| 00:04.30 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 00:04.42 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 00:09.49 | Innatech | Is anyone here experienced with connecting a Linksys RPT-300-NA ATA to * ? |
| 00:12.40 | carrar | We're gonna need you to move downstairs to the basement |
| 00:13.45 | fgwaller | Innatech: I did |
| 00:15.08 | Innatech | fgwaller: Did you have to do anything unusual to get it registered? I can't get it to register, and * fails it over to the default IVR. I can then reach softphone extensions, but if I try to call the RPT-300's assigned extensions, I get the out-of-service message. |
| 00:16.13 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:16.22 | fgwaller | switching off silence suppression and T38 I think, the rest was efault i think |
| 00:16.39 | fgwaller | Do you see in attempting to register at all? |
| 00:16.45 | Innatech | Do you recall the dial patter you set in the RPT? |
| 00:17.02 | Innatech | I see it attempting on the RPT's side, but not in the * output. |
| 00:17.20 | fgwaller | not off hand and the device is currently not hooked up anymore... |
| 00:17.48 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:18.06 | Innatech | ah. I've been nosing around list and forum archives, I imagine I'll find a dial pattern sooner or later. |
| 00:18.14 | Innatech | The registration thing is odd, though. |
| 00:18.17 | fgwaller | are your asterisk settings ok? |
| 00:18.38 | Innatech | My test X-ten extension and broadvoice trunk is working normally. |
| 00:18.56 | fgwaller | ohh... my faint memory comes back.... |
| 00:19.01 | Innatech | I'm trying now to add ATA's before I buy better multiple trunks and go live. |
| 00:19.32 | carl0s- | peice of shit. 192.168.253.200 0776608767 0a0a9cf7360 00102/00000 g729 No Init: INVITE |
| 00:19.41 | carl0s- | accept the INVITE goddamit |
| 00:19.50 | fgwaller | but the dialpattern only controlls on which dialed number the ATA tries to establish a connection instantly |
| 00:20.11 | Innatech | yes, I figure that's why I'm getting the default IVR somehow. |
| 00:20.45 | Zodiacal | anyone know how i can run something when a parked call times out? |
| 00:20.59 | fgwaller | I dont remember what the pattern syntax was, but it was in the online help of the AAT or on the linksys website |
| 00:21.33 | Innatech | Hmm. I'll check Linksys again. Didn't find anything very useful there earlier. |
| 00:21.43 | fgwaller | something like 2**,3**,1********** |
| 00:21.59 | Innatech | But you had no registration problems, eh? |
| 00:22.17 | fgwaller | no, not a bit... |
| 00:22.33 | Innatech | Hmm. Curious. Well, I'll fiddle with it some more. |
| 00:22.52 | fgwaller | but we had problems with the silence suppression on ;-) |
| 00:23.09 | Innatech | Nice to know. =) |
| 00:27.26 | Zodiacal | any ideas? |
| 00:28.51 | carl0s- | what does insecure=invite do? |
| 00:32.36 | *** join/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 00:34.25 | *** join/#asterisk jcaz (i=jcaz@the.depre.biz) |
| 00:34.31 | jcaz | Hello all |
| 00:35.18 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 00:35.28 | jcaz | i was just viewing a site when I saw this :( |
| 00:35.31 | jcaz | http://www.hotornot.com/r/?eid=AUHSAMR&key=LVW |
| 00:43.14 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 00:45.49 | Netgeeks | Anyone here running extremely high load asterisk systems? 400+ concurrent calls? |
| 00:47.23 | Netgeeks | insecure=invite causes asterisk to not challenge a user/peer upon an invite for the user/peer entry which you've set that directive. Otherwise if you have a secret specified, asterisk will challenge the invite |
| 00:48.07 | Netgeeks | I think is was added primarily to deal with situations where you might want to challenge a registration, but not challenge an invite for the same device |
| 00:48.17 | Netgeeks | but thats pure speculation by me |
| 00:48.53 | carrar | hahah jcaz |
| 00:50.27 | *** join/#asterisk leejohn (n=leejohn@210.213.240.109) |
| 01:01.46 | *** join/#asterisk trivex (n=trivex@CPE00112f8785ab-CM000f9f50281e.cpe.net.cable.rogers.com) |
| 01:05.03 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:05.17 | Zodiacal | anyone know of a way to get the parkandanounce cmd to speak the parked number on the same call and not call back again? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce |
| 01:05.38 | Zodiacal | parkandanounce = ParkAndAnnounce |
| 01:06.23 | Zodiacal | exten => _2XX,3,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/${EXTEN:1}|default,${EXTEN:1},1) |
| 01:06.42 | Zodiacal | is there a better way to dial than SIP/${EXTEN:1} |
| 01:06.50 | Zodiacal | or somthin |
| 01:07.26 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:09.19 | *** join/#asterisk Un1x (i=Sean@72.61.82.242) |
| 01:09.27 | Un1x | how much upload speed is needed for Ulaw? |
| 01:09.35 | Un1x | ~ulaw |
| 01:09.40 | jbot | extra, extra, read all about it, ulaw is pronounce "mu"-law and consumes 64 Kb/s in each direction. It is considered a loss-less CODEC with a sampling rate of 8,000 hz and is 8 bit. It delivers quality equivalent to that of a POTS line. |
| 01:10.06 | Un1x | not bad |
| 01:10.19 | Un1x | ~g.z29 |
| 01:10.23 | Un1x | ~g.729 |
| 01:10.24 | jbot | i guess g.729 is It was in November 1995 that the G.729 standard, also referred to as CS-ACELP was adopted by the ITU, a United Nations organization. Similar, quality-wise, to 32 kbps ADPCM, G.729 offers toll quality speech. Furthermore, being only an 8 kbps codec, G.729 offers opportunities for significant increases in bandwidth utilization to existing telephony ... |
| 01:10.49 | Un1x | ~g.723 |
| 01:10.51 | Un1x | ~g.711 |
| 01:11.38 | *** join/#asterisk hypnox (n=dan@62.49.107.66) |
| 01:12.10 | hypnox | anyone know if you can run a dialplan function from an agi? (That's function not application) |
| 01:13.20 | *** join/#asterisk lilalinux (i=e-trolle@deepthroat.deswahnsinns.de) |
| 01:13.30 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 01:13.30 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:13.48 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 01:17.01 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 01:23.55 | Qwell | hypnox: sure, just use Set() |
| 01:25.26 | JunK-Y | hypnox: yes u can. |
| 01:26.42 | mivck | hypnox: For example, I use: $AGI->stream_file('beep'); |
| 01:27.03 | Innatech | OK, so I've made some progress with the RTP-300. If I disable the second line, then the * extension assigned to the first line rings. If I enable the second line, then both extensions show as unavailable, and * sends the call to congestion limbo. |
| 01:27.25 | Innatech | Any suggestions/ideas would be welcomed. |
| 01:32.09 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 01:32.09 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 01:35.03 | leejohn | hi guys, does BLA/SLA stuff can be consider on 1.6 by developers? |
| 01:35.46 | russellb | there is some SLA stuff in the upcoming 1.4 release |
| 01:35.47 | *** join/#asterisk Hunter_SC (i=Junior@201.41.232.224) |
| 01:38.14 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-50-104.cybersurf.com) |
| 01:39.04 | Hunter_SC | Who has a SPA-3102? It could help in a dial plan me? |
| 01:41.12 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:42.48 | JunK-Y | is a demo for sla.conf is comin' too? |
| 01:55.13 | awk | what could cause mwi to stop working all the sudden on a polycom? |
| 01:55.16 | awk | the configs look fine |
| 01:55.18 | awk | there is just no mwi |
| 01:56.22 | awk | all polycoms are broken this way |
| 01:56.26 | awk | after a visit from a 'tech' |
| 01:57.15 | JunK-Y | and u still have mailbox=foo in ur sip.conf? |
| 01:57.20 | awk | yes |
| 01:57.43 | JunK-Y | does he changed something on the polycom config? |
| 01:57.48 | awk | egrep ^\\[\|mailbox /etc/asterisk/sip.conf|sed 's/\[//g;s/\]//g;s/mailbox\=//g'|awk '{print $1}'|uniq -c |
| 01:57.50 | awk | no |
| 01:57.53 | awk | this shows perfect results |
| 01:57.55 | awk | 2 of each |
| 01:58.20 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 02:01.51 | Innatech | Anyone know why Line 1 on a Linksys RTP-300 would register with the second line disabled, but fail to register both lines with both enabled? |
| 02:02.13 | Hunter_SC | Who has a SPA-3102? It could help in a dial plan me? |
| 02:09.48 | awk | grep msg *.cfg | grep mwi | grep -v phone[0-9]|sed 's/^.*.:/&\ /g;s/msg.mwi.[1-9].subscribe=//g;s/msg.*.$//g;s/phone.//g;s/.cfg//g;s/://g;s/\"//g'|uniq -c |
| 02:09.54 | awk | this has no problems either |
| 02:09.59 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 02:10.12 | awk | there is nothing in sip.cfg that resembles MWI functionality |
| 02:11.57 | *** part/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 02:14.01 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:14.02 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 02:14.52 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:14.56 | shmaltz | hi everyone |
| 02:16.16 | [TK]D-Fender | awk : Pastebin your configs (sip.conf sip.cfg) |
| 02:16.46 | [TK]D-Fender | awk : and the phones config from 2 phones that don't work. |
| 02:18.17 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 02:23.07 | *** join/#asterisk [koss] (i=koss@ppp-68-250-134-216.dsl.bcvloh.ameritech.net) |
| 02:23.39 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 02:23.39 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 02:26.19 | *** join/#asterisk nailbags|work (n=neilbags@149.171.94.134) |
| 02:28.11 | *** join/#asterisk shad0w1e (n=stjrox@ool-18b9247d.dyn.optonline.net) |
| 02:28.27 | shad0w1e | anyone familiar with this error?: NOTICE[5411]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'stl' |
| 02:29.16 | shmaltz | shad0w1e, why? you don't understand english? |
| 02:29.44 | shad0w1e | why didnt it come with my install |
| 02:29.48 | shad0w1e | and where do I Get it? |
| 02:29.57 | wunderkin | 99.95 |
| 02:30.00 | [TK]D-Fender | shad0w1e : thats a dialplan error |
| 02:30.27 | shad0w1e | ohhh |
| 02:30.31 | shad0w1e | it means im not registered/? |
| 02:30.57 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
| 02:30.59 | [TK]D-Fender | shad0w1e : When do you get this error? |
| 02:31.08 | tengulre11 | hi,all |
| 02:31.09 | shad0w1e | right after my client connects |
| 02:31.13 | shad0w1e | and it keeps spamming the log |
| 02:31.19 | shmaltz | tengulre, hie |
| 02:31.29 | [TK]D-Fender | shad0w1e : check your client config file |
| 02:31.37 | shad0w1e | oh the one i didnt configure |
| 02:31.38 | shad0w1e | heh |
| 02:31.39 | tengulre11 | can the asterisk as a gateway? |
| 02:31.45 | shad0w1e | i thought it was a missing file issue. thanx |
| 02:32.12 | shad0w1e | once I'm here... anyone know how to get a Linksys RTP300 to connect to this thing? I keep getting authentication errors |
| 02:32.51 | tengulre11 | anybody can help me? |
| 02:32.54 | shad0w1e | I read an article how to connect the PAP2 and this one doesnt seem to have that "use authentication" option |
| 02:32.59 | shad0w1e | as that does |
| 02:33.25 | tengulre11 | can the asterisk as a gateway? |
| 02:34.27 | tengulre11 | anybody here? |
| 02:34.37 | wunderkin | /ignore tengulre11 |
| 02:34.45 | tengulre11 | ? |
| 02:34.46 | shad0w1e | [TK]D-Fender thanks |
| 02:34.51 | tengulre11 | why? |
| 02:37.53 | awk | what change could one make to an asterisk polycom system that would hose mwi |
| 02:38.09 | awk | that would make mwi not appear to be operational |
| 02:38.19 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 02:38.44 | awk | gawk: dont just stand there staring, help |
| 02:39.01 | gawk | haha. i cant do anything! |
| 02:39.07 | gawk | thast why im here! |
| 02:39.50 | *** join/#asterisk Synyn_ (n=Synyn___@cpe-72-181-72-81.houston.res.rr.com) |
| 02:41.33 | awk | ~seen batphone |
| 02:41.48 | jbot | batphone <n=bugz@cpe-70-123-122-41.houston.res.rr.com> was last seen on IRC in channel #asterisk, 23h 6m 42s ago, saying: ' '. |
| 02:48.28 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 02:48.48 | [TK]D-Fender | awk : You might want to consider PB-ing your configs like I asked if you want our help... |
| 02:50.00 | *** join/#asterisk tempest1 (n=asf@adsl-144-60-181.chs.bellsouth.net) |
| 02:53.39 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 02:56.55 | *** join/#asterisk mrdigital (n=wild@pool-72-81-77-174.phlapa.east.verizon.net) |
| 02:58.33 | awk | [TK]D-Fender: didnt seee you ask |
| 02:58.35 | awk | what files? |
| 02:58.42 | awk | sip.conf i assume |
| 02:58.49 | awk | phone.1.cfg as well |
| 02:58.53 | [TK]D-Fender | [22:16] <[TK]D-Fender> awk : Pastebin your configs (sip.conf sip.cfg) |
| 02:59.02 | [TK]D-Fender | [22:16] <[TK]D-Fender> awk : and the phones config from 2 phones that don't work. |
| 02:59.04 | awk | ok |
| 02:59.19 | awk | would you like root access as well |
| 02:59.28 | awk | /etc/shadow perhaps |
| 02:59.32 | awk | ;) |
| 02:59.45 | mrdigital | file: would you mind helping me for a second? i need to configure the fxo port on my ata to talk to asterisk |
| 03:00.03 | [TK]D-Fender | awk : No, but I *would* like fries with that |
| 03:00.36 | file | I'm not a walking manual for ATAs, sadly enough |
| 03:00.47 | mrdigital | no idea huh? |
| 03:00.49 | file | but if you ask questions then others may join in |
| 03:00.53 | mrdigital | what about you [TK]D-Fender? |
| 03:00.57 | mrdigital | i got the fxs port working |
| 03:01.05 | mrdigital | i can make /recive calls to asterisk extenstions |
| 03:01.07 | [TK]D-Fender | mrdigital : What kind? |
| 03:01.12 | mrdigital | but now i need incoming pstn calls to goto asterisk |
| 03:01.15 | mrdigital | Zoom 5801 |
| 03:01.31 | [TK]D-Fender | mrdigital : Oh yeah, we talked about this one... never set one up. |
| 03:01.46 | mrdigital | tk: would you like vnc access to maybe look around? |
| 03:01.52 | [TK]D-Fender | mrdigital : Gotten ANYWHERE with the FXO side? |
| 03:02.14 | file | 867-5309! |
| 03:02.25 | [TK]D-Fender | For a good time call! |
| 03:05.46 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 03:08.01 | file | let the fun begin, hey! |
| 03:09.38 | *** join/#asterisk Clausian (i=reginald@203-206-65-20.dyn.iinet.net.au) |
| 03:10.28 | awk | this is way too much crap to PB |
| 03:10.48 | awk | as far as i know there are only 2 places to look for problems. the phone.x.cfg and sip.conf |
| 03:11.22 | awk | i could PB the output of `cat /dev/sda` |
| 03:11.44 | [TK]D-Fender | awk : give me sip.conf and the phone conf's to start |
| 03:12.27 | file | [TK]D-Fender: so how about that weather eh? |
| 03:12.45 | Qwell | eh? |
| 03:12.47 | Qwell | pfft |
| 03:13.32 | [TK]D-Fender | file : Craptastic! |
| 03:15.14 | Clausian | can i have more than one register => statment? |
| 03:15.26 | Qwell | Clausian: sure |
| 03:15.27 | file | sure |
| 03:16.05 | *** join/#asterisk fritz5150 (n=erik@72.174.226.238) |
| 03:16.26 | Clausian | thanks |
| 03:16.40 | fritz5150 | does anyone have any experience in setting up caller id spoofing on asterisk? |
| 03:17.20 | awk | sip add header |
| 03:17.36 | awk | works with some of my providers |
| 03:17.55 | file | fritz5150: Asterisk just allows you to set the callerid to what you want, it's up to where you send the call to allow it... ie: on your PRI your telco has to allow, on VoIP your carrier has to allow it |
| 03:18.05 | fritz5150 | i have followed the examples set out by the document on www.rootsecure.net |
| 03:18.39 | fritz5150 | What I need to do is set up a spoofing service. It looked easy at the start, but i'm in a little deep. |
| 03:19.17 | fritz5150 | basically i want to be able to call the did number and then enter the spoofing number , and then the number to call. |
| 03:19.29 | file | everything you need to do that is available in Asterisk |
| 03:19.59 | *** join/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 03:20.09 | fritz5150 | this is my first attempt at an asterisk configuration. quite familiar with linux, unix, and such, but asterisk is all new to me. |
| 03:20.16 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:20.35 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 03:21.02 | file | just off the top of my head you might want to make a flow chart of how it's going to work and break it down, and plan out each part... might help you learn better instead of looking at it overall all the time... |
| 03:21.29 | fritz5150 | I would be willing to trade product and or services for a little remote help. |
| 03:22.00 | awk | fritz5150: got any smack |
| 03:22.10 | file | awk: don't make me smack you :P |
| 03:22.24 | *** join/#asterisk Qwell[] (n=north@unaffiliated/qwell) |
| 03:22.24 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 03:22.24 | awk | but i need a fix. some zsh will do though. |
| 03:22.30 | Clausian | does anyone know why if i do 'exten => _X.,1,Answer' it works, but if i do 'exten => s,1,Answer' asterisk directs people straight to my isp's voicemail service? |
| 03:22.59 | file | Clausian: the s extension is for when the extension dialed is not known (like on analog lines) |
| 03:23.10 | fritz5150 | I want to be able to call the number assigned to me by voicepulse, have asterisk answer, then ask for the fake number, then the number to call. |
| 03:23.16 | file | Clausian: in this case your ISP is sending you the number the person actually dialed, so the _X. pattern match matches it |
| 03:23.38 | Clausian | ah |
| 03:23.57 | [TK]D-Fender | fritz5150 : Quite doable. |
| 03:24.10 | file | [TK]D-Fender is a respectable person who could probably set you up |
| 03:25.29 | fritz5150 | TK: I am willing to trade a website design, network configuration help, just about anything else you can think of :) |
| 03:26.47 | *** join/#asterisk RageMax (n=max@c-24-3-181-140.hsd1.pa.comcast.net) |
| 03:27.03 | fritz5150 | [TK]D-Fender: I am running the latest version of Asterisk, FreePBX, CentOS4.3. |
| 03:27.46 | RageMax | can someone exactly tell me what "Facility rejected" means (cause code 29) |
| 03:28.20 | *** part/#asterisk redondos (n=redondos@190.48.37.12) |
| 03:28.22 | awk | http://pastebin.ca/103112 |
| 03:29.17 | fritz5150 | [TK]D-Fender: do you have a skype id? |
| 03:29.30 | file | I'll pretend I didn't read that |
| 03:29.35 | awk | fritz5150: i dont think many people in here pay for their phone bro |
| 03:30.28 | fritz5150 | <awk>: sorry man. I just use skype to skype calls only. Kind of a newbie to VoIP |
| 03:30.47 | RageMax | file: any knowledge you can provide? ;) |
| 03:31.07 | file | RageMax: my knowledge of PRIs is ... not so good |
| 03:31.28 | RageMax | this isn't a PRI, it's a IAX registration |
| 03:31.42 | file | oh is it? |
| 03:31.42 | file | then I can probably help you! |
| 03:31.46 | file | do an iax2 debug and pastebin it |
| 03:32.05 | awk | haha. i set up some sip from teliax today. those guys are good (when they arent getting ddos'ed) |
| 03:32.40 | [TK]D-Fender | awk : Funny... I don't see any mailbox parms for those phones... |
| 03:32.48 | RageMax | file: http://pastebin.ca/103117 |
| 03:32.50 | [TK]D-Fender | fritz5150 : Nope..... I only do SIP/IAX |
| 03:32.56 | awk | oops. i deleted them. |
| 03:33.01 | awk | they are in the config though |
| 03:33.09 | awk | they all say mailbox=2299 |
| 03:33.12 | file | RageMax: oh that's easy, registration attempt was rejected |
| 03:33.24 | awk | egrep ^\\[\|mailbox /etc/asterisk/sip.conf|sed 's/\[//g;s/\]//g;s/mailbox\=//g'|awk '{print $1}'|uniq -c |
| 03:33.24 | RageMax | yes, but why is my question |
| 03:33.27 | fritz5150 | [TK]D-Fender: this is an IAX setup |
| 03:33.38 | file | RageMax: it won't say there |
| 03:33.45 | [TK]D-Fender | awk: Hard for me to trust them... pastebin "sip show peer [peer]" for your phones. |
| 03:33.46 | awk | [TK]D-Fender: that command shows 2 instances of each |
| 03:33.50 | file | RageMax: that would open it up for like brute force attacks and stuff... look on when you're registering if you can |
| 03:34.12 | RageMax | packet capture? |
| 03:34.24 | file | when? where... |
| 03:34.34 | Clausian | when it says 'ResponseTimeout is deprecated, please use Set(TIMEOUT(response)=timeout) instead.' where do i put Set(TIMEOUT(response)=timeout)? in the same place i had ResponseTimeout? |
| 03:34.55 | file | RageMax: just an iax2 debug server side, plus any applicable messages that show up on the CLI regularly about it |
| 03:35.14 | RageMax | can't, I'm registering with ipkall |
| 03:35.49 | file | well, there's not really much you can do then... if you're using the right information on your side and it's not working, it has to be something on their side |
| 03:36.06 | file | IAX2 registrations are pretty simple |
| 03:36.13 | RageMax | ok, let me ask you this, if this registration process doesn't work, then I won't be getting incoming calls right? |
| 03:36.17 | RageMax | unlike sip |
| 03:36.30 | [TK]D-Fender | awk : remove the subscribes for MWI in your phone.cfg file, and your "address" field should only be the username not including the IP or anything else. may be pollution things.... |
| 03:36.32 | file | why would SIP be any different? |
| 03:36.41 | RageMax | it's different in the way they have it setup |
| 03:36.43 | [TK]D-Fender | awk : and your config look really far off the beaten path.... |
| 03:36.45 | RageMax | they don't make you register |
| 03:37.00 | file | that's up to the way they have it configured, but one would assume |
| 03:37.14 | file | if they have no IP address in your IAX2 peer, then they can't send you a call |
| 03:37.22 | fritz5150 | Just so you all know, I'm not much for begging..... except this time. |
| 03:37.23 | awk | http://pastebin.ca/103121 |
| 03:37.43 | RageMax | well they have my dyndns address, at least they should |
| 03:38.16 | RageMax | the reason I got an IAX setup though is because I couldn't get SIP to work with my router |
| 03:38.37 | RageMax | it would drop the packet even when I had all ports forwarded, it was driving me mad |
| 03:38.37 | file | RageMax: well if they have a host specified for your peer, you can't register then since it's not dynamic |
| 03:39.12 | fritz5150 | RageMax: what router are you using? |
| 03:39.25 | RageMax | fritz5150: linksys wrt54GL with dd-wrt |
| 03:39.53 | RageMax | it didn't work with either the default firmware or dd-wrt, hence my confusion |
| 03:40.01 | RageMax | file: they just use a static IP?? |
| 03:40.01 | awk | RageMax: you might have to add some forwarding rules to get SIP to work on the linksys |
| 03:40.16 | fritz5150 | RageMax: if it was a Cisco, Nortel, or Enterasys, I could fix you up no problem. |
| 03:40.22 | file | RageMax: I'm not then, I don't know how they have it all setup - I'm just telling you how chan_iax2 works |
| 03:40.26 | RageMax | awk: I did, 5060, and 10000-20000 |
| 03:40.28 | file | eep, s/then/them/ |
| 03:40.57 | RageMax | awk: and again, I even forwarded all ports via the DMZ and it still wouldn't work |
| 03:41.09 | RageMax | the only time I actually got the packet was when I plugged straight into the modem |
| 03:41.09 | file | RageMax: no packets came into Asterisk? |
| 03:41.12 | awk | RageMax: i mean like generic forwarding rules |
| 03:41.18 | file | silly router |
| 03:41.36 | RageMax | yeah, who knows what it was doing |
| 03:42.11 | fritz5150 | Ragemax: instead of forwarding rules, can you do a static NAT Translation on the Linksys? |
| 03:42.13 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 03:42.14 | RageMax | then the clincher when I finally decided to throw SIP out the window, someone actually tried to call my asterisk box anonymously and *that* got through |
| 03:42.22 | RageMax | don't ask me how |
| 03:42.54 | eBody | hey guys. i keep getting crackle using a tdm2400 over POTS lines. which would be the best codec for this situation? |
| 03:43.01 | RageMax | fritz5150: not with the firmware I have on there now, you probably can with openwrt, which I didn't try yet |
| 03:43.16 | awk | RageMax: http://pastebin.ca/index.php |
| 03:43.28 | awk | this is a setup i built running a dozen or so linksys routers with openvpn on them |
| 03:43.40 | awk | you can ping any 192.168.0.0 address from any network |
| 03:43.44 | file | eBody: what's the call flow like? |
| 03:43.45 | fritz5150 | RageMax: static NAT should resolve your situation about packets being dropped. |
| 03:44.07 | RageMax | awk: wrong url ;) |
| 03:44.11 | fritz5150 | file: can you recommend anyone else who would be able to help me set up the spoofing? |
| 03:44.15 | awk | http://pastebin.ca/103132 |
| 03:44.17 | awk | heh |
| 03:44.23 | eBody | file, moderate |
| 03:44.25 | file | fritz5150: there are tons of consultants out there |
| 03:44.35 | fritz5150 | file: I have to get this set up tonight. |
| 03:44.51 | file | eBody: that's not quite the answer I was expecting, I meant are you calling from a SIP phone out it... or what |
| 03:44.58 | awk | fritz5150: what do you mean by 'spoof'? |
| 03:45.00 | file | fritz5150: well, it's tomorrow where I am so you've already past that deadline :D |
| 03:45.02 | fritz5150 | it has been kicking my rear for 4 days now. |
| 03:45.10 | file | passed |
| 03:45.11 | awk | if you want to send out a number in a trunk group you should be able to do so |
| 03:45.19 | eBody | file, yes. from a SIP phone through the asterisk box and tdm2400 through POTS lines |
| 03:45.37 | file | eBody: and what codec is the SIP phone using, and do you have any gains on the TDM2400? |
| 03:45.41 | RageMax | anyway, I'm going to get some rest, thanks for your help guys |
| 03:46.02 | fritz5150 | I want to call the DID assigned by my IAX Provider, have asterisk answer, then ask for the spoof number, then ask for the number to call, and place the call. |
| 03:46.05 | awk | fritz5150: if you need to use different routes for the calls you can set up contexts for each extension, per provider/account |
| 03:46.07 | file | awk: he just needs some dialplan stuff to run a service where someone calls in, types in the number to spoof as, number to dial, and it sends the call out |
| 03:46.12 | eBody | i was defaulted with ulaw, but switched to GSM and added a little rx and tx gain |
| 03:46.17 | eBody | seemed to fix it a whole lot. |
| 03:46.20 | eBody | but still some crackle |
| 03:46.34 | awk | ah, ok. |
| 03:47.01 | file | eBody: take the SIP phone out of the equation, perhaps set it up so you can call into the TDM2400 and use Echo... see if there is still crackle... or even Record it... anything to eliminate some variables |
| 03:47.18 | fritz5150 | awk: the only problem is that this problem has me running in circles. |
| 03:47.19 | awk | RageMax: my point was simply to show that i had to build some for loops to allow the routing to take place otherwise those packets get sent to /dev/null |
| 03:47.22 | eBody | ok i'm gonna actually check the lines right now. |
| 03:47.25 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 03:47.35 | eBody | file, u don't think it's a config issue? |
| 03:47.58 | file | eBody: crackle? meh, not really |
| 03:48.34 | file | if you can trace it down to the TDM2400 you should be able to give support a call, and they can try debugging the issue... might just mean tweaking your gains more, I dunno... I don't do hardware a lot |
| 03:48.39 | awk | fritz5150: to do that your carrier/voip provider needs to allow you to set the outbound caller id |
| 03:48.48 | file | if they let me near the hardware department stuff would explode |
| 03:49.04 | awk | fritz5150: if they are using a good implementation of SIP you can ask them about using extra sip headers to do so |
| 03:49.05 | fritz5150 | awk: they let me set my own caller id. |
| 03:49.19 | file | awk: I find it curious you're using extra SIP headers... |
| 03:49.20 | eBody | file, TDM support? |
| 03:49.25 | awk | fritz5150: but it sounds like it has to be in your trunk group to work |
| 03:49.25 | eBody | who would that be? digium? |
| 03:49.33 | file | eBody: Digium technical support |
| 03:49.48 | awk | fritz5150: adding a sip header allows you to send extra data for their end to parse while still not bypassing their own security measures |
| 03:49.58 | file | fritz5150: who is your outbound carrier? |
| 03:50.01 | fritz5150 | voicepulse |
| 03:50.06 | file | yeah they allow it |
| 03:50.14 | file | by regular methods, so you're fine |
| 03:50.15 | fritz5150 | that's why I chose them. |
| 03:50.30 | file | how do you want to authenticate callers? |
| 03:50.37 | awk | setcallerid(313337) |
| 03:50.38 | *** join/#asterisk bmg505 (n=leon@dsl-146-30-127.telkomadsl.co.za) |
| 03:50.41 | fritz5150 | 4 digit password. |
| 03:50.54 | file | fritz5150: let me type up something quickly |
| 03:50.58 | fritz5150 | I have a script, but I can't figure out why it's not working. |
| 03:51.03 | awk | aha... im doing something similar with a .call file atm |
| 03:51.04 | fritz5150 | I'll paste it here. |
| 03:51.08 | file | NO |
| 03:51.15 | fritz5150 | Oh, Ok. |
| 03:51.16 | file | that'll be fine... |
| 03:51.19 | file | hold! |
| 03:51.42 | awk | `fpm-calm-river.mp3` |
| 03:52.07 | file | argh |
| 03:52.09 | file | I just lost part of my net |
| 03:52.39 | fritz5150 | What part? Hopefully not the packets! ;) |
| 03:52.54 | file | my secured network gateway |
| 03:53.00 | file | and my development machine |
| 03:53.14 | awk | yowch |
| 03:53.26 | fritz5150 | yowch is right! |
| 03:53.53 | awk | file: do you have immediate physical access? |
| 03:53.58 | file | yes |
| 03:54.19 | awk | lucky you. if that happened to me right now id be s.o.l. |
| 03:54.30 | awk | and on the road |
| 03:54.35 | fritz5150 | file: are the packets being dropped? or some other type of outage? |
| 03:54.43 | file | dunno |
| 03:54.45 | file | restarting the box |
| 03:54.49 | file | it's back up :D |
| 03:54.58 | awk | i set up a 1u prototype today |
| 03:55.35 | fritz5150 | file: Lucky you! |
| 03:56.29 | file | fritz5150: http://pastebin.ca/103146 |
| 03:56.39 | file | needs error checking, but meh |
| 03:57.30 | awk | http://pastebin.ca/103148 |
| 03:57.54 | awk | it runs off of 2 CF cards, no moving parts whatsoever |
| 03:58.03 | awk | handles between 10 and 20 users |
| 03:58.12 | file | my devel box is a dual core Athlon64 4200+ with 1GB of RAM... does basic routing too |
| 03:58.19 | file | bridges two other networks to mine here |
| 03:58.42 | awk | file: i have a similar setup. my devel box, aptly named 'devel' does imaging as well |
| 03:59.07 | awk | its a dual xeon |
| 03:59.18 | awk | 80gb sata mirror |
| 03:59.46 | file | fritz5150: I even commented that for you so you could learn what it does ;) |
| 03:59.53 | file | and expand it... |
| 04:00.34 | fritz5150 | Does this go in the extensions.conf or extensions_custom.conf? |
| 04:00.51 | awk | haha thats pretty sweet |
| 04:01.40 | file | fritz5150: have you read the dialplan part of The Book? |
| 04:02.05 | fritz5150 | file: Yeah, but that's what's the most confusing to me. |
| 04:02.37 | file | ungood |
| 04:02.50 | fritz5150 | believe me, I know. |
| 04:03.02 | fritz5150 | I just want to get this thing done, and out of my hair. |
| 04:03.25 | fritz5150 | I know you will all laugh at me, but I even used FreePBX to set it all up so far. |
| 04:03.37 | file | yeah that's what's further ungood |
| 04:04.39 | fritz5150 | Give me a routing project, DNS, Web, Wireless, PKI, I'm your man. Just not with asterisk:( It'll take me a while to learn. |
| 04:05.24 | file | to quote the email I just read on asterisk-dev: |
| 04:05.26 | file | "Asterisk is a lot like sex, you fear it first, then once you get into it and its ALL you can think about!!!" |
| 04:06.10 | fritz5150 | It has me intrigued... That's probably not good though cause I'll be obsessed with it until I fully understand it. |
| 04:06.28 | file | that's how it starts... |
| 04:07.09 | fritz5150 | To be good... You must possess one important tool... an obsessive-compulsive personality! |
| 04:07.32 | fritz5150 | file: would you be up for a remote session? |
| 04:07.50 | *** join/#asterisk niteowldave (i=niteowlO@203.82.162.38) |
| 04:08.20 | file | no, I'm afraid not |
| 04:08.29 | file | if you wait till morning you might be able to snag someone to help you |
| 04:08.42 | file | niteowldave: it's Mr. T38 guy ;) |
| 04:08.48 | fritz5150 | I have 5 hours to get this completed. |
| 04:09.03 | niteowldave | file: Hi there, how r u going |
| 04:09.17 | file | sleeeeeeeepy |
| 04:09.25 | file | you? |
| 04:10.07 | niteowldave | I undersdtand, still no luck getting passthrough to work, I don't quite underdstand why the re-invite does not work. |
| 04:11.01 | niteowldave | file: Connected the ATAs back up to ser and all works OK, but I would rather use asterisk |
| 04:11.29 | niteowldave | file: must be about 1 in the morning there! |
| 04:11.39 | file | niteowldave: naturally it'll work with SER... it's a proxy, while Asterisk is a B2BUA... each side gets negotiated independently, and we want at the minimum audio |
| 04:11.43 | file | it is 1:11AM |
| 04:13.04 | niteowldave | File: I understand that much, I don't understand the whole NAT/re-invite issue though. Do you know where I can find a good explanation of that issue? |
| 04:14.07 | niteowldave | file: If i could get my head around the re-invite problem and how to resolve that I might get somewhere |
| 04:14.10 | file | niteowldave: not really, if I saw sip debugs though I might be able to say more... and it depends on what you mean by reinvite exactly... ie: reinviting the devices to talk directly, or them reinviting to Asterisk |
| 04:15.40 | niteowldave | eg, when the two devices are on the same lan the re-invite for the switchover to t.38 works ok, when I use the same ATAs behind a nat at different locations the re-invite for t38 fails |
| 04:16.29 | file | okay so you're talking about Asterisk sending a reinvite to each side so they talk directly? |
| 04:17.23 | niteowldave | yes, there is a sip trace at http://pastebin.ca/99174 that I did the other day |
| 04:19.34 | file | is that a full sip debug of the entire dialog from start to finish? |
| 04:19.48 | niteowldave | file: sorry that one is for a straight t38 only call |
| 04:19.55 | file | yeah |
| 04:20.12 | file | that's why I was confused |
| 04:21.21 | russellb | file: !!!!!!!!!!!!! |
| 04:21.40 | russellb | yay file, you so r0x0r |
| 04:21.47 | file | whyfor?!? |
| 04:21.50 | russellb | i don't know. |
| 04:21.55 | russellb | because it's late and i'm going insane |
| 04:22.00 | file | excellent |
| 04:22.38 | file | scary |
| 04:23.00 | *** join/#asterisk roving_prole (n=Harper@c-71-199-16-110.hsd1.co.comcast.net) |
| 04:24.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:24.34 | file | russellb: now if you change your mind again it'll be silly |
| 04:24.55 | russellb | :D |
| 04:24.58 | russellb | maybe I will! |
| 04:25.03 | russellb | probably not on that part, though |
| 04:29.23 | *** join/#asterisk docelmo (n=Snake@55-65.126-70.tampabay.res.rr.com) |
| 04:30.48 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:34.46 | Netgeeks | Heya Qwell, Russell! |
| 04:34.53 | JunK-Y | anyone knows how to sip notify a gxp-2000 to reboot it? since Event=>sys-control seems to go nowhere. |
| 04:35.46 | niteowldave | file: sorry, just running a new trace |
| 04:35.51 | Un1x | eh some help quick question wich is better to install asterisk on, Gentoo or Slackware? |
| 04:35.59 | russellb | Un1x: whatever you are most familiar with |
| 04:36.08 | russellb | doesn't matter in regards to asterisk ... |
| 04:36.28 | Un1x | what about zaptel drivers? |
| 04:37.07 | russellb | still doesn't matter |
| 04:37.58 | russellb | as long as you don't run SuSE 10 ... since they decided to make it near impossible to build custom kernel modules |
| 04:38.58 | file | Just say no! |
| 04:39.04 | Netgeeks | no |
| 04:39.27 | file | I guess I have powers over people... |
| 04:39.38 | russellb | just say yes |
| 04:39.49 | russellb | Now, everyone must choose between us! |
| 04:39.51 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 04:40.03 | file | russellb: unless they are undecided |
| 04:40.17 | russellb | or intent on lurking |
| 04:40.24 | russellb | or ... not here |
| 04:41.45 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 04:44.04 | Un1x | dylnes |
| 04:45.08 | niteowldave | file: try this trace http://pastebin.ca/103185..thanx |
| 04:48.06 | fritz5150 | file: I added that script to a custom context in extensions_custom.conf. |
| 04:48.12 | fritz5150 | Still no worky |
| 04:48.24 | file | doesn't like the SDP for T.38, eep |
| 04:48.58 | file | fritz5150: that dialplan logic was a base, it won't work outright - it has to be tweaked - for example the right DID needs to be specified, you need an entry in voicemail.conf to authenticate as, plus you need to make sure incoming calls are going to it |
| 04:49.46 | fritz5150 | Incoming calles are going to it, but I left out the authenticate for testing |
| 04:49.59 | file | have you looked at the CLI to see what it's doing? |
| 04:50.34 | fritz5150 | I set it up as exten 555 then I have the inbound route set to go to exten 555 |
| 04:50.52 | file | but what is it really doing? |
| 04:51.58 | niteowldave | file: asterisk or the ata? |
| 04:52.27 | file | niteowldave: Asterisk is sending the INVITE with T.38 SDP, but your ATA does not like the SDP |
| 04:55.00 | fritz5150 | file: the CLI isn't showing me anything |
| 04:55.59 | file | fritz5150: when you dial the number? |
| 05:00.08 | fritz5150 | file: nope it hust sits at the CLI> prompt |
| 05:01.06 | file | then you have other problems, such as incoming calls not getting to your box... and Voicepulse hopefully has configuration instructions to some capacity that you can double check because it's 2AM and my mind is gone |
| 05:01.30 | fritz5150 | I can see on the freepbx interface that the call is answered. |
| 05:01.31 | sharp | fritz5150, set verbose 3 |
| 05:01.39 | fritz5150 | I have the verbose set to 15 |
| 05:01.43 | sharp | oh |
| 05:01.58 | file | and you saw nothing on the Asterisk console? |
| 05:02.11 | fritz5150 | I am ssh'd in and not at the console. But that shouldn't matter, should it? |
| 05:02.52 | file | asterisk -r |
| 05:03.03 | fritz5150 | file: exten 555 answers the call. |
| 05:03.14 | fritz5150 | I did asterisk -rvvvvvvvvvv |
| 05:03.29 | file | oh, and you do realize I just put in imaginary filenames for the sounds? :) |
| 05:03.52 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.124) |
| 05:03.53 | file | and imaginary information for the actual dialing part |
| 05:04.07 | Un1x | file if i use Ulaw, then it wont use too much sys resources right, as in processor and sutff coz my server is, sempron 2200+ with 600megs ram... |
| 05:04.24 | fritz5150 | I know I customized the dialing part. |
| 05:04.57 | file | Un1x: you should be fine |
| 05:05.29 | fritz5150 | file: (IAX2/username:password@connect01.voicepulse.com/${DIAL_NUM}) |
| 05:06.30 | file | the console output would be nice though... so you can see what it is actually doing, not what it should be doing |
| 05:06.50 | fritz5150 | I will call the number again. and watch the cli |
| 05:12.45 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 05:13.25 | fritz5150 | file: exited non-zero on 'IAX2/Voicepulse1-1' |
| 05:13.50 | file | sorry, but I'm going to sleep |
| 05:14.22 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 05:15.01 | file | ugh |
| 05:15.15 | file | fritz5150: pastebin (http://www.pastebin.ca) full output |
| 05:22.01 | websae | anyone know where i can get a good deal on a 1 u server? |
| 05:22.14 | Qwell | websae: telcomjoshvoxmart |
| 05:22.22 | Qwell | (I want my commissions) |
| 05:22.22 | websae | who? |
| 05:22.26 | Qwell | websae: telcomjoshvoxmart |
| 05:22.31 | websae | hrm... |
| 05:22.35 | websae | is there a website |
| 05:22.46 | Qwell | no, just a guy in an alley |
| 05:24.52 | websae | where do i find him |
| 05:25.09 | Qwell | he finds you |
| 05:28.42 | niteowldave | file: Hi there, one of the ATAs had not taken a change to its settings correctly, here is the correct trace http://pastebin.ca/103214 |
| 05:35.56 | Un1x | file you know the caller id spoofing |
| 05:36.02 | Un1x | the set caller id in asterisk... |
| 05:36.12 | Un1x | was wondering is it possible i can like lift of the phone and press something like 9 or somethign |
| 05:36.18 | Un1x | and it ask me for a caller id number and dest number |
| 05:36.19 | Un1x | ? |
| 05:38.36 | fritz5150 | file: http://pastebin.ca/103220 |
| 05:48.19 | fritz5150 | file: can you see anything from the output? |
| 05:56.01 | *** join/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
| 05:57.48 | *** join/#asterisk speekac (n=alwin@60.51.217.58) |
| 06:02.08 | *** join/#asterisk tempest1 (n=asf@adsl-144-60-181.chs.bellsouth.net) |
| 06:12.46 | *** join/#asterisk RaHaiL (n=rahail1@209-19-88-238.detroit.mi.D-Conn.net) |
| 06:12.48 | *** join/#asterisk daysmen3 (n=primus@host86-138-238-236.range86-138.btcentralplus.com) |
| 06:12.59 | RaHaiL | any one alive that can do a small project |
| 06:14.38 | russellb | unless the project involves crawling into bed, not me |
| 06:14.51 | Qwell | russellb: It very well might |
| 06:14.56 | russellb | ha |
| 06:14.58 | Qwell | You'd regret that statement then |
| 06:15.03 | russellb | indeed, i would |
| 06:16.22 | *** join/#asterisk angom_h (n=papa@red-corp-200.38.15.233.telnor.net) |
| 06:16.40 | RaHaiL | noo |
| 06:16.46 | RaHaiL | lol i need some one to make interface for me |
| 06:16.54 | RaHaiL | where i can dump all my cdr |
| 06:17.12 | RaHaiL | and people aka client can login and see there usage and charge kind of like billing |
| 06:20.27 | RaHaiL | any one |
| 06:22.46 | *** join/#asterisk s0lid (n=jlq@210.213.199.246) |
| 06:31.16 | droops | hey RaHaiL thats not hard to do with a little php and a cron job |
| 06:31.31 | RaHaiL | 0 exprince on php |
| 06:31.37 | *** join/#asterisk nailbags|work (n=nailbags@149.171.94.134) |
| 06:31.44 | RaHaiL | just i know how to make website with html |
| 06:32.53 | droops | sounds like you need a book |
| 06:33.11 | RaHaiL | still reading anything i find about php |
| 06:33.15 | Clausian | RaHaiL: http://www.astpp.org/ |
| 06:33.33 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 06:35.12 | docelmo | what the hell does this mean vayansea la verga dejen de enviar spam |
| 06:36.36 | *** join/#asterisk sponix (i=family@host-66-205-123-177.classicnet.net) |
| 06:36.49 | Qwell | docelmo: means "I didn't want to be automatically added to the freeswitch mailing list, don't spam me" |
| 06:36.49 | Qwell | :P |
| 06:38.01 | MikeJ | heh.. funny thing is.. he manually subscribed... |
| 06:38.56 | docelmo | hay qwell coming to cluecon? |
| 06:39.03 | docelmo | anyone actually from this cannel going? |
| 06:39.57 | *** join/#asterisk TeePOG (n=TeePOG@dsl-145-154-108.telkomadsl.co.za) |
| 06:40.44 | *** join/#asterisk daysmen3 (n=primus@host86-138-238-236.range86-138.btcentralplus.com) |
| 06:41.06 | Qwell | MikeJ: I didn't :P |
| 06:41.09 | Qwell | docelmo: nope.. |
| 06:43.29 | MikeJ | to dev? |
| 06:47.34 | *** join/#asterisk tempest1 (n=asf@adsl-144-60-181.chs.bellsouth.net) |
| 06:48.44 | *** join/#asterisk daysmen3 (n=primus@host86-138-238-236.range86-138.btcentralplus.com) |
| 06:51.09 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 06:54.32 | *** join/#asterisk Un1x (i=Sean@72.61.82.242) |
| 06:55.02 | *** join/#asterisk mitcheloc (n=mitchelo@c-24-23-37-212.hsd1.ca.comcast.net) |
| 06:55.15 | Un1x | hey |
| 06:55.25 | Un1x | is there a problem if i use a pci nic in my asterisk box? |
| 06:55.48 | Un1x | it wouldn't cause any problems would it, because i have a PCI, linksys nic and dont want to use, the onboard nic wich is onthe mobo... |
| 06:57.52 | rob0 | The onboard NIC is surely PCI as well, FWIW. |
| 06:58.45 | rob0 | Yes, some zaptel hardware has trouble with shared interrupts. There's no way to say for sure that you will or won't have that problem. |
| 06:59.44 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 07:01.04 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 07:04.33 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 07:04.34 | Stephnie | hello |
| 07:05.34 | Stephnie | I want to connect TWO INBOUND calls to each other.....(just like SIP Peer to SIP Peer) but these inbound calls are not peers....any idea how to let them talk to each other? .... |
| 07:08.44 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 07:08.46 | RaHaiL | any one know where can i go change my defult codec |
| 07:09.00 | RaHaiL | i have my defult codec ulaw i want change to gsm |
| 07:09.08 | Assid | RaHaiL: in your context |
| 07:09.08 | Clausian | anyone in here running astpp? |
| 07:09.08 | *** join/#asterisk ma_dzen (n=ma_dzen@217.66.17.141) |
| 07:09.12 | RaHaiL | for the h323 route |
| 07:09.46 | RaHaiL | i mean what directory i can go and do that |
| 07:10.13 | Assid | i dunno about h323 |
| 07:10.54 | RaHaiL | thank you |
| 07:11.10 | Stephnie | Assid : what about my questions ? |
| 07:11.14 | Stephnie | question* |
| 07:12.57 | Assid | Stephnie: didnt see them.. i just logged in |
| 07:14.13 | Stephnie | I want to connect TWO INBOUND calls to each other.....(just like SIP Peer to SIP Peer) but these inbound calls are not peers....any idea how to let them talk to each other? .... |
| 07:14.20 | *** join/#asterisk vlrk (n=vlrk@202.65.134.119) |
| 07:14.29 | carrar | meetme |
| 07:14.34 | Assid | meetme.. |
| 07:14.41 | Assid | or use conference on your phone |
| 07:15.02 | carrar | call the phone directly |
| 07:15.17 | carrar | 867-5309 |
| 07:15.23 | Stephnie | okey thanks |
| 07:15.46 | benjk | meetme only works if you have zaptel hardware |
| 07:16.05 | benjk | if you don't, there's app_conference |
| 07:16.07 | Stephnie | no I dont have zaptel...both are inbound callers through SIP DID |
| 07:16.17 | benjk | app_conference then |
| 07:17.09 | Assid | meetme also works with ztdummy |
| 07:17.23 | benjk | which is a terrible hack that can bring your system down |
| 07:17.37 | Assid | really? |
| 07:17.41 | Stephnie | really |
| 07:18.38 | benjk | its a driver that runs inside the kernel mimicking hardware that you don't have |
| 07:18.54 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 07:19.09 | Assid | damn.. and i had these guys remove a zaptel wcfxo cause i wanted to use dummy |
| 07:19.24 | vlrk | how does intercom basically works with asterisk |
| 07:19.40 | benjk | use app_conference |
| 07:19.49 | benjk | its completely userland |
| 07:20.03 | benjk | intercom is obsoleted |
| 07:20.04 | Stephnie | benjk : but it was a terrible hack that can bring our system down ;) |
| 07:20.23 | benjk | use app_conference and throw out meetme |
| 07:20.33 | benjk | then you are safe |
| 07:20.44 | benjk | and don't load ztdummy |
| 07:20.58 | benjk | then there are no kernel modules involved |
| 07:21.17 | Stephnie | dont load ztdummy? u mean I shouldnt install zaptel if I dont have it ? |
| 07:21.32 | benjk | yeah precisely that's the point |
| 07:21.44 | benjk | why should you run zaptel drivers if you don't have zaptel hardware |
| 07:21.48 | benjk | doesn't make any sense |
| 07:22.18 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:22.35 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 07:22.45 | Assid | doesnt IVR need zaptel drivers for timng? |
| 07:22.54 | benjk | no |
| 07:23.16 | benjk | IAX trunking and meetme are the two things that use zaptel as a timing source |
| 07:23.34 | benjk | app_conference can be used in place of meetme |
| 07:24.14 | benjk | IAX trunking you won't be able to use, but it's not such a big issue, just set trunk=no in your iax.conf |
| 07:24.27 | Assid | hrmm |
| 07:24.43 | vooduhal | Hey guys. I've got a quick question. I've got a Sangoma A104 with two active PRIs and a Digium TDM card with 1 FXS and 1 FXO and I'm trying to setup my zatel.conf and zapata.conf. I'm having problems figuring out which channels are listed on which interface. Is there a way to tell which channels belong to which interface? |
| 07:25.04 | benjk | its only useful if you have many many concurrent IAX calls between the same two asterisk servers |
| 07:26.25 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:26.29 | *** join/#asterisk vgster (n=vgster@217.78.147.238) |
| 07:27.02 | benjk | then again, OpenPBX.org have a patch for chan_iax to use trunking without zaptel as a timing source, so if absolutely needed, it can be done too |
| 07:27.13 | Assid | i always thought background or something needed that |
| 07:27.31 | benjk | as I said, only IAX trunking and meetme |
| 07:28.28 | Assid | im guessing it can handle much more load as compared to the likes of meetme ? |
| 07:29.09 | benjk | I haven't noticed any difference in performance |
| 07:29.25 | benjk | there are a few options that meetme has which app_conference doesnt |
| 07:29.45 | benjk | on the other hand, app_conference creates conferences on the fly, so its easier on maintenance |
| 07:30.42 | *** join/#asterisk vlrk (n=vlrk@202.65.134.119) |
| 07:33.10 | Assid | looks interesting.. will play with it |
| 07:33.32 | Assid | no timing device and no resampling.. thatmeans everyone gets their native format |
| 07:33.36 | vlrk | iam using two snoms and asterisk to workout the intercom is . Is anybody successull in doing this ? |
| 07:37.35 | carrar | meetme is broken |
| 07:37.40 | carrar | but works mostly |
| 07:37.58 | benjk | use app_conference then ;) |
| 07:43.52 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 07:44.10 | Stephnie | forgot to ask....where do I get this app_conference from ? |
| 07:44.56 | Stephnie | information on wiki is out dated |
| 07:50.50 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 07:53.12 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 07:54.11 | *** join/#asterisk s0lid (n=jlq@210.213.240.222) |
| 07:58.36 | *** join/#asterisk thx2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 07:59.08 | thx2000 | Anyone know why my sipura would be trying to dial an IP address instead of the number im passing it? |
| 08:02.25 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B79E45.pool.t-online.hu) |
| 08:06.32 | Un1x | ~asteriskports |
| 08:06.39 | Un1x | wich ports does asterisk and zaptel use... |
| 08:06.44 | Un1x | so i can open them in, my firewall :) |
| 08:08.04 | rob0 | That depends which protocols you are using. |
| 08:08.09 | Un1x | sip |
| 08:08.24 | Un1x | well isn't there a list already |
| 08:08.31 | Un1x | of, Ports/protocals.. |
| 08:09.20 | _Vile | http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
| 08:10.39 | Un1x | 10x |
| 08:10.40 | Un1x | :p |
| 08:16.07 | *** join/#asterisk Formater (i=Formater@cable-87-116-143-43.dynamic.sbb.co.yu) |
| 08:16.25 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 08:17.16 | Formater | hi, I have problem with ooh323 + asterisk... when I make calls from SIP to H323, connection is made... but only one-way audio... I got audio from h323 endpoint... but nothing goes from SIP :( |
| 08:17.36 | Formater | so when there is audio at h323 endpoint I see: |
| 08:17.37 | Formater | Got RTP packet from 66.135.35.44:5002 (type 3, seq 36267, ts 77120, len 33) |
| 08:17.37 | Formater | Sent RTP packet to 212.183.41.33:45956 (type 18, seq 22305, ts 73600, len 20) |
| 08:18.02 | Formater | but when I talk or send dtmf in x-lite, nothing happens, no rtp packet on CLI |
| 08:18.04 | Formater | any idea? |
| 08:21.10 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 08:28.36 | *** join/#asterisk vlt (n=daniel@dslb-088-073-227-095.pools.arcor-ip.net) |
| 08:35.18 | *** join/#asterisk docelmo (n=Snake@55-65.126-70.tampabay.res.rr.com) |
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| 08:52.04 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:55.12 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
| 08:59.44 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
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| 09:12.02 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 09:14.22 | Stephnie | hi, i have installed app_conference ...now do I need to make conference.conf |
| 09:14.34 | *** join/#asterisk suma (n=kans@61.14.86.23) |
| 09:14.53 | suma | i want to have asterisk sending rtp to a single port |
| 09:15.01 | *** join/#asterisk andew (n=andew@84-45-170-202.no-dns-yet.enta.net) |
| 09:15.18 | suma | i configured rtp.conf , rtpstart=7078 & rtpend=7078 |
| 09:15.19 | folder | 136ms qualify for a locally connected (fast eth) SIP gateway is pretty poor isn't it? |
| 09:15.39 | suma | but when asterisk sends rtp it is sending to a different port in the SDP |
| 09:15.52 | folder | Stephnie: dunno. 1 sec. |
| 09:16.07 | Stephnie | folder: okey... |
| 09:16.32 | *** join/#asterisk parag_ast (n=root@dxb-b1751.alshamil.net.ae) |
| 09:16.34 | Stephnie | folder: in my asterisk ...exten=> 111,1,Conference(111/BLA) doesnt work |
| 09:16.35 | folder | Stephnie "There is no configuration file. Conferences are created on-the-fly. " |
| 09:16.47 | suma | anyone please help me regarding the rtp problem ? |
| 09:16.49 | Stephnie | folder: dont get any error too.. |
| 09:16.54 | folder | Stephnie: that's from http://www.voip-info.org/wiki/view/Asterisk+app_conference |
| 09:17.05 | parag_ast | Can anybody let me know that what are the parameters used in /usr/src/zaptel/zonedata.c... |
| 09:17.07 | parag_ast | ?? |
| 09:17.58 | folder | Stephnie: also have a look here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+conference |
| 09:18.35 | parag_ast | { 1, "au", "Australia", { 400, 200, 400, 2000 }, |
| 09:18.43 | parag_ast | what is this 400, 200, 400, 2000 |
| 09:18.44 | parag_ast | are |
| 09:18.45 | parag_ast | ?? |
| 09:19.26 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.125) |
| 09:21.21 | buzzdeee | hi, I have a problem with DID, I have overlapdial=yes and immediate=no in zapata.conf, I also tried to raise the matchdigittimeout to 10000 in channels/chan_zap.c, as suggested by benjk, but nothing helped, any idea what else I can do? |
| 09:21.40 | buzzdeee | is there a way to tell asterisk to wait for a given number of digits? as all have the same length? |
| 09:21.42 | parag_ast | Hello can anybody look at these frequencies http://www.geocities.com/virtualnetphone/zaptel_info.html and let me know where do i need to change in zonedata.c .... |
| 09:31.20 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 09:38.11 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 09:38.15 | Nobbie | hi, |
| 09:38.28 | parag_ast | Hello can anybody look at these frequencies http://www.geocities.com/virtualnetphone/zaptel_info.html and let me know where do i need to change in zonedata.c .... |
| 09:38.51 | Nobbie | i need some help with Digium PRI TE205 card. where can i get installation docs and help to connect it to telco ? |
| 09:42.15 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:42.26 | parag_ast | Hello can anybody look at these frequencies http://www.geocities.com/virtualnetphone/zaptel_info.html and let me know where do i need to change in zonedata.c .... |
| 09:43.33 | folder | Do we think it's ever likely that there'll be firmware updates for hardphones to support speex? |
| 09:47.10 | *** join/#asterisk tparcina (n=tparcina@lns02-0727.dsl.iskon.hr) |
| 09:47.29 | tparcina | hi channel! |
| 09:47.42 | folder | hi tparcina |
| 09:49.28 | folder | Nobbie: Have you tried http://www.digium.com/en/supportcenter/documentation/viewdocs/TE205P ? |
| 09:49.29 | *** join/#asterisk Zauephuaes (n=guido@196.37.100.13) |
| 09:49.52 | Zauephuaes | I'm having trouble getting Asterisk to play a file via AGI STREAM FILE |
| 09:50.12 | Zauephuaes | it accepts the command, returns 200 (ok) yet only silence ... |
| 09:52.12 | *** part/#asterisk angom_h (n=papa@red-corp-200.38.15.233.telnor.net) |
| 09:57.47 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 09:58.04 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 09:58.08 | tengulre | Hi,all |
| 09:58.11 | tengulre | I m backing! |
| 09:58.13 | folder | hiya |
| 09:59.08 | tengulre | who can tell me what's the 500 in username:sercet@mydomain.com/500 mean? |
| 09:59.31 | UlbabraB | hi, I have two ip addresses on my * box...is it possible to originate SIP calls to two different voip providers with the two different ip adresses as source addresses? |
| 09:59.51 | folder | tengulre: From what I remember I think it means that calls coming from that upstream provider which are to unknown extentions will be sent to extension 500. |
| 09:59.59 | UlbabraB | tengulre: the user is registered at extension 500 |
| 10:12.16 | RoyK | ~ping |
| 10:12.22 | jbot | pong |
| 10:12.22 | gr0mit | pong |
| 10:12.26 | RoyK | ~lart gr0mit |
| 10:13.52 | L|NUX | RoyK : how can we know that how many channels are allowed in DID ? |
| 10:20.30 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:21.34 | *** join/#asterisk folder (n=carl0s@compsup.demon.co.uk) |
| 10:21.46 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 10:35.09 | *** join/#asterisk tempy (n=tempy@c220-237-91-41.rochd1.qld.optusnet.com.au) |
| 10:37.48 | *** join/#asterisk NoNeo (n=ankamins@193.24.24.10) |
| 10:37.58 | tempy | hi all, does anyone know where to get some cheap clone FXO/FXS cards in australia? |
| 10:38.05 | *** part/#asterisk NoNeo (n=ankamins@193.24.24.10) |
| 10:38.38 | *** join/#asterisk MACscr (n=MACScr@adsl-75-23-81-217.dsl.peoril.sbcglobal.net) |
| 10:39.04 | folder | tempy: eBay |
| 10:39.16 | folder | tempy: You mean X100P / X101P of course |
| 10:39.42 | folder | That's not a recommendation to use them BTW. They're not very well supported really. |
| 10:39.56 | folder | (as in.. you don't get much help. Not saying whether or not they work well) |
| 10:41.01 | RoyK | tempy: use isdn instead. pots is evil |
| 10:41.19 | tempy | folder: only 3 on ebay.com.au and they are not cheap, nor clones |
| 10:41.37 | benjk | those cards are only good for one thing: zaptel timing |
| 10:41.45 | tempy | hmm, isdn ... |
| 10:41.52 | benjk | they are no good for sending phone calls through |
| 10:41.59 | *** join/#asterisk NoNeo (n=ankamins@193.24.24.10) |
| 10:42.04 | gr0mit | isdn is cheaper than pots in most civilised places |
| 10:42.13 | tempy | so what is? one of the vendor devices? |
| 10:42.46 | tempy | it isn't too big in australia, i wouldn't use isdn to connect for internet, surely wouldn't use it for VoIP |
| 10:42.50 | benjk | get a passive single port HFC PCI card and use BRIstuff |
| 10:42.52 | gr0mit | i have 3 isdn lines using billion |
| 10:43.02 | gr0mit | 15 euros/card |
| 10:43.09 | tempy | gr0mit: you in .au? |
| 10:43.13 | gr0mit | nah. |
| 10:43.13 | benjk | its not for data, its not for VoIP |
| 10:43.15 | gr0mit | .uk |
| 10:43.30 | folder | ISDN isn't cheaper here in the UK though. Calls cost the same but you have to buy an ISDN2e and then you need to keep a POTS line for your ADSL. So you end up paying three times the cost of a single POTS line. |
| 10:43.34 | benjk | its for connecting to a landline |
| 10:43.47 | gr0mit | i said in most civilised places. |
| 10:43.55 | gr0mit | i do not include .uk in that list. |
| 10:43.56 | tempy | i have broadband via cable tv system |
| 10:43.58 | benjk | sell DIDs and you get your line paid for |
| 10:44.22 | benjk | how much is an additional DID on your ISDN line in the UK? |
| 10:44.37 | folder | benjk: what, and use my ASDL so I act as a SIP provider for clients? |
| 10:44.41 | folder | benjk: £10 I think |
| 10:44.51 | benjk | what area code are you in? |
| 10:44.55 | folder | 0161 |
| 10:45.04 | benjk | Bristol |
| 10:45.11 | gr0mit | manchester! |
| 10:45.12 | folder | manchester |
| 10:45.13 | folder | :D |
| 10:45.15 | benjk | ah |
| 10:45.18 | benjk | silly me |
| 10:45.27 | folder | gr0mit: sure, uk certainly isn't civilised when it comes to this sort of stuff |
| 10:45.51 | gr0mit | actually for a business line ISDN does work out about the same as pots |
| 10:45.52 | benjk | anyway, it is quite possible that you can sell three or so DIDs for more than what you have to pay BT |
| 10:46.04 | benjk | that may cover the cost of your line rental |
| 10:46.15 | folder | benjk: but sipgate.co.uk offer them for nothing. |
| 10:46.18 | RoyK | tempy: anyway, even if isdn is a little more expensive, it works far better |
| 10:46.21 | RoyK | pots is ugly |
| 10:46.22 | RoyK | bad |
| 10:46.24 | RoyK | evil |
| 10:46.26 | RoyK | horrible |
| 10:46.29 | benjk | many people want DIDs over IAX |
| 10:46.37 | folder | benjk: good point |
| 10:47.09 | benjk | anyway, those PCI modem cards are no good |
| 10:47.19 | gr0mit | for anything other than mickeymouse installations do not use analogue trunks. |
| 10:47.32 | gr0mit | they are a waste of effort. |
| 10:47.35 | benjk | they are based on a chipset that is no longer manufactured |
| 10:47.44 | *** part/#asterisk NoNeo (n=ankamins@193.24.24.10) |
| 10:47.56 | RoyK | tempy: the 'standards' of pots vary between countries and unless you're really lucky, you'll end up using hours of debugging and at the end, call the telco, ask 'can you PLEASE give me ISDN' and then get a $15 ISDN card with an HFC chipset. |
| 10:47.58 | tempy | hmm, i might as well just buy an Engin box then, save the hassle, connect it to my Lan |
| 10:47.59 | benjk | but demand is there so the Chinese make new cards from refurbished chips |
| 10:48.20 | benjk | and also they have started using left over chips that didn't pass quality control |
| 10:48.25 | tempy | i have no idea of the cost of an ISDN line in .au |
| 10:48.31 | gr0mit | cheap. |
| 10:48.42 | gr0mit | we just installed a Telstra ISDN-10 |
| 10:48.56 | benjk | those cards were never really great, not even when they were made from new chips |
| 10:48.57 | macTijn | isdn-10 ? |
| 10:49.10 | benjk | but today, they are just trashy |
| 10:49.12 | folder | benjk: one of my 'genuine, original' X100P's came with a capacitor rolling around in the antistatic bag. The other one needed solder applying to two other capacitors as they were ready to fall out of the unsoldered holes. |
| 10:49.12 | Assid | hrmm i gotta put together a list of providers for UK and australia |
| 10:49.15 | macTijn | isn't that just an isdn-24/30 with only 10 channels enabled ? |
| 10:49.24 | gr0mit | a PRI with 10 chans enabled |
| 10:49.35 | macTijn | uhuh |
| 10:49.47 | macTijn | we only have 15 here in .nl as a minimum |
| 10:49.48 | benjk | if you really insist on analog, get a Sipura 3000 |
| 10:49.53 | folder | (I say 'genuine, original' sarcastically. That's what the eBay sellers describe them as, because they've changed the PCI IDs so it's recognised as an X100P) |
| 10:49.56 | macTijn | and it's only marinally cheaper |
| 10:50.07 | gr0mit | here BT do them min 8 chans |
| 10:50.18 | benjk | it has got one stage dialing and the newest firmware even supports UK caller ID now I think |
| 10:50.18 | gr0mit | then you can add chans 1 by one |
| 10:50.28 | macTijn | gr0mit: wow |
| 10:50.37 | benjk | there was never ever any such thing as an X100P |
| 10:50.50 | benjk | they are Ambient MD3200 softmodem PCI cards |
| 10:50.52 | macTijn | gr0mit: does it save much in comparison to a full PRI ? |
| 10:50.54 | benjk | all of them |
| 10:51.07 | gr0mit | yes - you pay per channel. |
| 10:51.20 | gr0mit | so a full pri they charge you for 30 channels. |
| 10:51.32 | gr0mit | for 8 chans they charge you for 8 channels. |
| 10:51.34 | macTijn | no base price ? |
| 10:51.38 | gr0mit | hmmm let me see |
| 10:52.10 | benjk | but you can use those modem PCI cards for zaptel timing |
| 10:52.50 | benjk | not as an FXO interface though |
| 10:53.00 | tempy | ISDN Home from telstra, one line after connection will cost 75% of my broadband line, stuff that |
| 10:53.11 | tempy | i'll go engin |
| 10:53.21 | *** part/#asterisk tempy (n=tempy@c220-237-91-41.rochd1.qld.optusnet.com.au) |
| 10:53.41 | benjk | here in Japan a BRI ISDN circuit costs exactly the same as an analog line |
| 10:54.02 | benjk | but on the analog line you pay extra for DTMF |
| 10:54.08 | macTijn | here in .nl it's around twice as expensive |
| 10:54.25 | folder | Europe as a whole seems to be mostly retarded. |
| 10:54.26 | benjk | so unless you want to use pulsedialing the ISDN circuit is actually less |
| 10:54.36 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 10:55.01 | folder | benjk: and the same for Caller-ID I seem to remember you saying.. |
| 10:55.05 | macTijn | folder: I think that's an overstatement, ever looked ad how cruddy .us is like nowadays ? :) |
| 10:55.17 | benjk | yeah, everything costs extra on the analog line |
| 10:55.30 | macTijn | analog lines here come with everything extra for free |
| 10:55.42 | gr0mit | http://www.downloads.bt.com/b4b/pdf/ISDN30e_pricing.pdf |
| 10:55.46 | benjk | including shitty sound quality :) |
| 10:55.53 | macTijn | benjk: no way :) |
| 10:55.54 | benjk | analog just sucks |
| 10:56.02 | macTijn | here it doesn't |
| 10:56.05 | folder | macTijn: well i meant everything is too expensive here (UK). Where are you located macTijn? |
| 10:56.10 | benjk | if I was in government, I would impose an analog phone tax |
| 10:56.17 | macTijn | folder: .nl, Amsterdam |
| 10:56.23 | folder | ah |
| 10:56.27 | macTijn | folder: phones are getting cheaper here |
| 10:56.35 | macTijn | and VoIP is getting quite popular |
| 10:56.38 | macTijn | anyway |
| 10:56.40 | macTijn | gtg |
| 10:56.46 | folder | well you chaps are quite relaxed overall really aren't you? |
| 10:56.53 | macTijn | ofcourse :) |
| 10:56.54 | benjk | obsolescence tax |
| 10:56.57 | folder | :D |
| 10:57.10 | folder | lol :D |
| 10:57.18 | macTijn | :P |
| 10:57.21 | macTijn | <- gone |
| 10:57.27 | folder | cya! |
| 10:57.33 | benjk | analog should be smoked out |
| 10:57.46 | benjk | double the tax every year |
| 10:58.03 | benjk | this is 150 year old crap that you are using there |
| 10:58.08 | folder | but it's required for the DSL. |
| 10:58.21 | benjk | DSL should be retired too |
| 10:58.39 | benjk | there is FTTH now |
| 10:58.48 | folder | There was a time when BT were pushing an ISDN2 variant into the homes. Pre-DSL. It was called "HomeHighway". It was basically (I think) ISDN2e but with a permanent POTS converter fitted to the second B-channel. |
| 10:59.12 | benjk | the Europeans were so retarded that they spend an outrageous amount on 3G wireless licenses |
| 10:59.36 | benjk | so much that it is mathematically impossible for any network to recover the cost of the license fee they paid |
| 10:59.46 | folder | 3g works well for me. Prices from Orange are more reasonable than most. 80p/megabyte. Vodafone charge £2.36/megabyte and no option to bundle. |
| 10:59.51 | gr0mit | yes they have written it off |
| 11:00.16 | benjk | the money would have been sufficient to connect every pig stall in Europe with fiber |
| 11:00.39 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 11:00.44 | folder | FTTH - Fibre to the Home. Is that any different to Cable? |
| 11:01.08 | vgster | not all the europeans, i thought the french refused to pay what the gov was asking for 3G licenses so they brought the price down |
| 11:01.21 | vgster | in the UK everyone had to jump on the 3G wagon, only for the wheel to come off |
| 11:01.40 | benjk | in the UK, in order for the license fees to be recovered during the validity period of those licenses, every UK citizen, including new born toddlers, would have to spend an average amount of 450 pounds per month for the next 20 years |
| 11:01.58 | vgster | yes it was crazy billions |
| 11:02.07 | vgster | ...how many new hospitals... |
| 11:02.22 | benjk | and like I said, the money would have been sufficient to completely cover all of Europe with fiber |
| 11:02.35 | folder | Why did they pay it then? |
| 11:02.39 | benjk | including every pig stall in the remotest mountain village |
| 11:02.54 | vgster | cos they thought they would miss out on the next big thing |
| 11:03.00 | folder | oh |
| 11:03.01 | vgster | so paid what the gov wanted |
| 11:03.08 | vgster | which in the UK was rediculous |
| 11:03.15 | benjk | because nobody thought they could afford to let their competitors have a license and not get one themselves |
| 11:03.25 | folder | well it is useful. I can sit in my car with my thinkpad and my k610i and connect to customers VPNs and do remote tech support. |
| 11:03.36 | benjk | they didn't pay "what the gov wanted" |
| 11:03.42 | benjk | it was an auctions |
| 11:03.44 | benjk | auction |
| 11:03.45 | folder | ah# |
| 11:03.58 | vgster | ok, but an auction with a starting price of X billion |
| 11:04.00 | benjk | they paid what they thought they had to pay to be on top of their competitors |
| 11:04.12 | vgster | but when they all have it whats the point |
| 11:04.14 | benjk | and the dynamics of the auction drove the cost through the roof |
| 11:04.45 | benjk | they had better spent that money on fiber |
| 11:04.56 | benjk | who needs DSL |
| 11:04.56 | vgster | didnt the money go to the gov for the airspace privelidge |
| 11:05.07 | benjk | DSL is just another transitional technology |
| 11:05.19 | folder | so where can I get my FTTH? |
| 11:05.36 | benjk | in Belgium, NL, Sweden, Korea, Japan |
| 11:05.41 | folder | oh |
| 11:05.42 | folder | :( |
| 11:05.53 | benjk | we three FTTH providers here |
| 11:05.53 | folder | what's the typical bandwidth provided? |
| 11:06.14 | benjk | one of them have a 50 USD per month all you can eat 100MBit full duplex package |
| 11:06.20 | folder | good greif |
| 11:06.26 | benjk | 100MBit/sec full duplex |
| 11:06.32 | folder | are you on that then? |
| 11:06.33 | Assid | i would love to have that kinda bandwith |
| 11:06.48 | benjk | and the ones with that package even give you a block of 8 ip addresses with it |
| 11:07.04 | benjk | all the others charge you quite a bit for the IP addresses |
| 11:07.09 | Assid | hrmm.. i wonder when ipv6 will mainstream |
| 11:07.16 | benjk | otherwise you get a dynamic ip |
| 11:07.21 | benjk | in 2035 |
| 11:07.31 | folder | that sucks. dynamic ip. |
| 11:07.33 | benjk | optimistically |
| 11:07.50 | benjk | yeah, but the base service is cheap, typically 50 USD per month |
| 11:08.00 | Assid | sucks.. we falling short of ips already.. i thought by 2007-2010 tops i'd see it |
| 11:08.01 | benjk | with one fixed ip it can be double that |
| 11:08.04 | folder | how much extra for a single static IP? |
| 11:08.11 | folder | pfft! |
| 11:08.13 | folder | crikey |
| 11:08.23 | benjk | in Japan ip addresses are what costs money |
| 11:08.29 | benjk | connectivity is dirt cheap |
| 11:08.43 | benjk | I pay 1000 yen for 8MBit ADSL per month |
| 11:08.50 | benjk | thats about 9 USD |
| 11:08.52 | mut | kinda like babies |
| 11:08.56 | mut | dirst cheap to make |
| 11:08.59 | mut | but hosting costs money |
| 11:09.00 | folder | that is cheap. Dynamic IP though? |
| 11:09.02 | Assid | what!?!?!?!? |
| 11:09.07 | Assid | 8mbit?!?!? |
| 11:09.12 | benjk | but I pay about 100 USD for 8 ip addresses |
| 11:09.42 | benjk | also, for reverse DNS, they charge you another 20 USD per month |
| 11:09.44 | Assid | i pay like $30/mo for 256kbit! |
| 11:09.56 | folder | My dsl is being upgraded to "up to 8mbps" (448kbps u/s) for free, but it still costs £25/month (about $40) |
| 11:10.16 | Assid | have your isp contact my isp! |
| 11:10.26 | benjk | our building just got fiber |
| 11:10.42 | benjk | so next month I am going to ditch my analog line which I had to keep for ADSL |
| 11:10.43 | folder | brb |
| 11:10.46 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.125) |
| 11:10.49 | MACscr | is DISA what is used when your voip phone is not connected directly to the asterisk system? ie, your remote |
| 11:11.08 | benjk | DISA == provide yourself a dialtone |
| 11:11.29 | Assid | man.. 1 day.. i will get 1mbit line.. i'd be soo damn happy |
| 11:12.00 | benjk | I think Singapore also has FTTH |
| 11:12.37 | MACscr | i guess i dont get that, usually dialtone is created by the phone system, which makes sense, but i get dialtone no matter if im hooked up to anything or not with my stupid grandstream gxp |
| 11:12.39 | benjk | so if you're in AU, you may want to consider moving to SG ;) |
| 11:12.52 | benjk | if you're in the UK, you may want to consider moving to BE or NL |
| 11:12.52 | Assid | whats FTTH again? |
| 11:12.57 | benjk | Fiber to the home |
| 11:13.04 | benjk | 100MBit full duplex |
| 11:13.10 | benjk | typcially |
| 11:13.20 | benjk | some providers may limit that bandwidth though |
| 11:13.39 | benjk | I have heard of some FTTH service which was limited to 20MBit |
| 11:13.56 | Assid | great i see all this technology .. i feel even more depressed cause of the way we are conned |
| 11:13.56 | benjk | and some other which was limited to 50Mbit |
| 11:14.49 | benjk | yeah, well, fiber is the natural way to send large amounts of digital data around over large distances |
| 11:15.00 | benjk | anything else is rather quaint |
| 11:15.10 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.125) |
| 11:15.35 | benjk | or an immense waste (ie wireless broadband) |
| 11:15.49 | mut | whats wrong with wireless broadband |
| 11:15.58 | benjk | its a waste of resources |
| 11:16.00 | *** join/#asterisk folder (n=carl0s@compsup.demon.co.uk) |
| 11:16.06 | mut | how ya figure? |
| 11:16.11 | MACscr | mut, if its the only option, then its ok =P |
| 11:16.11 | benjk | the wireless spectrum is limited by nature |
| 11:16.22 | mut | and? |
| 11:16.24 | benjk | you cant make it more than it is |
| 11:16.33 | mut | well sure, wireless broadband in new yourk city is crazy |
| 11:16.50 | mut | our company covers almost all northern michigan tho |
| 11:16.59 | benjk | wasting that on mobile data is as waste for most of the applications offered |
| 11:17.04 | mut | 60 some odd towers |
| 11:17.27 | benjk | it would make more sense to cover every bit of the inhabited surface of the planet with fiber |
| 11:17.44 | mut | and also be prohibitivly expensive to do so |
| 11:18.01 | benjk | then work out a scheme of unlicensed spectrum used for sharing the available bandwidth over fiber |
| 11:18.12 | benjk | less expensive than wireless |
| 11:18.18 | mut | hardly |
| 11:18.24 | benjk | why do we need wireless operators? |
| 11:18.33 | benjk | they are a waste of resources too |
| 11:18.38 | mut | operators? |
| 11:19.04 | benjk | just put a regulatory framework in place that makes every low range base station sharable |
| 11:19.22 | mut | heh |
| 11:19.32 | mut | you must have never been in a rural eara before |
| 11:19.34 | mut | area* |
| 11:19.34 | folder | grrr |
| 11:19.36 | benjk | then you walk down some alley some place where people live and you hop from base to base getting access to the underlying fiber |
| 11:20.07 | mut | sure, if you can cover a city in fiber more power to ya |
| 11:20.15 | benjk | this would be far more efficient and cost effective than all those mobile data services |
| 11:20.19 | mut | but what about the other 95% of the world |
| 11:21.06 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 11:21.07 | benjk | if you are in a really remote place, like a desert or a farm where there isn't any other living being for 100 miles, then wireless makes sense |
| 11:21.17 | benjk | but for 99% of the worlds pop that isn't the case |
| 11:21.17 | mut | 100 miles.. |
| 11:21.44 | mut | so you propose we run fiber down all these rural areas |
| 11:21.59 | benjk | I say it is cheaper than 3G mobile |
| 11:22.03 | mut | and that is less expensive somehow than wireless |
| 11:23.07 | benjk | its only about politics |
| 11:23.14 | Assid | great this stupid politics sucks |
| 11:23.18 | benjk | governments don't like anything that is decentralised |
| 11:23.27 | benjk | they want large centralised things |
| 11:23.37 | Assid | every phone prvider needs to have their own wires in the buildings to give us connetivity |
| 11:23.50 | Assid | they dont have it like in the US .. 1 line.. just change provider |
| 11:24.01 | benjk | if there is a fiber strand, you don't need that |
| 11:24.11 | mut | Assid: we run our own pairs |
| 11:24.12 | Assid | they dont wanna co-operate |
| 11:24.18 | mut | we don't run over verizons pairs |
| 11:24.43 | benjk | that's why I said "if the regulatory framwork was put in place" |
| 11:25.01 | benjk | its a political decision, not a technical one |
| 11:25.14 | mut | theres a pipe dream that won't happen for a very very long time |
| 11:25.35 | benjk | I was talking about what is economical and what is wasteful |
| 11:25.41 | benjk | has nothing to do with pipe dreams |
| 11:25.42 | Assid | well.. most providers over the world can just change the providers but still use the same carriers |
| 11:25.53 | benjk | fact is, the way things are organised is wasteful, that's all |
| 11:25.54 | mut | a fiber network everywhere is not economical |
| 11:26.03 | Assid | the same copper cable can be used for anyone else.. you just make it such that the network is expandablre to all |
| 11:26.07 | benjk | it is more economical than what we have now |
| 11:26.30 | Assid | imagine.. 3 telephone providers.. each has to fibre up.. cause they dont wanna share resources |
| 11:26.37 | mut | i dunno how ya figure that benjk but ok |
| 11:26.47 | mut | i'll just let ya think that |
| 11:26.55 | Assid | so every building needs 3 lines coming in.. means more equipment usage.. more lines.. etc |
| 11:27.00 | benjk | its not my figure, it was a study of the Yankee Group or Gartner, a few years back |
| 11:27.15 | mut | for urban areas probably |
| 11:27.16 | Assid | buildings dont like to have their ground dug up.. so they dont give permission.. means i lose out |
| 11:27.24 | benjk | no, for the whole of Europe |
| 11:27.39 | mut | urban areas |
| 11:27.40 | benjk | specifically every little corner in the whole of Europe |
| 11:28.07 | benjk | including the north of Norway and Sweden and Finland where there isn't a living soul for a 100 miles |
| 11:28.25 | mut | how did they propose funding it? |
| 11:28.35 | benjk | they didn't propose anything |
| 11:28.40 | mut | all the fiber, installation, interconnecting equipment, maintainance, etc.. |
| 11:28.56 | benjk | they calculated that the money spent on 3G infrastructure could have been used for such a fiber network |
| 11:29.11 | Assid | i justcalled up a provider ready to give me 256kbit for 30% cheaper than what im paying.. i cant chose it.. cause they dont have any cabling to me |
| 11:29.23 | benjk | and they gave very sane reasons why that would have been better value for money than a 3G network |
| 11:29.36 | mut | well |
| 11:29.48 | mut | i can't speak for what they're spending on the 3G network and it's coverage/costs |
| 11:30.07 | benjk | and I have to say it makes a lot of sense |
| 11:30.17 | mut | but i know building a fiber network compared to running wireless here is still prohibitivly expensive |
| 11:30.38 | benjk | but fiber has far better economies of scale |
| 11:30.50 | mut | right |
| 11:30.57 | mut | then you run into big business controlling it |
| 11:31.02 | benjk | its more universal |
| 11:31.02 | mut | and singular entities |
| 11:32.11 | benjk | couldn't be any worse than what the status quo is already |
| 11:33.12 | benjk | allow two or three providers in the big cities, force a single provider into universal coverage in return for a monopoly in the places with less pop density |
| 11:33.55 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 11:34.30 | mut | then you run businesses like ours right out of existence |
| 11:34.42 | mut | if a single company could provide fiber to all areas |
| 11:34.44 | benjk | that's the status quo already today |
| 11:34.49 | mut | hardly |
| 11:35.01 | benjk | not in the US, but in Europe it is |
| 11:35.15 | mut | there would be no need for any dialup companies anymore, no dsl companies, no wireless companies |
| 11:35.33 | benjk | there are those former PTTs who have a universal coverage obligation and a de facto monopoly in rural areas |
| 11:35.56 | benjk | and there are multiple providers in higher pop density areas |
| 11:36.41 | benjk | who needs DSL or dialup anyway |
| 11:36.49 | benjk | that should be phased out asap |
| 11:36.59 | mut | heh |
| 11:37.05 | mut | dialup is a cash cow for rural areas |
| 11:37.06 | benjk | seriously |
| 11:37.12 | mut | you have no idea man |
| 11:37.16 | MACscr | i like my dsl connection |
| 11:37.40 | mut | there is no infrastructure for anything faster in most of the US |
| 11:37.41 | MACscr | 6mb for $45 isnt bad |
| 11:37.50 | mut | just larger cities |
| 11:37.56 | mut | dsl is expanding fast tho |
| 11:38.03 | MACscr | which still pay out the wazoo as well |
| 11:38.07 | mut | but that only reaches to city limits |
| 11:38.14 | benjk | well of course the US is technologically behind |
| 11:38.27 | mut | no, the us is just larger than europe |
| 11:38.36 | benjk | they don't even have a functioning train system, get out of here |
| 11:38.41 | MACscr | and we have had A infrustructure a lot longer |
| 11:38.42 | mut | persons per mile is smaller |
| 11:38.45 | MACscr | your just getting yours |
| 11:39.11 | benjk | the reasons don |
| 11:39.13 | benjk | t matter |
| 11:39.20 | MACscr | lol |
| 11:39.30 | mut | ok |
| 11:39.30 | *** join/#asterisk Zauephuaes (n=guido@80.87.87.101) |
| 11:39.35 | benjk | fact is the US is run by lobbyists with exconomic interests that are not always for the best of the country |
| 11:39.37 | mut | thats like me putting you in jail |
| 11:39.40 | mut | cause the reasons don't matter |
| 11:40.13 | benjk | for example GM destroying the railways |
| 11:40.13 | MACscr | benjk, where are you from again? |
| 11:40.13 | benjk | that was good bbusiness for GM |
| 11:40.13 | Zauephuaes | AGI stream file doesn't output any sound for me (even with a gsm file from the asterisk distro). how can i troubleshoot this? |
| 11:40.13 | MACscr | benjk, we could care less about trains |
| 11:40.14 | benjk | but it was bad for the country |
| 11:40.26 | mut | i see plenty of trains around here.. |
| 11:40.26 | benjk | you better start caring about trains |
| 11:40.42 | MACscr | and why should we start caring? |
| 11:40.51 | benjk | because the oil price isn't going to stop rising |
| 11:40.56 | mut | they carry lots of crops around |
| 11:41.16 | benjk | it;ll go over 100 USD a barrel and further, you'll see |
| 11:42.35 | MACscr | maybe im retarded, bunt most of our trains here run on diesel |
| 11:42.48 | Zauephuaes | is there an asterisk channel anywhere around? I see this is politics :-) |
| 11:42.50 | MACscr | woops, meant, but |
| 11:42.52 | benjk | but the efficiency is way higher than with cars |
| 11:42.54 | Zauephuaes | and economics :-) |
| 11:43.24 | MACscr | benjk, we dont like trains because they arent convenient enough for us |
| 11:43.40 | benjk | in Australia they have road trains, they're essentially trains that are road based without rails |
| 11:43.45 | EyeCue | o_O |
| 11:43.51 | EyeCue | theyre called trucks. |
| 11:43.51 | MACscr | and if you havent figured it out yet, we have a lot more money than you =P |
| 11:43.51 | EyeCue | :) |
| 11:44.01 | benjk | that's for cargo only, but its efficient, far more efficient that individual trucks |
| 11:44.03 | MACscr | lol |
| 11:44.06 | mut | trucks?! |
| 11:44.07 | mut | no?! |
| 11:44.15 | EyeCue | yes :) |
| 11:44.25 | mut | omfg i'm going to die happy |
| 11:44.32 | MACscr | ha |
| 11:44.41 | benjk | they are not trucks, and they are not called trucks |
| 11:44.47 | benjk | they are called road trains |
| 11:44.52 | mut | pff |
| 11:45.05 | EyeCue | theyre called road trains, but theyre trucks with > 1 trailer. |
| 11:45.10 | folder | http://en.wikipedia.org/wiki/Image:Road_Train2.jpg |
| 11:45.20 | folder | ^^ that's mental! |
| 11:45.43 | EyeCue | yeh, trucks :) |
| 11:45.55 | mut | no |
| 11:45.57 | mut | it's a road train |
| 11:45.58 | MACscr | exactly, did u see below where it says "truck" |
| 11:46.00 | mut | get it right |
| 11:46.08 | mut | ;) |
| 11:46.33 | mut | it'de definatly suck driving one of those |
| 11:46.38 | MACscr | heck yeah |
| 11:46.42 | benjk | the longest road train is about a mile long and it has about 2000 wheels |
| 11:46.43 | EyeCue | talk about jack knifing. |
| 11:46.46 | EyeCue | good luck reversing. |
| 11:46.49 | benjk | that's clearly not a truck |
| 11:46.58 | mut | thats a fricking truck |
| 11:47.04 | MACscr | lol, its a truck |
| 11:47.06 | EyeCue | its a fscking truck. |
| 11:47.07 | benjk | its a road train |
| 11:47.10 | MACscr | it just has multiple trailers |
| 11:47.18 | folder | dude, they definately call them Road Trains |
| 11:47.20 | benjk | a mile long |
| 11:47.29 | folder | http://www.news.com.au/story/0,10117,18192186-29279,00.html |
| 11:48.03 | mut | ok |
| 11:48.07 | MACscr | i think truck was repeated about 20 times in that article |
| 11:48.16 | folder | "A road train is a truck design used in remote areas of Australia, United States and Western Canada to move bulky loads efficiently.". So it's a truck with many trailers, which is then called a Road Train :D |
| 11:48.55 | MACscr | Truck is the official name, road train is slang |
| 11:48.56 | MACscr | =P |
| 11:49.06 | folder | You're all correct. Lets have a group hug. |
| 11:49.08 | folder | :p |
| 11:49.10 | benjk | the point is however that there is one or two engines which pull engineless trailers/carriages |
| 11:49.33 | benjk | and that this is more efficient than a one engine per carriage scheme |
| 11:49.35 | mut | good job |
| 11:49.41 | MACscr | lol, duh |
| 11:49.47 | MACscr | what does that have to do with anything? |
| 11:49.56 | mut | it's not much more effcient tho |
| 11:50.02 | benjk | it has to do with efficiency and avoiding waste |
| 11:50.03 | mut | you still use more power to carry the load |
| 11:50.10 | benjk | thats what we were talking about |
| 11:50.30 | benjk | the cost per ton of cargo is still significantly less |
| 11:50.50 | benjk | and in passenger transport, the cost per passenger is significantly less |
| 11:51.03 | MACscr | correct, btu we dont WANT IT |
| 11:51.08 | MACscr | woops, i meant but |
| 11:51.09 | benjk | you soon will |
| 11:51.19 | MACscr | nope |
| 11:51.44 | benjk | lets see what happens when the oil prices go above 100 USD per barrel |
| 11:51.48 | MACscr | we will use other technologies to make cars more efficient |
| 11:51.59 | folder | I so looked like that dude who likes free software just then. I came downstairs with a keyboard in my hand tapping away at it and when I looked in the mirror I thought "geek!" |
| 11:52.01 | mut | ahh |
| 11:52.02 | mut | man |
| 11:52.05 | MACscr | we already have a hacked prius that does 100mpg |
| 11:52.09 | mut | did you guys see the tesla roadster? |
| 11:52.11 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:52.14 | mut | it's frickin badass |
| 11:52.28 | benjk | thats friggin wasteful |
| 11:52.39 | mut | http://www.autoblog.com/2006/07/20/tesla-roadster-unveiling-in-santa-monica/ |
| 11:52.58 | folder | yeah. I'm guessing they nicked the name from Nicholas Tesla.. |
| 11:53.24 | benjk | Tesla is going to spin in his grave |
| 11:53.33 | mut | yea folder thats the name of the company |
| 11:53.39 | MACscr | that car rocks |
| 11:53.43 | benjk | because he was all about efficiency |
| 11:53.44 | Zauephuaes | Hi everyone, I don't mean to be insistent and don't expect free help. it would be great if someone just said go away and check the mailing list or something |
| 11:54.03 | Zauephuaes | but hearing trucks when we want to talk asterisk is a bit ... |
| 11:54.05 | mut | Zauephuaes: i have no idea what the problem would be if that helps |
| 11:54.05 | Zauephuaes | frustrating |
| 11:54.16 | EyeCue | Zauephuaes, if you want to talk asterisk, then talk it. |
| 11:54.17 | benjk | he quit his job with Edison because he found Edison's philosophy too wasteful |
| 11:54.17 | Zauephuaes | mut: thanks! |
| 11:54.25 | mut | nor does anyone else if they aren't helping |
| 11:54.27 | EyeCue | all nerds must have downtime |
| 11:54.29 | benjk | then he went to Westinghouse |
| 11:54.30 | mut | EyeCue: he already did... |
| 11:54.42 | benjk | where his ideas of efficiency were appreciated |
| 11:54.44 | Zauephuaes | I'm using STREAM FILE via AGI but it isn't playing anything. no sound output to my softphone. |
| 11:55.01 | MACscr | yep, and who is remembered more? |
| 11:55.10 | benjk | Tesla of course |
| 11:55.18 | benjk | there ins't any unit called Edison |
| 11:55.19 | MACscr | lol |
| 11:55.26 | benjk | there is a unit called Tesla |
| 11:55.30 | Zauephuaes | i get a beep playing from get data, but no sound from stream file. grr |
| 11:55.44 | benjk | and whose system is standard today? |
| 11:55.47 | benjk | not Edisons |
| 11:55.52 | benjk | Teslas is |
| 11:55.52 | mut | pastebin the agi.. |
| 11:56.02 | Zauephuaes | ahh well. enjoy the trucks. I think I can solve my problem by myself! |
| 11:56.03 | Zauephuaes | lol |
| 11:56.04 | *** part/#asterisk Zauephuaes (n=guido@80.87.87.101) |
| 11:56.09 | MACscr | ha |
| 11:56.18 | benjk | and Edison himself later said not listening to Tesla was the biggest blunder in his life |
| 11:56.25 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 11:57.02 | benjk | that was about the power distribution in case you don't know |
| 11:57.04 | mut | heh |
| 11:57.06 | benjk | AC versus DC |
| 11:57.10 | mut | well i was going to try to help him and he left |
| 11:57.18 | benjk | DC distribution is inefficient and wasteful |
| 11:58.39 | mut | i best get back to work tho |
| 11:58.45 | mut | get some coffee |
| 11:58.46 | mut | adios |
| 11:58.49 | benjk | yeah, good idea |
| 11:58.51 | benjk | see you |
| 11:59.23 | *** part/#asterisk MACscr (n=MACScr@adsl-75-23-81-217.dsl.peoril.sbcglobal.net) |
| 12:00.20 | folder | brb again |
| 12:06.34 | *** join/#asterisk Swat2 (n=bler@dsl-202-72-187-123.wa.westnet.com.au) |
| 12:07.35 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 12:08.07 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 12:08.07 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 12:08.25 | *** join/#asterisk folder (n=carl0s@compsup.demon.co.uk) |
| 12:08.41 | folder | the train kept a rollin', all night long.. train kept a rollin' all night long.. |
| 12:08.48 | folder | or should that be truck? |
| 12:08.50 | folder | lol |
| 12:09.27 | rob0 | "Edison" will be the name of the first electric pickup truck. |
| 12:09.41 | folder | yeah right. Although it does have a sort of truck ring about it |
| 12:09.58 | rob0 | (If the marketers think the "Edsel" connection is distant enough.) |
| 12:10.12 | Swat2 | <PROTECTED> |
| 12:10.34 | folder | oh ffs. I need to shutdown again. I've left my USB <-> IDE cable on site and am fixing someones hd. |
| 12:10.35 | folder | brb |
| 12:10.37 | folder | again |
| 12:12.54 | *** join/#asterisk bkidney (n=bkidney@nat01-128.stratos.ca) |
| 12:13.33 | bkidney | Can anyone help me with a zaptel setup problem? |
| 12:16.54 | benjk | just ask your question, if somebody can and want to help they will answer |
| 12:18.05 | *** join/#asterisk Mashy (n=r@newcastle.rbip.net) |
| 12:18.29 | bkidney | I am trying to change my FXO port on a TDM11 card from fxsks to fxsgs, but when I run the "ztcfg -vv" I get the error "ZT_CHANCONFIG failed on channel 4: Invalid argument (22)." Anyone know what I might be doing wrong? |
| 12:19.10 | *** join/#asterisk truckdrivingman (n=carl0s@compsup.demon.co.uk) |
| 12:19.29 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:19.32 | benjk | have you rebooted the system? |
| 12:19.37 | bkidney | Yes |
| 12:19.50 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
| 12:20.35 | *** join/#asterisk javar (n=javar@69.79.216.179) |
| 12:20.51 | benjk | invalid argument (unless its a misleading error message) sounds a lot like some typo in the configuration file though |
| 12:20.53 | Modcuts | Anybody used voipGate in the uk as iax2 provider? |
| 12:21.07 | javar | hi benjk |
| 12:21.18 | benjk | hi javier |
| 12:21.28 | javar | how are you? |
| 12:21.36 | benjk | good |
| 12:21.52 | javar | i was trying contact to you... |
| 12:22.01 | benjk | I was overseas |
| 12:22.13 | javar | see |
| 12:22.20 | tzanger | benjk: your mom's basement isn't overseas |
| 12:22.41 | benjk | actually it is |
| 12:23.08 | bkidney | The config file works fine if I change the fxsgs to fxsks and make no other changes. |
| 12:23.21 | bkidney | I think it is a misleading error. |
| 12:23.50 | tzanger | benjk: :-) |
| 12:23.54 | benjk | then all I can suggest is google for the error message and/or look in the source code for clues |
| 12:24.38 | bkidney | I tried google, so I it's into the code for me. |
| 12:24.43 | bkidney | Thanks for the help. |
| 12:25.03 | benjk | somebody else here may know better |
| 12:25.33 | *** join/#asterisk antgel (n=antony@pdpc/supporter/monthlybyte/antgel) |
| 12:25.45 | antgel | hi all |
| 12:25.51 | folder | hiya |
| 12:26.02 | folder | a/s/l? |
| 12:26.14 | folder | (only kidding) |
| 12:26.14 | antgel | you're kidding, right? |
| 12:26.17 | folder | :D |
| 12:26.18 | antgel | :) |
| 12:26.26 | folder | think i've had too much coffee |
| 12:26.30 | antgel | hey, you're mirroring what i type |
| 12:26.39 | folder | heh yeah :) |
| 12:26.56 | benjk | the last TDM400 card I was laying my hands on has been blown up by the Israeli Air force |
| 12:27.18 | antgel | i'm new to asterisk, and new to complex telephony. a simple question... |
| 12:28.10 | antgel | a customer with a conferencing venue wants people to be able to dial in to hear the conference. is this a job for asterisk? |
| 12:28.31 | docelmo | yes you can do this in asterisk |
| 12:28.45 | antgel | i think i will recommend streaming it over the internet, but i'd like to explore this possibility as well |
| 12:28.56 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 12:29.03 | antgel | is this the "conference bridging" feature in the feature list? |
| 12:29.23 | [TK]D-Fender | antgel : Yes, and relatively easy to set up |
| 12:29.36 | benjk | how many people in the conference? |
| 12:29.42 | antgel | small. circa 5 |
| 12:29.48 | benjk | that's peanuts |
| 12:29.52 | antgel | indeed |
| 12:29.58 | benjk | and the setup is very simply |
| 12:29.59 | [TK]D-Fender | antgel : What you are talking about is *'s MeetMe application. I've done some decenlty sized conferences on my PRI over here with it. |
| 12:30.02 | benjk | simple |
| 12:31.11 | antgel | okay. i'm not sure if the lines will be pstn or isdn, but presumably it doesn't matter as long as i configure * properly :) |
| 12:31.17 | benjk | basically a single line in the configuration file |
| 12:31.29 | antgel | and presumably i configure it to take input from a sound card? |
| 12:32.30 | benjk | how the participants dial in doesn't matter, it could be VoIP as well |
| 12:32.39 | *** join/#asterisk kiddy (n=achu@220.225.191.38) |
| 12:32.55 | antgel | thanks a bunch, much appreciated all |
| 12:33.01 | Dr-Linux|work | anybody can tell me please what's wrong with brakets? >> exten => 9,1,GotoIf($["${DB(DISA/${CALLERID(num))}" = "1"]?2:5 |
| 12:33.04 | kiddy | can anybody tell me how to record calls (INand OUT) |
| 12:33.28 | kiddy | I see some configuration option for this in freepbx front end |
| 12:33.53 | benjk | I think there is a dedicated channel for freepbx |
| 12:33.59 | [TK]D-Fender | kiddy : Please read the channel topic... |
| 12:34.06 | benjk | #freepbx probably |
| 12:34.25 | kiddy | benjk : Can you tell me how manually we can do the call recording ? |
| 12:34.46 | benjk | monitor is your friend |
| 12:35.08 | benjk | search Voip-info.org for "asterisk monitor" there should be plenty of examples |
| 12:35.23 | kiddy | But what extension we have to dial for record the calls ? |
| 12:35.37 | benjk | as I said, there are examples available |
| 12:35.49 | kiddy | ok |
| 12:39.53 | benjk | Dr-Linux, your parentheses are off |
| 12:41.04 | [TK]D-Fender | Dr-Linux|work " exten => 9,1,GotoIf($["${DB(DISA/${CALLERID(num)})"="1"]?2:5) |
| 12:41.05 | *** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk) |
| 12:41.14 | benjk | should be GotoIf($["${DB(DISA/${CALLERID(num)}))}" = "1"]?2:5) |
| 12:41.32 | [TK]D-Fender | benjk : now yours are off ;) |
| 12:41.49 | benjk | no, you need to close the parens for (DB and (DISA |
| 12:42.05 | benjk | ah, no DB is { |
| 12:42.11 | [TK]D-Fender | benjk : ;) |
| 12:42.18 | benjk | but yours don't match either ;) |
| 12:42.56 | [TK]D-Fender | benjk : True... |
| 12:42.57 | *** join/#asterisk aRJAy (n=aRJAy@218-214-130-112.people.net.au) |
| 12:43.11 | aRJAy | re all |
| 12:43.11 | [TK]D-Fender | Dr-Linux|work " exten => 9,1,GotoIf($["${DB(DISA/${CALLERID(num)})}"="1"]?2:5) |
| 12:43.15 | [TK]D-Fender | there ! |
| 12:43.42 | aRJAy | I could do with a hand configuring a Sipura 3000. Is there anyone about that has some experience in that area? :) |
| 12:43.43 | benjk | CALLERID(num) then } for closing {CALLERID then ) for closing (DISA then } for closing {DB then "] for closing "[ |
| 12:43.54 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 12:43.58 | benjk | and finally ) at the end to close GotoIf( |
| 12:44.59 | benjk | Dr-Linux, you may want to use an editor that matches parentheses, braces and brackets for you |
| 12:45.20 | folder | does vi do that for the asterisk styles? |
| 12:45.27 | [TK]D-Fender | Bah... fingers & toes is where its at! |
| 12:45.49 | benjk | otherwise, you should close each opening paren/brace/bracket immediately and insert your stuff later |
| 12:46.08 | aRJAy | <--- Needs help with a SPA3000 config. Anyone here got experience on that product? |
| 12:46.10 | benjk | heh |
| 12:46.12 | aRJAy | :) |
| 12:46.24 | benjk | yeah, unfortunately |
| 12:46.31 | benjk | SPA is a nightmare to configure |
| 12:46.42 | aRJAy | So I'm finding out. |
| 12:46.56 | aRJAy | And there seems to be little support on their site too.. |
| 12:47.07 | benjk | best is to factory reset the damn thing and start over in very small steps |
| 12:47.40 | aRJAy | I'm trying to force a dial code that enables me to hook directly to a POTS connection (regular phone)... |
| 12:47.57 | aRJAy | I'm a total newb in the area... so eaaaasy on me :) |
| 12:48.04 | [TK]D-Fender | aRJAy : Go to www.voxilla.com Sipura forum. there are some excellent guides in th |
| 12:48.08 | aRJAy | I'm sure it's something simple |
| 12:48.13 | [TK]D-Fender | there* |
| 12:48.18 | aRJAy | Thanks Fender.. |
| 12:48.44 | [TK]D-Fender | aRJAy : Np. I'v seen them, very comprehensive by users who reall use these things for FAR more that ust * |
| 12:51.29 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 12:51.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:56.44 | docelmo | Tony, do you know anything about who fixed the issue in asterisk to pull back the 2nd VIA in a sip header? |
| 12:57.23 | aRJAy | [TK]D-Fender: looks like a good resource. Thanks again. |
| 12:57.35 | *** join/#asterisk roving_prole (n=Harper@72-254-127-253.client.stsn.net) |
| 12:59.04 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 12:59.44 | anthm | me tony or someone else? |
| 13:01.35 | benjk | wasn't that "me, my clones and Irene" ? |
| 13:01.55 | docelmo | you tony |
| 13:03.39 | anthm | i just made a patch the other day for doing a count on how many of a certian header there was and a way to access it by index # |
| 13:04.40 | anthm | oh yeah here |
| 13:04.41 | anthm | http://bugs.digium.com/view.php?id=7576 |
| 13:06.01 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 13:06.46 | jbalcomb | The internet is down. :( |
| 13:06.46 | *** part/#asterisk kiddy (n=achu@220.225.191.38) |
| 13:06.50 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 13:07.01 | anthm | i have not even run svn trunk but i made the patch against it since that is usually all they take it was for a guy who complaining about it in the dev channel and being told he didnt beling there asking such silly questions so I was motivated to toss it up there he was using 1.2 and he had to backport it a bit |
| 13:08.48 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.55 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:08.56 | *** join/#asterisk Alric (n=nbowyer@ppp-db.1stel.com) |
| 13:09.03 | Alric | Hello everyone |
| 13:10.47 | Alric | I upgraded a machine to 1.2.10 asterisk and 1.2.7 zaptel two days back, and the whole machine has deadlocked 2 times since then. Didn't seem to have problems on 1.2.6. There aren't any logs that I've seen indicating what the problem is. Any ideas? |
| 13:10.59 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 13:11.38 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 13:12.10 | benjk | yeah |
| 13:12.22 | benjk | go back to the version you were running before |
| 13:12.47 | *** join/#asterisk s41ted (n=christop@203.122.248.254) |
| 13:13.09 | s41ted | hi all! could any one help me with a quick astrisk voicemail question? |
| 13:13.36 | s41ted | Where would i be able to find documentation on the astrisk voicemail IVR? |
| 13:14.21 | benjk | for astrisk I don't know |
| 13:14.26 | [TK]D-Fender | s41ted : As in something like a "Quick User's Guide" for the menu structure? |
| 13:14.31 | s41ted | yep |
| 13:14.36 | benjk | but for asterisk, you may want to try voip-info.org |
| 13:14.37 | boddy | hii all I configured * with one port digium e1/t1 card to connect meridian but Ican make just one call when I am trying to 2 call in same time I have received "Jul 28 16:03:03 NOTICE[17121] app_dial.c: Unable to create channel of type 'Zap' |
| 13:14.37 | boddy | <PROTECTED> |
| 13:14.42 | [TK]D-Fender | benjk : Don't be anal... bkw may be lurking~! |
| 13:14.46 | boddy | have you any idea ? |
| 13:14.48 | RoyK | ~nickometer s41ted |
| 13:15.03 | *** join/#asterisk dacleric (n=dacleric@p54823D9B.dip0.t-ipconnect.de) |
| 13:15.17 | boddy | I am trying to sip call |
| 13:15.32 | s41ted | 41 has a meaning by the way.... not much of a leet speaker myself, but i just dont do it for kicks :) |
| 13:15.40 | [TK]D-Fender | RoyK : If your lack of lame-ness to the jbot nickometer is going to be your "claim to fame", then your 15 seconds ran our a LONG time ago... |
| 13:15.59 | RoyK | [TK]D-Fender: just playing with it... |
| 13:15.59 | [TK]D-Fender | out* |
| 13:16.24 | s41ted | in particular how long does the temporary message stay for? |
| 13:16.33 | coppice | ~nickometer [TK]D-Fender |
| 13:16.45 | s41ted | when you record it? |
| 13:17.02 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 13:17.09 | boddy | ? |
| 13:17.12 | [TK]D-Fender | jbot_ : What are you talking about?!?! I am FULLY mobile! |
| 13:17.47 | Mercestes | Is Jbot abusing the residents again? |
| 13:18.02 | coppice | what exactly does 76% lame mean? slightly more than 1.5 legs missing? |
| 13:18.38 | *** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
| 13:18.45 | RoyK | coppice: :) |
| 13:18.46 | Iam8up|lpy | can anyone tell me where the voicemail messages are stored? |
| 13:18.57 | Mercestes | /var/spool/asterisk/voicemail/default |
| 13:18.59 | Iam8up|lpy | i thought it was /var/lib/asterisk/vm but that dir doesn't exist |
| 13:19.25 | *** join/#asterisk |oranjia| (n=anban@dsl-146-58-127.telkomadsl.co.za) |
| 13:19.32 | Iam8up|lpy | Mercestes - could you be a huge help and give me a dir structure on those? |
| 13:19.33 | |oranjia| | hello world :) |
| 13:19.50 | Mercestes | Iam8up|lpy Yes.... |
| 13:19.52 | Iam8up|lpy | such as voicemail/default and voicemail/extension? |
| 13:20.00 | Iam8up|lpy | or is it voicemail/default/extension? |
| 13:20.10 | boddy | [TK]D-Fender have you any idea ? |
| 13:20.10 | Mercestes | give me a /msg. |
| 13:20.12 | Iam8up|lpy | our asterisk box has _no_ voicemail folder... |
| 13:20.13 | Mercestes | I can't spell your name..:P |
| 13:20.16 | boddy | about my problem |
| 13:21.05 | *** part/#asterisk Alric (n=nbowyer@ppp-db.1stel.com) |
| 13:21.43 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67B03.pool.t-online.hu) |
| 13:23.01 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 13:23.02 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 13:23.46 | boddy | ? |
| 13:23.47 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 13:25.38 | boddy | Jul 28 16:03:03 NOTICE[17121] app_dial.c: Unable to create channel of type 'Zap' |
| 13:25.38 | boddy | <PROTECTED> |
| 13:26.46 | Mercestes | boddy: check under /dev/zap and make sure ./zap and the channels under it aren't owned by root or something. |
| 13:26.56 | Mercestes | boddy: and make sure you *have* zap channels... |
| 13:27.36 | [TK]D-Fender | boddy :show CLI output of the successful & the failed call. |
| 13:28.35 | s41ted | does any one know the purpose of recording a temporary message in the voicemail IVR? |
| 13:28.40 | s41ted | what is the purpose? |
| 13:30.20 | boddy | Mercestes I can make one call without error |
| 13:30.41 | hmmhesays | so you can chroot your buildroot filesystem on this badboy |
| 13:30.45 | hmmhesays | so you can test stuff out |
| 13:31.03 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 13:32.39 | boddy | Mercestes How can I sure I have zap channels |
| 13:32.40 | boddy | <PROTECTED> |
| 13:32.41 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 13:33.22 | boddy | [TK]D-Fender :sip debug enable |
| 13:33.32 | boddy | and you want sip messages ? |
| 13:33.36 | boddy | on cli ? |
| 13:35.04 | *** part/#asterisk s41ted (n=christop@203.122.248.254) |
| 13:35.19 | Mercestes | boddy: /dev/zap/ should list zap channels. |
| 13:38.10 | *** join/#asterisk volp (n=volp@201.210.81.254) |
| 13:38.26 | Innatech | Can anyone explain/suggest why on a two-line Linksys RPT-300 ATA, the first line will register and work normally with the second line disabled, but with both lines enabled neither line will register with * ? |
| 13:39.20 | boddy | asterisk:/dev/zap # ls |
| 13:39.20 | boddy | . 10 13 16 19 21 24 27 3 4 7 channel timer |
| 13:39.20 | boddy | .. 11 14 17 2 22 25 28 30 5 8 ctl |
| 13:39.21 | boddy | 1 12 15 18 20 23 26 29 31 6 9 pseudo |
| 13:39.21 | boddy | asterisk:/dev/zap # |
| 13:40.39 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 13:41.07 | *** join/#asterisk useopenstupid (n=peter@cpc3-norw2-0-0-cust392.pete.cable.ntl.com) |
| 13:42.15 | *** part/#asterisk useopenstupid (n=peter@cpc3-norw2-0-0-cust392.pete.cable.ntl.com) |
| 13:43.09 | [TK]D-Fender | boddy : I wanted basic CLI output at verbose 10. |
| 13:44.29 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 13:44.51 | benjk | verbosity 10? |
| 13:44.59 | benjk | I think the highest is 3 |
| 13:45.43 | boddy | [TK]D-Fender output rushing |
| 13:45.45 | jbalcomb | benjk: 99 is the highest |
| 13:45.53 | benjk | since when is that? |
| 13:46.11 | jbalcomb | atleast 1.2.1 |
| 13:46.12 | benjk | I have never seen anything in the source that would look for anything higher than 3 |
| 13:47.09 | *** join/#asterisk andew (n=andew@84-45-170-202.no-dns-yet.enta.net) |
| 13:47.58 | jbalcomb | benjk: it'd be nice if anyone had any idea what is shown at each number regardless of how high the number goes |
| 13:48.31 | benjk | feel free to comb through the source code and document it |
| 13:49.16 | jbalcomb | feel free to not speak uselessly |
| 13:49.46 | benjk | ditto |
| 13:49.46 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:49.48 | boddy | ? |
| 13:50.17 | boddy | I am trying to make to g729 call in same time |
| 13:52.18 | *** join/#asterisk nortex (n=nortex@64.136.65.144) |
| 13:53.04 | boddy | [TK]D-Fender is there any way to redirect output to a file ? |
| 13:53.16 | boddy | because everything is rushing |
| 13:53.32 | Mercestes | boddy: yes.... |
| 13:53.38 | boddy | how ? |
| 13:53.39 | Mercestes | boddy: asterisk -r >> catchme.txt |
| 13:53.47 | boddy | ok |
| 13:55.01 | mut | just a little injection for the lull |
| 13:55.03 | mut | Harry Potter actor Daniel Radcliffe is to appear on the London stage next year, playing a stable boy who has an erotic relationship with his horses. |
| 13:55.23 | Mercestes | ... |
| 13:55.24 | benjk | seems appropriate |
| 13:55.31 | Mercestes | you've got to be kidding me. |
| 13:55.46 | hmmhesays | can I disable logger completely? |
| 13:56.08 | coppice | mut: a new use for his magic wand |
| 13:56.22 | *** join/#asterisk vlt (n=daniel@dslb-088-073-227-095.pools.arcor-ip.net) |
| 13:56.23 | mut | :P |
| 13:56.52 | jbalcomb | hmmhesays: i would think you could unload the logger and put load=no for the logger in modules |
| 13:57.00 | Mercestes | Is Hermironie going to be in it? I might want that on DVD> |
| 13:58.06 | mut | awww |
| 14:00.34 | [TK]D-Fender | boddy : How is basic dialplan flow "rushing" when you can't even get 2 calls in? |
| 14:02.29 | boddy | Please wait I pasteing pastebin |
| 14:04.34 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 14:04.53 | jbalcomb | [TK]D-Fender: did you test your connection yet? |
| 14:05.28 | aRJAy | hmm... still got probs with my Sipura |
| 14:05.47 | aRJAy | no matter what plan I put in there, I'm getting engaged signal... |
| 14:06.02 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 14:06.21 | cy3o3 | What do you guys think of voicepulse as a voip provider? |
| 14:06.35 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.241) |
| 14:06.40 | Kernel_core | hi all |
| 14:07.33 | [TK]D-Fender | jbalcomb : tonight |
| 14:07.48 | Kernel_core | I asked this question in #FreePBX but nobody answered , maybe you could help me :) |
| 14:08.01 | jbalcomb | [TK]D-Fender: alright. I'll be in tomorrow around noon if your available. |
| 14:08.05 | *** join/#asterisk rvn (n=danny@host-87-75-150-91.bulldogdsl.com) |
| 14:08.06 | Kernel_core | today I translated AMP to FARSI ... but when I choose FARSI , I get "???????" in my browser instead of correct characters ....( I used in my .po file UTF-8 as charset ) what is wrong ? and how could I fix it ? |
| 14:08.10 | jbalcomb | s/your/you're |
| 14:08.26 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:09.41 | [TK]D-Fender | jbalcomb : Tomorrow afternoon is eaten up for me. I'd like to do the SSH today with you actually. |
| 14:10.40 | hmmhesays | Kernel_core: sounds like you are missing a language pack |
| 14:10.49 | rvn | hi guys, just wondering if anyone on here has managed to get the Cisco 7936 conference phone working with asterisk using skinny.conf? |
| 14:11.27 | [TK]D-Fender | load chan_qwell.so ;) |
| 14:11.31 | jbalcomb | [TK]D-Fender: ok. Testing the connection houldn't take but two minutes. |
| 14:12.04 | [TK]D-Fender | jbalcomb : yup. hopefully we can synch up for a "group tour" of your layout. |
| 14:13.44 | dlynes_office | rvn: give qwell a demo phone, or pay him a few bucks to get it working; he's the developer of the chan_skinny.so module |
| 14:13.53 | trelane_ | is there a way to tell a digium te110p to not be the CPE end of the T!? |
| 14:13.55 | trelane_ | T1 even |
| 14:14.08 | jbalcomb | [TK]D-Fender: ok. i have soccer from 6 - 9, perhaps after? |
| 14:14.22 | dlynes_office | trelane_: I'm sure that's configurable in zaptel.conf is it not? |
| 14:14.33 | jbalcomb | [TK]D-Fender: unless we can do it before 5 |
| 14:15.00 | trelane_ | dlynes_office, I havn't seen an option for it as yet |
| 14:15.50 | rvn | dlynes_office: forgive my ignorance, but demo phone???? |
| 14:16.15 | dlynes_office | rvn: courier him a phone, temporarily, or donate one to him so he can use it for testing |
| 14:16.48 | dlynes_office | rvn: cisco phones are damned expensive; he doesn't have a whack of cash stored up to make sure every cisco phone works with chan_skinny.so |
| 14:17.06 | *** part/#asterisk mivck (i=1000@200.114.70.228) |
| 14:17.19 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 14:18.51 | jbalcomb | trelane: yeah, pri_cpe and pri_net |
| 14:19.05 | jbalcomb | trelane: for signalling on that zap channel |
| 14:19.06 | *** join/#asterisk Arnar (n=abi2@landi.oddi.is) |
| 14:19.13 | dlynes_office | trelane_: there ya go....jbalcomb's got the skinny on it |
| 14:19.25 | jbalcomb | woot |
| 14:19.38 | trelane_ | jbalcomb, thanks much :) |
| 14:20.40 | dlynes_office | trelane_: oh yeah...actually...it's in zapata.conf, not zaptel.conf |
| 14:20.51 | [TK]D-Fender | jbalcomb : before 5 |
| 14:21.17 | jbalcomb | sheesh, the watchguard mobile user VPN software REQUIRES the Zone Alarm firewall software... |
| 14:21.46 | dlynes_office | jbalcomb: dood...the zone alarm firewall software is like the totally best firewall software i've ever seen, bar none!!! |
| 14:22.25 | rvn | dlynes_office: i get you now |
| 14:22.52 | jbalcomb | dlynes_office: yeah but do i care to have firewall software on my pc when i spent $3,000.00 USD to have one for my whole network? |
| 14:23.03 | dlynes_office | jbalcomb: no kidding |
| 14:23.12 | *** join/#asterisk signuts (n=signuts@sig.triton.net) |
| 14:23.18 | dlynes_office | jbalcomb: or when you've got better firewall software such as iptables and netfilter? |
| 14:23.39 | dlynes_office | jbalcomb: i was being sarcastic...I absolutely hate zone alarm |
| 14:23.46 | signuts | Quick question, Hope easy answer, how do I set the CDR UserField from an AGI? Is it a enviornment variable? Or should I just $agi->exec("SetCDRUserField ..") |
| 14:23.59 | jbalcomb | dlynes_office: ;) true true. I'm working on a new project to set up Xen on my laptop |
| 14:24.10 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
| 14:24.28 | jbalcomb | dlynes_office: i want one XenU to be my firewall and bridge my net connection through for another XenU that acts as my Desktop |
| 14:24.39 | dlynes_office | funky |
| 14:24.42 | dlynes_office | whatever xenu is :) |
| 14:24.57 | jbalcomb | dlynes_office: dude, you don't know about Xen??!?! |
| 14:25.06 | dlynes_office | Yeah, i know wtf xen is |
| 14:25.09 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:25.10 | dlynes_office | just don't know wtf xenu is :) |
| 14:25.31 | jbalcomb | dlynes_office: Xen0 is the main Xen server and XenUs are the babies |
| 14:25.41 | dlynes_office | ah |
| 14:25.58 | jbalcomb | We use Xen quite a bit at my company |
| 14:26.00 | dlynes_office | yeah...i don't actually use vm's |
| 14:26.35 | jbalcomb | dlynes_office: we have these jerk ass filemaker servers. they can only handles 120 DB files a piece so we have to run 7 different servers to handle all our files. |
| 14:26.44 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67B3F.pool.t-online.hu) |
| 14:26.52 | dlynes_office | DB? |
| 14:27.02 | jbalcomb | dlynes_office: so we made a Xen server and put all 7 filemaker servers on the same physical machine. |
| 14:27.07 | jbalcomb | DataBase |
| 14:27.12 | dlynes_office | ah |
| 14:27.15 | dlynes_office | that's pretty lame |
| 14:27.31 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 14:27.41 | dlynes_office | Isn't filemaker some really old application, too? |
| 14:27.52 | jbalcomb | yeah, quite. we have three other servers that just run filemaker client so /hold/ the files open so the web server can access them. |
| 14:28.19 | jbalcomb | dlynes_office: its still in development but we are 3 major versions behind so kinda yeah. |
| 14:28.24 | *** join/#asterisk volp (n=volp@201.210.81.254) |
| 14:28.46 | jbalcomb | dlynes_office: its for one of our clients and they don't want to shell out the cash to have us move them to MySQL |
| 14:28.56 | dlynes_office | I wouldn't either |
| 14:29.04 | dlynes_office | I'd much rather use postgresql :) |
| 14:29.06 | mut | <3 our new tri-DS3 11ghz radios |
| 14:29.14 | jbalcomb | dlynes_office: i don't know who the moron is that started using filemaker to begin with but it happened... ;) |
| 14:29.37 | *** part/#asterisk Arnar (n=abi2@landi.oddi.is) |
| 14:29.54 | jbalcomb | dlynes_office: postgres is a bit slower than MySQL. Either is pretty damn good though and free-ish to boot. |
| 14:29.55 | dlynes_office | jbalcomb: probably a mac nut...I'm pretty sure filemaker originated on the mac |
| 14:30.10 | *** join/#asterisk Pj_ (n=pj@fernande.happycoders.org) |
| 14:30.23 | jbalcomb | dlynes_office: i think so too |
| 14:30.23 | dlynes_office | postgres is slower than mysql? Maybe if you're not doing any joins |
| 14:30.38 | dlynes_office | mysql is damned slow at doing joins |
| 14:30.56 | jbalcomb | dlynes_office: i dunno the low level stuff but the story i heard time and again was that postgres was more reliable but mysql is faster |
| 14:31.14 | *** part/#asterisk volp (n=volp@201.210.81.254) |
| 14:31.29 | jbalcomb | dlynes_office: i have heard the developers talking about not doing joins so much.. |
| 14:31.36 | dlynes_office | jbalcomb: yeah...mysql's faster only if you do all your logic in your interface language |
| 14:32.10 | dlynes_office | jbalcomb: if you place that logic in the database query language and/or stored procedures, mysql comes far behind postgresql |
| 14:32.30 | jbalcomb | dlynes_office gotcha. |
| 14:32.39 | dlynes_office | jbalcomb: ideally you don't want that code in a high level language...it's slower there than on the database |
| 14:32.59 | jbalcomb | dlynes_office: MySQL still seems more popular though, any reason for that? |
| 14:33.02 | dlynes_office | jbalcomb: mysql users generally return the entire result set, and then fiddle with the full set of data |
| 14:33.19 | dlynes_office | jbalcomb: it's simpler, and more people understand it |
| 14:33.31 | dlynes_office | jbalcomb: postgresql requires that you actually know how to use a database |
| 14:33.31 | Pj_ | jbalcomb: yes as you said it's faster for basic stuff, though totally not realiable, and there's no way to lock things properly |
| 14:33.32 | *** part/#asterisk parag_ast (n=root@dxb-b1751.alshamil.net.ae) |
| 14:34.26 | jbalcomb | Pj_ I heard something about improving locking by changing our tables formats? something about not using myisam I think? |
| 14:34.30 | Pj_ | which makes it unusuable for anything more than a news portal and I'm not even sure about that |
| 14:34.33 | *** join/#asterisk javar (n=javar@69.79.216.179) |
| 14:34.43 | jbalcomb | dlynes_office: Does Postgres have corporate support? |
| 14:35.04 | dlynes_office | jbalcomb: there's plenty of consulting companies that offer support for postgres, yes |
| 14:35.15 | jbalcomb | Pj_: =) perhaps but i know our DB files are 8 GB so I think we do a bit more than a new portal. |
| 14:35.24 | Pj_ | anyway this is #asterisk not #flamesql there are plenty of post Vs my articles on the net |
| 14:35.43 | Pj_ | jbalcomb: I can get you 8GB of news |
| 14:35.44 | Pj_ | :P |
| 14:36.01 | jbalcomb | dlynes_office: is there a main corporate sponsor? |
| 14:36.05 | hmmhesays | 8GB Boobies |
| 14:36.17 | Pj_ | hmmhesays: oh yeah baby |
| 14:36.38 | *** part/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 14:37.00 | jbalcomb | Pj_: umm.. no thanks. |
| 14:37.02 | [TK]D-Fender | jbalcomb : Pervasive Systems <- |
| 14:37.08 | dlynes_office | jbalcomb: check the postgresql web site...there's one or two main consulting companies (the two the contribute the most amount of code to the project) |
| 14:37.10 | hmmhesays | if only they would build a skype client for mipsel |
| 14:37.15 | dlynes_office | jbalcomb: yeah...what [TK]D-Fender said |
| 14:37.49 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 14:38.13 | jbalcomb | I want to start a company that only does enterprise level Linux support for apache, mysql, postgresql, sendmail, bind, and asterisk |
| 14:38.28 | jbalcomb | and maybe snort |
| 14:39.18 | dlynes_office | and you're looking for a company that specializes in postgres to do that? |
| 14:39.51 | jbalcomb | dlynes_office nah, just thinking outloud for the time being |
| 14:40.13 | dlynes_office | ah....anyways...just do a google search |
| 14:40.30 | dlynes_office | there's plenty of companies out there that have experience with most of those |
| 14:40.31 | jbalcomb | dlynes_office: i'm pretty sure i need four partners to make this work and about 100,000 in capital for the first year |
| 14:40.46 | dlynes_office | I, myself use all of the above |
| 14:40.55 | dlynes_office | I just try to avoid mysql whenever possible |
| 14:40.59 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:41.07 | *** join/#asterisk nighty_ (n=nighty@66-163-28-100.ip.tor.radiant.net) |
| 14:41.33 | nighty_ | would anyone know of a WIFI SIP Phone that can do PTT ? |
| 14:41.37 | dlynes_office | but when i'm testing new stuff, I usually use mysql because it's just easier to work with |
| 14:41.39 | mocker | jbalcomb: Hmm, do you think that other's who use that technology are likely to call for help? |
| 14:42.41 | jbalcomb | mocker: i should think so. we are getting ready to pay out 4,000 to have someone from mysql come out for 3 days. |
| 14:42.59 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 14:43.14 | SpaceBass | anyone got the new google talk working with *? |
| 14:44.33 | *** join/#asterisk n9urk (n=leonard@user-0ce2dhc.cable.mindspring.com) |
| 14:44.42 | n9urk | anyone running asterisk on tektonic? |
| 14:45.04 | *** join/#asterisk DrCool (n=DrKewl@202.125.113.10) |
| 14:45.35 | *** join/#asterisk s0lid (n=jlq@210.213.240.222) |
| 14:47.00 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 14:47.42 | X-Gen | google talk = jabber, SpaceBass search for jabber and * |
| 14:47.44 | X-Gen | :) |
| 14:49.00 | SpaceBass | thanks! |
| 14:49.44 | n9urk | Is it wise to use canreinvite=yes in 1.2.10? |
| 14:50.20 | n9urk | I set it to that and then started having problems with one way audio and such |
| 14:51.31 | *** join/#asterisk s0lid (n=jlq@210.213.240.222) |
| 14:51.39 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 14:51.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:52.53 | dlynes_office | bbiab... |
| 14:53.12 | Dovid | anyone know how to set up QOS for asterisk |
| 14:53.25 | Dovid | so that the asterisk box flags it on its end ? |
| 14:53.53 | Dovid | i looked at QOS on the wiki and it speaks about it but dosent mention how to use it |
| 14:54.05 | *** join/#asterisk aRJAy (n=aRJAy@218-214-130-112.people.net.au) |
| 14:54.11 | *** join/#asterisk Arnar_ (n=arnarb@landi.oddi.is) |
| 14:54.29 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 14:55.14 | harryvv | any comments on vitelity buying out sixtel or how good vitelity service is? seems thay are using 3coms backbone for its traffic. |
| 14:57.37 | Dovid | anyone on QOS |
| 14:57.59 | harryvv | explain |
| 14:58.14 | jbalcomb | Dovid: perhaps you could base your QoS on VLANs? |
| 14:58.46 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 14:58.51 | Dovid | jbalcomb: can u explain a little more |
| 14:58.59 | Dovid | do i have to set something up ont he linux box ? |
| 14:59.52 | jbalcomb | Dovid: your switches need to support VLANs, you configure a VLAN interface on your asterisk server, your setup the VLAN on your phone if it supports it |
| 14:59.58 | *** join/#asterisk brad6254 (n=brad6254@pool-72-72-199-176.altnpa.east.verizon.net) |
| 15:00.38 | jbalcomb | Dovid: configure a vlan interface on your router with routes from your normal network to and from the VLAN. all done |
| 15:00.40 | Dovid | the server is in a data center. will thsi still work ? |
| 15:01.06 | harryvv | jbalcomb is that to seperate his data network from the phone network? |
| 15:01.07 | Dovid | this* |
| 15:01.14 | jbalcomb | Dovid: do you mean that it is off premise? |
| 15:01.44 | Dovid | yes |
| 15:01.47 | jbalcomb | harryvv: not so much becase they will still run on the same switches but you can setup QoS based on VLAN ID |
| 15:02.03 | harryvv | I see |
| 15:02.15 | jbalcomb | Dovid: then no, this will certainly not work then |
| 15:02.16 | *** part/#asterisk Innatech (n=nospam@68-171-35-111.vnnyca.adelphia.net) |
| 15:02.33 | Dovid | so the setting on the phone for Vlan is only if the server si local ? |
| 15:02.35 | jbalcomb | Dovid: QoS via ToS if probably your best bet |
| 15:02.37 | Dovid | is local* |
| 15:02.41 | brad6254 | I'm having trouble dialing sip to zap using a grandstream ht-386. If i set it to inband, i can't get call parking to work, if i set it to rfc2833, i can't dial 9, wait for dial tone, then dial number. Do i need some other setting? |
| 15:02.46 | Dovid | ok. what us Qos via TOS ? |
| 15:02.50 | harryvv | I need a good switch that will handle poe and qos I guess the cisco series will work but are there others. |
| 15:03.21 | harryvv | brad, you can change the dialplan where you wont need to dial 9 |
| 15:03.23 | Dovid | harryvv: Yes a lot of good ones out there but expensive |
| 15:03.29 | jbalcomb | Dovid: its only for if you use VLANs and the way to use VLANs off premise is if you and a point-to-point connection |
| 15:04.00 | Dovid | ah so if i use the internet i am screwed ? |
| 15:04.02 | jbalcomb | harryvv: Dell PowerConnect is mostly what I use |
| 15:04.06 | brad6254 | harryvv: it seems the line doesn't recongnize the tones that are dialed. |
| 15:04.08 | *** join/#asterisk xbmodder_lappy (i=nobody@atarack/staff/xbmodder) |
| 15:04.09 | Dovid | is there any other solutions ? |
| 15:04.53 | jbalcomb | Dovid: probably the most important point for QoS is at your router |
| 15:04.59 | Dovid | ok |
| 15:05.03 | harryvv | dovid, well cost is a issue. I need to make a more professional voip demo unit instead of lugging everything in cardboard boxes. Was thinking of a aluminum frame with black plastic sides and wheels so it can be toted around. |
| 15:05.04 | Dovid | so if i get a good route |
| 15:05.19 | *** part/#asterisk brad6254 (n=brad6254@pool-72-72-199-176.altnpa.east.verizon.net) |
| 15:05.22 | jbalcomb | Dovid: you should be able to give SIP and RTP a higher priority and/or reservce a segment of bandwidth for them |
| 15:05.25 | harryvv | dell mmm |
| 15:05.43 | Dovid | jbalcomb: wht routers dot hat ? |
| 15:05.46 | Dovid | do that* |
| 15:05.59 | SpaceBass | X-Gen, thanks for the tip on searching for jabber....still not finding anything that suggests its clear cut...IE lots of folks working on it, but no solutions |
| 15:06.04 | jbalcomb | Dovid: anything non-residential mostly will |
| 15:06.22 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 15:06.23 | Dovid | ah. he wants to go the non residential route cause he has phones all over |
| 15:06.48 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.110.Dial1.SanJose1.Level3.net) |
| 15:07.13 | Dovid | jbalcomb: what do u say about this ? http://www.grack.com/news/2004/PrimusUpdates+WRT54GS.html and this ? http://www.voip-info.org/wiki/view/Linksys+WRT54G |
| 15:07.29 | jbalcomb | harryvv: have you considered a musicians mobile rack? they are generally the same format and footprint as standard IT racks |
| 15:07.30 | Dovid | harryvv: whay are you walkin around with boxes ? |
| 15:07.44 | harryvv | dovid, to demo my * system |
| 15:08.05 | harryvv | Would be nice to buy or build something a little more professional looking :) |
| 15:08.07 | jbalcomb | Dovid: WRT54GS is a residential wireless router |
| 15:08.31 | jbalcomb | harryvv: they probably make mobile IT racks as well |
| 15:08.33 | harryvv | jbalcomb ohh really never though of it. |
| 15:08.48 | harryvv | jbalcomb as long as its water proof |
| 15:08.50 | Dovid | harryvv: why not have the equipment in a date center and just bring a phon |
| 15:08.51 | jbalcomb | harryvv: the musicians racks look nice though |
| 15:09.07 | Dovid | jbalcomb: I know. do u think that will work ? |
| 15:09.17 | harryvv | dovid, customer firewall may not agree with my phone. |
| 15:09.17 | jbalcomb | harryvv: i'm sure they make them that way. removeable panels too. |
| 15:10.11 | jbalcomb | Dovid: i dont know what your doing, how much poower you need, or what your planning on doing in the future so i really couldn't say. |
| 15:10.15 | harryvv | heard a noise outside brb |
| 15:10.30 | jbalcomb | Dovid: i wouldn't consider it a professional solution. |
| 15:10.44 | Dovid | jbalcomb: server is in a data center, the max amount of phones per location is 2 |
| 15:11.37 | tzanger | musician's racks (1/4 rack) is nice and cna look really snazzy |
| 15:11.45 | Dovid | jbalcomb: i know it isnt profesional but its what he wants to spend. he dosent wana spend $300 - $400 a router for all his employees that are working out of home |
| 15:11.49 | tzanger | put hard rubber on the corners, nice set of caster wheels |
| 15:12.44 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 15:12.52 | Dovid | ? |
| 15:13.05 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 15:13.10 | nfi|ermes | hi all |
| 15:13.14 | tzanger | we have some really nice cases made by PMW case sales out of mississauga.. I think the case itself is a boxer case |
| 15:13.22 | nfi|ermes | how can i stop asterisk from a shell script ? |
| 15:13.27 | tzanger | rugged, pull-up handle, wheels.. very nice |
| 15:13.28 | tzanger | not cheap though |
| 15:13.36 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67C5F.pool.t-online.hu) |
| 15:13.39 | jero | hi |
| 15:13.41 | harryvv | do you have a web site? |
| 15:13.46 | tzanger | who |
| 15:13.58 | *** join/#asterisk Molotov (n=joe@unaffiliated/wiby) |
| 15:14.11 | harryvv | tzanger sorry do you have a site on that portable rack? |
| 15:14.22 | harryvv | or what ever it is :) |
| 15:14.59 | *** join/#asterisk Arnar_ (n=arnarb@landi.oddi.is) |
| 15:15.15 | tzanger | portable rack? any music store I'd imagine |
| 15:15.21 | SpaceBass | can anyone help me understand jabber functionality...basically is there a way to use googl talks' voice chat via asterisk? |
| 15:15.43 | tzanger | http://www.musiciansfriend.com/product/Gator-Deluxe-19-Inch-Rack-Case?sku=544792&src=3SOSWXXA |
| 15:15.46 | tzanger | gator makes EXCELLENT boxes |
| 15:16.06 | tzanger | snap off the ends and you have your connection points |
| 15:16.10 | [TK]D-Fender | tzanger : Most portable music rack cases don't have the necessary depth for computer enclosures, measure it off well.... |
| 15:16.34 | [TK]D-Fender | tzanger : ESP that one :) |
| 15:16.36 | nfi|ermes | how can i stop asterisk from a shell script or from a web link ? |
| 15:16.51 | harryvv | mmmm |
| 15:16.52 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 15:16.53 | [TK]D-Fender | nfi|ermes : "asterisk -rx "stop now" |
| 15:17.03 | nfi|ermes | thx [TK]D-Fender |
| 15:17.04 | MikeJ | asterisk -rx "stop now" |
| 15:17.10 | MikeJ | heh |
| 15:17.13 | *** part/#asterisk Arnar_ (n=arnarb@landi.oddi.is) |
| 15:17.46 | tzanger | [TK]D-Fender: don't get 1U full-depth servers |
| 15:17.51 | harryvv | tzanger I want it so I wont have to hookup all the ethernet connctions untangle the wires...everything is basicly ready to go once i come on site. Just plug in power and cable and go :) |
| 15:18.05 | tzanger | harryvv: see if you have the requisite depth |
| 15:18.07 | *** join/#asterisk IvyUK (n=mark@194.201.148.200) |
| 15:18.37 | harryvv | kinda like a min rack with wheels that wont get wet if I outside with it. |
| 15:18.44 | tzanger | 15" deep |
| 15:19.07 | harryvv | something like that. |
| 15:19.32 | tzanger | http://cgi.ebay.com/NEW-19-Mobile-Rack-Cisco-Router-Switch-CCNA-CCNP_W0QQitemZ200007984765QQihZ010QQcategoryZ64066QQcmdZViewItem |
| 15:19.39 | tzanger | something like that would work but is not really rugged |
| 15:20.23 | harryvv | yea going up stairs ramps bumping into things |
| 15:20.53 | tzanger | but yeah google for musician rack, mobile rack... nice rack... heh |
| 15:20.54 | harryvv | cute |
| 15:21.16 | harryvv | Or modify a portable lugage rack |
| 15:21.40 | tzanger | meh, I'm sure you can find a musician's rack with wheels and a sliding handle |
| 15:22.25 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 15:22.31 | tzanger | as soon as yo ustart modifying things you are going to put a lot of energy and time into making it look good, and likely could find something similar that would be cheaper if you factored your itme and materials into it |
| 15:22.34 | harryvv | a rack that tils back with extending handle |
| 15:22.35 | harryvv | ;) |
| 15:22.44 | harryvv | I know |
| 15:22.57 | tzanger | http://www.musiciansfriend.com/product/SKB-SKBRLX-RollX-Rack-Case-with-Wheels?sku=544590&src=3SOSWXXA |
| 15:23.00 | tzanger | perfect |
| 15:23.07 | harryvv | I am trying to get quotes to have a aux fuel tank made for my truck and its expensive. |
| 15:23.14 | Qwell | aren't music racks a little smaller? |
| 15:23.32 | Qwell | like, not as wide |
| 15:23.54 | harryvv | does not look big enough to hold at least one or two ip phones |
| 15:24.16 | tzanger | http://www.skbcases.com/product/pro_audio/rollx/skb-rlx-3.html |
| 15:24.19 | tzanger | there's the site |
| 15:24.22 | tzanger | that's a 3U |
| 15:24.30 | tzanger | 17" deep, 19" wide |
| 15:25.00 | tzanger | they go up to 6U with wheels and retractible handle |
| 15:25.26 | tzanger | 13.5" high |
| 15:25.39 | harryvv | that migh be just right |
| 15:25.48 | tzanger | put a 2U server in there, mount a tiny switch at the back, stuff a couple ip501s in there loose |
| 15:26.12 | tzanger | power bar and long (spring-coiled?) power cord and you're off to the races |
| 15:26.55 | file | email here, email there |
| 15:27.00 | harryvv | in fact, this would also be excellent to rent to a agency for a short time. Say rent it to a temp voting office with phones. |
| 15:27.23 | harryvv | temp office that has no phones i mean. |
| 15:27.24 | harryvv | ;) |
| 15:27.50 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 15:28.04 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:28.34 | [TK]D-Fender | tzanger : This is what I need now : http://www.musiciansfriend.com/product?sku=450341 |
| 15:28.40 | tzanger | yep |
| 15:28.54 | [TK]D-Fender | tzanger : Now only to find it reasonably priced shipped to me :) |
| 15:29.03 | tzanger | double keyboard stand? |
| 15:29.20 | tzanger | that looks like it should be relatively cheap to ship |
| 15:29.26 | tzanger | oversize (length) might kill you though |
| 15:29.30 | aRJAy | [TK]D-Fender: managed to fix my problem here.. Thanks for the redirect. |
| 15:29.49 | harryvv | aRJAy what was the problem |
| 15:30.32 | aRJAy | SPA3000... setting it up. Wasn't dropping down to gw0 (PSTN) |
| 15:30.58 | tzanger | [TK]D-Fender: ahh I see what you mean about the depth on that gator case.. oops, I thought it was deep too, but it's 12U and shallow. I figured it was 2U or 4U :-) |
| 15:30.59 | aRJAy | was putting the wrong string in the wrong place... if that makes sence :) |
| 15:33.38 | file | wrong string... wrong place... cool! |
| 15:33.49 | aRJAy | aye :) |
| 15:34.03 | *** join/#asterisk salviadud (n=ralfalfa@201.123.130.161) |
| 15:34.28 | aRJAy | not so cool though when you've banged your head so hard on the keyboard that you've created a new letter-ordering system!! |
| 15:35.11 | *** join/#asterisk javar (n=javar@69.79.216.179) |
| 15:36.07 | salviadud | aRJAy, come on! you didn't bang your head on the keyboard |
| 15:36.24 | salviadud | i'm wondering if anyone here uses a SafeType? |
| 15:36.49 | salviadud | i'm thinking about buying one, writing lines and lines of code, and chatting on irc will kill my hands eventually |
| 15:37.00 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 15:37.48 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:37.53 | thx2000 | Anyone know why my spa3102 would be trying to dial an IP address instead of the number im passing it? |
| 15:38.17 | *** join/#asterisk Ast001 (n=uros@212-200-196-125.adsl.sezampro.yu) |
| 15:38.25 | clyrrad | did you check your dialplan? |
| 15:38.37 | thx2000 | in the sipura or asterisk? |
| 15:38.39 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:38.40 | clyrrad | do you have the "Dial IP Address" box checked |
| 15:38.43 | clyrrad | in the sipura |
| 15:39.06 | salviadud | for you to dial an ip address on a sipura |
| 15:39.21 | salviadud | you need to dial * between the numbers |
| 15:39.32 | salviadud | like 200*33*122*0 |
| 15:39.51 | aRJAy | Why are people calling me QWERTY ? |
| 15:40.23 | harryvv | :) |
| 15:40.23 | thx2000 | well im trying not to dial one...lookin for the dial ip address option |
| 15:40.23 | file | qwerty... cutey... hrm |
| 15:40.24 | salviadud | does anyone do dvorak? |
| 15:40.24 | aRJAy | :) |
| 15:40.35 | Nugget | my girlfriend uses dvorak. |
| 15:40.41 | harryvv | I remember in the service the guys in the hanger called me pinger fresh out of tech school :) |
| 15:40.50 | Nugget | she says it's great. she never has to lock her workstation at the office -- nobody touches it. |
| 15:41.06 | Ast001 | Is it possible to install asterisk on point A and connect ordinary phone to it and then place and recieve calls from other ordinary phones which are not connected to Asterisk ? |
| 15:41.12 | salviadud | Nugget, your GF uses unix? |
| 15:41.16 | Nugget | yes |
| 15:41.34 | salviadud | well lucky you |
| 15:41.44 | Nugget | not really. when we met she used linux. :( |
| 15:41.47 | clyrrad | Nice! - been trying to convince the wife to do that for 2 years now :s |
| 15:41.49 | Nugget | but I fixed her |
| 15:41.56 | harryvv | btw, is there a device that when I plug it in will tell me if rtp/sip will pass though the customers firewall? |
| 15:41.58 | Nugget | now she uses a mac and afreebsd. |
| 15:41.59 | salviadud | linux, unix, whateva man |
| 15:42.08 | Nugget | s/af/f/ |
| 15:42.59 | salviadud | most girls i know that use mac, don't even know how it works, or how to open a damn terminal |
| 15:43.22 | salviadud | they're graphic designers... no clue about what goes on inside the box |
| 15:43.28 | Qwell | salviadud: you're supposed to wear it! |
| 15:43.33 | Qwell | like...on your face |
| 15:43.39 | Qwell | sheesh |
| 15:44.23 | salviadud | puta, no comprendo |
| 15:44.27 | xbmodder_lappy | Qwell, ! |
| 15:44.30 | salviadud | well, yeah, i agree |
| 15:44.53 | harryvv | I guess nmap is one free way to detect if a customer sip ports are open. but dont care to lug around a laptop for that. |
| 15:45.01 | *** join/#asterisk pdtwork (n=ptinsley@209.12.249.243) |
| 15:45.19 | pdtwork | anybody know of a little vpn router with PoE that is decent? |
| 15:45.24 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 15:45.43 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 15:46.15 | thx2000 | m0n0wall on a wrap |
| 15:46.55 | *** join/#asterisk CyberMad (n=cybermad@202.73.117.106) |
| 15:47.40 | clyrrad | Anyone know if its possible to force queues.conf to use extension based rules from extensions.conf to reach a member? I am trying to get call forwarding to work for queue members - but I need it to be done server side instead of phone side - has anyone done this before? |
| 15:48.09 | MikeJ | clyrrad, chan, local |
| 15:48.19 | [TK]D-Fender | clyrrad : You can use chan_agent driven agents to use your dialplan to determine who to call. |
| 15:48.40 | clyrrad | MikeJ - what do you mean chan local? |
| 15:48.49 | *** part/#asterisk Ast001 (n=uros@212-200-196-125.adsl.sezampro.yu) |
| 15:49.02 | clyrrad | TKD - you mean using agents.conf? Does that not mean the person has to log onto a queue? |
| 15:49.07 | harryvv | http://www.bandwidth.com/tools/voipTest# <-- test to see if sip port is blocked. Does also test for mgcp but not rtp or aix |
| 15:49.11 | MikeJ | you use chan_local. it's a proxy type channel.. lets a call go back into the dialplan |
| 15:49.39 | clyrrad | MikeJ - can you please PM me the syntax for that so i can research it - I have not seen this before |
| 15:49.44 | MikeJ | agent is a proxy channel as well prettty much |
| 15:49.58 | MikeJ | clyrrad, there will be stuff on the wiki about it. |
| 15:50.08 | MikeJ | LOCAL/ext@context |
| 15:50.16 | clyrrad | oh NICE :) |
| 15:50.19 | signuts | Any asterisk developers around? I am wondering why AGI (agi://localhost ) returns -1 and causes dialplan to stop execution. I thought jumping to priority n + 101 is deprecated but i need this support. Is there a way to check the return status of a AGI(...) command and proceed elsewhere in the dialplan upon failure? |
| 15:50.22 | clyrrad | and that can go in queues.conf? |
| 15:50.55 | MikeJ | that can go anywhere somthing like SIP/blah could go |
| 15:51.04 | *** part/#asterisk aRJAy (n=aRJAy@218-214-130-112.people.net.au) |
| 15:51.24 | clyrrad | MikeJ - I am going to try that in queues.conf - thanks for that information bud :) |
| 15:51.32 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 15:51.47 | MikeJ | np |
| 15:51.51 | MikeJ | ohyeah? |
| 15:51.52 | MikeJ | heh |
| 15:52.32 | hmmhesays | bah usb is pissing me off |
| 15:53.01 | mut | dewd usb is rawk |
| 15:53.02 | hmmhesays | if I plug any device into usb I should get *something* in dmesg right? |
| 15:53.09 | mut | i never realized how small usb flash mem got |
| 15:53.13 | hmmhesays | even if there is no driver for it? |
| 15:53.20 | mog | lsusb will show you whats connected |
| 15:53.23 | signuts | it's like 3 lines of code in res_agi.c to get my required behavior. Why would this not be implemented? |
| 15:53.23 | signuts | heh |
| 15:53.27 | mut | i got a 1 gig stick like 2mm thick, 3 in wide and 1 in long |
| 15:53.57 | *** join/#asterisk vlt (n=daniel@dslb-088-073-249-127.pools.arcor-ip.net) |
| 15:54.31 | mut | theres jumps now signuts |
| 15:55.30 | *** join/#asterisk nortex (n=nortex@64.136.65.144) |
| 15:56.16 | signuts | nut, is that only asterisk cvs? How does jumps work? I |
| 15:56.40 | signuts | nut, of course it's not in asterisk 1.2.10 |
| 15:57.29 | mut | well |
| 15:57.32 | mut | whats in your agi |
| 15:57.38 | mut | something is causing it to hangup in the agi.. |
| 15:58.08 | signuts | mut, I am using a socket agi and need to recover from connection refused messages. |
| 15:58.13 | signuts | agi://localhost |
| 15:58.14 | *** join/#asterisk fritz5150 (n=erik@208.15.8.26) |
| 15:58.35 | signuts | my agi is _intentionally_ not running |
| 15:58.54 | mut | you sure it's not running? |
| 15:59.00 | *** part/#asterisk fritz5150 (n=erik@208.15.8.26) |
| 15:59.06 | signuts | mut, 100% positive |
| 15:59.10 | Juggie | signuts, agi should return a connection refused. |
| 15:59.22 | signuts | Juggie, it does but it halts dialplan execution |
| 15:59.32 | Juggie | it shoudnt. |
| 15:59.35 | signuts | I want n + 101 to try a different method |
| 15:59.37 | Juggie | i've never seen that. |
| 15:59.44 | hmmhesays | i plug in a device and get nothing |
| 15:59.48 | hmmhesays | nothing at all |
| 15:59.48 | signuts | crazy.... pretty well defined behavior in the source code |
| 15:59.48 | Juggie | n+101 doesnt exist anymore unless yuo forcefully enable it |
| 16:00.55 | clyrrad | MikeJ - are you still there? |
| 16:01.52 | Juggie | signuts, only if you enable it. |
| 16:01.54 | Juggie | it was removed in 1.2 |
| 16:02.14 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:03.07 | *** join/#asterisk sponix (i=family@host-66-205-123-177.classicnet.net) |
| 16:03.30 | signuts | Juggie, well regardless n + 101 was my hack solution. that won't work anymore then. The main point is AGI(agi://localhost) returns -1 causing dialplan execution to halt upon a failure (connection refused) |
| 16:03.32 | *** part/#asterisk sponix (i=family@host-66-205-123-177.classicnet.net) |
| 16:03.52 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 16:04.08 | *** join/#asterisk cr0n (i=d@dsl-146-242-180.telkomadsl.co.za) |
| 16:04.10 | Juggie | signuts, are you sure |
| 16:04.52 | signuts | Juggie, are you? lemme show you my dialplan and its execution |
| 16:05.03 | Juggie | signuts, i'm pretty sure i've done that. |
| 16:05.11 | Juggie | pastebin your dialplan and the asterisk output |
| 16:05.18 | *** join/#asterisk sharp (n=sharp@c-68-45-160-72.hsd1.pa.comcast.net) |
| 16:05.19 | *** join/#asterisk folder (n=carl0s@compsup.demon.co.uk) |
| 16:05.22 | Juggie | www.pastebin.ca |
| 16:05.37 | harryvv | I know most of you are linux lovers but anyone here use windows and recomend system mechanic? |
| 16:06.00 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 16:06.07 | folder | harryvv: yes and no. |
| 16:06.12 | folder | respectively |
| 16:06.20 | harryvv | yes and no for ? |
| 16:06.23 | hmmhesays | lsusb should show something even if the driver is not found right? |
| 16:06.28 | folder | recommending 'system mechanic' |
| 16:06.42 | *** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) |
| 16:06.42 | *** mode/#asterisk [+o kram] by ChanServ |
| 16:06.46 | folder | yes I use windows, no I don't recommend system mechanic |
| 16:07.08 | harryvv | why did you have a bad experaince with it? |
| 16:07.42 | folder | no. just don't need it. do you really need yet more software to keep your computer working normally? |
| 16:07.55 | Juggie | signuts, waiting on that pastebin. |
| 16:08.51 | harryvv | folder, for some time my performance on my opteron system has been slow and yet to really pinpoint the problem. I have done all the required things like scandisk,defrag,check for errors, empty the cache, you name it. |
| 16:09.16 | signuts | Juggie, sorry. . |
| 16:09.18 | signuts | http://rafb.net/paste/results/lVlBmH24.html |
| 16:09.38 | signuts | Juggie, had to anonymize it :) |
| 16:09.45 | folder | I tend to find that the sort of people who use that kind of program, have windows XP machines clogged up with on average 5 or 6 of the following running: norton or mcafee antivirus, spamkiller, personal firewals, history cleaners, evidence scrubbers, cookie cleaners and on.. and on.. .(and ariston) |
| 16:10.58 | Juggie | signuts, you've modified res_agi.c? |
| 16:10.59 | *** join/#asterisk MamboKing (n=mambo@d38-23-38.commercial1.cgocable.net) |
| 16:11.01 | MamboKing | hey guys |
| 16:11.08 | signuts | Juggie, just one line.. |
| 16:11.15 | MamboKing | its been a while since I touch asterisk, did they get rid of zttool? |
| 16:11.19 | signuts | Juggie, it's only a printf on error |
| 16:11.22 | signuts | or wahteer |
| 16:11.23 | Juggie | no, make zttool |
| 16:11.32 | Juggie | in zaptel source. |
| 16:11.34 | MamboKing | i tried the compile fails :( |
| 16:11.45 | MamboKing | any alternative to it? |
| 16:11.45 | Juggie | your missing newt probally. |
| 16:11.53 | MamboKing | newt, kool i'll install that now |
| 16:12.00 | Juggie | be sure to install -devel |
| 16:12.09 | MamboKing | awesome, thanks man |
| 16:12.11 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 16:12.14 | Juggie | np |
| 16:13.34 | signuts | Juggie, you have an asterisk box to reproduce it? It should be simple. .I'm on 1.2.10 here |
| 16:13.39 | MamboKing | yup, that did it, thanks again |
| 16:14.14 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 16:14.27 | Juggie | signuts, trying to figure out if -1 causes a hangup. |
| 16:14.35 | Juggie | ok it does. |
| 16:14.36 | rene- | question: if i set a variable using the manager originate event, is this variable a global or a channel variable? |
| 16:14.37 | signuts | Juggie, I believe it does |
| 16:14.40 | Juggie | then i see the problem. |
| 16:14.40 | *** join/#asterisk oej (n=oej@65.197.203.67) |
| 16:15.57 | signuts | Returns -1 on hangup (except for DeadAGI) or if application requested hangup, or 0 on non-hangup exit. |
| 16:16.01 | Juggie | signuts, are you ok with editing some source? |
| 16:16.13 | signuts | Juggie, I have no problem with it |
| 16:16.21 | Juggie | go to line 166 |
| 16:16.25 | Juggie | or aronud there, in res_agi. |
| 16:16.35 | Juggie | res_agi.c within the res dir. |
| 16:16.37 | signuts | aye, there now |
| 16:17.48 | *** part/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) |
| 16:17.49 | signuts | But if doing what you say by changing that to return 0 won't do the trick. getbhostbyname isn't what is failed. |
| 16:17.59 | Juggie | yeah |
| 16:18.18 | Juggie | the proper fix might be to change them all. |
| 16:18.31 | Juggie | agi needs a return variable. |
| 16:19.00 | Juggie | change the one @ 210 |
| 16:19.04 | Juggie | and recompile, change it to 0. |
| 16:19.06 | Juggie | see if that helps. |
| 16:19.12 | Juggie | thats where its bailing on you. |
| 16:19.34 | signuts | my sugested change would be to add ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101); inside agi_exec(..) but I don't know what I need to return to make that work |
| 16:20.14 | harryvv | very cool voip test site |
| 16:20.17 | harryvv | http://myspeed.visualware.com/voip/ |
| 16:20.28 | *** join/#asterisk Seba_soy (n=s@64.76.126.29) |
| 16:20.32 | Seba_soy | hello all!! |
| 16:21.22 | Seba_soy | I am having a problem, I send a call to an asterisk with a zaptel card connected PSTN, when I hear ringing, I hear 2 rings one over other |
| 16:21.54 | Seba_soy | togheter, one is ring from pstn, Argentina, other is ring like USA |
| 16:22.07 | Juggie | signuts, * is moving away from +101 |
| 16:22.11 | Juggie | so now that isnt a solution |
| 16:22.21 | Juggie | the appropriate soludion would be a ${AGI_RESULT} variable |
| 16:22.25 | Juggie | which AGI() would fill |
| 16:22.28 | Juggie | with the reason it exited. |
| 16:22.33 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 16:22.59 | harryvv | anyone here sucsessfully use asterisk with skype |
| 16:23.21 | signuts | Juggie, aye. So use a GotoIf expression to check ${AGI_RESLT} |
| 16:23.26 | Nugget | I do, but you won't be enthused about how I do it. |
| 16:23.37 | EyeCue | skype isnt sip/iax right? |
| 16:23.42 | EyeCue | its its own protocol, so to speak? |
| 16:23.45 | Nugget | right, it's a closed protocol. |
| 16:23.55 | EyeCue | thought so |
| 16:23.57 | SpaceBass | its sip with a wraper from what I understand |
| 16:24.05 | SpaceBass | there is a gateway you can run |
| 16:24.20 | EyeCue | damn proprietary shit. |
| 16:24.24 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 16:24.26 | Nugget | *shrug* |
| 16:24.43 | Nugget | skype fills a void that asterisk and the open solutions haven't yet. |
| 16:24.52 | RoyK | <PROTECTED> |
| 16:25.00 | RoyK | i know |
| 16:25.11 | signuts | Juggie, the n + 101 behavior may be deprecated but it still functions. I just got it working |
| 16:25.16 | RoyK | i've had customers telling me that too |
| 16:25.29 | RoyK | signuts: even though, it's ugly |
| 16:25.44 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
| 16:25.44 | harryvv | Somone just told me skype will start charging 2 cents per call for skype to skype calls at the end of the year. |
| 16:25.54 | RoyK | sounds wonderful |
| 16:26.10 | RoyK | that way, people will focus more on open solutions |
| 16:26.15 | signuts | RoyK, I don't think so. it's uglier than any normal asterisk expression.. |
| 16:26.21 | signuts | ?? |
| 16:26.21 | Nugget | I really have a difficult time believing that. |
| 16:26.31 | Nugget | it doesn't make any sense at all |
| 16:26.39 | RoyK | what doesn't? |
| 16:26.47 | Nugget | that skype has plans to charge for internal calls. |
| 16:26.57 | RoyK | k |
| 16:27.04 | asterisk-dud | I have a tdm405p card for fxo ports and channel banks for fxs ports and asterisk keeps hanging up calls after about ten minutes when the come in for the fxo port and are routed to a fxs channel, can anyone help me? |
| 16:27.09 | RoyK | it might, though. time'll show :P |
| 16:27.20 | Nugget | I'm putting it in my "specious rumor" bucket. |
| 16:27.21 | signuts | Hopefully AEL saves asterisk dialplan garbage. How in the world can exten => XXX,101,HandlErrors() be uglier than exten => s,6,GotoIf($[ "${CALLERIDNAME}" : "Privacy Manager" ]?callerid-liar|s|1:s|7) |
| 16:27.59 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 16:27.59 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 16:28.02 | RoyK | signuts: i'd use agi :) |
| 16:28.11 | jbalcomb | [TK]D-Fender: http://integrics.com/products/enswitch/ |
| 16:28.20 | salviadud | skype will now kick the bucket |
| 16:28.29 | signuts | RoyK, AGI is the point of failure i'm trying to circumvent |
| 16:28.41 | salviadud | why charge for internal calls, when you can get them for free, with any other service |
| 16:28.49 | RoyK | signuts: i don't see the point of failure, really |
| 16:29.20 | [TK]D-Fender | jbalcomb : Ok... what about them? |
| 16:29.25 | RoyK | signuts: with huge amounts of calls, it may not be scalable, but we're running a rather large ITSP with AGI, and it works |
| 16:29.41 | signuts | if the daemon is down or it timesout doing a query inside agi things need to be able to recover |
| 16:30.15 | signuts | I to am running some fastAGI apps w/ much success. I just like covering my bases |
| 16:30.38 | *** join/#asterisk asterboy (n=root@S010600485480f4be.ed.shawcable.net) |
| 16:30.58 | RoyK | signuts: what daemon? |
| 16:30.58 | RoyK | I just AGI, not fastagi |
| 16:31.00 | signuts | I'd like having a IPVS setup or some load balancer. Scaling AGI isn't the problem, scaling asterisk probably is |
| 16:31.10 | signuts | RoyK, ahh. gotcha. Regular AGI |
| 16:31.17 | hmmhesays | bwhaha |
| 16:31.29 | asterboy | Putting together a quote and just want to verify that for a block of 25 DID #s I only need 1 T1 card priced at about $500? |
| 16:31.37 | eKo1 | I never use agi, I stick to c modules. |
| 16:31.39 | MrChimpy | it's easy to do fastagi, so don't do AGI |
| 16:31.57 | jbalcomb | [TK]D-Fender: nothing, potentially interesting for you. |
| 16:31.57 | MrChimpy | fastagi also doesn't have the odd sighups etc |
| 16:32.33 | harryvv | Free calling within the US and Canada. |
| 16:32.34 | harryvv | But remember, you can make free calls within the US and Canada to both landlines and mobile phones until the end of the year. |
| 16:32.37 | signuts | eKo1, good call, but development on those is harder and more time consuming. My perl daemon is incredibly smple |
| 16:32.42 | harryvv | http://www.skype.com/products/skypeout/ |
| 16:32.52 | MrChimpy | on a 2G Xeon 2x2.summinkGHz |
| 16:33.11 | eKo1 | signuts: yep, I had to sacrifice simplicity for scalability. |
| 16:33.37 | MrChimpy | i found I could start 240 threads of fastagi in about 0.2s using about 200meg, which is rather better |
| 16:33.50 | signuts | eKo1, understandable, i've written core components in C apps but I like prototyping out and seeing if sh!t works w/ scripting languages first. I'm at a time where I don't need to scale quite yet |
| 16:33.52 | harryvv | Any idea what voip engine skype uses? |
| 16:34.02 | MrChimpy | ...though still obscene compared to what C would do |
| 16:34.07 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 16:34.22 | asterboy | http://cgi.ebay.com/Like-New-Sangoma-A101-T-1-Card-for-Asterisk-VOIP-Server_W0QQitemZ110012128629QQihZ001QQcategoryZ51271QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 16:34.24 | eKo1 | Although my CPU is maxing out already :( |
| 16:34.28 | signuts | asterisk is pretty well written too the structures and functions all make pretty good sense |
| 16:34.35 | asterboy | Should be all I need for 1 channel bank. |
| 16:34.43 | MrChimpy | C for AGI apps is overkill IMHO - though I've spent years using C for CGI apps ;) |
| 16:34.45 | RoyK | signuts: scaling asterisk is indeed a problem... |
| 16:34.49 | asterboy | anyway, about $500 by the looks of it. |
| 16:34.54 | RoyK | I've just used perl for AGI |
| 16:35.08 | ringhals | I like php |
| 16:35.08 | RoyK | high memory footprint, but with a couple of gigs of RAM it works well |
| 16:35.14 | RoyK | that is, asterisk doesn't really use that much |
| 16:35.16 | Netgeeks | AGI's & Asterisk in comibination present some significant scaling issues |
| 16:35.17 | signuts | The bigest problem i've encountered with asterisk is load balancing across multiple servers and routing the calls to the correct server the UA is registered at |
| 16:35.36 | RoyK | Netgeeks: how many concurrent calls? calls per seconds? |
| 16:35.36 | Netgeeks | however, as you approach 400+ calls on a system, you begin to run into other problems |
| 16:35.38 | MrChimpy | not managed to load test properly yet, but looks promising |
| 16:35.56 | Netgeeks | RoyK: I really haven't ever pushed testing past 50 concurrent |
| 16:35.59 | RoyK | Netgeeks: asterisk itself scales horribly with 400+ concurrent calls with RTP bridging |
| 16:36.03 | Netgeeks | 50 calls per second I mean |
| 16:36.13 | RoyK | concurrent calls or call setups per sec? |
| 16:36.16 | eKo1 | Netgeeks: I've approached 130 simultaneous calls and I already have problems. |
| 16:36.20 | harryvv | signuts how many calls per min are going though those servers and how many channels do thay handle |
| 16:36.22 | MrChimpy | perl with standard agi simply won't work with more than 60 or so sessions |
| 16:36.28 | RoyK | eKo1: we've done 200 without problems so far |
| 16:36.36 | Netgeeks | I've pushed concurrent call testing to 1000+ (with alot of failures) |
| 16:36.48 | signuts | harryvv, two boxes in the lab w/ 4 T1 pri's I dont have any concrete numbers yet |
| 16:36.49 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:36.49 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:36.57 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 16:37.08 | eKo1 | Netgeeks: I've pushed about 300 calls in about 5 minutes with lots of FAILED and NO ANSWER calls. |
| 16:37.08 | Netgeeks | <-- on the phone - will rejoin thsi conversation in a sec |
| 16:37.15 | signuts | MrChimpy, I bet perl FastAGI would |
| 16:37.17 | harryvv | netgeeks, what cpu/ram did you use. |
| 16:37.42 | harryvv | ek01, did you use stress test software for that ? |
| 16:37.58 | eKo1 | No, this is on my production box :/ |
| 16:38.08 | ringhals | I have an issue with meetme rooms having extremely poor quality when I get more than about 5 iax remote clients in one room |
| 16:38.11 | MrChimpy | signuts: it probably does. as I said I started 240 clients ok of the same stuff written as fastAGI in 0.2s. Not actually tested it with asterisk itself under that sort of load yet. |
| 16:38.16 | harryvv | ek01, what would a opteron 244 with half a gig push for calls? |
| 16:38.28 | eKo1 | harryvv: no clue. |
| 16:38.30 | ringhals | with that said the cpu is under 1% util and load is at .02 |
| 16:38.31 | harryvv | k |
| 16:38.37 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 16:38.40 | ringhals | anyone have any suggestions |
| 16:38.50 | harryvv | eKo1 what disto are you using |
| 16:38.56 | Netgeeks | Harry: I've tested many different combinations, anywhere from a Dual Xeon 2.8G through a 12 Processor Sun system with 12G of memory |
| 16:39.11 | eKo1 | All I know is, I'm migrating * to a more powerful box with quad Intel Xeons and 2 GB of RAM. |
| 16:39.18 | eKo1 | harryvv: FC2 |
| 16:39.26 | Netgeeks | All my tests assumed that RTP traffic was carried as well as signalling traffic, and i only tested SIP to SIP or SIP to PRI |
| 16:39.32 | harryvv | I love fedora |
| 16:39.33 | harryvv | :) |
| 16:39.55 | ringhals | CentOS 4.3 :-) |
| 16:40.06 | eKo1 | ringhals: I'm using that as well. |
| 16:40.07 | harryvv | netgeeks which combo did you see that has the most calls with the least dropped or poor quality connections ;) |
| 16:40.11 | Netgeeks | RTP traffic begins to present a problem around 400 calls (800 legs of SIP), due to the packet per second interrupt rate |
| 16:40.19 | eKo1 | The quad CPU box is RHEL3 though. |
| 16:40.40 | *** join/#asterisk bartpbx (n=bartpbx@217.24.210.210) |
| 16:40.48 | eKo1 | Netgeeks: Is that a problem with * or the net interface? |
| 16:40.59 | Netgeeks | Even on the sun, I ran into the same issue, because a physical interrupt locks onto a single proc. You can move around which proc it locks on, but it still is locked to 1 |
| 16:41.03 | harryvv | netgeeks so are you saying to play it safe its best to go with a call rate of 350 per second? |
| 16:41.22 | Netgeeks | so then if you want to push more, you have to implement some kind of interrupt mitigation scheme |
| 16:41.37 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.157) |
| 16:41.49 | harryvv | I see |
| 16:41.55 | Netgeeks | Harry: yeah, unless I use the 400 number as tops for now |
| 16:42.01 | Netgeeks | 400 seems pretty reliable |
| 16:42.19 | Netgeeks | Thats on a dual 2.8G Xeon machine |
| 16:42.29 | harryvv | so what phone ratio have you seen that may work. I suspect call centers have a high call to phone ratio. |
| 16:42.46 | mut | how many raises should a person expect a year, a fairly high performance person at an IT job |
| 16:42.49 | Netgeeks | I'm going to have the chance to test a dual Intel Core Duo T2500 machine here shortly |
| 16:43.02 | harryvv | normal pstn network is 10 phones to one channel i think |
| 16:43.16 | ringhals | eKo1: most of my machines (distributed arch) are duel X 2.8 2 gig ram OR P4 2.8 1 gig |
| 16:43.51 | ringhals | mut: mine are bi anual |
| 16:44.16 | mut | salaried? |
| 16:44.18 | Netgeeks | For deployment in most situations I fall back on the old 10 to 1 ratio with average 3 minute call, and then modify that based on the clients guestimates or statistics and just tell them if things look different, be ready to spend some more money to bring the cluster up to requirements |
| 16:44.22 | *** part/#asterisk viLeR (i=1000@200.114.70.228) |
| 16:44.22 | ringhals | mut: yes |
| 16:44.25 | harryvv | so asuming you had to setup a 1,000 phone network as long as the call rate did not exceed 350 cps |
| 16:44.26 | hmmhesays | yyeaaahhaha |
| 16:44.40 | mut | what kinda raise ya get? |
| 16:44.49 | hmmhesays | my wrtsl54gs is now running asterisk with chan_oss working |
| 16:44.50 | mut | like size in comparison to what ya make |
| 16:44.53 | hmmhesays | and a usb sound card |
| 16:45.10 | ringhals | depends but on averarge between 3 and 5 percent every 6 months (over the last 3 years) |
| 16:45.14 | Netgeeks | 350 cps? what is the Siezure rate on those calls? |
| 16:45.29 | harryvv | Netgeeks so you use a cluster instead then a standalone server |
| 16:45.48 | tzanger | hmmhesays: what hardware for OSS on the wrt?! |
| 16:45.53 | tzanger | ahh usb sound card |
| 16:45.54 | Netgeeks | harry: yes, I use the cluster approach over the monolothic approach. |
| 16:46.03 | harryvv | intewresting |
| 16:46.15 | harryvv | beawolf cluster i guess |
| 16:46.23 | harryvv | or if thats the correct spelling. |
| 16:46.25 | harryvv | :) |
| 16:46.27 | Kernel_core | guys... today I translated AMP to FARSI ... but when I choose FARSI , I get "???????" in my browser instead of correct characters ....( I used in my .po file UTF-8 as charset ) what is wrong ? and how could I fix it ? |
| 16:46.28 | hmmhesays | tzanger yessah |
| 16:46.48 | Netgeeks | Oh, sorry, no, let me rephrase, I use a group of asterisk servers with extra scripts to provide a cluster like environment |
| 16:46.59 | Netgeeks | I don't use OS level clustering |
| 16:47.00 | salviadud | Kernel_core, duuuuuude, #freepbx |
| 16:47.02 | mut | so i guess theoretically i should be lookin forward to a $660/yr raise sometime this year |
| 16:47.03 | harryvv | thats interesting. |
| 16:47.14 | eKo1 | Netgeeks: I though you were using mosix or something. |
| 16:47.21 | Kernel_core | salviadud: nobody has knowledge about it there ! |
| 16:47.36 | Kernel_core | salviadud: I asked it many many times there , nobody answered |
| 16:47.51 | *** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.mn.comcast.net) |
| 16:48.05 | salviadud | Kernel_core, that's freakin' amazing... well, patience I guess, if they don't know, we don't know times 2 |
| 16:48.06 | Netgeeks | god no, I'd actually have to be pretty smart to run asterisk on a cluster platform! |
| 16:48.30 | harryvv | mut what kind of work do you do? im looking for a good designer or my self to make a good web page. Get my crap and services online :) |
| 16:48.38 | eKo1 | I like Mosix. Pretty easy to set up. |
| 16:49.18 | mut | couldn't design interfaces, just code them to do what ya want |
| 16:49.32 | ringhals | So with my meetme room issue. 5-10 remote iax clients using gsm codec attempt a conference call. the sound quality goes out the window almost instantly. However all 10 can be on concurent calls to a PRI int the same box and suffer 0 quality issues ... any help would be much appreciated |
| 16:49.40 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 16:49.40 | mut | i'de never sell my design work to others, tho i know people who do much worse that make money from it |
| 16:49.58 | tzanger | ringhals: what CPU |
| 16:50.07 | tzanger | hell what network card |
| 16:50.15 | tzanger | neither are used much with PRI meetme |
| 16:50.32 | ringhals | P4 2.8 no load on the server. Load is under .1 and CPU util is under 5% |
| 16:50.33 | jbalcomb | harryvv: if you are quite serious I may be able to help you with getting started |
| 16:50.45 | ringhals | not PRI meetme |
| 16:51.11 | ringhals | when placing standard calls there is no degredation of service.. but when they conference together is when it go to pot |
| 16:51.17 | Netgeeks | For my service platform I have a number of boxes I call media servers which run a modified asterisk and some otehr code related to load balancing, a true DB cluster for the backend, and a file system cluster for storage of voicemails/recordings, etc. |
| 16:51.40 | eKo1 | Netgeeks: Nice setup. |
| 16:51.47 | eKo1 | I just have on monolithic box running everything. |
| 16:51.49 | eKo1 | It sucks. |
| 16:51.53 | jbalcomb | Netgeeks: Are you familiar with this? http://integrics.com/products/enswitch/ |
| 16:52.16 | Netgeeks | We've got an installation going in right now that is designed to handle 300,000 accounts. |
| 16:52.30 | Netgeeks | jbalcomb: I'm aware of the product, I've never played with it |
| 16:52.46 | ringhals | tzanger I believe its a broadcom gig nic |
| 16:52.47 | jbalcomb | Netgeeks: You have a MySQL cluster? |
| 16:52.50 | mut | how in the world do you manage users |
| 16:53.14 | Netgeeks | jb: Yes, a shared memory cluster using SCI interfaces |
| 16:53.16 | Seba_soy | I am interestedo on that about a cluster using scripts. |
| 16:53.16 | tzanger | mut: flat files! |
| 16:53.35 | Seba_soy | Netgeeks how do you do that? |
| 16:53.45 | jbalcomb | Netgeeks: How does one go about setting that up? |
| 16:53.51 | mut | someone manually go in and say create this file |
| 16:53.56 | mut | and add it in the includes |
| 16:54.10 | foo | /x/ |
| 16:54.11 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 16:54.33 | Seba_soy | are you using some type of SIP REDIRECT? |
| 16:54.41 | Netgeeks | Seba: there is alot of documentation on that out there in the MySQL world. It's pretty neat, and has alot of gotchas. I'm not real familiar with it, as I have an expert on staff who is responsible for the file system cluster and the sql cluster |
| 16:55.01 | jbalcomb | Netgeeks: Ok, I see that it is covered in the MySQL documentation. Do you find it to be quite stable? Any big glitches to watch for? |
| 16:55.34 | jbalcomb | Netgeeks: What are you using for the file system cluster? |
| 16:55.39 | Seba_soy | Netgeeks: I will search more info |
| 16:55.45 | Netgeeks | jb: there are a shitload of gotchas... some really silly stuff that bites hard (like no field or table names can be over 32 characters, etc.) |
| 16:56.58 | eKo1 | that is one dumb gotcha |
| 16:56.59 | Netgeeks | jb: we looked at GFS (from RedHat) and Luster (Linux clUSTER) and chose Luster for it's redundancy, failover, and performance/growth benefits |
| 16:57.03 | eKo1 | typical mysql nonsense |
| 16:57.50 | ringhals | so anyone else have suggestions on my conferencing woes? |
| 16:58.06 | Netgeeks | jb: however once you get past all the gotchas, it seems to work great... it wasn't cheap to build, the share memory model requires all cluster nodes to have the memory to hold the entire DB in memory all the time, MySQL recommends 16+g per node... |
| 16:58.41 | Netgeeks | You can go with a non-shared memory cluster, but then you lose some reliability |
| 16:59.38 | jbalcomb | Netgeeks: ok, very good to know. both our current DB servers have 8GB so it's not too bad a jump. thank you very much for the notes. |
| 17:00.10 | Netgeeks | jb: no problem. |
| 17:01.01 | hmmhesays | not to try and compile zaptel on a wrt |
| 17:01.22 | Netgeeks | Seba: we are using both SIP 302 redirect and re-invite to do load balancing. |
| 17:05.00 | *** part/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 17:06.58 | ringhals | So with my meetme room issue. 5-10 remote iax clients using gsm codec attempt a conference call. the sound quality |
| 17:06.58 | ringhals | <PROTECTED> |
| 17:06.59 | ringhals | <PROTECTED> |
| 17:07.47 | *** part/#asterisk DrCool (n=DrKewl@202.125.113.10) |
| 17:08.41 | Juggie | cpu usuage? what kind of server? |
| 17:09.14 | ringhals | cpu is under 5% util and load is like .2 |
| 17:10.49 | *** join/#asterisk xheliox (n=gus@pdpc/supporter/active/xheliox) |
| 17:10.53 | xheliox | ~centosbug |
| 17:10.56 | jbot | centosbug is probably a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 17:11.07 | xheliox | ~centosbug |
| 17:11.08 | jbot | rumour has it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 17:11.11 | xheliox | Whoops |
| 17:11.14 | xheliox | sorry to do that twice :) |
| 17:14.29 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 17:15.11 | *** join/#asterisk cr0n (i=d@dsl-146-242-180.telkomadsl.co.za) |
| 17:15.13 | Juggie | signuts, are yuo still around? |
| 17:17.04 | Dr-Linux|work | question, my asterisk doesn't recognize CallerID for all incoming calls? what could be happend? |
| 17:17.13 | Dr-Linux|work | [TK]D-Fender, could you help? |
| 17:17.45 | hmmhesays | i can't seem to find a guide for configuring wcusb |
| 17:18.39 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:20.36 | Seba_soy | Dr-Linux|work from where are yuo accepting call |
| 17:20.44 | Seba_soy | SIP, ZAP, etc |
| 17:22.00 | Dr-Linux|work | Seba_soy, Zap |
| 17:22.37 | Seba_soy | and you know if callerid is coming? |
| 17:22.54 | Seba_soy | maybe it is blocked |
| 17:22.57 | Dr-Linux|work | it can get callid from most of callers, but for some callers it says something "warning" callerid returns with error on Zap/2 .." |
| 17:23.10 | Dr-Linux|work | Seba_soy, i see |
| 17:23.15 | Dr-Linux|work | Seba_soy, blocked where? |
| 17:23.21 | Seba_soy | from pbx |
| 17:23.49 | Dr-Linux|work | Seba_soy, you mean blocked from caller's provider comany? |
| 17:23.55 | Seba_soy | yes |
| 17:23.57 | Dr-Linux|work | or block on my lines? |
| 17:24.05 | bartpbx | hello |
| 17:24.05 | Seba_soy | from provider |
| 17:24.18 | Dr-Linux|work | Seba_soy, i see, it make sense |
| 17:24.32 | Dr-Linux|work | Seba_soy, so it's mean it's not my asterisk problem? |
| 17:24.39 | Seba_soy | ask you provider if he is blocking callerid |
| 17:24.53 | bartpbx | I'm looking for a asterisk Logo in good quality. I thought there was a logo download page somewhere on asterisk.org, but i cannot find it |
| 17:24.54 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 17:24.54 | Seba_soy | sure, if he is blocking then you will not receive it |
| 17:25.16 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 17:25.43 | Dr-Linux|work | Seba_soy, actually i'm prvoding outgoing calls via my asterisk on callerid basis |
| 17:26.01 | *** join/#asterisk alerios (n=alerios@201.244.242.109) |
| 17:26.26 | Dr-Linux|work | Seba_soy, i have 10 numbers that i've have added in DB |
| 17:26.40 | Dr-Linux|work | hhm.. |
| 17:26.45 | Seba_soy | maybe that is the problem, provider side problem |
| 17:27.02 | Dr-Linux|work | le me email them to enable callerID on their phone |
| 17:27.31 | Dr-Linux|work | Seba_soy, they are very old using it, so i never recieve callerids from their specific numbers .. |
| 17:27.39 | Dr-Linux|work | let me discuss with them |
| 17:31.09 | bartpbx | nobody has a logo for me? Or can point me to a logo? |
| 17:31.56 | tzanger | bartpbx: email digium |
| 17:32.01 | *** join/#asterisk tmartins (n=tmartins@200.192.160.100) |
| 17:32.56 | bartpbx | tzanger, thanks, I'll do this |
| 17:33.39 | hmmhesays | wildcard usb fxs anyone? |
| 17:35.39 | tzanger | nah |
| 17:35.48 | tzanger | I want an S518 with FXO integrated |
| 17:36.35 | *** join/#asterisk kss (i=kss@p54AEA102.dip0.t-ipconnect.de) |
| 17:37.21 | *** part/#asterisk kss (i=kss@p54AEA102.dip0.t-ipconnect.de) |
| 17:37.28 | *** join/#asterisk ComputerWarm (n=donc@209.29.156.97) |
| 17:37.29 | hmmhesays | do i even have to configure anything in zaptel.conf? |
| 17:38.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.233.171) |
| 17:38.12 | ComputerWarm | hello all question from with in a system() command how can i have ASterisk search a mysql database to check if a caller id is in there? |
| 17:38.36 | Qwell[] | ComputerWarm: Why not use the builtin mysql/odbc functionality? |
| 17:39.14 | ComputerWarm | that would be fine but i still need to figure out a way from in extensions.conf to check a caller id |
| 17:39.28 | Qwell[] | func_odbc |
| 17:39.38 | ComputerWarm | ok thanks i will look it up |
| 17:40.36 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 17:41.57 | angler | hmmhesays, what device are you using to use wcusb? |
| 17:42.42 | mountainm2k | does Asterisk Business Ediditon not support Realtime? |
| 17:42.45 | angler | hmmhesays, if it's a usb fxs device, you configure it in zaptel/zapata just like you would any other fxs/fxo device |
| 17:42.51 | hmmhesays | [root@fc50 ~]# ztcfg |
| 17:42.52 | hmmhesays | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 17:43.04 | hmmhesays | yeah but then I get that when I use fxoks=1 in zaptel.conf |
| 17:43.05 | philippel | ariel: where would you go to get a list of any new zapata.conf variables that I may want to mess with (other than looking throught he source)? and would that be in zaptel 1.2.7 or in libpri? |
| 17:43.09 | syzygyBSD | hmmhesays: did you modprobe your card first |
| 17:43.14 | ComputerWarm | you didn`t configure it correctly or you didn`t load the modules hmmhesays |
| 17:43.17 | syzygyBSD | try ztcfg -vvvvvvvvvv |
| 17:43.36 | hmmhesays | usbcore: registered new driver wcusb |
| 17:43.36 | hmmhesays | Wildcard USB FXS Interface driver registered |
| 17:43.36 | hmmhesays | Registered tone zone 0 (United States / North America) |
| 17:43.55 | *** part/#asterisk bartpbx (n=bartpbx@217.24.210.210) |
| 17:44.13 | tmartins | Anyone knows a better Debian package then the one from http://pkg-voip.buildserver.net / http://buildserver.net ? |
| 17:44.29 | hmmhesays | Module Size Used by |
| 17:44.29 | hmmhesays | wcusb 19328 0 |
| 17:44.29 | hmmhesays | zaptel 202244 1 wcusb |
| 17:44.33 | tmartins | or a channel asterisk debian specific ?! |
| 17:44.48 | bkw_ | hmmhesays, you doing USB on a WRT? |
| 17:45.03 | syzygyBSD | tmartins: why not just install from source or apt? |
| 17:45.17 | syzygyBSD | I install from apt, has everything I want |
| 17:45.38 | angler | mountainm2k, theres nothing really stopping you from using realtime with ABE |
| 17:45.47 | tmartins | syzygyBSD, bacause the last packages for asterisk 1.2.10 isn't stable |
| 17:46.03 | tmartins | the stable packages is an old version of asterisk |
| 17:46.08 | hmmhesays | bkw_ yeah |
| 17:46.15 | hmmhesays | but right now i'm just trying to get this to work in fc5 |
| 17:46.39 | mountainm2k | angler: It appears it does _not_ support res_mysql or cdr_mysql however |
| 17:46.51 | hmmhesays | bkw_ i got chan_oss running with a usb soundcard on my linksys here |
| 17:46.52 | mountainm2k | angler: but I did get cdr_odbc to work with MyODBC |
| 17:46.52 | angler | mountainm2k, odbc |
| 17:47.07 | bkw_ | ABE ships with cdr_odbc? |
| 17:47.13 | anthm | tsk tsk you should't have cancelled you cluecon reg you could have brougt it with |
| 17:47.14 | mountainm2k | yup |
| 17:47.26 | bkw_ | *FINGER* |
| 17:47.35 | mountainm2k | Eh? |
| 17:47.43 | bkw_ | I wrote that :P |
| 17:47.46 | mountainm2k | ;looks around, confused... |
| 17:47.51 | mountainm2k | O I C... |
| 17:47.54 | *** join/#asterisk cybergypsy (n=mark@APoitiers-157-1-65-64.w82-125.abo.wanadoo.fr) |
| 17:47.54 | hmmhesays | i wish i could make it to cluecon this year |
| 17:48.22 | mountainm2k | bkw_: Heh, did you write the ODBC for Realtime as well? Cause I can't get it to work.. :-P |
| 17:48.33 | bkw_ | anthm wrote that |
| 17:48.43 | angler | bkw_, :) |
| 17:48.55 | hmmhesays | so I have the driver loaded, but ztcfg errors |
| 17:49.24 | bkw_ | ABE is part of the reason I don't contribute to Asterisk any more... |
| 17:49.25 | anthm | hmmhe, then go |
| 17:49.36 | mountainm2k | <PROTECTED> |
| 17:49.49 | salviadud | what is ABE? |
| 17:49.53 | Qwell[] | ~abe |
| 17:49.56 | hmmhesays | anthm: cash flow problem |
| 17:50.03 | Qwell[] | jbot_: stupid bot |
| 17:50.09 | mountainm2k | bkw_: I'm somewhat disapointed that it's so far behind the open source version... |
| 17:50.18 | mountainm2k | <PROTECTED> |
| 17:50.29 | angler | mountainm2k, has to be due to testing and how fast Asterisk changes |
| 17:50.34 | mountainm2k | bkw_: I can see where you'd be ticked that they're selling your work, heh |
| 17:50.43 | salviadud | i still don't know what ABE is |
| 17:50.43 | bkw_ | well they had to strip out some stuff not considered stable yet |
| 17:50.47 | jarrod | anyone working with billing interfaces for cdr? |
| 17:50.53 | Qwell[] | salviadud: Asterisk Business Edition |
| 17:50.56 | mountainm2k | salviadud: ABE == Asterisk Business Edition |
| 17:51.08 | salviadud | o yeahhhh, i saw that at asterisk.org |
| 17:51.17 | salviadud | is it 1337 or somethin? |
| 17:51.29 | mountainm2k | For us, ABE seemed like a good idea -- tested, stable, supported |
| 17:51.32 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 17:51.45 | mountainm2k | I'll reserve judgement until I've actually used it for more than, say, an hour... :-P |
| 17:52.02 | Qwell[] | mountainm2k: You must be new here :p |
| 17:52.50 | hmmhesays | anyone see anything wrong with this? http://pastebin.ca/104155 |
| 17:52.59 | hmmhesays | beside the fact that ztcfg fails |
| 17:54.05 | *** join/#asterisk XARiUS (n=bdarcy@adsl-69-232-75-201.dsl.sndg02.pacbell.net) |
| 17:56.16 | XARiUS | anyone seen this before? |
| 17:56.23 | XARiUS | "channel.c:787 channel_find_locked: Avoided initial deadlock for...." |
| 17:59.41 | folder | XARiUS: Yes I've had that. Repeated up to about thirty times right after each other. It just went away though. |
| 18:00.23 | XARiUS | folder: yeah my customers had been reporting that their phones would just start ringing and not stop.. finally caught up with the issue |
| 18:00.35 | XARiUS | * console went nuts with ringing device.. etc.. and deadlock messages. |
| 18:00.58 | folder | hmm |
| 18:01.18 | XARiUS | of course initially I thought they were all on crack. |
| 18:01.22 | XARiUS | but I guess they had something there. |
| 18:01.23 | XARiUS | lol |
| 18:01.23 | folder | lol |
| 18:01.34 | lilalinux | is it still true, that MeetMe only works for Zaptel? |
| 18:01.51 | XARiUS | eh, nope? I use meetme quite a bit, no zap channels. |
| 18:02.00 | XARiUS | sip/iax, using kernel timer. |
| 18:02.03 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 18:02.15 | folder | apparently app-conference is the better choice, so I heard. |
| 18:02.34 | lilalinux | XARiUS, folder: thx will google for it |
| 18:02.34 | syzygyBSD | what do people here use as a tiff to pdf converter? |
| 18:03.11 | bkw_ | tiff2pdf |
| 18:03.43 | bkw_ | it comes with the libtiff stuff |
| 18:03.49 | syzygyBSD | k, thanks |
| 18:03.49 | lilalinux | syzygyBSD: tiff2pdf |
| 18:03.59 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:04.01 | mountainm2k | Qwell[]: New to ABE, perhapps, been using * for a month or two now... |
| 18:04.08 | syzygyBSD | sure enough, should have installed libtiff first |
| 18:04.19 | fgwaller | did anyone notice before that chan_alsa crashes asterisk in runtime if the soundcard ddi not detect a PCM device ;-) |
| 18:04.30 | folder | mountainm2k: is it any different to regular Asterisk? Other than the support of course? |
| 18:04.33 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 18:04.48 | mountainm2k | <PROTECTED> |
| 18:04.55 | syzygyBSD | of course, he paid for that... |
| 18:05.04 | folder | mountainm2k: posh front-end? |
| 18:05.30 | syzygyBSD | lol.. |
| 18:05.35 | mountainm2k | <PROTECTED> |
| 18:05.39 | mountainm2k | no front-end at all, heh |
| 18:05.44 | mountainm2k | under the covers, it's just asterisk |
| 18:05.45 | folder | right... |
| 18:06.03 | mountainm2k | and an older one at that |
| 18:06.23 | *** join/#asterisk innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 18:07.05 | *** join/#asterisk asteriskmonkey (n=phil@h216-235-8-130.host.egate.net) |
| 18:07.09 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 18:07.45 | sivana | anyone see this before? Zap/pseudo-687787378 s@longdistance_tdm:1 Rsrvd (None) |
| 18:07.53 | sivana | in "show channels" |
| 18:10.54 | *** join/#asterisk convey (n=kvirc@66.55.43.2) |
| 18:16.33 | jbalcomb | whats the command to send messages to other users on a linux box? |
| 18:16.46 | salviadud | write |
| 18:17.00 | salviadud | write user tty |
| 18:17.11 | salviadud | just type write and you'll get the syntax |
| 18:17.15 | salviadud | i forgot it |
| 18:18.11 | jbalcomb | so like `write root pts/0 "What's up?" |
| 18:18.34 | salviadud | kind of |
| 18:18.37 | salviadud | its more like |
| 18:18.43 | salviadud | write root |
| 18:18.45 | salviadud | then enter |
| 18:18.49 | salviadud | and then your message |
| 18:19.28 | jbalcomb | ah, ok |
| 18:20.10 | brad_mssw | hmm, iaxtel doesn't forward callerid information ? |
| 18:20.53 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 18:22.10 | asteriskmonkey | anyone knwo who provides cheapest trunks in canada that actually has working dtmf? |
| 18:22.25 | Zodiacal | anyone know the property i need to set so that the messsages button on my polycom phone dials into *97? |
| 18:22.35 | Zodiacal | right now it just dials the ext.. |
| 18:22.38 | Zodiacal | :/ |
| 18:23.58 | *** join/#asterisk bkidney (n=bkidney@nat01-128.stratos.ca) |
| 18:25.24 | hmmhesays | so anyone fix my wcusb problem yet? |
| 18:29.13 | cr0n | im getting a huge amount of echo but only from my side when talking to somebody over FXO, they cannot hear the echo and neither do they echo. its not my headset. any solutions or where could i start? ive read many tutorials and they dont really help. |
| 18:29.48 | tzanger | cr0n: your voice is bouncing off their hybrid. This can mean several things. |
| 18:30.18 | tzanger | First of all: What FXO device is this? Have you used fxotune? Have you dialed the telco's miliwatt number and adjusted your gains? |
| 18:30.22 | cr0n | tzanger: if i speak for long enough, theres no echo but a sudden "yes" or something and it echos |
| 18:31.09 | cr0n | FXO device: digium TDM400P |
| 18:31.34 | cr0n | miliwatt number? never heard of us having one of those |
| 18:31.34 | tzanger | cr0n: run fxotune, dial your telco's miliwatt number and adjust your gains. |
| 18:31.51 | tzanger | you need to corner a bell tech and ask them for the miliwatt and quiet term numbers |
| 18:32.11 | cr0n | i hope that even they would know |
| 18:32.11 | *** join/#asterisk froguz (i=froguz@200.54.67.53) |
| 18:32.31 | froguz | Hi ppl |
| 18:32.38 | Qwell[] | ~seen ppl |
| 18:32.50 | jbot | ppl <~ppl@CPE00e081260cf9-CM0011ae9233cc.cpe.net.cable.rogers.com> was last seen on IRC in channel #kde, 617d 11h 22m 37s ago, saying: 'I'm out to bed. Thanks aseigo.'. |
| 18:32.51 | tzanger | I'm telling you what you need to do; take the advice or don't, but that's what's needed. |
| 18:32.56 | tzanger | heh |
| 18:33.00 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 18:33.00 | Qwell[] | froguz: just missed him |
| 18:33.01 | tzanger | ~seen dead people |
| 18:33.05 | jbot | tzanger: i haven't seen 'dead people' |
| 18:33.40 | froguz | hahaa you're funny |
| 18:33.51 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:33.53 | tzanger | time for the classic |
| 18:34.03 | tzanger | ~seen my dick in three years, and god that's depressing |
| 18:34.11 | jbot | tzanger: i haven't seen 'my dick in three years, and god that's depressing' |
| 18:34.22 | froguz | LOL! |
| 18:35.29 | Corydon-w | Fat bot |
| 18:36.12 | froguz | ok, here we go... is there a way to send 'one by one' numbers to the zap channel? or even better, can i send block of numbers?? |
| 18:36.30 | syzygyBSD | froguz: senddtmf? |
| 18:37.01 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 18:37.34 | froguz | i'm sending 092885125 to a siemens hipath using E1 interface, but the hipath exoect to recieve 09 as the first blovk and then te rest of the number |
| 18:37.55 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 18:38.03 | eKo1 | the what now? |
| 18:38.19 | syzygyBSD | hmm.. I am guessing you just want Dial(Zap/1/09wwww2285125) |
| 18:38.22 | cr0n | tzanger: any guess to what these values should be? something along the lines of some most commonly used defaults? |
| 18:38.36 | *** join/#asterisk BugKham (i=CKGLOB@221.128.110.41) |
| 18:39.18 | *** join/#asterisk derekS (n=dereks@unaffiliated/dereks) |
| 18:39.22 | froguz | syzygyBSD, i've tried with w and ww... maybe i should play more with those pauses |
| 18:39.43 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 18:40.12 | derekS | hi. if i setup an asterisk server with no outside lines (so basically just allows internal calls)... is there an adapter that will allow me to use my cellphone to make calls? |
| 18:40.42 | syzygyBSD | derekS: look up gsm gateway |
| 18:40.58 | derekS | thanks |
| 18:41.18 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 18:42.12 | cr0n | syzygyBSD: without forwarding the call through a gsm gateway is there perhaps something that you can connect, an adapter of some sort as derekS asked? |
| 18:42.54 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt) |
| 18:42.58 | syzygyBSD | nope, not that I know of, if you are going to connect an adapter, why not just get a normal phone? |
| 18:43.01 | *** part/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt) |
| 18:43.20 | tzanger | cr0n: there aren't any, you need to tune it to your specific installation |
| 18:43.32 | cr0n | syzygyBSD: perhaps there are no landlines in the area.. or for whatever reasons |
| 18:43.38 | derekS | cr0n: what i wanted to do was forward through a gsm gateway :) |
| 18:43.40 | cr0n | tzanger: ill try |
| 18:43.49 | cr0n | derekS: ahh okay, i was wondering in anycase ;) |
| 18:44.03 | syzygyBSD | cr0n: wait.. what do you want to do? |
| 18:44.12 | derekS | ohh |
| 18:44.29 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 18:44.36 | cr0n | syzygyBSD: im just wondering if there is a way to connect a cellphone up so that if the <cell extenion> is dialed that it would rather go via the cellphone than the POTS? |
| 18:44.40 | syzygyBSD | what I can tell derekS wants to go: cellphone -> ?? -> asterisk |
| 18:44.54 | cr0n | syzygyBSD: for cost reasons of course. |
| 18:45.06 | syzygyBSD | cr0n: ya, that is backwards from what derekS wanted |
| 18:45.18 | derekS | :) |
| 18:45.19 | folder | There are some |
| 18:45.21 | derekS | thanks for oyur help |
| 18:45.22 | *** part/#asterisk derekS (n=dereks@unaffiliated/dereks) |
| 18:45.36 | folder | d'oh |
| 18:45.59 | syzygyBSD | cr0n: you want a gsm card, look up junghanns |
| 18:45.59 | folder | don't lots of people use those cellphone FXO port cradle thingies? |
| 18:46.16 | folder | I use a GSM <-> SIP gateway from Portech.com.tw |
| 18:46.36 | syzygyBSD | hmm, never heard of one of those |
| 18:46.39 | syzygyBSD | maybe |
| 18:46.46 | folder | having trouble with it though. It seems not to respond to the SIP INVITE requests sometimes. About 60% of calls come through (either in or out) |
| 18:47.12 | syzygyBSD | folder: you got reinvite=yes in sip.conf? |
| 18:47.26 | folder | syzygyBSD: it's like a single-SIM version of the 2N Voiceblue Lite. |
| 18:47.43 | syzygyBSD | k, nice |
| 18:47.49 | folder | syzygyBSD: No, I have specifically set reinvite=no, so that it doesn't try to do a direct link to the upstream sip provider. |
| 18:48.00 | folder | syzygyBSD: why would reinvite=yes help? |
| 18:48.06 | syzygyBSD | i dun know :) |
| 18:48.11 | folder | oh ok :) |
| 18:49.38 | folder | I dunno if pissing around with SIP Expire timers would help, but I set it to 24hrs and when I got in bed I called it, and it just rang.. and rang.. and rang. Then I tried again and it worked. |
| 18:50.11 | cr0n | syzygyBSD: problem with using a gateway over the net is bandwidth, so i guess something like a gsm card would be the way to go |
| 18:50.25 | cr0n | syzygyBSD: is this how leascostrouting is setup for mobile phones? |
| 18:50.56 | syzygyBSD | cr0n: k, just a question, can't you get more bandwidth? is a cellphone bill worth a increase in bandwidth bill? |
| 18:52.33 | folder | cr0n: this is exactly how GSM LCR is done. With traditional phone systems people use something like a Nokia Premicell. That's a cellphone module with an FXS (rj-11) port and its' used as a trunk by the phone system. This thing I have is the same but for SIP. There are also adapters which plug into your mobile phone and provide an RJ11, complete with dial-tone. You need to use one of those with an FXO card though. |
| 18:54.07 | *** part/#asterisk bkidney (n=bkidney@nat01-128.stratos.ca) |
| 18:54.13 | MamboKing | whats the name of the util for createing new mailboxes anyone know off the top of their head? |
| 18:54.58 | Nugget | there's a utility to creat mailboxes? |
| 18:55.02 | Bullseye_Network | MamboKing: In 1.2.x you dont have to create them anymore it will automatically create it just add it to voicemail.conf and call it once and it will do it |
| 18:55.30 | MamboKing | kool thanks |
| 18:55.34 | MamboKing | I figured as much |
| 18:56.14 | file | dan42: I just fixed bug 7552 |
| 18:56.26 | jbalcomb | [TK]D-Fender: you ready to check it out? |
| 18:57.11 | cr0n | syzygyBSD: more bandwidth is very expensive here, plus, since im very new to this im just viewing my options |
| 19:00.36 | *** join/#asterisk svenna_ (n=svenna@p548D0452.dip0.t-ipconnect.de) |
| 19:00.39 | MamboKing | anyone know if version 2 of the firefly softphone can be configured to point to your asterisk server? |
| 19:00.51 | MamboKing | it looks like the value is hard coded in |
| 19:01.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:01.35 | SpaceBass | ok....heres a question |
| 19:01.50 | SpaceBass | my company is moving to cisco phones, which includes the ability to use a softphone on my pc |
| 19:02.01 | SpaceBass | of course that only works when connected to the VPN and through a headset |
| 19:02.17 | SpaceBass | since my * box is not on the corporate VPN, is there anything I can do? |
| 19:02.27 | SpaceBass | Can I bridge that VPN connection though my laptop to my * Box? |
| 19:02.33 | denon | have them expose the pbx without vpn |
| 19:02.35 | Zodiacal | anyone know what i set the msg.mwi.1.callBack="" to on my polycom 601 so that the messages button will dial the users voicemail? |
| 19:02.35 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:02.37 | Zodiacal | *97? |
| 19:03.37 | SpaceBass | denon, I wish, but NOTHING happens outside of our VPN |
| 19:04.00 | *** join/#asterisk mitemous (n=sp@c-68-52-141-197.hsd1.tn.comcast.net) |
| 19:04.04 | SpaceBass | I'm convinced that we have no SMTP server...that someone actually prints out the mail outside the VPN and manually re-types it inside the corprate network |
| 19:04.14 | denon | hah |
| 19:05.15 | SpaceBass | I have a 7960 cisco on my desk...I'm not giving that up for a soft phone on a laptop....even for a local extension (I'm a remote employee) |
| 19:05.53 | ringhals | quit |
| 19:05.54 | ringhals | exit |
| 19:07.12 | svenna_ | does someone know, how ti get "Recording Calls With Asterisk" 2 work? voip-info.org says: "Asterisk 1.2 now comes with the new "automon" ... permits a user to turn on/off call recording ..." i enabled it in features.conf, but when i dial *1 nothing happens - so what have I forgotten? :-) |
| 19:07.38 | Qwell[] | svenna_: w or W to Dial() |
| 19:07.46 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 19:07.55 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 19:08.17 | SpaceBass | back....dropped my connection |
| 19:08.35 | svenna_ | jepp, i did that |
| 19:08.42 | *** join/#asterisk bofh42 (n=bofh42@p5482A735.dip0.t-ipconnect.de) |
| 19:08.42 | SpaceBass | so if that cisco VPN connection is just a network interface, and i bridge it to one of the two NICs on the laptop? |
| 19:08.44 | *** join/#asterisk Frogdude (n=FroggerD@c-24-16-72-159.hsd1.wa.comcast.net) |
| 19:08.56 | svenna_ | or, i did both of them - is this wrong?... |
| 19:09.16 | Qwell[] | svenna_: no, it's okay, if you want either side to be able to record the call |
| 19:09.29 | Qwell[] | svenna_: the most common reason I've seen that automon doesn't work... |
| 19:09.44 | Qwell[] | is that you aren't dialing *1 fast enough. You have literally half a second between the digits |
| 19:10.09 | svenna_ | ok, i give it a FAST try :-) |
| 19:10.12 | Qwell[] | You can change the timeout in features.conf, or just buy faster fingers :) |
| 19:10.56 | mitemous | anyone know what a telephone INVITE normally looks like in SIP? |
| 19:11.16 | mitemous | ie.. 12125551234@1.2.3.4 |
| 19:11.23 | eKo1 | It looks like a bunch of characters |
| 19:11.24 | mitemous | or maybe.. +12125551234@1.2.3.4 |
| 19:11.41 | *** join/#asterisk Frogdude (n=FroggerD@c-24-16-72-159.hsd1.wa.comcast.net) |
| 19:12.49 | file | or if you're crazy, sip:800551212;npdi=yes;phone-context=potato@127.0.0.1;user=phone |
| 19:13.11 | Frogdude | hi guys |
| 19:13.16 | file | hello |
| 19:13.17 | Frogdude | can someone please tell me what this message means? |
| 19:13.20 | Frogdude | Jul 28 12:40:07 WARNING[4638]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'outgoing' |
| 19:13.41 | file | Frogdude: something timed out, and normally it goes to the 't' extension... but there wasn't one, so it bailed out |
| 19:13.42 | Frogdude | I'm just trying to get asterisk working with voicepulse connect and x-lite |
| 19:13.56 | Frogdude | ahhh |
| 19:14.01 | mitemous | file: is that what a cisco phone or something else would use by default.. sip:2125551234 |
| 19:14.05 | cr0n | folder: thanks for the info on the GSM |
| 19:14.07 | file | pastebin your dialplan logic for the context outgoing and all the messages you see on the Asterisk console |
| 19:14.14 | Frogdude | file: thanks |
| 19:14.19 | file | mitemous: they would just do plain |
| 19:14.38 | Frogdude | something like a flakey internet connection? |
| 19:14.39 | Frogdude | :) |
| 19:16.41 | svenna_ | @Qwell : lol! i typed it in very fast and it worked :-) |
| 19:16.51 | svenna_ | thx ! |
| 19:17.20 | svenna_ | i guess i cant affort such fast fingers and will change that value :) |
| 19:18.41 | jarrod | are their dialing rules for intl country codes |
| 19:19.46 | file | jarrod: what do you mean? |
| 19:20.23 | jarrod | well, intl country codes are of variable length |
| 19:20.33 | jarrod | and i want to pull the country code from the cdr |
| 19:20.41 | *** join/#asterisk dacleric (n=dacleric@p54821534.dip0.t-ipconnect.de) |
| 19:20.46 | jarrod | but its not like an area code in the states where i can get the 3 digits |
| 19:20.57 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 19:20.58 | jarrod | it seems to be either 2, or 3 |
| 19:21.00 | file | ah |
| 19:21.02 | jarrod | i was wondering how to determine |
| 19:21.06 | hmmhesays | bah this is pissing me off |
| 19:21.43 | file | jarrod: I vaguely remember coming across something, lemme look |
| 19:21.48 | jarrod | thanks man |
| 19:21.48 | [TK]D-Fender | jbalcomb : PM |
| 19:23.10 | file | jarrod: can't find it :\ but I believe it was a C function that was able to take in a phone number and get the country code out of it |
| 19:23.31 | file | whether it was accurate or not, no clue |
| 19:23.55 | eKo1 | cc are 1 to 3 digits long |
| 19:24.36 | eKo1 | they all start with a digit in the range 1-9 |
| 19:25.30 | jarrod | im not worried about what the start with |
| 19:25.31 | jarrod | thats easy |
| 19:25.39 | jarrod | but determining how long they are is what is necessary |
| 19:26.02 | eKo1 | look here: http://en.wikipedia.org/wiki/List_of_country_calling_codes |
| 19:26.16 | eKo1 | Your going to need a function that compares and matches |
| 19:26.21 | eKo1 | so make one up |
| 19:26.37 | *** join/#asterisk stopher (n=business@cm-24-121-73-66.kingman.az.npgco.com) |
| 19:27.11 | mitemous | screw this..time to whip out a packet sniffer |
| 19:27.16 | stopher | Got a quick question, Can anything other than FXO/FXS work on a asterisk pbx, or do you NEED to have those? |
| 19:27.26 | Seba_soy | to get Country code you have to have stored all country codes on a database and then make a comparision |
| 19:27.28 | stopher | such as a modem /internal/external |
| 19:27.55 | mitemous | stopher: what would you use the modem for |
| 19:28.11 | mitemous | stopher: you cant terminate calls with a modem, if that's what you are asking |
| 19:28.19 | stopher | yeah thats what im asking |
| 19:28.26 | *** join/#asterisk ComputerWarm (n=donc@209.29.156.12) |
| 19:28.30 | eKo1 | a modem is not made for that |
| 19:28.31 | Seba_soy | you can make a C app and save al Country Codes on a matrix, and then make some function to find there... |
| 19:29.05 | eKo1 | you need an fxo/fxs card |
| 19:29.05 | ComputerWarm | could anyone take alook at this please |
| 19:29.05 | ComputerWarm | http://pastebin.ca/104255 |
| 19:29.05 | mitemous | stopher: if you havent already looked, just check out some of the internet voip companies |
| 19:29.05 | mitemous | you can connect asterisk to them for next to nothing |
| 19:29.06 | ComputerWarm | its a call back script i am working on. under the question i posted what i am getting from the cli |
| 19:29.25 | stopher | mitemous: I don't really want to have VoIP but i want to use the asterisk as a back-end to a REAL pbx |
| 19:29.55 | stopher | or KSU |
| 19:29.56 | Seba_soy | then do you need some digium cards? |
| 19:29.57 | eKo1 | * is a REAL pbx |
| 19:30.06 | mitemous | stopher: yeah, as far as i know, you need an interface card or you have to use a voip provider |
| 19:30.15 | stopher | lol * is a computer alternative eKo1 |
| 19:30.25 | stopher | okay, mitemous. thanks. |
| 19:30.25 | ComputerWarm | any phpagi scripters here? |
| 19:30.31 | Seba_soy | do you wanna interconnect actual pbx with a new * based pbx? |
| 19:30.47 | eKo1 | stopher: no, asterisk is a BETTER alternative |
| 19:30.51 | stopher | Seba_soy: that's my plan |
| 19:31.09 | Seba_soy | I did that some times |
| 19:31.12 | stopher | eKo1: For a smaller scale maybe, but for more than 25 phones, its a bit extreme don't you think? |
| 19:31.12 | Seba_soy | but is really bad |
| 19:31.32 | stopher | Seba_soy: really bad? how so |
| 19:31.56 | eKo1 | stopher: guess how many phones I run on one * box? |
| 19:32.09 | stopher | eKo1: I have no clue. |
| 19:32.10 | shido6 | 2000? |
| 19:32.16 | Seba_soy | it is difficult to configure |
| 19:32.22 | CunningPike | stopher: Guess how many we have at a local government site? |
| 19:32.23 | Seba_soy | and is not better quality |
| 19:32.31 | TommyTheKid | I honestly wouldn;t want to deploy even a 5,000 seat asterisk PBX, but I wouldn't want to have anything to do with any 5,000 user voice deployment :) |
| 19:32.35 | eKo1 | 80 - 90 |
| 19:33.07 | mitemous | eko1: did you deploy QoS in the network as well? (i'm assuming yes) |
| 19:33.10 | TommyTheKid | I was asked if I would want to deploy * company wide.. ~35K (at least till Aug 3) |
| 19:33.13 | stopher | eKo1: You have to have some strong networking then, background music alone robs bandwidth on IP phones |
| 19:33.13 | Seba_soy | stopher: how many lines on actual pbx? |
| 19:33.15 | eKo1 | mitemous: of course |
| 19:33.24 | stopher | Seba_soy: run that by me again? |
| 19:33.28 | eKo1 | stopher: strong networking? |
| 19:33.38 | mitemous | eko1: did you run them on their own vlan, or some other way? |
| 19:33.52 | Seba_soy | some switch 100mbps and good network cards I think |
| 19:33.58 | shido6 | g729 is g729, music or not :) |
| 19:34.14 | stopher | eKo1: yea, so you can get enough bandwidth on a system |
| 19:34.17 | eKo1 | mitemous: no vlans. they are all either in the same lan or different lans |
| 19:34.19 | Seba_soy | for music simply make it compatible with codec |
| 19:34.19 | stopher | *phone |
| 19:34.27 | TommyTheKid | its not the g729 or the g711 you need to watch out for, its FTP, HTTP and BitTorrent :) |
| 19:34.28 | ComputerWarm | any one interested in taking a look at a script for me please and maybe point out my mistakes |
| 19:34.29 | Seba_soy | so, if you use g729, transcode music to it |
| 19:34.49 | mitemous | sox works great for transcoding |
| 19:34.50 | eKo1 | I use g729. bandwidth is of no concern yet |
| 19:34.50 | TommyTheKid | Well and NFS, X11, etc :) |
| 19:35.05 | shido6 | and if you have iaxy's use adpcm |
| 19:35.17 | stopher | say this: metallica day on the radio, EVERY user in the building turns on BGM .. so you have 90 phones @ 40mbps / phone |
| 19:35.35 | stopher | wouldn't you think it would be a little hard on anything to put up with that much PLUS calls? |
| 19:35.53 | Un1x | where is dlynes |
| 19:35.56 | Un1x | did he die or something |
| 19:35.57 | Un1x | ? |
| 19:35.59 | mitemous | stopher: you dont get all 90 lines active at once |
| 19:36.13 | stopher | mitemous: if all of those phones have BGM on at once, you do :) |
| 19:36.20 | CunningPike | Un1x: I was talking to him last night |
| 19:36.29 | CunningPike | ~seen dlynes_office |
| 19:36.36 | jbot | dlynes_office <n=dlynes@216.251.149.66> was last seen on IRC in channel #asterisk, 4h 43m 43s ago, saying: 'bbiab...'. |
| 19:37.01 | stopher | annnnyway |
| 19:37.32 | Un1x | ok, well i gotta install asterisk |
| 19:37.41 | Un1x | so should i ninstall asterisk first |
| 19:37.45 | Un1x | or insert my card first |
| 19:37.48 | Un1x | ity's a tdm400p :p |
| 19:38.04 | stopher | I'm hoping that i can use asterisk as alternative VM and AA |
| 19:38.50 | eKo1 | AA? |
| 19:39.01 | stopher | Automated Attendant |
| 19:39.09 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 19:39.30 | eKo1 | Ah. The answer is yes. |
| 19:39.40 | eBody | in order for my sip phone to roll over incoming calls when the line is in use what do i need to do? |
| 19:40.20 | eKo1 | roll over? |
| 19:40.23 | TommyTheKid | Un1x: I'd suggest installing the card, dealing with kudzu (if you have that) as you boot, then installing the zaptel/libpri/asterisk/ast-sounds/etc |
| 19:40.26 | eBody | add incominglimit=8 to sip_additional.conf? |
| 19:40.31 | stopher | you don't know what roll over is eKo1? |
| 19:40.47 | eBody | if the line is in use and somebody else calls, i want another line to ring. |
| 19:41.01 | stopher | eBody: is it with POTS or VoIP? |
| 19:41.06 | eBody | POTS |
| 19:41.09 | TommyTheKid | ewww |
| 19:41.13 | stopher | You need to get that from yer Telco |
| 19:41.13 | eKo1 | eBody: that can be done with * |
| 19:41.19 | Un1x | cool |
| 19:41.20 | Un1x | :p |
| 19:41.36 | stopher | Telco will sense busy, and transfer to line two, if busy transfer to line three, etc, etc |
| 19:41.37 | Seba_soy | eBody: put dial statements one after other |
| 19:41.45 | Seba_soy | if lines is busy then asterisk will dial next |
| 19:41.56 | stopher | yeah, it will DIAL.. taking two lines, not just one |
| 19:42.11 | stopher | eBody: You are talking about INCOMING lines right? |
| 19:42.16 | stopher | *calls |
| 19:42.28 | Seba_soy | simply put dial after other |
| 19:42.32 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 19:42.35 | Seba_soy | if you need 10 phones, put 10 lines |
| 19:42.49 | Seba_soy | from exten => 1 to exten => 10 |
| 19:43.42 | stopher | aaadn eBody is silent |
| 19:43.42 | stopher | lol |
| 19:43.46 | Seba_soy | make a meetme room and put all lines inside... |
| 19:43.52 | eBody | yeah incoming lines |
| 19:44.01 | eBody | sorry guys, it's all hecktic here. |
| 19:44.17 | stopher | yeah, with asterisk, im assuming this, it will sense it is busy and will tell the caller its busy |
| 19:44.25 | stopher | Telco will sense busy, and transfer to line two, if busy transfer to line three, etc, etc |
| 19:44.33 | stopher | it will ring on yer asterisk as the line |
| 19:44.35 | stopher | two |
| 19:44.40 | stopher | instead of one as busy |
| 19:44.50 | eBody | i saw a line that could be added to the sip_addtional.conf |
| 19:45.01 | stopher | for that same feature? |
| 19:45.16 | stopher | a POTS line can't be answered twice and forwarded while yer on it. |
| 19:45.25 | Seba_soy | what exactly do you want to do eBody? |
| 19:45.48 | stopher | he wants incoming call on line one, to go to line two if line one is busy... i think... right eBody? |
| 19:45.52 | eBody | Seba_soy, we have POTS lines coming to our *box using a tdm2400. |
| 19:46.02 | eBody | stoffell, exactly |
| 19:46.04 | Seba_soy | yes.. then? |
| 19:46.19 | eBody | i have the gxp-2000 and the manual says it's supposed to do just that!! but it's not |
| 19:46.26 | stopher | hmm |
| 19:46.40 | eBody | when a call comes from the outside world i want it to goto one place and one extension. |
| 19:46.50 | stopher | oooh |
| 19:46.51 | eBody | if that person is on the phone, i want it to "roll over" her her next "line" |
| 19:46.53 | stopher | like DID |
| 19:47.13 | stopher | you want it to intercom differently |
| 19:47.19 | Seba_soy | ok, so you have 4 pots lines with rotative |
| 19:47.20 | stopher | ignore everything i said then. |
| 19:47.40 | Seba_soy | a person call you and you make a internal phone to ring |
| 19:47.55 | Seba_soy | if that phone is busy you want to make another internal phone to ring |
| 19:47.57 | Seba_soy | that is? |
| 19:48.09 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 19:48.15 | eBody | yeah, but the same extension. |
| 19:48.18 | stopher | he wants the same internal phone to ring on a different intercom line -- thats what i got from it |
| 19:48.27 | eBody | these phones have 4 virtual lines on them. |
| 19:48.38 | stopher | yeah each virtual line is an 'intercom line' |
| 19:48.43 | eKo1 | VoIP phones? |
| 19:49.50 | Seba_soy | well I think if it is only 1 phone with 4 virtual lines, it should not return BUSY instad all 4 lines are busy |
| 19:50.08 | Seba_soy | maybe you can assign different sip account to each intercom line |
| 19:50.40 | eBody | i have but it doesn't really solve the problem. :( |
| 19:51.17 | XARiUS | ebody: I've never used the gxp, but if it works like cisco and polycom's, it should roll over to the next "virtual" line. Try toying with the call waiting features on the gxp. |
| 19:51.24 | Seba_soy | so if you have it registered with 4 sip different accounts, just make ring each one. |
| 19:51.35 | ComputerWarm | any php agi scripters in here if so could you please take a look at http://pastebin.ca/104255 and see if you can help me figure out why it won`t execute. |
| 19:52.39 | eBody | XARiUS, just did, man this is messed up. |
| 19:52.47 | XARiUS | ebody: I forget the specifics, but on the cisco, disabling call waiting caused it to roll over to the next virtual line. or something like that, it's been awhile. |
| 19:53.06 | XARiUS | ah. |
| 19:53.10 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:53.38 | XARiUS | while I'm not afk, anyone seen this msg before? |
| 19:53.46 | XARiUS | NOTICE[2108]: chan_sip.c:11245 handle_request: Unknown SIP command 'SI17676P/2.0' from 'xxx.xxx.xxx.xxx' |
| 19:54.00 | XARiUS | I keep getting them spammed in the console.. it's coming from a sip 7960 |
| 19:54.30 | *** join/#asterisk GerjanT (n=gerjan@frontgate.watchthe.net) |
| 19:54.33 | XARiUS | it doesn't have vad/garp enabled, so not sure what it's all about. |
| 19:56.12 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 19:56.16 | asterboy | Any suggestions on printable material about Asterisk to add to my quote? |
| 19:56.54 | asterboy | Pamplets and what nots |
| 20:00.45 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 20:01.50 | stoffell | uhm? lol |
| 20:01.56 | folder | my eyes hurt and I was all nervous and tense and twichy when I went to Tesco just then. It's all Asterisk's fault. |
| 20:01.57 | Un1x | hey |
| 20:02.03 | Un1x | when i try login into, cvs for digium |
| 20:02.07 | Un1x | it say's unknown host |
| 20:03.07 | XARiUS | wow tesco.. must be UK :) |
| 20:03.11 | folder | yeh |
| 20:03.30 | XARiUS | haven't been in one of those in a while.. the buggy's kill me, in the US the rear wheels are fixed. |
| 20:03.39 | XARiUS | tesco's go sideways, I was a maniac, running into everyone. |
| 20:03.55 | folder | I've been glaring at this computer for probably a minimum of 12hours per day, all week, and it makes me feel really wierd when I go out in public. |
| 20:04.03 | XARiUS | lol |
| 20:04.07 | XARiUS | need to get out more then! |
| 20:04.08 | folder | actually, more like 16 - 17 hrs per day |
| 20:04.18 | nortex | Un1x, cvs is no longer used. Check the website for svn instructions |
| 20:04.18 | folder | need to give up on this shit for a week. |
| 20:04.27 | stoffell | folder: get a 2nd monitor ;) |
| 20:04.49 | folder | how does that help? :D I could try wearing sunglasses,.. |
| 20:05.12 | stoffell | folder: you could do the same stuff in half the time, so... 8hrs a day in front.. and 8 hours in the sun? ;) |
| 20:05.34 | folder | lol :D |
| 20:05.53 | *** join/#asterisk jbsolutios (n=jbenson@217.144.148.26) |
| 20:05.56 | XARiUS | oh man.. breaking news, mel gibson arrested for DUI in LA County. |
| 20:06.01 | XARiUS | Jesus is going to be very dissapointed :( |
| 20:06.01 | stoffell | folder: it IS hot in uk isn't it? ;) |
| 20:06.10 | folder | stoffell: Is it right now, hell yeah! |
| 20:06.10 | clyrrad | When using local_chan is there anyway to access the variables that were set in the originating channel? |
| 20:06.38 | folder | stoffell: but 90% of the time is miserable. Right now, and this last 2 - 3 weeks, it's been unbearably hot (for us.. who aren't used to it). |
| 20:07.13 | jbsolutios | Hi. I am using a Junghanns QuadBRI card with 3 x ISDN2e lines in the UK. When I set it to use Point to Point, it complains that there is no D-channel. Does anyone else have any experience with BRI please? |
| 20:07.14 | stoffell | folder: tell me 'bout it, I'm in belgium, same story.. hot hot hot and sometimes flooding a few hours.. lol |
| 20:07.27 | clyrrad | ... Is there some way to propigate the variables into the local_chan from the originating channel? |
| 20:07.47 | stoffell | jbsolutios: pastebin.ca your zaptel.conf ? |
| 20:07.58 | eBody | i'm not getting callerid info from the outside via my zap trunks....... |
| 20:08.17 | eBody | should i change the context = in the sip.conf??? |
| 20:08.40 | clyrrad | what do your phones say who is calling? |
| 20:08.40 | eBody | from like context = from-sip-external to context = from-trunk ?? |
| 20:08.43 | Bullseye_Network | Is there a way to disable manager transfers for a specific SIP phone? SO a call CANNOT be transfered via manager API? |
| 20:08.45 | eBody | Unknown |
| 20:09.06 | jbsolutios | stoffell: http://pastebin.ca/104299 |
| 20:09.07 | clyrrad | do you have Caller ID="Unknown" in your [general] section? |
| 20:09.23 | eBody | yes |
| 20:09.26 | clyrrad | thats why |
| 20:09.28 | clyrrad | take it out |
| 20:09.37 | eBody | just take it out and the callerid info will come through? |
| 20:09.42 | clyrrad | yep |
| 20:09.49 | clyrrad | you are overriding the caller id with Unknown |
| 20:09.50 | eBody | nice! :) |
| 20:09.53 | jbsolutios | stoffell: I am guessing that BT have set up the line as PTMP instead of PTP. What do you think? |
| 20:10.04 | eBody | clyrrad, then context= is fine?? |
| 20:10.12 | clyrrad | context=incomming |
| 20:10.17 | clyrrad | thats how i have mine set |
| 20:10.19 | stoffell | jbsolutios: could very well be, try the zapata.conf sample included with bristuff, and try other "signalling" modes.. |
| 20:10.39 | jbsolutios | stoffell: do you use BRI in the UK with BT? |
| 20:11.01 | stoffell | jbsolutios: no,using it in belgium, but it's pretty much the same |
| 20:11.21 | jbsolutios | stoffell: I have a feeling that BT may have set up the line using PTMP rather than PTP. Have you used this before? |
| 20:12.00 | stoffell | jbsolutios: try the sample (zapata.conf) of bristuff, it shows a ptmp example.. |
| 20:12.34 | clyrrad | Does anyone use chan_local?? |
| 20:12.51 | TommyTheKid | people who install * on their desktop? :) |
| 20:13.12 | eBody | in sip.conf.....which context is the context= referring to?? |
| 20:13.15 | eBody | in zapata.conf? |
| 20:13.29 | clyrrad | extensions.conf |
| 20:14.16 | eBody | thank you clyrrad |
| 20:14.58 | *** join/#asterisk Seba_soy (n=s@64.76.126.29) |
| 20:15.20 | clyrrad | NP |
| 20:15.24 | TommyTheKid | eBody: in sip.conf, the default for unregistered users |
| 20:15.37 | TommyTheKid | eBody: in zapata the context of inbound calls |
| 20:15.53 | TommyTheKid | refer to extensions.conf for more details :) |
| 20:16.55 | jbsolutios | stoffell: many thanks |
| 20:17.36 | jbsolutios | stoffell: I have exactly the same config on another system working fine with PTP |
| 20:19.28 | asterboy | how many DID #s can * handle? say 3 blocks of 25 for 75 total on a regular PC, 1Gb Ram, Dual core 3.4GHz? |
| 20:19.45 | Seba_soy | somebody knows why ANI is not received on zaptel PRI if it have mark of Not Screened |
| 20:19.45 | [TK]D-Fender | asterboy : Easily. |
| 20:19.52 | TommyTheKid | asterboy: my LX50 had 400 did's |
| 20:19.56 | asterboy | holy shit |
| 20:19.56 | TommyTheKid | coming in over 4 PRIs |
| 20:20.00 | TommyTheKid | itwas less than that |
| 20:20.07 | *** join/#asterisk n3tim (n=n3tim@BHE200139189010.res-com.wayinternet.com.br) |
| 20:20.08 | TommyTheKid | dial P3 1.4GHz |
| 20:20.11 | TommyTheKid | dual that is |
| 20:20.28 | TommyTheKid | our provider is dorky, they give us 100 DID's per PRI |
| 20:20.36 | CunningPike | asterboy: Concurrent calls is the key.......... |
| 20:20.37 | asterboy | In the business setting, I've not been too happy with the way VOIP lines work. |
| 20:20.38 | TommyTheKid | s/give us/make us take/ |
| 20:20.44 | TommyTheKid | thx jbot :) |
| 20:20.58 | asterboy | so the DID and rotary trunk thing looks attractive. |
| 20:21.26 | asterboy | glad I can get them in blocks of 25 |
| 20:21.31 | asterboy | 100 is a bit much |
| 20:21.36 | TommyTheKid | I am getting them from my internal PBX |
| 20:21.48 | asterboy | anyone else unhappy with VOIP termination quality |
| 20:21.52 | TommyTheKid | via cross-over T1 to my ast box |
| 20:22.04 | asterboy | just does not seem to be good for business...great for home. |
| 20:22.06 | [TK]D-Fender | asterboy : I run about 110 on my PRI |
| 20:22.18 | [TK]D-Fender | asterboy : Zeon single 2.8 |
| 20:22.21 | asterboy | on 1 T1? |
| 20:22.50 | asterboy | must have a couple of dual channel PRIs |
| 20:22.50 | eKo1 | asterboy: VoIP is not as reliable as regular PSTN so it is natural that people complain. |
| 20:22.59 | TommyTheKid | essentially, you can probably get away with about a 10:1 ratio of users to lines, in most cases, people call eachother and it doesnt cost a line :) |
| 20:23.02 | asterboy | ya, thats what I'm finding |
| 20:23.15 | Un1x | fuck |
| 20:23.19 | Un1x | now i have to get a free domain |
| 20:23.19 | Un1x | lol |
| 20:23.29 | TommyTheKid | easy there |
| 20:23.51 | asterboy | ok, thanks for the input |
| 20:23.53 | x86 | Un1x: i'll give you one of mine, but it's close to expiring |
| 20:24.20 | TommyTheKid | I had the unfortunate experience of renewing a .nu domain the other day... 70 EUR .. ouch |
| 20:24.33 | rene- | i am getting Ext: 1 Cause: Unallocated (unassigned) number (1), in my incoming calls via PRI ISDN... does anybody know what the F that means? |
| 20:24.38 | TommyTheKid | time to move to another suffix |
| 20:25.02 | TommyTheKid | I am getting missing halding for madatory IE 12 :) |
| 20:25.04 | [TK]D-Fender | asterboy : no.. 110 DIDs on 1 PRI |
| 20:25.06 | *** join/#asterisk gchaix (n=gchaix@osuosl/staff/gchaix) |
| 20:25.16 | TommyTheKid | but I know why, cause the dipshit wont set their caller id right |
| 20:25.25 | rene- | fucking caller id woes |
| 20:25.29 | [TK]D-Fender | asterboy : Low occupancy. this is so I have 2 blocks of 50#'s. 1 for extensions, 1 for direct fax (SpanDSP) |
| 20:25.54 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) |
| 20:26.12 | Seba_soy | i am very happy with voip on my company |
| 20:26.44 | Un1x | x86 whats the name |
| 20:26.45 | Un1x | ? |
| 20:26.53 | TommyTheKid | is there something in AST that I could (from a web app for example) cause two numbers to be dialed and brided to gether? |
| 20:27.33 | rene- | an originate event to the new AJAM manager interfase? |
| 20:27.42 | TommyTheKid | we have a "click to dial" button on our corporate directory, but it uses a java conferencing server that doesnt set the caller ID right |
| 20:28.26 | clyrrad | can anyone help me with a Local Chanel issue? |
| 20:29.10 | eKo1 | TommyTheKid: You can with the manager API. |
| 20:30.01 | TommyTheKid | so.. I need to write another webapp to accept the numbers, do some input validation and submit them to the local manager interface... |
| 20:30.12 | Seba_soy | TommyTheKid: what if someboyd answer line before another one? |
| 20:30.14 | eKo1 | Yep. |
| 20:30.26 | asterboy | [TK]D-Fender, that sounds like a nice setup |
| 20:30.30 | TommyTheKid | Seba_soy: you wnat to dial 1.. wait till they answer then dial the second |
| 20:30.49 | Seba_soy | I supposse, yes... |
| 20:30.56 | Seba_soy | put it on a room. |
| 20:31.02 | TommyTheKid | so when people are searching the "namefinder" (searches ldap) they can click the little phone icon and they will be connected to the other party |
| 20:31.32 | unmanaged | So here is the question, CDR MYSQL adds the 'uniqueid' field to the SQLDB but I need to know how to record a file with this 'uniqueid' in the filename, it looks like epoch time but in ther DB they end in .1 (32523535.1, 21312312331.2, and so on) I need the filenames and uniqueid to match... any idea? is this a asterisk var? or is there anyway to c ontrol the format of the field 'uniqueid'? |
| 20:31.34 | Mercestes | can someone msg me a working page example? Polycom phone does not pick up only rings. I have set(ALERT_INFO=AUTO_ANSWER) and it's not happy..:( |
| 20:32.12 | eKo1 | unmanaged: not that I know of. |
| 20:33.38 | [TK]D-Fender | asterboy : I'm running all my favourite gear, you bet that I like it :) |
| 20:33.57 | TommyTheKid | the unique id is created (most likely) by the database as the record is inserted, so it wouldnt exist till after the call was done (and the CDR was inserted to the DB) |
| 20:34.23 | unmanaged | Tommy I know that... :) |
| 20:35.00 | unmanaged | I am just trying to figure out how to get epoc time into that db |
| 20:35.08 | unmanaged | with the call record |
| 20:35.21 | eKo1 | just convert the calldate to epoch |
| 20:35.31 | unmanaged | hmmm |
| 20:35.35 | clyrrad | ..TDK-Fender are you farmiliar with local_chan? |
| 20:35.51 | eKo1 | clyrrad: you mean chan_local |
| 20:35.55 | clyrrad | yep |
| 20:35.57 | clyrrad | LOL |
| 20:36.07 | clyrrad | are you farmiliar with it? |
| 20:36.13 | eKo1 | Yes. |
| 20:36.24 | clyrrad | Ok- how can i propigate varables? |
| 20:36.28 | clyrrad | with out using globals? |
| 20:36.41 | Un1x | fuck man no-ip is slow |
| 20:36.46 | [TK]D-Fender | clyrrad : Somewhat.. whats your question? |
| 20:37.02 | [TK]D-Fender | clyrrad : explain how you're using it... |
| 20:37.15 | clyrrad | I have my queues.conf file set to member=Local/200@context..... |
| 20:37.34 | clyrrad | problem is when the call jumps back into the dial plan - all the preset varibles that were set before are gone... |
| 20:37.40 | Un1x | yo you know during installation of slackware, slackware asks, you for a Server name, and a domain name i put in canucks and then it asked me domain name i put in foob.tar |
| 20:37.44 | [TK]D-Fender | clyrrad : Let me guess... you want to "push" info to the client PC for queue calls right? |
| 20:37.53 | clyrrad | so when a call comes in on the toll free line - i have tollfree=1..... |
| 20:37.55 | clyrrad | no... |
| 20:37.57 | Un1x | but obviously it doesn't, resolve because it's no domain, but now my domain is active how do i change it, ... |
| 20:38.23 | Mercestes | Can someone give me a working page syntax please? set(alert_info="auto_answer") isn't working on a polycom.. |
| 20:38.35 | clyrrad | now when you dial a queue and you reach a queue member.... it sends you back into the dialplan becase of the use of "Local" - but all the variables are reset - so tollfree="" |
| 20:38.37 | TommyTheKid | clyrrad: you could use astdb :) |
| 20:38.38 | clyrrad | follow me? |
| 20:38.51 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 20:38.56 | Netgeeks | anyone here give me a quick tip on getting the current head of asterisk? svn checkout http://svn.digium.com/svn gives me an error |
| 20:38.58 | clyrrad | how would astdb help in this situation? |
| 20:38.58 | Mercestes | Sip.cfg reads: <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/> |
| 20:39.22 | TommyTheKid | it wouldn't be a global variable, you could set it in there .. probably "misuse" but :) |
| 20:39.54 | [TK]D-Fender | clyrrad : cheap trick for you : Push the vars into AstDB, then change the callerID name before entering the queue. inside all "Local" dials you would then strip the family/key info from the callerid name and use to putt the "pushed values". You'd naturally want to run a clean-up program once in a while to ensure it doesn't flood. |
| 20:40.11 | [TK]D-Fender | TTK : short version, but yeah :) |
| 20:40.34 | clyrrad | ekkkkk - is there not a better way to do this? |
| 20:40.50 | TommyTheKid | SIP phone instead of local |
| 20:40.51 | TommyTheKid | ? |
| 20:41.02 | [TK]D-Fender | clyrrad : Have you tried referencing the vars with "_" in front for inheritance? |
| 20:41.06 | clyrrad | no - becase then call forwarding wont work from queue |
| 20:41.18 | clyrrad | no - I have not tried that |
| 20:41.20 | *** part/#asterisk stopher (n=business@cm-24-121-73-66.kingman.az.npgco.com) |
| 20:41.21 | *** join/#asterisk stopher (n=business@cm-24-121-73-66.kingman.az.npgco.com) |
| 20:41.26 | clyrrad | let me give that a shot |
| 20:42.46 | clyrrad | like this? ${_tollfree} |
| 20:42.53 | clyrrad | or like this _${tollfree} |
| 20:43.05 | rene- | rule #1 of IT management: blame the provider |
| 20:43.11 | *** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 20:43.56 | [TK]D-Fender | ok, I'm outta here... later all. |
| 20:44.00 | [TK]D-Fender | BBIAB |
| 20:44.17 | *** part/#asterisk ComputerWarm (n=donc@209.29.156.12) |
| 20:45.44 | Un1x | Could not resolve for cvs.digium.com. |
| 20:45.52 | Un1x | there we go cvs is down |
| 20:45.53 | file | we don't use CVS anymore |
| 20:45.59 | file | we haven't for quite some time |
| 20:46.05 | Un1x | heh i was following directons from ther book i received with the card... |
| 20:46.26 | Un1x | file so should i just download the tar.gz? |
| 20:46.31 | Un1x | and do it that way or use ftp... |
| 20:46.34 | jbsolutios | stoffell: I have checked. it is PTP and I keep getting this error with the BRIstuff: WARNING[16281]: chan_zap.c:2506 pri_find_dchan: No D-channels av |
| 20:46.34 | jbsolutios | ailable! Using Primary channel 3 as D-channel anyway! |
| 20:46.36 | file | you can if you wish |
| 20:47.23 | eKo1 | Un1x: use svn |
| 20:48.37 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 20:48.58 | Un1x | «file» wich one is the latest version of Asterisk... |
| 20:49.06 | folder | Is it normal for SIP messages to be coming through every 30 or so seconds? Mostly "CSeq: 102 OPTIONS" ? |
| 20:49.28 | folder | (with sip debug enabled) |
| 20:49.33 | file | 1.2.10 is the most recent 1.2 |
| 20:49.38 | Un1x | okays |
| 20:50.24 | Netgeeks | hrm, is it coincidence that the three people talking now are Unix file folder? |
| 20:50.38 | folder | LOL |
| 20:50.40 | file | maybe! |
| 20:50.48 | folder | file.. come here let me encompass you :) |
| 20:51.18 | Un1x | lol |
| 20:51.21 | E-bola | lol |
| 20:51.30 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 20:51.52 | folder | I changed mine to folder for a laugh after seeing file |
| 20:52.03 | folder | that's how exciting I am |
| 20:52.33 | folder | lol |
| 20:52.43 | gchaix | inode, anyone? |
| 20:52.46 | gchaix | heh |
| 20:52.47 | inode | lol :D |
| 20:52.50 | inode | fancy that! |
| 20:52.57 | directory | I have lots of nicks :D |
| 20:53.07 | directory | file, filesystem, devicenode, partition, directory, symlink |
| 20:53.25 | gchaix | devnull |
| 20:54.59 | mountainm2k | OK, another ABE question -- or maybe not strictly ABE... musiconhold doesn't seem to work... |
| 20:55.08 | gchaix | Here's one for y'all: anyone ever tried to run * on a Gentoo Xen slice? More specifically, gotten the zaptel stuff to compile so the MeetMe app can be used? |
| 20:55.19 | folder | This is frustrating. I've got sip debugging on, ready to log a failed call through to my GSM SIP gateway. But the calls aren't failing :(. I guarantee that when I leave the house or try it from bed, it won't work. sip show channels says "Last Message: tx INVITE" when it's not working. it's like the gateway box isn't answering the INVITE. |
| 20:57.49 | gchaix | I suspect zaptel is unhappy because Xen is not letting it talk to the hardware clock |
| 20:58.06 | Zodiacal | polycom's kick cisco's butt |
| 21:00.18 | folder | Is there any point in me messing about with SIP Expire times to try to fix this? |
| 21:00.36 | nighty_ | would anyone know of a WIFI SIP Phone that can do PTT ? |
| 21:00.49 | mountainm2k | <PROTECTED> |
| 21:00.54 | folder | PTT? like a walkie talkie? |
| 21:00.54 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-8.getnet.net) |
| 21:00.55 | nighty_ | Push to Talk |
| 21:00.57 | mountainm2k | as in push-to-talk, like the nextel? |
| 21:01.11 | mountainm2k | I don't know of any... |
| 21:01.30 | folder | what purpose does PTT have on a phone? |
| 21:01.40 | nighty_ | walkie talkie like |
| 21:01.48 | eKo1 | might as well buy a walkie talkie |
| 21:01.50 | mountainm2k | I would guess for hooking Asterisk up to a radio system... |
| 21:01.56 | Qwell[] | eKo1: app_rpt! |
| 21:01.58 | mountainm2k | which it is capable of |
| 21:02.00 | mountainm2k | yeah, that |
| 21:02.08 | nighty_ | eKo1: no, not when you also need phone caps |
| 21:02.21 | eKo1 | nighty_: buy a cell phone |
| 21:02.36 | nighty_ | eKo1: would that works ? I just need the wifi part |
| 21:02.38 | unmanaged | folder, use can use asterisk as a repeater control |
| 21:03.03 | folder | is that a radio term? |
| 21:03.12 | eKo1 | nighty_: i dunno |
| 21:03.22 | Johnnie | apt_rpt kicks butt! |
| 21:03.40 | nighty_ | what I need is SIP multicast PTT |
| 21:03.42 | Zodiacal | anyone know if polycom hints can display caller id? |
| 21:03.50 | nortex | What kind of hardware is needed for app_rpt? |
| 21:04.16 | Johnnie | You basically need to build a custom board to interface with COR/COS, then apply that to a TDM card. |
| 21:04.21 | nortex | Zodiacal, So you can see who someone is on the phone to? |
| 21:04.31 | unmanaged | hmm |
| 21:04.41 | mountainm2k | nortex: They also make a quad pciradio board |
| 21:04.42 | Zodiacal | nortex yeah |
| 21:04.49 | Un1x | hey you know in zaptel.conf you have to add youre, loadzone.. and defaultloadzone, well mines is canada i added CA and deleted the US |
| 21:04.54 | Un1x | but it says no such loadzone as ca |
| 21:04.55 | unmanaged | I had seen someplace that someone was useing a channel bank to do something like that, johnnie |
| 21:04.56 | Un1x | why is that? |
| 21:05.06 | mountainm2k | nortex: but that's for hooking asterisk into a radio system -- like ham radio... |
| 21:05.11 | Zodiacal | nortex now it just displays "line in use" |
| 21:05.16 | nortex | Zodiacal, I have not, I have only seen status. |
| 21:05.19 | Un1x | somoene |
| 21:05.26 | Zodiacal | nortex okie thanks |
| 21:06.00 | CunningPike | Zodiacal: Consider FOP for that sort of thing |
| 21:06.13 | Un1x | loadzone = us |
| 21:06.13 | Un1x | defaultzone=us |
| 21:06.18 | Un1x | i changed them to CA |
| 21:06.23 | Un1x | and it dont work someone.. file? |
| 21:06.28 | unmanaged | http://www.allstarlink.org/ asterisk & networked repeaters |
| 21:06.35 | CunningPike | Un1x: Just use US |
| 21:06.40 | Un1x | ok |
| 21:06.49 | *** join/#asterisk woolbeo (n=woolbeo@toby.stoneflytech.com) |
| 21:07.41 | folder | dear god. George Bush "wants a lebanon with a free and pro-american governmen" |
| 21:07.43 | folder | t |
| 21:07.45 | woolbeo | Is it possible to configure * so that one can forward voice mail between different voice mail contexts? |
| 21:07.45 | nortex | What about hooking it to a motorala walkie talkie sysem? is that possible without some custom boards? |
| 21:07.51 | folder | that sounds like just what they were saying about Iraq |
| 21:07.56 | Mercestes | Can anyone give me a hand with extension app Page()? |
| 21:08.28 | Johnnie | unmanaged: I've heard of that being done... |
| 21:08.49 | unmanaged | nortex, look here http://www.zapatatelephony.org/app_rpt.html, also check out fcc.goc for what you can leagly do on non licenced bands |
| 21:08.50 | Johnnie | I have some UHF amateur repeaters in service, and I have them tied together with Asterisk. |
| 21:08.55 | unmanaged | fcc.gov |
| 21:09.20 | unmanaged | johnnie, KE4TVV here what is your call? |
| 21:09.30 | Johnnie | The only thing I haven't bothered with is GMRS, since you can't really piddle with telephony on GMRS anyway. |
| 21:09.31 | Johnnie | K3JDL |
| 21:10.01 | Johnnie | A friend of mine has a crossband link running Asterisk too...it's a lot of fun. |
| 21:10.26 | unmanaged | how much does it cost? |
| 21:10.28 | Johnnie | The fact that we can link repeaters on IAX2 sort of fascinates the hell out of me. |
| 21:10.34 | Johnnie | What part? |
| 21:10.49 | unmanaged | radio interface? |
| 21:11.09 | Johnnie | Well, a friend of mine built mine... I gave him $150 per interface, not sure what the actual cost was. |
| 21:11.13 | Un1x | isn't Digiums tech support 24/7 |
| 21:11.14 | Johnnie | I can ask...be right back. |
| 21:11.16 | Un1x | or customers timwe zone |
| 21:11.20 | Un1x | i could say im on PST lol |
| 21:11.38 | folder | Where is SIP Expire time set in Asterisk? |
| 21:12.34 | syzygyBSD | Un1x: Mon. - Fri., 8 am - 5 pm (customer's time zone) |
| 21:13.26 | Un1x | heh yea,. if im in PST or MST im i can stil call them :)_ |
| 21:13.42 | Un1x | i wish zaptel deves would hurry up and fix the drivers for freebsd ;) |
| 21:13.46 | Un1x | then i wouldn't need linux :P |
| 21:13.59 | Johnnie | unmanaged: I think he bought the boards for $25 and $55 for the parts. |
| 21:14.26 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:14.33 | Johnnie | He went on site and did the interfacing for me when I was away, plus he's a heck of a good guy anyway, so I gave him extra...haha |
| 21:14.56 | Un1x | ?. i bought the tdm22B :P |
| 21:14.57 | Johnnie | So, you're talking about $80 or so. I think he got all of his parts from DigiKey and Mouser, perhaps Jameco. |
| 21:15.03 | Un1x | 2 fxs and 2 fxo digium card brand new hehe :) |
| 21:15.38 | directory | Un1x: someone else has taken it upon themselves to port the drivers to FreeBSD |
| 21:16.22 | folder | yeah |
| 21:17.15 | *** join/#asterisk c4t3l (n=c4t3l@cpe-70-116-156-139.houston.res.rr.com) |
| 21:18.00 | hads|home | 8 am - 5 pm (customer's time zone) <- That's bizzare, I'm GMT+12 - who wants to be up in the middle of the night to take calls from here. |
| 21:19.09 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 21:19.39 | angler | Support is 24/7 for customers that have a 24x7 maintainance agreement |
| 21:19.46 | folder | What does having 'a crossband link running Asterisk' do? Is this Asterisk doing radio stuff, like for radio-freaks, or is it the radio gear doing phone stuff, like for phone-geeks? |
| 21:23.21 | Mercestes | Can someone please help me with Application Page() on Asterisk 2.10 with a Polycom 501 using Sip_1.6.6? |
| 21:23.38 | Mercestes | POlycom just rings..:( |
| 21:23.46 | CANO-1982 | nortex,I thik you could use the phonepatch project for asterisk |
| 21:24.37 | eKo1 | Asterisk 2.10? |
| 21:25.01 | Mercestes | Asterisk 1.2.10 |
| 21:25.49 | nortex | CANO-1982, Never heard of it |
| 21:26.00 | CANO-1982 | wait a minute |
| 21:26.10 | nortex | CANO-1982, k |
| 21:27.09 | CANO-1982 | nortex, Im curretly trying to use it |
| 21:27.22 | CANO-1982 | http://www.nongnu.org/asterisk-phpatch/ |
| 21:27.46 | CANO-1982 | but it seems that i cant control my serial port, yet |
| 21:27.56 | CANO-1982 | take a look |
| 21:30.10 | CANO-1982 | nortex, I think it cold help you |
| 21:30.38 | nortex | Looks useful. I may have to bookmark this in case someone ask if we could do this here. |
| 21:33.18 | unmanaged | folder. both |
| 21:33.45 | folder | unmanaged: ah I see. |
| 21:34.00 | CANO-1982 | ok, how could I know if it worked for you? |
| 21:34.18 | CANO-1982 | nortex, because Im having some troubles |
| 21:36.41 | unmanaged | oh this is cool |
| 21:37.03 | unmanaged | I have a 12v truck mount pc that I got out of a old cop car |
| 21:37.04 | unmanaged | ! |
| 21:37.32 | unmanaged | hmmm |
| 21:39.00 | nortex | a mobile pbx :) |
| 21:39.37 | macTijn | hmm, build in a nice gauge that moves like KITT ;) |
| 21:39.57 | CANO-1982 | nortex,aj aja, cool |
| 21:40.17 | CANO-1982 | could I email you? |
| 21:40.28 | CANO-1982 | sorry, my english sucks! |
| 21:41.32 | nortex | CANO-1982, Are you having problems with phonepatch stuff? I am probably no help since I just read the page a few minutes ago. |
| 21:42.11 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:43.16 | *** join/#asterisk mikea (n=mike@66.77.162.99) |
| 21:43.53 | mikea | Anyone know of any issues with the latest svn snapshot (as of a week ago) and problems recognizing RFC2833 DTMF in the voicemail app? |
| 21:44.06 | mikea | * isn't recognized, but # is. |
| 21:44.57 | syzygyBSD | is it possible to not allow users to change their voicemail messages? either a parameter passed to voicemailmain or in voicemail.conf |
| 21:45.40 | *** join/#asterisk mikea (n=mike@66.77.162.99) |
| 21:45.55 | Bullseye_Network | How is letting agents login based on the callerid or the sip phone? I changed the callerid in sip.conf for a phone and it would not let them agent login. |
| 21:46.28 | mikea | sorry, my IRC client is half broken.. any problems w/ recent snapshots and rfc2833 digits? |
| 21:47.36 | CANO-1982 | nortex, teah, but maybe its a problem with my PC |
| 21:47.47 | nortex | Bullseye_Network, The sip information is recorded in to the astdb for dynamic agents. |
| 21:47.55 | CANO-1982 | nortex, well, Im leaving now |
| 21:48.07 | nortex | CANO-1982, Good Luck. |
| 21:48.13 | Bullseye_Network | im having problems with some sip phones somehome getting the callerid from another phone |
| 21:48.17 | CANO-1982 | thanks |
| 21:48.37 | Bullseye_Network | s/somehome/somehow/g |
| 21:49.24 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 21:49.24 | folder | what does the 'g' at the end of the s/i asoicasjcs/i can't spell/ line mean? |
| 21:49.44 | Bullseye_Network | in vi its global... |
| 21:49.53 | *** join/#asterisk mikea (n=mike@66.77.162.99) |
| 21:50.00 | mikea | alright, my client is fixed. |
| 21:50.17 | folder | oh. i'm non the wiser then. :) |
| 21:50.37 | *** part/#asterisk unmanaged (n=unmanage@64.89.118.139) |
| 21:50.47 | mikea | I am having a problem hitting * to access my voicemail login with recent versions of asterisk. |
| 21:50.53 | Bullseye_Network | It replaces all occurences on the current line. without g it only replaces the first occurance. |
| 21:51.06 | folder | ahhh. now I understand. |
| 21:51.08 | mikea | I'm not sure if it's a DTMF problem, or if the voice mail app just doesn't recognize * now. I can use # to hang up. |
| 21:51.09 | folder | cool |
| 21:51.13 | nortex | later all have a good weekend. |
| 21:51.14 | *** join/#asterisk adorah (n=Administ@84.94.133.192.cable.012.net.il) |
| 21:51.22 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:52.14 | mikea | ah, you know what, my sip phones can't hit * locally. |
| 21:52.45 | folder | how come? |
| 21:53.18 | mikea | When i dial voice mail |
| 21:53.33 | mikea | on older versions of asterisk, you could hit * to log in and check your voicemail |
| 21:53.43 | mikea | with the snapshot I'm using, that's not working. |
| 21:54.30 | angler | mikea, do you have the "a" extension in extensions.conf? |
| 21:54.31 | folder | oh yeah I understood that, but I thought you were saying it was because your sip phones couldn't send it or something |
| 21:54.45 | mikea | angler, no. |
| 21:54.47 | *** join/#asterisk Kernel-Kris (n=kkirklan@lfkn-fw.angelinacounty.net) |
| 21:55.04 | Kernel-Kris | $20 USA PayPal to someone that will build a dial plan for me |
| 21:55.20 | mikea | I'm not using AMP or anything. I built my extensions.conf by hand. |
| 21:55.21 | angler | mikea, woops mis-read what you had typed |
| 21:55.23 | mikea | Am I missing something? |
| 21:55.52 | Kernel-Kris | the dial plan is for only 6 people with one incoming call attendant and voice mail |
| 21:55.56 | angler | mikea, you should be able to just make an extension "*" that dumps into voicemail |
| 21:56.00 | mikea | Has the voicemail application changed since 1.2.10? |
| 21:56.04 | sharp | i get Unable to open '/dev/zap/pseudo': No such file or directory |
| 21:56.08 | sharp | ztdummy is loaded |
| 21:56.40 | mikea | angler: it will dial that extension if I hit it even during the recording? |
| 21:56.49 | adorah | <Kernel-Kris>it is sooo easy to do with freepbx interface..DIY.. |
| 21:57.30 | Kernel-Kris | adorah: freepbx ...can i have a link please.....also i cant get the zaptel drivers to compile in sarge |
| 21:57.38 | angler | mikea, is your setup basically you dial into your own extension which gives you your own voicemail and then you want to press * to login? |
| 21:57.50 | mikea | yeah |
| 21:58.25 | angler | mikea, then you need "a" extension in the context the Voicemail app is called which calls VoicemailMain |
| 21:58.54 | mikea | angler: what does the "a" extension do? |
| 21:59.03 | angler | mikea, it's been this way for awhile, not sure how you were doing it before without it |
| 21:59.06 | adorah | <Kernel-Kris>either http://www.freepbx.org/trac or www.trixbox.org to get a bootable build linux=*=freepbx in one run.. |
| 21:59.07 | hads|home | a = * |
| 21:59.42 | angler | mikea, you will route the "a" extension to voicemailmain. The voicemail app itself jumps to "a" when * is pressed |
| 22:00.06 | mikea | ah, that's what I am missing |
| 22:00.21 | mikea | That works perfectly. |
| 22:00.44 | mikea | I'm trying to move away from AMP and build my own extensions.conf. I didn't see the "a" extension in there. |
| 22:00.45 | Kernel-Kris | adorah: well i would like a fully functional non cripled debain distro this box will be doing other things |
| 22:00.55 | mikea | AMP probably hid it away someplace. |
| 22:01.31 | adorah | Kernel-Kris: u better use a dedicated machine for *.. |
| 22:02.05 | mitemous | ever use a VPS for a deployment with a few phones |
| 22:02.28 | Kernel-Kris | adorah: its a 1.2ghz dual processor box with 1gig of ram.....and 9 raid 5 scsi drives....i would like for it to do more than just PBX |
| 22:02.33 | adorah | I have that annoying bug that when an incoming call dial an extension and no-replay the line can keep busy for hours-zap lines mainly.. |
| 22:02.35 | mitemous | in case anyone was wondering..NAT is a bitch |
| 22:02.55 | eKo1 | mitemous: use iax |
| 22:03.26 | adorah | Kernel-Kris: than get a cheaper box even PIII and use it only for *.. |
| 22:03.38 | dlynes_laptop | mitemous, i see people bitching about it constantly on here...once i figured out how to get asterisk to work with it, i've never had an issue with it, save for 1 or 2 routers with flaky firmware |
| 22:04.13 | mitemous | i'm not actually working with * |
| 22:04.22 | Kernel-Kris | adorah: and do you recomedn the above linx for *.... i would like somthing i didnt have to compile all these drives for my one port card, and i will be ading 1 more card in the future |
| 22:04.22 | mitemous | i'm playing with a SIP component in visual studio |
| 22:04.32 | dlynes_laptop | heh |
| 22:04.33 | hads|home | Kernel-Kris: You will run into sound issues if you try and use your Asterisk box for other things. |
| 22:04.49 | dlynes_laptop | a .NET sip component? |
| 22:04.50 | mitemous | gotta figure out how to add the rport parameter into the Via field ;) |
| 22:05.03 | mitemous | dlynes: exactly.. Sip.NET actually |
| 22:05.10 | dlynes_laptop | ah...never heard of it |
| 22:05.12 | dlynes_laptop | but sounds cool |
| 22:05.19 | eKo1 | Sip.NET?! oh boy... |
| 22:05.22 | mitemous | yeah, just went surfing for one..and that one came up |
| 22:05.25 | dlynes_laptop | It's extension to the Socket class? |
| 22:05.28 | woolbeo | anyone had problems with a dsl modem causing noise on a line even with a dsl filter before the tdm card? |
| 22:05.34 | mitemous | yeah, it sits on top of it |
| 22:05.43 | mitemous | sipclient.connect(); |
| 22:05.48 | dlynes_laptop | cool |
| 22:05.49 | mitemous | sipclient.register(); |
| 22:05.52 | mitemous | sipclient.invite(); |
| 22:05.53 | mitemous | etc |
| 22:06.24 | mitemous | i'm about to head home to the cable modem so i can do some more programming without being behind NAT though |
| 22:06.30 | dlynes_laptop | cool...rewrite asterisk in C#, so it'll run on both windows and mono :) |
| 22:06.31 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 22:06.34 | teknoprep | hi all |
| 22:06.44 | teknoprep | i can't connect to my asterisk server over the inet |
| 22:06.46 | teknoprep | very odd |
| 22:06.52 | teknoprep | yet i can connect to it form the local network |
| 22:06.57 | teknoprep | using IAX or SIP |
| 22:07.00 | AvoidingDeadlock | you don't have a default gateway set? |
| 22:07.07 | teknoprep | for the asterisk box? |
| 22:07.11 | AvoidingDeadlock | yes |
| 22:07.13 | teknoprep | like a network defauilt gateway |
| 22:07.15 | teknoprep | of course i do |
| 22:07.25 | teknoprep | it has an INET ip not nat'd |
| 22:07.53 | dlynes_laptop | teknoprep, do you have port 5060, 4569, and your rtp ports blocked on the firewall? |
| 22:07.56 | teknoprep | yes |
| 22:08.01 | teknoprep | i opened all ports for testing |
| 22:08.03 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:08.08 | dlynes_laptop | yes, or no? |
| 22:08.13 | mitemous | dlynes: that'd be an awesome project..i cant imagine how long it would take..lol |
| 22:08.13 | teknoprep | no i block nothing |
| 22:08.16 | dlynes_laptop | You're telling me yes, they are bocked |
| 22:08.20 | teknoprep | no |
| 22:08.22 | teknoprep | they are not blocked |
| 22:08.22 | dlynes_laptop | but now you're saying they're not |
| 22:08.25 | dlynes_laptop | which is it? |
| 22:08.26 | dlynes_laptop | ok |
| 22:08.26 | teknoprep | i read your question wrong at first |
| 22:08.58 | x86 | anyone know of a place that will manufacture calling cards for me? |
| 22:09.13 | x86 | just physically make the cards |
| 22:09.15 | mitemous | dlynes: i'm actually just trying to build a dirty prototype of a click-to-call app |
| 22:09.25 | eKo1 | x86: where? |
| 22:09.45 | x86 | eKo1: in the US preferrably, but if cheaper overseas, so be it ;) |
| 22:09.49 | dlynes_laptop | x86, www.goldline.net |
| 22:10.05 | dlynes_laptop | x86, It's a North Vancouver, BC, Canada company |
| 22:10.37 | mikea | angler: that fixed my problem. It's working fine. Thanks! |
| 22:10.49 | dlynes_laptop | mitemous, what for? |
| 22:10.56 | dlynes_laptop | mitemous, why reinvent the wheel? |
| 22:10.59 | x86 | dlynes_laptop: i clicked on the cards and got a list of companies they resell phone cards for |
| 22:11.08 | x86 | dlynes_laptop: i want want that will print MY cards for me |
| 22:11.08 | mikea | dlynes_laptop: Goldline is good? |
| 22:11.17 | dlynes_laptop | x86, they do |
| 22:11.19 | mitemous | dlynes: because i dont have an extra server laying around to install * on :) |
| 22:11.22 | dlynes_laptop | x86, they do private labelling |
| 22:11.33 | dlynes_laptop | mikea, yeah..he's been around for a while |
| 22:11.45 | mikea | dlynes_laptop: They might be one of my customers :-) |
| 22:11.49 | dlynes_laptop | mikea, he sells about 20 different brands of cards himself in the local vancouver market |
| 22:11.56 | mikea | oh |
| 22:12.01 | x86 | dlynes_laptop: private labelling of companies they resell for right |
| 22:12.04 | mikea | There's a voip provider called Goldline. :- |
| 22:12.05 | x86 | dlynes_laptop: not my service ;) |
| 22:12.16 | x86 | dlynes_laptop: they'll put my name on some other company's card |
| 22:12.22 | x86 | dlynes_laptop: i want my name on MY OWN cards ;) |
| 22:12.33 | dlynes_laptop | x86, oh...that I don't know...I thought they did private labelling for your own company |
| 22:12.40 | x86 | dlynes_laptop: nope |
| 22:12.44 | Kernel-Kris | ok any recomendations on a asterisk distro.....one that works better than *@Home |
| 22:12.56 | x86 | Asterisk ;) |
| 22:13.26 | macTijn | ubuntu + apt-get install asterisk-bristuff |
| 22:13.48 | Mercestes | Anyone have paging working in 1.2.10? |
| 22:13.55 | dlynes_laptop | Kernel-Kris, how about ftp://ftp.digium.com/pub/telephony/asterisk/asterisk-1.2.10.tar.gz? |
| 22:14.09 | mikea | I just had thw worst experience with a zaptel card.. I rebooted.. after reboot my T1s wouldnt come up.. the driver wouldn't use any interupts.. couldn't get anything to work |
| 22:14.10 | dlynes_laptop | Mercestes, not yet, but i will tonight |
| 22:14.16 | mikea | until I moved it to a different PCI slot |
| 22:14.20 | mikea | then boom, works perfectly |
| 22:14.21 | Mercestes | dlynes_laptop: I'm having issues... |
| 22:14.27 | mikea | that's the second time it's happened. |
| 22:14.36 | dlynes_laptop | Mercestes, you're using regular overhead paging, or phone paging? |
| 22:14.38 | Kernel-Kris | asterisk disrto as in OS+Asterisk not just the package itself |
| 22:15.21 | mikea | Personally I just installed CentOS and installed compiled asterisk myself. AMP is great if you're just looking to setup a PBX.. but if you want to make asterisk do some fun stuff, it's frustrating. |
| 22:15.22 | Mercestes | dlynes_laptop: Nevermind..got it. |
| 22:15.34 | dlynes_laptop | Kernel-Kris, ftp.slackware.com and grab slackware, install it, then go to ftp://ftp.digium.com/pub/telephony/asterisk/asterisk-1.2.10.tar.gz and install asterisk |
| 22:15.53 | hads|home | Kernel-Kris: Whatever distro you are comfortable with. |
| 22:16.21 | hads|home | Hey dlynes_laptop |
| 22:16.32 | dlynes_laptop | hads|home, world domination by slackware is coming to an irc channel near you, soon |
| 22:16.40 | hads|home | :) |
| 22:18.23 | mikea | I used to love slackware when I was into being being a geek and doing things manually. |
| 22:18.27 | mikea | Now I'm old and lazy. |
| 22:18.28 | mikea | :D |
| 22:20.03 | *** join/#asterisk HolyGod (i=nobody@got.securebinary.com) |
| 22:22.46 | XARiUS | so whats better now days, just compiling from svn or using the binary releases? |
| 22:23.13 | dlynes_laptop | x86, yeah, they do exactly what you want to do |
| 22:23.21 | dlynes_laptop | x86, they just haven't added that info to the website yet |
| 22:23.43 | *** join/#asterisk esculapio__ (n=ESCulapi@reserved-231-1.tricom.net) |
| 22:23.48 | dlynes_laptop | x86, You can email the president of western canada operations (vp of glprint) at zargaran@goldline.net |
| 22:24.18 | dlynes_laptop | x86, just email him telling him what you want to do, and that you want pricing on it |
| 22:24.47 | dlynes_laptop | x86, if you want, just tell him daniel at 24/7 communications sent you |
| 22:24.50 | dlynes_laptop | x86, :) |
| 22:29.54 | x86 | dlynes_laptop: thanks |
| 22:31.25 | *** part/#asterisk Kernel-Kris (n=kkirklan@lfkn-fw.angelinacounty.net) |
| 22:34.04 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 22:39.58 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 22:44.14 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 22:44.28 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 22:44.30 | SpaceBass | hey folks |
| 22:44.35 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 22:44.39 | SpaceBass | anyone registering with gizmo and doing inbound routing? |
| 22:50.33 | Un1x | ~fxsfxo |
| 22:50.34 | jbot | methinks fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 22:51.22 | Un1x | so i gotta plug my phone into a FXS port then |
| 22:51.26 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-225.cybersurf.com) |
| 22:55.01 | Un1x | anyone here? |
| 22:55.19 | hads|home | Yes. Phone into FXS |
| 22:58.56 | *** part/#asterisk woolbeo (n=woolbeo@toby.stoneflytech.com) |
| 23:04.46 | *** join/#asterisk Amilcar_ (n=amilcar@201.34.202.17) |
| 23:04.49 | SpaceBass | anyone registering with gizmo and doing inbound routing? |
| 23:05.30 | Un1x | anyone aroudn to help with extensions.conf |
| 23:05.37 | Un1x | it's the only problem im having,.... |
| 23:05.45 | Amilcar_ | Anyone here knows why digium have choose Mantis as the tracker?? |
| 23:05.46 | Un1x | other configs where fine and easy but extensions.conf is weird... |
| 23:08.33 | Un1x | anyone... |
| 23:08.44 | Un1x | bah wish phone worked id call tech support :P |
| 23:08.54 | SwK | un1x ask about the problem |
| 23:09.10 | SwK | someone might answer |
| 23:09.14 | Un1x | err well, on page 33 of the book they send with there tdm22b cards |
| 23:09.23 | Un1x | says i should add this dail plan in extensions.conf... |
| 23:09.25 | directory | yessss? what? |
| 23:09.36 | Un1x | but see here, the problem, is imnot going to use PSTN lines... |
| 23:09.40 | *** kick/#asterisk [Un1x!n=twisted@pdpc/supporter/active/twisted] by twisted[asteria] (Wake up, call them.) |
| 23:09.40 | *** join/#asterisk Un1x (i=Sean@72.61.82.242) |
| 23:09.42 | Un1x | this is direct to voip.. |
| 23:09.56 | SwK | heh |
| 23:09.58 | Un1x | that wasnt nice :/ |
| 23:10.00 | directory | haha |
| 23:10.02 | SwK | well then dial the sip channel you need |
| 23:10.03 | Un1x | heh directory |
| 23:10.05 | Un1x | can ya help |
| 23:10.10 | twisted[asteria] | then dont' send me private messages telling me to wake up |
| 23:10.13 | Un1x | well i need a dailplan according to the book |
| 23:10.17 | Un1x | heh k twis |
| 23:10.28 | directory | your dialplan doesn't have to follow the book, you can make it do whatever you want |
| 23:10.36 | Un1x | ive never made the dailplan |
| 23:10.39 | directory | and if my house burns down because I wasn't watching what I'm cooking, I'll blame you |
| 23:10.40 | Un1x | directory do you have the book |
| 23:10.43 | Un1x | look at page 33 |
| 23:10.47 | directory | I do not |
| 23:10.51 | Un1x | :/ |
| 23:10.57 | Un1x | gr8 whyd they send me this then :/ |
| 23:11.04 | directory | and I'm also not in technical support |
| 23:11.09 | Un1x | anyway directory any help on where or how i can create a dailplan |
| 23:11.12 | Corydon-w | Un1x: you do that again and you'll be removed. You can interrupt me when you're paying me at my going rate. |
| 23:11.25 | twisted[asteria] | Corydon-w, haha, you too? |
| 23:11.39 | directory | he probably messaged every op |
| 23:11.40 | Un1x | twisted yea |
| 23:11.56 | Un1x | ok guys, im sorry was stupid thing of me to do... |
| 23:12.05 | Un1x | so now back to my dailplan, can i get help directory... |
| 23:12.06 | directory | Un1x: ask a specific question and someone may answer it |
| 23:12.14 | Un1x | page 33 on the book... |
| 23:12.19 | directory | Un1x: otherwise you can call Digium technical support for installation assistance |
| 23:12.20 | twisted[asteria] | what book? |
| 23:12.29 | Un1x | a dailplan at the end of extensions.conf i dont need the pstn.. |
| 23:12.32 | hads|home | Un1x: That isn't a question. |
| 23:12.36 | Un1x | the digium book |
| 23:12.47 | Un1x | users manual i did everything following it but now i cant do what it's asking.. |
| 23:12.58 | twisted[asteria] | oh, you bought ABE? |
| 23:13.04 | *** join/#asterisk roving_prole (n=Harper@c-71-199-16-110.hsd1.co.comcast.net) |
| 23:13.17 | twisted[asteria] | (asterisk buisness edition) |
| 23:13.19 | Un1x | no i bought the, TDM22B kit... |
| 23:13.22 | Un1x | nopes. |
| 23:13.30 | twisted[asteria] | oh, then you should be talking to digium tech support |
| 23:13.32 | twisted[asteria] | this is the asterisk channel |
| 23:13.41 | twisted[asteria] | :P |
| 23:13.50 | directory | my french fries are done! |
| 23:13.52 | Un1x | lol comon, asterisk is kinda 'owned by them' ;P |
| 23:13.53 | twisted[asteria] | yay |
| 23:13.56 | twisted[asteria] | uh |
| 23:14.02 | twisted[asteria] | asterisk is an open source project |
| 23:14.04 | hads|home | mmm fries |
| 23:14.05 | directory | owned? |
| 23:14.14 | Un1x | well yes, but there managing or supporting it or something ;P |
| 23:14.22 | twisted[asteria] | only if you buy it from them |
| 23:14.23 | Un1x | anyway there help thing says, join asterisk on freenode hehe |
| 23:14.27 | Un1x | i did... |
| 23:14.33 | Un1x | i bought from digium direct :/ |
| 23:14.34 | directory | you have installation support with the card, why don't you call? |
| 23:14.37 | twisted[asteria] | you bought the card |
| 23:14.42 | twisted[asteria] | you get install support |
| 23:14.44 | Un1x | because i cant dail the number ... |
| 23:14.54 | twisted[asteria] | you don't have a phone line? |
| 23:14.57 | Un1x | the number only allows us calls not, from canada ;/ |
| 23:14.58 | directory | erm, I cooked these too long |
| 23:15.12 | twisted[asteria] | parse error 271 |
| 23:15.35 | Un1x | directory; can you pointme to where i can create my own dailplan or already created one, ... |
| 23:15.52 | twisted[asteria] | ~docs |
| 23:15.53 | 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 |
| 23:16.03 | twisted[asteria] | have fun. |
| 23:16.57 | Un1x | :\ |
| 23:17.08 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 23:17.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:22.28 | *** join/#asterisk tenlet (n=tenlet@pool-141-153-215-248.mad.east.verizon.net) |
| 23:26.03 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 23:28.30 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 23:34.25 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 23:47.08 | SpaceBass | anyone registering with gizmo and doing inbound routing? |
| 23:50.09 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 23:50.11 | teknoprep | hey all |
| 23:50.19 | teknoprep | how do i change the codec used with freepbx? |
| 23:50.21 | teknoprep | on asterisk |
| 23:50.37 | teknoprep | or is that all client based? |
| 23:51.36 | hads|home | ~freepbx |
| 23:51.37 | jbot | rumour has it, freepbx is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 23:51.43 | teknoprep | sorry |
| 23:51.44 | teknoprep | wrong channel |
| 23:52.01 | hads|home | :) |
| 23:52.41 | SpaceBass | lol |
| 23:56.39 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 23:59.21 | Un1x | fuck man |
| 23:59.27 | Un1x | digium needs to get another number, |
| 23:59.31 | Un1x | that 1877 linuxme doesn't work |
| 23:59.50 | Un1x | i get a ring after a bit it say's `sorry youre call could not be completed as dailed' |
