| 00:00.28 | dave-outlaw | Hey guys, I'm having a problem with asterisk. Using the latest asterisk and AMP. I have some phones configured, and they all work very well, until I try to use ad-hoc mode with no default user. Logging in seems to work, but I can't ring that extension... it just goes to a busy message. Anyone know how to fix this? |
| 00:00.33 | *** join/#asterisk psi_force (n=mark@marksnb.eng.unimelb.edu.au) |
| 00:00.39 | psi_force | hi all |
| 00:02.15 | psi_force | does anyone know how a person might be changing his channel id? we allocate users a 6 digit username and we are seeing someone having a channel SIP/incoming-xxxx |
| 00:03.57 | psi_force | but all the users should have channel names like SIP/YYYYYY-xxxx |
| 00:05.05 | *** join/#asterisk calculus (n=SPAM@24-176-224-55.dhcp.snlo.ca.charter.com) |
| 00:05.49 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
| 00:05.58 | dlynes | cassio: so your service provider is you? I though the logs you were showing me were from a colo server at godaddy? |
| 00:06.44 | calculus | is it possible to use a standard dialup modem (line in, line out) to connect a hard-wired phone in to use with asterisk? |
| 00:06.45 | cassio | dlynes, its a box that I rented at godaddy.com |
| 00:07.09 | dlynes | cassio: the log that you showed me, right? |
| 00:07.14 | cassio | yes |
| 00:07.35 | dlynes | cassio: and it's trying to register some sip devices at your home, or some sip upstream ports? |
| 00:07.38 | WasPhantom | hey is anyone maintaining a "current" package for asterisk on debian? |
| 00:07.52 | calculus | cassio: was that a 'yes' to me? |
| 00:08.00 | dlynes | calculus: no, to me |
| 00:08.06 | psi_force | anyone have any ideas about this channel issue? |
| 00:08.14 | cassio | calculus its to dlynes |
| 00:08.24 | Abydos313 | WasPhantom i tried to apt-get asterisk..no go |
| 00:08.31 | cassio | dlynes http://pastebin.com/540936 |
| 00:08.58 | cassio | it looses its registration, not only for broadvoice, but for my pap2 also |
| 00:09.53 | WasPhantom | hmmm perhaps I should have a look into maintaining one.... just starting to deploy it myself, so as good an excuse as any |
| 00:10.30 | dlynes | cassio: on your pap2, there should be a line 1 page |
| 00:10.56 | dlynes | cassio: on there, there should be a setting 'NAT Keep Alive Enable' |
| 00:11.03 | dlynes | make sure that's enabled |
| 00:11.52 | dlynes | cassio: also on your SIP page, down at the bottom, there's a setting 'NAT Keep Alive Intvl' |
| 00:12.00 | dlynes | cassio: make sure that's set to 3, instead of 15 |
| 00:12.55 | dlynes | cassio: also, for your outbound and inbound sip registrations on your asterisk box at godaddy.com, make sure qualify= is set to qualify=300 |
| 00:13.15 | dlynes | or even qualify=180 is good too...it's a bit overkill, but it would also work |
| 00:13.55 | cassio | dlynes I aready tried to change for 300 but happens the same thinh |
| 00:13.56 | dlynes | cassio: then reboot your pap2, and on your server at godaddy, do a sip reload on the asterisk cli |
| 00:14.11 | robin_sz | OK, so .. assuming I have some sort of FXO clone card in, wcfxo modprobed .. how can I tell if its "working" ?? |
| 00:14.44 | dlynes | If that still doesn't work for you, the probably is probably with broadvoice |
| 00:14.44 | WasPhantom | well - if it modprobed, you're well on your way, as it detected the card as valid heh |
| 00:15.04 | dlynes | cassio: other than that, I can't help you much more...I'm on my way out the door |
| 00:15.07 | robin_sz | ie it should autoanswer, I think it used to in 1.0.7 ... but it havent touched this box in 6 onths or more, and ive forgotten what I conf'd |
| 00:15.47 | robin_sz | yeah, its modprobed |
| 00:17.39 | robin_sz | context=external |
| 00:17.39 | robin_sz | channel => 1 |
| 00:17.47 | *** join/#asterisk Pinnen (i=pinnen@jultomten.luktar.bajs.nu) |
| 00:17.48 | robin_sz | that should do it huh? |
| 00:19.07 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 00:24.11 | wunderkin | JUSTIN YOU |
| 00:24.31 | robin_sz | hmmm ... nothing about zap in show modules .. ?? |
| 00:24.33 | *** join/#asterisk M-I-A-- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 00:26.11 | robin_sz | hmm and nothing about zap in showchanneltyoes?? |
| 00:30.29 | tronix | is zapata.conf set up? |
| 00:30.37 | tronix | and chan_zap.so in modules dir, readable? |
| 00:30.43 | tronix | try starting * with -vvvvvc |
| 00:30.49 | tronix | it'll tell you if it saw an error when trying to |
| 00:30.52 | *** join/#asterisk jovan (n=jovan@host181-98.pool8710.interbusiness.it) |
| 00:30.52 | tronix | load chan_zap. |
| 00:30.57 | jovan | hi |
| 00:31.09 | tronix | good evening |
| 00:31.26 | robin_sz | ah, no chan_zap.so |
| 00:31.27 | jovan | :) |
| 00:32.17 | robin_sz | chan_zap comes from zaptel right? |
| 00:32.20 | tronix | yes |
| 00:32.41 | tronix | I'm not as familiar with how clone hardware do it for the software stuff on * server |
| 00:33.20 | tronix | but if it's true clones, should work with zaptel installation |
| 00:33.30 | robin_sz | just pretend its a T100P ... |
| 00:33.34 | robin_sz | it USED to work .. |
| 00:33.44 | tronix | you need two parts config'd: |
| 00:33.46 | robin_sz | but I did the upgrade to 1.0.2.4 |
| 00:33.49 | tronix | kernel side and * side |
| 00:33.53 | tronix | sounds like you have kernel side up |
| 00:33.55 | tronix | but not * side |
| 00:33.59 | robin_sz | right ... |
| 00:34.18 | robin_sz | just makeing linux26 again in zaptel ... |
| 00:34.22 | tronix | cool |
| 00:34.28 | robin_sz | paying attention this time :) |
| 00:34.31 | tronix | :) |
| 00:35.29 | robin_sz | I guess its a debain thing and its putting it somewhere other than /usr/lib/asterisk/modules/ |
| 00:35.53 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
| 00:35.58 | tronix | q for #asterisk: in olden days, we were told not to plug modems into hotel room PBX ports. does this still hold true, for various analog telephony devices? |
| 00:36.07 | tronix | (in the U.S.) |
| 00:36.13 | WasPhantom | which model are you looking for? |
| 00:36.35 | tronix | robin_sz: ahhh. haven't tried * on Debian, myself, though wouldn't surprise me because Debian is strict about the FSSTND |
| 00:37.03 | WasPhantom | I've got it going on a couple of debian boxes |
| 00:37.09 | robin_sz | I think some hotel PBX ports are a bit "weird" .. have "other" signallign going on, as wel as regular phane crap |
| 00:37.46 | WasPhantom | neil@king:~$ locate chan_zap.so |
| 00:37.46 | WasPhantom | /usr/lib/asterisk/modules/chan_zap.so |
| 00:37.55 | robin_sz | some for example, ring an alarm if you disconnect the phone ... |
| 00:37.58 | tronix | big danger, I seem to vaguely recall, was from power frying stuff |
| 00:38.09 | robin_sz | shrug ... |
| 00:38.28 | robin_sz | any half-decent hotle provided a LAN these days |
| 00:38.47 | tronix | indeed |
| 00:38.59 | WasPhantom | robin_sz, how did you install * on the linux box, as you can see above, mine has installed it in the correct location |
| 00:39.20 | robin_sz | well, I dl'd tjhe tow lates lumps * and zaptel |
| 00:39.26 | robin_sz | removed my old .debs |
| 00:39.41 | robin_sz | untarred, maked and make installed |
| 00:41.02 | robin_sz | hmm /// dont see anything about chan_zap.so in zaptel ... |
| 00:41.02 | WasPhantom | have you configured /etc/zaptel.conf ? |
| 00:41.11 | robin_sz | ahh, its NOT in zaptel, chan_zap coems from * |
| 00:41.12 | WasPhantom | and /etc/asterisk/zapata.conf |
| 00:41.32 | *** join/#asterisk azzie (i=az@cpe-24-168-17-173.si.res.rr.com) |
| 00:41.37 | tronix | ahh my bad. sorry |
| 00:42.05 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 00:42.07 | robin_sz | you are forgiven :) |
| 00:42.24 | Snake-Eyes | has any one here used scopeserv before ? |
| 00:44.26 | robin_sz | well, lots. |
| 00:45.02 | robin_sz | Snake-Eyes: ive tried scopdex once ... |
| 00:45.06 | robin_sz | similar. |
| 00:45.22 | robin_sz | well, not that similar |
| 00:45.37 | WasPhantom | ok - not to sound silly, but you're using debian yeah? why are you building * from source? ( I do realise that * deb packages aren't "current" at the moment, but they're stable ) |
| 00:46.27 | robin_sz | WasPhantom: because 1.0.7 is ancienne, and I wanna try the blleding edge |
| 00:46.43 | litage | does asterisk load into ram all of the settings read from /etc/asterisk/ ? |
| 00:46.51 | mzo | if i stab you a lot, will that be bleeding edge? :) |
| 00:47.04 | WasPhantom | robin_sz, fair enough, just asking ;-) |
| 00:47.17 | robin_sz | and we run 1.0.2.x in the "production" machine, I wnat to develop a dialplan that stands a chance of working |
| 00:47.37 | robin_sz | maybe even with ael ... |
| 00:47.49 | wunderkin | >:( chanspy is acting really weird for me |
| 00:48.10 | Snake-Eyes | robin_sz, do you know why the dial plan info isnt stored in mysql database like the rest of info for extension with scopeserv ? |
| 00:48.26 | robin_sz | Snake-Eyes: no. |
| 00:49.20 | robin_sz | scopdex is a mixture of scopalamine and dexedrine ... try some, it might not help, but at least you won;t worry about it at the time ;) |
| 00:50.58 | robin_sz | argh .. chan_zap.c craps itself badly on compile ... |
| 00:51.10 | mzo | did it leave smears? |
| 00:51.38 | robin_sz | oh , need a newer libpri |
| 00:51.45 | robin_sz | smears? nah it lefts LUMPS! |
| 00:51.54 | mzo | eww, |
| 00:51.58 | mzo | core dumps, ewww. |
| 00:52.20 | wunderkin | hooooowdy hooooooo |
| 00:54.32 | robin_sz | merry christmas :) |
| 00:55.42 | psi_force | does anyone know how to restrict sip users from not trunking or how to stop them from overwritting their channel name |
| 00:55.57 | wunderkin | yey now if i could get sphinx and chanspy figured out id be all set |
| 00:56.01 | robin_sz | from not trunking? easy. |
| 00:56.04 | file | they don't overwrite their channel name... it's generated by chan_sip based on a few factors |
| 00:56.22 | robin_sz | just put them in a context that doesnt have trunk access |
| 00:58.33 | robin_sz | so .. in sip.conf put each sip user in "context=sipusers" or soem such |
| 00:58.48 | robin_sz | in extensions.conf [sipusers] |
| 00:59.01 | robin_sz | include => localphones |
| 00:59.14 | robin_sz | or whatever context you put your local phoens in ... |
| 00:59.46 | robin_sz | just dont include international or whatever trunk contexts you use for non-sip users |
| 01:00.39 | robin_sz | ahh, now I have zap channels :) |
| 01:01.34 | cassio | does anyone know why asterisk looses all its registration, for peers and lines in about 1 min after running it? |
| 01:02.45 | robin_sz | it doesnt does it? |
| 01:02.53 | cassio | all connections get time out on 1 minute |
| 01:02.57 | robin_sz | it starts off with NO regiostrations ... |
| 01:03.05 | robin_sz | and then connections log in ... |
| 01:03.26 | robin_sz | let me guess, you are behind a NAT firewall, with your phones on the other side right? |
| 01:05.14 | cassio | robin_sz is that to me? |
| 01:05.25 | robin_sz | yes |
| 01:05.37 | *** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net) |
| 01:07.08 | cassio | no, I am not behind a nat, and it looses registration for all lines and phones, 1 minute later after registered |
| 01:08.26 | rene- | it is said that provisioning of sipura/linksys voip devices takes place in two steps, second stage can be http/tftp; is it possible to have the sipura pull its first stage boot conf from an http server? |
| 01:09.26 | robin_sz | cassio: ok, then not knowing .. i was assuming your NA was dropping all the conections .. but maybe not |
| 01:09.40 | rene- | i am just wondering, since people around here always say that tftp is insecure and everything, if it was possible to remove the tftp part in a provisioning system alltogether |
| 01:10.18 | robin_sz | arghh ... cassio said LOOSES |
| 01:10.55 | cassio | robin_sz got a clue? |
| 01:11.24 | robin_sz | I can only think its a network/firewall thing ... thats would be my first clue |
| 01:11.36 | robin_sz | and its LOSES by the way |
| 01:12.04 | cassio | there is no firewall |
| 01:12.18 | robin_sz | then I am as lost as you are |
| 01:13.42 | *** join/#asterisk mihaela (n=mihaelam@83-131-23-16.adsl.net.t-com.hr) |
| 01:13.49 | psi_force | file: ok so how do I enforce channel names to SIP/yyyyyy-xxxx (where yyyyyy is their username/extension number) |
| 01:14.28 | wunderkin | y will be the peer name |
| 01:15.26 | psi_force | file: it seems that if someone has connected as a sip trunk, the client asterisk box decides the channel name which is not what I want |
| 01:16.01 | *** join/#asterisk Garak_ (n=garak@209.5.171.170) |
| 01:16.02 | robin_sz | HMM WHATS THE "BEST" FORMAT TO rECORD(FILENAME) IN?? |
| 01:16.06 | robin_sz | GSM? |
| 01:16.29 | *** join/#asterisk KryoStoffer (n=kri@helium.kri.dk) |
| 01:16.34 | file[laptop] | psi_force: the channel name is not meant for actual use because the channel driver makes it, it's not an absolute that it's always going to look how you want - various things can change it |
| 01:16.52 | file[laptop] | so if you're depending on it to always look a specific way, good luck |
| 01:16.56 | Garak_ | Are there anyproducts out there that will automaticly switch over POTS lines to a single CO provided line in a power failour(to maintain 911 service) |
| 01:17.38 | psi_force | <file> so whats the best way to allocate billing then, src (that can also change) |
| 01:17.44 | psi_force | opps |
| 01:17.50 | psi_force | file: so whats the best way to allocate billing then, src (that can also change) |
| 01:17.57 | file[laptop] | use accountcode, or cdr variables |
| 01:20.09 | litage | does asterisk load into ram all of the settings read from /etc/asterisk/ ? |
| 01:20.27 | *** join/#asterisk a1fa||64 (n=a1fa@24.144.49.62) |
| 01:21.14 | psi_force | litage: yes |
| 01:21.22 | SwK | is it just me or was that like the lamest halftime show |
| 01:21.56 | psi_force | litage: you will need to run "/etc/init.d/asterisk reload" if you make changes to the config files |
| 01:22.16 | litage | psi_force: or just run ``reload'' within the aserisk console |
| 01:23.41 | psi_force | litage: yes, but be aware that sometimes reloading can play havoc with zaptel drivers in older versions of asterisk |
| 01:24.29 | psi_force | file[laptop]: thanks btw |
| 01:24.45 | *** join/#asterisk ravsi (n=ravsi@pool-71-108-178-182.lsanca.dsl-w.verizon.net) |
| 01:24.56 | litage | psi_force: since asterisk stores in ram all of /etc/asterisk/ , if you have like 5000 tenants, will you need to split those tenants up across multiple asterisk servers, or can they all be located on a single * box? |
| 01:27.29 | Snake-Eyes | if you direct CDRs to be sent to the Manager interface, how do you grab each CDR as it comes into the Manger interface? |
| 01:27.45 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
| 01:28.29 | *** join/#asterisk dave-outlaw (n=dave@fyshwick.fwpclient.officelink.net.au) |
| 01:29.35 | dave-outlaw | Hey guys. After further invesigation, it appears that when I log into a phone, it just reports it as busy. If I enable call waiting, I can then ring it from other extensions, though. Is this a known bug, or something I'm doing wrong? :) |
| 01:30.12 | robin_sz | OK, so ... I have a "recpetion" context with the usual menu "for sales press one .. " etc .... works fine from external |
| 01:30.17 | psi_force | litage: it all depends on how powerful for server (cpu) and your call occupacy |
| 01:30.28 | dave-outlaw | investigation, even |
| 01:30.52 | dave-outlaw | It's important that I get adhoc mode to work, too :/ |
| 01:30.55 | robin_sz | now .. how can I arrange to test it from inside .. ie arrange a weird extnsion number I dont use that puts a call into recpetion context? |
| 01:33.37 | Darwin35 | cool festival 1.96 built on fbsd |
| 01:35.25 | ravsi | anyone recommend a good cheap fxo/fxs device? |
| 01:35.57 | ravsi | I am trying to build a voicemail |
| 01:37.22 | newl | What's the phrase? You can have good, you can have cheap, but you cannot have both. :) |
| 01:37.57 | trixter | that depends if it fell of a truck or not :P |
| 01:38.14 | trixter | good, cheap, legal, pick 2 |
| 01:38.15 | trixter | :P |
| 01:38.39 | ravsi | the first 2 :) |
| 01:38.39 | tronix | SwK: didn't bother watching halftime. Janet was missed, though. ;) (that'd probably have had spiced things up a bit...) |
| 01:39.01 | SwK | tronix, it sucked |
| 01:39.06 | ravsi | it did |
| 01:39.10 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 01:39.13 | ravsi | badly |
| 01:39.28 | SwK | and trust me... unless you wanted to see keith richards naked you REALLY REALLY REALLY didnt want them to have another wardrome malfunction |
| 01:39.29 | websae | anyone here familiar with faxing in asterisk? does it work? |
| 01:39.40 | tronix | SwK: HAHAHA that's a very good point! |
| 01:41.49 | ravsi | a fxs port cannot behave as a fxo correct? |
| 01:41.56 | Katty | mew. |
| 01:42.03 | websae | anyone familiar with asterisk and faxign? |
| 01:42.26 | glm2k | coppice isn't around websae ... he would be the one to ask |
| 01:42.40 | websae | thank you |
| 01:42.48 | glm2k | you're welcome |
| 01:43.50 | SwK | hey katty |
| 01:43.59 | Katty | hiya. |
| 01:44.37 | trixter | hi |
| 01:47.02 | psi_force | later all |
| 01:47.08 | *** join/#asterisk KryoStoffer (n=kri@helium.kri.dk) |
| 01:48.02 | robin_sz | websae: I use an Eicon Divas server card with asterisk for fax |
| 01:48.10 | robin_sz | for fax rx anywya |
| 01:48.32 | *** join/#asterisk jarnaud (n=jarnaud@c-67-191-4-38.hsd1.fl.comcast.net) |
| 01:48.32 | robin_sz | works fine, uses an AGI script to email the fax to someone |
| 01:52.44 | websae | Eicon Divas server card..hrm |
| 01:52.52 | websae | what's that card do? |
| 01:56.38 | Qwell | ravsi: no |
| 01:56.49 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:01.57 | dave-outlaw | Thanks anyway guys. I might try asking a little later |
| 02:01.59 | dave-outlaw | ]/quit |
| 02:03.12 | rene- | websae: i think it it is a card with fax dsp and fax ports |
| 02:06.09 | tronix | does anyone have a particular favorite softphone for windows, for either sip or iax2? somebody elsewhere was asking, and I wasn't sure what was popular with Windows |
| 02:06.26 | tronix | I suggested SNOM, xten, and idefisk |
| 02:06.39 | WasPhantom | I use xten here, works wel |
| 02:08.05 | ariel_ | I use xten and idefisk here both work well |
| 02:08.49 | tronix | cool, thanks. sounds like I wasn't too far off the mark with my suggestion, then. ;) |
| 02:09.35 | *** join/#asterisk annonimous (n=annonimo@201.152.124.242) |
| 02:09.48 | annonimous | hello |
| 02:13.17 | *** part/#asterisk calculus (n=SPAM@24-176-224-55.dhcp.snlo.ca.charter.com) |
| 02:17.45 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 02:18.47 | De_Mon | idefisk? does that do iax? |
| 02:18.51 | ariel_ | yes |
| 02:18.56 | De_Mon | I want an IAX2 phone with video |
| 02:18.58 | ariel_ | seems like a slow night tonight |
| 02:19.09 | justinu | stupid bowl is on |
| 02:19.14 | Qwell | justinu: we're geeks |
| 02:19.23 | Qwell | our kind don't watch sports |
| 02:19.28 | justinu | i like motorsport |
| 02:19.33 | mzo | iour sports are battleing robots |
| 02:19.34 | ariel_ | hay can't you do two things watch and be on a laptop at the same time.. |
| 02:19.36 | De_Mon | justinu beauty and the geek is on too IIRC |
| 02:19.36 | mzo | and cyberbowl 2064 |
| 02:19.37 | *** part/#asterisk jarnaud (n=jarnaud@c-67-191-4-38.hsd1.fl.comcast.net) |
| 02:19.51 | De_Mon | <-- watching ESPN ice skating |
| 02:20.03 | ariel_ | well game is good 10/14 |
| 02:20.08 | WasPhantom | I'm a bit pissed really, the F1 won't be free to air in New Zealand |
| 02:20.10 | WasPhantom | sky only |
| 02:20.14 | *** join/#asterisk rehan_Linuxer (n=rehan_Li@p54A7FA97.dip.t-dialin.net) |
| 02:20.20 | justinu | WasPhantom: that sucks :( |
| 02:20.32 | justinu | it's not free here either |
| 02:20.38 | De_Mon | racing is worse than golf |
| 02:20.40 | justinu | you need speed channel, which is only on pay cable or satellite |
| 02:20.45 | WasPhantom | ahh okies.... first session I've had to look at paying for it |
| 02:20.54 | Qwell | justinu: and torrent networks :p |
| 02:20.58 | justinu | yeah, that too |
| 02:21.04 | justinu | f1 is not popular here in the states |
| 02:21.07 | WasPhantom | ahh, I never thought of downloading the torrent.... |
| 02:21.12 | Katty | beep. |
| 02:21.14 | De_Mon | are there many wrecks in f1? |
| 02:21.16 | justinu | most people watch cars go around in circles at NASCAR races |
| 02:21.24 | justinu | yeah, there's always accidents |
| 02:21.31 | ariel_ | what is a f1 |
| 02:21.37 | justinu | formula 1 |
| 02:21.44 | ariel_ | never mind don't watch car racing anyway |
| 02:21.59 | justinu | i'm a gearhead as well as a geek |
| 02:22.08 | WasPhantom | likewise :-) |
| 02:22.18 | justinu | my dad just bought a pontiac GTO (holdon |
| 02:22.19 | justinu | ) |
| 02:22.21 | justinu | holden |
| 02:22.30 | justinu | i was just driving it |
| 02:22.33 | justinu | very fast :) |
| 02:22.43 | justinu | i fly |
| 02:22.47 | De_Mon | theres some longhair'ed football player in the superbowl, hair down to his knees or waist, super long |
| 02:22.50 | WasPhantom | too many holdens here |
| 02:22.50 | WasPhantom | heh |
| 02:23.00 | justinu | this car has 400horsepower |
| 02:23.06 | justinu | impressive |
| 02:23.26 | De_Mon | I wish there was more arial racing or acrobatics on tv |
| 02:23.38 | ariel_ | there was a wings channel |
| 02:23.40 | justinu | you guys hear the news about that rocket racing league? |
| 02:23.43 | *** join/#asterisk Alric (n=nbowyer@ppp-db.1stel.com) |
| 02:23.47 | De_Mon | those are acidents you don't walk away from |
| 02:23.52 | ariel_ | but got pulled for the war channel now |
| 02:23.53 | WasPhantom | justinu, hell yeah, I want to see that |
| 02:24.00 | justinu | that's my next hobby |
| 02:24.01 | justinu | ;) |
| 02:24.02 | mzo | but you leave good smears in the rocket league |
| 02:24.11 | De_Mon | mzo I bet! |
| 02:24.16 | tronix | heh my "other car" has a 85 hp Continental engine and no muffler. *cough* |
| 02:24.22 | justinu | heh |
| 02:24.26 | justinu | piper cub? |
| 02:24.41 | tronix | actually, 152. couldn't remember exact specs offhand |
| 02:24.53 | justinu | ah, Lycoming O160 iirc |
| 02:24.56 | tronix | ahh yes |
| 02:25.02 | justinu | 120hp, iirc |
| 02:25.13 | justinu | i got my private ticket in a 152 |
| 02:25.16 | justinu | first solo :) |
| 02:25.19 | justinu | in a 152 |
| 02:25.24 | ariel_ | I got mine in a 172 |
| 02:25.31 | ariel_ | 152 is just too small |
| 02:25.35 | justinu | yep, toy airplane |
| 02:25.38 | WasPhantom | actually, speaking of F1, I might have to buy F1 Grand Prix for the PSP |
| 02:25.40 | justinu | but it was the cheapest way to get the ticket |
| 02:25.45 | tronix | solo'd in a 152. it's good for hours-building during training, but post-training, so out of the 152. |
| 02:25.47 | De_Mon | putting gass in my car is bad enough :( |
| 02:25.55 | justinu | right after the 152, i went to a twin piper seminole |
| 02:26.04 | ariel_ | now they have a new cheap ticket. Sports Pilot |
| 02:26.06 | tronix | too bad I can't fly a proper 18-wheeler (747-400) |
| 02:26.10 | ariel_ | no medical needed |
| 02:26.14 | justinu | you probably could |
| 02:26.17 | ariel_ | 20 hours is all you need |
| 02:26.28 | justinu | i flew the A320 sim at united training center in denver |
| 02:26.33 | justinu | piece of cake |
| 02:26.40 | justinu | V1 cuts are easy, engine out lands, easy |
| 02:26.42 | justinu | everythign is easy |
| 02:26.48 | tronix | justinu: sweet. been pining about doing a sim session sometime... just need to get around to making arrangements |
| 02:26.58 | WasPhantom | justinu, should I be worried that you're doing some training in large aircraft with no intention of piloting them?? ;-) |
| 02:27.09 | justinu | i'm already a licensed pilot :P |
| 02:27.27 | Qwell | justinu: Fly me to VON :p |
| 02:27.35 | WasPhantom | phew |
| 02:27.42 | ariel_ | Largest plane I have flown is the C-130E |
| 02:27.46 | justinu | that's big |
| 02:27.54 | justinu | 4 engine too |
| 02:27.58 | justinu | nice |
| 02:28.14 | tronix | De_Mon: 100LL aviation gas is about USD $3.50/gallon here iirc, and the Cessna 152 only chews 6 gal/hour... so about $20/hr in fuel costs alone |
| 02:28.16 | rene- | has anyone used the SAS and MOH options for the sipura products with asterisk? |
| 02:28.18 | De_Mon | justinu have you ever used a flight simulator to crash into a building? |
| 02:28.26 | justinu | no :P |
| 02:28.30 | tronix | hahaha |
| 02:28.32 | justinu | i did fly down the vegas strip in the a320 tho |
| 02:28.50 | justinu | and under the golden gate :) |
| 02:29.05 | rene- | <cliche>did you got any virgins?</cliche> |
| 02:29.24 | wunderkin | did you crash into microsoft? |
| 02:29.34 | De_Mon | *cough* |
| 02:29.37 | justinu | a virgin daqauri (8 hour bottle to throttle rule) |
| 02:29.50 | wunderkin | oops the fbi is watching |
| 02:31.32 | websae | anyone here using the sipura 841? |
| 02:32.07 | rene- | im playing with several sipura atas but no 841s |
| 02:34.34 | websae | how do you like the atas? |
| 02:34.42 | websae | i have one running |
| 02:35.30 | rene- | i like them, they sound good and they are cheap. but i like better ip phones because of less cabling |
| 02:35.58 | websae | how do you hook a ata adapter into a network interface box? |
| 02:37.43 | tuxinator_linux | justinu, bottle to throttle, must be a pilot. |
| 02:40.45 | *** join/#asterisk deidre (i=uyiii@stevanus.centrin.net.id) |
| 02:42.21 | deidre | hi, is it possible to use intel 536EP as X100P clone? I noticed that intel 536EP use MD1724 chipset. So far I get no luck use it with asterisk :( |
| 02:42.35 | Qwell | a generic generic, eh? |
| 02:42.45 | rene- | i believe that the right one should be 537ep |
| 02:42.48 | tuxinator_linux | I thing most close will only connect to telco, not telephones |
| 02:42.56 | tuxinator_linux | s/thing/think |
| 02:43.11 | tuxinator_linux | s/close/clones |
| 02:43.35 | tuxinator_linux | Qwell would know, he knows everything |
| 02:43.41 | deidre | yup, I read from the wiki that the 'real' x100p clone is intel 537ep.. |
| 02:44.19 | tuxinator_linux | why not just buy a digium card? |
| 02:44.27 | tuxinator_linux | I did |
| 02:44.33 | deidre | but since this intel 536ep is all I got right now, maybe there are possibilities changing it into x100p clone too :P |
| 02:44.34 | rene- | deidre: i havent heard that the 536 is compatible |
| 02:44.35 | *** join/#asterisk boa2 (n=jkr888@c-69-248-26-95.hsd1.nj.comcast.net) |
| 02:44.51 | rene- | get a 537 is the cheapest option |
| 02:46.13 | *** join/#asterisk Kab0b (n=asdf@S0106001195772f19.ed.shawcable.net) |
| 02:46.14 | brookshire | http://jared.degraffenried.net/asteriskoids/ |
| 02:46.32 | brookshire | lol. |
| 02:47.16 | deidre | oh thanks rene for the information. Maybe you're right. Maybe I should dump this 536EP to the trash can and get some intel 537ep ;) |
| 02:47.17 | rene- | to go extra sure buy it from a vendor that claims compatibility with *, it will cost you twice or more than getting the clones from someone who doesnt know that this can be used as very cheap pbx parts but its still a lot less expensive than the current entry level asterisk hardware |
| 02:48.24 | *** join/#asterisk relyuhcs (n=Bosco@c-69-247-188-78.hsd1.al.comcast.net) |
| 02:49.04 | *** join/#asterisk MYing (n=Ming@mying.enta.net) |
| 02:52.09 | *** join/#asterisk chaos1 (i=chaos1@klover226.bitnet.nu) |
| 02:53.01 | chaos1 | I am looking for some tutorial,docs on asterisk 1.2.4 |
| 02:53.04 | chaos1 | please suggest |
| 02:53.15 | chaos1 | i want to know what has changed and in detail |
| 02:53.31 | wunderkin | read the changelog |
| 02:53.50 | *** join/#asterisk CaT[tm] (n=cat@CPE-144-136-105-206.nsw.bigpond.net.au) |
| 02:53.59 | CaT[tm] | what's the netsec version of asterisk for? |
| 02:54.10 | brookshire | <PROTECTED> |
| 02:54.53 | brookshire | cat: the netsec version of asterisk works with a firewall product from ranch networks that can transparently open or close sip ports |
| 02:56.05 | CaT[tm] | brook: ahhh. thanks. a README on the download site to that affect would be just lovely, btw :) |
| 02:56.20 | Qwell | Like the README.netsec? :p |
| 02:56.32 | file | brookshire: MattyBrooks!!! |
| 02:56.39 | CaT[tm] | http://ftp1.digium.com/pub/asterisk/ |
| 02:56.53 | wunderkin | brooky wooky |
| 02:57.53 | chaos1 | I need some assitance with setting up a priority queue for remote agents on asterisk |
| 02:58.16 | chaos1 | i know asterisk can do that as seen from its feature list |
| 02:59.51 | chaos1 | whata is a-number profile btw |
| 03:00.51 | chaos1 | :( |
| 03:02.28 | chaos1 | inte bra |
| 03:03.00 | wunderkin | well describe what you want a little more |
| 03:04.52 | chaos1 | thanks wunderkin, here goes it |
| 03:05.19 | tuxinator_linux | Superbowl is over |
| 03:05.57 | chaos1 | I have users registered on other places |
| 03:06.08 | brookshire | who won? |
| 03:06.12 | tuxinator_linux | yellow |
| 03:06.14 | Qwell | me |
| 03:06.26 | tuxinator_linux | steelers |
| 03:06.29 | justinu | bud light |
| 03:06.44 | tuxinator_linux | really stupid godaddy.com ads |
| 03:06.48 | tuxinator_linux | really hate that company |
| 03:07.03 | chaos1 | the incoming calls on asterisk need to be queues based on the incoming number (the agents are remote as there are registered on some sip proxy) |
| 03:07.49 | chaos1 | read queues as queued |
| 03:08.10 | *** join/#asterisk rehan_Linuxer (n=rehan_Li@p54A7CE87.dip.t-dialin.net) |
| 03:08.24 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 03:08.26 | chaos1 | the sip users or iax clients are the callers |
| 03:09.22 | chaos1 | how will asterisk check whether the remote agent is busy or not before forwarding a call to it from queue |
| 03:09.29 | chaos1 | i have been thinking a lot over it |
| 03:09.48 | justinu | FUCK this cable modem |
| 03:09.48 | chaos1 | because for some calls, it has to ring one remote agent |
| 03:10.14 | chaos1 | while for some other calls it has to ring more than one remote agents simultaneousy |
| 03:10.37 | chaos1 | can i make group of remote agents on asterisk? and then just forward call to that group |
| 03:11.00 | chaos1 | these remote agents are registered on sip proxy. they are sip users or iax clients |
| 03:11.55 | *** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 03:13.38 | *** join/#asterisk rt (n=markv@c-67-180-32-90.hsd1.ca.comcast.net) |
| 03:14.02 | chaos1 | any thoughts on my problem |
| 03:14.41 | chaos1 | okay an easy question now, |
| 03:14.59 | chaos1 | would http://www.digium.com/handbook-draft.pdf version 2 document, be valid for asterisk 1.2 as well or is there some major change |
| 03:16.26 | wunderkin | so you are saying that the caller id determines what queue the person is put into |
| 03:17.01 | wunderkin | hopefully you can take care of that part |
| 03:18.04 | wunderkin | the other question, use agentcallbacklogin, it will know when someone is on a call from the queue, but if they make outgoing calls you need to use pause/unpausequeuemember |
| 03:18.57 | wunderkin | read up on queues, this is all basic stuff |
| 03:19.08 | wunderkin | as far as the handbook im sure it is out of date, probably based on 1.0 |
| 03:21.05 | chaos1 | wunderkin:you rock big time! |
| 03:22.25 | chaos1 | what would you recommend for reading up on asterisk 1.2 |
| 03:22.35 | chaos1 | i want to understand how things are done on that |
| 03:23.03 | chaos1 | wunderkin:please suggest some documentation/tutorial beside its /doc |
| 03:26.43 | *** join/#asterisk sack (n=sack@96.Red-83-50-158.dynamicIP.rima-tde.net) |
| 03:27.39 | wunderkin | ~docs |
| 03:27.41 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 03:27.42 | wunderkin | ~thebook |
| 03:27.44 | jbot | thebook is, like, Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 03:35.01 | chaos1 | anyone here tried radius with asterisk |
| 03:35.05 | chaos1 | that portaone patch |
| 03:35.08 | chaos1 | ;) |
| 03:51.21 | *** join/#asterisk silly (n=silly@cpe-24-174-162-34.satx.res.rr.com) |
| 03:51.36 | *** join/#asterisk silly_ (n=silly@cpe-24-174-162-34.satx.res.rr.com) |
| 03:51.52 | *** join/#asterisk silly (n=silly@cpe-24-174-162-34.satx.res.rr.com) |
| 03:59.10 | CaT[tm] | bleh. broken installs cript: if (!file_exists(AMP_CONF)) { out(AMP_CONF." does not exist, copying default"); copy("amportal.conf", "/etc/amportal.conf"); ... }' |
| 04:08.43 | rene- | have people tried feeding the sample configuration for the sipura 841 phone available in the sipura support site to devices like spa 2002 or the pap2? was it successful? |
| 04:09.46 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 04:10.05 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 04:10.14 | *** join/#asterisk sancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 04:10.30 | *** part/#asterisk rene- (i=rene@dsl-201-128-115-222.prod-infinitum.com.mx) |
| 04:10.35 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 04:10.37 | *** join/#asterisk rene- (i=rene@dsl-201-128-115-222.prod-infinitum.com.mx) |
| 04:12.48 | *** join/#asterisk miguel3239 (n=chatzill@dsl092-074-055.bos1.dsl.speakeasy.net) |
| 04:16.32 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 04:24.12 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:34.24 | *** join/#asterisk newl (n=newlook@203-59-122-96.dyn.iinet.net.au) |
| 04:42.59 | *** part/#asterisk boa2 (n=jkr888@c-69-248-26-95.hsd1.nj.comcast.net) |
| 04:51.07 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 04:52.13 | rene- | ~/o |
| 04:58.46 | *** join/#asterisk miguel3239 (n=chatzill@dsl092-074-055.bos1.dsl.speakeasy.net) |
| 05:00.06 | *** join/#asterisk FastJack_ (i=fastjack@p5091E7A5.dip.t-dialin.net) |
| 05:05.39 | *** join/#asterisk xachen (i=justin@magnum.thisgeek.com) |
| 05:08.41 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:08.45 | *** join/#asterisk daguerro (i=Co_Care@lss-67-28.ee.itb.ac.id) |
| 05:09.23 | daguerro | hello |
| 05:10.08 | *** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
| 05:11.13 | daguerro | Is there anyone that have been succesfully using asterisk realtime?? |
| 05:11.22 | Qwell | daguerro: yep |
| 05:12.45 | daguerro | i'm trying to connect my sip.conf qith mysql, but the asterisk wont read from the database. i've followed the instructions at voip-info.. can u help me?? |
| 05:13.03 | Qwell | pastebin any errors you get |
| 05:14.33 | daguerro | hold on a sec |
| 05:15.56 | daguerro | Feb 6 11:48:25 NOTICE[4085] chan_sip.c: Registration from 'MM <sip:12345@167.205.67.33>' failed for '167.205.67.28' - Username/auth name mismatch |
| 05:16.01 | mdave | ok.. tcpdump shows my spa sending packets to port 5060 of the * box, but even tho i have sip debug on im not seeing any sip debug output |
| 05:16.37 | mdave | so how do I force * to give some feedback as to wtf it is doing with the packets |
| 05:17.07 | mdave | hrm.. i tried 'sip debug yes really i mean it' |
| 05:17.10 | mdave | without any luck |
| 05:17.20 | mdave | i have debug 255 qnd verbose 255 |
| 05:17.23 | mdave | yet nothing |
| 05:17.50 | daguerro | what should i do qwell? |
| 05:17.52 | mdave | it was working before, i had lost the * console and killed it and restart it, then all of a sudden its gone wonky |
| 05:18.09 | Qwell | daguerro: wait for somebody who knows SIP errors better.. |
| 05:20.25 | mdave | hrm.. heres something interesting.. tcpdump does not show * sending any udp back to the spa.. |
| 05:20.31 | mdave | at least nothing to or from port 5060 |
| 05:20.39 | mdave | to anywhere, for that matter |
| 05:20.49 | *** join/#asterisk slan (n=lba@user-12lml5g.cable.mindspring.com) |
| 05:20.50 | mdave | all 5060 traffic is inbound, nothing going out |
| 05:21.32 | mdave | almost as though there is some 'sip suppress all response' set or something |
| 05:21.40 | mdave | (no, ipfw is not blocking it) |
| 05:27.55 | *** part/#asterisk help (i=Co_Care@lss-67-28.ee.itb.ac.id) |
| 05:28.03 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.254.80.204.Dial1.Chicago1.Level3.net) |
| 05:31.11 | mattwj2005 | ask not what you can do for Asterisk.....ask what Asterisk can do for you :P |
| 05:33.07 | mdave | at the moment id be happy if asterisk went back to doing what it was before i restarted it eralier today |
| 05:33.26 | mattwj2005 | what happened mdave? |
| 05:33.35 | mdave | but it seems to not want to, nor am I able to discern anything to explain why it isnt |
| 05:33.43 | mdave | well.. i had lost the console window for my * |
| 05:33.45 | mdave | so I killed it |
| 05:33.47 | mdave | and restarted it |
| 05:33.51 | mdave | now, my spa wont register |
| 05:33.55 | mdave | sip debug shows nothing |
| 05:33.58 | mdave | well, usuually |
| 05:34.06 | mdave | every now and then i get a few messages |
| 05:34.19 | mdave | but i cant seem to force it to speak sip |
| 05:34.31 | mdave | eg i reboot the spa, it claims to be trying to register, yet nothing on sip debug |
| 05:34.37 | mdave | tcpdump shows udp traffic on port 5060 |
| 05:34.41 | mdave | but no output from * |
| 05:34.56 | mdave | im not sure wether its successfully registering with bv, either |
| 05:35.23 | mattwj2005 | bv? spa? |
| 05:35.26 | mdave | the one time I did see a sip debug mesage regarding the spa, there was a 401/unauthorized in one of them |
| 05:35.33 | mattwj2005 | I am a noob....what I can I say |
| 05:35.39 | mdave | but ive double and tripled chcked the passwords, they are right |
| 05:35.46 | mattwj2005 | but maybe I can be some help anyways |
| 05:35.47 | mdave | spa = Sipura spa-2000 |
| 05:35.54 | mdave | bv = broadvoice voip provider |
| 05:35.56 | *** join/#asterisk sancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 05:36.08 | mattwj2005 | oh okay |
| 05:37.47 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 05:37.48 | mattwj2005 | how did you configure the spa? |
| 05:37.59 | mattwj2005 | config files right? |
| 05:38.58 | mattwj2005 | or did you use the cli? |
| 05:39.53 | mdave | the spa was a web interface |
| 05:39.57 | *** part/#asterisk Alric (n=nbowyer@ppp-db.1stel.com) |
| 05:39.59 | mdave | s/was/has |
| 05:40.49 | mattwj2005 | okay what happened to the console window for asterisk? |
| 05:40.57 | mattwj2005 | just crashed? |
| 05:41.31 | mdave | actually, i think the cablemodem got a reboot, so my workstation lost connectivity |
| 05:41.39 | mdave | but i dont think thats relevant |
| 05:41.44 | mattwj2005 | oh okay |
| 05:41.58 | mattwj2005 | how did you restart asterisk? |
| 05:42.03 | mdave | the thing is I started asterisk fresh, now it isnt letting the spa register |
| 05:42.09 | mdave | the same way i had before |
| 05:43.01 | mdave | ok theres some sip debugs |
| 05:43.04 | mdave | 401/unauthorized |
| 05:43.05 | mdave | sigh |
| 05:43.20 | mattwj2005 | hmmm |
| 05:43.26 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 05:44.24 | litage | if you direct CDRs to be sent to the Manager interface, how do you grab each CDR as it comes in? |
| 05:45.15 | mattwj2005 | I am not sure mdave |
| 05:45.23 | mattwj2005 | anyone else have an idea? |
| 05:47.00 | rene- | mdave: i would got the spa factory reset it and start over |
| 05:47.27 | rene- | and it sounds like your asterisk installation was somehow upgraded |
| 05:48.06 | mdave | unless it upgrades itself spontaneously i dont see how that would be possible |
| 05:48.09 | Corydon76-home | litage: you'd write an interface to do so? |
| 05:48.26 | mdave | i had downloaded and installed v 1.2.3 |
| 05:48.37 | mdave | and afaik thats the most current |
| 05:48.56 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 05:49.06 | mattwj2005 | 1.2.4 is currently out |
| 05:49.13 | mdave | well the running version is still 1.2.3 |
| 05:49.27 | mdave | which is the version i installed |
| 05:49.32 | mattwj2005 | did 1.2.3 have a memory leak like 1.2.2? anyone know? |
| 05:49.34 | mdave | i am the only one with root to the box |
| 05:49.36 | Qwell | mattwj2005: yes |
| 05:49.59 | mdave | so unless * did it on its own, it didnt happen |
| 05:50.19 | litage | Corydon76-home: where can i find documentation on writing such an interface? there' no mention of playing with CDRs in the manager interface on voip-info, and i've trolled google to no avail |
| 05:50.26 | mattwj2005 | okay....I wasn't sure if that notice in the topic was from 1.2.2 or 1.2.3 or both |
| 05:50.44 | rene- | you have 1.2.3 and today is topic is about upgrading |
| 05:50.52 | rene- | s/i/'/ |
| 05:50.56 | Corydon76-home | litage: why don't you telnet to the interface, login, and see what comes across? |
| 05:51.07 | Corydon76-home | It IS only text, afterall |
| 05:51.22 | mdave | right now what id like to do is get the version I have working the way it was yesterday |
| 05:51.29 | mdave | then maybe think about upgrading |
| 05:51.58 | mdave | why would * say 401 unauthorized to my spa? the password in the spa matches the one in sip.conf |
| 05:52.01 | litage | Corydon76-home: you mean if you configure * to send CDRs to the manager interface, they're just written to the manager rather than being extracted via some API function call? |
| 05:52.04 | Corydon76-home | Hmmm, let's think about that... memory leak in 1.2.3, fixed in 1.2.4... hmmmm |
| 05:52.04 | mdave | ive double and triple checked that |
| 05:52.20 | Corydon76-home | litage: Bingo |
| 05:53.32 | Corydon76-home | mdave: now why would you want to run a version with a big fat memory leak? |
| 05:53.45 | mdave | so.. any idea why a previously working * setup would start saying 401 unauthorized to sip registrations? |
| 05:53.59 | mdave | i want to fix the problem i have right now, then i'll worry about the upgrade |
| 05:54.07 | rene- | memory is cheap nowadays |
| 05:54.16 | mattwj2005 | what happens with the memory leak anyones? system crash, application crash, or what happens? |
| 05:54.21 | mdave | unless * checks to see if there is a new version and refuses to run properly if you dont upgrade |
| 05:54.21 | mattwj2005 | *anyways |
| 05:54.26 | mdave | which would be just crap |
| 05:54.30 | Corydon76-home | mdave: the clients changed the password in their configs? |
| 05:54.36 | mdave | the client is me |
| 05:54.37 | mdave | my spa |
| 05:54.40 | mdave | connecting to my * |
| 05:54.43 | mdave | the password matches |
| 05:54.50 | mdave | ive checked like 6 times |
| 05:54.57 | Corydon76-home | Has your SPA upgraded itself? |
| 05:55.00 | mdave | and it matched before * was restarted |
| 05:55.03 | a1fa||64 | mattwj2005: depends how it leaks.. buffer over-runs.. someone can root your box if you are runing * as root |
| 05:55.19 | mattwj2005 | okay thanks |
| 05:55.32 | a1fa||64 | application may force other apps to run unstable, etc.. |
| 05:55.39 | mdave | Corydon-w, no |
| 05:55.43 | Corydon76-home | Like, say, upgraded to try to start using Vonage's servers? |
| 05:55.45 | a1fa||64 | depending where it leaks.. (what memory address,e tc) |
| 05:56.06 | mdave | lol |
| 05:56.07 | mdave | no |
| 05:56.21 | litage | Corydon76-home: thanks for that bit of info :) |
| 05:56.38 | Corydon76-home | mattwj2005: no, the memory leak is limited to just crashing |
| 05:56.44 | mdave | the spa is unlocked, and not under control of any voip provider |
| 05:56.47 | mdave | its under my control |
| 05:57.05 | mdave | and its trying to register with my * box.. i can see the packets on 5060 with tcpdump |
| 05:57.11 | Corydon76-home | mdave: have you run a SIP debug? |
| 05:57.18 | mdave | sip debug is on |
| 05:57.26 | mdave | for most of the time, nothing is output |
| 05:57.33 | mdave | once in a while, it blurts a spurry of messages |
| 05:57.40 | mdave | and the only thing notable is the 401 unauthorized |
| 05:57.44 | Corydon76-home | Well, pastebin the messages |
| 05:57.45 | Corydon76-home | ~pb |
| 05:57.50 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 05:57.50 | *** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4163113.sympatico.ca) |
| 05:58.07 | mdave | i cant reliably get them tho.. |
| 05:58.19 | mdave | i can restart *, reboot the spa, and wait and wait and no sip debug |
| 05:58.24 | *** part/#asterisk jbroome (n=jbroome@63-168-10-93.celito.net) |
| 05:58.27 | mdave | its like its saving them all up to spit them out when i least expect it |
| 05:58.33 | mdave | instead of outputting them in realitme |
| 05:58.41 | litage | does SIP split the signalling and RTP streams? |
| 05:58.41 | Corydon76-home | That's because after you restart, sip debugging turns off |
| 05:58.49 | mdave | no, even after I turn sip debug on |
| 05:58.52 | mdave | and i reboot the spa |
| 05:58.58 | mdave | and it shows registration failed |
| 05:59.01 | mdave | next retry in |
| 05:59.02 | Corydon76-home | Then your SPA is not re-registering |
| 05:59.03 | mdave | and it retires |
| 05:59.10 | mdave | and i see udp packets on 5060 |
| 05:59.13 | mdave | coming in, over and over |
| 05:59.24 | Corydon76-home | The Asterisk console outputs SIP debug in realtime |
| 05:59.29 | mdave | and all the meanwhile (we are talking several minutes) nothing output from * sip debug |
| 05:59.50 | mdave | i get a whole screenful of tcpdump output showing packets from the spa to * on 5060 |
| 05:59.56 | mdave | and still nothing from * |
| 06:00.11 | Corydon76-home | I need to see this to believe it |
| 06:00.37 | mdave | then, usually right before im ready to stop and restart it ahain, i get a few sip debug messages |
| 06:00.39 | Corydon76-home | You're not running something else on that box, aren you? |
| 06:00.45 | mdave | nothing of any note |
| 06:00.50 | mdave | and nothing it wasnt running before |
| 06:00.54 | mdave | when * was working just fine |
| 06:00.55 | *** join/#asterisk daguerro (i=Co_Care@lss-67-28.ee.ITB.ac.id) |
| 06:01.01 | Corydon76-home | Something that might be listening to port 5060? |
| 06:01.05 | mdave | no |
| 06:01.12 | mdave | lsof -i confirms * is listening to 5060 |
| 06:01.21 | Corydon76-home | Is anything else? |
| 06:01.23 | mdave | or claims to be, anyway, sigh |
| 06:01.39 | Corydon76-home | Give me remote root into your box |
| 06:01.50 | mdave | no. nothing else is listening to 5060 |
| 06:01.57 | mdave | theres no other telephony apps running |
| 06:02.33 | mdave | Corydon-w, give me your credit card and bank account numbers and pin |
| 06:02.35 | mdave | :P |
| 06:02.50 | mdave | i appreciate you want to help, but i cant go there |
| 06:02.56 | Corydon76-home | You're the one looking for help |
| 06:03.12 | Corydon76-home | Digium asks for the same privileges if you call for tech support |
| 06:03.13 | mattwj2005 | sorry cory.....I am with dave on this one |
| 06:03.27 | mdave | and theyd get the same answer |
| 06:03.50 | Corydon76-home | in which case you'd get a nice "Sorry, but without that access, we can't help you." |
| 06:03.54 | mdave | I *might* consider a dual-screen with me monitoring.. |
| 06:04.07 | mdave | if I had something like that setup, which I dont at the moment |
| 06:04.17 | mdave | then their tech support is useless.. and sure nothing id pay for |
| 06:04.26 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:05.03 | Corydon76-home | If you're not willing to give access to your computer, then don't complain about it not working |
| 06:05.11 | mdave | tech support to me doesnt mean 'just let them come in and fix it'.. it means they tell *me* how to fix it.. if I was paying, that is |
| 06:05.16 | mdave | im not complaining |
| 06:05.23 | mdave | just seekeing help |
| 06:05.31 | mdave | but not help in the 'here let me just do it for you' variety |
| 06:06.03 | Corydon76-home | It's not that. You're describing behavior I've never seen before |
| 06:06.27 | mdave | in any case, anyone that would just give out their root password to a vendor to come in and play with it deserves to get hacked |
| 06:06.35 | Corydon76-home | Could be something rather simple, and I'd be happy to point it out, if I had access |
| 06:06.44 | mogorman | ooh starting to? |
| 06:06.57 | CaT[tm] | yes. it's my first time. lucky I'm sitting. |
| 06:06.59 | mogorman | no mdave |
| 06:07.04 | mogorman | we ssh into peoples boxes all the time |
| 06:07.08 | mogorman | you just make a temp pass |
| 06:07.14 | mogorman | and watch em in screen if your worried |
| 06:07.16 | mdave | if it was that simple, you could tell me what command or info to pull up and see the same thing |
| 06:07.20 | Math` | or you add his key into your authorized_keys |
| 06:07.22 | Corydon76-home | There's also a way for you to give me access without giving me root, if you like |
| 06:07.24 | mogorman | no its never that simple |
| 06:07.34 | mogorman | and if it was or you know better |
| 06:07.36 | Qwell | normal user + screen = <3 |
| 06:07.38 | mogorman | you arent calling support |
| 06:07.43 | mogorman | qwell is right |
| 06:07.54 | mattwj2005 | I am probably parod....but I know you can kill a machine in 9 characters |
| 06:08.02 | Qwell | You know, I've only ever given one person root on my box... |
| 06:08.04 | mdave | in any case, at this point since theres apparently an upgrade addressing a major problem, im probably going to save my extensions and sip conf, blow this all away, and reinstall from scratch |
| 06:08.10 | mogorman | and what can be solved over the phone can be solved over ssh in 1/100th time on average |
| 06:08.22 | mdave | i dont know or care what qwell has or is saying, i have him on ignore since hes an asshole |
| 06:08.27 | Corydon76-home | mdave: why would you do that? |
| 06:08.46 | Qwell | I'm not that bad, am I? |
| 06:08.50 | Corydon76-home | There's absolutely no need to wipe out an install to do an upgrade |
| 06:08.56 | mdave | Corydon-w, becuase I made it work from scratch once, and since i apparently need to upgrade anyway, i might as well |
| 06:09.11 | mdave | i'll just save the bits of config ive tweaked on |
| 06:09.16 | *** join/#asterisk johnrage (n=jabetong@82-167-4-43.odsplus.com) |
| 06:09.25 | Corydon76-home | mdave: are you running AMP? |
| 06:09.30 | Qwell | That's why astlinux is good...key disk |
| 06:09.41 | mdave | Corydon-w, yes, but if some bit of something setup wrong is whats causing the problem, clearing what i have out completely may help |
| 06:09.50 | mdave | Corydon-w, not even sure what it is |
| 06:10.07 | Corydon76-home | If you're running AMP, it's no wonder you're having problems |
| 06:10.19 | *** join/#asterisk eDitor (i=XTeAm@server.ivinskis.kursenai.lm.lt) |
| 06:10.21 | Corydon76-home | Everybody using AMP has huge problems |
| 06:10.24 | Qwell | ~amp |
| 06:10.27 | jbot | amp is, like, NOT supported here! people using it should join #amportal |
| 06:10.31 | mdave | wtf is amp? |
| 06:10.33 | Qwell | ;] |
| 06:10.40 | CaT[tm] | so AMP is fun is it? |
| 06:10.44 | Qwell | ~say test |
| 06:10.45 | jbot | test |
| 06:10.47 | mdave | i dont know what it is, how could i be running it? |
| 06:10.58 | Corydon76-home | Or Asterisk@Home ? |
| 06:10.59 | Qwell | ~say <Qwell> mdave: AMP is junk. Don't use it. |
| 06:11.01 | jbot | <Qwell> mdave: AMP is junk. Don't use it. |
| 06:11.04 | Qwell | I love that bot |
| 06:11.26 | mdave | Corydon-w, i have no idea what AMP is |
| 06:11.56 | *** join/#asterisk blkremedy (n=ur3rdeye@240M06.oasis.mediatti.net) |
| 06:12.12 | CaT[tm] | qwell: seriously, is it that bad? how can it break things? |
| 06:12.24 | Qwell | CaT[tm]: because it's very poorly written |
| 06:12.36 | CaT[tm] | yes. I can see that. :/ |
| 06:12.40 | Qwell | Corydon76-home: Wanna be my say bot? :P |
| 06:12.53 | Qwell | Maybe he'll end up ignoring everybody |
| 06:13.02 | Qwell | CaT[tm]: no, it's asstacular |
| 06:13.06 | mdave | Corydon-w, only the truly clueless would allow some random stranger off the net direct root access to his box.. or expect someone else to, for that matter |
| 06:13.42 | JamesDotCom | get over it man ;( |
| 06:13.44 | Qwell | CaT[tm]: and it's got a bunch of useless macros that everything goes through...it's just ugly |
| 06:13.47 | blkremedy | question....what would be the ideal size of HDD to use with asterisk@home? |
| 06:13.50 | CaT[tm] | qwell: yaay. anything I can read on its interactions with asterisk that I can show ppl who might want to use it? |
| 06:13.58 | Corydon76-home | Qwell: what's really craptacular is that he can't even seem to specify my nickname correctly |
| 06:14.07 | Qwell | Corydon76-home: well, it is two tabs :p |
| 06:14.10 | mattwj2005 | anyone want a Diet Dew? you know chill out |
| 06:14.12 | mattwj2005 | :P |
| 06:14.15 | mogorman | so my code was segfaulting 4 times in a row |
| 06:14.17 | mogorman | and now |
| 06:14.18 | mogorman | no problem |
| 06:14.22 | mogorman | no change to code |
| 06:14.26 | Qwell | mogorman: awesome |
| 06:14.32 | Corydon76-home | mogorman: yay, coredumps |
| 06:14.34 | Qwell | was it in your code? |
| 06:14.36 | Qwell | bkw__: ACK! |
| 06:14.43 | Corydon76-home | coredumps make me happy |
| 06:14.59 | mogorman | oog coredumps |
| 06:15.05 | mogorman | might be smart to turn on.... |
| 06:15.06 | Corydon76-home | Well, at least if the compile is with dont-optimize |
| 06:15.06 | Qwell | Corydon76-home: I bet. ;) |
| 06:15.26 | mogorman | yeah i have been doing all that |
| 06:16.24 | *** join/#asterisk Equinox (n=secret@pool-70-110-76-69.tampfl.fios.verizon.net) |
| 06:16.26 | litage | does SIP split the signalling and RTP streams? |
| 06:16.39 | Qwell | litage: RTP is separate from SIP. Entirely different RFCs |
| 06:16.40 | Math` | yes |
| 06:16.47 | Corydon76-home | litage: yes |
| 06:16.51 | Qwell | SIP uses RTP, but.. |
| 06:16.59 | Qwell | so do mgcp, skinny, etc |
| 06:17.05 | mattwj2005 | that is how that works |
| 06:17.16 | Corydon76-home | Pretty much everything except IAX |
| 06:17.26 | Qwell | h323? |
| 06:17.34 | Qwell | I know 0 about h323.. |
| 06:17.45 | litage | when someone says "SIP splits the signalling and voice streams", they mean that sip does the signalling, and uses rtp for carrying the voice data, and that sip doesn't bundle both into one stream? |
| 06:17.59 | Corydon76-home | litage: correct |
| 06:18.01 | Math` | exact |
| 06:18.03 | litage | nice |
| 06:18.22 | Math` | litage: SIP uses SDP to negotiate which codec to use and the location of the rtp stream |
| 06:18.39 | Corydon76-home | IIRC, SIP RTP uses a single stream for each kind of media |
| 06:18.49 | Qwell | each kind, like video vs audio? |
| 06:18.51 | Math` | correct |
| 06:18.52 | Qwell | vs dtmf |
| 06:18.52 | litage | does H.323 do the exact same thing? |
| 06:19.08 | Math` | Qwell: audio vs video, dtmf are in the same rtp as voice |
| 06:19.12 | Corydon76-home | I believe so, with the exception that H.323 control is over TCP |
| 06:19.23 | Math` | h323 can do udp too |
| 06:19.25 | litage | ah |
| 06:19.44 | Corydon76-home | Math`: for control? Or for media only? |
| 06:19.50 | Math` | both |
| 06:20.13 | Math` | that was with a gk tho |
| 06:20.31 | Corydon76-home | as opposed to a gw |
| 06:20.50 | Math` | asterisk (ooh323) was registering as gateway to gnugk |
| 06:20.53 | Math` | using udp |
| 06:21.05 | Math` | (iirc) |
| 06:22.09 | litage | if SIP uses RTP for its media stream, does each stream use its own port? |
| 06:22.20 | Math` | obviously |
| 06:22.35 | litage | just wanted to explicitly clear that up :) |
| 06:22.40 | Math` | or else it would have been the same stream :) |
| 06:23.08 | Math` | I'm wondering... how's video support on IAX? |
| 06:23.27 | litage | Math`: how is it then that you only poke a hole in your firewall for port 5060 to allow SIP through? how does the RTP stream get through? |
| 06:23.47 | Qwell | litage: some firewalls are smart enough, but with many, you need to open the rtp ports too |
| 06:24.00 | Math` | litage: rtp uses a specified range |
| 06:24.08 | litage | Qwell: how are some "smart enough"? |
| 06:24.08 | Math` | (in rtp.conf) |
| 06:24.12 | Corydon76-home | Your firewall basically has to know to read the control stream to open up a specified port and forward it to the right host |
| 06:24.20 | Math` | some recognize sip traffic |
| 06:24.22 | Qwell | litage: well, for instance, the ranch networks gear |
| 06:24.36 | Qwell | asterisk says "hey, open this up, and give it exactly this much bandwidth" |
| 06:24.41 | litage | Qwell: how about iptables? |
| 06:24.42 | Qwell | something like that anyhow |
| 06:24.50 | Math` | as long as your server is not firewalled/nat'ed, the NAT tables of the clients are going to be automaticly ajusted |
| 06:24.53 | Qwell | litage: There are probably conntrack's for SIP |
| 06:24.56 | Math` | the same way TCP is forwarded |
| 06:25.04 | Corydon76-home | Sometimes, though, they detect that there's a NAT and allow the inner host to establish the stream connection |
| 06:25.50 | Corydon76-home | Once a path is established, the packets may flow in both directions |
| 06:25.58 | Qwell | unless you've got a junk d-link. ;] |
| 06:26.47 | Corydon76-home | Even some of the top-end firewalls are junk when it comes to establishing two-way connections |
| 06:27.26 | Corydon76-home | *cough*SonicWall*cough* |
| 06:27.29 | techie | any familiar with cha_btp? |
| 06:27.32 | techie | chan |
| 06:28.28 | *** join/#asterisk hadi57 (i=al_moghr@62.3.44.62) |
| 06:29.19 | litage | i opened port 5060 on my workstation's firewall and allowed 5060 to pass through my router. why was i able to send and receive sip calls if i didn't open ports 10,000 to 20,000? |
| 06:29.53 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 06:30.07 | Corydon76-home | litage: Linux workstation? |
| 06:30.07 | Math` | litage: as we said, some firewalls are smart enough |
| 06:30.18 | GeneG | Does anyone know whether Asterisk 1.2 does a better job of native bridging IAX2 connections through a NAT? Doesn't seem to work with 1.0.10 (unable to transfer). |
| 06:30.33 | Corydon76-home | Do you have an iptables rule that contains ESTABLISHED,RELATED ? |
| 06:31.20 | litage | Corydon76-home: yes, using iptables |
| 06:31.24 | Corydon76-home | Those two keywords allow the firewall to let through SIP RTP sessions established by the control port |
| 06:31.36 | Corydon76-home | since the connections are related... |
| 06:32.34 | litage | Corydon76-home: yes |
| 06:33.04 | litage | ah, i see |
| 06:33.46 | Math` | Corydon-w: so iptables parses SIP to properly map RTP ? |
| 06:34.22 | Corydon76-home | Math`: I think so, yes. |
| 06:34.40 | Math` | I wonder is pf does so too |
| 06:34.43 | Math` | s/is/if/ |
| 06:35.16 | Corydon76-home | It may not fully parse the messages, but it may only detect the IP:port text portions internal to the protocol and open ports accordingly |
| 06:35.33 | Math` | that'd be nice |
| 06:36.01 | Math` | broadband routers should be using iptables |
| 06:36.51 | *** join/#asterisk mred (n=edm@ppp167-250-67.lns2.syd6.internode.on.net) |
| 06:37.19 | mred | Hey all |
| 06:37.39 | mred | Got a quick question is any body available? |
| 06:40.30 | glm2k | mred: just ask...if someone knows, it will be answered. |
| 06:40.41 | mred | ok thanks |
| 06:40.59 | litage | Corydon76-home: when you said that iptables allows the rtp stream through because of the ESTABLISHED,RELATED clause, were you referring to UPnP? |
| 06:41.34 | mred | I have a section in extensions.conf that looks like this: |
| 06:41.38 | Corydon76-home | Not necessarily |
| 06:41.38 | mred | [inbound-analog] |
| 06:41.39 | mred | exten => s,1,Answer |
| 06:41.39 | mred | exten => s,2,Goto(waitexent,1,1) |
| 06:41.39 | mred | ;exten => s3,1,Hangup |
| 06:41.39 | mred | [waitexent] |
| 06:41.39 | mred | exten => 1,1,Background(play-extension-message) |
| 06:41.41 | mred | exten => 1,2,WaitExten(15) |
| 06:41.43 | mred | exten => 1,400,Goto(local,400,1) |
| 06:41.45 | mred | exten => i,1,Playback(invalidextension) |
| 06:41.47 | mred | exten => i,2,Goto(waitextent,1,2) |
| 06:41.49 | litage | mred: use a pastebin |
| 06:41.49 | Corydon76-home | DON'T PASTE INTO THE CHANNEL |
| 06:41.49 | mred | exten => i,2,Hangup |
| 06:41.53 | Corydon76-home | USE PASTEBIN |
| 06:41.56 | Corydon76-home | ~pb |
| 06:41.57 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 06:41.58 | litage | mred: rafb.net/paset |
| 06:42.03 | litage | mred: rafb.net/paste |
| 06:42.20 | mred | Ah ok sorry |
| 06:42.25 | *** join/#asterisk rehan_Linuxer (n=rehan_Li@p54A7E7CD.dip.t-dialin.net) |
| 06:42.35 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 06:42.46 | Corydon76-home | litage: don't know if upnp is necessarily parsed by iptables |
| 06:42.46 | Math` | Corydon76-home: when using the iAX2 protocol, in a full frame, the "Frame type" and "Subclass" fields, are they the same value as AST_FRAME_* ? |
| 06:43.02 | Math` | iptables doesnt do upnp |
| 06:43.11 | Math` | there is a project called upnpd tho |
| 06:43.15 | mred | Could someone explain what I need to adjust to alow more than 1 number to be dialed for an extension? |
| 06:43.38 | Math` | define your extensions in that contex |
| 06:43.42 | Math` | context* |
| 06:43.55 | Corydon76-home | Dial(SIP/one&SIP/two) |
| 06:44.02 | CaT[tm] | AMP has drained all my energy. this bites. |
| 06:44.23 | Qwell | CaT[tm]: give it up |
| 06:44.33 | Qwell | vi is a far better config tool |
| 06:44.36 | Corydon76-home | Math`: the subclass is dependent upon what the Frame type is |
| 06:44.42 | CaT[tm] | for me, yeah. |
| 06:44.51 | CaT[tm] | but this is not necessarily for me :/ |
| 06:44.57 | Corydon76-home | but yes, the frame type is generally the same as AST_FRAME_* |
| 06:45.32 | Math` | ok |
| 06:46.46 | mred | Thanks Math. Got it |
| 06:47.18 | litage | on my router, i allowed port 5060 through, but i didn't allow any RTP through.somehow i was able to send and receive SIP calls. how was that able to work? |
| 06:47.44 | Corydon76-home | litage: haven't we already gone through that? |
| 06:47.51 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.200) |
| 06:47.57 | glm2k | heh, and i was about to answer him too :) |
| 06:48.14 | litage | Corydon76-home: my router is a cisco router; it doesn't use iptables |
| 06:48.21 | Corydon76-home | litage: short term memory loss? |
| 06:48.25 | glm2k | lol |
| 06:48.35 | Math` | litage: same thing anyways |
| 06:48.42 | glm2k | different name |
| 06:48.42 | litage | Corydon76-home: it uses ACLs and only ESTABLISHED, *no* RELATED clauses |
| 06:48.52 | Corydon76-home | litage: well, you're out of luck, because I don't have the source to your Cisco router handy |
| 06:49.02 | litage | heh |
| 06:49.09 | Math` | heh I wonder if anybody not working for cisco has that :P |
| 06:49.21 | Corydon76-home | Mike Lynn might |
| 06:49.31 | Corydon76-home | but he can't say (at least, not publically) |
| 06:49.58 | glm2k | oy, one hit for stolen ios source code... |
| 06:50.02 | Math` | got plenty of matches for 11.2-8 ios source :o |
| 06:50.23 | glm2k | i got one for 12.3 |
| 06:50.27 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 06:50.34 | Math` | oh |
| 06:50.53 | glm2k | ah heck, the uk police nabbed the guy just a few links down |
| 06:51.22 | glm2k | heh, talk about reading google like newspaper with a timeline |
| 06:52.22 | glm2k | i don't get it, the leak spawned security concerns? does it have that many exploits? |
| 06:52.46 | glm2k | wouldn't a leak actually make something more secure? (yeah i'm from _that_ camp) |
| 06:52.55 | dpryo | If you read the source, and discovers bugs, yeah, sure? :) |
| 06:53.12 | dpryo | Depends on how you use it |
| 06:53.22 | johnrage | I am looking for a developer who can help me build our asterisk..contact me offlist |
| 06:53.36 | glm2k | that's my point, after all this time, you'd expect cisco to have a a bulletproof implementation |
| 06:53.38 | Mavvie | johnrage: this is not a list. |
| 06:53.59 | dpryo | glm2k: Nothing is bulletproof ;) |
| 06:54.09 | glm2k | johnrage: please post on the biz list |
| 06:54.15 | glm2k | Mavvie: true that. |
| 06:54.17 | johnrage | thanks guys |
| 06:54.35 | Qwell | http://www.37signals.com/svn/archives/001064.php |
| 06:54.41 | Qwell | THAT is bulletproof ^^ |
| 06:55.06 | glm2k | lol |
| 06:59.13 | Mavvie | I don't understand this channel juggling: |
| 06:59.15 | Mavvie | <PROTECTED> |
| 06:59.19 | Mavvie | why does it do that? |
| 06:59.47 | *** join/#asterisk pobre (n=seymore@203.215.73.192) |
| 07:01.00 | *** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net) |
| 07:04.41 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 07:28.13 | *** join/#asterisk Ingress (n=asdf@CPE00121708156f-CM00111ae5ca70.cpe.net.cable.rogers.com) |
| 07:28.37 | *** part/#asterisk Ingress (n=asdf@CPE00121708156f-CM00111ae5ca70.cpe.net.cable.rogers.com) |
| 07:33.10 | KeX_WorX | good morning |
| 07:33.37 | KeX_WorX | is it possible to dial a number during a call is beeing sestablished? |
| 07:34.29 | KeX_WorX | if the callee is busy I want to dial somethin so, the caller gets automatically called back (is this understandable?) |
| 07:35.41 | *** join/#asterisk Astar (n=astar@ANantes-154-1-16-70.w81-53.abo.wanadoo.fr) |
| 07:40.47 | *** join/#asterisk EriSan (n=erisan@151.8.109.85) |
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| 07:48.37 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 07:51.53 | *** join/#asterisk Tond (n=t@69.158.18.148) |
| 07:53.06 | Tond | How can I trim the firt 2 digits of an extension before i redirect it to a Cisco GW? ex. i want to translate 011 into 322 and send it to the terminating gw |
| 07:53.47 | *** join/#asterisk kamileon (n=kamileon@68.62.190.253) |
| 07:54.51 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:56.01 | *** join/#asterisk koperniqs (n=koperniq@129.187.12.96) |
| 07:56.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:56.04 | koperniqs | hi |
| 07:56.25 | kamileon | hello everyone |
| 07:59.47 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-214.claranet.co.uk) |
| 08:03.15 | *** join/#asterisk kamileon_ (n=kamileon@68.62.190.253) |
| 08:04.53 | litage | while a call is taking place, would the RTP stream go through Asterisk or SER or GnuGK, or would it go straight from caller to callee? |
| 08:05.33 | Qwell | litage: depends |
| 08:05.51 | JamesDotCom | ser doesnt touch media streams |
| 08:05.53 | JamesDotCom | it's a sip proxy |
| 08:06.37 | mover | good morning:-) |
| 08:06.41 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 08:06.48 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 08:06.53 | litage | Qwell: depends on what? |
| 08:07.07 | Qwell | litage: with Asterisk, it depends on whether you allow reinvites |
| 08:07.38 | litage | Qwell: why/how does that affect it? |
| 08:07.42 | *** join/#asterisk svenna (n=svenna@p548D363E.dip0.t-ipconnect.de) |
| 08:07.55 | Qwell | if you reinvite, the media won't go through * |
| 08:08.19 | litage | Qwell: and if you don't reinvite, the RTP stream goes through *? |
| 08:08.24 | [av]bani | reinvite means the rtp streams can be redirected |
| 08:08.27 | litage | ah |
| 08:09.01 | [av]bani | * tells the remote end to establish rtp connection directly with the device, only sip remains on * |
| 08:09.13 | [av]bani | but of course, nat prevents that from working |
| 08:09.33 | FuriousGeorge | for some reason, one of my mailboxes greetings have become inaccessible nor can i set them in the "advanced options" |
| 08:14.10 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:15.31 | litage | how does a media proxy help/solve NAT issues? |
| 08:17.10 | *** join/#asterisk daguerro (i=Co_Care@167.205.67.28) |
| 08:17.47 | *** join/#asterisk puzzled (n=yeahrigh@a80-127-234-176.dial.xs4all.nl) |
| 08:21.03 | *** join/#asterisk Bambr (n=Bambr@213-35-241-181-dsl.end.estpak.ee) |
| 08:24.33 | af_ | skinny uses tcp or udp? |
| 08:25.39 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:25.51 | litage | af_: http://www.protocols.com/pbook/VoIPFamily.htm#Skinny |
| 08:28.02 | *** join/#asterisk shekhar_ (n=shekhar@221-128-138-24.exatt.net) |
| 08:28.09 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:28.36 | KeX_WorX | anyone uses the Read function in dialplan? |
| 08:28.58 | af_ | oh thanks lilo |
| 08:29.01 | af_ | oh thanks litage |
| 08:29.02 | af_ | :) |
| 08:29.31 | KeX_WorX | exten => _X,103,Read(INP,tt-monkeys,2,1) <-- I've this, but asterisk waites until timeout (not 1 sec) and reads nothing : / |
| 08:29.32 | af_ | oh ok, so tcp for signaling and rtp for audio..... |
| 08:29.38 | litage | np af_. i just recommend that you google your questions before asking them in irc. i found that page by searching for something like "skinny tcp udp" |
| 08:29.46 | af_ | mhh |
| 08:30.29 | af_ | why sip so popular? |
| 08:30.57 | af_ | it's because it's a very well described standard? |
| 08:34.35 | JamesDotCom | among other reasons |
| 08:35.16 | JamesDotCom | it's designed to work well over the internet |
| 08:37.37 | *** join/#asterisk [av]bani (n=[av]bani@washuu.anime.net) |
| 08:41.04 | litage | how does a media proxy help/solve NAT issues? |
| 08:41.31 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:44.57 | *** part/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4163113.sympatico.ca) |
| 08:45.04 | Qwell | litage: it can know where to send the packets internally |
| 08:45.04 | JamesDotCom | 2 nat'ted clients connect to the media proxy which has a real ip |
| 08:45.23 | JamesDotCom | instead of 2 nat'ted clients trying to connect to private ip's |
| 08:45.28 | JamesDotCom | but it's really not an issue |
| 08:45.41 | JamesDotCom | a decent sip proxy (SER) can rewrite the sdp packets when they have a private ip in themn |
| 08:47.11 | litage | if 2 NAT'd clients connect through SER, they don't need a media proxy? |
| 08:47.18 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:47.54 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 08:47.57 | JamesDotCom | basically |
| 08:49.25 | litage | if you're using SER or GnuGK, in what situation(s) would you/clients need a media proxy? |
| 08:51.38 | *** join/#asterisk brookshire (n=mbrooks@gateway.digium.com) |
| 08:53.08 | JamesDotCom | i cant speak for gnugk |
| 08:53.21 | litage | how about for SER? |
| 08:54.15 | JamesDotCom | with regards to sip, only if they're behind a symmetrical nat will there be problems i believe |
| 08:54.29 | JunK-Y | brookshire: !!! |
| 08:54.37 | brookshire | goto bed! |
| 08:54.39 | brookshire | :D |
| 08:55.16 | cypromis | yah ulaws go to sleep |
| 08:55.17 | cypromis | ;) |
| 08:55.33 | brookshire | blues #1 |
| 08:55.34 | brookshire | :D |
| 08:55.36 | JunK-Y | im on phone with gf |
| 08:55.44 | brookshire | o |
| 08:55.45 | JunK-Y | blues and chicks yeah! |
| 08:55.50 | JunK-Y | :) |
| 08:56.00 | cypromis | ;) |
| 08:56.09 | brookshire | <-- eating ramen |
| 08:57.18 | litage | JamesDotCom: symmetrical nat? |
| 08:57.38 | *** join/#asterisk Genman (n=hansenhl@atlrel1.hp.com) |
| 08:57.43 | Genman | Hi people |
| 08:57.56 | JunK-Y | <--- eating nothing |
| 08:58.27 | Qwell | <--- chewing on cellphone antenna |
| 08:58.28 | Qwell | mmm |
| 08:58.29 | Genman | Could anyone help me with some HW advise? |
| 08:58.34 | Qwell | Genman: ask away |
| 08:58.51 | Qwell | It's advice, btw.. |
| 08:58.55 | Qwell | advise is a verb |
| 08:59.03 | Genman | thx |
| 08:59.10 | JunK-Y | im going to bed now |
| 08:59.13 | JunK-Y | see ya guys |
| 08:59.17 | Qwell | JunK-Y: night |
| 08:59.19 | JunK-Y | see ya tomorrow brookshire |
| 08:59.23 | Qwell | oh wait |
| 08:59.28 | Qwell | JunK-Y: Are you going to VON? |
| 08:59.38 | JunK-Y | Qwell: not sure yet |
| 08:59.40 | JunK-Y | u? |
| 08:59.44 | Qwell | likely |
| 08:59.54 | *** join/#asterisk shekhar_ (n=shekhar@221-128-138-92.exatt.net) |
| 09:01.15 | JunK-Y | i will c if i could be there. |
| 09:01.44 | Qwell | I'm just gonna take vacation time |
| 09:02.09 | trixter | you get vacation time? |
| 09:02.13 | Qwell | if I do that, I'll have better chances of going to Astricon, paid |
| 09:02.19 | Qwell | trixter: 5 weeks a year. :D |
| 09:02.26 | trixter | I oinly get 52 weeks a year |
| 09:02.43 | Genman | I need to make a PBX, and we have 4 ISDN2 connections and 30 PSTN connections, what card(s) should I use for the server? |
| 09:02.49 | Qwell | yikes |
| 09:03.02 | trixter | define pstn connections |
| 09:03.03 | Qwell | Genman: Why not get like two PRIs? Where are you located? |
| 09:03.07 | trixter | analog lines? |
| 09:03.18 | Genman | alalog lines yes |
| 09:03.23 | Qwell | hp...us? |
| 09:03.24 | Genman | analog lines |
| 09:03.36 | Qwell | Get some PRIs, that's just nuts |
| 09:03.45 | Genman | Qwell: not for HP - Spare time fun :-) |
| 09:03.53 | trixter | depending on the plan it may actually be cheaper to get analog |
| 09:04.05 | Qwell | plus the ISDN? |
| 09:04.13 | trixter | what if its idsl? |
| 09:04.20 | Qwell | I'd be willing to put money on the fact that it's cheaper with PRI :p |
| 09:04.29 | trixter | which is basically isdn but dedicated instead of switched |
| 09:04.32 | litage | do you set the packetization (Eg: 20ms) on the enduser device (Eg: softphone), on *, or on SER? |
| 09:04.34 | Genman | Yea, see the lines are what we have, cannot really change that |
| 09:04.35 | trixter | I just dont know |
| 09:04.52 | trixter | and like where I live its more for a pri than analog even though my phone company ran fiber past my driveway :/ |
| 09:04.58 | Qwell | Genman: are the ISDN's BRI? |
| 09:05.01 | *** join/#asterisk jan__ (n=jan@ip22.ds1-saen.adsl.cybercity.dk) |
| 09:05.04 | Qwell | 4x BRI? |
| 09:05.34 | trixter | just a few miles away though pris get really really cheap |
| 09:05.45 | Genman | BRI ? |
| 09:05.53 | Qwell | ~bri |
| 09:05.54 | jbot | methinks bri is the Basic Rate Interface , an ISDN access interface type composed of two B-channels each at 64 kbps and one D-channel at 16kbps (2B+D). |
| 09:06.13 | Qwell | that? |
| 09:06.20 | Genman | Yea, sound like that |
| 09:06.31 | *** join/#asterisk GordonF (n=SumDude@dsl-146-18-17.telkomadsl.co.za) |
| 09:06.47 | Qwell | What's an E1, 30 or 32? |
| 09:07.18 | Qwell | If it were just a few less, I'd say get a channelbank, and put it to an E1 card |
| 09:07.21 | Genman | none of either i think, we have 4 lines |
| 09:07.23 | WasPhantom | depends on how you get it delivered. |
| 09:07.27 | trixter | 32 but one is dead and the other is signalling |
| 09:07.32 | trixter | so 30 bearer |
| 09:07.35 | Qwell | oh, hmm |
| 09:07.38 | Qwell | So that would work |
| 09:07.46 | trixter | the dead one is used for syncxing or something lame like that |
| 09:08.41 | Qwell | so like...if you could get something that'll take 4x BRI, and output a PRI, you could get a dual T1/E1 card, that box, and a channelbank...put the T1/E1's through * |
| 09:08.56 | Qwell | bit complex, but it's probably better than the alternatives |
| 09:09.25 | Genman | Qwell: and the analog? |
| 09:09.32 | Qwell | through the channelbank to an E1 |
| 09:09.35 | trixter | that is what the channel bank is for |
| 09:09.37 | *** join/#asterisk huseyin8{ (n=pieter44@60.48.210.93) |
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| 09:09.49 | Qwell | trixter: let me know if I'm going too far fetched |
| 09:09.51 | trixter | we have botnet! |
| 09:09.52 | *** join/#asterisk daryouch22{ (n=rldept@221.126.255.159) |
| 09:09.54 | *** join/#asterisk pierre47{ (n=alvin_Y@211.41.217.182) |
| 09:09.54 | Qwell | indeed |
| 09:09.56 | *** join/#asterisk Ducobra (n=sovaIyeA@85.108.234.28) |
| 09:10.01 | *** join/#asterisk yYkEv (n=harullah@61.106.75.239) |
| 09:10.03 | *** join/#asterisk tecnoIogy1[ (n=Exinghao@85.102.206.67) |
| 09:10.03 | Genman | ah, I need to read up on this before I start working on it, and recommended linkis ? |
| 09:10.03 | Qwell | no staffers avail, of course |
| 09:10.03 | *** join/#asterisk IsebnemK (n=LPsJp@85.101.234.56) |
| 09:10.04 | *** join/#asterisk onIine1Y (n=roberta4@85.108.171.12) |
| 09:10.06 | *** join/#asterisk isanhH (n=minyeAY@60.49.236.146) |
| 09:10.06 | *** join/#asterisk cobrag] (n=AjzOw@210.213.190.58) |
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| 09:10.14 | trixter | Genman: www.voip-info.org |
| 09:10.14 | *** join/#asterisk ashish21h (n=kIavye1F@85.106.225.123) |
| 09:10.15 | *** join/#asterisk adrianne34| (n=pink33_@85.103.7.4) |
| 09:10.18 | *** join/#asterisk wparkinso (n=fiIiz34X@62.162.198.109) |
| 09:10.18 | Qwell | ~docs |
| 09:10.20 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 09:10.20 | *** join/#asterisk muhpirq| (n=hiII34J@81.215.28.146) |
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| 09:10.22 | *** join/#asterisk arzuJ} (n=xiaogang@85.103.211.109) |
| 09:10.24 | Qwell | hmm, large |
| 09:10.32 | *** join/#asterisk estepania7I (n=wshaw^@85.103.53.28) |
| 09:10.33 | *** join/#asterisk DEwPr (n=alberto2@85.106.233.38) |
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| 09:11.02 | trixter | the botnet skills are weak |
| 09:11.07 | trixter | they had 100 the other day |
| 09:11.44 | *** join/#asterisk zvdamrongs (n=lsantiag@85.99.61.144) |
| 09:11.53 | Genman | Cheers, |
| 09:12.29 | *** join/#asterisk cliff8] (n=rnanette@211.41.217.182) |
| 09:13.26 | Qwell | Isn't this what +f is supposed to fix? heh |
| 09:14.08 | *** join/#asterisk YchienP (n=SIsOs@221.124.131.34) |
| 09:14.21 | tronix | aww... and I was *so* hoping they just wanted to learn something about *... ;) |
| 09:14.50 | Qwell | That was kinda...pathetic |
| 09:15.43 | Qwell | I'm unimpressed |
| 09:15.54 | trixter | people on aol have better |
| 09:16.08 | *** join/#asterisk soundguy (n=soundguy@mel-gw-pt1536.blendtek.com.au) |
| 09:16.09 | Qwell | my grandma has a bigger botnet |
| 09:16.23 | Qwell | and she's been dead for 20 years :p |
| 09:17.00 | soundguy | Hi |
| 09:17.07 | Qwell | hi |
| 09:17.09 | soundguy | Hey. How do I change the key you need to press to transfer calls in asterisk. By default it is "#", but I want to make it "##" -- I know it is possible, just not sure how. Any help would be greatly appreciated |
| 09:17.50 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:18.20 | Qwell | I think new versions can be set from features.conf |
| 09:18.29 | soundguy | Because often on some phones I need to enter numbers followed by the hash/pound key, and when I press that it tries to transfer it |
| 09:18.46 | *** join/#asterisk cryter (n=l@219.95.82.2) |
| 09:19.37 | *** join/#asterisk eivindtr (n=wingnut-@cC3012269.inet.catch.no) |
| 09:19.54 | cryter | hello |
| 09:20.01 | cryter | i'm new to asterisk |
| 09:20.43 | cryter | i just bought a TDM card and a IP-Phone |
| 09:21.03 | Qwell | off to bed |
| 09:21.19 | cryter | the Card and IP-Phone has been successfully installed but i don't know what should I do now |
| 09:21.25 | Genman | Qwell: Thanks for the help, enjoy your sleep |
| 09:21.34 | soundguy | Configure it :) |
| 09:21.40 | soundguy | Have you compile asterisk with the zaptel drivers? |
| 09:21.44 | cryter | yes |
| 09:21.49 | cryter | i compiled it already |
| 09:22.01 | soundguy | So have you configured zaptel.conf and zapata.conf? |
| 09:22.07 | cryter | yes |
| 09:22.16 | cryter | modprobe wctdm |
| 09:22.18 | cryter | success |
| 09:22.22 | soundguy | what module do you have, FXO or FXS? |
| 09:22.27 | cryter | modprobe zaptel also success |
| 09:22.35 | cryter | 1 FXO and 1 FXS |
| 09:22.46 | cryter | modprobe wcfxo also success |
| 09:22.47 | soundguy | ok |
| 09:22.49 | cryter | modprobe wcfxs also success |
| 09:22.55 | Qwell | cryter: don't load those two |
| 09:22.58 | Qwell | just wctdm |
| 09:22.59 | cryter | my IP phone is AT-323 |
| 09:23.01 | soundguy | so in extensions.conf have you configured it? |
| 09:23.20 | Qwell | cryter: wctdm takes care of the fxo and fxs ports on the tdm card. zaptel is a dep, so it'll be autoloaded |
| 09:23.26 | cryter | the extensions.conf is the one that I dont know what should I do with it |
| 09:23.51 | *** join/#asterisk Sajid_Khan (n=human@203.145.159.37) |
| 09:23.53 | cryter | my boss tell me to use IAX protocol |
| 09:23.56 | Sajid_Khan | Hi |
| 09:24.24 | Sajid_Khan | Any one intrested in doing one Asterisk Project |
| 09:26.34 | cryter | my TDM card I connected with telephone line, then my AT-323 IP-Phone is connected to the network cable |
| 09:26.42 | rene- | Sajid_Khan: i might be |
| 09:26.43 | soundguy | Use SIP |
| 09:26.46 | *** join/#asterisk zachary10s (i=nikki10n@85.98.106.144) |
| 09:26.46 | cryter | i can see the IP-Phone on the network |
| 09:26.48 | *** join/#asterisk ATnikolaos (i=robinds@81.215.74.93) |
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| 09:27.01 | cryter | how can i make phone call to this IP-phone? |
| 09:27.03 | rene- | Sajid_Khan: what are you looking for |
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| 09:27.18 | soundguy | hmm...lots of joins |
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| 09:27.34 | Sajid_Khan | Rene- would u mind if we talk in private...?? |
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| 09:27.41 | cryter | i woulr appreciate if somebody can give me a hint, i need a kick start |
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| 09:27.51 | cryter | i would appreciate if somebody can give me a hint, i need a kick start |
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| 09:28.02 | cryter | netsplit |
| 09:28.06 | Sajid_Khan | Any one else intrested in doing one Asterisk Project |
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| 09:32.17 | trixter | Sajid_Khan: what type of asterisk project? I am doing many right now ... |
| 09:32.46 | trixter | this botnet is quite lame |
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| 09:32.50 | basty | Hi |
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| 09:33.43 | basty | I want to connect from an Asterisk 1.0.10 to an Asterisk 1.2.4 via SIP (not IAX). After setting up an registry line into the sip.conf it replys "wrong passwort". Anyone has an idea ? |
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| 09:36.57 | Sajid_Khan | !ping $me |
| 09:38.07 | jaike | am compiling asterisk-addons...make clean make make install..but the cdr_addon_mysql.so isnt compiled..do i have to edit the Makefile? |
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| 09:38.24 | jaike | its not in /usr/lib/asterisk/modules |
| 09:38.57 | cypromis | do you have the mysql development stuff installed ? |
| 09:39.02 | cypromis | otherwhiles it will just ignore it |
| 09:39.22 | Genman | What is the name of analog lines? |
| 09:39.28 | jaike | mysql is on another server |
| 09:39.40 | cypromis | you need the development headers/libs to compile the ccdr stuff |
| 09:39.49 | Genman | I mean digital lines are DSx , but what is analog lines called? |
| 09:41.20 | jaike | hmm..tnx |
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| 09:43.58 | rene- | ~seen jerjer |
| 09:44.02 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 18d 12h 13m 8s ago, saying: 'thanks again'. |
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| 09:45.47 | cypromis | sweet |
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| 09:46.29 | jaike | that did it..thanks |
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| 09:48.56 | basty | I want to connect from an Asterisk 1.0.10 to an Asterisk 1.2.4 via SIP (not IAX). After setting up an registry line into the sip.conf it replys "wrong password". Anyone has an idea ? In the debug I can see that Asterisk 1.0.10 is seding a register string like: <sip:customer@XXXXX>;tag=as3e7326b4' <- the tag ?! |
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| 10:16.00 | jaike | have just upgraded from 1.0.9 to 1.2.4....i try to make a call to a queue..it goes into the queue, plays MOH for a few seconds...then hangs up the call |
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| 10:16.28 | jaike | ive already set joinempty = yes and leavewhenempty = no |
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| 10:21.31 | Sajid_Khan | Hm |
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| 10:23.26 | PoWeRKiLL | hi |
| 10:23.46 | PoWeRKiLL | I have lot of error like this Feb 6 11:10:49 WARNING[6178] chan_zap.c: Ring requested on channel 0/7 already in use on span 1. Hanging up owner. any idea ? |
| 10:24.06 | Johan | Question: When a call enters asterisk there are several variables like 'uniqueid'. When I use queues and asterisk calls back an agent, the original uniqueid is lost and a new one is created. Is there a way to retrieve the original uniqued back when an agent is called back? Or a way to pass the var's? |
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| 10:29.10 | PoWeRKiLL | Johan get it when you get the call and store it on a variable |
| 10:30.25 | RoyK | hm... |
| 10:32.05 | RoyK | with callingcards, prepaid stuff, you usuall have a small area covered with soft plastic or so, so you can just remove that with a coin or your nails or something. what do you call this area? in english? |
| 10:32.09 | *** join/#asterisk [Lv]archeens (n=a@84.237.140.136) |
| 10:32.17 | [Lv]archeens | Join New MMORPG Online Game http://www.afelhem.com ... Join forum .. and fill out beta ! |
| 10:32.19 | [Lv]archeens | Join New MMORPG Online Game http://www.afelhem.com ... Join forum .. and fill out beta ! |
| 10:32.27 | *** part/#asterisk [Lv]archeens (n=a@84.237.140.136) |
| 10:34.37 | *** join/#asterisk pipa (i=user@210.16.41.221) |
| 10:34.41 | Johan | PoWeRKiLL: I do but when an agent is called back I need to find out what call that is, you see what I men? |
| 10:35.29 | PoWeRKiLL | so make an entry in astdb uniqueid and callerid |
| 10:36.08 | Johan | PoWeRKiLL: ok, but the uniqueid changes when the agent is called back |
| 10:36.18 | Johan | that's the problem |
| 10:36.18 | *** join/#asterisk fulgas (n=fulgas@209.8.233.10) |
| 10:37.43 | PoWeRKiLL | Johan so use the callerid for find it back |
| 10:38.53 | Johan | PoWeRKiLL: Ok, but I thought the callerid isn't alway's known, or I am wrong? Lemme take a look at the docs again |
| 10:38.54 | pipa | hi guys, new to installing asterisk |
| 10:39.12 | pipa | whats the best distro for asterisk? thanks |
| 10:39.16 | PoWeRKiLL | Johan it's true but I don't see another solution |
| 10:39.24 | CaT[tm] | the one you like the most. |
| 10:39.57 | pipa | CaT[tm], so any distro will do as long as it is working.. thanks! |
| 10:40.02 | Johan | PoWeRKiLL: Hmmz I will keep on searching ;) |
| 10:40.08 | CaT[tm] | pipa: prettymuch. |
| 10:40.17 | pipa | okidoki! |
| 10:40.21 | Astar | someone knows actos ? |
| 10:41.32 | knobo | Does anyone have example-data for a group_member_table for a database? |
| 10:42.07 | knobo | I'm a litle bit unshure on how to add entries |
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| 10:44.44 | knobo | sorry that is group |
| 10:44.51 | knobo | argh. |
| 10:44.57 | knobo | group_member_table |
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| 10:47.10 | knobo | quehe_member_table |
| 10:48.22 | knobo | . o O (need more coffee) |
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| 10:54.37 | RoyK | knobo: wtf are you doing here? :) |
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| 10:56.06 | EriSan | hi guys, if i get a SIP/2.0 403 Forbidden, what would be causing that ? |
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| 10:57.54 | *** kick/#asterisk [YIfAc!n=mark@pdpc/sponsor/digium/kram] by kram ((*@81.213.99.129) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 10:57.54 | *** kick/#asterisk [ann-mari36\!n=mark@pdpc/sponsor/digium/kram] by kram ((*@81.213.99.129) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 10:57.54 | *** kick/#asterisk [atlanta25e!n=mark@pdpc/sponsor/digium/kram] by kram ((*@81.213.99.129) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 10:57.54 | *** kick/#asterisk [maureenzO!n=mark@pdpc/sponsor/digium/kram] by kram ((*@81.213.99.129) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
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| 11:02.17 | RoyK | EriSan: wrong password? |
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| 11:03.00 | EriSan | no, i was able to make a call before, after that i allways get that |
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| 11:11.02 | Anonty | Hi |
| 11:11.16 | Anonty | One of my peers always says 'UNREACHABLE' |
| 11:11.25 | Anonty | Is there a way to see why it is 'UNREACHABLE'? |
| 11:11.40 | *** join/#asterisk jan__ (n=jan@ip22.ds1-saen.adsl.cybercity.dk) |
| 11:12.39 | jan__ | I look for a way to connect 4xISDN2 (digital/BRI) lines to my Asterisk, do you know any good digium hardware? |
| 11:14.23 | Zeeek | Anonty maybe it doesn't like responding to a qualify |
| 11:14.32 | Zeeek | remove qualify and see if that helps |
| 11:17.32 | ckruetze | jan__: Try http://www.beronet.com |
| 11:19.16 | *** join/#asterisk Kittie (n=G@pD953591D.dip.t-dialin.net) |
| 11:19.17 | jan__ | does digium not have any hardware to a 4xISDN BRI line |
| 11:19.30 | Kittie | join 3d chat |
| 11:19.32 | Kittie | It's free, and doesn't take long to set up. Check it out at: |
| 11:19.34 | Kittie | http://www.imvu.com/catalog/web_registration.php?userId=508500 |
| 11:19.36 | *** part/#asterisk Kittie (n=G@pD953591D.dip.t-dialin.net) |
| 11:20.00 | ckruetze | jan__: I don't know, but I think they only do PRI and analog |
| 11:20.14 | jan__ | ok |
| 11:20.50 | jaike | whew. just finished upgrading 1.0.7 to 1.2.4. so many configs to overhaul |
| 11:22.49 | *** join/#asterisk hcandacej (n=chao36_@85.106.233.38) |
| 11:23.31 | Anonty | Zeeek >> Then the connection still doesn't work. I mean I don't get any incomming calls |
| 11:23.50 | Anonty | Zeeek >> So I have no idea how to check whether it is working or not |
| 11:27.12 | *** join/#asterisk Etop (n=Etop@80.72.33.130) |
| 11:27.28 | Etop | hiya |
| 11:27.36 | Etop | i need some help |
| 11:27.42 | iDunno | that's nice. |
| 11:27.46 | Etop | :) |
| 11:27.47 | Etop | i know |
| 11:27.55 | Etop | and im sure some of you people are so nice |
| 11:28.00 | Etop | and will answer a simple question |
| 11:28.01 | iDunno | (or better yet: ask and people may respond :) |
| 11:28.06 | Etop | hehe |
| 11:28.07 | Etop | ok |
| 11:28.15 | Etop | i have a cisco as5300 |
| 11:28.25 | Etop | and asterisk |
| 11:28.35 | Etop | i already have h323 configured on as5300 |
| 11:28.57 | Etop | and i want to add sip |
| 11:29.02 | Etop | using asterisk... |
| 11:29.12 | Etop | the question is - if i add one more dialtopeer on cisco |
| 11:29.14 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 11:29.19 | Etop | will h323 get screwed up ? |
| 11:30.15 | iDunno | right - not a clue, lost you after "cisco" then lost you again when you said "h323" :) |
| 11:30.47 | Etop | :) |
| 11:30.50 | Etop | ok |
| 11:31.00 | Etop | i have no choice |
| 11:31.07 | Etop | i have a bunch of clients connected through h323 |
| 11:31.15 | Etop | and i can't disconnect them just like that |
| 11:31.38 | thazza | Etop: Can't you find the power plug? |
| 11:31.48 | Etop | i think i could |
| 11:31.56 | Etop | i can unplug it and look for a new job |
| 11:32.05 | Etop | :) |
| 11:32.40 | thazza | Etop: Unplug, look for a new job.. Or don't fix. quit, and then when they call you back up in a month.. install sip phones. |
| 11:33.35 | iDunno | thazza: you are evil. :) |
| 11:33.58 | Etop | bbl |
| 11:33.59 | Etop | ty |
| 11:34.00 | *** part/#asterisk Etop (n=Etop@80.72.33.130) |
| 11:34.02 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 11:34.16 | mut | yesssssss |
| 11:34.21 | mut | 3 cars stuck in the driveway to work this mroning |
| 11:34.24 | mut | one of them being mine |
| 11:35.01 | thazza | mut: Build a bigger driveway. |
| 11:35.29 | *** join/#asterisk Newbie___ (i=me@211.24.146.12) |
| 11:35.34 | mut | whats that goin to do |
| 11:35.45 | Newbie___ | hi guys, is there a way to find out if i have successfully installed h323 ? |
| 11:35.55 | iDunno | mut: make them not-stuck? |
| 11:36.10 | mut | how is a bigger driveway goin to make em not get stuck? |
| 11:36.30 | mut | just more room for more to be stuck |
| 11:38.17 | iDunno | limit to 3 cars, make bigger driveway. |
| 11:39.45 | thazza | mut: Stop putting super glue on your driveway. ;-) |
| 11:39.56 | mut | heh |
| 11:40.01 | mut | snow drifts this morning were horrible |
| 11:40.11 | mut | even on the road to work i almost got stuck cause of a drift |
| 11:40.24 | thazza | mut: Much better to put caster suger mixed with cooking oil. |
| 11:40.26 | *** join/#asterisk saftsack (n=saftsack@IP-213188106101.dialin.heagmedianet.de) |
| 11:40.33 | saftsack | hi |
| 11:40.38 | saftsack | _Sam--, hi are you here? |
| 11:41.50 | trixter | snow drifts arent usually a problem for me, infact people have followed me down unplowed streets cause I kinda clear it as I drive |
| 11:42.07 | trixter | depends on how high the drift is though |
| 11:43.18 | mut | well |
| 11:43.35 | mut | it's about 7-10 inches of snow drift |
| 11:43.49 | mut | for ~500 yards of driveway |
| 11:44.12 | mut | i made it about halfway til i saw another stuck car and hesitated |
| 11:44.22 | mut | that lil bit of slow down stopped me cold |
| 11:44.44 | mut | i wouldn't have made it around it anyway so i guess it didn't matter |
| 11:45.09 | dpryo | lol, you newbies in snow! |
| 11:45.54 | trixter | oh that is nothing |
| 11:45.56 | trixter | for my jeep anyway |
| 11:46.01 | trixter | 31 inch tires, 2 inch lift |
| 11:46.06 | mut | for a jeep sure |
| 11:46.14 | mut | the ground clearance is more than 7 inches |
| 11:46.15 | trixter | I have driven through higher snow on the road to get to the chain check point :P |
| 11:46.29 | mut | and 4wd |
| 11:46.35 | dpryo | I live in Norway. We have snow 12 months a year, and penguins in the street. |
| 11:46.38 | trixter | well its not just that I raised my jeep up with larger tires and all that.. had to put a 2 inch lift on so they fit |
| 11:46.43 | trixter | its fun off roading though! |
| 11:46.54 | trixter | live near the rubicon trail which is quite a bit of fun |
| 11:46.57 | mut | looks like the arctic right now tho |
| 11:46.59 | trixter | 22 miles takes 2 days to drive |
| 11:47.06 | mut | all ya see is headlights down the drivewa |
| 11:47.13 | mut | with snow blowing past contantly with no let up |
| 11:47.21 | mut | bbfew |
| 11:47.29 | mut | gotta make my way back to my car to get it out |
| 11:48.55 | Johan | PoWeRKiLL: I found a solution. All queue actions are written to a log file. Parsing that makes it possible to see 'who' answerd 'wich' phonecall. |
| 11:49.38 | Zeeek | Anonty ? |
| 11:49.54 | Zeeek | slt PoWeRKiLL |
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| 12:00.03 | mut | man i love oversized coffee cups |
| 12:00.33 | saftsack | does my asterisk need some special functions for redirection? |
| 12:01.17 | blkremedy | Is 40 gigs too much for an asterisk@home box? |
| 12:01.45 | blkremedy | I mean what's the ideal HDD size? |
| 12:01.52 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
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| 12:03.34 | *** part/#asterisk sambal (n=hartger@ip51cf10fa.direct-adsl.nl) |
| 12:03.59 | *** join/#asterisk sambal (n=hartger@ip51cf10fa.direct-adsl.nl) |
| 12:04.01 | blkremedy | I'm trying to determine if I should use a 10 gig or a 40 gig but, it seems like 40 gigs would be an over kill. To all of the experts in the room; does asterisk really need a lot of space? |
| 12:04.21 | Zeeek | You never have enough, so why not use the 40 |
| 12:04.42 | Zeeek | The bigger disk is usually faster anyway |
| 12:05.32 | mut | less it's a few 9gig scsi's |
| 12:05.42 | sambal | hi, i have a question about outgoing callerid with the Digium TE210P Card(2x e1 card), Can this be set in the extensions.conf with Set(CALLERID(number)=12345) ? or are there other settings to do this? |
| 12:06.04 | mut | it can be set in the extensions |
| 12:06.10 | mut | or zapata.conf |
| 12:06.16 | blkremedy | That is a fact. Just didn't want to waste a 40 gig for something that's not going to really use all of that space |
| 12:06.17 | mut | callerid=12345 |
| 12:06.21 | mut | channe=>1 |
| 12:06.25 | mut | callerid=54321 |
| 12:06.27 | mut | channe=>2 |
| 12:06.31 | sambal | yeah, with zapata.conf you must set it to an channel |
| 12:06.32 | blkremedy | I'll go with the 40 gig just in case |
| 12:06.41 | mut | or set it to call channels |
| 12:06.45 | mut | callerid=54321 |
| 12:06.51 | mut | channe=>1-24 |
| 12:06.54 | mut | er whatever |
| 12:06.55 | RoyK | ~pb |
| 12:06.57 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 12:07.11 | sambal | k, and with extensions is it the way to do it with Set(CALLERID)? |
| 12:07.20 | mut | yea |
| 12:07.40 | sambal | okey great i know enough going to try it again :-) |
| 12:07.44 | sambal | thanks for your info mut |
| 12:07.50 | sambal | and your time ofcourse |
| 12:08.12 | sambal | why? |
| 12:08.17 | mut | echo |
| 12:09.11 | sambal | hmm |
| 12:09.23 | [av]bani | echo, the curse of cheap hardware :( |
| 12:09.47 | mut | yep |
| 12:10.03 | RoyK | mut: zaptel echocancel.... |
| 12:10.25 | mut | havn't gotten it to work |
| 12:10.49 | *** join/#asterisk diegonix (n=diego@201.6.253.99) |
| 12:11.34 | [av]bani | mut: http://search.ebay.com/search/search.dll?cgiurl=http%3A%2F%2Fcgi.ebay.com%2Fws%2F&fkr=1&from=R8&satitle=%2Btellabs+%2B2572&category0= |
| 12:11.57 | [av]bani | http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers |
| 12:12.21 | sambal | hmm, not working with callerid in zapata i think my E1 provider (versatel) has closed it for us |
| 12:12.22 | mut | something you use? |
| 12:12.41 | *** join/#asterisk zotz (n=zotz@24.244.133.10) |
| 12:13.37 | [av]bani | mut: seems the best way to deal with echo, the asterisk software ec's are not very good |
| 12:14.15 | mut | i don't need 23 of em tho |
| 12:14.15 | [av]bani | mut: and the tellabs ECs seem the best, its what ILECs use |
| 12:14.16 | mut | heh |
| 12:14.27 | mut | you ever done this? |
| 12:14.36 | [av]bani | well you have a te405p, so you're presumably using >1 channel |
| 12:15.15 | mut | yea.. it's a t1 canceller tho.. not a per channel |
| 12:15.20 | mut | or is it? |
| 12:15.29 | [av]bani | its a t1 canceller, so it cancels every channel on the t1 |
| 12:15.54 | mut | well for right now i'm only using 1 of em |
| 12:16.00 | [av]bani | doesnt matter then |
| 12:16.00 | mut | maye in the future i'd euse 4 |
| 12:16.15 | [av]bani | itll work the same way regardless |
| 12:16.41 | [av]bani | your t1 is still clocked at 1.54mbps, you're just only using 1 timeslot of it |
| 12:17.04 | saftsack | Feb 6 13:17:47 debian kernel: a 5 33 0 4 0 3 0 0 tr: 33 r 16 100 2 11 1 11 2 2 |
| 12:17.08 | saftsack | what is that? |
| 12:18.25 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
| 12:20.49 | mut | so has anyone in here actually done that? |
| 12:22.05 | I-MOD | a hardware echocan? |
| 12:22.29 | mut | the tellabs hack |
| 12:25.53 | *** join/#asterisk reth (i=reth@2001:16d8:20:2:211:11ff:fe58:35cb) |
| 12:27.37 | mut | http://cgi.ebay.com/Tellabs-253-Echo-Canceller-B-1_W0QQitemZ5863050110QQcategoryZ3309QQssPageNameZWD1VQQrdZ1QQcmdZViewItem |
| 12:27.48 | mut | that maybe be better than the other? |
| 12:27.52 | saftsack | are some isdn experienced people here? |
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| 12:34.03 | mut | maye if i get good at it i can build some homebrews up and sell em to people |
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| 12:34.10 | mut | premade echo cans for asterisk |
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| 12:34.15 | CaT[tm] | that the user/group? |
| 12:34.15 | mut | kline flood! |
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| 12:34.16 | CaT[tm] | oops |
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| 12:34.43 | mut | so what happened to +r on the chan |
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| 12:34.56 | *** mode/#asterisk [+b *!*@85.101.70.178] by kram |
| 12:34.57 | saftsack | were there a netsplit? |
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| 12:35.17 | iDunno | nope - looks like people are trying to flood :/ |
| 12:35.32 | *** kick/#asterisk [oZfCy!n=mark@pdpc/sponsor/digium/kram] by kram ((*@85.101.70.178) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 12:35.33 | *** kick/#asterisk [YEnGr!n=mark@pdpc/sponsor/digium/kram] by kram ((*@85.101.70.178) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 12:35.33 | *** kick/#asterisk [yaomin28O!n=mark@pdpc/sponsor/digium/kram] by kram ((*@85.101.70.178) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 12:35.33 | *** kick/#asterisk [alexandrgW!n=mark@pdpc/sponsor/digium/kram] by kram ((*@85.101.70.178) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 12:35.33 | *** kick/#asterisk [phearosa!n=mark@pdpc/sponsor/digium/kram] by kram ((*@85.101.70.178) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 12:35.33 | *** kick/#asterisk [lzeynep20r!n=mark@pdpc/sponsor/digium/kram] by kram ((*@85.101.70.178) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 12:36.15 | vattern | lol |
| 12:36.15 | mut | his clone kick thing sure takes a bit to kick in |
| 12:36.24 | sambal | damn bots |
| 12:36.37 | Poincare | what happened to the "+r" on #asterisk? |
| 12:36.41 | saftsack | omg ^ |
| 12:37.09 | CaT[tm] | nope |
| 12:37.24 | mut | has anyone actually put together one of these telllab modular echo cancellers for asterisk? |
| 12:38.06 | mut | there it is1 |
| 12:38.08 | mut | ! |
| 12:40.13 | saftsack | is it possible to define a special context for every channel on my tdm card with zap? |
| 12:40.19 | I-MOD | yep |
| 12:43.25 | dudes | setup a group for each zap channel, and assign the context in the group |
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| 12:49.22 | saftsack | dudes, thanks |
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| 12:49.37 | Morex | Hello all |
| 12:49.50 | Morex | Anybody got experience hooking up * to Avaya switches? |
| 12:50.00 | saftsack | dudes, do you mean zapata.conf? |
| 12:50.10 | Morex | We've got a problem with outbound calls |
| 12:50.30 | saftsack | dudes, ok works |
| 12:50.32 | saftsack | thanks |
| 12:51.24 | znoG | just wondering... |
| 12:51.29 | znoG | is this Joseph Tanner guy serious? |
| 12:51.38 | znoG | does he really think Walmart is going to give away free IP phones? |
| 12:52.47 | mut | hm? |
| 12:55.10 | mut | what about the te406p |
| 12:55.15 | mut | echo cancel work on that? |
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| 13:06.04 | CaViCcHi | free ip phones? |
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| 13:08.03 | joelsolanki | Hello all, I want some guidance. I am using asterisk. I have take onnet usa. it consists of more than 50,000 area codes. Now how do i route only the area codes of onnet on that provider and the area codes which are not covered in onnet to other provider ? |
| 13:08.23 | joelsolanki | can anybody provide me hints for this ? |
| 13:08.58 | Ahrimanes | joelsolanki: you need sometihng like least cost routing |
| 13:09.39 | joelsolanki | but least cost routing can only work if i m having billing system integrate in asterisk. |
| 13:09.42 | hypa7ia | joelsolanki: that kind of complex question might be better on the mailing list |
| 13:09.56 | joelsolanki | :) |
| 13:10.06 | hypa7ia | just sayin' :) |
| 13:10.09 | Ahrimanes | joelsolanki: http://www.voip-info.org/wiki/view/Application+LCDial |
| 13:10.18 | Ahrimanes | joelsolanki: you can just set billing cost to 0 |
| 13:10.31 | joelsolanki | I hope some experienced person will be here. and i dont think this complex question. lot of people require this type of setup. |
| 13:10.47 | znoG | how do you get the listing of the 50,000 area codes for onnet? |
| 13:11.01 | joelsolanki | My provider gave me. |
| 13:11.07 | znoG | in what format? |
| 13:11.26 | joelsolanki | dont u think extension+macros should work ? |
| 13:11.46 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 13:11.51 | [av]bani | mut: yea i think you could make money reselling hacked tellabs EC's :) |
| 13:11.52 | Ahrimanes | well it could easily work, but a lot of maintenance |
| 13:12.04 | joelsolanki | hmm ok. |
| 13:12.14 | joelsolanki | i will try out something :). thanks for hints |
| 13:12.25 | znoG | as for the best way to do it, probably writing app_onnet.c or something similar. Another way is to get that list, parse it and stick it in a DB, write an AGI so that when you go to Dial someone, it checks if the area code is part of the onnet listing, otherwise dial using whoever you wanna use. |
| 13:13.16 | Ahrimanes | the goal here is the same as in LCD so why not use that and set billing to zero or something like that? that gives you db contained codes and easy way to check the db |
| 13:13.45 | *** join/#asterisk kamileon (n=kamileon@68.62.190.253) |
| 13:13.57 | znoG | not sure how many calls you plan on making per day, but it could get a bit of heavy load if you make a lot of them |
| 13:14.11 | mut | do the 406p cards echo cancel very well [av]bani? |
| 13:15.25 | *** join/#asterisk saggief (n=dequinuL@85.96.123.247) |
| 13:16.03 | *** join/#asterisk cpm (n=Chip@border0.avitecture.net) |
| 13:16.31 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:17.07 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 13:17.10 | mishehu | oh fuck, please tell me that I'm not the only person having this problem with voicemail |
| 13:17.28 | *** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 13:17.37 | *** join/#asterisk _Paulo_ (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 13:17.46 | mishehu | anytime somebody leaves me a message, the msgXXXX.txt file is 0 bytes |
| 13:18.54 | joelsolanki | hmm ok znog:i got your point |
| 13:19.01 | *** join/#asterisk Flyboy-SR22 (i=rsears@gateway.adnc.com) |
| 13:19.53 | [av]bani | mut: no idea. only one way to find out! |
| 13:20.00 | _Paulo_ | somebody uses gnu-bayonne with *??? |
| 13:20.15 | mut | yep, keep asking in here |
| 13:21.12 | Lurr | can anyone help me with a ringing indication problem? |
| 13:21.59 | darkskiez | Lurr: no, but our * server is called lrrr |
| 13:22.18 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:22.37 | *** join/#asterisk GchanghoW (n=Zmarrucc@81.215.3.152) |
| 13:22.44 | [av]bani | mut: but tellabs does work with anything... |
| 13:23.49 | mut | i may as well just use my cisco 5350 then |
| 13:25.07 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:26.39 | *** join/#asterisk _tdonahue (n=tdonahue@208.51.101.201) |
| 13:28.32 | *** join/#asterisk SparFux (n=player@tor/session/x-9d3771c51dcf5afc) |
| 13:28.53 | SparFux | Hello. How can I determine wether a channel has been hung up by the peer? |
| 13:29.04 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 13:29.16 | iDunno | you should go. probably. |
| 13:29.24 | iDunno | for singing that line, of that song, evil. |
| 13:29.27 | SparFux | Hi elg. |
| 13:31.05 | *** join/#asterisk pohik (n=daryl6G@81.215.3.152) |
| 13:31.16 | SparFux | When my context is executed, which I have an "h" prio in, the extension once dialed is first terminated and AFTER THAT the "h" context is executed. But this is too late for me, I have to have it executed immediately after peer hung up. Or I have to check, wether it has hung up the line. |
| 13:31.49 | elg | jbot, onjoin -elg |
| 13:31.49 | jbot | ok, elg |
| 13:31.58 | elg | jbot, onjoin elg |
| 13:32.26 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:33.46 | [av]bani | mut: someone just posted on * ml that they had bad experience with te411p |
| 13:33.58 | mut | yea, hm |
| 13:34.09 | mut | maybe i'll have to bite the bullet and sell this pos |
| 13:34.14 | mut | get a sangoma |
| 13:34.26 | [av]bani | just get a tellabs and hack it |
| 13:34.34 | Bambr | hey guys |
| 13:34.44 | mut | then i need to find all the other crap for it too |
| 13:34.49 | Bambr | i got question about configuration stored in db |
| 13:34.49 | [av]bani | its good enough for AT&T its good enough for you |
| 13:35.02 | [av]bani | mut: no, just 2 RJ45's and a powersupply |
| 13:35.06 | mut | heh they use it as intended tho |
| 13:35.14 | mut | and rack to put it in |
| 13:35.16 | Bambr | how do i write to extensions table what used to be include directive in file extensions.conf? |
| 13:35.23 | [av]bani | no, thats only if you want it to look pretty |
| 13:35.43 | mut | well i'de rather not have cards strewn across the floor |
| 13:35.50 | [av]bani | your loss :) |
| 13:35.54 | [av]bani | some of us LIKE it that way |
| 13:36.07 | mut | and some of us like to look professional |
| 13:36.12 | mut | *shrug* |
| 13:36.16 | *** join/#asterisk walhala (n=walhala@stardust.noc.frontier.fr) |
| 13:36.18 | [av]bani | but if you get a tellabs then it will work no matter what you have.. digium, sangoma, cisco, etf |
| 13:36.19 | walhala | hi all |
| 13:36.21 | [av]bani | etc |
| 13:36.40 | walhala | i have a strange error on my asterisk i have this "Failed to grab lock" what does it mean ? |
| 13:36.52 | [av]bani | but a rack seems cheap |
| 13:36.55 | mut | what happens when i use it for fxo? |
| 13:37.09 | walhala | does anyone know how to loadbalance sip ? |
| 13:37.14 | mut | hm |
| 13:37.23 | mut | i dunno, i'll look at it a bit more |
| 13:37.25 | [av]bani | mut: what do you think it _is_, when you connect t1 to ilec? |
| 13:37.30 | [av]bani | it's fxo |
| 13:37.40 | mut | cause the cards on ebay and whatever |
| 13:37.53 | mut | yea |
| 13:38.02 | [av]bani | all it does is echo cancel, whether its fxo or fxs is up to your channelbank |
| 13:38.37 | [av]bani | and really, fxs sucks you should use ip phones instead... |
| 13:38.44 | [av]bani | and those usually have local EC in the phone |
| 13:39.09 | mut | yea |
| 13:39.19 | mut | we have a whole town hooked up via fxs tho |
| 13:39.27 | [av]bani | EC is just some headache you dont want to mess around with, and a good hardware EC seems the best way to go |
| 13:39.38 | mut | our own copper house to house |
| 13:39.45 | [av]bani | kill all humans |
| 13:40.37 | jaiger | mut, you want hardware EC if you have that many users |
| 13:40.42 | mut | i could get this... |
| 13:40.47 | mut | http://cgi.ebay.com/Tellabs-253-Echo-Canceller-B-1_W0QQitemZ5863050110QQcategoryZ3309QQssPageNameZWD1VQQrdZ1QQcmdZViewItem |
| 13:40.51 | mut | then get those cards |
| 13:40.55 | jaiger | you don't want the headache of software ec |
| 13:40.55 | mut | and stick em in there |
| 13:41.00 | *** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 13:42.01 | jaiger | mut, is that the 2 card shelf? |
| 13:42.05 | mut | yea |
| 13:42.11 | mut | with 2 cards in it |
| 13:42.15 | mut | but not the models i need aparently |
| 13:42.30 | mut | least that guide on voip-info doesn't list the 2551 model |
| 13:42.37 | jaiger | that should work. I have the 16 card shelf w/ one 2572 in it. would have liked the 2 card shelf |
| 13:42.49 | *** join/#asterisk newl (n=newlook@203-59-210-244.dyn.iinet.net.au) |
| 13:42.59 | mut | so you've done this before? |
| 13:43.02 | jaiger | yeah |
| 13:43.09 | jaiger | for my own office here |
| 13:43.13 | *** join/#asterisk cypromis (n=michael@82.103.129.176) |
| 13:43.16 | mut | doing all the ping outs not much of a deal then? |
| 13:43.22 | mut | pin |
| 13:43.37 | jaiger | it was a pain that I wish I didn't have to do but it works |
| 13:43.46 | mut | hm |
| 13:43.57 | mut | did it using this? |
| 13:43.57 | mut | http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers |
| 13:44.15 | jaiger | you only need like 6 pins. power, ground, and TX/RX |
| 13:44.33 | jaiger | mut, yeah I've been watching that page |
| 13:44.45 | iCEBrkr | yo yo |
| 13:45.23 | jaiger | ok, maybe 10 pins. I did this 1.5 years ago |
| 13:46.24 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
| 13:46.28 | jaiger | mut, the most annoying thing was that my shelf is wire wrap and I couldn't find a wire wrap tool large enough for the pins. I ended up finding/buying some Molex connectors that I could modify to fit |
| 13:46.45 | *** join/#asterisk crich1999 (n=crich@p54BF99B1.dip0.t-ipconnect.de) |
| 13:48.06 | jaiger | took me a while to find the connectors |
| 13:48.44 | _Sam-- | [av]bani: you dont have any problems to report with the .26 firmware? |
| 13:49.05 | [av]bani | _Sam--: just the looping bug still |
| 13:49.11 | _Sam-- | same |
| 13:49.13 | *** join/#asterisk misty (n=misty@oh-65-40-78-243.sta.sprint-hsd.net) |
| 13:49.24 | [av]bani | oh, and ringtone volume is borked in the phone UI |
| 13:49.29 | queuetue | Hi. My cable company had to swap out a deceased cable modem/router in the last few days, and now I cannot call out to another sip phone outside of my NAT (and behind another NAT.) This config worked fine before, so I assume it is a port forwarding problem. He can call me fine (* inside my NAT) and I can call another SIP phone fine also behind my NAT, but If I call him, his phone rings, he answers, and there is just dead air unti |
| 13:49.31 | _Sam-- | i remotely upgraded some client phones, but i didnt know about the big issue with the display problems |
| 13:49.37 | _Sam-- | Important Information: |
| 13:49.41 | _Sam-- | Due to a firmware bug in 1.0.2.6 which causes display problems..... |
| 13:49.43 | misty | Good morning :) I am doing initial support with the thought of upgrading our dinosaur of a phone system with Asterisk |
| 13:50.12 | [av]bani | _Sam--: duped the bug then? |
| 13:50.13 | _Sam-- | they implemented, though, a feature i asked for |
| 13:50.28 | _Sam-- | the disable missed call log on screen |
| 13:50.28 | misty | I'm trying to find some info about whether our handsets may possibly interface with Asterisk |
| 13:50.36 | mut | jaigerL wire wrap? |
| 13:51.29 | jaiger | mut, some of the shelves are wire wrap |
| 13:51.47 | queuetue | misty: You'd have to give more information than that. |
| 13:52.19 | misty | of course I would :) I was just making sure it was the right place to ask. They are Toshiba DKT2010 phones. |
| 13:52.28 | *** join/#asterisk Defraz_ (n=t0tal@tim.mychoice.cc) |
| 13:52.40 | mut | ahve some pics? |
| 13:55.24 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 13:55.34 | jaiger | mut, it's not too clear but here's the back of my 255 shelf showing the wire wrap... http://magneto.innovationsw.com/~jaiger/images/Tellabs-255A.jpg |
| 13:55.48 | jaiger | taken w/ a phone camera |
| 13:56.18 | misty | It looks like no toshiba phones are voip capable |
| 13:58.20 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 14:00.00 | jaiger | mut, looks like the 2 card shelf would be nicer to work with |
| 14:00.07 | mut | yea |
| 14:00.14 | mut | where'd ya buy the power supply |
| 14:00.19 | [av]bani | jaiger: you wired up all the slots? |
| 14:00.29 | jaiger | [av]bani, no, just the one I need |
| 14:00.31 | misty | oh my word, this is confusing! Any links on choosing phones / boards to go with an asterisk solution? |
| 14:00.42 | jaiger | mut, watched ebay for months to collect all the pieces |
| 14:00.58 | mut | so it's a tellab supply? |
| 14:01.03 | jaiger | mut, yes |
| 14:01.42 | [av]bani | afaik you just need a 48v supply, which isnt too hard to find |
| 14:01.46 | mut | ya |
| 14:01.55 | [av]bani | lots of them around for telco equipment |
| 14:02.00 | jaiger | the docs on my shelf say 10A supply but I only have 1 card so I went with 1A and it seems to work fine |
| 14:02.18 | jaiger | I would go for a telco brand though |
| 14:02.35 | *** join/#asterisk CleanerX (n=nix@p54A3B16F.dip0.t-ipconnect.de) |
| 14:02.44 | mut | yea |
| 14:02.54 | jaiger | it puts off a lot of heat too. make sure you have cooling |
| 14:03.01 | mut | i have em for my dslams |
| 14:03.10 | mut | they take the same thing |
| 14:03.15 | *** join/#asterisk trelane_ (n=trelane@asterisk.sosdg.org) |
| 14:04.10 | queuetue | misty: Any SIP or IAX phone will work with asterisk - the best thing to do is read reviews and ask people that have specific models how they like them. I use sipura SPA2000 and am pretty happy with them. |
| 14:04.37 | mut | thot hey don't say how many amps they are |
| 14:04.39 | misty | I've a feeling looking at phones first is the wrong way around |
| 14:04.46 | mut | other than "no more than 20amps" |
| 14:05.00 | misty | we currently have 50 lines, I'd like to have 150 or 200 potential lines |
| 14:05.01 | [av]bani | spa2000 isnt a phone |
| 14:05.17 | queuetue | I had to replace a dead DSL modem/router in the last few days, and now when I call another sip phone outside of my NAT (and behind another NAT), I get no audio after the ring and pickup. This config worked fine before, so I assume it is a port forwarding problem. He can call me fine (* inside my NAT) and I can call another SIP phone fine also behind my NAT, but If I call him, his phone rings, he answers, and there is just dead ai |
| 14:05.23 | misty | how do I find the appropriate cards that provide that? not sure of the terminology |
| 14:05.53 | [av]bani | misty: youll end up spending about as much to interface phones with voip as you will just buying voip phones. and youll get far more functionality from a voip phone. |
| 14:06.08 | misty | I am sure you're right |
| 14:06.16 | misty | but first I have to see what I would put into my server :) |
| 14:06.46 | queuetue | spa2000 I'm not sure why you want to make the distinction, but yes, the SPA2000 is an adapter that lets me use any phone I want. We actually use GE 27977ge-3 cordless headsets, which we all love with the sipura. |
| 14:06.48 | mut | so like |
| 14:06.49 | mut | http://cgi.ebay.com/4-New-Tellabs-48v-24v-Power-Supplies-model-8035_W0QQitemZ7587000115QQcategoryZ36323QQrdZ1QQcmdZViewItem |
| 14:06.51 | mut | that'de be perfect eh |
| 14:07.01 | *** join/#asterisk Defraz_ (n=t0tal@tim.mychoice.cc) |
| 14:07.23 | KeX_WorX | how can I set callerid from within an agi script? |
| 14:07.47 | [av]bani | queuetue: spa2000 is a 'gateway' |
| 14:07.52 | [av]bani | or rather, an ATA |
| 14:07.55 | KeX_WorX | is this correct that I've to print SET CALLERID "name <nr>" to standart out? |
| 14:08.14 | misty | I can get a phone cheaper than that adaptor |
| 14:08.17 | [av]bani | queuetue: the distinction between a gateway/ata and a voip phone is HUGE. which is why i make the distinction |
| 14:08.20 | queuetue | [av]bani: Again, no clue why you want to make the distinction, but you are free to. |
| 14:09.54 | jaiger | mut, looks good to me |
| 14:10.38 | misty | ok what is the thing inside the computer called, for voip? it's called a voice board on our systems |
| 14:11.28 | [av]bani | a cpu? |
| 14:11.34 | misty | I don't know much at all about our current system, to give you an idea it is running on a 486 |
| 14:11.37 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:12.00 | misty | you don't have to have interface boards?? these are what the 25-pair plugs into I think |
| 14:12.16 | *** join/#asterisk coppice (n=chatzill@138.162.17.210.dyn.pacific.net.hk) |
| 14:12.54 | *** join/#asterisk zotz (n=zotz@24.244.133.10) |
| 14:13.06 | queuetue | misty: This may be a project you will have a hard time deploying on your own.... At least purchase the asterisk book before starting a 150 extension rollout. :) |
| 14:13.22 | *** join/#asterisk zotz (n=zotz@24.244.133.10) |
| 14:13.34 | misty | of course, I am just doing initial research now to learn the terminology etc |
| 14:13.44 | misty | it isn't something that will probably happen within the next year even |
| 14:14.23 | Modcuts | does anyone in here use the grandstream gxp2000? |
| 14:14.32 | queuetue | misty: OK, the basics are best learned via google (www.voip-info.org) , and then come here for specifics. No one likes repeating stuff that's very easy to learn with a web search. |
| 14:14.47 | misty | thank you |
| 14:14.49 | [av]bani | Modcuts: yes, _Sam-- and i do |
| 14:14.57 | *** join/#asterisk CoKane (n=CoKane@87.192.246.56) |
| 14:15.15 | *** part/#asterisk misty (n=misty@oh-65-40-78-243.sta.sprint-hsd.net) |
| 14:15.55 | queuetue | Modcuts: I'm planning to, but have not. |
| 14:16.10 | Modcuts | [av]bani: well have u ever got custom tones to work or have any better tones then the one it ships with? |
| 14:16.54 | [av]bani | yes, ive got them to work |
| 14:17.09 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 14:17.39 | CoKane | hy guys I installed the quadBRI PCI ISDN card from junghanns getting the following error "ZT_SPANCONFIG failed on span 1: No such device or address (6)" |
| 14:17.47 | CoKane | anyone experience this before |
| 14:17.50 | queuetue | Can I hook "normal" ethernet gear up to a POE system? |
| 14:18.21 | KeX_WorX | anyone uses asterisk 1.2 and rewrites the callerid od the calleridname? |
| 14:18.34 | KeX_WorX | I tried to do that, but without success : / |
| 14:18.41 | KeX_WorX | can anyone help me? |
| 14:19.25 | queuetue | RTP requires forwarding of ports 8766 to 35000? Isn't that a mite excessive? |
| 14:20.19 | saw | guys, is there something like an analog-to-ip box to connect analog devices to asterisk? |
| 14:20.38 | Ahrimanes | saw: ATA ? |
| 14:20.46 | saw | whats that? |
| 14:20.52 | [av]bani | queuetue: some phones let you choose the ports |
| 14:21.20 | [av]bani | saw: http://www.voip-info.org/wiki-ATA |
| 14:21.22 | Ahrimanes | saw: analogue telephone adapter |
| 14:21.59 | [av]bani | ~ATA |
| 14:22.05 | jbot | from memory, ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
| 14:22.28 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:23.08 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 14:23.08 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:23.55 | *** join/#asterisk fsa (i=fsa@221-128-182-98.exatt.net) |
| 14:25.14 | fsa | Hello everyone. I've a PBX system setup at my office and I just came across Asterisk. Can anyone tell me how can I benefit from it! |
| 14:25.49 | Ahrimanes | what a question |
| 14:25.55 | CaViCcHi | yeah! |
| 14:26.02 | cpm | If it isn't obvious, then perhaps you wouldn't benefit from t. |
| 14:26.43 | knobo | RoyK: searching for help. |
| 14:26.47 | fsa | wat hardware do I need to setup Asterik |
| 14:26.48 | coppice | fsa: is your PBX broken? |
| 14:27.05 | fsa | coppice: no my PBX system is working properly |
| 14:27.14 | knobo | <PROTECTED> |
| 14:27.25 | coppice | then it doesn't need fixing. seems you have no use for Asterisk |
| 14:28.06 | fsa | does Asterisk need any special hardware? if not I'd like to experiment with it and set it up at my home |
| 14:28.39 | Ahrimanes | fsa: basically no, but if you need to connect to pstn you will need some specific hardware |
| 14:28.50 | queuetue | What's the easiest way to test MOH with only 1 phone on hand? |
| 14:29.12 | fsa | I read that Asterisk supports VoiceMail and Custom Hold music which my current PBX system does not support |
| 14:29.22 | RoyK | knobo: #asterisk-no |
| 14:29.41 | Ahrimanes | fsa: true, http://www.voip-info.org/wiki/view/Asterisk will give you a good overview and voip-info.org is a great reference |
| 14:29.46 | CaViCcHi | it doesnt support? wow... |
| 14:29.49 | fsa | Ahrimanes: then how can I test it? |
| 14:29.53 | *** join/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 14:29.54 | queuetue | fsa: It does, and no special hardware is required - unless you want to hook it up to POTS lines. |
| 14:30.04 | skeffling | queuetue, something like exten = 234,2,MusicOnHold then dial 234 |
| 14:30.26 | Ahrimanes | fsa: you need to install it, there's something called asterisk@home which can help ease the testing, google for it |
| 14:30.32 | fsa | queuetue: wat are POTS lines? |
| 14:30.41 | Ahrimanes | ~pots |
| 14:30.43 | jbot | i heard pots is Plain Old Telephone Service as in "Old Analogue Crap" |
| 14:31.24 | fsa | I am currently reading Asterisk Wiki but was just anxious to know if I can install it without special hardware? |
| 14:31.26 | queuetue | fsa: Regular phone lines from the telco, unlike VOIP lines (which come over the Internet via DSL.) |
| 14:31.38 | *** join/#asterisk jan__ (n=jan@ip22.ds1-saen.adsl.cybercity.dk) |
| 14:32.23 | queuetue | fsa: You will need a computer that has a network card. You can smoke-test 98% of asterisk without any special hardware. When (if) you want to hook it into the "normal" phone system, you will need some hardware. |
| 14:32.38 | jan__ | What would you recommed as channel bank to a A104 card and E1? |
| 14:32.51 | Ahrimanes | fsa: but really, the wiki knows much more than most of us in here.. reading it is good |
| 14:33.13 | fsa | so I can setup a system thru which I can read caller number and can block it or do a specified task? |
| 14:33.41 | _Paulo_ | fsa, this is what * is for. |
| 14:34.12 | fsa | I've a network card and also internal modem. Will this suffice for now? |
| 14:34.24 | _Paulo_ | fsa, yes |
| 14:34.53 | Modcuts | [av]bani: how did you convert some or with sox then use tftp server on your comp to upload them on ours it uploads them but the tone don't change does it have to be a specific size? |
| 14:35.03 | Ahrimanes | fsa: well internal modem might not be much use |
| 14:35.08 | queuetue | fsa: Not if you wish to connect it to phone lines (unless you have a very specific modem.) If you simply wish to test asterisk with VOIP, etc then yes. |
| 14:35.48 | _Paulo_ | fsa, you can contract some Voip service... |
| 14:35.56 | fsa | thn what kind of modem do I need to have? |
| 14:36.03 | queuetue | fsa: Please go read just a little bit of the asterisk docs - one or two pages should suffice to answer every question you've asked so far. |
| 14:36.22 | fsa | I also have an external modem |
| 14:36.23 | *** join/#asterisk tzafrir_laptop (n=tzafrir@85-64-243-145.barak-online.net) |
| 14:36.35 | fsa | Thnkx guys for clearing my doubts. I'll go and read the Wiki now |
| 14:36.37 | _Paulo_ | fsa, go install * |
| 14:36.47 | _Paulo_ | :-) |
| 14:37.01 | fsa | Thnkx guys |
| 14:37.05 | tzafrir_laptop | Am I the only one who had problems with the missing .version file in the zaptel tarball? |
| 14:37.22 | *** join/#asterisk bails (n=bails@81.168.76.189) |
| 14:37.44 | bails | anyone in here using a@home 1.5 |
| 14:38.17 | Bambr | how do i write to extensions table what used to be include directive in file extensions.conf? |
| 14:38.44 | bails | cos i just did a ls -lah /var/mail/admin and found out its 2gb? |
| 14:39.29 | bails | if i remove it and touch it in again i can see it growing in size in massive amounts |
| 14:39.43 | bails | till in under a minute its 2gb again |
| 14:40.02 | iCEBrkr | bails: Ok? So open those emails up and read the contents |
| 14:40.10 | bails | LOL oh yeh |
| 14:40.12 | wunderkin | thats what i was thinking.. |
| 14:40.28 | bails | it appears to be an mplayer thing |
| 14:40.43 | iCEBrkr | ~a@h |
| 14:40.53 | iCEBrkr | >: | |
| 14:40.59 | iCEBrkr | ~asterisk@home |
| 14:41.01 | jbot | asterisk@home is, like, http://asteriskathome.sourceforge.net/, or http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home |
| 14:41.09 | iCEBrkr | hrrm. |
| 14:42.10 | bails | ok I'll p**s off there then |
| 14:42.13 | queuetue | How do I set asterisk up to restart when all calls are completed? |
| 14:42.36 | Bambr | is here a person who knows well that database stored configuration stuff? |
| 14:42.47 | queuetue | "restart when convenient" ? |
| 14:42.48 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 14:42.50 | Bambr | a.k.a. realtime conf |
| 14:42.54 | _Paulo_ | besides app_txfax is there any other application to send a fax trhough asterisk? |
| 14:42.59 | iCEBrkr | queuetue: eh? |
| 14:43.27 | queuetue | iCEBrkr: I'm asking if the command "restart when convenient" will restart when all calls are completed. |
| 14:44.28 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 14:45.05 | Ahrimanes | queuetue: that would be a yes |
| 14:45.19 | iCEBrkr | queuetue: If that's what the description says... |
| 14:45.40 | queuetue | iCEBrkr: thanks for the not-quite-help. |
| 14:45.54 | iCEBrkr | queuetue: Umm. that's like asking if "ls" will list files. |
| 14:46.05 | iCEBrkr | <PROTECTED> |
| 14:46.43 | queuetue | iCEBrkr: I suppose you were born with the knowledge of what "empty call volume" was, but some of us non-savants might ask for clarity from time to time. |
| 14:46.58 | iCEBrkr | It just makes sense.. |
| 14:47.04 | coppice | _Paulo_ with iaxmodem you can use HylaFAX |
| 14:47.25 | iCEBrkr | queuetue: ...and when all else fails.. Experiment. |
| 14:47.35 | queuetue | It sure does, once you know what it means. Which means someone told you once - probably someone less snotty. |
| 14:48.00 | iCEBrkr | queuetue: No, I'm just able to guess good. |
| 14:48.10 | queuetue | iCEBrkr: I would prefer to not cancel any sales calls because some dude on irc wants to feel superior. Thanks. |
| 14:48.15 | iCEBrkr | TRIAL AND ERROR! EXPERIMENT AND SEE |
| 14:48.46 | iCEBrkr | queuetue: LOL! Maybe you shouldn't be on IRC asking for help on a production box. |
| 14:48.46 | Bambr | is here a person who knows well that database stored configuration stuff? |
| 14:49.06 | queuetue | iCEBrkr: Once gain, thanks for not-quite-helping. |
| 14:49.12 | iCEBrkr | Sure! No problem! :) |
| 14:49.20 | *** join/#asterisk TonyM_ (n=TonyM@adsl-solo-80-168-225-214.claranet.co.uk) |
| 14:49.39 | *** join/#asterisk donnib (n=aaa@0x555281d0.adsl.cybercity.dk) |
| 14:49.46 | tzafrir_laptop | queuetue, yes, only when there are no active calls |
| 14:50.22 | queuetue | tzafrir_laptop: Thank you. (Ahrimanes Already answered me, though - I just forgot to thank him. :) ) |
| 14:50.42 | donnib | hi all |
| 14:50.53 | _Paulo_ | coppice, thanks. |
| 14:51.07 | donnib | maybe somebody can help me out with a SIP registry which doesn't show up under sip show registry at all |
| 14:51.17 | tzafrir_laptop | again: nobody had a problem building zaptel 1.2.2/1.2.3 from a clean tarball due to an empty version number? |
| 14:51.26 | iCEBrkr | donnib: Does it show up under 'sip show peers' |
| 14:51.27 | iCEBrkr | ? |
| 14:51.46 | donnib | yes it does |
| 14:51.53 | iCEBrkr | There ya go :P |
| 14:52.11 | donnib | well i don't get it |
| 14:52.24 | donnib | it should be under registration or ? |
| 14:52.37 | iCEBrkr | donnib: I assume this is a phone you're registering? |
| 14:52.52 | donnib | no. it's my VoIP provider |
| 14:53.05 | iCEBrkr | oh. |
| 14:53.21 | *** join/#asterisk fhqLn (n=seth2o@heim-032-176.raab-heim.uni-linz.ac.at) |
| 14:53.22 | *** join/#asterisk chungenw\ (n=Kbchinpa@85.102.231.56) |
| 14:53.30 | *** join/#asterisk sinavQ (n=Zclaudia@85.96.154.22) |
| 14:53.30 | *** join/#asterisk soner9t (n=_matarya@211.41.223.225) |
| 14:53.33 | *** join/#asterisk diane27s (n=ynbirget@85.103.74.37) |
| 14:53.35 | *** join/#asterisk arash36m (n=o_erdem@85.103.33.54) |
| 14:53.37 | iCEBrkr | donnib: then you should have a register => line in your sip.conf for it.. |
| 14:53.40 | *** join/#asterisk pdewayneg (n=kxneal@85.106.187.20) |
| 14:53.50 | donnib | i did that already |
| 14:53.53 | *** join/#asterisk curtvU (n=cathisX@85.98.149.150) |
| 14:53.59 | donnib | but still nothing happends |
| 14:53.59 | iCEBrkr | donnib: Ok, yea, that's kinda odd |
| 14:54.00 | *** join/#asterisk ^salone^ (n=ronen40l@188025.uninet.lv) |
| 14:54.06 | *** join/#asterisk fIiper38A (n=tcarrol_@85.96.219.181) |
| 14:54.11 | donnib | i can ping out everything looks ok |
| 14:54.12 | *** join/#asterisk ramchandra6a (n=fagnese@p54956EAF.dip.t-dialin.net) |
| 14:54.13 | *** join/#asterisk AdianneY (n=_Nabhira@213.197.129.54) |
| 14:54.13 | *** join/#asterisk Esguitar (n=changkyu@85.104.156.181) |
| 14:54.26 | *** join/#asterisk OdieterM (n=Bmtichel@85.102.147.84) |
| 14:54.27 | donnib | i have put sip debug and verbose 9 and debug 10 but no information at all in the CLI |
| 14:54.29 | iCEBrkr | donnib: It shows up in 'peers' but not 'registry'? |
| 14:54.34 | donnib | yup |
| 14:54.35 | Ahrimanes | queuetue: yes, please remember to do so ;) |
| 14:54.46 | iCEBrkr | Bambr: Please don't /msg me. |
| 14:54.50 | _Paulo_ | coppice, IAXModem uses libspandsp also? |
| 14:54.54 | wunderkin | did you add the register line after you started asterisk, or did a reload? |
| 14:54.56 | Bambr | nobody answers me anyway |
| 14:54.59 | Bambr | :( |
| 14:55.02 | donnib | i did a reload |
| 14:55.05 | donnib | nothing happend |
| 14:55.09 | iCEBrkr | Bambr: Cuz not many people use realtime. |
| 14:55.23 | Bambr | but still, somebody do :) |
| 14:55.26 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 14:55.49 | donnib | pretty strange.... |
| 14:56.01 | *** join/#asterisk CleanerX1 (n=nix@p54A3B16F.dip0.t-ipconnect.de) |
| 14:56.20 | wunderkin | donnib, make sure that the line isnt commented out with a ; and try a sip reload |
| 14:56.40 | donnib | it's not :) |
| 14:56.44 | iCEBrkr | ha |
| 14:57.27 | donnib | i've put the line in before my provider options and it's not commented out |
| 14:57.43 | donnib | evrything is at the end of the sip.conf |
| 14:57.59 | donnib | must be something i am missing |
| 14:58.01 | iCEBrkr | donnib: your register => line is in [general] right? |
| 14:58.19 | *** join/#asterisk selin9L (n=beyda1_@85.106.225.123) |
| 14:58.44 | wunderkin | doesnt sound like it |
| 14:59.00 | donnib | it's afeter the [general]. isn't that ok ? |
| 14:59.01 | *** join/#asterisk chan5z (n=sabrina3@85.106.225.123) |
| 14:59.03 | FuriousGeorge | hey all |
| 14:59.08 | donnib | there is alot of stuff between it |
| 14:59.18 | FuriousGeorge | so one of my voicemailboxes' greetings broke |
| 15:00.20 | donnib | actually i see a [authenticatication] before my settings |
| 15:00.24 | donnib | is that ok ? |
| 15:00.52 | queuetue | Ahrimanes: Yes, sir! :) |
| 15:01.13 | Ahrimanes | queuetue: :) |
| 15:01.47 | FuriousGeorge | and in the "advanced options" the option to set greetings or listen to them is non existant |
| 15:02.25 | iCEBrkr | donnib: That's a problem |
| 15:03.05 | mut | anyone used an Airspan softswitch? |
| 15:04.47 | queuetue | Can anyone give the rough legal requirements to playing commercial music as music on hold? I assume owning the CD isn't quite enough... Is getting proper permission impossible? Impractically expensive? |
| 15:04.50 | donnib | thanx. i got it fixed |
| 15:05.19 | donnib | sip debug no |
| 15:05.25 | donnib | ups :) |
| 15:05.33 | iCEBrkr | donnib: wrong window :P |
| 15:06.00 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:06.15 | hypnox | queuetue no idea but why not look for some creativecommons licensed music, there's lots |
| 15:06.25 | cpm | queuetue, Roughly, you can't do it. |
| 15:06.37 | queuetue | hypnox: In my (limited) experience, it's mostly junk, as well. :) |
| 15:07.03 | cpm | It's doable, but the headaches of getting the 'mechanical reproduction' licenses are awful. Are you in the US? |
| 15:07.29 | queuetue | cpm: I'm a US citizen, but reside (as does the server) in Quebec. |
| 15:08.16 | *** part/#asterisk fsa (i=fsa@221-128-182-98.exatt.net) |
| 15:08.22 | cpm | Ca,US are on the same page riaa/copyright wise. So no help there. hypnox's suggestion is a good one. |
| 15:08.36 | *** join/#asterisk coppice (n=chatzill@197.197.17.210.dyn.pacific.net.hk) |
| 15:09.12 | queuetue | Straying off-topic here, but is there a CC music rating system anywhere? |
| 15:09.49 | donnib | is reload not enough to re-read the config files ? |
| 15:10.08 | znoG | so music from www.sounddogs.com can legally be played in MOH? |
| 15:10.10 | SparFux | Q: How to determine, wether a channel has been hung up by the peer? |
| 15:10.45 | tzafrir_laptop | queuetue, it may be some sort of music redistribution or public playing. Depending on the type of the PBX |
| 15:10.49 | *** join/#asterisk gvag11 (n=gvag11@ipa51.4.tellas.gr) |
| 15:11.06 | gvag11 | Hi all... |
| 15:11.21 | tzafrir_laptop | Anyway, you can get some free classical music |
| 15:11.30 | cpm | znoG, Read the sounddogs page, "These music tracks are owned by Sounddogs.com, Inc., or were produced and made available under license. All rights reserved. Unauthorized duplication is a violation of applicable federal and state laws of the United States and international treaties." |
| 15:11.54 | cpm | So, you need to license the work from them to reproduce it. |
| 15:13.06 | mdave | anyone know of a *good* tutorial on spa-2000 dialplan syntax ? |
| 15:14.20 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0dh.dialup.mindspring.com) |
| 15:15.05 | *** part/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:15.43 | znoG | mdave: search google for the spa2k manual, it has the dialplan syntax |
| 15:15.50 | znoG | too bad it doesn't support regexs |
| 15:16.58 | Falle | How do i configure a queue to play ringing tones to the client instead of MoH? |
| 15:17.11 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-214.claranet.co.uk) |
| 15:17.13 | mdave | yeah I was just looking for that at siprua's site and not finding it |
| 15:17.15 | gvag11 | Coppice: i am trying to set the Head info for TxFax, and before the txfax command, i set LOCALHEADERINFO="BLABLABLABLABLABLA" |
| 15:17.30 | queuetue | mdave: Like "jingle bells" in DTMF tones? :) |
| 15:17.34 | gvag11 | but it doesn't work |
| 15:17.35 | mdave | all they have on the 2000 is a 'quickstart' guide |
| 15:17.44 | *** join/#asterisk CleanerX (n=nix@p54A3B16F.dip0.t-ipconnect.de) |
| 15:17.45 | iCEBrkr | mdave: Oh, the PDF is up on Sipura's site. |
| 15:17.45 | znoG | mdave: no, there's a user guide with that info |
| 15:17.51 | queuetue | Falle: Meant for you, not mdave . |
| 15:17.53 | mdave | queuetue, i can play 'marry has a little lamb' tho |
| 15:18.03 | znoG | it really is a shame it doesn't support regex (the dialplan syntax) |
| 15:18.11 | znoG | makes writing a dialplan a little harder |
| 15:18.15 | mdave | iCEBrkr, well its not under 'support', at least as far as i can see |
| 15:18.25 | iCEBrkr | mdave: Ya gotta dig for it.. It's kinda lost on their page. |
| 15:18.28 | Falle | queuetue: hehe, no.. just normal ringtones like when you call the PSTN :) |
| 15:18.38 | iCEBrkr | mdave: I had the same problem |
| 15:18.43 | mdave | ugh.. and even that is in some proprietary unknown format |
| 15:18.47 | iCEBrkr | znoG: I dunno if it's 'harder' |
| 15:19.04 | iCEBrkr | znoG: Cuz people like me don't know regexp off the top of their head, so the current method is quite simple :P |
| 15:19.09 | wunderkin | Falle: show application queue |
| 15:19.41 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-89.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 15:20.56 | Falle | wunderkin: hmm, that works too i guess :) But i was looking for a queues.conf solution. |
| 15:20.58 | gvag11 | Coppice: i am trying to set the Head info for TxFax, and before the txfax command, i set LOCALHEADERINFO="BLABLABLABLABLABLA" but it doesn't work... Any idea? |
| 15:21.36 | znoG | iCEBrkr: oh, yea, but they could "support" it, doesn't mean people have to necessarily use it :) |
| 15:21.51 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:22.14 | *** join/#asterisk TonyM_ (n=TonyM@adsl-solo-80-168-225-214.claranet.co.uk) |
| 15:22.20 | iCEBrkr | znoG: It's an embeded system, cramming all that stuff in there?? :P |
| 15:22.47 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:23.32 | znoG | iCEBrkr: aww it wouldn't be that much :) |
| 15:23.38 | coppice | gvag11: that should be right. are you sure it is spelled correctly? |
| 15:23.40 | iCEBrkr | haha |
| 15:24.48 | gvag11 | coppice i check that 100 times ... first Set(LOCALHEADERINFO="bldslkfdkg") and then txfax(${FILE}|caller) but it doesn't work ... Any idea ? |
| 15:24.49 | rustyb | queuetue: you want to custom ringtones to the caller? |
| 15:24.57 | *** join/#asterisk bigjb (n=nnnbigjb@195.60.10.114) |
| 15:25.00 | rustyb | send* |
| 15:25.07 | queuetue | rustyb: Not me, Falle |
| 15:25.29 | rustyb | Falle: it's easy |
| 15:25.40 | *** join/#asterisk Modcuts (n=info@ppwood.gotadsl.co.uk) |
| 15:27.47 | *** part/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 15:28.19 | madounet | Hi, is there someone using V.22 from spandsp with * ? |
| 15:30.30 | *** join/#asterisk secure75 (n=mic@host-82-135-30-151.customer.m-online.net) |
| 15:31.51 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 15:33.56 | coppice | I doubt it. its not finished |
| 15:34.49 | Modcuts | [av]bani : What did you use to create the files and do that have to be a certain size to work? |
| 15:34.52 | *** join/#asterisk [Atlas] (n=whois@216.190.144.90) |
| 15:35.17 | [Atlas] | anyone know where i can get a free sip or iax account for testing or one that accepts paypal? |
| 15:35.28 | madounet | coppice, is it in the spandsp roadmap ? |
| 15:35.35 | *** join/#asterisk santiago (n=santiago@63.245.86.215) |
| 15:36.10 | coppice | well, sort of. |
| 15:36.35 | queuetue | [Atlas]: Free World Dialup |
| 15:36.37 | *** part/#asterisk santiago (n=santiago@63.245.86.215) |
| 15:37.09 | queuetue | [Atlas]: You could also always call yourself to test. |
| 15:37.47 | [Atlas] | so i would register myself as a peer of myslef? |
| 15:38.05 | queuetue | [Atlas]: I never have, but it should be possible. |
| 15:38.40 | FuriousGeorge | is it possible my tinkering in queue.conf is responsible for my voicemail greeting having disappeared and the inability to set a new one (the option isnt there) ive tried restarting asteriskik and reloading music on hold |
| 15:38.56 | [Atlas] | and FWD will let me connect to it with my asterisk system? |
| 15:39.10 | [Atlas] | sorry to sound like a n00b its just that i am one ;p |
| 15:39.12 | *** join/#asterisk Bambr (n=Bambr@213-35-237-161-dsl.end.estpak.ee) |
| 15:39.27 | queuetue | [Atlas]: yes. |
| 15:39.55 | [Atlas] | thanks! |
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| 15:41.08 | *** join/#asterisk CleanerX1 (n=nix@p54A3B16F.dip0.t-ipconnect.de) |
| 15:42.11 | Skkip | atlas: ipkall.com as well |
| 15:45.35 | FuriousGeorge | this is kinda serious. when someone calls they get the temp (default) message (just allsion saying "please leave a message..."), and, besides the fact that it tookus a while to get hte original message just right, there appears to be now way to reset it |
| 15:45.53 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc0dh.dialup.mindspring.com) |
| 15:46.15 | Math` | FuriousGeorge: you want to change it? |
| 15:47.43 | *** join/#asterisk swb (n=swb@cornelyn.force9.co.uk) |
| 15:48.41 | swb | Hullo |
| 15:48.46 | EriSan | hi, can someone take a look at http://asterisk.pastebin.com/541718 and see what could be wrong ? |
| 15:49.11 | FuriousGeorge | Math`: we used to be able to go to "advanced options" after authenticating and set busy and away messages |
| 15:49.17 | FuriousGeorge | i assume we are still supposed to be able to |
| 15:49.29 | FuriousGeorge | i dont know exactly whe n this stopped working |
| 15:50.31 | swb | I have a question about the nature of execution in a macro |
| 15:50.32 | FuriousGeorge | i see a 56 meg wav in the dir, which must be the greeting we recorded, date seems right |
| 15:51.19 | Math` | works here.. |
| 15:51.42 | swb | I understnad that when a macro finishes executing it goes back to whre it was called from, providing ${MACRO_OFFSET} wasnt set, I also understand that when you use a Goto from a macro executino of the macro ends there and execution continues at whichever extension/priority you did Goto to |
| 15:51.56 | swb | my question is this: If you do a GoSub from a Macro |
| 15:52.25 | swb | does the execution come back into thre Macro when the sub routine is finished, and will it then go back to the original context when the Macro subsequently finishes? |
| 15:53.37 | _Sam-- | bani you there? |
| 15:53.45 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-216-21.rockynet.com) |
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| 15:55.34 | *** join/#asterisk outIook25A (n=saifalla@85.96.123.247) |
| 15:57.43 | swb | Anyone used Gosub() in a Macro? |
| 15:58.17 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 15:58.18 | kippi | hi |
| 15:58.53 | *** part/#asterisk jefrey (n=jefrey@202.190.203.200) |
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| 16:02.36 | _Paulo_ | swb, this kind of question is easier to find out by trial/error than to type an answer here |
| 16:03.17 | swb | _Paulo_, I guess that |
| 16:03.22 | swb | trialling now :P |
| 16:04.51 | Alric | Anyone have the problem with IP phones where the MWI comes on before the caller is done leaving a message? |
| 16:05.22 | rajiv | how can i have outbound calls fail over from an iax channel to a zap channel ? i have Dial(IAX/..) then Dial(Zap/..) but if the called party is busy, * tries to dial twice |
| 16:05.29 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-214.claranet.co.uk) |
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| 16:08.18 | kippi | on the voicemail, when you recorde your own message to play to people before they leave you a message, after your message has played you get the lady saying please leave a message after the beep, is there away to remove the ladys voice and just have your recording? |
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| 16:08.55 | *** mode/#asterisk [+b *!*@81.214.158.251] by kram |
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| 16:09.01 | *** kick/#asterisk [honey10^!n=mark@pdpc/sponsor/digium/kram] by kram ((*@81.214.158.251) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 16:09.01 | *** kick/#asterisk [chien36M!n=mark@pdpc/sponsor/digium/kram] by kram ((*@81.214.158.251) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 16:09.01 | *** kick/#asterisk [pinkyx!n=mark@pdpc/sponsor/digium/kram] by kram ((*@81.214.158.251) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 16:09.01 | *** kick/#asterisk [dshleeo!n=mark@pdpc/sponsor/digium/kram] by kram ((*@81.214.158.251) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 16:09.02 | *** join/#asterisk mahmutQ^ (n=XbrendaD@81.213.136.137) |
| 16:09.02 | *** join/#asterisk ahmed27_ (n=celil13c@81.215.61.130) |
| 16:09.03 | *** kick/#asterisk [edawnm!n=mark@pdpc/sponsor/digium/kram] by kram ((*@81.214.158.251) CLONEBOTS DETECTED.. AUTO KICK ACTIVATED) |
| 16:09.04 | *** join/#asterisk aviNL (n=KTderrek@81.213.169.185) |
| 16:09.05 | *** join/#asterisk GparadiseH (i=DhqEa@196.217.144.78) |
| 16:09.06 | rajiv | 'show application VoiceMail' says how. use 's' |
| 16:09.09 | *** join/#asterisk seIda31w (n=ScooI_@85.106.178.231) |
| 16:09.13 | *** join/#asterisk bryn1i (n=alcZa@85.100.73.63) |
| 16:09.17 | twisted[asteria] | WOOHOO |
| 16:09.20 | *** join/#asterisk emeksizfX (n=ban27t@85.108.105.97) |
| 16:09.23 | *** join/#asterisk ZWnJz (n=_wsamuel@88.224.81.16) |
| 16:09.24 | *** join/#asterisk stamos4W (n=AdaqingK@81.214.227.229) |
| 16:09.31 | *** join/#asterisk mmagamenr (i=phil4|@adsl196-78-144-217-196.adsl196-13.iam.net.ma) |
| 16:09.31 | docelm0 | nice |
| 16:09.32 | *** join/#asterisk youhanseCX (n=alayne_S@81.215.31.108) |
| 16:09.32 | *** join/#asterisk alderek (n=miuKi@85.99.210.69) |
| 16:09.33 | *** join/#asterisk dhan-na1G (n=nmaryamc@85.108.90.248) |
| 16:09.34 | *** join/#asterisk darrenkw (n=QgoIdO@85.103.53.28) |
| 16:09.36 | *** join/#asterisk pakhil^ (n=nina3_@85.102.123.240) |
| 16:09.40 | *** join/#asterisk pingVI (n=balas9_@85.101.95.186) |
| 16:09.44 | *** join/#asterisk tVxHd (n=santosD@85.108.84.47) |
| 16:09.44 | *** join/#asterisk \ZeMl (n=QGpCa@212.156.168.230) |
| 16:09.47 | *** mode/#asterisk [+r] by twisted[asteria] |
| 16:09.50 | twisted[asteria] | i hate to do that. |
| 16:09.55 | docelm0 | What's +r? |
| 16:09.58 | *** mode/#asterisk [+r] by twisted[asteria] |
| 16:10.10 | twisted[asteria] | forces you to be registered with nickserv to join |
| 16:10.11 | docelm0 | registered users only? |
| 16:10.28 | twisted[asteria] | takes care of most of the spambots |
| 16:10.29 | docelm0 | ya.. That does suck but if it keeps the channel asshole free then I will take the extra step.. |
| 16:10.46 | *** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4163113.sympatico.ca) |
| 16:10.47 | docelm0 | until someone does a script work around |
| 16:10.47 | kippi | rajiv: was your message show app.. to me? |
| 16:10.57 | rajiv | kippi: yes |
| 16:10.59 | *** join/#asterisk eivindtr (n=wingnut-@062016241059.customer.alfanett.no) |
| 16:11.01 | kippi | ok cool |
| 16:11.14 | rajiv | kippi: you want something like Voicemail(s3000) |
| 16:11.19 | kippi | ah |
| 16:11.32 | rajiv | s for skip instructions |
| 16:11.48 | twisted[asteria] | i never would have noticed had kram's spambot kicker not started up and growl informed me |
| 16:13.51 | mdave | anyone have any idea how you signup or even contact voxbone.com? |
| 16:14.00 | mdave | their site seems to have no contact info, no signup, nothing |
| 16:14.35 | rajiv | mdave: there is a signup link on the left side |
| 16:14.57 | mdave | ? |
| 16:15.01 | mdave | theres nothing on the left side |
| 16:15.07 | mdave | you are talking at http://voxbone.com/home.jsf ? |
| 16:15.14 | rajiv | http://www.voxbone.com/register.jsf |
| 16:15.19 | mdave | weird |
| 16:15.21 | mdave | thanks |
| 16:16.37 | rajiv | how much do they charge ? |
| 16:16.45 | mdave | oh cripe.. i had disabled javascript becuase some site was doing something stupid, and apparently the menu is completely invisible with html |
| 16:16.46 | mdave | stupid |
| 16:16.48 | mdave | absolutely stupid |
| 16:16.52 | mdave | anybrowser.org |
| 16:18.11 | rajiv | mdave: if you sign upp, let me know how much 1 did from their nyc pop to a US # is |
| 16:18.33 | synthetiq | a sip response 400 "Invalid From" usually indicates what? |
| 16:21.07 | mdave | ugh.. their did numbers in the us are sorted first by city |
| 16:21.11 | mdave | rather than state or arecode |
| 16:21.13 | mdave | what a royal pia |
| 16:21.33 | mdave | and you cant even display them all at once and search, it breaks them into 5 pages |
| 16:21.48 | mdave | pretty monstrous, as far as UI goes |
| 16:22.01 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 16:22.24 | mdave | and as near as I can tell they dont have anything in my areacode, and even if they did all i could do is pick an areacode, i dont see any option to see available xchanges within an ac |
| 16:22.33 | mdave | and it looks like they charge $9 to setup nd 7.50 a month |
| 16:22.43 | mdave | for US numbers |
| 16:22.47 | mdave | dunno if they charge minutes or not |
| 16:23.17 | mdave | just for grins ill pick one randomly and see if it offers that on a second page |
| 16:23.55 | mdave | yeah, apparently you gotta pay before you even find out the exchange, let alone get to pick it |
| 16:24.09 | mdave | but since they dont even have my ac, it dont matter to me anyway |
| 16:24.38 | oogle | does anyone have experience with Voip Reach? Are they reliable and non-evil? |
| 16:24.58 | _Sam-- | hey file how long does it take for a toll free port? |
| 16:25.18 | mdave | also, i would think, being in the internet telephony market, they would have an inbound sip number, fwd, something |
| 16:25.32 | Falle | hmm, another quetion here.. Callprogress dont show up in the CLI anymore even thogh verbose is set to 5. What do i need to do after upgrading to 1.2.* for this to work? |
| 16:25.41 | mdave | they dont even give a phone # on their site |
| 16:25.53 | mdave | whois gives a number, but its not listed in fwdout, at the very least |
| 16:26.01 | mdave | er |
| 16:26.01 | mdave | wait |
| 16:26.04 | MikeJ[Laptop] | _Sam--, 2-3 weeks to be safe... extra $25 gets it expedited to be within a week |
| 16:26.08 | mdave | scratch that, the same # is on the contact page |
| 16:26.24 | _Sam-- | thats isquick really |
| 16:26.31 | _Sam-- | i think i may to fill out the LOA |
| 16:26.39 | _Sam-- | do you have an LOA on asterlink anyplace? |
| 16:26.52 | MikeJ[Laptop] | ummmm |
| 16:26.52 | *** join/#asterisk Cinen (n=Cinen@vpn.triadtelecom.com) |
| 16:26.53 | file | http://www.asterlink.com/transfer.doc |
| 16:26.59 | _Sam-- | ty |
| 16:27.01 | MikeJ[Laptop] | yes.. there ^^^^^ |
| 16:27.03 | MikeJ[Laptop] | heh |
| 16:30.16 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:30.49 | *** join/#asterisk pjz (n=pj@66.219.59.183) |
| 16:31.24 | pjz | anyone know why my polycom 500s have an initial 2s pause before you can hear anything once outgoing calls are connected? |
| 16:31.40 | pjz | or, more importantly, how to fix it? |
| 16:35.29 | *** join/#asterisk coppice (n=chatzill@87.155.17.210.dyn.pacific.net.hk) |
| 16:36.46 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 16:36.58 | websae | what's a good cisco interface to connect a PRI to? |
| 16:37.18 | _Sam-- | hey file does a toll free port have to have a ring to number? i think my toll free stands alone |
| 16:37.28 | _Sam-- | like it doesnt ring to another number |
| 16:37.30 | file | nah |
| 16:38.02 | file | just need a good bill copy |
| 16:38.06 | websae | any suggestions for a PRI gateway, perhaps by cisco? |
| 16:38.45 | _Sam-- | fair enough...im giving the teliax guy the benefit of the doubt still...its not looking promising for him. |
| 16:39.13 | file | _Sam--: did you find your routing issue? |
| 16:39.18 | xachen | heh |
| 16:39.23 | xachen | Teliax is still billng me |
| 16:39.26 | xachen | even though I cancelled |
| 16:39.29 | _Sam-- | yeah, sorry for your headache...it was 2 hops out of my upstream provider |
| 16:39.33 | xachen | well requested a cancel + refund |
| 16:39.34 | xachen | assholes |
| 16:39.39 | GerbilWrk | Teliax support took a week to answer my e-mail |
| 16:40.14 | *** join/#asterisk ahattar (n=ahattar@static-68-236-175-229.ny325.east.verizon.net) |
| 16:40.15 | _Sam-- | they do have a good intentions |
| 16:40.21 | GerbilWrk | xachen, why did you cancel with them? |
| 16:41.22 | xachen | They ingored my tickets and closed them a week later with no reponse |
| 16:41.22 | xachen | at least a reply would have been nice |
| 16:41.23 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 16:41.43 | *** join/#asterisk Garak_ (n=garak@209.5.171.170) |
| 16:42.35 | sevard | I'm running asterisk@home and an application I need only supports the 2.4 kernel, being a slackware user I'm really unsure how to use this CentOS deal.. Is there a "download/compile new kernel application", there seems to be an application for everything on here |
| 16:42.48 | Garak_ | I got a TDM400P with two FXS modules, I can call the second line but no audio is being passed, I can call the console from either and audio passes, I'm also not hearing the demo from the phones |
| 16:42.52 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 16:43.03 | _Sam-- | GerbilWrk: did you cancel teliax? |
| 16:43.16 | file | well, I'm always here since I'm a geek... |
| 16:43.17 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 16:43.38 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 16:43.59 | Garak_ | Any ideas on what I might have done wrong |
| 16:44.12 | iDunno | too much alcohol at lunchtime? |
| 16:44.32 | file | cosmic rays |
| 16:45.04 | iDunno | martians. |
| 16:45.05 | sevard | anyone? |
| 16:45.21 | websae | PRI gateways/routers--any suggestions? |
| 16:45.34 | _Sam-- | xachen : did you try contacting teliax about your refund, they are usually pretty good with me |
| 16:45.35 | GerbilWrk | _Sam--, no we just got an 800 number and corporate account from them |
| 16:45.51 | iDunno | sevard: sorry, have you checked to see if there's a 2.4 kernel in the yum repository? |
| 16:46.01 | iDunno | yum search kernel |
| 16:46.41 | sevard | iDunno: checking, sorry.. package management systems are beoyond me. |
| 16:46.46 | sevard | beyond* |
| 16:47.00 | iDunno | erm, slakware has one, they're very handy things. |
| 16:47.28 | sevard | sure, installpkg, not this fancy "hey application, give me this and install it while i take a crap" |
| 16:47.42 | _Sam-- | [av]bani : you around? |
| 16:48.42 | tzafrir_laptop | sevard, if you're a slackware guy, simply install asterisk yourself on slackware from source |
| 16:48.49 | sevard | iDunno: It doesn't seem to have one. |
| 16:48.57 | tzafrir_laptop | Asterisk@Home uses no package management whatsoever |
| 16:49.07 | GeneG | Question: has IAX2 native bridging through a NAT been improved from 1.0.10 to 1.2? With 1.0.10 Asterisk isn't native bridging my IAX2 connection and I suspect it's sending an internal IP instead of the external to the outside party. I can provide a IAX2 debug trace if anyone things they can help? |
| 16:49.45 | sevard | Tzafrir_laptop: I did have it on slackware but I didn't understand enough about asterisk to having it totally working, and since docs wern't helping me and everyone says RTFM i decided to take a look at asterisk @ home |
| 16:49.56 | *** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 16:50.39 | Garak_ | Any hints on why an FXS interface on a TDM400P might be sending voice but not receiving |
| 16:52.53 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 16:53.04 | mdave | any idea where nufone.net's list of did prefixes are? |
| 16:53.28 | mdave | i tried calling them, got voicemail.. you'd think that a business that was legit would answer their phone during business hours |
| 16:53.57 | mdave | i suspect they are a fly-by-night operating out of their garage, when a company doesnt have anyone avaiable to answer their business phone number |
| 16:54.06 | mdave | during normal daytime business hours |
| 16:54.27 | tronix | think it's an one person operation, not sure. they've been around for at least two years? more? not sure. |
| 16:54.42 | tronix | I don't know exact prefixes but they have at least 866 and 989 DIDs |
| 16:54.48 | tronix | (989 = michigan) |
| 16:55.13 | stack_ | I'm setting up a queue where the members have different penalties with the idea that after the queue dials the lowest priority people, it dials the next highest priority... this doesn't seem to work and it just bounces between the members with the lowest priority... am I doing something wrong? |
| 16:55.58 | mdave | also sems very suspicious when the first thing their 'signup' does is want to take yur money, without even a 'select what you want, and choose options' |
| 16:56.12 | mdave | I was hoping to find 616 |
| 16:56.46 | rajiv | mdave: they have an xls to download but it lists oonly area codes not prefix |
| 16:57.15 | *** join/#asterisk Alric (n=nbowyer@l-69-148-121-117.stu.swau.edu) |
| 16:57.16 | mdave | it boggles me why the hell people put stuff like that in secret proprietary formats instead of plain ascii text |
| 16:57.43 | mdave | even the crap software that makes that format is capable of making a text file |
| 16:57.49 | *** join/#asterisk buxy (n=nnnnnnra@arrakeen.ouaza.com) |
| 16:58.04 | steve___ | crap_software(tm) |
| 16:58.05 | steve___ | :) |
| 16:58.10 | mdave | rajiv, but I suppose I can run strings on it, if you can link me to it, i dont see it linked anywhere on their site |
| 16:58.24 | rajiv | mdave: turn on js already. heh |
| 16:58.29 | mdave | rajiv, already did |
| 16:58.36 | mdave | nufone.net |
| 16:58.40 | mdave | not voxbone.com |
| 16:58.48 | rajiv | oh |
| 16:58.50 | rajiv | heh sorry |
| 16:58.51 | |vinsik| | Hi all! I have a little problem. im using nvbackgrounddetect and rxfax to receive faxes in tiff file. Now fax line is detected, but gets disconnected with NOTICE: channel.c:1906 ast_read: Dropping incompatible voice frame on SIP/konekh-ac5d of format slin since our native format has changed to ulaw |
| 16:58.55 | rajiv | i was talking abotu voxbone |
| 16:58.55 | |vinsik| | how to resolve this? |
| 16:59.08 | mdave | nah, voxbone doesnt have 616, ive already skipped past them |
| 16:59.13 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 16:59.14 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 16:59.27 | mdave | id like to find a 616 did in a specific exchange |
| 16:59.37 | mdave | and id like to not have to pay per minute for inbound calls delivered over SIP |
| 16:59.38 | |vinsik| | using 1.2.4 |
| 17:01.40 | [TK]D-Fender | sevard : Just install from source. f you installed all the default devel libs off the install CD you'll be just fine... |
| 17:02.07 | _Sam-- | sevard 'install from source' sounds so daunting |
| 17:02.13 | _Sam-- | but its like 2 commands |
| 17:03.13 | mdave | its not daunting, but sometimes its nice if the files something installs are tracked so as to facilitate complete removal if desired |
| 17:03.34 | mdave | especially when instead of defaulting to installing *all* its files under one common subdir, it puts them all over in system dirs |
| 17:03.37 | GerbilWrk | Anyone know of a way to open or close a phone system with Asterisk? |
| 17:03.47 | mdave | define 'open' or 'close' ? |
| 17:04.12 | GerbilWrk | at 7 a.m. we open the phone system to take calls, and at 10 p.m. we close it so all calls get a recording saying we are closed |
| 17:04.22 | mdave | hrm |
| 17:04.32 | mdave | im sure its possible |
| 17:04.47 | mdave | extensions.conf i beleive allows setting time parameters on when lines are or arent valid |
| 17:04.56 | _Sam-- | its easy |
| 17:05.01 | _Sam-- | GOTOIFTime |
| 17:05.02 | sevard | Question: If I install a 2.4 kernel will I have problems with Zaptel drivers? |
| 17:05.05 | rajiv | show application GotoIfTime |
| 17:05.29 | _Sam-- | this is mine: exten => 8772942920,6,GotoIfTime(11:00-19:00|mon-fri|*|*?open,877294XXXX,1) |
| 17:05.36 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 17:05.41 | _Sam-- | er |
| 17:05.49 | _Sam-- | i guess i forgot to strip the number :) |
| 17:05.53 | GerbilWrk | :) |
| 17:05.59 | sevard | nice |
| 17:06.32 | _Sam-- | then if we're not open, its goes to the next priority for that extension |
| 17:06.56 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 17:06.59 | _Sam-- | which in my case is this: exten => 877294XXXX,7,Goto(closed,s,1) |
| 17:06.59 | mdave | _Sam--, your page looks rather empty if someone doesnt have flash installed |
| 17:07.11 | _Sam-- | mdave: good thing thats not 97% of the public |
| 17:07.28 | mdave | good thing you dont want that last 3% of the market |
| 17:07.29 | [TK]D-Fender | sevard : Nope, I've been running * on "stock" Slackware for the past 2 years |
| 17:07.30 | _Sam-- | i cater to the 97-98% |
| 17:07.45 | _Sam-- | if you dont want flash, that is your own stupid fault |
| 17:07.46 | mdave | pity, when you could cater to 100% trivially |
| 17:07.53 | _Sam-- | we have text only links |
| 17:08.06 | [TK]D-Fender | Va-t'ens buxy! Il n'y a personne! |
| 17:08.30 | _Sam-- | if you want to buy something i'll be glasd to provide it |
| 17:09.10 | *** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 17:09.23 | mdave | if I was looking for something you sold and couldnt use your site, id probably just look for another site that was more standards compliant |
| 17:09.47 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 17:09.47 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 17:09.50 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 17:09.53 | mdave | but enough on that.. i'll stick to topic now |
| 17:09.55 | _Sam-- | thats fine...but you would end up on someone's site who has less product, worse service, and higher prices...there is a reason we are #1 in our market. |
| 17:10.06 | mdave | since this isnt a 'sucky websites' channel, but a asterisk suppose channel |
| 17:10.09 | _Sam-- | but if you want to browse our site with lynx, cant help ya. |
| 17:10.13 | mdave | supporT |
| 17:10.38 | mdave | I use firefox, but I refuse to allow a site to run programs on my browser - run them on your server, display the data |
| 17:10.51 | mdave | but enough on that.. i'll stick to topic now |
| 17:11.03 | _Sam-- | your business will be dearly missed |
| 17:12.14 | *** join/#asterisk cassio (n=cassio@c91102b1.rjo.virtua.com.br) |
| 17:12.32 | cassio | I am getting too much of this, what could this be? |
| 17:12.33 | cassio | Feb 6 15:11:10 WARNING[4243]: chan_sip.c:9599 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 17:13.08 | azzie | cassio, let me guess - old version of * ? :) |
| 17:13.23 | cassio | azzie well your wrong, its up to date |
| 17:16.25 | *** join/#asterisk stormfr (n=StorM@stardust.noc.frontier.fr) |
| 17:17.03 | stormfr | hello, after upgrading to last version i still have many sip lock with realtime. Any idea how to find the problem as there is no coredump ? |
| 17:18.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:18.38 | *** part/#asterisk ahattar (n=ahattar@static-68-236-175-229.ny325.east.verizon.net) |
| 17:19.24 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 17:20.21 | stack_ | I'd like to make a queue that would dial through a group of people once, is this possible? |
| 17:20.40 | [TK]D-Fender | stack_ : Probably bast to jsut do it in dial-plan logic.. |
| 17:20.58 | znoG | has anyone experienced a box running Asterisk suddenly reboot because of it? |
| 17:21.11 | znoG | i can't imagine what else would cause it to reboot all of a sudden, unless it was hardware. |
| 17:21.30 | stack_ | [TK]D-Fender: well, i'd like it to do a bit of round robit too, so that the incoming calls get distributed among a group of people evenly |
| 17:22.06 | [TK]D-Fender | stack_ : Not much of a round robin if it only goes once :) But you can do that in dial-plan as well |
| 17:22.47 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 17:23.05 | stack_ | [Tk]D-Fender, how would I create a list of people and randomly dial them then? |
| 17:23.48 | [TK]D-Fender | stack_ : Well there is the "random" function now. As for the list, just shove it in ASTDB with incrementallyt names family/key pairs |
| 17:24.12 | dudes | random's cool |
| 17:24.21 | stack_ | [TK]D-Fender, k thanks for the start... I'll start plugging around |
| 17:25.17 | CoffeeIV_ | In my dialplan, when an incoming call selects an extension, I have it use System(myscript.sh) to keep some custom logs. I would like to also do this if one of my internal users transfers the call to another extension, but there is no place in the dialplan where it does Transfer( ) . . . is there a way to do this without hacking C code ? |
| 17:27.21 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 17:27.40 | austinnichols101 | znoG: no, but it sounds like a great feature |
| 17:30.26 | CoffeeIV_ | to have an external action trigger on a call transfer, do I have no choice but to write an application to monitor messages through the telnet API ? |
| 17:33.38 | tainted_ | CoffeeIV_ pretty much |
| 17:34.23 | *** part/#asterisk buxy (n=nnnnnnra@arrakeen.ouaza.com) |
| 17:34.29 | CoffeeIV_ | tainted_: I was getting that impression in my searches . . . . thanks |
| 17:37.21 | lo_tech | . |
| 17:41.09 | Flyboy-SR22 | mdave - saw your post re: NuFone - I have used them for over a year with very little problems...for what I do, they seem pretty good. Just some info. |
| 17:41.30 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:41.52 | mdave | Flyboy-SR22, that may be, but there seems to be no way to get in touch with them or to find the npa/nxx they offer for DID |
| 17:42.27 | Flyboy-SR22 | mdave - I will log in to my account and see if they list their DID there..I have about 10 866 DIDs with them |
| 17:42.31 | mdave | they could provide the best service in the world, but if they dont have any useful sales contacts, they might as well not exist |
| 17:42.47 | *** join/#asterisk voipme (n=root@193.120.103.128) |
| 17:42.51 | rajiv | mdave: nufone does only toll free and US michagan |
| 17:43.04 | koperniqs | cu |
| 17:43.05 | mdave | a company that desires to gain new customers isnt going to get far if they dont make it easy for new customers to get in touch with them |
| 17:43.17 | mdave | rajiv, I know, I was specifically looking for 1-616 in michigan |
| 17:43.30 | rajiv | mdave: i'm using gizmoproject (sipphone.com) for 1 did now but they dont support dtmf properly |
| 17:43.33 | rajiv | oh ok |
| 17:43.35 | Flyboy-SR22 | mdave - I only show my current DIDs - says their DID ordering system is offline right now |
| 17:44.01 | mdave | Flyboy-SR22, ah well, but apparently to be able to log in to get to it you have to pay them $ up front.. not gonna happen |
| 17:44.10 | rajiv | was just going to suck it up and go with junction and pay the incoming per min charge |
| 17:44.43 | mdave | a customer-friendly order process allows the customer to see and select all options related to what they are ordering, *then* displays a total charge, *then* prompts for personal information and payment |
| 17:44.54 | stack_ | Are AstDB values per session or global? |
| 17:44.55 | rajiv | mdave: agreed |
| 17:45.07 | Flyboy-SR22 | WOW - 3 cents per minuute |
| 17:45.08 | iCEBrkr | mdave: This is why I STILL don't have a Nufone account... |
| 17:45.12 | voipme | anyone having trouble with $HANGUPCAUSE on a EUROISDN pri, im seeing 0 always being returned on BT and eircom pri's |
| 17:45.12 | Flyboy-SR22 | Junction = expensive!! |
| 17:45.22 | mdave | requiring personal info up front is poor. requiring payment up front is a deal-breaker |
| 17:45.32 | rajiv | mdave: have you looked at www.sellvoip.net ? |
| 17:45.38 | mdave | in any case, i seem to have found another co that has 616 |
| 17:46.02 | iCEBrkr | mdave: Yeah, paying for that candy bar before you can eat it, is a a total deal breaker :P |
| 17:46.06 | mdave | they had a number i liked, and I selected it, but then their site gave me a blank page.. apparently one has to create a login first |
| 17:46.24 | mdave | no, having to pay before you even *see* what candy bars they have is the dealbreaker |
| 17:47.02 | iCEBrkr | mdave: I'm partially thinking you're expecting too much. |
| 17:47.05 | iCEBrkr | But whatever |
| 17:47.15 | iCEBrkr | I use VoicePulse and it works well for me. |
| 17:47.21 | mdave | anyway, i made a login, then tried to select the same number, but its missing from their list.. i suspect what I did at first 'reserved it' |
| 17:47.22 | rajiv | mdave: http://www.sellvoip.net/RateForm.php has form for area and prefix |
| 17:47.24 | mdave | so I sent them an email |
| 17:47.40 | mdave | iCEBrkr, expecting to see what you are buying before you but it is too much? lol |
| 17:47.40 | rajiv | iCEBrkr: voicepulse connect? |
| 17:47.40 | iCEBrkr | NuFone is always 'busy' or 'broken' or whatever. I've been attempting to get an account from them for over a year and still nothing on their site |
| 17:47.47 | iCEBrkr | rajiv: yeah |
| 17:48.08 | mdave | voicepulse connect may be an option |
| 17:48.15 | iCEBrkr | mdave: You have a package plan, you know the cost and what it'll do up front.. What's the issue here? |
| 17:48.38 | mdave | iCEBrkr, the key is being able to see what npa/nxx they have. im not looking for outbound, im looking for inbound |
| 17:48.56 | mdave | and what number i can have is very important, and in fact is even a deciding factor on wether it will work |
| 17:49.25 | iCEBrkr | Welp, VoicePulse Connect is 'pay as you go' and pretty plain-jane easy to setup. |
| 17:49.41 | mdave | yeah.. if you only use their inbound, is it basically free? |
| 17:49.48 | mdave | your prepaid balance just sits there unused? |
| 17:49.52 | iCEBrkr | Yup |
| 17:49.53 | iCEBrkr | well |
| 17:49.57 | iCEBrkr | wait. |
| 17:50.03 | iCEBrkr | You have a monthly for the DID |
| 17:50.05 | rajiv | $11 / month |
| 17:50.07 | tainted_ | it's not free |
| 17:50.10 | tainted_ | it's 11/month |
| 17:50.16 | tainted_ | and the quality isn't reliable |
| 17:50.16 | mdave | hrm |
| 17:50.20 | rajiv | gizmo is $12 / 3 months |
| 17:50.20 | mdave | their site says 'No monthly charges or minimums' |
| 17:50.29 | iCEBrkr | tainted_: Maybe in your neck of the woods, but it works just fine over here.. |
| 17:50.40 | iCEBrkr | mdave: For a DID you pay $11/mo |
| 17:50.50 | mdave | ah ok |
| 17:50.54 | tainted_ | iCEBrkr i used them for various rate centers |
| 17:50.56 | mdave | the no monthly is on the outbound |
| 17:50.58 | iCEBrkr | mdave: If you're all outbound, then there isn't a monthly. |
| 17:51.04 | mdave | well this one ive found says 7/m |
| 17:51.04 | tainted_ | iCEBrkr wasn't one DID connected to my desk phone dude |
| 17:51.07 | mdave | so i'll try that |
| 17:51.19 | mdave | but i have to go |
| 17:51.20 | mdave | ttyl |
| 17:51.23 | iCEBrkr | tainted_: eh? |
| 17:52.06 | iCEBrkr | tainted_: I'm using VoicePulse exclusively for ALL my calls.. I don't own a landline for phone calls. The only issues I have are bandwidth issues on my end.. like at my apartment. |
| 17:52.13 | saftsack | many, hi are you here? |
| 17:52.35 | tainted_ | what kind of issues.. could be them |
| 17:53.22 | iCEBrkr | tainted_: Naa, I'm pretty sure it's my bandwidth. |
| 17:53.40 | tainted_ | how are u so sure |
| 17:53.42 | tainted_ | tcpdump? |
| 17:53.44 | iCEBrkr | tainted_: I'm in an apartment complex.. And you know bandwidth is sketchy. |
| 17:53.56 | tainted_ | ummm |
| 17:53.58 | iCEBrkr | tainted_: I know cuz I'm on shitty Roadrunner. |
| 17:54.04 | tainted_ | are u riding your neighbor's wifi connection? |
| 17:54.22 | *** join/#asterisk Morex (n=blah@host86-137-22-82.range86-137.btcentralplus.com) |
| 17:54.34 | Morex | Anybody have any problems with DTMF over ZAP? |
| 17:54.37 | iCEBrkr | tainted_: Hello! Apartment = Concentrated number of users. More so than on a typical city block.. |
| 17:55.06 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 17:55.13 | iCEBrkr | tainted_: During the day when people are at work, I get a solid 500k/sec, other times, I'm lucky to get 300k/sec |
| 17:55.46 | iCEBrkr | The problems are kinda rare so I haven't spent any time looking into it. |
| 17:55.57 | iCEBrkr | My friends claim the call quality isn't any worse than a cellphone call. |
| 17:56.19 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 17:57.17 | tainted_ | i thought that cable modem worked around the old 'shared bandwidth' model |
| 17:57.27 | tainted_ | but maybe roadrunner hasn't |
| 17:57.29 | *** join/#asterisk toMACINA (n=toMACINA@port-83-236-223-18.static.qsc.de) |
| 17:57.34 | toMACINA | hi all |
| 17:57.41 | Himeko | isn't that exactly what he is talking about |
| 17:58.16 | tainted_ | yea i suppose he is |
| 17:58.34 | Himeko | 300k/sec is more than enoug, it's latency goign up that will mess with the call |
| 17:58.46 | iCEBrkr | tainted_: Roadrunner in Florida sucks ass.. |
| 17:59.01 | iCEBrkr | tainted_: The modem continues to reset itself at random times.. Just because. |
| 17:59.26 | tainted_ | well what kind of problems are you having with calls |
| 17:59.41 | Astar | hum dont remember how to monitor asterisk |
| 17:59.45 | Astar | what is the command ? |
| 17:59.45 | tainted_ | you still might be misdiagnosing |
| 17:59.52 | _Paulo_ | asterisk -r ??? |
| 18:00.17 | Astar | thanks |
| 18:00.58 | Astar | hum it's not started pfff |
| 18:03.20 | *** join/#asterisk ToTo (n=ToTo@host103-158.pool874.interbusiness.it) |
| 18:09.05 | Astar | where can i see why its not running ? |
| 18:10.32 | tronix | logfile. maybe /var/log/asterisk/messages |
| 18:10.40 | tronix | or start up by hand with: |
| 18:10.46 | tronix | # asterisk -vvvvc |
| 18:10.56 | tronix | (and add -u asterisk -g asterisk if you run under asterisk user/group) |
| 18:11.48 | *** join/#asterisk Genman (n=hansenhl@atlrel2.hp.com) |
| 18:12.00 | Genman | Hi people |
| 18:12.03 | _Sam-- | safe_asterisk works too |
| 18:12.16 | _Sam-- | and you should probably run it as a user other than root |
| 18:13.15 | Genman | I'm looking for a Channel Bank 30 channels -> E1 |
| 18:13.18 | Genman | Does anyone have any suggestions? |
| 18:13.33 | Genman | Where i can find one? |
| 18:13.34 | sevard | in asterisk@home on tty9 I'm getting a non readable font for colored text |
| 18:13.54 | *** join/#asterisk overridex-work (n=override@69-161-57-4.sbtnvt.adelphia.net) |
| 18:14.13 | Astar | thanks |
| 18:14.20 | *** join/#asterisk steelcase (n=stevec@63.173.198.31) |
| 18:14.55 | Genman | brb |
| 18:15.21 | overridex-work | hi all, this question isn't asterisk specific, but has anyone had a problem of calls being dropped when on hold using T1? (using analog lines split out of it) |
| 18:15.56 | *** part/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 18:17.56 | *** join/#asterisk _4d4m_ (n=adam@176-40-101-159.adsl.legend.co.uk) |
| 18:18.58 | _Sam-- | genman: fxo or fx? |
| 18:19.00 | _Sam-- | er or fxs |
| 18:20.11 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 18:21.02 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 18:26.11 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 18:27.10 | *** join/#asterisk zamsler (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 18:28.37 | saftsack | _Sam--, hi |
| 18:32.35 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 18:33.06 | *** join/#asterisk telcoDE (n=A_mail@pool-68-238-244-251.phlapa.fios.verizon.net) |
| 18:33.20 | _Sam-- | hi sack |
| 18:33.28 | _Sam-- | how is your faxing? |
| 18:33.49 | _Sam-- | telcoDE: how like your fios? i just ordered the 15/2 package |
| 18:34.14 | _Sam-- | and what telco? |
| 18:34.58 | telcoDE | I love it, although I only went with the 5 |
| 18:35.09 | saftsack | _Sam--, faxing is good here :) |
| 18:35.10 | _Sam-- | you were on a cable modem on friday, fios today...what gives! |
| 18:35.17 | _Sam-- | fios at your house? |
| 18:35.20 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:35.20 | _Sam-- | cable modem at work? |
| 18:35.36 | *** join/#asterisk juanjoc (n=jcomella@222-32-235-201.fibertel.com.ar) |
| 18:35.37 | saftsack | _Sam--, but i dont use faxgetty so its a little bit hard to set the modemstatus as idle and ready |
| 18:35.44 | _Sam-- | i live down in middletown...where is your fios? |
| 18:35.54 | telcoDE | this is FIOS at home. I'm off work today |
| 18:36.10 | telcoDE | middletown area as well |
| 18:36.15 | Katty | file: :> |
| 18:36.17 | _Sam-- | im close to boyds corner |
| 18:36.21 | _Sam-- | my house |
| 18:36.41 | _Sam-- | telcoDE: do you work for a telco / phone integrator? |
| 18:36.59 | telcoDE | I'm here logging, so I can pick up on Asterisk questions/answers |
| 18:37.28 | _Sam-- | do you run asterisk for any businesses or do any asterisk commercial work? |
| 18:37.37 | telcoDE | yep a telecom equipment seller/installer |
| 18:37.45 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 18:37.57 | telcoDE | looking into ASterisk just to get up to speed on the future |
| 18:38.17 | _Sam-- | i do commercial asterisk deployments in DE,PA and NJ |
| 18:38.22 | telcoDE | traditional PBXs are loosing major market share to voip |
| 18:38.32 | tronix | telcoDE: ahh, nice. I used to live in Newark and Wilmington for about 12 years. Liked living in NCCo but areas near the C&D was decent, too |
| 18:38.47 | *** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
| 18:38.55 | _Sam-- | tronix: its not too bad here, still. |
| 18:39.24 | tronix | sweet |
| 18:39.33 | _Sam-- | how long has it been since you were here? |
| 18:40.01 | tronix | hmm I went to western NY for school then got sucked into a job, and haven't been 'home' since... about 12 years now. :) but still do visit DE every few years |
| 18:40.07 | tronix | go to the Pumpkin Chunkin' and stuff |
| 18:40.19 | _Sam-- | how did you end up in DE? |
| 18:40.34 | tronix | dad got transferred by DuPont to their Newark office |
| 18:40.50 | telcoDE | anyone here used your * box to do double duty for home/office security or home/office automation? |
| 18:40.52 | tronix | well, one of all their NCCo offices/labs. :) |
| 18:41.13 | _Sam-- | nice, one of my best friends is a dupont :) |
| 18:41.18 | _Sam-- | his grandfather is THE dupont :) |
| 18:41.22 | tronix | telcoDE: I haven't personally done it but was reading about a X10 module that can be integrated |
| 18:41.26 | tronix | _Sam--: niiiice! |
| 18:41.30 | _Sam-- | for him! |
| 18:41.35 | tronix | hahaha I can imagine ;) |
| 18:43.55 | queuetue | I was just considering picking up a little epia board to build a combination home * server and media PC. But then I remembered hearing about some sort of limitation with "exotic" motherboards not working with the zaptel cards ... Did I imagine that problem? Higher voltage something-or-other? |
| 18:44.18 | fugitivo | telcoDE: powerlinc usb or powerlinc serial |
| 18:44.29 | telcoDE | I heard about X10 as well. I have a dialer box thar used to be at my old job. It calls out a predefined lict of numbers based on changes in circuit state from open->closed or closed->open. Wondered how tough it would be to integrate it's alarming function with an open source security product |
| 18:44.44 | fugitivo | telcoDE: there's an opensource project that adds /dev/xxx nodes for controlling devices |
| 18:44.44 | sevard | In voicemail I have 140 => 12345,Bob Smith,,,attach=no|saycid=yes|envelope=no|delete=no yet *411 says No directory entries match your search |
| 18:45.16 | fugitivo | telcoDE: for example echo 1 >> /dev/x10/A1 will power on a device |
| 18:45.21 | telcoDE | I'll have to look into that thanx |
| 18:45.28 | sevard | oh god, i'm retarded, never mind |
| 18:49.35 | telcoDE | time to check in with my work email, bbl |
| 18:51.27 | *** join/#asterisk Spida (i=Spida@p508A27EF.dip0.t-ipconnect.de) |
| 18:51.38 | Spida | hi |
| 18:52.06 | Spida | can I get help here for getting my fritz pci to work with mISDN, too? |
| 18:53.53 | saftsack | are any german voip user here? |
| 18:55.13 | Spida | I hope to become one *g* |
| 18:55.41 | steelcase | Linux admin, learning about Asterisk, interested in suggesting it to my company: can they use their old Toshiba phones? The one on my desk says it's a "DTK2010-SD." |
| 18:57.48 | steelcase | Make that "DKT2010-SD." |
| 18:58.01 | crich1999 | Spida, i could help you but i must go home now |
| 18:58.10 | file | no, no you can't steelcase |
| 18:58.36 | steelcase | file: That's a bummer, why not? |
| 18:58.40 | crich1999 | Spida, I'm at home in around 2 hours, if you're still here i can try to help ya |
| 18:58.43 | file | proprietary |
| 18:58.50 | fafnir | FOG |
| 18:59.23 | steelcase | file: Dang! Hafta get some different phones maybe. |
| 18:59.41 | [av]bani | you'd spend as much money to get new phones anyway |
| 19:00.14 | steelcase | [av]bani: You mean, we'd spend as much money to get Asterisk set up? |
| 19:00.38 | steelcase | [av]bani: With old phones as well as new ones? |
| 19:00.40 | rene- | hi [av]bani |
| 19:01.17 | rene- | hows grandstream auto provisioning working out for you? |
| 19:02.16 | rene- | oh wait, it is that time of the day |
| 19:03.04 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 19:03.20 | *** join/#asterisk j0n (n=jellis@206-169-48-226.gen.twtelecom.net) |
| 19:04.56 | j0n | does anyone know how to use hints in AEL for presence? |
| 19:06.59 | *** join/#asterisk apardo (n=apardo@87.218.44.253) |
| 19:11.18 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 19:13.10 | *** join/#asterisk donnib (n=aaa@0x555281d0.adsl.cybercity.dk) |
| 19:13.28 | *** join/#asterisk moy (n=kvirc@201.145.203.195) |
| 19:14.01 | donnib | i am running asteriskwin32 and my asterisk keeps telling me that there is no such host. why ? is it possible to make a ping or nslookup on asteriskwin32 ? |
| 19:14.07 | *** join/#asterisk GerbilNut (i=GerbilNu@65.88.144.41) |
| 19:14.08 | [av]bani | steelcase: yes, you'd spend as much on equipment to interface your old phones, as you would on completely new voip phones |
| 19:14.12 | *** part/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 19:14.39 | [av]bani | rene-: havent got around to that yet, still working on snom and sipura |
| 19:14.45 | [av]bani | and polycom :x |
| 19:14.57 | _Sam-- | hey bani you still on teliax? |
| 19:15.09 | GerbilNut | _Sam--, i've got a question about the GOtoIf statement you showed me earlier |
| 19:15.14 | hypnox | surely a 2 line sipura ATA which costs half that of a phone means you can connect old phones for 25% of the cost of a new IP phone |
| 19:15.16 | _Sam-- | GerbilNut: ok |
| 19:15.28 | GerbilNut | Anyway to make it do it at different times on different days? |
| 19:15.32 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:15.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:16.11 | donnib | any ideas ? |
| 19:16.44 | *** part/#asterisk secure75 (n=mic@host-82-135-30-151.customer.m-online.net) |
| 19:16.44 | _Sam-- | GerbilNut: sure |
| 19:16.53 | _Sam-- | i just set mine monday-friday 11-7 |
| 19:16.54 | [av]bani | _Sam--: yea, but havent used it lately |
| 19:16.58 | _Sam-- | cause that is our business hours |
| 19:17.16 | donnib | anyone using asteriskwin32 ? |
| 19:17.24 | GerbilNut | oh crap, i didn't even see the mon-fri statement in there : / |
| 19:17.35 | _Sam-- | wow i never even know about asteriskwin32 |
| 19:18.08 | donnib | i think it works ok when you got it up and running just for test purposes. |
| 19:18.09 | [av]bani | _Sam--: its just asterisk on linux running inside a vmware or something like that |
| 19:18.35 | fugitivo | it's something you won't do if you look for stability |
| 19:18.41 | fugitivo | ;) |
| 19:18.43 | _Sam-- | i see |
| 19:19.29 | donnib | does anybody have an idea on my problem ? |
| 19:19.31 | [av]bani | if you can believe it, snom firmware feels even more beta than grandstream |
| 19:19.52 | rene- | av[bani]: im working with sipura right now, xml configuration tru cgi bin works well |
| 19:19.55 | _Sam-- | wow, i was going to get a snom just to see if i could hear any audible differences in the sound qualty vs. the gx |
| 19:20.00 | _Sam-- | vs. the gxp too |
| 19:20.12 | _Sam-- | i like my gxps, i cant help it |
| 19:20.18 | _Sam-- | they friggin work fine |
| 19:20.18 | [av]bani | every bug you send them, they claim complete astonishment and assume user error by default |
| 19:20.20 | GerbilNut | we just got two snom 360's in |
| 19:20.42 | [av]bani | i have to ride them, give them excruciatingly detailed bug reports, tcpdumps, digital photos, etc |
| 19:20.45 | GerbilNut | can't complain, just can't get the voicemail working properly yet |
| 19:20.48 | [av]bani | before they fix bugs |
| 19:20.56 | _Sam-- | grandstream is listening |
| 19:21.02 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 19:21.06 | _Sam-- | they implemented the disable missed call feature i asked for |
| 19:21.08 | _Sam-- | took them about 1 week |
| 19:21.11 | [av]bani | yeah, its one way though. at least snom has email which answers |
| 19:21.14 | rene- | its kind of a downer that grandstream only takes binay configuration |
| 19:21.15 | justinu | who else is having problems with roadrunner here? |
| 19:21.29 | _Sam-- | i have good contact at GS if i had a serious prblem |
| 19:21.43 | [av]bani | you dont know what grandstream is listening to (eg, ignore sendtext, custom ringtones, minibrowser, but implement nat router? guh...) |
| 19:22.10 | _Sam-- | what would you do wtih a mini browser? give me the killer app for it |
| 19:22.18 | steelcase | [av]bani: Got it -- thanks |
| 19:22.21 | _Sam-- | i dont know why i need it, beecause i dont know what it can do |
| 19:22.31 | [av]bani | queue displays, interfacing with directory lookups for phone numbers or customer data |
| 19:22.32 | fugitivo | _Sam--: a lot of things |
| 19:22.49 | justinu | i just cancelled RR and ordered speakeasy 6.0/768 service |
| 19:22.52 | fugitivo | _Sam--: give only a phone to the recepcionist with the right app, not a pc |
| 19:23.00 | _Sam-- | i see |
| 19:23.33 | fugitivo | _Sam--: cisco has a great app for hospitals, they can see the pacient history on the phone |
| 19:23.34 | [av]bani | let receptionists look up business/customer-specific data on the phone itself |
| 19:23.37 | _Sam-- | we have a nice front end for lookups of customer and other data, with a nice click to dial setup |
| 19:23.43 | [av]bani | pull up customer specific info on the screen when customers call |
| 19:23.50 | _Sam-- | it would take longer to do it on the phone, than on the comptuer she uses |
| 19:23.57 | [av]bani | [TK]D-Fender implemented inventory lookup on the phone |
| 19:24.15 | _Sam-- | whats the point...every desk with a phone, has a computer. |
| 19:24.17 | [av]bani | so you have phones in your warehouse |
| 19:24.24 | [av]bani | few desks in a large warehouse |
| 19:24.27 | _Sam-- | our warehouse has computers, and phones |
| 19:24.27 | fugitivo | _Sam--: easy of use |
| 19:24.29 | [av]bani | but you can hang phones everywhere |
| 19:24.52 | _Sam-- | i wouldnt want our warehouse guys checking inventory on a phone |
| 19:24.59 | rene- | [av]bani: the directory lookup was on what phones? |
| 19:25.08 | [av]bani | i guess your warehouse guys suck :) |
| 19:25.15 | _Sam-- | because it would take them twice as long as it does now |
| 19:25.20 | _Sam-- | its a matter of efficiencies |
| 19:25.21 | rene- | s/directory/inventory |
| 19:25.25 | fugitivo | _Sam--: the pc itself is an old design, say welcome to lcd screens and no mouse :) (movies) |
| 19:25.27 | _Sam-- | a phone isnt always the most efficient way |
| 19:25.31 | [av]bani | shrug, if your inventory is numeric only, i dont see how |
| 19:26.00 | _Sam-- | we have so many ways to check our inventory...but often times, we need to see all items from a specific manufacturer |
| 19:26.12 | _Sam-- | and a phone would be slower...same with looking up customer numbers |
| 19:26.12 | [av]bani | cisco has an entire website of xml applications, for medical financial etc. |
| 19:26.21 | fugitivo | _Sam--: you're right, a phone isn't always the best way, but sometimes it is |
| 19:26.24 | _Sam-- | the examples you have sasid would be less efficient using a mini browser |
| 19:26.27 | rene- | [av]bani: this was on cisco phones. via xml? |
| 19:26.33 | _Sam-- | than using a full web browser |
| 19:26.36 | [av]bani | _Sam--: ok, so xml doesnt work for you. but it works for tens of millions of other companies :) |
| 19:26.54 | _Sam-- | maybe they should hire me to show them how to save time! |
| 19:27.21 | [av]bani | for me, i can interface with argus and respond to ISP alarms from my nightstand |
| 19:27.25 | fugitivo | yeah, cisco will hire you to show them that they're wrong and you're right |
| 19:27.47 | _Sam-- | the companies using minibrowsers to look up customer phone numbers...i could certainly save them minutes per employee per day. |
| 19:27.59 | rene- | whats the deal with presence, is it something costumers are requesting? |
| 19:28.17 | [av]bani | rene-: you can do it with any phone that has xml or minibrowser |
| 19:28.31 | rene- | that would be mostly cisco and polycom right? |
| 19:29.02 | [av]bani | rene-: aastra, snom also |
| 19:29.09 | rene- | i see |
| 19:29.18 | [av]bani | mitel too, but those are way expensive |
| 19:29.20 | *** join/#asterisk DrData (n=michael@p54B25F2D.dip.t-dialin.net) |
| 19:29.55 | [av]bani | _Sam--: http://www.o2m8.com/modules.php?name=News&file=article&sid=25 |
| 19:30.01 | rene- | they arent that expensive, well the ones with mini screens arent, i remember seeing one that has something that resembled a color palm computer with touchscreen and stylus |
| 19:30.02 | [av]bani | maybe not useful to _you_, but it's useful to others |
| 19:30.14 | rene- | that one is probably very expensive |
| 19:30.15 | [av]bani | not having to have a PC everywhere for things you want to interface to is nice |
| 19:30.34 | [av]bani | especialyl places you dont necessarily want a PC, eg the kitchen |
| 19:30.44 | fugitivo | it's useful, nice and cool |
| 19:31.22 | [av]bani | being able to display the queue status on the _phone itself_ is nice, rather than having to refer to a pc all the time |
| 19:31.36 | fugitivo | or controling your x10 house with the phone :) |
| 19:31.51 | [av]bani | you can implement phone functions on the phone itself, like being able to intercept conferences, pick up other extensions via the phone, etc |
| 19:32.04 | [av]bani | any function the phone can't currently do, but you can implement via asterisk |
| 19:32.06 | _Sam-- | im sorry for seeming like i was saying the technology isnt useful....i just understand for myself how it would do anything good for the things that i do daily. |
| 19:32.25 | rene- | what about the presence stuff, from my understanding is getting your IM status and have your phone react according to it, a separate jabber like service seems to be required... is it something your customers are requesting? |
| 19:32.28 | [av]bani | so it would suck for you, but everyone else on the entire planet finds it usefl :) |
| 19:32.51 | _Sam-- | [av]bani: if they implemented it tomorrow, what is the first thing you would do/setup with it? |
| 19:32.53 | DrData | can anybody help me with asterisk+sipgate+capi? |
| 19:33.03 | [av]bani | _Sam--: if who did? grandstream? |
| 19:33.06 | _Sam-- | yeah |
| 19:33.20 | [av]bani | i'd add interface to argus |
| 19:33.30 | [av]bani | and queue status for our receptionists |
| 19:33.44 | [av]bani | along with customer lookup of data when they call in |
| 19:34.05 | _Sam-- | we do that now...but we just modify caller ID bsaed on the data |
| 19:34.28 | _Sam-- | if the calleridnum is in our system, we modify the callerid |
| 19:34.29 | [av]bani | account info, passwords, billing info, address |
| 19:34.43 | [av]bani | you can control the formatting with xml |
| 19:34.55 | [av]bani | not the wraparound mess that would be with cid :) |
| 19:35.31 | _Sam-- | i hear what you're saying, and i can see how that would be useful to some...but having customer info on the lcd of the phone isnt that practical for what we do...it would still need to be displayed on the computer anyway |
| 19:35.46 | overridex-work | hi all, this question isn't asterisk specific, but has anyone had a problem of calls being dropped when on hold using T1? (using analog lines split out of it) |
| 19:35.50 | _Sam-- | but i am starting to hear what you're saying...slowly :) |
| 19:36.15 | [av]bani | so you cant think of a single use for xml that would be useful to your company, that sucks |
| 19:36.22 | [av]bani | your loss :) |
| 19:36.39 | _Sam-- | i'll ask our web guy see what says :) |
| 19:36.46 | fugitivo | _Sam--: you don't need to have a pc on each position or maybe you want a phone on a hall where you don't put a pc |
| 19:36.55 | [av]bani | it's not so much "OMG SAVED 3 HOURS PER CALL", its more about making your receptionists happy |
| 19:37.11 | _Sam-- | i am not about making people happy...i am about saving time per employee :) |
| 19:37.14 | [av]bani | or not having to get out of bed and reach for a pc keyboard and mouse |
| 19:37.39 | cpm | Just as an aside, for all of this, I could have gone to radio shack, bought a $20 answering machine, thrown it on the ground, stomped on it, and had a telephone device that didn't work either, but it would have only cost $20 and only taken a few moments, rather than weeks and kilo buks. |
| 19:37.47 | fugitivo | it's the same quetion, why you need a palm or pda if you have a pc? |
| 19:38.11 | [av]bani | cpm: i feel like that when i use microsoft windows |
| 19:38.16 | cpm | heh |
| 19:38.42 | cpm | Well, in that case, It's a $.69 switch, stomped on the ground. |
| 19:39.10 | _Sam-- | fugitivo: i dont think thats a fair comparison, since a pda has more functionality than 1 feature (minibrowser)... |
| 19:39.19 | _Sam-- | you cant send email through your phone (ok i bet YOU could somehow) |
| 19:39.36 | _Sam-- | through your SIP phone...or wouldnt want to receive a jpg attachment to your phone |
| 19:39.45 | fugitivo | _Sam--: you're wrong, a minibrowser could have any functionality you want |
| 19:40.00 | _Sam-- | my pda ...i can do alot more than a mini browser |
| 19:40.01 | fugitivo | _Sam--: that's why it's a "browser" and not an app inside the phone |
| 19:40.11 | rene- | does the poly 301 has this mini browser? |
| 19:40.18 | fugitivo | _Sam--: a mini browser will do what the app server says |
| 19:40.27 | fugitivo | _Sam--: so your pda is more limited than a mini browser |
| 19:40.42 | _Sam-- | if you say so it must be! |
| 19:41.09 | _Sam-- | given a choice between my PDA...and a mini browser on my phone, i will pick my PDA each and every time :) |
| 19:41.14 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 19:41.22 | _Sam-- | <my pda is also a pcs phone > |
| 19:41.43 | fugitivo | _Sam--: ok |
| 19:41.56 | fugitivo | _Sam--: just remember that the future are the webapps :) |
| 19:42.26 | _Sam-- | i am just a few steps behind...when the killer app comes for XML for my phone ...i will know i need it. |
| 19:42.39 | _Sam-- | but i dont know i need it because i dont know what i need it for |
| 19:42.45 | _Sam-- | when i know, i will now i guess! |
| 19:43.19 | fugitivo | you think that the minibrowser is an integrated app and it's not, it's just like any other webapp |
| 19:43.27 | [av]bani | rene-: only the 601 has minibrowser |
| 19:44.13 | fugitivo | the idea of the minibrowser is coding what you need, maybe there'll be some apps that'll fit your needs, but the real power is developing apps for your specific needs |
| 19:44.57 | [av]bani | i would say it is more that _Sam-- lacks imagination, rather than xml lacks application :) |
| 19:45.09 | fugitivo | maybe it's not usefull for you or your company or you still don't have the vision of how it'll help in your company |
| 19:45.13 | [av]bani | ask your receptionists what would save them time on the phone |
| 19:45.23 | [av]bani | ask your warehouse guys what would be useful |
| 19:45.30 | fugitivo | i agree with bani |
| 19:45.32 | _Sam-- | i spend alot of deep thought on our phones and what will save our people time...i have 10 people that do inbound phone sales work all day |
| 19:46.03 | fugitivo | a warehouse guy will be more comfortable with a phone with a little screen, not a pc |
| 19:46.07 | *** join/#asterisk lorinc (n=ang@caracas-1512.adsl.interware.hu) |
| 19:46.14 | fugitivo | just a few keys to press |
| 19:46.21 | _Sam-- | it doesnt accomplish anything better than we do now |
| 19:46.47 | _Sam-- | in fact, unless you could hook up a bar code scanner to the phone, i would argue it will take longer. |
| 19:46.58 | _Sam-- | but that is ME and my specific situation. |
| 19:47.12 | fugitivo | _Sam--: you were the guy that insult me for not using a webinterface for asterisk? |
| 19:47.21 | _Sam-- | not for asterisk, no. |
| 19:47.26 | _Sam-- | for not using any GUI for linux. |
| 19:47.27 | _Sam-- | like X |
| 19:47.31 | fugitivo | you don't seem to be so open minded |
| 19:47.44 | fugitivo | i use X |
| 19:47.57 | _Sam-- | you said all your asterisk stuff is just done from console / vi |
| 19:48.01 | fugitivo | yes |
| 19:48.02 | fugitivo | that's right |
| 19:48.05 | fugitivo | but i use x |
| 19:48.08 | fugitivo | dual head |
| 19:48.11 | fugitivo | with kde |
| 19:48.19 | _Sam-- | my argument was that i have more tools available under X than you have just working from a console |
| 19:48.28 | fugitivo | i do my work from the consoles |
| 19:48.33 | _Sam-- | and that for development work, using an interface outside the console is quicker. |
| 19:48.43 | _Sam-- | having many open windows / screens / terms on a desktop |
| 19:48.49 | _Sam-- | rather than screen and flopping back and forth |
| 19:49.14 | fugitivo | well, you should see the minibrowser in that way and you'll understand why it's usefull |
| 19:49.44 | _Sam-- | just because it exsits doesnt mean that its useful for everyone |
| 19:50.21 | fugitivo | it exists and it's used by millions of people, big brands like cisco and polycom includes it on their phones |
| 19:50.37 | _Sam-- | so, my car comes with an AM radio...that is useful to many people that want that feature |
| 19:50.40 | _Sam-- | but i dont need it |
| 19:50.53 | _Sam-- | its the same thing...many people may use it and like it... |
| 19:50.57 | _Sam-- | but it doesnt mean that i need it |
| 19:50.59 | _Sam-- | or have any use for it |
| 19:51.18 | fugitivo | not for you, maybe it's useful for people on your company |
| 19:51.26 | _Sam-- | i am not trying to be difficult or argumentative by any means. |
| 19:51.33 | _Sam-- | and if i DID have the killer app for it, i would certainly admit that |
| 19:51.35 | _Sam-- | and try to figure it out |
| 19:52.05 | fugitivo | the killer app would be a self developed app that fits your needs |
| 19:52.06 | _Sam-- | and maybe tonite when i cant sleep, i will be like "that bastard fugitivo was right..i cant beleive i never thought of using it that way" |
| 19:52.18 | _Sam-- | but as of now, i dont have that though! |
| 19:52.50 | fugitivo | google it, maybe looking at the apps you'll have an idea of why it's usefull |
| 19:53.37 | fugitivo | i saw some apps from cisco and i think it's great |
| 19:53.58 | fugitivo | the apps for medical companies are great |
| 19:56.26 | *** join/#asterisk apardo (n=apardo@87.218.44.253) |
| 19:56.30 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:59.03 | *** join/#asterisk areski (n=areski@184.Red-83-60-97.dynamicIP.rima-tde.net) |
| 20:04.01 | rustyb | is anyone else having garbled inward audio problems with connect.voicepulse.com? |
| 20:04.49 | cpm | they keep dropping me, 'too lagged' today. /me thinks they are congested today |
| 20:04.57 | cpm | over selling their bandwidth yet again |
| 20:05.39 | fugitivo | wait until they buy more |
| 20:05.59 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 20:07.02 | cpm | Microsoft OSs running critical medical apps, are the true killer apps |
| 20:07.21 | fugitivo | lol |
| 20:07.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-112-192.red.bezeqint.net) |
| 20:07.45 | fugitivo | like controlling robot arms with a laser? |
| 20:08.06 | kuku5 | anyone else having major problems with cisco 7960 phone hanging up ? |
| 20:10.12 | [av]bani | _Sam--: given up on teliax yet? |
| 20:11.56 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.9) |
| 20:12.06 | _Sam-- | bani not yet, almost. |
| 20:12.21 | _Sam-- | ive got our outgoing from 2 places going over asterlink |
| 20:12.27 | _Sam-- | that guy file is a hack, but so far so good :) |
| 20:12.50 | _Sam-- | im still debating pulling the trigger on porting our main toll free over to asterlink |
| 20:12.54 | Dr-Linux | anyone tell me please that why there is too much active channels from "3001" and 3002 extension? |
| 20:12.54 | Dr-Linux | http://pastebin.com/542067 |
| 20:15.12 | austinnichols101 | kuku5: no problems with 7960's here (7.5 fw) |
| 20:15.23 | kuku5 | ah |
| 20:15.26 | kuku5 | im using 7.3 |
| 20:16.34 | austinnichols101 | I've heard there are problems with 7.5, but haven't seen any |
| 20:18.11 | [av]bani | _Sam--: i've got a ticket open for some days now, basically screaming 'i want to give you money but you dont want to take it' so heh ... |
| 20:18.20 | [av]bani | i guess that pretty much says it all |
| 20:18.35 | _Sam-- | the main guy from teliax was here earlier |
| 20:18.47 | _Sam-- | he said if i found anyone with problems to tell them to email him directly |
| 20:18.51 | [av]bani | on irc? |
| 20:18.54 | _Sam-- | yeah |
| 20:18.57 | [av]bani | heh |
| 20:19.05 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 20:19.07 | _Sam-- | hold on i'll get you his email |
| 20:19.17 | [av]bani | well given the stuttering issues combined with the non repsonse to tickets... |
| 20:19.22 | [av]bani | i dont think im going to bother |
| 20:19.43 | [av]bani | fwiw it seems ITSP quality is universally suck |
| 20:19.45 | file | who called me a hack? :) |
| 20:19.51 | _Sam-- | lol |
| 20:19.54 | _Sam-- | jk obviously! |
| 20:20.05 | _Sam-- | [av]bani: asterlink has worked well for me |
| 20:20.12 | _Sam-- | i mean, the calls are like 10X clearer |
| 20:20.22 | _Sam-- | but i am 10ms to them, 50ms to teliax |
| 20:20.36 | _Sam-- | you already know the downside to asterlink...no local DID |
| 20:20.49 | sevard | Is Playfile(current-time-is.gsm); correct syntax? |
| 20:21.00 | twisted[asteria] | no |
| 20:21.14 | *** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net) |
| 20:21.21 | [av]bani | _Sam--: i wish grandstream would implement setting the provisioning url via dhcp |
| 20:21.37 | [av]bani | thats the only thing preventing me from having a truly out-of-the-box-pnp solution |
| 20:21.40 | [av]bani | 0 config |
| 20:21.48 | _Sam-- | [av]bani: one other thing about asterlink....check this out |
| 20:21.54 | sevard | twisted[asteria]: is System(current-time-is.gsm); correct syntax? |
| 20:21.59 | _Sam-- | it was 11:30 on a friday night....i had some crazy problem forget what.. |
| 20:22.01 | twisted[asteria] | except that your customers dhcp is beyond your control |
| 20:22.05 | twisted[asteria] | sevard, second time: no. |
| 20:22.06 | _Sam-- | i message the asterlink guy, and he actually writes back! |
| 20:22.11 | cpm | Err, is there a way to ask a zap fxo port if it detects dialtone? |
| 20:22.15 | _Sam-- | 11:30 on a friday night |
| 20:22.19 | sevard | twisted[asteria]: I changed Playfile to System. |
| 20:22.24 | twisted[asteria] | sevard, still not right. |
| 20:22.26 | [av]bani | well some companies ahve hardcore geeks running them, i think tahts the case with alot of ISTPs now |
| 20:22.30 | twisted[asteria] | ~docs |
| 20:22.35 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:22.35 | sevard | twisted[asteria]: want to give me a hint? |
| 20:22.37 | sevard | twisted[asteria]: got it |
| 20:23.15 | _Sam-- | part of the teliax guys problem (not customer service wise)...is that he is in bed with rockynet it looks like |
| 20:23.27 | _Sam-- | and until he gets off there, i think the service wont improve |
| 20:23.27 | [av]bani | _Sam--: how long that situation lasts is anyones guess, but it was like that back in the late 70s/early 80s too. you could call PC mfg's and find guys still hacking away at 11pm |
| 20:23.58 | [av]bani | yea rockynet seems like some small time isp, and teliax is joined at the hip or something |
| 20:24.13 | _Sam-- | rockynet does have OK connectivity, but not great. |
| 20:24.18 | _Sam-- | they are not just a tiny ISP |
| 20:24.23 | _Sam-- | but maybe a small/medium one :) |
| 20:24.25 | [av]bani | not what it takes for a reliable ITSP |
| 20:24.45 | [av]bani | either that or teliax needs better equipment |
| 20:24.54 | [av]bani | it doesnt seemt o handle jitter at all |
| 20:25.07 | _Sam-- | asterlink: better customer service, better call quality, close to industry standard pricing.....the only thing teliax has better is caller id with name. |
| 20:25.25 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-5.dynamic.qsc.de) |
| 20:25.27 | Dr-Linux | anybody get a chance to see my pastbin? |
| 20:25.29 | Dr-Linux | http://pastebin.com/542067 |
| 20:25.39 | _Sam-- | and i m going to test something from asterlink that may make the caller id with name functional , or somewhat functional |
| 20:26.34 | *** join/#asterisk bweschke (n=bweschke@ip67-91-35-121.z35-91-67.customer.algx.net) |
| 20:27.34 | [av]bani | _Sam--: you have any use for auto-provisioning ? |
| 20:28.00 | _Sam-- | it would all be done from the DHCP parameters? |
| 20:28.12 | _Sam-- | i would have a use for a 0 touch auto-config for the phones |
| 20:28.23 | _Sam-- | like you turn it on, it gets DHCP, and configs itself somehow |
| 20:28.23 | [av]bani | no, you cant do that with grandstream yet though |
| 20:28.36 | [av]bani | its 1-touch config right now, till gs lets you config the path via dhcp |
| 20:28.43 | [av]bani | for sipura, snom its 0 |
| 20:29.00 | _Sam-- | my config is pretty quick anyway, it takes me literally 2 minutes or less per phone from the web interface... |
| 20:29.14 | _Sam-- | i think i could probably do it just as fast using that method as any other current method? |
| 20:29.16 | [av]bani | well this would be like 2 sec of cut+paste in a text editor :) |
| 20:29.28 | _Sam-- | you would have to open the web interface still |
| 20:29.32 | _Sam-- | and then reboot the phone? |
| 20:29.36 | _Sam-- | and have cfg-mac |
| 20:29.37 | _Sam-- | ? |
| 20:29.37 | [av]bani | and if you wanted to change the config of _all_ the phones at once.... |
| 20:29.44 | [av]bani | its 2 seconds, plus a reboot |
| 20:29.57 | _Sam-- | how do you generate mac-cfg.txt? |
| 20:30.01 | [av]bani | magic :) |
| 20:30.08 | _Sam-- | somehow you still have to know the mac of the phone |
| 20:30.14 | _Sam-- | do you have to manually enter it anyplace? |
| 20:30.15 | Nugget | well my ghetto asterisk-to-skype gateway appears to be pretty functional. |
| 20:30.31 | [av]bani | well, i can auto gen extensions from macs |
| 20:30.38 | [av]bani | that would be a bit icky though :) |
| 20:31.09 | _Sam-- | what im saying...is you have to generate the mac-cfg files...which means you need to know the mac and enter it (maybe) someplace...my arguiment is while you are doing that i can do the web interface just as fast? |
| 20:31.33 | [av]bani | unlikely |
| 20:31.43 | [av]bani | i can cut+paste faster than you can do web |
| 20:31.50 | _Sam-- | i see |
| 20:31.53 | [av]bani | and when you want to change config of all phones at once.... |
| 20:31.54 | _Sam-- | go to the status page of the phone |
| 20:31.56 | _Sam-- | copy mac address |
| 20:31.59 | _Sam-- | paste it |
| 20:32.01 | [av]bani | in mine its 2 seconds |
| 20:32.10 | [av]bani | yours... go to each phone... change config.. reboot |
| 20:32.34 | _Sam-- | luckily, my phone configs never change (havent in 1 year or more) |
| 20:32.39 | _Sam-- | just the firmware changes |
| 20:32.47 | [av]bani | plus, you have the config of all phones in a central location, unified format nomatter what hte vendor |
| 20:32.58 | _Sam-- | but i think you are right,...it would be faster maybe your way, but you still have to go to each phone and get the mac |
| 20:33.11 | [av]bani | no.. i could make it auto gen extensions from the mac |
| 20:33.15 | [av]bani | it already auto gens passwords |
| 20:33.58 | _Sam-- | in order to create the mac-cfg...you have to use the grandstream configuration thing? |
| 20:34.03 | [av]bani | no |
| 20:34.09 | [av]bani | which is the whole point i think :) |
| 20:34.15 | _Sam-- | they had a tool for creating the mac-cfg |
| 20:35.26 | _Sam-- | you just use the template? |
| 20:36.14 | [av]bani | no, it will emit the config data directly |
| 20:36.16 | *** join/#asterisk trelane_ (n=trelane@asterisk.sosdg.org) |
| 20:36.48 | trelane_ | I am using two cisco 7905's trying to call one phone from the other gives me the message "Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 20:36.48 | [av]bani | so no having to juggle config files, no need to clutter directories with piles of files |
| 20:36.52 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 20:37.26 | [av]bani | also.. if you ever change a phone from grandstream to something else.. just change the mac addr in the config file |
| 20:37.31 | [av]bani | and plug the new phone in |
| 20:37.38 | [av]bani | all settings are kept |
| 20:38.09 | sevard | twisted[asteria]: I can't seem to find the document you're recommending, can you help me? |
| 20:38.09 | _Sam-- | i didnt realize those setting were that universal |
| 20:38.14 | svartalfheim | here |
| 20:38.14 | sevard | find it, that is. |
| 20:38.27 | [av]bani | nat, extensions, etc. are |
| 20:38.32 | [av]bani | stun |
| 20:38.34 | [av]bani | proxies |
| 20:38.37 | [av]bani | context |
| 20:38.40 | [av]bani | overlap dialing |
| 20:38.42 | [av]bani | syslog |
| 20:38.48 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 20:39.15 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 20:39.23 | [av]bani | i could add more if i find more to share |
| 20:39.53 | [av]bani | of course theres nothing preventing vendor-specific stuff, in fact i let you populate the snom phonebook |
| 20:40.08 | [av]bani | i dont know if grandstream lets you do that yet |
| 20:44.00 | synthetiq | when using rf2c833 for dtmf, and you dial to fast, (using auto dialer) is it asterisks fault or what, because it only works if you dial slowly ...? |
| 20:44.30 | Mavvie | hmm... seems to be that nobody knows what channel juggling is! |
| 20:45.19 | Mavvie | (and I see it as "Moving call from channel 94 to channel 101") |
| 20:45.44 | trelane_ | I am attempting to get two internal extensions with cisco phones to call each other, http://pastebin.com/542156 displays the extensions.conf info for the phones. Has anyone seen this error before? |
| 20:46.13 | shido6 | cisco100 and cisco101 , are they registered? |
| 20:46.21 | shido6 | what do you have for them in sip.conf |
| 20:46.24 | shido6 | got a host? |
| 20:46.24 | trelane_ | shido6: yes |
| 20:46.31 | trelane_ | shido6: no they're dynamic |
| 20:46.39 | shido6 | pastebin the sip.conf for these 2 phones AND the [general] at the top |
| 20:47.30 | shido6 | dynamic is good - I'll wait for the pastebiin |
| 20:47.32 | shido6 | -i |
| 20:48.53 | trelane_ | shido6: updated |
| 20:50.02 | trelane_ | that by the way is all of extensions.conf, I've got a paired down version I'm using to try to fix this |
| 20:50.42 | trelane_ | all these phones are on the local subnet |
| 20:50.46 | DrData | does anybody know, how to transfer the CALLERIDNUM with a call transfer through capi? |
| 20:53.20 | trelane_ | shido6: ping? |
| 20:54.02 | *** join/#asterisk elg (n=fugalh@hfugal.NMSU.Edu) |
| 20:55.35 | tronix | Mavvie: I only know that * can move channels occasionally... I just don't know the exact criteria it uses |
| 20:56.09 | tronix | trelane_: hmm it's saying 'all circuits are busy' as reason for failure. 7905 has only one line? what does sip.conf look like? (minus the secret) |
| 20:56.14 | Mavvie | tronix: aha, thanks. |
| 20:56.38 | tronix | Mavvie: this is the reason why some people warn about not hardcoding channel names in AGI scripts or other tools, btw |
| 20:57.12 | Mavvie | tronix: that's not the problem here, the problem is that sometimes it tries to juggle it to an occupied channel and starts to panic and drops the call. |
| 20:58.03 | trelane_ | tronix: it's in the updated pastebin |
| 20:58.03 | Dr-Linux | how to kill active sip peer? |
| 20:58.14 | docelm0 | Anyone know of any place that has asterisk jobs posted? |
| 20:58.16 | trelane_ | tronix: and I have 2 7905's |
| 20:58.24 | tronix | trelane_: in 542156? I don't see sip.conf info there |
| 20:59.04 | trelane_ | http://pastebin.com/542165 |
| 20:59.06 | trelane_ | 165 :) |
| 20:59.08 | trelane_ | not 156 |
| 20:59.23 | Mavvie | docelm0: -biz mailinglist |
| 20:59.24 | trelane_ | odd it updated |
| 20:59.25 | trelane_ | ooh well |
| 21:02.19 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 21:03.19 | docelm0 | I figured that.. thanks mavvie |
| 21:04.24 | Dr-Linux | how to kill active sip channel ? |
| 21:04.33 | docelm0 | soft hangup channel |
| 21:06.08 | RoyK | is it possible to run zaptel 1.2 with asterisk 1.0_ |
| 21:06.09 | RoyK | ? |
| 21:06.23 | docelm0 | I have by accident.... |
| 21:06.47 | RoyK | docelm0: run zap 1.0? |
| 21:06.50 | RoyK | er |
| 21:06.50 | RoyK | 1.2 |
| 21:07.06 | docelm0 | 1.2 w/ ast 1.0.. Worked fine for the most part |
| 21:07.29 | *** join/#asterisk flavour (n=InveneoR@88-111-125-168.dynamic.dsl.as9105.com) |
| 21:07.41 | RoyK | most? |
| 21:09.05 | tdonahue | hi all |
| 21:09.07 | docelm0 | I had some issues with hung ZAP channels |
| 21:09.29 | tdonahue | for dtmf, if our carrier offers inband or RFC2806, which is the best option for asterisk? |
| 21:09.54 | *** part/#asterisk flavour (n=InveneoR@88-111-125-168.dynamic.dsl.as9105.com) |
| 21:10.02 | *** join/#asterisk mzo (i=user@ool-435193b3.dyn.optonline.net) |
| 21:10.11 | mzo | clone idiots suck :P |
| 21:11.11 | docelm0 | tdonahue, I would use RFC |
| 21:11.26 | docelm0 | Inband only works with g711 and well that eats TONS of bandwidth |
| 21:12.56 | RoyK | docelm0: i was only thinking of using the rtc ztdummy |
| 21:13.16 | tdonahue | docelm0: well, the channels are coming in ulaw anyway because of a customer requirement (problems with 729 compression on his IVR) |
| 21:13.47 | MikeJ[Laptop] | tdonahue, don't use inband and 729 will work fine I bet |
| 21:14.06 | MikeJ[Laptop] | wait.. 2806? |
| 21:14.10 | MikeJ[Laptop] | what is that |
| 21:14.23 | tdonahue | MikeJ[Laptop]: that is what i'm trying to figure out... |
| 21:14.26 | MikeJ[Laptop] | heh |
| 21:14.57 | MikeJ[Laptop] | that's not dtmf is what that is |
| 21:15.01 | tdonahue | we are currently having DTMF problems, which is why we are I'm asking about 2806 |
| 21:15.05 | MikeJ[Laptop] | URLs for Telephone Calls |
| 21:15.05 | MikeJ[Laptop] | <PROTECTED> |
| 21:15.21 | MikeJ[Laptop] | 2833 is dtmf in the rtp |
| 21:15.33 | MikeJ[Laptop] | either that or sip info will solve your issue |
| 21:16.05 | RoyK | g729 and inband dtmf does not work.... |
| 21:16.22 | tdonahue | i don't think our carrier supports sip info, but i'm currently trying to find out about 2833 |
| 21:16.32 | MikeJ[Laptop] | RoyK, is there an echo in here? |
| 21:16.59 | MikeJ[Laptop] | those are pretty much your options with asterisk.. |
| 21:17.09 | MikeJ[Laptop] | there are a few more out there.. but they are pretty obscure |
| 21:17.24 | MikeJ[Laptop] | ok... who can name all of the dtmf methods you can do with sip? |
| 21:17.45 | MikeJ[Laptop] | I can name 4... |
| 21:17.46 | tdonahue | with that statement i now understand why our carrier is using 2806 :/ |
| 21:17.46 | MikeJ[Laptop] | hmm |
| 21:17.50 | MikeJ[Laptop] | there are more |
| 21:17.54 | RoyK | erm |
| 21:18.07 | RoyK | what part of asterisk is it that's using ztdummy? chan_zap? |
| 21:18.09 | MikeJ[Laptop] | 2806 is not a dtmf method tdonahue |
| 21:18.50 | trelane_ | I am attempting to get two internal extensions with cisco phones to call each other, http://pastebin.com/542165 displays the extensions.conf info for the phones. Has anyone seen this error before? |
| 21:20.26 | tdonahue | trelane_: is the network the phones are on routable to from your asterisk box? |
| 21:21.01 | tdonahue | can you ping the phones from your asterisk box? |
| 21:22.41 | trelane_ | tdonahue: they're all local lan |
| 21:22.59 | Zodiacal | does asterisk benefit from 64Bit cpus or Hyperthreading? |
| 21:23.01 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.80.101.Dial1.Chicago1.Level3.net) |
| 21:23.59 | trelane_ | tdonahue: both are quite pingable |
| 21:24.28 | mattwj2005 | so what happened? irc got attack? |
| 21:27.28 | mattwj2005 | I heard something about the clone wars.....I thought they were talking Star Wars ;) |
| 21:27.30 | tdonahue | trelane_: try adding "context=internal-extension" to your sip accounts. that is the only other thing I can see. |
| 21:28.12 | mzo | it wasn't just this network, seems a lot of places got hit |
| 21:29.34 | mattwj2005 | thanks mzo....darn people anyways...thanks for the info |
| 21:29.53 | trelane_ | tdonahue: ok noted |
| 21:29.55 | mattwj2005 | is a hacker not a cracker.....I am only a good guy with computers :) |
| 21:32.42 | j0n | does anyone know how to set up hints for presence in AEL? |
| 21:36.38 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 21:40.16 | *** part/#asterisk mattwj2005 (n=Matt@dialup-4.159.80.101.Dial1.Chicago1.Level3.net) |
| 21:41.05 | docelm0 | RoyK, thats doable.. I was using it for the TDM410 |
| 21:41.28 | docelm0 | tdonahue, If they are coming ulaw then get it rfc will make it easier to convert down the road if need be. |
| 21:42.43 | RoyK | ok |
| 21:42.50 | RoyK | i just need ztdummy with rtc support |
| 21:44.45 | *** join/#asterisk postel (n=jk@unaffiliated/postel) |
| 21:46.42 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 21:51.04 | *** join/#asterisk saftsack (n=saftsack@p54A7DB63.dip.t-dialin.net) |
| 21:55.25 | *** join/#asterisk Qwell (i=north@outboxes.com) |
| 21:55.31 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.80.101.Dial1.Chicago1.Level3.net) |
| 21:56.38 | *** join/#asterisk bjames (n=bjames@67-102-228-17.adsl.lbdsl.net) |
| 21:56.41 | bjames | hi |
| 21:57.20 | bjames | I just got in my Rhino Channel bank |
| 21:57.24 | Qwell | you're up late :p |
| 21:57.31 | shido6 | $50 Million, Qwell |
| 21:57.39 | Qwell | wow |
| 21:57.42 | bjames | I'm trying to figure out what modules I need to load on my Linux machine |
| 21:57.43 | shido6 | the ip phones alone are 8 |
| 21:57.50 | *** join/#asterisk festr_ (n=festr@gw-sitel.lam.cz) |
| 21:57.54 | shido6 | you want in? |
| 21:58.07 | shido6 | i think I may pull Digium in on this one. |
| 21:58.10 | Qwell | of course |
| 21:58.20 | bjames | I got a Red-Fone bridge connected to the Asterisk Box |
| 21:58.51 | bjames | ztcfg is telling me Zaptel dynamic span creation failed: Function not implemented |
| 21:58.56 | shido6 | how do you like that thing bjames ? |
| 21:58.57 | festr_ | hello, i've strange crashes with last asterisk svn and probably all versions since 1.2.0. |
| 21:58.57 | festr_ | (gdb) bt |
| 21:58.58 | festr_ | #0 0xb7dd5c17 in malloc () from /lib/tls/libc.so.6 |
| 21:58.58 | festr_ | #1 0xb76f993b in iax_frame_new (direction=1, datalen=1840) at iax2-parser.c:920 |
| 21:58.58 | bjames | this is Asterisk 1.0.10 |
| 21:59.10 | Qwell | shido6: I'll be home in a few hours if you want to discuss it more |
| 21:59.19 | RoyK | festr_: bugs.digium.com? |
| 21:59.20 | festr_ | this bt, does it mean, that last func. was malloc? |
| 21:59.30 | shido6 | I have to write up some questions and submit them TODAY |
| 21:59.31 | shido6 | so hurry |
| 21:59.32 | shido6 | <PROTECTED> |
| 21:59.43 | Qwell | shido6: msg me? |
| 21:59.47 | festr_ | RoyK before commit bug i need to understand something |
| 21:59.52 | RoyK | ok |
| 22:00.05 | festr_ | RoyK what this gdb bt mean? |
| 22:00.14 | RoyK | pastebin the whole thing |
| 22:00.15 | festr_ | RoyK #0 is last "instruction" ? |
| 22:00.16 | RoyK | also |
| 22:00.19 | festr_ | RoyK ok |
| 22:00.27 | RoyK | is this compiled with -O6 or -O0? |
| 22:00.35 | RoyK | if -O>0, recompile with -O0 |
| 22:00.43 | RoyK | the optimisation can fsckup the bt |
| 22:00.56 | saftsack | ai |
| 22:00.59 | Qwell | shido6: I'll just take lunch, and we can talk now if you'd like |
| 22:01.05 | saftsack | some germans with isdn here? |
| 22:01.20 | shido6 | # ? |
| 22:01.34 | Qwell | shido6: I don't have my cell with me today. Is IRC okay? |
| 22:01.35 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 22:01.40 | festr_ | RoyK http://pastebin.com/542304 |
| 22:01.44 | RoyK | but the first (#0) is the call that crashed the box. the #1 is the functiona that called #0 etc |
| 22:01.45 | shido6 | ok |
| 22:02.03 | festr_ | RoyK i've made make valgrind, but somtimes i see -O6 |
| 22:02.18 | RoyK | make valgrind is dead on 1.2 |
| 22:02.26 | RoyK | edit the makefile and change O6 |
| 22:02.34 | festr_ | there are many makefiles |
| 22:02.39 | festr_ | :( |
| 22:02.41 | RoyK | the root one |
| 22:02.44 | festr_ | ok |
| 22:02.53 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 22:02.58 | RoyK | most rely on that |
| 22:03.14 | festr_ | it start doing since 2006/25/01 |
| 22:03.25 | festr_ | i thing, this day was timebomb |
| 22:03.54 | RoyK | festr_: iirc, yes |
| 22:03.57 | RoyK | festr_: what version? |
| 22:04.30 | RoyK | anyway |
| 22:04.35 | RoyK | that trace doesn't look good |
| 22:04.42 | RoyK | recompile with -O0 and try again |
| 22:05.14 | festr_ | RoyK i've some CVS HEAD version which does not affect this bug but preventively, i've upgraded to latest 1.2 SVN and from this point it starts crashing |
| 22:06.28 | RoyK | festr_: try this to find all makefiles with optimisation |
| 22:06.29 | RoyK | find . -name Makefile -exec grep -H -- -O[0-9] {} \; |
| 22:06.30 | festr_ | RoyK but i've return to older CVS HEAD but still coredumps. so upgrade again to latest SVN and still coredumps, very strange |
| 22:06.45 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 22:06.50 | RoyK | create a new dump with -O0 and call again, please :P |
| 22:06.58 | festr_ | ok :) |
| 22:06.59 | RoyK | it's quite hard to debug without a good backtrace |
| 22:07.43 | RoyK | bloody kernel upgrades |
| 22:07.46 | festr_ | btw whats the purpose of O6? there is only O3 deined in gccc |
| 22:07.54 | RoyK | i know |
| 22:07.57 | RoyK | not my fault |
| 22:08.25 | RoyK | some people do -O9 just in case the compiler can do it |
| 22:08.25 | festr_ | i've post this question to dev list with strange answer :) |
| 22:08.39 | RoyK | without thinking of what the compiler really _can_ do with the code, doing that |
| 22:08.55 | festr_ | dont understand |
| 22:10.18 | elg | so with digium fxo card, apparently the calling end can hear call waiting indication too? |
| 22:10.26 | elg | is there a way to turn that off? |
| 22:10.38 | festr_ | RoyK libpri -O0 too.. ? |
| 22:10.59 | RoyK | festr_: since the dump is in chan_iax2..... doubt it'll be necessary |
| 22:11.05 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
| 22:11.17 | RoyK | festr_: but i doubt it'll hurt |
| 22:11.20 | g4m | what is the easiest way to figure out what sip clients are currently in use? (hook off) |
| 22:11.44 | Qwell | g4m: sip show channels, should show you |
| 22:12.09 | g4m | perfect thank you |
| 22:12.11 | Qwell | there isn't really an "offhook" state for SIP though, so it'll only be active calls |
| 22:12.31 | RoyK | g4m: you can't sense hook status, since that isn't reported to asterisk, sip show channels shows those ringing or talking |
| 22:12.47 | Qwell | it is with sccp ;] |
| 22:13.15 | RoyK | heh |
| 22:13.25 | trelane_ | any idea how to make a 7905 not try to login as phone? sip debug is starting to give me an idea as to what hte issue is? |
| 22:13.26 | RoyK | iirc the sccp documentation is quite hard to get |
| 22:13.36 | Qwell | eh...who needs it? |
| 22:13.41 | festr_ | RoyK btw, always last log in full.log before crash was: |
| 22:13.43 | festr_ | RoyK Feb 5 16:48:04 DEBUG[27479] chan_iax2.c: Ooh, voice format changed to 8 |
| 22:14.07 | Qwell | trelane_: Isn't it a phone? |
| 22:14.09 | RoyK | it changed?? |
| 22:14.13 | RoyK | 8 is alaw iirc |
| 22:14.19 | festr_ | RoyK i know |
| 22:14.33 | Qwell | festr_: Is there a zap channel in there anywhere at all? |
| 22:14.43 | festr_ | RoyK all calls was to or from 4xE1 |
| 22:14.47 | Qwell | I just saw a bug a minute ago... |
| 22:15.01 | RoyK | should be alaw in the first place..... |
| 22:15.03 | RoyK | strange |
| 22:15.11 | Qwell | http://bugs.digium.com/view.php?id=6421 |
| 22:15.13 | RoyK | festr_: just reproduce it with -O0 and full debug |
| 22:15.18 | Qwell | festr_: Go look at that bug... |
| 22:15.44 | festr_ | RoyK i hope it will reproduce. i'm just in process of chanign makefiles |
| 22:16.08 | trelane_ | Qwell: yeah but it's just bloodyminded to think it's going to use the username "phone"! |
| 22:16.28 | Qwell | trelane_: That is in the settings of the phone |
| 22:16.57 | RoyK | festr_: jus a sec :) |
| 22:17.01 | festr_ | RoyK anyway, another box with same hardware with 3x bigger load and no crash for weeks (exactly same version of CVS HEAD). but different kernel 2.6.14 (gcc 4.0.3). server where i could reproduce crashes is 2.6.12 gcc 4.0.2 |
| 22:17.04 | bjames | ztcfg is telling me Zaptel dynamic span creation failed: Function not implemented << that module I needed was ztd-eth and ztdynamic |
| 22:17.08 | Qwell | RoyK: Make him look at that bug :P |
| 22:17.09 | trelane_ | Qwell: not according to what the web config engine says |
| 22:17.19 | _Sam-- | hey file, by default, we cant call hawaii? |
| 22:17.33 | Qwell | _Sam--: us48 iirc |
| 22:17.39 | _Sam-- | ugh |
| 22:17.44 | _Sam-- | they allow incoming from HI |
| 22:17.55 | Qwell | because it doesn't cost extra for them to call you |
| 22:18.01 | Qwell | or, maybe it does...dunno |
| 22:18.01 | _Sam-- | sure it does |
| 22:18.24 | _Sam-- | its an option...to allow inbound from HI and AK |
| 22:18.29 | _Sam-- | HI = like 10c / min |
| 22:18.34 | RoyK | festr_: find . -name Makefile -exec perl -pe 's/\-O\d/-O0/g' -i {} \; |
| 22:18.35 | _Sam-- | AK = worse |
| 22:18.36 | festr_ | Qwell it seems that this bug is different |
| 22:18.53 | festr_ | RoyK nice one thanks |
| 22:19.25 | _Sam-- | hey file, im going home...if you get this, and its possible...please allow us to call out to HI...i will pay the extra per minute for those calls. |
| 22:20.13 | festr_ | RoyK it could be done with sed |
| 22:20.19 | festr_ | RoyK you like perl ? :) |
| 22:20.21 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:20.28 | RoyK | it could be done with lots of other stuff as well |
| 22:20.37 | RoyK | but i tend to use perl |
| 22:20.42 | RoyK | why not? :) |
| 22:20.47 | mattwj2005 | are there any disadvantages of using iax for voip service providers? |
| 22:21.08 | RoyK | it's not like it takes a lot more time to fork/exec perl as compared to sed when you're doing it like 10 times :) |
| 22:21.08 | festr_ | only advantages :) |
| 22:21.36 | mattwj2005 | I was thinking like security....reliablity...anything like that |
| 22:22.06 | festr_ | RoyK :) i have seen scripts in perl: `echo "$FOO" >> /tmp/1`... this is very unusual :) |
| 22:22.08 | RoyK | mattwj2005: you will miss one great opportunity if you use IAX |
| 22:22.27 | mattwj2005 | what is that? |
| 22:22.29 | RoyK | mattwj2005: the hours after hours of fighting NAT problems with SIP |
| 22:22.39 | mattwj2005 | lol |
| 22:22.40 | mattwj2005 | :P |
| 22:23.17 | mattwj2005 | I am currently using voipjet for outgoing calls...they are iax based |
| 22:23.29 | RoyK | SIP was created by people who thought the world was nice, all people kind and official IP addresses to everyone, no firewalls and peace on earth and prolly just about enough pot |
| 22:23.40 | mattwj2005 | I really like it......I don't know about incoming though |
| 22:23.59 | *** join/#asterisk adibar (n=adibar@217-162-123-170.dclient.hispeed.ch) |
| 22:25.02 | mattwj2005 | I did once configure incoming sip....but I am sure iax incoming can't be that bad |
| 22:25.38 | festr_ | mattwj2005 with trunking and using g729 or ilbc you can safe a lot of packets :) |
| 22:26.13 | mattwj2005 | by using iax, festr_? |
| 22:26.19 | festr_ | mattwj2005 yes |
| 22:26.35 | mattwj2005 | okay...sounds good |
| 22:26.37 | festr_ | mattwj2005 for more then 10 calls you save 10kbits per call. |
| 22:26.58 | mattwj2005 | per call....1kbit per second? |
| 22:27.09 | festr_ | mattwj2005 10 calls with g729 ~ 250kbit with trunking it is 128kbit |
| 22:27.38 | mattwj2005 | nice.....voice quality too? |
| 22:27.53 | festr_ | mattwj2005 and second major save with trunking is that packets are sent every 20ms so less overhead |
| 22:28.09 | file | you're.... you're my number one |
| 22:30.21 | PoWeRKiLL | hi |
| 22:30.34 | adibar | Hi everyone. |
| 22:31.21 | PoWeRKiLL | I'm trying to compile chan_bluetooth.c with asterisk 1.2.4 and it's complaining about channel_pvt.h but I can't find this file in * any idea ? |
| 22:31.40 | festr_ | btw CLI> stop when convenient |
| 22:31.45 | festr_ | console will freez |
| 22:31.49 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 22:32.00 | festr_ | so i've to reconnect |
| 22:32.07 | ManxPower | Anyone know of a cheap 1010xxx dial around service? |
| 22:32.19 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 22:33.08 | Qwell | ManxPower: In 2006? Probably not very many left |
| 22:33.22 | *** join/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m) |
| 22:33.26 | ManxPower | Qwell, well my IP service is 1000ms latency.... |
| 22:33.26 | adibar | PoWeRKiLL: Use the CVS-version of chan_bluetooth. That did compile for me... But none of my handies seems to be compatible :-( |
| 22:33.53 | ManxPower | I'll give my grandmother a call, she uses one of those alot. |
| 22:34.00 | ManxPower | Qwell, and there are zillions of them |
| 22:34.01 | *** part/#asterisk steelcase (n=stevec@63.173.198.31) |
| 22:34.06 | Qwell | oh |
| 22:34.14 | *** part/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m) |
| 22:34.40 | harryvv | anyone here know of a voip wholsaler of voip 604 or 778 numbers? |
| 22:35.13 | GerbilNut | any recommendations for putting in a delay before a wav is played? |
| 22:35.26 | Qwell | ManxPower: Depending on the price, it might be cheaper to just setup a tollfree account on a server somewhere, and dial out through that |
| 22:35.27 | adibar | Wait(() ;-) |
| 22:35.39 | Qwell | figure 4c/min tops for that |
| 22:35.52 | PoWeRKiLL | adibar I try to get it via SVN but I get an error |
| 22:36.40 | adibar | err... adibar="quite newbie"... SVN ? |
| 22:36.53 | [av]bani | anyone got native moh working in * ? |
| 22:39.34 | Mark_Halverson | anyone using * with SS7 ??? saw some dead projects on the wiki...need the ability to do a TCAP dip on 800 call and log CIC in the CDR |
| 22:40.06 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-216-21.rockynet.com) |
| 22:42.56 | synthetiq | Feb 6 17:44:26 NOTICE[5284]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 22:44.22 | ManxPower | Mark_Halverson, you won't be able to do that with Asterisk AFIK |
| 22:45.27 | adibar | PoWeRKiLL: Sorry, my fault. It was the app_voicechanger that has been usable from CVS. For chan_bluetooth use the following URL: http://www.thetechguide.com/howto/asterisk/chanbluetooth.html |
| 22:46.22 | adibar | PoWeRKiLL: with that it compiles fine... |
| 22:48.00 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 22:48.09 | *** join/#asterisk kn0x (n=atlantic@71.194.235.251) |
| 22:50.28 | *** join/#asterisk MoutaPT (i=MoutaPT@85.139.183.206) |
| 22:50.57 | MoutaPT | Hello, does any one has configured any CheckPoint Firewall with Asterisk |
| 22:50.59 | MoutaPT | on DMZ |
| 22:51.01 | MoutaPT | ? |
| 22:51.02 | _Sam-- | file: THANKS. |
| 22:52.06 | g4m | Is there a way to make a call file wait for the creator to start a connection, i.e. after the call file is grabbed by asterisk not start outbound call until the person making the call picks up their phone. |
| 22:52.12 | ManxPower | MoutaPT, not any different than any other firewall |
| 22:52.38 | ManxPower | you need to open up the SIP port (5060) and the RTP ports (defaults to 10,000 - 20,000, but can be changed), all ports are UDP, of course. |
| 22:52.46 | MoutaPT | ManxPower, in fact i don't know what is happening all the ports are opened |
| 22:52.57 | MoutaPT | some times i register well sip phones |
| 22:53.01 | MoutaPT | sometimes not |
| 22:53.09 | MoutaPT | also only get audio with |
| 22:53.12 | ManxPower | MoutaPT, Any NAT involved? |
| 22:53.13 | MoutaPT | nat=route |
| 22:53.15 | MoutaPT | no NAT |
| 22:53.36 | adibar | g4m: give the file a date in the future with "touch" |
| 22:54.34 | Mark_Halverson | anyone having problems or success conecting * to a MetaSwitch ??? |
| 22:54.45 | MoutaPT | i really get in troubles with network team, they don't know about asterisk ... |
| 22:55.23 | *** join/#asterisk [Atlas] (n=whois@216.190.144.90) |
| 22:55.43 | MoutaPT | and i'm the "client" of them |
| 22:55.54 | [Atlas] | is there any tftp servers i can use with linux that i can specify an interface to listen on rather than an ip address |
| 22:55.56 | [Atlas] | ? |
| 22:55.58 | MoutaPT | they just answer me, everything is opened |
| 22:55.59 | adibar | outch |
| 22:56.21 | MoutaPT | but CheckPoint system has a kind of SmartDefence |
| 22:56.34 | MoutaPT | i've been looking and it is just monitoring |
| 22:56.52 | MoutaPT | that's why i asked if some one has worked with this firewall |
| 22:56.57 | ManxPower | if there's no nat, then leave nat= out. |
| 22:57.03 | MoutaPT | nat=out? |
| 22:57.10 | MoutaPT | or empty? |
| 22:57.20 | WasPhantom | no ;nat= |
| 22:57.24 | WasPhantom | I'm guessing |
| 22:57.36 | ManxPower | um, remove any nat= lines |
| 22:57.52 | MoutaPT | ok, i only get sip phones registring |
| 22:57.57 | MoutaPT | until now with nat=no |
| 22:58.02 | MoutaPT | or nat=route |
| 22:58.10 | MoutaPT | nat=route audio is ok |
| 22:58.22 | MoutaPT | otherwise no sound for calls to asterisk services |
| 22:58.27 | ManxPower | if there is no NAT involved then you do not need nat= lines. |
| 22:58.30 | ManxPower | if you need them, then there is nay involved. |
| 22:58.34 | MoutaPT | i mean 8200 no sound |
| 22:58.42 | MoutaPT | zapata call sound is ok |
| 22:58.54 | ManxPower | If nat is involved then there is one set of things you need to do, if there is no NAT involved then you have to do a different set of things. |
| 22:59.12 | MoutaPT | my info available is no NAT |
| 22:59.26 | MoutaPT | i've made externalip |
| 22:59.30 | MoutaPT | and localnet |
| 22:59.34 | MoutaPT | in sip conf |
| 22:59.38 | ManxPower | MoutaPT, you can do a "sip debug" in the Asterisk CLI. If you see private addresses then you know they lied. |
| 22:59.58 | ManxPower | MoutaPT, those options will break things if there is no NAT. |
| 23:00.07 | MoutaPT | Thanks |
| 23:00.18 | MoutaPT | yes my local ip sip client appears |
| 23:00.21 | MoutaPT | on sip headers |
| 23:00.28 | [av]bani | hmm... moh is stuttering, and i can't see any reason for it |
| 23:01.23 | *** join/#asterisk wunderkin (n=kev@wsip-24-120-65-156.lv.lv.cox.net) |
| 23:03.47 | MoutaPT | is there any difference between nat=no |
| 23:03.52 | MoutaPT | or no lines about nat? |
| 23:05.30 | Qwell | bbl |
| 23:05.32 | j0n | does anyone know how to use hints for presence in ael? |
| 23:05.34 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 23:06.29 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net) |
| 23:07.17 | file | hail |
| 23:07.29 | trelane | hrm I only thought it was supposed to rain today |
| 23:09.21 | MoutaPT | Does any one knows the difference between nat=no ; nat=route and no line about nat? |
| 23:10.33 | *** join/#asterisk exstatica (i=exstatic@redline.mednor.net) |
| 23:11.55 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 23:12.52 | ManxPower | MoutaPT, nat=route is to work around bugs in one model of the Uniden SIP phone. |
| 23:13.11 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:13.17 | shmaltz | is the list down? |
| 23:13.22 | adibar | Could anyone use wildcards for the called number on a construct like: exten => _1234X./5551234,2,blah ? |
| 23:13.26 | CaT[tm] | is there a nice alternative to AMP (other then vi :)? :) preferably something I wont have to do open-heart surgery on to install right. |
| 23:13.53 | crich1999 | CaT[tm], try voiceone.it |
| 23:14.53 | crich1999 | i mean www.voiceone.it |
| 23:14.59 | CaT[tm] | crich1999: thanks. I'll give it a look (btw http://v- yeah :) |
| 23:15.22 | MoutaPT | thanks ManxPower!! |
| 23:15.35 | shmaltz | anybody know if the list is down? |
| 23:15.48 | MoutaPT | could you just tell me diference between nat=no nat=never and no lines about nat ? |
| 23:17.17 | *** join/#asterisk apardo (n=apardo@87.218.44.253) |
| 23:18.50 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 23:20.00 | g4m | adibar: thanks |
| 23:21.15 | adibar | g4m: welcome |
| 23:21.25 | MoutaPT | is there any way else than sip debug |
| 23:21.42 | MoutaPT | to see what * is doing while trying to register sip clients |
| 23:22.18 | adibar | ethereal or ngrep ;-) but that's even worse |
| 23:23.03 | *** join/#asterisk tuxinator_linux (n=tuxinato@m090e36d0.tmodns.net) |
| 23:23.10 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 23:23.24 | *** join/#asterisk airdog (n=kvirc@S01060007e9584bcd.vs.shawcable.net) |
| 23:23.36 | wundaboy | what is the default username/password for the web control of a polycom ip500? |
| 23:23.36 | g4m | adibar; although 'date --date "1 minute" +%c' might work better |
| 23:23.46 | [TK]D-Fender | wundaboy : Polycom / 456 |
| 23:23.59 | [TK]D-Fender | wundaboy : Althrough you really should provision it... |
| 23:24.18 | wundaboy | what do you mean provision it? |
| 23:24.35 | [TK]D-Fender | wundaboy : Configure it from a FTP/TFTP server |
| 23:24.45 | wundaboy | oh |
| 23:24.50 | [TK]D-Fender | wundaboy : Thats where all the real settings are |
| 23:24.54 | wundaboy | good call, can i find info about that on voip-info ? |
| 23:25.27 | adibar | g4m: If it works... fine. But I followed the syntax for touch from the man-page |
| 23:26.11 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 23:26.48 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-9.cybersurf.com) |
| 23:26.52 | *** part/#asterisk mattwj2005 (n=Matt@dialup-4.159.80.101.Dial1.Chicago1.Level3.net) |
| 23:27.52 | adibar | g4m: btw. in the newer man-pages of date you don't find the hint for future or past date-usage anymore. So hopefully you still have a box with stone-age linux on it where you will find the hint inside "man date" |
| 23:28.36 | *** join/#asterisk kamileon (n=kamileon@68.62.190.253) |
| 23:28.55 | g4m | adibar: it works with debian/linux 2.6 |
| 23:29.01 | *** join/#asterisk lithi (n=interp3@Toronto-HSE-ppp3858329.sympatico.ca) |
| 23:29.04 | kamileon | hello #asterisk |
| 23:29.30 | adibar | g4m: it works, but it is not mentioned inside the man ;-) |
| 23:30.27 | shido6 | anyone live in oklahoma? |
| 23:30.32 | lithi | Whats a good way of limiting calls? Groups? |
| 23:31.02 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 23:31.27 | MoutaPT | any one Works with asterisk Portugal? |
| 23:32.18 | airdog | my asterisk server (v 1.1 on Mandriva 2006) won't authenticate incoming calls from my VOIP provider. Everything else works fine |
| 23:32.29 | airdog | would appreciate help |
| 23:32.38 | rob0 | kamileon, that IP of yours is just 5 hops away from me, hey, notice that Comcast has fubar'ed their rDNS? |
| 23:32.56 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net) |
| 23:32.56 | rob0 | (or turned it off maybe)' |
| 23:33.24 | *** join/#asterisk CodeGuru (i=CodeGuru@82.201.227.209) |
| 23:33.28 | kamileon | rob0: weird so youre hsv local too? |
| 23:33.39 | CodeGuru | hello everybody |
| 23:33.40 | rob0 | Florence actually. |
| 23:33.48 | *** join/#asterisk fiber0pti (n=John@invinine.com) |
| 23:33.53 | kamileon | great, this area is the best. |
| 23:34.16 | CodeGuru | any expert could give me a hand here, i just need some advise on using asterisk on my upcomming solution ?! |
| 23:34.26 | kamileon | im about to throw my * boxen out the window over a simple issue i cant figure out, and i should be able to |
| 23:34.33 | kamileon | nice to meet you rob0 btw |
| 23:34.33 | fiber0pti | I'm using polycom 500s and I'm getting a lot of messages about how extensions are getting lagged and are unreachable and then a couple seconds later they become reachable again. They are all on a lan going through a managed switch which is not heavily used. Any ideas? |
| 23:35.06 | rob0 | kamileon: let me know so I can be standing under the window :) |
| 23:35.06 | CodeGuru | :) nice rob0 , but can you help me ? |
| 23:35.14 | Mavvie | kamileon: you have to give a little bit more information than that. |
| 23:35.27 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-89.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:35.31 | kamileon | Mavvie : i will when i break down and request assistance ;) |
| 23:35.37 | kamileon | heh |
| 23:35.42 | rob0 | CodeGuru: not likely, but I can listen :) |
| 23:35.49 | Mavvie | aha, you're in the "at least threaten myself with it" phase. |
| 23:35.55 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
| 23:36.08 | CodeGuru | thanks rob0, here is my question: |
| 23:36.12 | Umaro | Hey guys.. anyone know of a digium supplier/somewhere to get digium cards in india? |
| 23:36.26 | kamileon | simply put, i just have a tdm40b and a x101p in one box and im just trying to match 8XXXXXXX to send my call out zap/5 * |
| 23:36.38 | kamileon | see why i feel stupid now |
| 23:36.46 | RoyK | just upgraded to 2.6.15.2 and it seems asterisk is only scheduled on two logical cpus |
| 23:36.48 | rob0 | 40b is what, a single FXS? |
| 23:36.52 | RoyK | it doesn't only seem so.... |
| 23:36.54 | kamileon | 4 fxs |
| 23:37.01 | kamileon | i only use 1 though |
| 23:37.23 | rob0 | hey, wanna sell some of the others? :) |
| 23:37.25 | kamileon | want some cheap fxs module(S) |
| 23:37.28 | rob0 | YES |
| 23:37.29 | CodeGuru | i have requirement to provide the sales department an integrated solution of receiving calls from pbx to thier computers (voip) and the ability to record or transfere or conference those calls ? can this be done with asterisk as the phone server ? |
| 23:37.37 | kamileon | yes i only *need* two of them |
| 23:37.58 | rob0 | Two would be wonderful for me! I have a TDM with one. |
| 23:38.01 | PoWeRKiLL | thanks adibar |
| 23:38.20 | kamileon | module 3 is failing to calibrate or whatever now, it worked 3 days ago fine, dont know how to diagnose the issue |
| 23:38.20 | rob0 | ok, so I am here to help you kamileon :) |
| 23:38.21 | adibar | PoWeRKiLL: Did it help ? |
| 23:38.42 | rob0 | ztcfg -vvv |
| 23:38.47 | CodeGuru | rob0 -> did u read my question ? |
| 23:38.52 | Umaro | CodeGuru: Definately. |
| 23:39.29 | kamileon | rob0: i just took modules 3 and 4 out of zapata.conf all together, so im running only 2 of them |
| 23:39.34 | CodeGuru | umaro: im not a linux guy, so is there a complete fool proof guide or tutorial on installing and configuring Asterix ? |
| 23:40.00 | RoyK | hm |
| 23:40.54 | airdog | my asterisk server (v 1.1 on Mandriva 2006) won't authenticate incoming calls from my VOIP provider. Everything else works fine. Would really appreciate help!! |
| 23:41.10 | distortion | anyone seen a limit to the number of extensions defined? |
| 23:41.14 | rob0 | kamileon: maybe it's no longer zap/5? |
| 23:41.29 | adibar | airdog: firewalled ? |
| 23:41.32 | *** join/#asterisk bryan2 (n=Miranda@c-67-164-201-80.hsd1.ut.comcast.net) |
| 23:41.36 | CodeGuru | guys, could any1 volanteer for my help here :(( |
| 23:41.45 | distortion | I have 10,000 extensions for a context i want to send to Congestion() heh |
| 23:41.50 | airdog | adibar: yes but as said outgoing calls work fine |
| 23:41.58 | ManxPower | kaldemar, that is a simple matter of your dialplan |
| 23:42.15 | airdog | adibar: 5060 is open |
| 23:42.15 | bryan2 | I have a small problem that I havn't been able to figure out. How do I configure asterisk so that it does not answer a line? |
| 23:42.16 | rob0 | CodeGuru: there's tons of information at the wiki, but it's not a simple thing. |
| 23:42.46 | CodeGuru | a link will be good here |
| 23:42.47 | adibar | airdog: Did you open the desired ports (UDP 5060 + UDP 10000-20000) ? |
| 23:42.50 | kamileon | rob0: ztcfg still reports it as 5 even though only 2 modules are loaded, i guess since theyre still physically onboard |
| 23:42.55 | rob0 | ~docs |
| 23:42.56 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:42.56 | bryan2 | I've messed with the context settings of the zaptel config files but have had no luck. |
| 23:43.04 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 23:43.06 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 23:43.32 | airdog | adibar: not the 10000 and up... that necessary? Then I have to open a LOT of ports |
| 23:43.33 | CodeGuru | ok, i will be back in a few minutes after seeing the site |
| 23:43.42 | airdog | codeguru: try http://www.voip-info.org/wiki/index.php?page=Asterisk |
| 23:43.47 | rob0 | kamileon: what do you see in the console when trying to make a call? |
| 23:44.05 | trelane | has anyone ever had a cisco 7905 registration problem with asterisk where the phone was sending invalid sip login info? |
| 23:44.12 | adibar | airdog: U still can restrict it to the asterisk-box. But for that works. |
| 23:44.22 | adibar | me |
| 23:44.29 | kamileon | it just shows the line picking up, then i dial 2 digts, any 2, and it hangs up |
| 23:45.28 | trelane | adibar, what was your solution? mine insists on loging in with the username "phone" |
| 23:45.44 | airdog | adibar: but if the ports were closed, would I still get the log message saying Failed to authenticate user in the logs? |
| 23:45.52 | rob0 | kamileon: what verbosity level? |
| 23:45.59 | rob0 | (increase it) |
| 23:46.01 | adibar | airdog: Zyxel 2000 ? |
| 23:46.14 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 23:46.15 | kamileon | rob0: i think i ran * with like 7 v's |
| 23:46.25 | airdog | adibar: no, just sip.conf no other hardware |
| 23:46.31 | kamileon | im just mismatching my extentions is all i believe |
| 23:46.52 | airdog | tried insecure=very or insecure=port,invite to no avail... |
| 23:46.57 | rob0 | hmmm, should be enough v's. |
| 23:47.02 | ManxPower | ~docs |
| 23:47.04 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:47.24 | litage | when a call goes softphone -> Asterisk -> softphone, only * needs g729 licenses. where are the g729 licenses need when the call goes softphone -> SER -> softphone, or softphone -> SER -> Asterisk -> Softphone ? |
| 23:48.02 | adibar | airdog: Some providers don't like the userame "asterisk". So you have got do put something like this inside the general section of sip.conf: useragent="da VoIPsta" |
| 23:48.28 | ManxPower | stupid bot |
| 23:49.05 | adibar | airdog: for the firewall-stuff check : http://www.voip-info.org/wiki/index.php?page=Asterisk%20firewall%20rules |
| 23:49.10 | airdog | adibar: well as said outgoing goes fine so provider accepts my stuff. The fail to authenticate are in my log |
| 23:49.30 | airdog | adibar: ok will do |
| 23:49.50 | adibar | airdog: hope that helps |
| 23:49.58 | distortion | heh, nice.. 10k extensions loaded and its working, load time wasnt long at all either |
| 23:52.12 | rene- | dummy question: is mac address supposed to be universally unique and can it be used to identified a vendor of a technology? |
| 23:52.25 | Qwell | rene-: yes and yes |
| 23:52.30 | Qwell | BUT... |
| 23:52.40 | Qwell | That isn't quite true in reality |
| 23:52.41 | airdog | adibar: thanks for the help, appreciate... |
| 23:52.49 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 23:52.58 | Qwell | There is nothing stopping A) MAC cloning, B) Vendors using MACs they shouldn't |
| 23:53.01 | adibar | airdog: welcome |
| 23:53.08 | *** join/#asterisk bryan2 (n=Miranda@c-67-164-201-80.hsd1.ut.comcast.net) |
| 23:53.16 | Qwell | rene-: in general they can be trusted though |
| 23:53.22 | adibar | ...and who helps me ? |
| 23:53.22 | bryan2 | How do I turn answering on a zaptel port off? |
| 23:53.33 | bryan2 | I'm soory. Off. |
| 23:53.33 | Qwell | bryan2: turn off answering? |
| 23:53.33 | rene- | Like the guidelines for private ip space, mosat people follow them but i have meet people that dont |
| 23:53.46 | Qwell | rene-: basically |
| 23:53.49 | bryan2 | Yeah. I don't want asterisk to pick up the line when it rings. |
| 23:53.55 | rene- | thanks men |
| 23:54.18 | ManxPower | bryan2, if you don't tell Asterisk to answer the line then it won't answer it./ |
| 23:54.29 | airdog | adibar: don't think I can help much but what's your issue? |
| 23:54.47 | adibar | airdog: just joking ;-) |
| 23:54.51 | bryan2 | Brilliant. I've installed Asterisk at home and I've got the default context pulled up. What do I put in it? |
| 23:55.11 | rob0 | rene-: I spoof a MAC at home to keep the same IP address from the ISP. |
| 23:55.25 | Qwell | rob0: I've had to do the opposite |
| 23:55.42 | airdog | adibar: well good thing, cause..... anyway, the firewall thing doesn't help, it still refuses to authenticate.... shit |
| 23:55.51 | kamileon | rob0: on my dials right, if i press the 8 (key to send call out zap) it waits for another digit before hanging up, anything but 8 immediatly hangs up |
| 23:56.15 | Qwell | rob0: once, my ISP wouldn't give me a new IP because of my MAC, so I had to spoof a random MAC to get a new one |
| 23:56.20 | rob0 | kamileon, sounds like a dialplan problem. |
| 23:56.33 | adibar | airdog: did ya try a sip debug peer <whatever> ? |
| 23:56.34 | CaT[tm] | wow. the voiceone.it install procedure is, well, wow. so clean. |
| 23:56.41 | kamileon | bryan2 : exten => s,1,noop(${CALLERID(num)}) |
| 23:56.44 | kamileon | i believe |
| 23:56.52 | kamileon | something like that |
| 23:56.59 | bryan2 | Excellent. Thanks Kam. |
| 23:57.02 | airdog | adibar: well it's on debug 10 and verbose 10, what more can I do? |
| 23:57.10 | ManxPower | kamileon, exten => s will ONLY EVER be called when there are NO digits received |
| 23:57.25 | airdog | adibar: not sure I know about this sip debug stuff |
| 23:57.32 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 23:57.35 | adibar | type: sip debug peer <name-of-peer> |
| 23:58.25 | adibar | airdog: on to get rid of that amount of data type : sip no debug |
| 23:58.27 | kamileon | bryan2 : im not sure if that will work, i got it from someone else helping me with my dialplan, i want my x100p NEVER to pick up the pstn line, thats what my guy at digium told me to use |
| 23:58.43 | Spida | can I get help here for getting my fritz pci to work with mISDN, too? |
| 23:58.46 | ManxPower | so basically exten => s is only matched when a call comes in on an analog FXO port, or when you are stupid and put immediate=yes. There are a couple of crappy VoIP providers that use it to, but not many. |
| 23:58.54 | kamileon | bryan2 : that exten was in context [incoming-zap] btw |
| 23:59.09 | bjames | My * server is crashing after I load the ztd_eth module and start * |
| 23:59.09 | airdog | adibar: done thanks |
| 23:59.15 | ManxPower | kamileon, don't put an exten => s in the context= in zapata.conf |
| 23:59.23 | Qwell | What is ztd_eth? |
| 23:59.39 | ManxPower | bjames, why do you want to use TDMoE? |
| 23:59.43 | adibar | airdog: with this you should see, where it is failing |
| 23:59.53 | bjames | ManxPower, I have a Redfone |
