| 00:49.04 | ManxPower | the context= line in sip.conf is what links that device to the dialplan (extensions.conf) |
| 00:49.18 | d00gster | ok |
| 00:50.05 | ManxPower | Strom_C: but if a client is unhappy anyway, it is good for them to be panicky. |
| 00:51.03 | MACscr | hmm, i have read the wiki for debuggin, but i still cant seem to get my asterisk 1.2x system to show me better debug data |
| 00:51.17 | MACscr | anyway to filter out the registration info that comes every 60 seconds |
| 00:52.49 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 01:12.22 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 01:17.14 | mmartinn | Anyone know if there is a "Gentoo way" for getting asterisk addons for 1.4.x built in the voip overlay? An ebuild anywhere? |
| 01:17.33 | Qwell | mmartinn: packages are kinda...meh |
| 01:17.47 | Qwell | It's useless to debug if you have problems |
| 01:18.13 | mmartinn | Qwell: An ebuild will build it from source, and I can just leave the artifacts from the build process |
| 01:18.19 | *** join/#asterisk borisyaltsin (n=chris@wificlient-52-69.publicwireless.ualberta.ca) |
| 01:18.53 | mmartinn | I'm about to test the jabber addon... does that exist/sound futile for 1.4? |
| 01:19.02 | Qwell | it works.. |
| 01:19.18 | *** part/#asterisk borisyaltsin (n=chris@wificlient-52-69.publicwireless.ualberta.ca) |
| 01:19.19 | mmartinn | looks like i'm building from svn ;) |
| 01:21.19 | frized17 | what port will be used in asterisk? |
| 01:21.38 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:21.46 | Qwell | frized17: many |
| 01:22.39 | frized17 | Qwell im new in asterisk where can i find a basic configuration of asterisk? |
| 01:22.44 | Qwell | ~wikis |
| 01:22.53 | jbot | i guess wikis is http://www.voip-info.org |
| 01:22.53 | Qwell | ~book |
| 01:22.55 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:22.55 | frized17 | ah ok... |
| 01:22.57 | Qwell | stupid slow bot |
| 01:23.10 | frized17 | thanks |
| 01:23.37 | mmartinn | Does Asterisk-Addons 1.4.0 go with Asterisk 1.4.1? |
| 01:24.46 | Qwell | mmartinn: yeah |
| 01:24.49 | fbffff | freeat12five, make samples |
| 01:25.17 | mmartinn | Qwell: thanks |
| 01:30.59 | [TK]D-Fender | mmartinn, If you're starting new stuff may as well go CURRENT. |
| 01:31.50 | mmartinn | [TK]D-Fender: aye... just deleted the checkouts to do that |
| 01:33.02 | mmartinn | Still, doesn't appear anything after 1.4.0 is released for the addons |
| 01:33.12 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 01:33.25 | [TK]D-Fender | mmartinn, I'm referring to * 1.4.2 |
| 01:33.32 | BigCanOfTuna | How can I run a script as 'asterisk' on my linux machine? |
| 01:33.44 | mmartinn | Right, I did go to that :) |
| 01:33.53 | BigCanOfTuna | BigCanOfTuna: from the command line. |
| 01:34.15 | mmartinn | You answered your own question! |
| 01:34.45 | [TK]D-Fender | mmartinn, Don't interrupt him when he's talking to himself! |
| 01:34.51 | mmartinn | lol :) |
| 01:35.26 | mmartinn | BigCanOfTuna: Different distributions of linux have different ways they've standardized doing it -- what distro? |
| 01:35.35 | BigCanOfTuna | mmartinn: ubuntu |
| 01:36.08 | BigCanOfTuna | mmartinn: I would imagine it would be something like: su - asterisk -c 'ruby ./myscript' |
| 01:36.13 | BigCanOfTuna | mmartinn: ...but it doesn't work. |
| 01:36.32 | mmartinn | Does Ubuntu have an /etc/conf.d/? |
| 01:36.50 | eryc | how do i disable dundi? |
| 01:36.51 | mmartinn | BigCanOfTuna: I know you could do it from /etc/init.d/asterisk or some such, by editing that file |
| 01:37.17 | mmartinn | BigCanOfTuna: Assuming you're using the init script to start it |
| 01:37.29 | mmartinn | BigCanOfTuna: Did you just want from the command line? |
| 01:37.42 | BigCanOfTuna | mmartinn: yes, that is rigth, from the command line. |
| 01:39.45 | mmartinn | BigCanOfTuna: Can I ask what you are doing with 'ruby ./myscript' ? Have you tried successfully with just asterisk? |
| 01:40.12 | mmartinn | BigCanOfTuna: like su -c /usr/sbin/asterisk asterisk |
| 01:40.27 | mmartinn | BigCanOfTuna: Assuming 'which asterisk' tells you it was in /usr/sbin |
| 01:40.44 | BigCanOfTuna | mmartinn: it's a simple bot that moves call files to the outgoing directory....but I need it to run as asterisk so asterisk can manipulate the file at it's final destination. |
| 01:41.38 | mmartinn | BigCanOfTuna: Ah. Try su -c 'ruby ./myscript' asterisk |
| 01:42.45 | BigCanOfTuna | mmartinn: nah, didn't work....does it have anything to do with asterisk not having a shell? |
| 01:42.50 | mmartinn | BigCanOfTuna: You may also be interested in start-stop-daemon which can change users and create pid files for simple servers |
| 01:43.49 | mmartinn | BigCanOfTuna: Does this command show something with nologin? 'grep -i asterisk /etc/passwd' |
| 01:44.35 | mmartinn | BigCanOfTuna: If so, you may need to give it a shell; if you can't 'su asterisk' as root, fix that first. |
| 01:51.49 | *** join/#asterisk tonyb2006 (n=tonyb@2002:4571:29c2:0:0:0:0:1) |
| 01:52.17 | tonyb2006 | if I have a meetme conference running, and an SIP provider set up to call PSTN lines, can I call a PSTN line on the meetme conference |
| 01:53.51 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:57.11 | [TK]D-Fender | tonyb2006, You can use a phone on your system to call outside and then transfer them to the MeetMe, or you can use a Call-file or AMI originate event to callout and bridge them. |
| 01:57.24 | tonyb2006 | hmm |
| 01:59.08 | *** join/#asterisk b00mer (n=b00mer@pool-71-178-30-192.washdc.fios.verizon.net) |
| 01:59.14 | aricorn | can I connect asterisk to cisco call manager using SCCP like I would a skinny enabled phone? |
| 01:59.20 | *** join/#asterisk red9012 (n=marc3234@206-248-178-169.dsl.teksavvy.com) |
| 01:59.28 | BigCanOfTuna | Why would this happen? I start asterisk using the the init script, and as root, I move a call file into the outgoing directory....the call file disappears, but the call is never made. I start asterisk as root, copy the same file into the same directory, call is successfully made. |
| 02:00.44 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:01.33 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 02:03.20 | *** join/#asterisk Fieldy (i=GgT3WGhM@gentoo/contributor/Fieldy) |
| 02:04.06 | b00mer | Does anybody know if I can use chan_cellphone with AsteriskNow? or do I need roll my own? |
| 02:05.32 | b00mer | is there any asterisk distro/build that supports chan_cellphone? |
| 02:07.52 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:07.54 | tengulre | hi,all |
| 02:08.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:09.00 | *** join/#asterisk bawb2 (n=bawb2@17.14.124.24.cm.sunflower.com) |
| 02:09.01 | tengulre | which ports I need opened when I using Asterisk behind firewall?? |
| 02:09.26 | fbffff | all of them |
| 02:09.39 | JT | not true |
| 02:09.57 | JT | for sip: udp 5060 udp 10000-20000 depending on rtp.conf |
| 02:10.05 | tengulre | I don't know which RTP ports need opened? |
| 02:10.06 | JT | for iax: udp 4569 |
| 02:10.09 | tengulre | JT, yes! |
| 02:10.15 | JT | i just said |
| 02:10.15 | aricorn | udp 5060! |
| 02:10.27 | tengulre | JT: 10000-20000 all opened? |
| 02:10.37 | JT | yes |
| 02:11.08 | *** join/#asterisk mivck (n=mv@190.83.23.93) |
| 02:18.07 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 02:18.40 | bintut | what tool do you use for testing asterisk (sip)? |
| 02:19.50 | flenders | bintut: a softphone? |
| 02:20.04 | dc3aes | i know this isnt a nufone support channel but anyone else all of a sudden all outgoing calls dont send callerid? (nufone call summary confirms that I am sending it properly, just from nufone -> dialed number it gets stripped of) |
| 02:20.24 | *** join/#asterisk Fieldy (i=EmwYiVBZ@gentoo/contributor/Fieldy) |
| 02:20.45 | *** join/#asterisk AbsTradELic (n=vldmr@201.79.156.219) |
| 02:22.09 | AbsTradELic | anyone here from digium ? |
| 02:22.40 | tengulre | I m not, ;) |
| 02:22.56 | tengulre | anyone here from PlayBoy? |
| 02:23.06 | dc3aes | from nufone? |
| 02:23.17 | AbsTradELic | ok... its a test |
| 02:23.30 | AbsTradELic | one two tree... experience |
| 02:23.45 | AbsTradELic | testing.... sound one two tree |
| 02:23.51 | Corydon76-home | Why would someone from Digium need to be here? |
| 02:24.07 | tengulre | anybody know which softphone have soucecode and can be running under MS platform? |
| 02:24.17 | [TK]D-Fender | AbsTradELic, How about you ask you actual question and see if one NEEDS to be here to answer it... |
| 02:25.40 | AbsTradELic | huMruM |
| 02:26.11 | AbsTradELic | [TK]D-Fender: thank you |
| 02:26.18 | AbsTradELic | X:) |
| 02:26.39 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 02:29.15 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 02:29.30 | sumasuma | is there is Asterisk realtime for 1.4 ? |
| 02:29.40 | sumasuma | i had it for 1.2, but looks no more in 1.4 |
| 02:29.47 | [TK]D-Fender | AbsTradELic, So..... what is it? |
| 02:32.16 | AbsTradELic | [TK]D-Fender: sorry |
| 02:34.56 | mmartinn | Anyone know how to make jabber client module connect? Everything looks good but it says nothing about connecting -- just "not connected" when I do anything |
| 02:35.38 | b00mer | Does anybody know if I can use chan_cellphone with AsteriskNow? or do I need roll my own? |
| 02:35.50 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 02:36.51 | AbsTradELic | [TK]D-Fender: can you tell me your opition about a good asterisk book ? I have one here with me |
| 02:37.06 | [TK]D-Fender | ~book |
| 02:37.11 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:37.56 | *** join/#asterisk Fieldy (i=4PF190xx@gentoo/contributor/Fieldy) |
| 02:37.56 | AbsTradELic | oH... a bot, thank you |
| 02:38.00 | [TK]D-Fender | AbsTradELic, If you've got that, you've got enough to get off the ground. once you've got a basi system up, and have worked some phones ad some different kinds of lines and learned the dialplan you can extend your knowledge with the WIKI to go into further detail. |
| 02:38.12 | [TK]D-Fender | AbsTradELic, Yes, jbot is as it says... |
| 02:38.13 | [TK]D-Fender | ~jbot |
| 02:38.16 | jbot | jbot is, like, only marginally useful at best, He got a C- on his Turing Test, or dating the fembots, or a potty mouth, or some dimwit who learns crud. |
| 02:38.42 | AbsTradELic | X:) |
| 02:38.55 | [TK]D-Fender | ~lart mmartinn |
| 02:38.55 | jbot | beats the living hamstercrap out of mmartinn |
| 02:39.01 | mmartinn | ! |
| 02:39.04 | mmartinn | that's a new one |
| 02:39.07 | [TK]D-Fender | pwned |
| 02:39.20 | mmartinn | ~lart [TK]D-Fender |
| 02:39.20 | jbot | moos at [TK]D-Fender |
| 02:39.23 | mmartinn | d'oh |
| 02:39.28 | [TK]D-Fender | mooooooooo! |
| 02:39.33 | mmartinn | ~lart [TK]D-Fender |
| 02:39.33 | jbot | breaks out the Hoover and sucks up [TK]D-Fender |
| 02:39.53 | [TK]D-Fender | mmm... a free suck.... so much cheaper than down-town.... |
| 02:40.08 | AbsTradELic | [TK]D-Fender: I have this book... buy it today !! X;) |
| 02:40.27 | [TK]D-Fender | AbsTradELic, Congratulations. May I suggest you start READING it? :) |
| 02:40.46 | AbsTradELic | X:D |
| 02:40.59 | AbsTradELic | thank you again |
| 02:41.04 | AbsTradELic | X;) |
| 02:41.09 | *** join/#asterisk vishmaker (n=vishmake@202.83.33.94) |
| 02:41.52 | AbsTradELic | [TK]D-Fender: sure !! |
| 02:42.00 | AbsTradELic | X;) |
| 02:42.15 | mmartinn | ~piglatin jbot, do you have any other ninja commands? |
| 02:43.04 | AbsTradELic | muito gay |
| 02:43.12 | AbsTradELic | ahauahauaha |
| 02:43.14 | [TK]D-Fender | AbsTradELic, Anyways, what was your intended question earlier? Or was it just about other recommended books? |
| 02:43.21 | mmartinn | ~lart AbsTradELic |
| 02:43.21 | jbot | explains, ever so gently, that if AbsTradELic doesn't give the channel more information, they can't help |
| 02:43.31 | mmartinn | hmm |
| 02:43.48 | [TK]D-Fender | Strangely appropriate.... |
| 02:43.48 | mmartinn | I give up on the app_jabber thing; no support for it in Asterisk-Java from me... |
| 02:43.57 | AbsTradELic | [TK]D-Fender: intended question earlier |
| 02:44.41 | [TK]D-Fender | AbsTradELic, then you have the best/nly book I'd suggest fornow. |
| 02:44.42 | [TK]D-Fender | AbsTradELic, Start with that and take it step by step |
| 02:45.26 | AbsTradELic | ok... started. lets go !! |
| 02:45.30 | AbsTradELic | X;) |
| 02:47.20 | flenders | astra: brazilian? |
| 02:47.27 | flenders | AbsTradELic: brazilian? |
| 02:47.54 | *** join/#asterisk BBRodriguez (n=BBRodrig@p54B002E5.dip0.t-ipconnect.de) |
| 02:49.32 | AbsTradELic | flenders: yes !! |
| 02:49.39 | AbsTradELic | I am |
| 02:50.26 | AbsTradELic | well... english is not my native language, but I understand a lot |
| 02:51.16 | flenders | AbsTradELic: I'm a brazo too |
| 02:51.27 | AbsTradELic | too ? |
| 02:51.40 | file | In-A-Gadda-Da-Vida! |
| 02:52.11 | AbsTradELic | huM.... then I dont need to use english with you |
| 02:52.19 | AbsTradELic | X:D |
| 02:52.44 | flenders | yeah, but try not to write in portuguese here... not very polite I would say |
| 02:52.59 | voiper1 | !! Unexpected Channel selection 3 |
| 02:52.59 | voiper1 | <PROTECTED> |
| 02:53.21 | AbsTradELic | flenders: huM... I understand |
| 02:53.58 | ManxPower | voiper1: using BRI or PRI? |
| 02:54.06 | AbsTradELic | flenders: where are you from brazil ? |
| 02:54.22 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 02:54.22 | voiper1 | ManxPower: PRI te110p |
| 02:54.22 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 02:54.33 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 02:54.50 | flenders | sorry, by bitchx bitched |
| 02:54.58 | flenders | I was born in SP |
| 02:55.06 | ManxPower | voiper1: the restart message is normal. Asterisk does that to idle channels every once in a while or some telcos forget about the connection |
| 02:55.06 | *** part/#asterisk voltagex (n=voltagex@124-254-89-30-dsl.ispone.net.au) |
| 02:55.11 | flenders | but been living in sydney/australia for about 5 years |
| 02:55.12 | JT | Sao Pao? |
| 02:55.16 | ManxPower | bbiaw |
| 02:55.27 | flenders | JT: sao paulo |
| 02:55.38 | AbsTradELic | and actually live there ? |
| 02:55.45 | voiper1 | manxpower: thanks mate |
| 02:55.46 | flenders | yeah |
| 02:56.06 | flenders | going home for holidays in 4 weeks though |
| 02:56.07 | JT | flenders: damn, that's what my other bet was on, bad memory |
| 02:56.11 | AbsTradELic | I born in RJ... but just born !! |
| 02:56.45 | flenders | JT: it was close enough |
| 02:57.12 | AbsTradELic | flenders... I actually live in PB |
| 02:57.13 | flenders | AbsTradELic: I can see you don't like rio too much do you? :o) |
| 02:57.32 | flenders | AbsTradELic: pretty far |
| 02:58.35 | AbsTradELic | yeag |
| 02:58.38 | AbsTradELic | yeah |
| 02:58.44 | AbsTradELic | pretty far |
| 02:59.25 | AbsTradELic | I live here by a long time |
| 03:03.06 | *** join/#asterisk ManxPower (n=manxpowe@46.sub-75-203-156.myvzw.com) |
| 03:06.34 | *** join/#asterisk ctooley (n=ctooley@tx-71-2-127-90.dhcp.embarqhsd.net) |
| 03:09.58 | ctooley | Is there a good source of dialtone audio somewhere? |
| 03:12.04 | mmartinn | night folks :) |
| 03:12.16 | mmartinn | ~lart jbot |
| 03:12.16 | jbot | whacks mmartinn upside the head |
| 03:12.19 | mmartinn | d'oh |
| 03:12.28 | *** part/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 03:15.28 | ManxPower | ctooley: run Answer, Monitor, and Playtones |
| 03:16.21 | *** join/#asterisk voltagex (n=voltagex@124-254-89-30-dsl.ispone.net.au) |
| 03:17.16 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 03:23.51 | ctooley | ok, I'll give that a shot ManxPower |
| 03:24.04 | voltagex | anyone active right now? |
| 03:24.56 | ctooley | voltagex, depends on how you define "active" |
| 03:25.12 | [TK]D-Fender | voltagex, Odds are those in front of a computer are SEDENTARY... but thats just mea... |
| 03:25.12 | ctooley | I'm activating my "get another beer routine", does that count? |
| 03:25.18 | voltagex | lol |
| 03:25.21 | voltagex | ok. |
| 03:25.38 | voltagex | Asterisk doesn't seem to be able to find the phone attached to my ATA |
| 03:25.47 | voltagex | I can call from it, just not to it |
| 03:26.49 | ctooley | sounds to me like Asterisk can't find your ATA |
| 03:26.55 | ctooley | not the phone attached to it |
| 03:27.09 | ctooley | is your ATA registering or is it a fixed IP address? |
| 03:27.15 | voltagex | fixed |
| 03:28.44 | voltagex | ok changing host= to the IP makes asterisk able to ring it, but I can't pick the call up |
| 03:28.44 | ctooley | do you have qualify=yes? |
| 03:29.02 | ctooley | you can't pick it up? |
| 03:29.09 | ctooley | you mean the handset is glued to the base? |
| 03:29.14 | voltagex | as in, the phone doesn't ring |
| 03:29.31 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:29.45 | voltagex | app_dial.c:1090 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 03:30.23 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:30.52 | ctooley | How is it ringing if you get that error? |
| 03:31.18 | voltagex | ok I'm trying to call it from a softphone, it's ringing on the softphone's end |
| 03:31.39 | voltagex | phone1/phone1 10.1.1.1 N 5060 UNREACHABLE so how do I make it reachable? |
| 03:31.41 | ctooley | yeah, it can't find the ip address |
| 03:31.54 | ctooley | can you ping 10.1.1.1 from the server that asterisk is on? |
| 03:31.58 | voltagex | but I'm calling from within my own network, to my own network :/ |
| 03:32.06 | ctooley | is it going through NAT or any kind of firewall? |
| 03:32.12 | voltagex | no |
| 03:32.12 | ctooley | are you sure you can ping the device? |
| 03:32.16 | voltagex | yes |
| 03:32.28 | voltagex | 64 bytes from 10.1.1.1: icmp_seq=1 ttl=64 time=1.79 ms |
| 03:33.13 | ctooley | sip debug peer phone1 |
| 03:33.31 | ctooley | do you have "qualify=yes" set in sip.conf for that peer? |
| 03:33.50 | voltagex | yes |
| 03:34.32 | voltagex | ok now it goes straight to error |
| 03:34.37 | voltagex | sip phone doesn't even ring |
| 03:35.11 | [TK]D-Fender | voltagex, reset your ATA. |
| 03:35.44 | voltagex | ok |
| 03:35.56 | voltagex | may knock me offline as the ATA is in my DSL modem |
| 03:36.02 | JT | haha |
| 03:36.30 | [TK]D-Fender | voltagex, Keep in mind several like that will not let you ue the ATA ion the INTERNAL INTERFACE. |
| 03:37.20 | voltagex | [TK]D-Fender: I'm sure I had this working before I wiped my configs in a drunken stupor |
| 03:37.42 | [TK]D-Fender | voltagex, Sober up, and pastebin what you've got. |
| 03:37.52 | voltagex | lol, this was several days ago |
| 03:38.01 | voltagex | ok one minute |
| 03:38.21 | JT | sometimes it takes several days... |
| 03:39.03 | voltagex | I officer assure you I'm judge as a sober. |
| 03:39.34 | JT | alright, step out of the vehicle sir |
| 03:41.09 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 03:45.40 | *** join/#asterisk bawb2 (n=bawb2@17.14.124.24.cm.sunflower.com) |
| 03:48.24 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 03:52.36 | *** join/#asterisk bmg505 (n=leon@c1-208-7.rndf.isadsl.co.za) |
| 03:54.19 | bintut | flenders: sorry, i just came back.. nope, not a softphone. a testing tool for asterisk(sip) pbx. like, stress testing tool, vulnerability, etc. |
| 03:58.45 | flenders | voltagex: what router you have? |
| 03:59.00 | flenders | voltagex: is it the WRT54GP2? |
| 04:08.40 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 04:09.11 | bsd_tech | wrt54gp2 v1 or v2 |
| 04:09.13 | bsd_tech | ? |
| 04:11.53 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 04:16.42 | sumasuma | how can i inject a wav file into a bridged call ? |
| 04:19.09 | flenders | bsd_tech: I didn't even know that there were 2 versions |
| 04:19.37 | bsd_tech | there is the old ble and black cased unit |
| 04:19.47 | bsd_tech | and the new silver cased unit |
| 04:21.44 | flenders | bsd_tech: oh ok... mine is the old blue/black |
| 04:21.56 | bsd_tech | ok its a pain |
| 04:22.06 | flenders | bsd_tech: can you use the voip ports on the internal interface on the new one? |
| 04:22.07 | bsd_tech | I have one and there is a bug in the firmware |
| 04:22.58 | bsd_tech | the wifi is off by 1 channel |
| 04:23.03 | bsd_tech | in the old one |
| 04:23.28 | bsd_tech | the new one openwrt is workingon a port with asterisk and voip port support |
| 04:23.40 | De_Mon | how do I find out why a resournce was unable to be unloaded? |
| 04:24.01 | JT | sumasuma: you modify dial, i think |
| 04:25.03 | JunK-Y | sumasuma: you throw them in a meetme conf and u stream on the meetme conf. |
| 04:25.24 | JunK-Y | De_Mon: which ressources? |
| 04:25.48 | JunK-Y | u cant unload them if usecount > 0 or it some other mod depends on it. |
| 04:26.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:28.15 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:32.07 | AbsTradELic | ~book |
| 04:32.29 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:37.07 | *** join/#asterisk brian (i=brian@unaffiliated/brian) |
| 04:38.30 | De_Mon | JunK-Y it was chan_sip, I was looking for a screen that told me what was using it, or how many things |
| 04:38.47 | De_Mon | there wern't any calls, or modules loaded (that I know of) that would depend on it... |
| 04:39.04 | De_Mon | Geez! autoload just bit me in the ass, HARD |
| 04:39.26 | De_Mon | chan_sipTCPtest.so chan_sipBroken.so and chan_sip.so were all loading and asterisk did not like that one bit |
| 04:39.26 | JunK-Y | no actives calls and ya were unable to unload chan_sip.so? sounds odd. |
| 04:39.38 | *** join/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
| 04:39.44 | De_Mon | JunK-Y ya, thats what I said ;) |
| 04:40.36 | De_Mon | ah well, server is back up with all the bad modules moved somewhere safe... back to sleep. |
| 04:42.11 | sumasuma | when i put the TDM400P card inside this motherboard it did not work, any help? http://resources.mini-box.com/online/MBD-J-J7F2WE1G5D/moreimages/image1.jpg |
| 04:43.42 | Strom_M | define "did not work" |
| 04:44.03 | sumasuma | when i do lspci, it did not show the card itself |
| 04:44.51 | Strom_M | are you sure the card is seated and the PCI bus isn't deactivated? |
| 04:44.57 | JT | De_Mon: did you enjoy the bite? |
| 04:45.28 | sumasuma | Strom_M: I need to check whether it is deactivated or not, i will check in bios and be back |
| 04:45.49 | sumasuma | thanks for that pointing |
| 04:49.43 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 04:50.31 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 04:50.31 | IOscanner | anyone know if there is an FCC id for Asterisk? |
| 04:50.37 | JT | ... |
| 04:50.40 | JT | it's software |
| 04:50.49 | JT | software doesn't generally have FCC IDs |
| 04:51.00 | ZX81 | the TDM cards might |
| 04:51.02 | IOscanner | if you use asterisk as a Dialer it requires it to register it with the state. |
| 04:51.17 | JT | what? |
| 04:51.19 | ZX81 | the hardware or software? |
| 04:51.27 | IOscanner | that is true. We could put a card in and use the ID. |
| 04:51.37 | ZX81 | yep |
| 04:51.39 | JT | what requires what to register? |
| 04:52.02 | IOscanner | Each state if you are going to use a dialer for a call center |
| 04:52.11 | IOscanner | it has to be registered with the state |
| 04:52.15 | putzz | does any one know how fast dtmf tones can be received? is it 100ms in between them or is that too fast? |
| 04:52.23 | ZX81 | I require every to register with me - I call it Free Cash Cheques - you send me a cheque written out for cash and its free :) |
| 04:52.29 | ZX81 | *every1 |
| 04:52.43 | IOscanner | lol |
| 04:52.55 | ZX81 | putzz: should be sweet whatever unless its inband |
| 04:53.03 | ZX81 | overlapping might screw things |
| 04:53.26 | putzz | im using RFC2833 |
| 04:53.33 | ZX81 | yeah should be sweet then |
| 04:53.42 | putzz | its not catching them at 100ms in between so it should be something else then |
| 04:53.43 | putzz | ;-) |
| 04:53.50 | ZX81 | maybe the lack of VLDTMF in 1.2 could cause issues |
| 04:53.59 | ZX81 | what version of asterisk you running? |
| 04:54.05 | putzz | 1.2 |
| 04:54.18 | putzz | 1.2.13 |
| 04:54.19 | ZX81 | talking to another asterisk machine or what? |
| 04:54.21 | putzz | to be exact |
| 04:54.35 | putzz | well its receiving call from pstn to did to asterisk |
| 04:54.38 | putzz | then catching the tones |
| 04:54.47 | *** join/#asterisk Fieldy (i=kJjRJqjO@gentoo/contributor/Fieldy) |
| 04:54.57 | ZX81 | you can edit /etc/asterisk/logger.conf |
| 04:55.02 | ZX81 | and turn on dtmf logging |
| 04:55.05 | *** join/#asterisk un_j (n=andrzej@216.70.188.32) |
| 04:55.11 | ZX81 | then restart asterisk or do a logger reload |
| 04:55.16 | ZX81 | and see what happens |
| 04:55.18 | un_j | hey how to create extension that will give you a signal - e.g. after dialing "9"] |
| 04:55.36 | ZX81 | un_j you mean like play the dialtone again? |
| 04:55.40 | ZX81 | ignorepat |
| 04:55.52 | ZX81 | think ignorepat => 9 |
| 04:55.59 | un_j | yes dial tone again |
| 04:56.09 | putzz | ZX81: what do u think the most I can push in ms between tones? |
| 04:56.20 | putzz | I want to try 50ms is that pushing it? |
| 04:56.37 | *** join/#asterisk TsimShaTsui (n=BBRodrig@p54B02654.dip0.t-ipconnect.de) |
| 04:56.39 | ZX81 | putzz try it and see |
| 04:56.45 | putzz | k thanks |
| 04:56.51 | ZX81 | the only likely problem is the length |
| 04:56.55 | ZX81 | not the time between |
| 04:57.18 | ZX81 | i.e. if you start them every 300ms with 100ms between and asterisk makes a 350ms tone it wont work |
| 04:57.36 | ZX81 | just turn on logging for dtmf |
| 04:57.41 | ZX81 | and see how it goes :) |
| 05:00.11 | *** join/#asterisk Fieldy (n=toon@gentoo/contributor/Fieldy) |
| 05:00.53 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 05:00.55 | zeeesh | hi |
| 05:05.11 | IOscanner | I am creating calls via Asterisk Manageer port. Anyone know how fast and how many this can support? |
| 05:05.54 | *** join/#asterisk FieldySnuts (n=toon@gentoo/contributor/Fieldy) |
| 05:05.56 | JunK-Y | a lot. |
| 05:25.14 | Juggie | do them async so the manager doesnt wait for a connection before returning control to you |
| 05:27.45 | *** join/#asterisk Fieldy (n=toon@gentoo/contributor/Fieldy) |
| 05:41.14 | dc3aes | i notice 99% of the volume of posts is in 'support', I am curious if there is a "repository' or otherwise organized collection of user projects/scripts/addons/etc? |
| 05:42.24 | *** join/#asterisk annonimous (n=annonimo@189.136.64.167) |
| 05:42.29 | annonimous | hi! |
| 05:46.19 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 05:48.24 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 05:50.50 | *** join/#asterisk newbie1 (n=root@202.161.189.44) |
| 05:55.42 | annonimous | hiya! |
| 05:57.02 | annonimous | i have troubles with a sip trunk cause the trunk registers and make calls but if im talking about 10 minutes or 4 minutes, the trunk gets "sticked" and if i need to redial the asterisk tells "all circuits are busy now pls try again later" and i need to reload the asterisk to get the trunk working, what command i need to put in my box to get right this trouble? |
| 06:04.10 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:04.52 | *** join/#asterisk SaiNt_D (n=saint@161.139.212.82) |
| 06:05.37 | SaiNt_D | hello guys |
| 06:06.00 | SaiNt_D | i hv a problem with the voicemail interface...can anyone help? |
| 06:09.36 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 06:09.36 | SaiNt_D | i hv a problem with webvmail interface....can anyone help? |
| 06:10.34 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:11.02 | SaiNt_D | i hv a problem with webvmail interface....can anyone help? |
| 06:12.30 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 06:12.31 | JT | SaiNt_D: wtf, stfu |
| 06:12.38 | annonimous | wtf? |
| 06:12.42 | annonimous | stfu? |
| 06:12.50 | SaiNt_D | ? |
| 06:12.52 | JT | what the fuck, shut the fuck up :) |
| 06:13.02 | JT | repeating like that is extremely rude |
| 06:13.02 | SaiNt_D | y must u b so mean? |
| 06:13.06 | annonimous | JT, lol |
| 06:13.13 | JT | because you are making people irritated here |
| 06:13.23 | JT | please learn some netiquette |
| 06:13.37 | SaiNt_D | i c u r the only one making noises |
| 06:13.39 | annonimous | humm JT i have posted a question but nobody answer i need to republish or only wait?? |
| 06:14.11 | JT | SaiNt_D: everyone else is either not here, or silently groaning to themselves and ignoring you |
| 06:14.25 | JT | there's a saying |
| 06:14.27 | SaiNt_D | thanks for the info |
| 06:14.36 | JT | ask a specific question and you might get a specific answer |
| 06:14.48 | SaiNt_D | i'm sorry |
| 06:14.51 | JT | the chances of people helping you without enough info, is low |
| 06:15.02 | SaiNt_D | i just started on this asterisk |
| 06:15.18 | SaiNt_D | so i cant give u exactly what the prob is |
| 06:15.18 | JT | SaiNt_D: no probs, sorry i snapped :) |
| 06:18.48 | annonimous | JT, do you have experience with sip trunks? |
| 06:19.19 | JT | annonimous: are you using freepbx or trixbox? |
| 06:19.40 | annonimous | JT, freepbx why? |
| 06:20.25 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 06:20.56 | *** join/#asterisk DoDaT69 (i=Tracer@66-240-110-115.atl.fdn.com) |
| 06:21.04 | DoDaT69 | anyone install asterisk in fbsd? |
| 06:21.23 | DoDaT69 | I am having a hellofa time with a sangoma card in fbsd |
| 06:24.49 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:29.00 | JT | DoDaT69: umm is that even at all supported? |
| 06:30.09 | DoDaT69 | yea |
| 06:30.12 | DoDaT69 | they have drivers for it |
| 06:30.30 | DoDaT69 | but the problem I am having is after the driver install it patches zaptel, which has to be recompiled |
| 06:30.41 | DoDaT69 | and well, installing from ports, its kinda hard to recompile |
| 06:32.22 | DoDaT69 | Error: Analog card only supported in TDM mode! |
| 06:32.26 | DoDaT69 | thats the error that ends it all |
| 06:32.52 | JT | what card? |
| 06:33.27 | DoDaT69 | a200d |
| 06:33.47 | JT | ah ok |
| 06:35.26 | DoDaT69 | its making me batty |
| 06:38.22 | *** join/#asterisk garreel (n=garreel@host143-34-dynamic.17-87-r.retail.telecomitalia.it) |
| 06:51.27 | *** join/#asterisk Zaphod_Biblbroks (n=use@80.93.248.52) |
| 07:02.46 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 07:06.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:09.51 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 07:10.28 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 07:10.41 | linagee | is there a distributed real time blacklist for caller IDs of telemarketers? ;) |
| 07:11.10 | gordonjcp | don't know, but if not it's a good idea! |
| 07:11.11 | *** join/#asterisk Avochelm (n=damien__@gw-morphett.koalatelecom.com.au) |
| 07:11.29 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 07:11.44 | gordonjcp | I know if I google some of the numbers that crop up in my logs as missed calls from American numbers, they seem to be telemarketers |
| 07:12.15 | linagee | gordonjcp: i'd also like simple "are you a human" tests. ;) "what is 1+1? please press the appropriate key on your phone." |
| 07:13.06 | linagee | gordonjcp: it's interesting that when advertisers actually use a caller ID instead of just "blank", calling it back often leads to a "call could not go through" message. lol. (likely just spoken by their asterisk box or something) |
| 07:13.11 | *** join/#asterisk wwq222 (n=chatzill@c-71-231-5-6.hsd1.or.comcast.net) |
| 07:13.45 | gordonjcp | linagee: my UK landline is on the do not call list, which is good |
| 07:13.52 | linagee | gordonjcp: in that way, asterisk is kind of evil. lol. putting the ability to do anything with the phone in the hands of spammers. :( |
| 07:13.55 | wwq222 | Hi does anyone know how to get Asterisk to queue outgoing calls (we only have a fixed number of lines)without writing your own app that controls the number of files in the outgoing directory? |
| 07:14.08 | gordonjcp | however I have a premium-rate number formatted like a mobile number which I put on survey forms, etc. |
| 07:14.15 | JT | linagee: heh, one can imagine evil schemes ;) |
| 07:14.25 | gordonjcp | linagee: I try and get that one out there as much as possible |
| 07:14.39 | JT | gordonjcp: rofl! |
| 07:14.51 | gordonjcp | ;-) |
| 07:14.55 | JT | are the monthlies expensive for a premium rate number? |
| 07:15.01 | gordonjcp | nope |
| 07:15.03 | linagee | JT: "hello, this is the president. we are in a dire crisis. if you'd like to donate $1 to your government and help out, please touch tone your CC numbers now" hahha. ;) |
| 07:15.21 | linagee | JT: watch their eyes roll when a $500 transaction appears. lol |
| 07:15.30 | JT | gordonjcp: what country premium rate? |
| 07:15.44 | gordonjcp | JT: it's from voipuser, it's (I think) 50p/minute |
| 07:15.48 | linagee | JT: you have to ask, what is reality? heh |
| 07:15.54 | linagee | totally perceived |
| 07:16.04 | JT | linagee: heh, yeah i was thinking you could cause panic by making a fake "emergency broadcast system" |
| 07:16.06 | gordonjcp | but it starts 07031, and 07xxx is generally used for mobiles... |
| 07:16.16 | JT | call a whole street or suburb, at once, or almost at once |
| 07:16.20 | JT | with a recorded message |
| 07:16.45 | linagee | JT: and be banned from all telephone switches faster than sh!t hits a fan. ;) |
| 07:16.53 | JT | yeah no shit |
| 07:16.59 | linagee | (or at least you'd hope that's the way it would work) |
| 07:17.01 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:17.02 | JT | you'd need some anonymitiy ;) |
| 07:17.33 | linagee | JT: how is making people panic profitable though? |
| 07:17.50 | JT | i dunno, ask fox news |
| 07:18.02 | linagee | they just sell commercials |
| 07:18.13 | gordonjcp | JT: isn't it illegal to ring people up with a recorded message? |
| 07:18.13 | JT | and hysteria |
| 07:18.20 | linagee | JT: so sell prime time sky writting time. lol |
| 07:18.22 | JT | gordonjcp: most likely |
| 07:18.32 | JT | gordonjcp: it's probably also illegal to cause mass panic |
| 07:18.38 | linagee | JT: as everyone runs outside because of the disaster message, they see a coke ad in the sky |
| 07:18.39 | linagee | or something |
| 07:18.40 | gordonjcp | JT: I'm pretty sure it is in the UK, don't know about the US though |
| 07:18.49 | JT | gordonjcp: and make false statements about national security |
| 07:19.08 | linagee | JT: unlikely actually. people are already addicted to coke. it doesn't need to be advertised any more. maybe some other dire compant |
| 07:19.10 | linagee | y |
| 07:19.10 | gordonjcp | JT: that seems a pretty open thing |
| 07:19.20 | JT | heh |
| 07:19.29 | JT | a radio station here did a pretty well done prank |
| 07:19.45 | JT | they got someone to record a bunch of prompts, and they chucked them all on a pc |
| 07:19.58 | linagee | JT: it's illegal to cause a mass panic? LOL. how about putting circuit boards strategically placed around a town with LED messages on them? lol. it's a bomb! it's a bomb! hahaha |
| 07:20.54 | linagee | all it would take is analyzing one board by someone competant who would say, "this isn't a bomb....?" to stop the idiocy |
| 07:21.28 | gordonjcp | linagee: you missed out the not noticing it for two weeks bit |
| 07:21.31 | JT | they called up the announcer's aunt who is outside of the radio coverage area, and had a recording "hi, you have been randomly selected to donate $50 to kids with cancer, if you would like to donate, say yes" "no" "i'm sorry, i did not understand that, do you want to donate $50?" "NO" "press 1 or hang up to donate $50" |
| 07:21.34 | linagee | gordonjcp: lol |
| 07:22.21 | JT | she calls out to her husband to whinge about the stupid thing, then hangs up, then they ring her back "hi, and thanks for donating $50 for kids with cancer, if you would like to donate another $50, say yes" |
| 07:22.35 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 07:22.53 | linagee | JT: pissing people off is a good way to get sued |
| 07:23.04 | linagee | just keep going down that road long enough. jheh |
| 07:23.18 | JT | linagee: it was the announcer's aunt, they admitted to it after that bit, and she had a chuckle over it |
| 07:23.30 | JT | it was funny as hell though |
| 07:23.34 | JT | she was so annoyed |
| 07:23.43 | linagee | JT: what if they did it to her like 100 times? hah |
| 07:23.49 | linagee | "hang up again to donate another $50" |
| 07:23.52 | JT | and the announcer knew she hated IVRs/speech recognition |
| 07:23.55 | linagee | likely she would just stop answering |
| 07:24.09 | JT | yeah but then it would've stopped being funny |
| 07:24.12 | JT | and just tedious |
| 07:25.20 | linagee | huh? |
| 07:25.28 | linagee | JT: tedious? for a human or for a computer? |
| 07:25.39 | JT | listeners |
| 07:25.44 | JT | of the radio station |
| 07:26.03 | JT | they'd probably start to empathise with the victim too much |
| 07:26.16 | linagee | interesting. but they wouldn't if it was just a few times |
| 07:26.29 | JT | yeah but airtime is expensive i guess |
| 07:26.50 | linagee | it's a fixed cost. heh |
| 07:27.14 | linagee | only if you divide it and say the equipment has costs does it become "expensive" |
| 07:27.44 | linagee | wait, maybe it always is. does clearchannel lease out it's licensese? |
| 07:28.00 | linagee | that's why i stopped listening to radio like 5 years ago. it's all a stupid scam by a company called clearchannel |
| 07:29.07 | JT | not in australia, but we have other alliances |
| 07:29.23 | linagee | s/alliances/monopolies/ |
| 07:30.53 | *** join/#asterisk harlequin516 (n=sham@styk.net) |
| 07:31.39 | harlequin516 | How many SIP ulaw calls can I run through to my asterisk box through my 384Kup DSL? |
| 07:32.03 | JT | harlequin516: no more than 4 |
| 07:32.45 | harlequin516 | Is it simply #calls / bandwidth? Or is it more complicated? |
| 07:33.04 | JT | it also depends how good your link is in reality |
| 07:35.09 | harlequin516 | My speed test shows 717Kb/s Upload |
| 07:35.37 | JT | sounds like bullshit |
| 07:36.04 | putzz | in bits maybe kbits then its = 100kb |
| 07:36.13 | putzz | isnt it? |
| 07:36.20 | putzz | almost 100kb |
| 07:43.46 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 07:44.30 | harlequin516 | I stated wrong before about 384Kb... I just upgraded. |
| 07:45.26 | JT | anyone here know trelane? |
| 07:46.04 | *** join/#asterisk lasse123 (n=Lasse@int-gw.algitech.com) |
| 07:47.25 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:50.05 | wwq222 | Hi does anyone know how to get Asterisk to queue outgoing calls (we only have a fixed number of lines)without writing your own app that controls the number of files in the outgoing directory? |
| 07:51.25 | hermuli | can anyone help me with Festival? |
| 07:51.47 | lasse123 | Hi, does anyone have a clue why the hangup/busy does not get through to the PSTN provider in the following case? I make a call: Cellphone->PSTN-provider->(SIP)->My Asterisk->(SIP)->My OpenSER->(SIP)->My Sip phone, everything works fine so far sound in both directions and whatnot, but when i hangup the call on the SIP phone, Asterisk does not forward the hangup to the PSTN provider...it just kills the legs of the call but no SIP message is sent to |
| 07:51.47 | lasse123 | the PSTN provider. |
| 07:53.51 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 07:54.35 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
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| 08:03.50 | *** join/#asterisk AJaymn (i=AJ14@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 08:04.11 | AJaymn | Anyone know if someones came up with a script to be able to take credit card payments by touch tones? |
| 08:06.13 | harlequin516 | MGCP is on by default... Why do I need this? |
| 08:07.26 | harlequin516 | Does it provide anything to SIP or IAX channels? |
| 08:07.36 | harlequin516 | Or Dundi? |
| 08:10.39 | patrickv0x | any sysmaster user here ? |
| 08:11.49 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 08:14.08 | hermuli | could anyone help me with Festival? |
| 08:15.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:16.35 | *** join/#asterisk dj-fu (i=ajc@unaffiliated/dj-fu) |
| 08:19.07 | badcfe | WARNING[2704]: channel.c:760 ast_channel_alloc: Channel |
| 08:19.09 | badcfe | allocation failed: Can't create alert pipe! |
| 08:19.20 | badcfe | anyone know what that means? |
| 08:19.34 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 08:19.34 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:20.46 | badcfe | WARNING[2704]: chan_sip.c:3782 sip_new: Unable to |
| 08:20.47 | badcfe | allocate AST channel structure for SIP channel |
| 08:21.13 | *** join/#asterisk juice (i=juice@mo-76-0-10-6.dhcp.embarqhsd.net) |
| 08:21.49 | badcfe | WARNING[2704]: channel.c:760 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! WARNING[2704]: chan_sip.c:3782 sip_new: Unable to WARNING[2704]: chan_sip.c:3782 sip_new: Unable to allocate AST channel structure for SIP channel NOTICE[2704]: chan_sip.c:13646 handle_request_invite: |
| 08:21.56 | badcfe | Unable to create/find SIP channel for this INVITE |
| 08:30.09 | lasse123 | Hi, does anyone have a clue why the hangup/busy does not get through to the PSTN provider in the following case? I make a call: Cellphone->PSTN-provider->(SIP)->My Asterisk 1.4.2->(SIP)->My OpenSER->(SIP)->My Sip phone, everything works fine so far sound in both directions and whatnot, but when i hangup the call on the SIP phone, Asterisk does not forward the hangup to the PSTN provider...it just kills the legs of the call but no SIP message is se |
| 08:30.09 | lasse123 | nt to the PSTN provider. Is there some setting to control what messages that are forwarded? |
| 08:30.18 | lasse123 | no |
| 08:31.38 | *** join/#asterisk apardo (n=apardo@87.217.145.14) |
| 08:39.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:44.39 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 08:54.47 | *** join/#asterisk denzs (n=denzs@carbon.gonicus.de) |
| 08:55.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:55.45 | denzs | hi, iam using *-1.2.14 bristuffed, after updating my snom 360 firmware to 6.5.8 dpickup isnt working anymore.....is there something i have to change? |
| 09:10.21 | denzs | before upgrading i used fw 4.1 on the snom |
| 09:19.54 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
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| 09:34.12 | *** join/#asterisk xtr-II (i=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 09:35.34 | hermuli | why do i get this? "Function CHANNEL not registered" when i try this: "exten=1004,1,Set(CHANNEL(language)=fr)" |
| 09:36.21 | *** join/#asterisk Corydon76-home (i=brown@pdpc/supporter/sustaining/Corydon76-home) |
| 09:36.21 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 09:36.53 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 09:37.32 | *** join/#asterisk tparcina (n=tparcina@cisco16.fesb.hr) |
| 09:40.35 | badcfe | WARNING[2704]: channel.c:760 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! WARNING[2704]: chan_sip.c:3782 sip_new: Unable to WARNING[2704]: chan_sip.c:3782 sip_new: Unable to allocate AST channel structure for SIP channel NOTICE[2704]: chan_sip.c:13646 handle_request_invite: Unable to create/find SIP channel for this INVITE |
| 09:40.44 | badcfe | anyone known what trigger this messages? |
| 09:54.51 | *** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
| 09:56.17 | SoftIce | hi, please can somebody show me how to this using did (exten) in the begining, this is my current string exten => s,1,Gotoif($[ "${DNID:0:4}" = "2224" ]?ContextName,s,1) how do I do it like exten => 222,1, GOTO ContextName ? |
| 09:56.31 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-214247.home.otenet.gr) |
| 09:59.21 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:02.26 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 10:05.57 | *** join/#asterisk kgx (n=karuna@60.234.196.160) |
| 10:06.16 | hermuli | why do i get this: "Function CHANNEL not registered" when i try this: "exten=1004,1,Set(CHANNEL(language)=fr)"??? |
| 10:14.26 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 10:23.52 | hermuli | *CLI> show functions |
| 10:23.52 | hermuli | No such command 'show functions' (type 'help' for help) |
| 10:23.58 | hermuli | what is going on with my asterisk? |
| 10:24.10 | hermuli | i'm totally utterly lost |
| 10:25.04 | nasls_lsa | ? |
| 10:25.22 | nasls_lsa | you can create functions |
| 10:25.26 | SoftIce | hermuli :) |
| 10:25.29 | nasls_lsa | in the configuration files |
| 10:25.32 | SoftIce | what happend to your show functions ? |
| 10:25.43 | hermuli | if i only knew... |
| 10:25.56 | SoftIce | what version u using? |
| 10:26.16 | hermuli | good question... |
| 10:26.20 | hermuli | 1.4.something |
| 10:26.28 | SoftIce | maybe its changed thats why |
| 10:27.05 | hermuli | core show functions |
| 10:27.10 | hermuli | thank you SoftIce |
| 10:27.22 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 10:27.27 | SoftIce | ahh, yes, i remeber its all moved to core :) |
| 10:27.33 | SoftIce | core show version, etc |
| 10:27.57 | hermuli | no wonder the wiki stuff is outdated |
| 10:28.22 | hermuli | is there any such help and instruction thing that might be even closely up to date? |
| 10:29.18 | SoftIce | im waiting for the new book |
| 10:29.27 | hermuli | ok |
| 10:29.44 | SoftIce | when ever that happens:) |
| 10:29.58 | SoftIce | specially waiting for it to get to south africa :) |
| 10:29.58 | hermuli | hmmm |
| 10:30.11 | hermuli | i'd put my money on never... |
| 10:30.24 | hermuli | or when it does it's already badly outdated when it comes out of print... |
| 10:33.59 | SoftIce | ye :( |
| 10:34.11 | SoftIce | its pretty blind, as I would love to have a book with the updated commands |
| 10:34.21 | SoftIce | ive upgraded to 1.2.17 and it tells me all my syntaxes are incorrect |
| 10:34.47 | SoftIce | and are deprecated |
| 10:35.52 | hermuli | i see |
| 10:37.33 | hermuli | hmm... i wonder how i actually do change the language on fly... |
| 10:39.16 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 10:41.07 | *** join/#asterisk lorinc (n=ang@pool-5513.adsl.interware.hu) |
| 10:41.09 | nasls_lsa | hello , I put a BeroNet BN4S0 ( ISDN - 4 ports ) on my linux .. and now what ? I install isdn4linux.de ? is there anything else better ? |
| 10:43.49 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
| 10:46.20 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 10:47.16 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 10:49.53 | nasls_lsa | I download thaat ? ftp://ftp.isdn4linux.de/pub/isdn4linux/kernel/v2.6 ? and install it ? |
| 10:51.14 | *** join/#asterisk donkilla (n=rogers@196.200.26.174) |
| 10:55.49 | *** part/#asterisk donkilla (n=rogers@196.200.26.174) |
| 11:02.33 | nasls_lsa | great .. I lost network after I put the card on the pc ... |
| 11:03.01 | SoftIce | mwahhaha |
| 11:03.06 | nasls_lsa | :) |
| 11:03.39 | nasls_lsa | probably suse found it as network device - I found the card somewere in the configs of network |
| 11:04.16 | SoftIce | suse eeeeek :) |
| 11:04.21 | SoftIce | maybe thats your problem allready |
| 11:06.52 | nasls_lsa | after I solve that I will continue the installation :P |
| 11:16.07 | SoftIce | bonjour :( |
| 11:18.08 | *** join/#asterisk friedrich| (n=friedric@e177244060.adsl.alicedsl.de) |
| 11:21.19 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
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| 11:32.31 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 11:32.33 | unice | hi |
| 11:34.39 | nasls_lsa | hiii |
| 11:36.11 | *** join/#asterisk Athare (n=athare@c1-221-14.tbnb.isadsl.co.za) |
| 11:36.43 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 11:36.55 | Athare | Can anyone assist with a sirrix PCI 4S0 card installation? |
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| 11:44.28 | nasls_lsa | aahahaha , from Beronet ? |
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| 12:13.18 | hermuli | why doesn't my asterisk have the CHANNEL function? |
| 12:13.36 | hermuli | what on earth could i have done to lose it? |
| 12:13.48 | hermuli | has it ever even been there? |
| 12:16.44 | hermuli | is there anyone there or is everyone out drinking beer already? |
| 12:18.11 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 12:25.03 | cpm | bit early for all that |
| 12:27.36 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:30.28 | hermuli | it's never too early |
| 12:30.48 | d00gster | folks, I have configured a sip provider and could make outbound calls but when I receive a call I see the log indicating the a call is in progress. I receiver the following message "Executing NoOp("SIP/xxxxxxx-xxxxxxx", "Received incoming SIP connection from unknown peer to 4162222222") in new stack" |
| 12:30.57 | d00gster | what could be the problem? |
| 12:31.09 | d00gster | for the unknown peer |
| 12:33.12 | hermuli | arrrr... all i had to do was load the module... why didn't anyone say so!!!!??? :P |
| 12:33.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:36.24 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 12:36.36 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
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| 12:39.26 | freeat12five | I am ready to pull my hair out! I am trying to get asterisk 1.4.2 on my Via C3 800 box (running Debian Sarge 2.6.8 kernel compiled with the Cyrix III processor). When I run .configure straight, asterisk compiles but dies when I run it (Illegal Instruction). When I add the flag --build=i586, it dies in the middle of the 'make'. Can anyone send me any pointers? I am on the verge of a meltdown! |
| 12:39.26 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 12:39.41 | anonymouz666 | zZzZ... |
| 12:40.34 | stoffell | any hands-on experience or recommendations on the linksys wip330 phone? |
| 12:40.40 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 12:42.29 | fbcit | If the voice prompts in * are gsm and a user connects with a phone that does not support gsm, shouldn't * transcode the prompts into the format the phone supports? |
| 12:43.07 | anonymouz666 | fbcit: i think so. |
| 12:43.11 | shido6 | if you have the codec set in the config for the phones protocol |
| 12:43.24 | fbcit | mmm... |
| 12:43.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:43.34 | shido6 | in /etc/asteirsk/*.conf (SIP, IAX, H323, etc) |
| 12:45.57 | fbcit | I'm using sccp and debug output shows that it is looking for g723|ulaw|alaw. My prompts are the default and are gsm. When I dial into voicemail everything looks good on * CLI, but no audio gets to the phone. |
| 12:46.21 | fbcit | sccp.conf has deny=all, allow=ulaw, allow=alaw. |
| 12:46.57 | fbcit | opps... disallow=all |
| 12:46.57 | *** join/#asterisk misk0 (n=misk0@62.48.116.68) |
| 12:47.01 | d00gster | folks, I have configured a sip provider and could make outbound calls but when I receive a call I see the log indicating the a call is in progress. I receiver the following message "Executing NoOp("SIP/xxxxxxx-xxxxxxx", "Received incoming SIP connection from unknown peer to 4162222222") in new stack" |
| 12:47.04 | d00gster | what could be the problem? |
| 12:47.12 | tzafrir_laptop | freeat12five, what's the problem? |
| 12:47.36 | tzafrir_laptop | freeat12five, in older versions you had to set PROC=i586 in the Makefile |
| 12:48.05 | stoffell | why does every manufacturer use the term SIP v2 in their specs?? |
| 12:48.27 | tzafrir_laptop | a VIA C3 is not a Cyrix III, I believe. Your processor is probably VIA C3, and not C3-2 |
| 12:48.48 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 12:48.50 | freeat12five | tzafrir_laptop: I don't think you set that anymore. but you can pass the --build=i586 |
| 12:49.24 | misk0 | any link for SIPS implementation on asterisk? |
| 12:49.28 | *** part/#asterisk tparcina (n=tparcina@cisco16.fesb.hr) |
| 12:49.29 | freeat12five | tzafrir_laptop: really? Boy do I feel like an ass. |
| 12:49.42 | freeat12five | I thought C3 stood for Cyrix III |
| 12:50.19 | freeat12five | which processor do you compile it for then? |
| 12:50.25 | freeat12five | the kernel that is.. |
| 12:50.26 | tzafrir_laptop | no. C3 is Centaur or something. A company Via bought separately |
| 12:50.36 | freeat12five | ahh... |
| 12:51.07 | anonymouz666 | why dial(local/5@blah) and in [blah] 5,1,queue()... I got an ast_queue_frame error |
| 12:51.11 | freeat12five | Which processor should I compile for, do you know? |
| 12:51.26 | stoffell | misk0, http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+channels this maybe? |
| 12:52.14 | freeat12five | tzafir_laptop: That is the devil with it. It worked fine with the Cyrix III kernel selection, so I never knew better. |
| 12:52.35 | freeat12five | I will poke through the menuconfig and see if there is a better choice. |
| 12:54.08 | freeat12five | The vendor is CentaurHauls (per Google) |
| 12:55.16 | misk0 | stoffell: not SIP.... SIPS - secure SIP, or SIP over TLS |
| 12:55.49 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:56.53 | stoffell | oops, sorry ... no idea :-) |
| 12:57.14 | misk0 | stoffell: thx ..... anyone else? |
| 12:58.27 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 13:00.59 | freeat12five | tzafir_laptop: I am going to try something. I compiled the kernel for CyrixIII/Via C3 (they both have the same option), but I never recompiled gcc which was originally compiled against a 686 instruction set. I will try to recompile gcc and let you know what happens. |
| 13:03.07 | nasls_lsa | any ideas why I lost network and how I fix it , after I put the Beronet BN4S0 cards ? |
| 13:03.33 | tzafrir_laptop | freeat12five, there should be no need for that (unless you don't use the original gcc of Debian) |
| 13:03.43 | tzafrir_laptop | dpkg -l libc6-i686 |
| 13:03.54 | tzafrir_laptop | maybe you have that one |
| 13:06.09 | freeat12five | tzafrir_laptop: not there. |
| 13:06.14 | *** join/#asterisk Fieldy (n=toon@gentoo/contributor/Fieldy) |
| 13:07.06 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 13:07.27 | freeat12five | the only one is libc6 |
| 13:07.55 | freeat12five | version 2.3.2.ds1-22sa |
| 13:09.57 | freeat12five | tzafrir_laptop: Still don' |
| 13:10.13 | freeat12five | still not sure if I need to recompile. |
| 13:11.05 | tzafrir_laptop | I have not tried 1.4 on a via. Just 1.0 and 1.2 . But I don't recall that issue |
| 13:11.32 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:11.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:11.32 | freeat12five | Let me try 1.2 |
| 13:11.55 | *** join/#asterisk apardo (n=apardo@87.217.144.231) |
| 13:11.57 | freeat12five | I will grab the source and go in and set the PROC in config |
| 13:12.52 | *** join/#asterisk zeeesh (i=GrTB0T@202.38.55.125) |
| 13:12.54 | zeeesh | hi |
| 13:13.47 | zeeesh | in this line what does mean by extension_number??? "extension_number => voicemail_password,user_name,user_email_address,user_pager_email_address,user_option(s)" |
| 13:13.56 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 13:14.22 | freeat12five | does anybody know if zaptel 1.4 will work with asterisk 1.2? |
| 13:18.20 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:19.32 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 13:20.36 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
| 13:21.30 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 13:21.34 | saint_ | hi all |
| 13:23.47 | nasls_lsa | I got a beronet BNS40 card , what to I do after that ? download ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/mISDN-CVS-2007-03-29.tar.bz2 ??? |
| 13:26.58 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 13:27.28 | vooduhal | In which context does the 'o' extension need to be defined in order for operator=yes in voicemail.conf to work? |
| 13:27.33 | badcfe | what does it mean when asterisk says "Channel |
| 13:27.36 | badcfe | allocation failed: Can't create alert pipe!"? |
| 13:27.58 | saint_ | i have a question.. i am reading the asterisk book, and they are talking about compiling zaptel... Actually, I just downloaded the asterisk archive, compiled, and installed, and it did not give me any error.. does this mean that zaptel was included in it ? Or do I still have to go ahead and download it and compile it ? |
| 13:30.14 | *** join/#asterisk Jubei (n=Stormtro@147.27.47.68) |
| 13:34.02 | Jubei | guys I'm trying to build mISDN and I get "/bin/sh: scripts/genksyms/genksyms: No such file or directory" any ideas? |
| 13:34.04 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 13:34.28 | misk0 | saint_: you are probably reading Asterisk 1.2 and you have installed asterisk 1.4 |
| 13:34.51 | misk0 | saint_: in 1.4 it's little bit different organized |
| 13:35.00 | saint_ | misk0, I have installed asterisk 1.4 , yes |
| 13:35.38 | saint_ | ok.. can i go ahead and continue the book or is there something more up to date outthere ? All I want is to configure a SIP trunk with an Alcatel PBX and see if it works |
| 13:36.21 | misk0 | yes you can, but that isn't only thing that is different |
| 13:37.24 | saint_ | ok .. |
| 13:37.31 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:39.05 | heison | anyone here using rx_fax with asterisk 1.4 over SIP? |
| 13:40.51 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 13:41.17 | vooduhal | In which context does the 'o' extension need to be defined in order for operator=yes in voicemail.conf to work? |
| 13:43.05 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:43.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:45.47 | badcfe | what does it mean when asterisk says "Channel allocation failed: Can't create alert pipe!"? |
| 13:46.27 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 13:46.52 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 13:48.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:54.17 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 13:54.53 | *** join/#asterisk JoeMoes (n=JoeMoes@wolverine.vcc.de) |
| 13:56.30 | *** join/#asterisk Rhab (n=Rhab@p3EE0CDA5.dip.t-dialin.net) |
| 13:56.45 | Rhab | hi |
| 13:57.04 | Rhab | how to register a iaxmodem with asterisk? |
| 13:57.16 | Rhab | send me a link pls or smth |
| 13:57.37 | tzafrir_laptop | Rhab, it is an IAX client |
| 13:57.46 | Rhab | ok |
| 13:58.03 | Rhab | but when i start asterisk |
| 13:58.09 | Rhab | with iaxmodem running |
| 13:58.10 | tzafrir_laptop | it has a username and a password (secret) like any iax client. |
| 13:58.18 | Rhab | it says "resitration failed" |
| 13:58.28 | Rhab | ok i see |
| 13:58.57 | Rhab | which config file i have to put this username+pw? |
| 13:59.03 | Rhab | iax.conf? |
| 14:00.54 | Rhab | i defined a owner in the iaxmodem config |
| 14:01.43 | Rhab | have i to put this in iax.conf from asterisk? |
| 14:03.41 | Rhab | ok seems like i found it |
| 14:04.19 | *** join/#asterisk b00mer (n=b00mer@208.74.12.42) |
| 14:04.43 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 14:09.32 | *** join/#asterisk shinux__ (n=shinux@193.220.50.75) |
| 14:10.54 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:10.54 | *** join/#asterisk Fieldy (n=toon@gentoo/contributor/Fieldy) |
| 14:10.54 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 14:13.11 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 14:13.11 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:14.41 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:15.27 | *** join/#asterisk Splat (n=splat@eth112.tas.adsl.internode.on.net) |
| 14:20.23 | Rhab | some1 here who has a running iaxmodem |
| 14:20.25 | Rhab | ? |
| 14:21.15 | jm|work | anyone using experimental and madwifi and found problems lately? |
| 14:21.46 | jm|work | wrong channel |
| 14:21.48 | jm|work | sozzy. |
| 14:23.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:25.56 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 14:27.17 | *** join/#asterisk ppyy (i=ppyy@218.93.157.163) |
| 14:27.38 | *** join/#asterisk adam_vollrath (n=adam@pc152.BISonline.com) |
| 14:27.44 | *** join/#asterisk hedge77 (n=ambray@209.42.192.214) |
| 14:27.44 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 14:30.52 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 14:32.54 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 14:33.01 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 14:35.04 | *** join/#asterisk _DAW (n=chatzill@adsl-156-109-78.msy.bellsouth.net) |
| 14:36.09 | *** join/#asterisk irule (n=chatzill@189.164.43.19) |
| 14:36.41 | irule | hi guys, where in URL can I learn the very basics of dialplans so I can really understand it? |
| 14:36.55 | adam_vollrath | irule, the O'Reilly book. |
| 14:37.08 | irule | done, what else? |
| 14:37.36 | Zeeek | blitzrage irule in the Asterisk.org bookget the book |
| 14:37.46 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
| 14:37.54 | Zeeek | mixup of several buffers! |
| 14:38.19 | Zeeek | I flooded myself! |
| 14:38.44 | gambolputty | I wet myself! |
| 14:38.46 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:38.46 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:39.04 | Zeeek | flooding is worse, it's all over my feet |
| 14:39.35 | Zeeek | How will I go to the conference in two hours (http://x2z.eu) with wet feet? |
| 14:41.13 | anonymouz666 | what you would use to read() dtmf from a IVR and based on its result build another context with these values ? |
| 14:41.13 | anonymouz666 | dynamically |
| 14:41.21 | anonymouz666 | realtime extensions? |
| 14:42.50 | Rhab | Can i stop iaxmodem telling me every time when it registered successfully |
| 14:43.03 | Rhab | it just floods my screen every minute |
| 14:43.39 | Mahmoud | edit your logging i think |
| 14:44.10 | Mahmoud | logger.conf file |
| 14:44.15 | *** join/#asterisk antoniobrandao (n=yytrttry@201-93-242-248.dsl.telesp.net.br) |
| 14:44.17 | Mahmoud | you can set what outputs should appear to the console |
| 14:44.18 | Rhab | ok |
| 14:44.34 | JunK-Y | someone know how to register and be able to send an outbound proxy? |
| 14:44.38 | Rhab | well the output appears not to the CLI only |
| 14:44.54 | Rhab | it apears everywhere |
| 14:45.04 | Mahmoud | Rhab, even when you are not connected to asterisk -r? |
| 14:45.14 | antoniobrandao | hellou, how can i make libpri to set a variable in asterisk channel? I made some debugs and notice that libpri detects a type of call. I want to pass it to asterisk. |
| 14:45.29 | Rhab | Mahmoud yes even then |
| 14:45.53 | Rhab | and when asterisk is not running it tells me regitration timed out |
| 14:46.21 | Mahmoud | what app does send that notification? |
| 14:46.42 | Rhab | the problem lies at the iaxmodem itself |
| 14:46.49 | Mahmoud | ah |
| 14:47.24 | Rhab | i should add something to my iaxmodem config i huess |
| 14:47.37 | Mahmoud | i didn't use iaxmodem |
| 14:47.50 | JunK-Y | antoniobrandao: yes, but you have to modify libpri. |
| 14:47.53 | Mahmoud | i think so, since iaxmodem is the one causing these messages, not asterisk |
| 14:48.01 | Rhab | y |
| 14:48.08 | JunK-Y | whats ur output exactly? |
| 14:48.51 | antoniobrandao | JunK-Y, do u think i have to hack just libpri, just zaptel or both? |
| 14:49.26 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:49.30 | Rhab | JunK-Y Its a time stamp + "Registration completed successfully" |
| 14:49.35 | JunK-Y | probably just chan_zap.c yeah. |
| 14:49.59 | *** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net) |
| 14:50.00 | JunK-Y | Rhab: iaxmodem generate this. |
| 14:50.03 | Rhab | appearing every minute since i set refresh= 60 |
| 14:50.11 | Rhab | ye i know |
| 14:50.29 | JunK-Y | so just change iaxmodem, isnt * the problem. |
| 14:50.58 | *** join/#asterisk HexDump (n=a@31.Red-217-126-215.staticIP.rima-tde.net) |
| 14:51.00 | HexDump | hi all |
| 14:52.09 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 14:54.01 | *** join/#asterisk shinux_ (n=shinux@196.1.179.225) |
| 14:54.58 | HexDump | could anybody have a look at this little dialplan and tell why when I dial 200 (after the background message) it is not calling allex but if I dial 200 directly it is called correctly? thnaks in advance. |
| 14:55.06 | HexDump | here is the dialplan -> http://rafb.net/paste/ |
| 14:55.18 | HexDump | sorry -> http://rafb.net/p/CKhqI465.html |
| 14:56.27 | Rhab | means >/dev/null 2>&1 that the output is not shown? |
| 14:57.05 | Mahmoud | yeah |
| 14:57.43 | HexDump | ummmm, could it be the background command can only be used with the "s" extension?. ALl examples I have seen work this way. |
| 14:57.45 | saint_ | hey.. if I just want SIP trunk, I do not need the zaptel stuff, right ? |
| 14:57.54 | Mahmoud | the error is not shown, but it is still generated =] |
| 14:58.01 | HexDump | saint_: right :) |
| 14:58.09 | saint_ | thank you |
| 14:58.18 | saint_ | anyone tried SIP trunk with Alcatel OmniPCX / OXE ? |
| 14:58.27 | Rhab | Mahmoud there is no error ;) |
| 14:58.30 | HexDump | saint_: well, I think it is the only question I could answer :), I´m newbie too |
| 14:58.39 | saint_ | lol |
| 14:58.45 | Mahmoud | Rhab, it's not shown, but generated and sent to /dev/null |
| 14:59.16 | Mahmoud | Rhab, is there any option other than /dev/null? |
| 14:59.29 | Rhab | Mahmoud if thers ever a error i wont see it |
| 14:59.38 | Rhab | thats right |
| 14:59.45 | Rhab | but idc |
| 15:00.06 | Rhab | I can also logg it in some file |
| 15:00.12 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:00.13 | Rhab | mbe ill do this |
| 15:00.33 | Mahmoud | you should rotate the log file or it may get big and waste your disk space |
| 15:00.42 | Rhab | but every minute senseless log is too much flood imo |
| 15:00.42 | HexDump | no one could help here please? |
| 15:01.21 | Mahmoud | Rhab, the application is still wasting CPU and generating that error message, but since it is sent to /dev/null it's unseen |
| 15:01.37 | Rhab | ya |
| 15:01.40 | *** join/#asterisk ctooley (n=ctooley@tx-71-2-127-90.dhcp.embarqhsd.net) |
| 15:01.49 | Rhab | btw its NO error msg |
| 15:01.58 | Mahmoud | lol |
| 15:02.02 | Mahmoud | dude.. whatever |
| 15:02.16 | Rhab | it registrated SUCCESSFULLY |
| 15:03.08 | Rhab | i found: >/tmp/iaxmodem.stdout 2>/tmp/iaxmodem.stderr |
| 15:03.17 | Rhab | this will do a rotated log? |
| 15:03.38 | *** join/#asterisk Fieldy (n=toon@gentoo/contributor/Fieldy) |
| 15:04.13 | drako | hwo I can know which SIP channels are bien used, |
| 15:04.20 | JunK-Y | ask the iaxmodem author will be the perfect solution i guest. |
| 15:04.23 | drako | or extensions being used from the console |
| 15:04.27 | JunK-Y | drako: core show channels |
| 15:05.53 | *** join/#asterisk Pazzo (n=ugelt@dialin-225136.rol.raiffeisen.net) |
| 15:05.53 | kovger | HexDump, i just copy-pasted your extension into my test 1.2.17 * and it worked without problem, i dialled 100 then under the background message (or after it) dialed 200 and it worked |
| 15:06.43 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 15:06.48 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-214247.home.otenet.gr) |
| 15:10.39 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 15:10.57 | drako | JunK-Y, ty |
| 15:15.11 | *** join/#asterisk Igniz (n=lcarneva@200.68.64.9) |
| 15:15.15 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 15:16.34 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 15:16.40 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:17.11 | Igniz | Hi, I need some help with SIP headers |
| 15:18.27 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 15:18.39 | dhill | how do i match extension 455-4432 |
| 15:18.55 | dhill | or anything with - in it |
| 15:18.55 | Qwell[] | exten => 4554432,1,DoStuff() |
| 15:19.01 | Qwell[] | You don't - you ignore it |
| 15:19.12 | dhill | i have a loop going |
| 15:19.13 | dhill | <PROTECTED> |
| 15:19.13 | dhill | <PROTECTED> |
| 15:19.16 | dhill | over and over and over |
| 15:19.30 | Qwell[] | why? |
| 15:19.54 | dhill | someone put that number in their Cfwd on their Sipura |
| 15:20.11 | dhill | and its not matching anything in [outbound] |
| 15:20.12 | dhill | i guess? |
| 15:20.13 | dhill | i dunno |
| 15:20.17 | Zeeek | you'll have to remove the - |
| 15:20.21 | Qwell[] | if it wasn't matching, it would fail |
| 15:20.22 | dhill | how do i remove the - |
| 15:20.40 | Zeeek | cluebat to luser? |
| 15:20.47 | Qwell[] | definitely |
| 15:20.56 | dhill | so why is it looping |
| 15:21.00 | dhill | shouldn't it match |
| 15:21.03 | dhill | _NXXXXXX |
| 15:21.05 | dhill | ? |
| 15:21.16 | red9012 | is there a way to have a accept/refuse call option when answering calls from queue? |
| 15:21.27 | badcfe | what does it mean when asterisk says "Channel allocation failed: Can't create alert pipe!"? |
| 15:21.32 | Zeeek | dhill maybe _NXX. would |
| 15:21.59 | Zeeek | but then calling NNN-NNNN won't do |
| 15:22.36 | dhill | well, it is a customer.. i dont have the ability to change it |
| 15:22.52 | adam_vollrath | You should. |
| 15:23.01 | dhill | well, i can if i want |
| 15:23.05 | dhill | with http |
| 15:23.14 | dhill | but i want to make sure they dont type it in again |
| 15:23.17 | Igniz | I need to change the SIP Peer IP address to match the one of the SIP Server before the call takes place, how can i do that? |
| 15:24.49 | HexDump | can I define variables inside contexts? I mean, not in global context but the others that create the user |
| 15:28.18 | Zeeek | HexDump you mean like Set(XYZ=value) ? |
| 15:29.31 | dhill | ok, how do i drop a call then? |
| 15:29.43 | dhill | soft hangup |
| 15:29.43 | dhill | ok |
| 15:29.45 | dhill | :P |
| 15:32.48 | HexDump | Zeeek: ohhh, that way... I was thinking of Var=value. |
| 15:33.10 | JunK-Y | HexDump: no u cant. |
| 15:33.34 | Zeeek | a minimum effort of reading is in order |
| 15:34.22 | Zeeek | I recommend http://www.asteriskdocs.org as the first stop |
| 15:34.23 | *** join/#asterisk Corydon76-home (i=brown@pdpc/supporter/sustaining/Corydon76-home) |
| 15:34.23 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 15:34.59 | Zeeek | possibly complimented by http://www.voip-info.org/wiki-Asterisk+variables |
| 15:35.00 | HexDump | Zeeek: I have read it, but at this time I have 10000 of things on the head :) |
| 15:35.20 | Zeeek | But then setting a variable is one of the most important things to GET |
| 15:35.33 | Zeeek | even I know how to tdo that |
| 15:35.48 | Zeeek | and I know nothing... nothing |
| 15:36.45 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 15:36.56 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
| 15:40.52 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 15:40.55 | zdrulio | hello all |
| 15:41.02 | zdrulio | i have 1 question |
| 15:41.15 | bulle | zdrulio: the answear is 42 |
| 15:41.22 | zdrulio | aahah |
| 15:41.24 | zdrulio | nica |
| 15:41.27 | zdrulio | nice |
| 15:41.32 | zdrulio | but other question ;) |
| 15:41.54 | Zeeek | the answer is http://x2z/eu |
| 15:42.15 | Zeeek | better yet http://x2z.eu |
| 15:42.50 | zdrulio | hum |
| 15:43.08 | Zeeek | so ask already |
| 15:44.05 | zdrulio | my english is bad, but how can i test asterisk work on a weak PC |
| 15:44.16 | syzygyBSD | zdrulio: try it? |
| 15:44.28 | Zeeek | how weak? |
| 15:44.44 | syzygyBSD | what kind of traffic... |
| 15:44.46 | Zeeek | in the above sentence, your english is perfect by the way |
| 15:44.48 | zdrulio | i mean any program or scrip who imitate 10or 20 calls |
| 15:45.19 | syzygyBSD | Zeeek: I would have added an s to work |
| 15:45.26 | Zeeek | 10 or 20 simultaneous calls won't work well ojn very old hw |
| 15:45.36 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:45.57 | syzygyBSD | he left? |
| 15:46.00 | Zeeek | syz just trying to lighten the tension :) |
| 15:46.23 | syzygyBSD | oh.. it was much better english than most geeks |
| 15:46.32 | Zeeek | indeed |
| 15:46.42 | Zeeek | my own be cool tho |
| 15:46.55 | syzygyBSD | too bad he left before getting a real recomendation |
| 15:47.10 | aydiosmio | I like sipsak |
| 15:47.12 | Zeeek | a few calls on a pentium 3 would be my guess |
| 15:47.30 | Zeeek | not 10-20 |
| 15:47.35 | Zeeek | but that's my guess |
| 15:47.39 | syzygyBSD | come on, the centrinio is based on P III, it can't be that bad |
| 15:48.11 | Zeeek | I have a PIII with 512M at what, 400Mhz? |
| 15:48.17 | Zeeek | I can't remeber |
| 15:48.18 | syzygyBSD | I will go home and throw a ubuntu disk in my laptop (PIII 1Ghz) and benchmark it |
| 15:48.40 | Zeeek | laptop disks suck tho |
| 15:48.46 | Zeeek | so it's not fair |
| 15:48.54 | syzygyBSD | um... it should all be in memory, right? |
| 15:49.03 | Zeeek | true, mostly |
| 15:49.18 | syzygyBSD | not like i am recording all 20 calls |
| 15:49.24 | Zeeek | heh |
| 15:49.28 | Zeeek | why not? |
| 15:49.40 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:49.42 | Zeeek | you should, and to an offsite NFS server |
| 15:49.48 | Zeeek | for the benchmark |
| 15:49.53 | syzygyBSD | because I really don't want 20 copies of louie louie |
| 15:49.57 | Zeeek | haha |
| 15:50.16 | Zeeek | louielouie must be public doman by now! |
| 15:50.17 | syzygyBSD | guess what I test with... |
| 15:50.31 | Zeeek | that song was popular when I was in HS |
| 15:50.35 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 15:50.37 | zdrulio | Re |
| 15:50.38 | Zeeek | and THAT was a while ago |
| 15:50.45 | syzygyBSD | wb |
| 15:50.47 | zdrulio | i say |
| 15:51.12 | zdrulio | samo program who imitate 10-20-30 calls |
| 15:51.15 | zdrulio | or more |
| 15:51.36 | zdrulio | to view memory and CPU stats in moment of calls |
| 15:51.38 | tclark | anyone using hpec with t1 connected to channel banks ? |
| 15:52.16 | *** join/#asterisk Connor- (n=billy@198-144-174-5.knx.tn.nxs.net) |
| 15:52.19 | syzygyBSD | well, I meant Playback(lyrics-louie-louie) |
| 15:53.02 | Zeeek | there are lyrics to louie louie? |
| 15:53.14 | Connor- | Hey guys, I started recently having a issue with some of my 7960's coming unregistered from my Asterisk box.. I've not changed anything with the phones nor with asterisk.. and I'm completly at a loss.. what do I need to look at? |
| 15:54.08 | zdrulio | Zeeek, syzygyBSD: are some of you know program ? |
| 15:54.36 | syzygyBSD | I wrote my own, let me see if I can find it |
| 15:56.16 | quidpro | Connor: As one famous telecom adveritisng line says "It's in the network"/ |
| 15:57.41 | Connor- | Well, that's the strange part.. I've got 2 7960's and a ATA coming off of the same ethernet switch, (and wire) into a smaller switch (which has been changed out) and they're NOT unregistering |
| 15:58.18 | quidpro | weird |
| 15:58.50 | quidpro | try switching ports between what works fine and what doesn't and watch |
| 15:59.13 | Connor- | it's so hit and miss, it's just strange. |
| 15:59.13 | *** join/#asterisk apardo (n=apardo@87.217.145.137) |
| 16:01.31 | *** join/#asterisk FieldySnuts (n=toon@gentoo/contributor/Fieldy) |
| 16:02.39 | Qwell[] | quidpro: "in" the network? |
| 16:03.52 | *** join/#asterisk andy-andy (n=andrew@ool-18b9164b.dyn.optonline.net) |
| 16:05.29 | andy-andy | hey guys, which linux distro for asterisk you can recommend ? |
| 16:05.58 | bsd_tech | any linux its your choice |
| 16:06.07 | bsd_tech | linux is linux is linux |
| 16:06.08 | quidpro | Qwell, I think that's the slogan. But I'm Canadian so I don't get to see the commercial that often. |
| 16:06.16 | quidpro | Or maybe it's "it's the network" |
| 16:06.17 | Qwell[] | "It's the network" |
| 16:06.32 | Qwell[] | bsd_tech: hey, that's my line |
| 16:06.46 | bsd_tech | lol |
| 16:06.58 | quidpro | Yeah, the *is* isn't supposed to be... |
| 16:06.59 | bsd_tech | Qwll us it and stop just sitting there |
| 16:07.30 | Connor- | We'll what I'm finding is, the phones unregister in asterisk, but, the phone doesn't know that it's unregisted. |
| 16:07.31 | quidpro | What do you guys think of surge protection for incoming analog lines... hype or must-have? |
| 16:07.43 | Qwell[] | quidpro: always better to be safe |
| 16:07.44 | Connor- | allmost like asterisk's timers are messed up. |
| 16:07.50 | bsd_tech | must have if your in the midwest |
| 16:08.05 | bsd_tech | use ztdummmy |
| 16:08.10 | bsd_tech | its your friend |
| 16:08.35 | Connor- | bsd_tech who you talking to? |
| 16:08.43 | bsd_tech | if your in a high thunderstorm / electric storm area use them |
| 16:08.45 | quidpro | Yeah... I might just go for one that goes in at the panel... There aren't too many surge bars that can handle 4 analog lines (there is one, but heck of alot cheaper to buy a dedicaed module). |
| 16:08.46 | andy-andy | linux is linux sounds good , but it really doesnt matter ? i remember glibc version issues with other software... |
| 16:09.07 | bsd_tech | I use bsd |
| 16:09.14 | bsd_tech | I am happy with it |
| 16:09.34 | aydiosmio | quidpro: I worked for a company in florida with a just a few analog POTS and T1s, we lost a few peices of equipment each year |
| 16:09.43 | aydiosmio | ATAs, Fax machines, modems |
| 16:09.52 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:10.18 | andy-andy | i never use bsd , and since i start playng w asterisk , i prefer to play on my playground, but thnks 4 advice |
| 16:10.37 | aydiosmio | we even lost ethernet hubs + hubs downstream to surges in electric service -- but that's anothe rmatter |
| 16:11.04 | Connor- | andy-andy: CentOS is a good choice |
| 16:11.31 | quidpro | haha... anybody ever notice the Verizon "network guy" is reminiscent of the phone guy from the 80's TV show "Alice"? |
| 16:11.48 | andy-andy | what you think about slackware ? |
| 16:12.00 | bsd_tech | slack is good |
| 16:12.08 | Connor- | andy-andy: I think it's too much work. |
| 16:12.20 | aydiosmio | Debian is better than Slack for * imho |
| 16:12.33 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 16:12.38 | tonyb2006 | about reminicent advertising characters in modern time: http://phib3r.com:443/Original-Mac-vs-PC.jpg |
| 16:12.38 | JunK-Y | i love debian/ubuntu too. |
| 16:12.47 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 16:12.50 | andy-andy | which debian based distro you prefer ? |
| 16:13.19 | andy-andy | will ubuntu fit server requirements ? my wife use it as desktop |
| 16:14.13 | Zeeek | http://www.pr.com/press-release/34406 |
| 16:14.26 | JunK-Y | ubuntu-server. |
| 16:14.38 | quidpro | tony: nICE |
| 16:14.42 | quidpro | eer, damn caps |
| 16:14.44 | ezway` | ;) |
| 16:15.10 | reber | choose the original, choose debian |
| 16:15.20 | Zeeek | or BSD |
| 16:15.27 | reber | or BSD |
| 16:15.59 | Zeeek | or BSD |
| 16:16.09 | reber | or BSD |
| 16:16.12 | Zeeek | or... |
| 16:16.32 | andy-andy | Zeeek/reber , loop ? |
| 16:16.37 | reber | or women |
| 16:16.48 | reber | (sorry it was another movie ;) ) |
| 16:16.59 | Zeeek | women is good. I'm ok with that |
| 16:17.04 | bsd_tech | BSD |
| 16:17.14 | bsd_tech | I want my BSD |
| 16:17.28 | Zeeek | http://x2z.eu is on FreeBSD |
| 16:17.28 | Igniz | is there any way to modify the contents of a sip header? in the dialplan i mean |
| 16:17.39 | bsd_tech | if your running staright asterisk it works great |
| 16:17.45 | bsd_tech | and we have the gui working also |
| 16:17.52 | bsd_tech | I have the patch for it |
| 16:17.52 | Zeeek | I tries asterisk a couple of times on FreeBSD |
| 16:18.09 | bsd_tech | its in the ports |
| 16:18.21 | Zeeek | that version didn't work for me |
| 16:18.22 | bsd_tech | dont build from source |
| 16:18.35 | Zeeek | but I tried to make it from scratch and it worked |
| 16:18.35 | bsd_tech | we have 1.4.2 working |
| 16:18.44 | bsd_tech | and zaptel 1.4 |
| 16:18.58 | heison | does anyone know why I'm getting the drop frame? |
| 16:18.58 | bsd_tech | we have a all new 1.4 team |
| 16:19.06 | heison | <PROTECTED> |
| 16:19.06 | heison | [Mar 30 12:17:50] NOTICE[3711]: channel.c:2287 __ast_read: Dropping incompatible voice frame on SIP/sipura-fxo-081f7e20 of format slin since our native format has changed to alaw |
| 16:20.04 | Zeeek | I haven't yet seen if receiving a fax will kill asterisk |
| 16:20.09 | Zeeek | since the update |
| 16:20.52 | heison | Zeeek: if fax is received on Zap (FXO), it will crash asterisk, which is why i'm now doing this over linksys 3102 - which doesn't seem to work either... |
| 16:21.42 | Zeeek | with spandsp some faxes would work and some wouldn't |
| 16:22.06 | Zeeek | however, since updating I haven't tried. We tell people to join the 2&st century and use email |
| 16:22.51 | heison | Zeeek: i have had very good luck with 1.2, spandsp, 100% of fax is coming thru, same hardware doesn't work very well with 1.4.2 |
| 16:24.04 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 16:24.51 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 16:27.10 | HexDump | bye all peopple |
| 16:28.48 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 16:29.44 | badcfe | what does it mean when asterisk says "Channel allocation failed: Can't create alert pipe!"? |
| 16:29.53 | bsd_tech | and zaptel and libpri and the gui |
| 16:30.09 | bsd_tech | even thou the gui needs loads of work |
| 16:30.15 | bsd_tech | its lacking so much |
| 16:34.35 | *** join/#asterisk Vec (n=Vec@dsl-244-221-22.telkomadsl.co.za) |
| 16:35.23 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 16:36.07 | Vec | When I do a module show like codec I can't find speex, is speex a module or is it built into asterisk directly ? |
| 16:37.46 | bsd_tech | its a sep ppkg |
| 16:37.50 | andy-andy | why nobody remind about http://www.astlinux.org/ ? |
| 16:37.53 | bsd_tech | you have to build the lib |
| 16:38.21 | Vec | bsd_tech : r u refferring to speex ? |
| 16:38.32 | bsd_tech | yes |
| 16:39.07 | bsd_tech | you have to grab the speex source build it then rebuild asterisk so that it finds it and includes it |
| 16:39.33 | Zeeek | http://x2z.eu - the conference is on ! |
| 16:39.52 | bsd_tech | why do people not read before doing |
| 16:40.06 | Vec | oh, ok got it |
| 16:40.07 | Vec | thanks |
| 16:41.31 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 16:41.45 | bsd_tech | need a pin |
| 16:41.56 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 16:41.59 | genz | What's the maximum power that I can send to a phone? |
| 16:42.27 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75b:0:240:f4ff:fe6b:7650) |
| 16:42.29 | Zeeek | 2007 2007 22 |
| 16:42.36 | genz | Management is complaining that people aren't paying attention to the "Please pay attention to our options, as they have recently changed." |
| 16:42.39 | Zeeek | 2007 2007 00 |
| 16:43.00 | genz | So I was thinking the best option would be to send a surge of electricity to the user and just give the a quick friendly zap. |
| 16:43.06 | *** join/#asterisk AbsTradELic (n=vldmr@201.79.156.219) |
| 16:43.15 | Strom_M | genz, sorry, it isnt going to work |
| 16:43.22 | bsd_tech | ? |
| 16:43.25 | Zeeek | PINs also 666 5551212 123123123 3 |
| 16:43.27 | Strom_M | you can't code for electric zaps in SS7 |
| 16:43.28 | genz | I was going to call it the fixUser feature, and you'd send 2 variables. 1) amount of charge. 2) timeout |
| 16:43.31 | AbsTradELic | anyone here from digium ? |
| 16:43.42 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 16:43.43 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:43.48 | andy-andy | may be somebody on conference , say couple words about recommended linux distro for hosted asterisk ? |
| 16:43.49 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:43.52 | genz | Strom_M, but is it really that horrible of an idea? |
| 16:44.06 | Strom_M | genz, yeah, because you'll fry the equipment before you fry the user |
| 16:44.28 | genz | Strom_M, i'll just have to figure out the tone they used it Lost to make the guy keel over... |
| 16:44.43 | Strom_M | AbsTradELic, the digium folk are here from time to time; what's your question? |
| 16:45.11 | hedge77 | are you allowed to pattern match like _[IO]13X or would I have to break it down for _[IO]130.._[IO]139? |
| 16:45.19 | bsd_tech | pin is not working |
| 16:45.28 | bsd_tech | I cant get in |
| 16:45.31 | Strom_M | hedge77, you can pattern match like that |
| 16:45.32 | irule | I just set a nice topic in ltsp-es :) |
| 16:45.42 | Strom_M | _[10]13X is valid |
| 16:45.44 | Zeeek | you need to enter show id fiorst: 22622# then one of the PINS end with # |
| 16:45.48 | irule | oops sorry wrong # |
| 16:46.15 | bsd_tech | ? |
| 16:46.16 | *** join/#asterisk tigmmi (n=tigmmi@adsl-245-184-192-81.adsl2.iam.net.ma) |
| 16:46.20 | tigmmi | hello |
| 16:46.35 | bsd_tech | wich pin |
| 16:46.37 | hedge77 | the ael2 parser isn't in 1.4.2 is it? |
| 16:46.44 | bsd_tech | irule wasthat you |
| 16:46.45 | Qwell[] | hedge77: sure it is |
| 16:47.17 | tigmmi | is there a way to control the MOH volume? it's saturated |
| 16:47.20 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:47.29 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:47.31 | generalhan | hey all ! |
| 16:47.48 | tigmmi | mpg123 -g dose nothing |
| 16:47.57 | bsd_tech | none of the pins work |
| 16:48.28 | bsd_tech | none of the oins seem to work I keep getting invalid |
| 16:48.37 | bsd_tech | I am streaming the stream |
| 16:48.41 | bsd_tech | but I want in |
| 16:49.44 | Cybertoy | why don't you register with the web-site and get your pin? |
| 16:49.47 | Cybertoy | it only takes 2 minutes |
| 16:49.58 | Zeeek | I gave good PINs |
| 16:50.01 | hedge77 | Qwell[]: do I have to do something to use it or is it on all the time for .ael files now? |
| 16:50.10 | Qwell[] | hedge77: it's always on |
| 16:50.36 | generalhan | can some one please help me out with a soxmix issue? i have this perl script that i run each night to mix the -in/-out recorded calls, its worked for the longest time, and now for some reason its not getting rid of the -in/-out files after it creates the mixed file. can someone look at the script and see if you notice anything really wrong please : http://generalhan.pastebin.ca/416961 |
| 16:50.39 | hedge77 | well then i'd hate to have seen what few errors you would get with the first one |
| 16:51.32 | tigmmi | is there a way to control the MOH volume? it's saturated |
| 16:51.47 | Qwell[] | tigmmi: Don't use mpg123 |
| 16:51.52 | bkruse | no! |
| 16:52.00 | bkruse | use mpg123 then do top, you will be surprised. |
| 16:52.35 | *** join/#asterisk Fieldy (n=toon@gentoo/contributor/Fieldy) |
| 16:52.38 | tigmmi | what to use so |
| 16:52.46 | wunderkin | and use sox to reduce the volume |
| 16:52.50 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 16:52.56 | Zeeek | come on into the conference guys |
| 16:53.11 | Zeeek | bsd_tech so you made it :) |
| 16:53.12 | aydiosmio | dial in and get crunk |
| 16:53.28 | Strom_M | there's a conference? :) |
| 16:53.34 | tigmmi | bbl |
| 16:53.39 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
| 16:53.54 | Qwell[] | I'd wager that there is always a conference call going on somewhere in the world |
| 16:54.11 | Strom_M | yeah, i missed the url |
| 16:54.17 | Qwell[] | hey Strom_M, we should petition to get rid of timezones |
| 16:54.30 | Strom_M | erhmmmm....no. |
| 16:54.32 | Qwell[] | just had a thought.. how would time zones work when we colonize Mars? |
| 16:54.43 | bkruse | Qwell[]: automagically |
| 16:54.43 | Strom_M | mars will be on its own time |
| 16:54.51 | bkruse | or some moron will split it up, one of the two |
| 16:54.54 | Zeeek | http://x2z.eu |
| 16:54.58 | hedge77 | i take back what i said the parser was exactly right as to what was wrong I was just blind to it. derp derp derp |
| 16:54.58 | Qwell[] | it would be a real PITA to schedule meetings with folks on Mars |
| 16:55.37 | Strom_M | Qwell, yeah, not to mention that radio waves take 20 minutes to travel each direction |
| 16:55.40 | Zeeek | go to talkshoe.com and get a pin |
| 16:55.52 | Qwell[] | Strom_M: Don't bring logic into this |
| 16:56.00 | Strom_M | Zeeek, yeah, im not gonna do it if its recorded :) |
| 16:56.10 | Zeeek | why ? |
| 16:56.18 | Qwell[] | FBI is looking for him. |
| 16:56.22 | Qwell[] | I mean...crap, sorry |
| 16:56.25 | Strom_M | because im not in the mood at the moment |
| 16:56.33 | Zeeek | ok |
| 16:56.33 | Strom_M | Qwell[], great, you blew my cover |
| 16:56.47 | Zeeek | listen to the stream |
| 16:56.52 | Strom_M | Zeeek, you should also specify AM / PM on the site |
| 16:56.54 | *** join/#asterisk jdchaiken (n=jdchaike@dsl254-079-108.nyc1.dsl.speakeasy.net) |
| 16:57.00 | Zeeek | or a recorded mp3 |
| 16:57.08 | Zeeek | PM EDT |
| 16:57.15 | generalhan | anyone have any ideas about my stupid soxmix issue ? |
| 16:57.18 | bsd_tech | I am in |
| 16:57.19 | Strom_M | Zeeek, ON THE SITE |
| 16:57.20 | Strom_M | not here |
| 16:57.21 | Strom_M | :) |
| 16:57.30 | Zeeek | what site ? |
| 16:57.42 | Strom_M | the one you linked to? |
| 16:57.43 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 16:57.47 | blitzrage | cogo dot what? |
| 16:57.53 | Zeeek | ok I'll look |
| 16:57.55 | codazoda | Anyone here use LumenVox voice recognition? |
| 16:58.37 | codazoda | I've got it working, returning the words that I spoke, but I would like it to return a keyword instead of the phrase I spoke. Trying to figure out how to setup the grammar to do that. |
| 16:58.46 | Zeeek | Sorry I though I did say PM. Done. |
| 16:59.06 | jdchaiken | I have a dumb question. I'm playing around w/ * trying to learn what I'm doing. I have a channel bank, but no T1 interface card on my * box. I have a Cisco router w/ a spare WIC slot. Cisco T1 cards are much cheaper than the Digium T1 interfaces, Can I somehow make this work? |
| 16:59.19 | Strom_M | jdchaiken, probably not. |
| 16:59.46 | Strom_M | unless the cisco router speaks SIP or something |
| 16:59.55 | aydiosmio | not likekly, only a few cisco devices do |
| 16:59.58 | Strom_M | in which case.....kluuuuudge-o-riffic |
| 17:00.29 | aydiosmio | our AS5300 is okay though. it works. and speaks g723, unlike some solutions which will remain namesless |
| 17:00.49 | bkruse | aydiosmio: hehe ;] |
| 17:01.38 | jdchaiken | Ok. I guess I'm buying a Digium or Rhino card then. |
| 17:01.41 | jdchaiken | Thanks |
| 17:02.08 | bsd_tech | what is the webgui like cobalt for unix |
| 17:02.56 | jdchaiken | I already have a spare Channel Bank though. I'm willing to spend $500 to play, maybe not $2000. |
| 17:03.39 | hedge77 | who was the guy plugging idefisk yesterday? |
| 17:03.46 | Qwell[] | hedge77: probably zoa |
| 17:03.59 | blitzrage | Zeeek: I'm gonna drop off here since I gotta go work on the book again :) |
| 17:04.00 | hedge77 | yeah that sounds right |
| 17:04.25 | blitzrage | Zeeek: ping me next week and I'll try and jump in there again to answer any questions |
| 17:04.58 | codazoda | No LumenVox users, eh? |
| 17:05.25 | hedge77 | is there a particular thing that typically triggers "405 method not allowed" sip errors |
| 17:06.24 | coppice | aydiosmio: you seem to have a deep fascination with this crappy codec |
| 17:07.01 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:08.26 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:08.59 | aydiosmio | I seem to work for a company that services dial-up customers |
| 17:09.21 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:09.51 | coppice | and how does G.723.1 help with that? |
| 17:10.42 | e-milio | hello guys |
| 17:11.31 | bkruse | coppice: it sucks! |
| 17:11.53 | Ng | hmm, music on hold is stopping right after it starts, and not playing any music |
| 17:11.59 | Ng | any ideas? |
| 17:12.10 | bkruse | crashing? |
| 17:12.13 | bkruse | no verbose output? |
| 17:12.18 | bkruse | are you using mpg123? |
| 17:12.32 | Ng | [default] |
| 17:12.32 | coppice | ah, I see. use a nasty codec to drive away the users, and you have less bandwidth problems. I guess that will work |
| 17:12.32 | Ng | mode=mp3 |
| 17:12.32 | Ng | directory=/usr/share/asterisk/mohmp3 |
| 17:12.35 | Ng | is the config |
| 17:12.53 | Ng | but I noticed that "moh show files" shows me nothing. there are files in that dir and the asterisk user can read them |
| 17:12.54 | tigmmi | MOH music very bad!! after reducing mp3 volume using sox, the sound get even worst!!!!!! |
| 17:13.01 | *** join/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 17:13.22 | Strom_M | tigmmi, well yeah, because you're re-encoding it |
| 17:13.52 | tigmmi | so what to do, the ekiga to ekiga sound is very clean |
| 17:13.54 | heison | does anyone know why this is happening? |
| 17:13.55 | heison | [Mar 30 12:17:50] NOTICE[3711]: channel.c:2287 __ast_read: Dropping incompatible voice frame on SIP/sipura-fxo-081f7e20 of format slin since our native format has changed to alaw |
| 17:14.05 | bsd_tech | http://www.sineapps.com/news.php?rssid=1503 |
| 17:14.05 | tigmmi | only the MOH is umpleasent to hear |
| 17:14.19 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 17:15.29 | *** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net) |
| 17:15.33 | Ng | oh, I was being a spanner |
| 17:15.42 | Ng | no mp3 support installed and I wasn't even using mp3 files |
| 17:15.44 | Ng | sorry for the noise ;) |
| 17:16.18 | bkruse | heison: what are you transcoding to? |
| 17:16.23 | bkruse | your endpoint has changed codecs |
| 17:16.28 | bkruse | is this in 1.4 and are you using a transcoder? |
| 17:16.58 | Ng | fwiw, I've seen those dropping incompatible voice frame errors when calling someone calls me and my phone is forwarded to my mobile phone :/ |
| 17:17.00 | heison | bkruse: this is a fax call.. rxfax() |
| 17:17.14 | bkruse | hmm |
| 17:17.19 | bkruse | not sure |
| 17:17.23 | heison | bkruse: k |
| 17:18.07 | heison | with zaptel 1.4, i'm also experiencing problems with FXO/FXS channels, ztcfg would sometimes report error with ZT_CHANCONFIG |
| 17:18.16 | heison | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 17:18.20 | bkruse | lol |
| 17:18.25 | bkruse | is that not obvious? :\ |
| 17:19.33 | heison | bkruse: it's quite obvious, but rebooting the machine the problem goes away and then comes back |
| 17:19.51 | bkruse | ohrly? |
| 17:19.51 | bkruse | hmm |
| 17:20.00 | *** join/#asterisk ppyy (i=ppyy@218.93.157.163) |
| 17:20.30 | *** join/#asterisk ikear (n=angelsz@202.59.70.162) |
| 17:20.32 | heison | bkruse: the funniest part is... i have not physically move the PCI slots of the cards, but they have been sometimes probed in the wrong order |
| 17:20.41 | e-milio | Anybody knows if there is in the digium plans to release an appliance with T1/E1 ports ? |
| 17:20.43 | heison | ie. FXO becomes FXS and vice versa |
| 17:20.43 | ikear | anyone familiar with mjua? |
| 17:20.52 | generalhan | can some one please take a look at my soxmix command to make sure that im at least running the correct syntax to get this working:: http://generalhan.pastebin.ca/417002 |
| 17:20.56 | *** join/#asterisk punkgode (n=Punkgode@rev-200-40-119-222.netgate.com.uy) |
| 17:21.02 | bkruse | heison: super wierd. |
| 17:21.53 | heison | bkruse: is there a way to force the loading order of the wcfxo and wctdm modules? |
| 17:21.57 | aydiosmio | e-milio: they had a development device, but I don't think it's in digium's best interest to offer appliances now |
| 17:22.10 | bkruse | well, you should be able to load on a per card basis, they should load up the same everytime |
| 17:22.14 | Qwell[] | aydiosmio: why? |
| 17:22.14 | bkruse | its a system problem if they do not |
| 17:22.54 | e-milio | aydiosmio:why? |
| 17:23.24 | Qwell[] | e-milio: and if that's something you'd be interested in, I would recommend contacting sales and expressing an interest. |
| 17:23.26 | heison | bkruse: wondering if this is a kernel problem... it's the same hardware for the last 2 years |
| 17:23.35 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 17:24.07 | e-milio | Qwell[]: Do youi think there would be interest on makin one with T1port ? |
| 17:24.18 | aydiosmio | just my opinion, developing and supporting something like that would take away from their core business |
| 17:24.44 | Qwell[] | e-milio: I'd bet people would like to see that, sure. |
| 17:24.51 | bkruse | lol |
| 17:24.52 | Qwell[] | aydiosmio: which is? |
| 17:24.54 | bkruse | Qwell[]: run! |
| 17:25.04 | Qwell[] | bkruse: fix my grep! |
| 17:25.06 | aydiosmio | developing asterisk |
| 17:25.12 | GreyFoxx | Can the Asterix Manager (or maybe a proxy app of some kind) be configured so that a user logged in can only see events and such related either directly to them or to some sort of defined "group" they are part of ? |
| 17:25.16 | Qwell[] | I can totally segfault busybox grep with a ^, heh |
| 17:25.17 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 17:25.34 | bkruse | Qwell[]: whats wrong, cannot grep -r? :[ |
| 17:25.35 | e-milio | aydiosmio: I think you are a bit confused |
| 17:25.41 | Qwell[] | bkruse: grep -v "^blah" |
| 17:25.44 | coppice | aydiosmio: clue: they get money for their hardware, and give away their software |
| 17:25.45 | Qwell[] | uber-hosed |
| 17:25.53 | bkruse | so your removing all lines that begin with blah? |
| 17:25.58 | Qwell[] | yeah |
| 17:26.00 | bkruse | is this on the aadk? |
| 17:26.07 | Qwell[] | yep |
| 17:26.11 | bkruse | and it didnt work!? |
| 17:26.12 | Qwell[] | on my new code |
| 17:26.21 | bkruse | i thought i used that a couple days ago, for removing comments in conf files |
| 17:26.21 | ikear | i'm using mjua to make an outgoing call but asterisk refuses with a "request pending" message - what could be wrong? |
| 17:26.27 | aydiosmio | coppice: yup |
| 17:26.43 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 17:26.43 | aydiosmio | don't forget support |
| 17:26.52 | aydiosmio | and selling copies of business asterisk |
| 17:27.05 | bkruse | duh! |
| 17:27.07 | heison | bkruse: i'm looking at /var/log/messages... and sometimes i would see the wcfxo driver being loaded first, sometimes i would see the wctdm messages first |
| 17:27.17 | heison | bkruse: that is definitely my problem |
| 17:27.43 | coppice | aydiosmio: don't you think selling a whole system box, instead of just an interface, might bring more revenue? :-) |
| 17:28.35 | bkruse | heison: hmm |
| 17:28.42 | bkruse | coppice: yes. |
| 17:29.08 | bsd_tech | lost the call |
| 17:29.13 | bkruse | :[ |
| 17:29.26 | bsd_tech | it just rings now |
| 17:29.35 | bsd_tech | like thier server died |
| 17:30.05 | tigmmi | MOH sound quality is caused by the mpg123 -r 8000, use custom app by removing that stupid default thing make things softer !!!!!!!!!!!! |
| 17:30.07 | tigmmi | lol |
| 17:30.12 | tigmmi | it's ok |
| 17:30.17 | bsd_tech | thats the number again |
| 17:30.25 | *** join/#asterisk svenna_ (n=svenna@p548D318C.dip0.t-ipconnect.de) |
| 17:30.41 | aydiosmio | coppice: maybe, but diversification at such a young stage causes the overall quality to suffer from what I've seen |
| 17:30.47 | bkruse | it doesnt make it softer, it puts it into the right hertz value |
| 17:30.55 | aydiosmio | if Digium were, say, GE, it would be different |
| 17:31.15 | Qwell[] | aydiosmio: well, we don't have any plans to make fridges or give home loans |
| 17:31.30 | tigmmi | bkruse, Qwell it's -r 8000 of the mpg123 that caused the problem |
| 17:31.37 | tigmmi | thanks anyway |
| 17:31.47 | aydiosmio | but you know, just my opinion |
| 17:31.54 | coppice | aydiosmio: I suppose you do realise they are now 7 years into doing this? they aren't exactly beginning |
| 17:32.09 | bsd_tech | I get dead air on the conf |
| 17:32.17 | bkruse | lol |
| 17:32.37 | bsd_tech | that sucks |
| 17:32.44 | bsd_tech | it just hung up on me calling in |
| 17:32.50 | aydiosmio | yet they |
| 17:33.03 | aydiosmio | are still on very logn development schedules for their hardware |
| 17:34.01 | e-milio | aydiosmio: why do you think that ? |
| 17:34.51 | bsd_tech | O well I am out it is not answering now |
| 17:35.19 | bsd_tech | looking for the chat client |
| 17:36.14 | GreyFoxx | lnx: You can configure each of the views independantly to browse the filesystem and not just the database contents |
| 17:36.17 | GreyFoxx | oops |
| 17:36.58 | bsd_tech | the text client is not supported in bsd |
| 17:37.21 | bsd_tech | are we ending the call |
| 17:37.28 | bsd_tech | ok |
| 17:38.05 | ikear | could anyone help me out with mjsip here? |
| 17:38.21 | tzafrir_laptop | ikear, what is mjsip? |
| 17:38.30 | ikear | a java-based sip stack |
| 17:38.48 | ikear | i'm using their user agent and when i try to make a call using it, asterisk returns "request pending" |
| 17:39.05 | ikear | but it can receive calls fine |
| 17:39.12 | tzafrir_laptop | 'sip debug' in asterisk can help you |
| 17:39.20 | tzafrir_laptop | If not, pastebin the trace |
| 17:39.26 | Zeeek | bsd_tech thanks for joining us - it was fun! |
| 17:39.37 | bsd_tech | yeah |
| 17:39.46 | ikear | pastebin? |
| 17:39.52 | tzafrir_laptop | ~pb |
| 17:39.54 | 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 |
| 17:39.54 | Zeeek | I'll post some notes - see you later |
| 17:39.58 | bsd_tech | I will make it next friday |
| 17:40.55 | ikear | ok i'll give sip debug a shot |
| 17:41.00 | ikear | thanks |
| 17:44.33 | tzafrir_laptop | ikear, if you have other sip phones: sip debug peer <name> or: sip debug ip <address> |
| 17:44.52 | tzafrir_laptop | sip debug is v e r y verbose |
| 17:44.58 | *** join/#asterisk snowy_owl (n=snowy_ow@200.218.192.10) |
| 17:44.58 | hedge77 | what would * be doing that makes idefisk return sip method not allowed? Debug output -> http://pastebin.ca/417022 |
| 17:45.13 | tzafrir_laptop | isn't idefisk iax? |
| 17:45.19 | hedge77 | does sip now too |
| 17:45.26 | tzafrir_laptop | use iax |
| 17:45.55 | hedge77 | and it looks like it works pretty well so if I can get it to stop spamming me with errors.... |
| 17:46.40 | hedge77 | i'm using it for testing what will eventually be an extension for a sip phone, using iax would defeat the purpose |
| 17:46.51 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 17:46.56 | tzafrir_laptop | hedge77, somebody doesn't like Content-Type: application/simple-message-summary ? |
| 17:47.16 | snowy_owl | a little question: 1.4.0 DOESNT send a reinvite to update the session. Why 1.4.2 do that? Look the sample sip.conf (canreinvite): "In Asterisk 1.4 this setting also affect direct RTP at call setup (a new feature in 1.4 - setting up the call directly between the endpoints instead of sending a re-INVITE)" |
| 17:47.41 | e-milio | is idefisk os ? |
| 17:48.11 | tzafrir_laptop | e-milio, no, emacs is an OS. |
| 17:48.32 | tzafrir_laptop | and no. it is not |
| 17:48.42 | tigmmi | bye |
| 17:50.02 | e-milio | tzafrir_laptop: os=opensource rght ? |
| 17:50.20 | tzafrir_laptop | snowy_owl, I'm afraid I missed you. But is your question based solely on the documentation of Asterisk or also on your experince with it? |
| 17:50.31 | *** join/#asterisk stmille (n=Steve@204-88-250-240.co.slo.ca.us) |
| 17:50.41 | tzafrir_laptop | e-milio, in the second reply: yes |
| 17:51.49 | aydiosmio | oss = open source software, OS = operating system |
| 17:52.04 | aydiosmio | ambiguity-- |
| 17:52.34 | bkruse | lol, agreed |
| 17:53.54 | *** join/#asterisk jshardy (n=jhardy@74.211.24.166) |
| 17:54.38 | Hmmhesays | ossossosso? |
| 17:54.55 | eald | well, if you think it emacs is an OS |
| 17:55.01 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-142296.home.otenet.gr) |
| 17:55.35 | eald | tzafrir_laptop: I saw you post in the voip-pkg list, do you know which version of * will be shipped in etch? |
| 17:56.05 | hedge77 | it's debian so probably 0.9-beta2 or something |
| 17:56.28 | tzafrir_laptop | eald, http://packages.debian.org/ |
| 17:56.29 | eald | %-) |
| 17:56.38 | tzafrir_laptop | it's not going to change |
| 17:57.04 | eald | ok, thanks |
| 17:57.07 | tzafrir_laptop | hedge77, woody (3.0) had Asterisk 0.1 |
| 17:57.25 | tzafrir_laptop | Sarge had 1.0.7 . Etch has 1.2.13 |
| 17:57.30 | ikear | tzafrir_laptop: http://pastebin.ca/417030 |
| 18:01.59 | hedge77 | tzafrir_laptop: changed context-type to text/plain and it still does it. I'll have to wait around for zoa i guess |
| 18:02.59 | tzafrir_laptop | eald, for the moment I'm backporting stuff to Etch . |
| 18:03.53 | tzafrir_laptop | From what I saw in the Etch cycle, you cannot always rely on backports to be as timely as the main packages |
| 18:04.19 | ikear | tzafrir_laptop: |
| 18:04.28 | ikear | the bugger works on asterisk1.12 but not on 1.14 |
| 18:04.29 | eald | tzafrir_laptop: do you have a repository for that? |
| 18:04.31 | ikear | blah |
| 18:05.04 | tzafrir_laptop | eald, deb http://updates.xorcom.com/rapid etch main |
| 18:05.28 | eald | thanks |
| 18:05.43 | tzafrir_laptop | Most packages (except zaptel) are built from pkg-voip |
| 18:06.02 | tzafrir_laptop | it also has freepbx that is not yet in Sid |
| 18:06.10 | tzafrir_laptop | (but is in pkg-voip) |
| 18:07.30 | eald | thanks for the info |
| 18:08.26 | ars247 | Does anyone know if Asterisk 1.2.16 support Call Pickup? |
| 18:09.23 | eald | with sources, nice |
| 18:11.26 | Hmmhesays | call pickup? |
| 18:11.37 | *** join/#asterisk Jon335 (i=jon335@unaffiliated/jon335) |
| 18:12.18 | Jon335 | does anyone here have experience with the Sipura SPA-3000? |
| 18:12.37 | ars247 | yeah call pickup |
| 18:12.44 | ars247 | Direct or Group call pickup |
| 18:13.47 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:14.21 | paolob | Hi guys! is asterisk capable of speaking fax's T38 protocol? thank you! |
| 18:15.23 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:17.49 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
| 18:17.57 | zoa | yo yo |
| 18:18.56 | bkruse | yoyo! |
| 18:19.20 | *** join/#asterisk kinsey (n=bal@58.85-200-224.bkkb.no) |
| 18:19.32 | bsd_tech | yoyoyo |
| 18:19.42 | bsd_tech | Yugio |
| 18:20.37 | bkruse | yay |
| 18:21.41 | tzafrir_laptop | hedge77, still here? |
| 18:22.39 | kinsey | when sending an incoming call to a client phone, is there a way to keep asterisk from sending @ipofasterisk as FROM? |
| 18:22.44 | tzafrir_laptop | paolob, hopfully Asterisk 1.4 will let T.38 flow unharmed |
| 18:22.46 | kinsey | I would like only the external number.. |
| 18:22.50 | *** join/#asterisk Fieldy (i=4IJV7ebx@gentoo/contributor/Fieldy) |
| 18:24.06 | *** join/#asterisk canapa (i=canapa@091-141-058-213.dyn.one.at) |
| 18:24.15 | paolob | tzafrir_laptop, that means that 1.2 still doesn't feature it...? or does it feature it with problems? |
| 18:24.33 | zoa | 1.2 doesnt do it at all |
| 18:24.51 | *** join/#asterisk Meaty` (n=meaty3@office.abi.ca) |
| 18:27.25 | *** join/#asterisk SGN (n=a@adsl-66-122-88-147.dsl.renocs.nvbell.net) |
| 18:28.10 | *** join/#asterisk MNSain-AZ (n=MNSain@66.218.245.199) |
| 18:28.47 | MNSain-AZ | Good afternoon. |
| 18:28.50 | paolob | zoa, tzafrir_laptop, thank you! |
| 18:29.00 | JunK-Y | is there any way to register with a particular outbound-proxy? |
| 18:29.15 | aydiosmio | what's the channel name format for show channel? I paste SIP/vz-outbound-0a18 but it says "not a known chanel" |
| 18:29.25 | aydiosmio | (that channel name is in show channels) |
| 18:29.47 | SGN | Question to anyone that can help. I've got a TDM800P installed, with Asterisk 1.4.2 & Zaptel 1.4.1. Its all configured (to the best on my knowledge), but the Zap channel doesn't register on the start of Asterisk. Any help would be much appreciated. |
| 18:30.01 | JunK-Y | show channel <tab> |
| 18:30.15 | aydiosmio | JunK-Y: thanks |
| 18:31.54 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 18:31.59 | MNSain-AZ | Anyone here run a Digium PRI card? |
| 18:32.32 | chefrs | Why is it that if I set callprogress=yes in zapata.conf whenever I now dial out it just keeps ringing and ringing and ringing, even if the other line *does* answer. But if I comment out callprogress=yes, Asterisk fails to notice when a line is hungup. |
| 18:33.09 | aydiosmio | because callprogress is unreliable with zap channels |
| 18:33.16 | aydiosmio | :( sad panda |
| 18:33.43 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:33.59 | chefrs | Gah. |
| 18:34.37 | chefrs | But if I switch the context to from-pstn, then I can't do inbound routes stop working correctly. |
| 18:34.49 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-d32f25b720915495) |
| 18:34.52 | chefrs | Err... they stop working.. |
| 18:37.09 | zoa | JunK-Y: i dont think there is a way with to register with a particular outbound proxy |
| 18:37.13 | zoa | im not sure though |
| 18:37.24 | zoa | olle might have a special branch that actually does it |
| 18:37.29 | zoa | you might want to check with him |
| 18:37.44 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net) |
| 18:40.19 | *** join/#asterisk sysreq (n=sysreq@193.203-ppp.3menatwork.com) |
| 18:41.17 | MNSain-AZ | Any one seen this: |
| 18:41.20 | MNSain-AZ | "PRI got event: HDLC Abort (6) on Primary D-channel of span 1" |
| 18:43.06 | Corydon-w | Yep |
| 18:43.37 | MNSain-AZ | How do ya fix it? ;-) |
| 18:44.07 | Corydon-w | Well, could be a number of things, but it usually indicates that you have line noise on your T1 circuit |
| 18:44.45 | MNSain-AZ | <nod> I'll call the Telco and get them to check it. |
| 18:45.22 | MNSain-AZ | I did have the card on a shared IRQ, which I discovered is a BAD plan. |
| 18:45.58 | JunK-Y | zoa: i know we could register and throw the register with some context. |
| 18:46.36 | MNSain-AZ | Cory: Is this related to the HDLC error: "chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway!" |
| 18:47.04 | hedge77 | tzafrir_laptop: i'm back |
| 18:47.04 | Corydon-w | Yes |
| 18:47.21 | MNSain-AZ | It just so happens that 24 is actually my dchan and it's configured as such. |
| 18:47.25 | MNSain-AZ | Ok. Thanks. |
| 18:47.40 | MNSain-AZ | I'm off to harrass the Telco. Thanks for your time. |
| 18:47.53 | Corydon-w | That error means that it could not find D-channel signalling in the 24th channel |
| 18:48.18 | Corydon-w | could be due to frame slips or other problems |
| 18:53.10 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 18:54.20 | andy-andy | quit |
| 18:54.22 | andy-andy | exit |
| 18:55.26 | *** join/#asterisk Fieldy (i=T96V9CzO@gentoo/contributor/Fieldy) |
| 19:01.03 | Ifaistos | does anyone know if we can use the blf function of Grandstream gxp-2000 to monitor zap channels instead of sip extensions ? |
| 19:01.47 | zoa | Ifaistos: yes |
| 19:01.51 | zoa | it should work fine |
| 19:01.52 | zoa | with hint |
| 19:02.41 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 19:02.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:03.48 | Ifaistos | so i need to add something like subscribecontext=ext-local-custom to the zapata.conf ? |
| 19:04.51 | aydiosmio | can someone give me an idea of what needs to be reconfigured in the case * sends back 603 Declined? |
| 19:08.43 | punkgode | hi, this is not directly related to Asterisk, but maybe someone can help me. I'm provisioning several SPA3201 but now I need to remotely setup Port Forwarding, but I can't find the syntax to define this on the xml file. Anyone knows how to do this? |
| 19:14.15 | Voice2 | wahts version of libspandsp works with 1.4.1 |
| 19:14.16 | Voice2 | 1.4.2 |
| 19:15.19 | SGN | I can't get chan_zap to be compiled in asterisk. I have zaptel hardware installed (TDM800P). I compiled the zaptel packages (1.4.2). In addition it is configured and running properly on the system. Even still, every time I try to compile asterisk, it doesn't compile chan_zap. Any thoughts? Would be greatly appreciated. |
| 19:19.40 | *** join/#asterisk mick-w (n=mick@85.189.195.131) |
| 19:23.16 | *** join/#asterisk xenon4356 (i=4852f16e@gateway/web/cgi-irc/ircatwork.com/x-a5dbfcbcd108dcef) |
| 19:28.59 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:32.50 | bsd_tech | I had this issue |
| 19:33.04 | bkruse | SGN: make menuselect |
| 19:33.09 | bsd_tech | no |
| 19:33.21 | bsd_tech | it has to do with zaptel and the proper dir |
| 19:33.21 | bkruse | go to channels -> and see why it says chan_zap ISNT getting built |
| 19:33.41 | SGN | well in menu select it is selected |
| 19:33.49 | bkruse | oh |
| 19:33.49 | bkruse | hmm |
| 19:33.52 | bkruse | nvm then :] |
| 19:33.52 | bsd_tech | it looks for zaptel.h in /usr/include/zaptel |
| 19:34.20 | bsd_tech | if it does not find that dir when builing it wont build zap |
| 19:34.39 | SGN | the dir is there and there is tonezone.h and zaptel.h there |
| 19:36.50 | SGN | Is there any other files that asterisk compile looks for? |
| 19:36.56 | SGN | from zaptel? |
| 19:37.38 | mick-w | hey all :) |
| 19:37.53 | tzafrir_laptop | <PROTECTED> |
| 19:38.10 | tzafrir_laptop | In older versios it was /usr/linux/zaptel |
| 19:38.18 | SGN | Well I have zaptel 1.4.1 and asterisk 1.4.2 |
| 19:38.22 | tzafrir_laptop | sorry: /usr/include/linux/zaptel/h |
| 19:39.23 | mick-w | i have a q: when i call an extension that is configured to go to sip and the sip account is not logged on, it takes about 4 or 5 secs to refuse the call |
| 19:39.25 | mick-w | whys that? |
| 19:39.27 | tzafrir_laptop | what version of zaptel do you hve? |
| 19:39.45 | SGN | 1.4.1 |
| 19:40.08 | SGN | I just downloaded it yesterday |
| 19:40.42 | tzafrir_laptop | mick-w, what do you mean by "reject the call"? Have you looked at the CLI trace? |
| 19:41.17 | mick-w | tzafrir_laptop: sorry no i haven't, i'm very new to asterisk, just finding my feet i'm afraid! |
| 19:41.30 | tzafrir_laptop | asterisk -rvvv |
| 19:41.31 | mick-w | tzafrir_laptop: by reject it i mean my sip phone says call refused |
| 19:41.47 | tzafrir_laptop | (from the linux terminal) |
| 19:41.51 | tzafrir_laptop | and then call |
| 19:41.54 | mick-w | tzafrir_laptop: ahh sorry yeh im used to that, sorry i didnt think CLI = command line lol |
| 19:42.04 | mick-w | one sec |
| 19:42.13 | tzafrir_laptop | Command Line Interface |
| 19:42.26 | mick-w | yeh sorry i know that part, just didnt twig for some reason :P |
| 19:43.07 | hedge77 | is there a good guide to picking which codecs to use anywhere? |
| 19:43.08 | mick-w | ok so right we get this: |
| 19:43.26 | mick-w | got a pastebin you guys like to use? |
| 19:43.36 | tzafrir_laptop | SGN, what exactly do you see on the configure output regarding zaptel? |
| 19:44.00 | SGN | sec |
| 19:44.37 | mick-w | ok heres the trace: |
| 19:44.38 | mick-w | http://pastebin.ca/417276 |
| 19:45.26 | mick-w | so it takes about 10 seconds to find out that no-one has registered to SIP/paul.mckenzie |
| 19:45.44 | mick-w | welll in fact it doesnt, it figures that out straight away |
| 19:45.45 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 19:45.53 | mick-w | but takes 10 seconds to come up with the timeout |
| 19:46.20 | mick-w | it references a rule t, sounds as if the t rule in the internal context would sort it? |
| 19:46.50 | SGN | tzafrir_laptop: http://pastebin.ca/417283 |
| 19:47.00 | aydiosmio | ick, freepbx chokes on E.164 formatted DIDs |
| 19:47.56 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 19:48.55 | *** join/#asterisk tmjb (n=tmjb@195.252.125.18) |
| 19:49.24 | tmjb | could someone tell me what is this app_voicemail.c:2299 inboxcount: Failed to obtain database object for 'asterisk'! ? |
| 19:49.43 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 19:50.35 | tzafrir_laptop | SGN, hmm.. but it finds zaptel.h |
| 19:50.46 | bkruse | russellb: let me know if you wana hit up some bowling tonight |
| 19:51.08 | jer | if i have a number that i want when someone calls it, to ring on a bunch of different numbers (SIP phones, and lines outside the PBX) ... how might i go about doing that? |
| 19:51.27 | bsd_tech | not anymore |
| 19:51.32 | bsd_tech | there have been changes |
| 19:51.47 | bsd_tech | freepbx has grown alot |
| 19:52.04 | bsd_tech | I use it on a few systsms adn have no problem with enum |
| 19:52.05 | mick-w | tzafrir_laptop: did you see that pastebin? :) |
| 19:53.10 | hedge77 | jer: for just multiple sip numbers it's like Dial(SIP/100&SIP/101&SIP/102) but i don't know if you can put a Zap/whatever in there also |
| 19:53.51 | jer | hedge77, yeah, i know how to do multiple sip numbers |
| 19:53.58 | tzafrir_laptop | mick-w, so it's a dialplan issue |
| 19:54.42 | mick-w | tzafrir_laptop: i reckon i just saw something which points me in the right direction, i didnt realise dial() had moe arguments |
| 19:54.43 | tzafrir_laptop | in 20:42:55 the dial failed, but only in 20:43:05 it timed out on the dialplan |
| 19:55.04 | mick-w | tzafrir_laptop: seems as though if i amend the timeout on the rule i use in extensions.conf i could change it? |
| 19:56.03 | russellb | bkruse: i'll school you. |
| 19:56.38 | tzafrir_laptop | Try putting an explicit Hangup after htat Dial |
| 19:56.45 | hedge77 | jer: it works like you'd expect here I just tried it |
| 19:57.11 | jer | hedge77, what does? multiple sip phones ringing? |
| 19:57.17 | tmjb | I get this warning and for some reason i can not write to spool voice mail box i run asterisk as a root so permission should not be a problem |
| 19:57.18 | hedge77 | jer: and outside lines |
| 19:57.21 | jer | oh |
| 19:57.24 | bkruse | russellb: bring it! |
| 19:57.31 | jer | interesting; here it just keeps ringing and ringing |
| 19:57.35 | jer | (with no phones ringing) |
| 19:57.49 | mick-w | tzafrir_laptop: ok ill try the hangup |
| 19:57.56 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 19:58.32 | hedge77 | jer: 5551212 => Dial(${GLOBAL(TRUNK)}/${EXTEN:${TRUNKMSD}}&SIP/101&SIP/102); called all three, but i didn't try and talk on them |
| 20:00.48 | mick-w | tzafrir_laptop: hey that hangup worked |
| 20:00.51 | hedge77 | jer: just tried it with my cell and the first phone to answer gets the call, not all three. |
| 20:01.03 | mick-w | tzafrir_laptop: so if you dont do a hangup it just waits for the dial to timeout? |
| 20:03.15 | jer | hedge77, right |
| 20:04.57 | mick-w | any way of making a hangup line apply to multiple extensions? |
| 20:06.08 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:06.50 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 20:07.13 | TripleFFFF | i got 1 way audio on asterisk 1.4.2 .. doing the echo test.. no audio at all |
| 20:07.24 | *** join/#asterisk ccesario (n=ccesario@201-0-124-24.dsl.telesp.net.br) |
| 20:07.29 | TripleFFFF | localnet and external ip set ok |
| 20:07.30 | tzafrir_laptop | TripleFFFF, which phone? |
| 20:07.41 | TripleFFFF | eyebeam |
| 20:07.42 | TripleFFFF | to test |
| 20:07.42 | ccesario | Hi peoples, how to I capture one call that is ring in other extension ? |
| 20:07.52 | TripleFFFF | eb --> WAN -> ASTERISK |
| 20:08.10 | TripleFFFF | but.. EB ==> WAN ==> DEMO i hear the voices |
| 20:08.44 | tzafrir_laptop | what is EB? |
| 20:08.49 | TripleFFFF | eyebeam |
| 20:09.07 | tmjb | does anyone know why i am getting this warning app_voicemail.c:2299 inboxcount: Failed to obtain database object for 'asterisk'! ??? |
| 20:09.32 | tzafrir_laptop | ccesario, chanspy? |
| 20:09.42 | tzafrir_laptop | ccesario, in zaptel: zapbarge |
| 20:09.57 | Corydon-w | tmjb: because you've enabled ODBC in voicemail and have failed to configure res_odbc ? |
| 20:12.11 | TripleFFFF | hmm |
| 20:12.14 | tzafrir_laptop | what is the "demo" vs. the "asterisk"? |
| 20:12.17 | TripleFFFF | so.. weird |
| 20:12.23 | TripleFFFF | i meant |
| 20:12.27 | ccesario | tzafrir_laptop, I don't know the correct trem.... but I want cature the call ringing in other extension |
| 20:12.28 | TripleFFFF | demo context thing |
| 20:12.43 | Corydon-w | If you hear voices, you might want to try professional help |
| 20:12.50 | TripleFFFF | [demo] |
| 20:13.02 | TripleFFFF | exten => s,n,WaitExten ; Wait for an extension to be dialed. |
| 20:13.02 | TripleFFFF | exten => 2,1,BackGround(demo-moreinfo) ; Give some more information. |
| 20:13.05 | TripleFFFF | this i here |
| 20:13.06 | tmjb | Corydon-w,nop i did not is there any I say voicemail.conf no user odbc ? |
| 20:13.07 | TripleFFFF | hear |
| 20:13.09 | TripleFFFF | but not echo() |
| 20:13.33 | Corydon-w | tmjb: rephrase, please |
| 20:13.58 | TripleFFFF | so if i dial 2 i hear congratulations you have succefully got to the point your weekend is ruined.. |
| 20:14.00 | TripleFFFF | ;) |
| 20:14.14 | TripleFFFF | but exten => 999,1,Echo() |
| 20:14.18 | TripleFFFF | i get nothing niet |
| 20:14.59 | TripleFFFF | it executes ok.. no audio.. also the rtp sends to my lan ip in all ips i see.. it should use natted |
| 20:15.39 | mick-w | how do i go about getting a timing source on debian running 2.4.27-3-386 ? |
| 20:15.39 | *** join/#asterisk ZefK (n=m@81.181.249.106) |
| 20:15.54 | tmjb | tmjb,problaby the problem is that i builded support for odbc but i did not say anywhere to use ;odbcstorage=asterisk so is there way i can say user just write to filesystem ? |
| 20:16.16 | tzafrir_laptop | mick-w, do you have a uhci USB controller? |
| 20:16.31 | mick-w | on this mobo im not sure, let me check |
| 20:16.34 | tzafrir_laptop | grep hci /proc/interrupts |
| 20:17.13 | TripleFFFF | ill try musiconhold |
| 20:17.24 | tzafrir_laptop | generally: if you have a USB chip of intel or via |
| 20:17.56 | mick-w | that grep didnt return anything |
| 20:18.04 | mick-w | its an epia 5000 so thats via |
| 20:18.05 | tzafrir_laptop | mick-w, another option is: apt-get install kernel-image-2.6-686 |
| 20:18.12 | TripleFFFF | music on hold works |
| 20:18.19 | tzafrir_laptop | (or 386, if you actually prefer that) |
| 20:18.23 | TripleFFFF | so eho test broken ?\ |
| 20:18.49 | tzafrir_laptop | so kernel-image-386 , indeed |
| 20:18.51 | mick-w | i think its best to stick to 386 as the epia 5000 has a cpu that doesnt have a certain execution, 686 kernels apparently can crash out |
| 20:18.56 | tmjb | Corydon-w, the problem is that i builded support for odbc but i did not say anywhere to use ;odbcstorage=asterisk so is there way i can say user just write to filesystem ? |
| 20:19.00 | JunK-Y | TripleFFFF: MOH has a generator, a playback is okay? |
| 20:19.11 | TripleFFFF | seems like it |
| 20:19.14 | mick-w | ok does the 686 have the ztdummy driver built in? |
| 20:19.15 | TripleFFFF | only echo is not working |
| 20:19.18 | mick-w | sorry not 686 |
| 20:19.22 | TripleFFFF | and one way on rest |
| 20:19.22 | mick-w | 2.6 |
| 20:19.24 | Corydon-w | tmjb: No, you must rebuild app_voicemail with the option turned off |
| 20:19.34 | TripleFFFF | cat /etc/redhat-release |
| 20:19.34 | TripleFFFF | Enterprise Linux Enterprise Linux AS release 4 (October Update 4) |
| 20:19.38 | Qwell[] | tmjb: right now, it's an all or none deal |
| 20:19.44 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 20:19.46 | ZefK | I'm looking for a provider that offers DIDs in Rusia. Could anyone advise me? |
| 20:19.52 | tzafrir_laptop | mick-w, the zaptel-source package of Sarge is still broken, so building it with m-a will not work |
| 20:20.07 | Qwell[] | What's m-a? |
| 20:20.24 | tzafrir_laptop | add deb http://updates.xorcom.com/rapid sarge main |
| 20:20.41 | tzafrir_laptop | and install zaptel and zaptel-modules-`uname -r` |
| 20:20.49 | tmjb | Corydon-w,thats what I was afraid off ok then lets build the damn thing ;) Tnx guys very mutch should this be in some faq or wiki or something ? |
| 20:21.03 | tzafrir_laptop | m-a is modules-ssistant, to build kernel modules from kernel-sources packages |
| 20:21.07 | Qwell[] | ahh |
| 20:21.30 | mick-w | hmmm cheers tzafrir_laptop |
| 20:21.33 | mick-w | :) |
| 20:22.17 | tzafrir_laptop | Good night folks |
| 20:23.18 | TripleFFFF | waht could be breaking echo but not playback or hold ? |
| 20:23.26 | *** part/#asterisk adam_vollrath (n=adam@pc152.BISonline.com) |
| 20:23.40 | Qwell[] | TripleFFFF: answer? |
| 20:24.05 | garreel | wich is the code to make callerid visible on zap channels even if it defaults to hidden? |
| 20:24.17 | garreel | anyone knows? |
| 20:26.14 | *** join/#asterisk mmartinn (n=martins@128.227.123.22) |
| 20:26.25 | *** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
| 20:26.32 | mmartinn | Is there any way to get a command like "zap show channels" to only show active calls on them? |
| 20:26.47 | mmartinn | Similar to what zttool could do before analog lines |
| 20:28.14 | TripleFFFF | ah |
| 20:28.15 | TripleFFFF | no |
| 20:29.59 | *** join/#asterisk Pepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
| 20:30.09 | Pepse | Afternoon, gents |
| 20:30.26 | TripleFFFF | if i asnwer first.. i here some kind of WIND noise |
| 20:30.29 | TripleFFFF | nothing else |
| 20:31.21 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 20:31.29 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:33.32 | [hC] | Qwell: get anywhere with that 7970? my chan_sccp just decided recently to start doing one way audio. randomly. |
| 20:34.33 | Pepse | I can't believe nobody's made an Aim Phoneline to Asterisk gateway yet |
| 20:34.38 | Pepse | with the API being open and all |
| 20:36.03 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 20:36.14 | TripleFFFF | ah |
| 20:36.23 | TripleFFFF | echo doest say this is an echo test !!! lol |
| 20:36.30 | tuan_modulis | what version of asterisk is asterisknow compatible with? |
| 20:36.33 | TripleFFFF | duh |
| 20:36.48 | tuan_modulis | like... only the 1.4 ? |
| 20:38.14 | Pepse | yeah |
| 20:39.39 | mick-w | hmmm ive got the zaptel modules for my kernel which is cool :) seeing as i have kernel 2..4 i cant use ztdummy, so i have to use zaprtl right? |
| 20:42.07 | mick-w | hmm i dont think i have the right usb controller in fact |
| 20:42.13 | mick-w | dont have usb_uhci loaded |
| 20:43.26 | *** part/#asterisk mmartinn (n=martins@128.227.123.22) |
| 20:50.37 | *** join/#asterisk monstertruck (n=monstert@74.167.124.204) |
| 20:51.19 | monstertruck | Hi |
| 20:51.31 | monstertruck | i have a weird issue with agi and get_variable |
| 20:51.48 | monstertruck | first call to get_variable returns invalid command |
| 20:51.50 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:52.05 | monstertruck | but the second call, returns the value of the variable requested on the first call |
| 20:52.26 | monstertruck | and from there, every other call returns the value the previous call should have returned |
| 20:53.02 | monstertruck | anybody seen this before? |
| 20:53.40 | TripleFFFF | how do i enable outside perms to httpd conf ? |
| 20:54.04 | TripleFFFF | its lcokign down on localnet hosts only |
| 21:02.45 | *** join/#asterisk wcfxswtf (i=patrick@notchill.com) |
| 21:03.19 | wcfxswtf | any idea what happened to the wcfxs module? doesnt seem to be in src and im getting "Module 0: FAILED FXS (FCC)" on boot |
| 21:07.34 | *** join/#asterisk JoeDeveloper (n=jdevel@www.airlinksystems.com) |
| 21:09.17 | *** part/#asterisk JoeDeveloper (n=jdevel@www.airlinksystems.com) |
| 21:10.01 | *** part/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:10.47 | bsd_tech | bind to 0.0.0.0 |
| 21:13.06 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
| 21:13.10 | zoa | yoyo |
| 21:13.11 | hedge77 | is there a reason the times in 'show translation' would be significantly higher from a 1.0.something to a 1.4.2 system on similar hardware? |
| 21:13.14 | zoa | im back |
| 21:14.16 | zoa | not really |
| 21:14.27 | zoa | its not like they could have optimised that a whole lot |
| 21:14.38 | hedge77 | that's what I can't figure out |
| 21:14.51 | zoa | maybe one is with HT off and the other one ON ? |
| 21:14.55 | zoa | or different compile options ? |
| 21:16.22 | hedge77 | they're both kernel 2.6. hmm one is a 3ghz p4 one is 2.4 but i didn't think it would make a difference at that speed |
| 21:18.14 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
| 21:18.59 | hedge77 | * -> ilbc times are 6ms worse (15 vs 21) |
| 21:20.26 | hedge77 | and * -> gsm are 2-3 times worse (2ms on 1.0, 4-6ms on 1.4.2) |
| 21:27.21 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 21:37.48 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 21:46.13 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:49.13 | *** join/#asterisk gapcaller (n=m@adsl-75-57-186-122.dsl.emhril.sbcglobal.net) |
| 21:50.01 | gapcaller | hey guys I need a little help. I am trying to setup an SPA-3102 PSTN gateway and I get the message "all circuits are busy now" when I try to dial out |
| 21:54.17 | gapcaller | anyone here? |
| 21:55.15 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 21:57.27 | tonyb2006 | maybe |
| 21:57.30 | tonyb2006 | maybe not |
| 21:57.35 | tonyb2006 | <_< |
| 21:58.07 | gapcaller | any suggestions? |
| 21:58.53 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 22:01.15 | *** join/#asterisk kannan (n=kannan@210.211.178.193) |
| 22:02.10 | *** part/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 22:11.55 | *** join/#asterisk Defraz (n=t0tal@209.137.240.88) |
| 22:13.23 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
| 22:13.25 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 22:16.52 | *** join/#asterisk Erik2300 (n=ewilhelm@67-132-248-66.dia.static.qwest.net) |
| 22:17.42 | Erik2300 | Can anyone help with getting IMAP to work in Asterisk? |
| 22:17.57 | Erik2300 | I have it installed against the UW C-Client |
| 22:18.08 | Erik2300 | but I get errors |
| 22:21.56 | JunK-Y | specifying the error will help if we dont want to be here in 2300 :) |
| 22:23.15 | drfreeze | I'm having problems getting a Polycom501 to be recognized by asterisk. Can anyone help? |
| 22:23.28 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 22:25.55 | un_j | :-) |
| 22:26.28 | un_j | hello exten => *3, 1, ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${DIALEDPEERNUMBER}|numberplan-custom-1,${DIALEDPEERNUMBER},1) - how to make it call back if called from zaptel? |
| 22:26.28 | Erik2300 | the error i get it this can't connect to IMAP server {192.168.8.15:143/imap/notls/user=ewilhelmsen} INBOX |
| 22:27.04 | Erik2300 | I have tried setting up voicemail.conf different ways and I always get a similar error message |
| 22:30.47 | *** join/#asterisk andylauppe (n=alauppe@c-71-230-37-238.hsd1.pa.comcast.net) |
| 22:31.34 | andylauppe | greetings all; I own 12 g729 licenses but need to get them transfered to a different machine. is this difficult? |
| 22:31.51 | wcfxswtf | is there an asterisk 1.4 package for debian yet? |
| 22:32.50 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 22:33.04 | drfreeze | Why would a Polycom phone get a different SNTP address than that which is specified inthe admin config |
| 22:33.26 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net) |
| 22:33.27 | Erik2300 | anyone have any idea on the error |
| 22:33.49 | fx0 | a |
| 22:34.27 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 22:36.05 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 22:37.16 | drfreeze | anyone remember the default user/passwd for Polycom web interface? |
| 22:37.30 | wunderkin | it is in the admin guide |
| 22:37.37 | Corydon-w | Polycom/456 |
| 22:38.20 | drfreeze | Corydon-w: thx |
| 22:38.20 | zoa | andylauppe: just call digium support |
| 22:39.17 | _DAW | drfreeze - there is a sntp dhcp override switch in the sip.cfg |
| 22:39.25 | drfreeze | Corydon-w: gotta polycom501 that is getting a different SNTP address than in the config. Any ideas why? |
| 22:39.26 | *** join/#asterisk ak_hepcat (n=ak_hepca@patroler.gci.net) |
| 22:39.37 | Corydon-w | DHCP |
| 22:40.12 | drfreeze | _DAW: yeah, it is getting an address we don't recognize. not what we setup in the phone |
| 22:40.17 | andylauppe | zoa: i looked on their webpage for a moment and couldn't find the number. anyone have it on the top of their mind? |
| 22:42.33 | *** part/#asterisk andylauppe (n=alauppe@c-71-230-37-238.hsd1.pa.comcast.net) |
| 22:42.43 | zoa | something .xxx-linuxsupport or something |
| 22:43.22 | wcfxswtf | any big advantage running 1.4 over 1.2? |
| 22:43.31 | Erik2300 | i am getting an IMAP error working with asterisk 1.4.2 |
| 22:43.56 | Qwell[] | he left, but it's 877-linux-me |
| 22:44.08 | Erik2300 | it is Can't connect to imap server {192.168.8.15:143/imap/notls/user=ewilhelmsen}INBOX |
| 22:44.12 | Erik2300 | any help |
| 22:44.55 | un_j | exten => *3, 1, ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${DIALEDPEERNUMBER}|numberplan-custom-1,${DIALEDPEERNUMBER},1) / why won't work if call placed from zap? |
| 22:45.46 | TripleFFFF | 1.4.2 broken |
| 22:45.58 | *** part/#asterisk ak_hepcat (n=ak_hepca@patroler.gci.net) |
| 22:46.11 | TripleFFFF | i just confirmed.. daling 10 times 999. (echo.. 3 out of 5 wont work.. same as dialing anywhere.. |
| 22:46.24 | TripleFFFF | wiiper mode.. intermittent |
| 22:46.25 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 22:48.07 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 22:48.07 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:51.22 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:03.21 | wcfxswtf | <PROTECTED> |
| 23:03.28 | Qwell[] | very |
| 23:04.10 | Erik2300 | when is the new oreilly book slated to be released? |
| 23:07.25 | CunningPike | Is http://bugs.digium.com/view.php?id=4865 post 1.2.1? |
| 23:07.26 | wcfxswtf | yeah seems a bit "old" |
| 23:07.33 | *** join/#asterisk _DAW (n=chatzill@adsl-156-109-78.msy.bellsouth.net) |
| 23:07.50 | Erik2300 | it is dated now |
| 23:08.36 | wcfxswtf | 2nd edition due out soon? |
| 23:08.39 | wcfxswtf | if so ill hold off |
| 23:08.46 | wcfxswtf | and just read the pdf ... but i def prefer treeware |
| 23:09.04 | Erik2300 | there is a wiki for the new book under development |
| 23:09.13 | wcfxswtf | url? |
| 23:09.34 | Erik2300 | http://etel.wiki.oreilly.com/wiki/index.php/Main_Page |
| 23:10.37 | wcfxswtf | thanks |
| 23:10.41 | Erik2300 | yup |
| 23:10.44 | wcfxswtf | oh its a cookbook |
| 23:10.47 | Erik2300 | yeah |
| 23:10.54 | wcfxswtf | not 2nd edition |
| 23:10.55 | wcfxswtf | got it |
| 23:11.00 | Erik2300 | sorry yeah |
| 23:11.03 | Erik2300 | not a 2nd |
| 23:15.52 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 23:23.26 | *** join/#asterisk mick-w (n=mick@ip-217.146.108.113.merula.net) |
| 23:23.30 | mick-w | lo lo |
| 23:32.56 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 23:35.15 | wcfxswtf | whats the best web interface? |
| 23:36.49 | robin_sz | define best |
| 23:37.18 | robin_sz | there are many ... al suck |
| 23:37.24 | wcfxswtf | k |
| 23:37.28 | wcfxswtf | buying ora book to read this weekend |
| 23:37.36 | wcfxswtf | just wanted to add extension asap to check it out today |
| 23:40.31 | *** part/#asterisk garreel (n=garreel@host48-2-dynamic.16-87-r.retail.telecomitalia.it) |
| 23:58.55 | wcfxswtf | erm ive got the sip phones connecting, but how can i set up just simple test extensions? |
| 23:59.03 | wcfxswtf | i tried exten = 1234,1,Dial(Sip/1234) |
| 23:59.09 | wcfxswtf | s/Sip/SIP |
