| 00:00.04 | Bentley | hey all, does areskicc have any advantages over astcc (aside from the eye candy)? |
| 00:00.34 | tzanger | wtf is areskicc? |
| 00:00.35 | jerlique | redder86: do you mean that I need to setup another extension to my agent login extention with # as the argument to agentlogincallback ? |
| 00:01.01 | Bentley | tzanger: http://www.voip-info.org/wiki-AreskiCC+CallingCard+Application |
| 00:01.14 | Luke-Jr | Does anyone have MyPhoneCompany working thru Asterisk? |
| 00:01.14 | Mneumonic | can anyone help me? :) |
| 00:02.10 | bjohnson | Mneumonic: 1. ditch vonage as fast as you can |
| 00:02.17 | tzanger | amen to that |
| 00:02.36 | bjohnson | Mneumonic: 2. get 2 Sipura SPA 3000 units from voipsupply or voxilla |
| 00:02.44 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
| 00:03.23 | bjohnson | Mneumonic: 3. buy as many Sipura SPA 2000 units as phones you want (they will allow you to use any analog phone you want) OR buy voip phones |
| 00:03.37 | bjohnson | Mneumonic: 4. digital PBX phones will not work |
| 00:03.56 | tzanger | bjohnson: or forget all that and use a T100P+channel bank and save yourself from drowning in a sea of wall warts, cabling and heat dissipation |
| 00:04.07 | bjohnson | depends how many phones |
| 00:04.18 | tzanger | 24-port FXS Access Bank 1 + T100P will run you about US$750 |
| 00:04.27 | bjohnson | just noticed something .. he doesn't have anaolog lines .. he does not need the SPA 3000s |
| 00:04.30 | tzanger | and let you hook up 24 analog phones |
| 00:04.53 | Mneumonic | bjhonson - what other options do i have besides vonage? Also, the SPA 3000 units work with asterisk? |
| 00:05.18 | bjohnson | I was thinking 2 analog lines .. which would normally max out with about 10 handsets .. but those rules do not apply to a pure voip system |
| 00:06.06 | bjohnson | Mneumonic: the SPA 3000 units are 1 fxo and 1 fxs .. of course they will work with * but the system you describe does not need the fxo ports |
| 00:06.08 | tzanger | there are many providers. Personally I put all my calls through nufone -- they just work. there are others too but the more I've screwed around with them the more money I save just by terminating to nufone. Unfortunately they are *not* very new user friendly |
| 00:06.28 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 00:06.40 | ta[i]nted | does /outgoing/ respect any form of file locking? |
| 00:06.48 | *** part/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 00:06.48 | tzanger | ta[i]nted: eh? |
| 00:06.53 | tzanger | no I don't think so |
| 00:07.00 | ta[i]nted | asterisk is so aggressive at picking up files in /outgoing/ |
| 00:07.12 | Mneumonic | i just dont really *get* the FXS and FXO stuff... they connect to the asterisk server? how do analog phones know when to put the call on hold and all those features? |
| 00:07.28 | tzanger | ta[i]nted: that is why you move the files from another dir to there (keep it all in the same filesystem) since rename is an atomic operation on almost all linux filesystems |
| 00:07.29 | ta[i]nted | even if i'm in the middle of copying/moving a file into the directory |
| 00:07.36 | Luke-Jr | I just found a nice 1.3c/min outgoing-only provider :) |
| 00:07.57 | tzanger | Luke-Jr: use them for 30 days for at least 1000 minutes worth of time and come back and report |
| 00:08.15 | tzanger | ta[i]nted: you're likely moving across filesystems -- don't do that |
| 00:08.16 | Luke-Jr | tzafrir: I won't use 1000 minutes for at least 3 months... |
| 00:08.27 | tzanger | Luke-Jr: I'm saying to load test it before saying they're nice |
| 00:08.29 | Luke-Jr | tzafrir: Voipjet |
| 00:08.31 | ta[i]nted | tzanger good point, however, i'm putting the file into the directory from a samba share |
| 00:08.32 | shido6 | ok Mneumonic, its not that difficult :) |
| 00:08.42 | tzanger | I've found that the first few calls are great but when you run into issues it blows |
| 00:08.47 | Luke-Jr | tzafrir: I don't do loads; Just VoIP in my home :) |
| 00:08.50 | tzanger | voipjet seems to be alright |
| 00:08.58 | tzanger | you can get cheaper than that though |
| 00:09.02 | shido6 | Mneumonic FXS devices use FXO signalling and FXO devices use FXS signalling a handset that will plug into a FXS card |
| 00:09.04 | znoG | voipjet is damn cheap |
| 00:09.05 | Luke-Jr | cheaper than 1.3c/min?? |
| 00:09.11 | shido6 | your Co line from the telco plugs into a FXO card |
| 00:09.12 | ta[i]nted | tzanger what would you suggest, move file to a /tmp/ on the same filesystem and then cron job it into /outgoing/? |
| 00:09.13 | tzanger | and for your home, just get an all you can eat, it's usually cheaper for residential people |
| 00:09.14 | shido6 | with FXS signalling |
| 00:09.20 | tzanger | ta[i]nted: something like that, yes |
| 00:09.26 | ta[i]nted | that is ass |
| 00:09.29 | eipi | derkomissar any news? |
| 00:09.31 | tzanger | ta[i]nted: rename (move) is only atomic on the same filesystem |
| 00:09.40 | tzanger | ta[i]nted: or use fuser and a daemon |
| 00:10.14 | Luke-Jr | I just need to get my receiving provider to work :/ |
| 00:10.29 | ta[i]nted | Luke-Jr which provider? |
| 00:10.33 | Luke-Jr | MyPhoneCompany |
| 00:10.44 | Luke-Jr | seems to be the only one w/ POTS #s in Atchison, KS |
| 00:11.16 | Druken | POTS #'s? |
| 00:11.19 | bjohnson | Mneumonic: check the wiki for voip providers |
| 00:11.21 | bjohnson | ~docs |
| 00:11.22 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 00:11.39 | bjohnson | you will need to know what fxs do before you can use * |
| 00:11.48 | Luke-Jr | Druken: Like (913) 555-1212 |
| 00:12.05 | Druken | oh... you mean DID's :) |
| 00:12.10 | tzanger | i.e. a combination of bash's -N and fuser to move files that have nobody touching them to /outgoing/ |
| 00:12.15 | Luke-Jr | Druken: yea |
| 00:12.50 | Druken | Luke-Jr: my city is unserviced as well, do don't feel bad |
| 00:13.10 | Luke-Jr | Druken: Well, I already paid for the service... just can't get it to register me |
| 00:13.42 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 00:13.57 | ta[i]nted | tzanger i don't see how that would be better than a script running to move files |
| 00:15.13 | Druken | Luke-Jr: oh, so you do have service... you just can't figure out how to interface with it :) got ya |
| 00:15.40 | Luke-Jr | Druken: right |
| 00:15.57 | *** join/#asterisk dolson (~dana@Sudbury-HSE-ppp3980676.sympatico.ca) |
| 00:16.02 | Druken | i can't even get local numbers |
| 00:16.14 | liquidno2 | damn it |
| 00:16.28 | tzanger | ta[i]nted: it would be a script that does it |
| 00:16.41 | liquidno2 | why am I getting double console entries for off and on hook events in the asterisk console |
| 00:17.00 | dolson | is there a variable that returns the duration of the call upon hangup? I've been reading the mailing list and wiki and I don't see it (apart from cdr, but I just want it echoed with noop in the console) |
| 00:17.12 | tzanger | just a while [ 1 ] ; do if -N ... ; then for a in ls /var/spool/asterisk/tmp/* ; do fuser $a && if $? -eq 0 then mv... |
| 00:17.20 | tzanger | sleep 1 inbetween and it'll take zero CPU |
| 00:17.23 | Druken | liquidno2: do you have it echoing to consol as well as asterisk -r ? |
| 00:17.44 | liquidno2 | Druken: how can I verify that? |
| 00:17.55 | liquidno2 | I started asterisk in background mode... |
| 00:17.59 | liquidno2 | so you might be right... |
| 00:18.06 | Druken | ctl c, then see if it still echo's :) |
| 00:18.22 | liquidno2 | somebitch |
| 00:18.25 | liquidno2 | :) |
| 00:18.28 | liquidno2 | danke |
| 00:18.36 | Druken | hehehe |
| 00:18.49 | ta[i]nted | what's if -N? |
| 00:19.16 | liquidno2 | ta[i]nted in reference to what? shell script? |
| 00:19.29 | ta[i]nted | liquidno2 yea |
| 00:20.16 | *** join/#asterisk chaoscon (stormy@chaoscon.user) |
| 00:20.19 | liquidno2 | per my bash cheeat sheet |
| 00:20.31 | Luke-Jr | http://pastebin.ca/5432 <-- can anyone see any obvious problems? |
| 00:20.43 | liquidno2 | "-N file: true if file exists and was modified since last read |
| 00:20.51 | Luke-Jr | particularly with the sipmedia/MyPhoneCompany part |
| 00:21.20 | Druken | Luke-Jr: yeah... are you behind a nat ? |
| 00:21.55 | Luke-Jr | Druken: yes and no |
| 00:22.02 | Druken | ok..... |
| 00:22.06 | Luke-Jr | Druken: eth0 is behind a NAT w/ DMZ; eth1 is direct |
| 00:22.19 | tzanger | ta[i]nted: returns true if file/dir has changed since last read |
| 00:22.24 | Druken | so the asterisk box is the gateway ? |
| 00:22.29 | Luke-Jr | No |
| 00:22.51 | Luke-Jr | eth1 goes from my cable modem to the USB on the * box |
| 00:23.00 | Luke-Jr | eth0 is from cable modem -> WRT54G -> * box |
| 00:23.08 | Druken | usb.... |
| 00:23.14 | liquidno2 | wow what in the hell |
| 00:23.16 | Luke-Jr | Only got one Ethernet port |
| 00:23.20 | *** part/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 00:23.23 | liquidno2 | uhm |
| 00:23.26 | liquidno2 | but why? |
| 00:23.39 | Luke-Jr | I use eth1 for VoIP, generally |
| 00:23.39 | Druken | Luke-Jr: so spend 20 bux and get a second card |
| 00:23.44 | Qwell | You have two links to the modem for your * box? |
| 00:23.53 | Luke-Jr | Druken: the cable modem also only has 1 ethernet jack |
| 00:23.57 | Luke-Jr | Qwell: yes |
| 00:24.01 | liquidno2 | Luke-Jr: do this |
| 00:24.01 | Qwell | umm, why? |
| 00:24.11 | Luke-Jr | Qwell: round-robins? |
| 00:24.14 | liquidno2 | Luke-Jr: connect two ethernet ports to the WRT54G |
| 00:24.28 | liquidno2 | put a NAT IP on teh second ethernet card |
| 00:24.46 | liquidno2 | and then do a dmz on that IP |
| 00:24.53 | Luke-Jr | liquidno2: The USB is working just fine.... >.> |
| 00:25.03 | Luke-Jr | unless I unplug it |
| 00:25.10 | Druken | then why do you need the router? |
| 00:25.11 | Luke-Jr | then it screws my routing tables |
| 00:25.17 | Luke-Jr | Druken: other LAN systems |
| 00:25.43 | Druken | i follow, but it's fucked up |
| 00:25.43 | Luke-Jr | ... |
| 00:25.56 | liquidno2 | I am sorry man... but that is one messed config |
| 00:25.58 | jerlique | Is there a way to change the voicemail greeting, so it doesnt say "wlecome to comedian voicemail" |
| 00:26.21 | Luke-Jr | liquidno2: also, each IP gets seperate bandwidth caps; does that make more sense? |
| 00:26.21 | Druken | yeah, replace the gsm file |
| 00:26.22 | Druken | hehehe |
| 00:26.39 | liquidno2 | So install the sveasoft firmware and cap at the vlan |
| 00:26.47 | liquidno2 | that is how I run my network at the house |
| 00:27.09 | liquidno2 | works great and I don't have some magic mojo configuration with USB cables and stuff |
| 00:27.10 | Luke-Jr | liquidno2: USB-IP gets one 3MBit cap and Router gets a seperate 3MBit cap |
| 00:27.27 | Qwell | Luke-Jr: Is that "supposed" to work? |
| 00:27.29 | liquidno2 | HAH a /32 block? :) |
| 00:27.35 | Luke-Jr | Qwell: what? |
| 00:27.48 | Druken | liquidno2: no... a block of 32 |
| 00:27.57 | Druken | 29 useable ip's.. |
| 00:27.57 | Qwell | My ISP threw a fit when I was getting 2 IPs using a little trick I found, heh |
| 00:28.05 | liquidno2 | oh... So a /27 |
| 00:28.15 | Sedorox | wha little trick? share! |
| 00:28.16 | Sedorox | :-p |
| 00:28.25 | Druken | Qwell: oops :) |
| 00:28.33 | Qwell | Sedorox: Earthlink DSL, a pppoe dialer, and a hub |
| 00:28.42 | Sedorox | lol |
| 00:28.45 | Qwell | If you connect both dialers at the right time, you'll get two seperate IPs |
| 00:28.48 | liquidno2 | nice |
| 00:28.52 | Sedorox | hehe |
| 00:28.54 | Luke-Jr | Qwell: I doubt it's actually true, but I'm using it to justify my configuration ;) |
| 00:29.26 | Qwell | I got a call about 45 minutes later...they weren't happy. |
| 00:29.42 | liquidno2 | hehe, surprised the even noticed |
| 00:29.50 | Druken | Qwell: i would have told them tough shit, fix your broken config :) |
| 00:29.52 | liquidno2 | somebody must have been watching the DHCP logs |
| 00:29.58 | *** join/#asterisk _daver_ (~daver@ns1.tmok.com) |
| 00:30.09 | Qwell | so were we. Hell, we were surprised it worked. We just kinda "found" it, on accident. We were both new to networking, and assumed we had it connected right, heh |
| 00:30.16 | Luke-Jr | Druken: LOL... all cable ISPs have broken configs ;) |
| 00:30.19 | file[laptop] | fyi I can connect multiple PPPoE sessions with my ISP... |
| 00:30.29 | file[laptop] | most I've had is 8 sessions, 8 IPs |
| 00:30.30 | Qwell | file[laptop]: "allowed" to? |
| 00:30.40 | file[laptop] | they tried to block it but ending up bringing the entire province to a halt |
| 00:30.41 | liquidno2 | Yeah... I encoutnered that today with a Verizon customer |
| 00:30.41 | Druken | Luke-Jr: PPPOE isn't cable |
| 00:30.42 | Qwell | see, Earthlink reamed me |
| 00:30.58 | liquidno2 | she had a Westel router... and was also connected via PPPoE on her PC |
| 00:31.00 | file[laptop] | haven't fixed it in... oh... must be two years |
| 00:31.16 | Qwell | file[laptop]: similar setup to what I described? |
| 00:31.31 | file[laptop] | single ADSL modem... just run different PPPoE sessions on the systems |
| 00:31.34 | Luke-Jr | anyway, what is wrong with my config and a NAT? |
| 00:31.43 | file[laptop] | I can even do it on Linux if I wanted... multiple PPPoE sessions... |
| 00:31.49 | Nugget | NAT blows goats. |
| 00:31.55 | file[laptop] | routing goes a little evil though |
| 00:31.57 | Qwell | with ours, we actually had to click "dial" at specific times, or it would fail authentication on the second one |
| 00:31.59 | liquidno2 | Buit wouldn't that be easily fixed with a radius tweak? |
| 00:32.03 | Qwell | and we used two different systems |
| 00:32.10 | liquidno2 | I know we did that with our radius |
| 00:32.16 | znoG | anyone else's messenger just die? |
| 00:32.16 | file[laptop] | my ISP tried... |
| 00:32.18 | liquidno2 | block dual logins |
| 00:32.26 | liquidno2 | it is a one liner in freeradius |
| 00:32.29 | Qwell | liquidno2: Thats what earthlink did. He's the problem though |
| 00:32.31 | file[laptop] | they tried two times |
| 00:32.35 | file[laptop] | both times failed ;) |
| 00:32.40 | liquidno2 | amateur night |
| 00:32.49 | Qwell | I click connect, it says, "Hey, here I am", I click connect again, says, "Hey, here I am" |
| 00:32.56 | liquidno2 | although I am sure their radius servers were more complex than ours... |
| 00:32.57 | liquidno2 | :) |
| 00:32.59 | Qwell | first connection gets authenticated, then the second connection does |
| 00:33.17 | Luke-Jr | liquidno2: Druken: would you mind telling me what was broken w/ NAT in my config? even if my setup is screwy? =p |
| 00:33.18 | Qwell | it ONLY worked if the second client connected while the first was authenticating |
| 00:33.33 | liquidno2 | you nkow... |
| 00:33.34 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 00:33.37 | Druken | liquidno2: they probably use radiator |
| 00:33.38 | liquidno2 | that actaully makes sense |
| 00:33.57 | liquidno2 | since you have yet be authenticated... |
| 00:33.58 | Qwell | znoG: yeah |
| 00:34.02 | liquidno2 | you aren't added to the raddb |
| 00:34.06 | Qwell | liquidno2: right, and I'm already in the queue |
| 00:34.13 | Luke-Jr | znoG: MSN did; normal IM didn't :) |
| 00:34.13 | liquidno2 | so a second one... man |
| 00:34.14 | Qwell | so it probably checked at the start |
| 00:34.18 | liquidno2 | that has to be a bug :) |
| 00:34.22 | Qwell | liquidno2: surely |
| 00:34.23 | liquidno2 | yeah |
| 00:34.35 | liquidno2 | makes perfect sense when you think about it though |
| 00:34.41 | Qwell | but yeah, they were flaming mad when they called |
| 00:35.07 | liquidno2 | I would have said "give me a /29 and I won't publish it on my blog" |
| 00:35.10 | Qwell | it was just a standard Fujistu modem that you would get with Verizon |
| 00:35.15 | Luke-Jr | LOL |
| 00:35.29 | liquidno2 | I don't have a blog, but they don't know that :) |
| 00:35.36 | Qwell | then we had the modem connected to the uplink on the modem, and PC1 and PC2 were connected from there |
| 00:35.43 | Qwell | uplink on the hub, rather |
| 00:36.01 | liquidno2 | I use to scerw with BellSouth |
| 00:36.23 | liquidno2 | DHCP RFC specifies that the request can request their own IP address |
| 00:36.34 | liquidno2 | as long as it isn't taken |
| 00:36.44 | *** join/#asterisk freat[laptop]_ (~freat[lap@147.126.10.251) |
| 00:36.46 | liquidno2 | the DHCP server should honor the request |
| 00:36.49 | tzanger | liquidno2: yeah but half the PPPoE servers ensure you do NOT get your same IP |
| 00:36.54 | liquidno2 | So I had the same IP for 2 years :) |
| 00:37.03 | tzanger | it's not a requirement that they follow that |
| 00:37.04 | Luke-Jr | liquidno2: I don't think that's unusual |
| 00:37.27 | mgeorge | I am having some troubles with kphone and "maximum retires exceeded" errors. Anyone familiar w/ SIP configuration? |
| 00:37.31 | Luke-Jr | liquidno2: to abuse that, you should have requested some easy-to-remember IP |
| 00:37.35 | Derkommissar | im looking for a fake ring, anyone here have a ring.wav or .gsm or .mp3 ? |
| 00:37.36 | liquidno2 | Well it worked right up until they hacked up the DHCP server to not follow the spec and started rejecting my requests :) |
| 00:38.12 | Qwell | to go back to my point, heh |
| 00:38.12 | Luke-Jr | A few years ago, I set my IP to 68.9.80.80 (static) and it worked fine :) |
| 00:38.23 | liquidno2 | Derkommissar: I have one |
| 00:38.23 | Qwell | Luke-Jr: Are you sure your ISP supports multiple connections to the modem? |
| 00:38.29 | liquidno2 | Derkommissar: hold on |
| 00:38.41 | mgeorge | I am getting the chan_sip maximum retries exceeded and any click on a button on the kphone interface will take over a minute to do anything. |
| 00:38.44 | Luke-Jr | Qwell: My ISP doesn't support my modem at all; that's not related |
| 00:38.46 | Derkommissar | thanls |
| 00:38.52 | liquidno2 | Derkommissar: http://slim.expresshosting.net/In.wav |
| 00:39.02 | buddah | anyone know how to upgrade firmware for cisco ata-186? |
| 00:39.05 | Qwell | Luke-Jr: So get a splitter, and connect two modems. :P |
| 00:39.11 | Luke-Jr | Qwell: Generally, I restrict VoIP to eth1, but I don't know how to do that w/ * |
| 00:39.14 | mgeorge | However,w hen calling kphone from the * box, it works fine and dtmf will send fine. A hangup, though, takes very long... |
| 00:39.18 | Luke-Jr | Qwell: 2 modems costs twice the price |
| 00:39.26 | _daver_ | You're better off just getting a business cable connection and stop worrying about DHCP IP's. |
| 00:39.26 | shido6 | well each interface has an ip |
| 00:39.28 | Manipura | Other than popularity, anyone know of disadvantages for using IAX2? |
| 00:39.31 | shido6 | u can set the ip to listen on |
| 00:39.35 | Luke-Jr | _daver_: Not offered here |
| 00:39.35 | liquidno2 | Derkommissar: that work for you? |
| 00:39.47 | shido6 | in the iax.conf or sip.conf or h323.conf or mgcp.conf whatever protocol you are using |
| 00:39.50 | Luke-Jr | shido6: The problem is the IP to send registrations on |
| 00:40.00 | Cresl1n | Hey!!!! ;-) |
| 00:40.04 | shido6 | ? |
| 00:40.06 | _daver_ | Luke: Ah, too bad. I'm in Cox territory. A business cable connection for a "home office" is $69/month. |
| 00:40.15 | liquidno2 | man |
| 00:40.19 | liquidno2 | cheaper than TW |
| 00:40.21 | Luke-Jr | _daver_: that's too expensive for me, anyway |
| 00:40.26 | shido6 | Luke-Jr what do you mean? you're going to accept registrations on a different ip then placing the call from? |
| 00:40.27 | liquidno2 | they want $99 for a business |
| 00:40.31 | _daver_ | for 5 megabits down/512k up. |
| 00:40.39 | _daver_ | i have 3 additional static IP's so I'm paying $99. |
| 00:40.42 | Luke-Jr | shido6: no |
| 00:40.54 | Luke-Jr | shido6: I mean the binding doesn't affect the source IP of registrations, from what I can tell |
| 00:41.19 | Luke-Jr | _daver_: I have 6MBit for ~$40/mo |
| 00:41.28 | Luke-Jr | _daver_: advertised at 3MBit, but the config they send my modem is 6MBit... |
| 00:41.33 | _daver_ | Luke: Nice. ;) |
| 00:41.42 | Luke-Jr | so I don't bother uncapping since I'm happy w/ it |
| 00:42.07 | liquidno2 | I want to upgrade just to get the 768K up... |
| 00:42.24 | liquidno2 | I almost did... but then all my favorite BT sites went down :) |
| 00:42.30 | netsurfer | lol @ uncapping.. thats like getting $10 of gas and paying for $5 |
| 00:42.45 | shido6 | wow |
| 00:42.47 | redder86 | is there a Bug marshall here? I need http://bugs.digium.com/bug_view_page.php?bug_id=0003518 reopened, and I'm not the original poster. |
| 00:42.56 | tzanger | uncapping's great until they cut you off |
| 00:43.59 | Luke-Jr | tzafrir: cut me off how? I could uncap and get service if I wasn't a customer... |
| 00:44.07 | Luke-Jr | their loss if they cut me off |
| 00:44.26 | *** join/#asterisk stibbons (~peter@c220-237-17-139.belrs3.nsw.optusnet.com.au) |
| 00:44.30 | tzanger | Luke-Jr: they have a thousand people who aren't pissing about with their business model, they won't even miss you |
| 00:44.57 | Luke-Jr | tzafrir: I can bypass their unregistration of my account, though |
| 00:45.11 | Luke-Jr | tzafrir: besides, as I said, it's not worth my trouble to uncap |
| 00:45.18 | tzanger | Luke-Jr: then they'll just have you arrested for theft of service |
| 00:45.22 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 00:45.25 | tzanger | really, how far do you want to take it? |
| 00:45.36 | tzanger | exactly -- it's not worth the trouble |
| 00:45.41 | _daver_ | don't fuck with the cable company, especially if they're the only decent broadband provider in town. ;) |
| 00:45.55 | liquidno2 | Yeah... |
| 00:46.03 | liquidno2 | most places do have overlay though |
| 00:46.19 | eipi | anyone from SouthAmerica??? |
| 00:46.22 | znoG | eipi: yo |
| 00:46.51 | _daver_ | around here (rhode island) cable outnumbers DSL by like 10 to 1. |
| 00:46.55 | liquidno2 | If I want one context to be completely seperate from the other context in the system... I just assign that channel it its own context right? |
| 00:47.02 | tzanger | liquidno2: yup |
| 00:47.09 | liquidno2 | Thought so |
| 00:47.29 | liquidno2 | the wife complained about dialing 9 to get an outside line |
| 00:47.34 | liquidno2 | damn her |
| 00:47.37 | netsurfer | lmao |
| 00:47.40 | Luke-Jr | the way I see it, if they don't mess with me (eg: ignore my use of an unauthorized modem), I won't mess with them (eg: uncap) |
| 00:47.41 | tzanger | liquidno2: ewhy the hell would you have that at home anyway |
| 00:47.44 | netsurfer | lazy bi@tch |
| 00:47.45 | Beave | haha. |
| 00:47.51 | Beave | my wife complains about the same thing. |
| 00:47.56 | tzanger | Luke-Jr: you sound like your 13 |
| 00:47.58 | liquidno2 | tzanger: home based business |
| 00:48.02 | Luke-Jr | tzafrir: ... |
| 00:48.06 | liquidno2 | I have ring detection |
| 00:48.09 | tzanger | liquidno2: again, why bother with 9 :-) |
| 00:48.14 | liquidno2 | so I push my business line to the asterisk |
| 00:48.18 | Luke-Jr | tzafrir: I don't like other people controlling my stuff |
| 00:48.23 | Beave | I set it up so she can dial NPA+thenumber. |
| 00:48.28 | Beave | she's okay with that, for some reason. |
| 00:48.29 | tzanger | Luke-Jr: you're on their network. If you don't like it, start up your own. |
| 00:48.45 | liquidno2 | tzanger: well I have other extensions in teh system |
| 00:48.49 | tzanger | Beave: I rather like 10-digit dialing, and have * insert the 1 in front of anything that's long distance for me |
| 00:48.50 | _daver_ | Luke: your provider doesn't authenticate modems based on their RF MAC address or anything? |
| 00:49.00 | tzanger | liquidno2: meh, use *exten or something then :-) |
| 00:49.01 | liquidno2 | remote KVM over IP and stuff like that |
| 00:49.07 | Beave | tzanger: nice. |
| 00:49.07 | Luke-Jr | _daver_: they do |
| 00:49.16 | jsolares | anyone here use AMP for setting up asterisk? i'm having a bit of a problem getting CDR to work |
| 00:49.23 | liquidno2 | so I can dial the PBX and then connect to the KVM if my office is offline |
| 00:49.37 | Luke-Jr | tzafrir: I'm on their network; as long as I obey their rules, it shouldn't matter that i have control of my equipment |
| 00:49.37 | netsurfer | _daver_ - just reprog the flash chip ;) |
| 00:49.54 | tzanger | Luke-Jr: so long as you follow the TOS they had you sign, absolutely |
| 00:50.01 | tzanger | Luke-Jr: they won't give a shit if you're following their TOS |
| 00:50.07 | Luke-Jr | tzafrir: Lucky my ISP is contractless |
| 00:50.13 | Luke-Jr | I signed nothing |
| 00:50.19 | tzanger | Luke-Jr: but uncapping or "repurposing" an old modem is hardly allowed under their TOS, I'd imagine |
| 00:50.29 | Luke-Jr | tzafrir: I'm not uncapping |
| 00:50.31 | tzanger | Luke-Jr: yeah so they'd just cut you off then |
| 00:50.55 | tzanger | like I said, they have a thousand customers that aren't giving them grief, so if they don't get your $40/mo I doubt they'd care |
| 00:51.34 | liquidno2 | I am the biggest PITA to my local cable company... |
| 00:51.44 | liquidno2 | the instant my internet connection goes down... I call them |
| 00:52.02 | liquidno2 | They don't even bother to trouble shoot me anymore :) |
| 00:52.04 | terrapen | my cablemodem is solid as hell |
| 00:52.06 | liquidno2 | they just send a tech :) |
| 00:52.16 | terrapen | i can't remember it ever going down |
| 00:52.26 | terrapen | my power goes out more. |
| 00:52.28 | tzanger | liquidno2: sounds like you're saving them money in the long run |
| 00:52.28 | liquidno2 | mine was really good for about 6-8 months |
| 00:52.34 | Luke-Jr | terrapen: power out != cable out for me :) |
| 00:52.37 | liquidno2 | tzanger: probably |
| 00:52.51 | terrapen | i could put a UPS on my Soekris gateway |
| 00:52.55 | *** join/#asterisk chaoscon_ (stormy@chaoscon.user) |
| 00:52.55 | terrapen | but not much of a point |
| 00:53.03 | liquidno2 | every once in a while they will add a stub off the cable loop in my neighborhood and I will know it :) |
| 00:53.04 | terrapen | if the power goes out, i read a book |
| 00:53.06 | terrapen | or go for a run |
| 00:53.42 | liquidno2 | I am pretty sure I am the only high speed customer in my neighborhood |
| 00:54.00 | liquidno2 | cause everytime I call... they are like "Oh we didn't know there was a problem... we will send somebody out." |
| 00:54.26 | liquidno2 | and most of the time the tech doesn't even come to the house... it is normally upstream somewhere |
| 00:54.32 | netsurfer | liquidno2 - nah they are just BSing |
| 00:54.40 | liquidno2 | netsurfer: could be |
| 00:54.47 | liquidno2 | although I am pretty back woods here |
| 00:55.00 | tzanger | meh |
| 00:55.05 | tzanger | I'm being joe-jobbed on asterisk-users |
| 00:55.16 | tzanger | someone's spamming russia from akohlsmith-asterisk@benshaw.com |
| 00:55.27 | netsurfer | my DSL has been fine since I got it in sept.. but this last week its been off/on every night |
| 00:55.28 | terrapen | I wonder how my little Soekris box would do as a * server, doing SIP <--> IAX2 |
| 00:55.32 | tzanger | so -users gets the bounces and I get a probe message |
| 00:55.40 | terrapen | does that use much CPU? (SIP <--> IAX2) |
| 00:55.47 | tzanger | terrapen: probably pretty good, that's not a heavy CPU task I don't think |
| 00:55.47 | jsolares | bleh, it was the mysql-devel package that was missing |
| 00:55.48 | liquidno2 | tzanger: that is why the asterisk list goes to the trash in thunerbird and I read it from there :) |
| 00:55.59 | tzanger | liquidno2: heh |
| 00:56.15 | liquidno2 | man that is a high volume list |
| 00:56.20 | tzanger | yes, yes it is |
| 00:56.23 | netsurfer | brb.. hopefully |
| 00:56.31 | tzanger | the only higher-volume list I was regularly reading was lkml |
| 00:56.43 | liquidno2 | not familiar with that one |
| 00:56.51 | liquidno2 | linux kernel |
| 00:56.53 | liquidno2 | ? |
| 00:56.54 | tzanger | yup |
| 00:56.59 | tzanger | I was on it about 5 years ago |
| 00:57.00 | liquidno2 | ah never seen that notation before |
| 00:57.10 | Luke-Jr | liquidno2: it's pretty common... |
| 00:58.55 | terrapen | i wonder what codec my Polycom IP500 is using by default |
| 00:59.00 | *** join/#asterisk VoIPMasta (~John@201.133.111.242) |
| 00:59.03 | VoIPMasta | Hi |
| 00:59.10 | *** join/#asterisk dizzydiffi (dizzydiffi@adsl-70-240-241-42.dsl.hstntx.swbell.net) |
| 00:59.13 | modulus_ | jbot babelfish de en die kinderszenen |
| 00:59.20 | terrapen | shit...how do i tell? |
| 00:59.24 | VoIPMasta | I need someone in the USA to help me test some toll free dids, it won't cost you anything to dial :) |
| 00:59.26 | *** join/#asterisk netsurfer (netsurfer@82.133.113.162) |
| 00:59.33 | dizzydiffi | Hello people, I have a question on OpenH323 |
| 00:59.39 | Luke-Jr | VoIPMasta: ? |
| 00:59.42 | terrapen | voip, sure |
| 00:59.43 | Beirdo | jbot who is your owner? :) |
| 00:59.45 | jbot | I think you lost me on that one, Beirdo |
| 00:59.48 | Beirdo | heh |
| 00:59.52 | dizzydiffi | have anyoone installed OpenH33 |
| 01:00.41 | VoIPMasta | terrapen: can I send you the number through query (pm)? |
| 01:00.56 | ariel_ | hello everyone. |
| 01:01.25 | terrapen | sure...i hope i don't have to do very much |
| 01:01.31 | terrapen | because im really lazy |
| 01:02.36 | liquidno2 | what is a standard wait time before handing over to Voicemail? |
| 01:02.39 | liquidno2 | 20 seconds? |
| 01:02.47 | liquidno2 | or 15? |
| 01:03.09 | tzanger | liquidno2: I don't think there's a standard |
| 01:03.14 | tzanger | I usually leave 4 rings which is 20s |
| 01:03.18 | liquidno2 | Okay. |
| 01:03.26 | liquidno2 | I guess that should have been the question :) |
| 01:03.55 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
| 01:03.57 | *** join/#asterisk file (~symlink@mctn1-1056.nb.aliant.net) |
| 01:04.04 | bjohnson | hmm .. e164.org has to call numbers to register them and I'm not there!! Time for a softphone extension |
| 01:05.09 | *** join/#asterisk ccfiel (~chatzilla@210.213.139.142) |
| 01:06.37 | bjohnson | anything like e164.org that would work as a global follow me system? |
| 01:08.05 | tzanger | suh-weet |
| 01:08.10 | tzanger | trolltech is allowing GPL win32 apps |
| 01:08.17 | tzanger | I am a very happy camper |
| 01:09.38 | tzanger | ok, now I *know* you must be around 13 years old |
| 01:09.46 | Luke-Jr | nope |
| 01:10.05 | tzanger | I enjoy and use Linux as much as the next guy but I'm simply not that naive |
| 01:10.23 | Luke-Jr | Sarcasm. |
| 01:10.29 | liquidno2 | anybody that does anything with IT has to know that windows is the defacto standard for the desktop |
| 01:10.53 | liquidno2 | I shudder at the mear thought of some of my users using linux on the desktop |
| 01:11.08 | Luke-Jr | liquidno2: why? |
| 01:11.21 | tzanger | I would actually enjoy that |
| 01:11.31 | tzanger | my bank uses OS/2 |
| 01:11.37 | Luke-Jr | O.o |
| 01:11.38 | Beave | hehe. |
| 01:11.40 | liquidno2 | I would to, but the bell curve is fighting against me |
| 01:12.02 | hermie | anybody here use AMP? |
| 01:12.03 | tzanger | then again I don't technically work in IT |
| 01:12.22 | liquidno2 | hermie: I tried installing it in Debian... and it went badly |
| 01:12.52 | liquidno2 | the directions are highly RH/etc centric |
| 01:13.18 | liquidno2 | and near as I can tell spandsp won't compile under debian |
| 01:13.36 | tzanger | liquidno2: I find that unbelievably hard to believe |
| 01:13.38 | *** join/#asterisk ionix (ionix@MTL-HSE-ppp186113.qc.sympatico.ca) |
| 01:13.46 | netsurfer | lol, I dont |
| 01:13.50 | netsurfer | debian is a bitch |
| 01:13.53 | liquidno2 | at least it wouldn't on unstable... and I tried 5 versions of cpp from 2.95 to 3.4 |
| 01:13.54 | netsurfer | :P |
| 01:14.21 | liquidno2 | I prefer Debian over RH... but I manage more RH servers |
| 01:14.31 | liquidno2 | so I know where my bread is buttered :) |
| 01:14.31 | tzanger | hahaha |
| 01:14.39 | tzanger | apparently the photo fo the day by shutterline has stopped in November |
| 01:15.02 | liquidno2 | Oh and todays geek word of the day: badmin |
| 01:15.06 | _daver_ | I'm using ArchLinux. It's a rather minimalist BSD-like distro... |
| 01:15.16 | tzanger | I'm a staunch slackware supporter |
| 01:15.22 | *** join/#asterisk OzJames79 (~James@203.208.64.29) |
| 01:15.23 | Beave | Slackware and Gentoo .... |
| 01:15.30 | liquidno2 | "That guy sucks. He is such a badmin." |
| 01:15.43 | netsurfer | eek... now the distro wars have started |
| 01:15.44 | Beave | I tend to try to stay away from RH, as I hate it - but if the client wants it.. oh well. |
| 01:15.44 | _daver_ | I first installed Slackware in 1994, on a pre-1.0 kernel. .99.11, I think. ;) |
| 01:15.44 | tzanger | liquidno2: ugh that sounds like something out of that terrible book "Sniglets" |
| 01:15.57 | tzanger | _daver_: :-) I started in '96 |
| 01:16.05 | tzanger | christ 9 years |
| 01:16.12 | Beave | _daver_: So you're new to slackware. I was a SLS user pre-slackware days.. |
| 01:16.19 | Beave | j/k about the new part. |
| 01:16.20 | OzJames79 | Hi all when i make a call asterisk dies ... anyone have any idea its a fresh install of debian http://www.pastebin.com/238975 |
| 01:16.21 | _daver_ | Beave: I ran SLS before that. ;) |
| 01:16.27 | _daver_ | It came on 12 floppies, I think. |
| 01:16.39 | Beave | SLS later was addoptied by Pat and became slackware. |
| 01:16.43 | _daver_ | I recall install it on a 386SX-16 with 4 megs of RAM. |
| 01:16.44 | Luke-Jr | netsurfer: 5.2 here |
| 01:16.47 | hermie | is anybody here mogorman? |
| 01:16.53 | Beave | yeah.. SLS.. boy, those where the days! |
| 01:17.00 | _daver_ | I couldn't run X... but everything else ran great! |
| 01:17.04 | tzanger | OzJames79: your problem is that asterisk segfaulted |
| 01:17.17 | tzanger | OzJames79: first things first, go burn yourself a memtest86 CD or floppy and boot from it |
| 01:17.23 | liquidno2 | I actually ran QNX in 1995 |
| 01:17.30 | tzanger | give it a half hour. If it comes back clean look for a dead fan or something |
| 01:17.34 | _daver_ | Before that, I messed around with Coherent Unix. |
| 01:17.40 | liquidno2 | and we migrated to Minix shortly after that |
| 01:17.47 | tzanger | I played with qnx... it still is a pretty nice little RTOS |
| 01:18.07 | dizzydiffi | Hello has anyone installed OpenH33 |
| 01:18.07 | liquidno2 | It was for an optical imager |
| 01:18.10 | tzanger | Linux and its RT patches are, IMO, eating its lunch though... QNX still has superior support though |
| 01:18.16 | liquidno2 | god I forget what for... |
| 01:18.19 | tzanger | dizzydiffi: no |
| 01:18.19 | Beave | I was more of a VMS guy before moving to Unix in the late 80's/early 90s. |
| 01:18.20 | dizzydiffi | I need a little help |
| 01:18.23 | liquidno2 | something at a doctors office I think |
| 01:18.34 | Beave | I'm still a VMS guy, but it doesnt pay any bills - obviously. |
| 01:18.35 | OzJames79 | thank you |
| 01:18.41 | tzanger | OzJames79: I also notice that you are running with broadvoice |
| 01:18.41 | liquidno2 | Digital all the way! :) |
| 01:18.52 | tzanger | are you sure you installed the patch they require properly? |
| 01:18.55 | _daver_ | I have an AlphaStation 400 here, that was running OpenVMS until its power supply died. ;) |
| 01:19.01 | Beave | http://deathrow.vistech.net is one of my OpenVMS projects. |
| 01:19.07 | _daver_ | VMS is an interesting OS... |
| 01:19.10 | tzanger | if the patch is installed wrong I could see sigsegvs being a symptom |
| 01:19.11 | Beave | public openvms cluster.. |
| 01:19.16 | liquidno2 | I miss my Alpha 2100 |
| 01:19.24 | liquidno2 | the washing machine we called it |
| 01:19.30 | dizzydiffi | so no one has installed OpenH323 |
| 01:19.34 | Beave | I just got a DS10L (Alpha). Its a 1U rack mount box. rocks. |
| 01:19.43 | tzanger | dizzydiffi: not me. I avoid h323 even more than I avoid SIP |
| 01:19.45 | Beave | It's going into the cluster soon. |
| 01:19.53 | _daver_ | I worked at an early Internet provider that ran its whole operation off of a MicroVAX II back in the 1989/90... |
| 01:19.59 | liquidno2 | that is what I really liked about openvms |
| 01:20.00 | tzanger | dizzydiffi: I ran h323 about 3 or 4 years ago and it was sucked the hind tit back then, too |
| 01:20.13 | liquidno2 | it all integrated and clusteres so well |
| 01:20.33 | liquidno2 | I mean you can access hard drives on different machines all over the network |
| 01:20.33 | Beave | Clustering rocks in vms.... Dave: I've owned my share if uVAX II's.. heh. |
| 01:20.40 | liquidno2 | even striping them and stuff... |
| 01:20.49 | tzanger | liquidno2: nifty.. I have to play with Andrewfs more |
| 01:20.53 | liquidno2 | just flat out crazy stuff that modern OS can't do right |
| 01:20.59 | tzanger | (my name is Andrew too, incidentally) |
| 01:20.59 | Beave | liquidno2: yep. Its even nicer if you have shared scsi. |
| 01:21.03 | _daver_ | In 1993 they finally upgraded to a VAXStation 4000 ;) |
| 01:21.04 | liquidno2 | and this was 20 years ago |
| 01:21.27 | liquidno2 | tzanger: I like afs |
| 01:21.35 | liquidno2 | used it in a mail cluster I made |
| 01:21.41 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 01:21.57 | liquidno2 | 350GB of AFS storage |
| 01:21.59 | Beave | We have a Alpha 500/333, uVAX 3100, and a emulated box (Via SIMH on intel hardware). I have a brand new DS10L going in soon. |
| 01:22.42 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 01:22.48 | liquidno2 | Beave: I think I ahve been to your site when it was /.ed |
| 01:22.50 | dontmsgme | Does Diamondcard.us let you set CID |
| 01:23.08 | _daver_ | Beave: I just logged on to one of your boxes under a demo account. It's bringing back memories. ;) |
| 01:23.12 | Beave | liquidno2: yeah, that was a while back. |
| 01:23.41 | Beave | _daver_: yeah - go for it. Hit dahmer.vistech.net under the demo account.. Obviously the Alpha is much faster than the vaxen. |
| 01:23.57 | _daver_ | Beave: Yeah, that's the one I'm on. |
| 01:24.04 | liquidno2 | VAX gets mega uber geek points though :) |
| 01:24.06 | Beave | whats great, I almost never, never have to touch that cluster. |
| 01:24.08 | dontmsgme | Does Diamondcard.us let you set CID |
| 01:24.27 | SuperAlex | asterisk = teh win :D |
| 01:24.35 | tzanger | I love reading boren's diary on k5 |
| 01:24.37 | Beave | stable.. and just works. now if I could port asterisk to it.. heh. |
| 01:24.45 | tzanger | what his stories don't provide is more than made up by someof the comments |
| 01:25.01 | tzanger | he talks about trying to save a rabbit by giving it water orally... someone writes in "as ooposed to what, anally?" |
| 01:25.17 | *** part/#asterisk SuperAlex (~SuperAlex@adsl-19-19-207.asm.bellsouth.net) |
| 01:25.19 | _daver_ | VMS is rock solid. |
| 01:25.25 | tzanger | and then he talk sabout increasing the light he gives his turkeys to encourage them to lay and someone else writes in "normally people decrease lighting to get laid" |
| 01:25.27 | Beave | yeppers. |
| 01:25.40 | hermie | wouldn't be very hard to port asterisk with all of HP's tools |
| 01:25.43 | znoG | hey guys, is it possible to stick "#asterisk-es now open" in the topic? |
| 01:25.52 | znoG | for the asterisk gringos |
| 01:25.53 | znoG | :) |
| 01:26.33 | Beave | hermie: Famous last words.. Porting to VMS , even with the right tools, isn't always fun. |
| 01:27.02 | _daver_ | Heh. I remember writing a text editor in VAX BASIC back in HS. |
| 01:27.20 | terrapen | daver... |
| 01:27.21 | Beave | _daver_: BASIC on the cluster, as well as a crap load of other compilers |
| 01:27.24 | terrapen | did you work at @Home |
| 01:27.34 | hermie | Beave: I know, I know... but it wouldn't be _that_ difficult :) |
| 01:27.37 | _daver_ | terrapen: No. |
| 01:27.41 | dizzydiffi | ahhhhh |
| 01:27.43 | terrapen | heh, guess not :) |
| 01:28.07 | _daver_ | Beave: Cool. I'm a little rusty with my VMS. |
| 01:28.38 | Beave | hermie: I assited Larne in porting Epic (IRC client). T'was actually a bitch. While DEC C use to be cutting edge, it's not terribly forgiving. |
| 01:28.38 | liquidno2 | Okay kiddies |
| 01:28.47 | liquidno2 | VERY IMPORTANT LESSON JUST LEARNED |
| 01:28.54 | liquidno2 | if you are testing a station |
| 01:28.54 | tzanger | liquidno2: don't eat yellow snow? |
| 01:28.58 | liquidno2 | better |
| 01:29.01 | liquidno2 | and more on topic |
| 01:29.04 | liquidno2 | if you are testing a station |
| 01:29.08 | tzanger | liquidno2: dont' stick your dick in light sockets? |
| 01:29.13 | terrapen | np |
| 01:29.15 | liquidno2 | make sure the station has the ringer ON |
| 01:29.16 | Beave | yeah. |
| 01:29.16 | letherglov | or your dick |
| 01:29.19 | letherglov | that's even worse |
| 01:29.23 | Beave | haha |
| 01:29.32 | tzanger | liquidno2: hahaha |
| 01:29.39 | tzanger | 15 minutes? bah that was an easy fix |
| 01:29.47 | hermie | if you stick your dick in a light socket, make sure the station has the ringer on??? |
| 01:29.47 | liquidno2 | I am here only as a lesson to others. |
| 01:30.05 | tzanger | hermie: only if the ringer's hooked up to oneof those light flashers |
| 01:30.13 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 01:30.19 | hermie | liquindno2: Perhaps the purpose of your life is to serve as a warning to others? |
| 01:30.41 | letherglov | will that safetly and effectively cause your six hour erection from cialis to go away? |
| 01:30.49 | *** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc) |
| 01:30.59 | hermie | no, they drain the blood from your penis with a needle for that |
| 01:31.15 | liquidno2 | hermie: you know it |
| 01:31.24 | tzanger | letherglov: it might blow the damn thing right off depending on how good a connection you managed ot make |
| 01:31.43 | letherglov | time to become a priest... |
| 01:32.04 | liquidno2 | now to just adjust the settings in the context to allow outbound dialing... |
| 01:32.32 | liquidno2 | and the wife will be a happy camper |
| 01:32.38 | liquidno2 | happy wife == happy liquidno2 |
| 01:33.00 | liquidno2 | or at least > miserable hell liquidno2 |
| 01:33.06 | Qwell | upset wife == happy ..nm |
| 01:33.10 | letherglov | WHY DOESN'T THE PHONE WORK liquidno2 !!! |
| 01:33.11 | dontmsgme | Once you have a vonage account how do you use it with Xlite to make calls, they arent giving me a server to point to |
| 01:33.19 | hermie | doesn't VMS have gcc? |
| 01:33.32 | liquidno2 | letherglov: I have had my fair share of that sadly |
| 01:33.42 | liquidno2 | I need to get an UPS for the pbx machine |
| 01:33.47 | letherglov | oh, for sure |
| 01:33.52 | letherglov | I've got a 3000 VA one on my box |
| 01:33.52 | shido6 | thats odd, that should be the first thing u get when u sign up for a service |
| 01:33.56 | letherglov | the hard drive recently took a shit |
| 01:33.59 | liquidno2 | that and the print server |
| 01:34.02 | letherglov | but asterisk seems to be happily loaded in ram |
| 01:34.06 | letherglov | so as long as nobody touches it.... |
| 01:34.07 | letherglov | ;-) |
| 01:34.10 | tzanger | letherglov: that's quite a trick, I'd like to see that |
| 01:34.22 | liquidno2 | What benefit would that provide? |
| 01:34.32 | liquidno2 | does it store most everything in RAM anyways? |
| 01:34.39 | letherglov | yes, until it swaps |
| 01:34.43 | letherglov | point being I can't ssh into it |
| 01:34.45 | letherglov | because the disk is dead |
| 01:34.47 | letherglov | I need to fix it |
| 01:34.51 | liquidno2 | Ah I see |
| 01:34.53 | letherglov | but in the meantime, it still passes calls |
| 01:34.54 | letherglov | ;-) |
| 01:35.04 | liquidno2 | You can swap the drive without power cycling |
| 01:35.08 | letherglov | I was burning it in anyway |
| 01:35.12 | letherglov | so I don't care too much |
| 01:35.14 | liquidno2 | I have done it before |
| 01:35.15 | letherglov | it was supposed to do that |
| 01:35.57 | tzanger | what the fuck is a Bmin chord? |
| 01:36.01 | tzanger | oh |
| 01:36.02 | tzanger | minor |
| 01:36.02 | tzanger | duh |
| 01:36.04 | liquidno2 | minor |
| 01:36.09 | liquidno2 | need the tab for that? :) |
| 01:36.12 | tzanger | no |
| 01:36.23 | tzanger | I am used to seeing Bm for minor |
| 01:36.40 | tzanger | and min ..> I was going through my head for all the 7ths and augments and stuff |
| 01:36.44 | tzanger | and 'min' wasn't fitting anything |
| 01:36.47 | terrapen | anybody else seeing jitter on their NuFone line right now? |
| 01:37.04 | tzanger | terrapen: I'll give you (and all of #asterisk) a very valuable hint |
| 01:37.07 | Cresl1n | anybody have any echo problems w/ the fxo modules on a TDM card? |
| 01:37.07 | tzanger | actually two |
| 01:37.09 | terrapen | this is the first time that it has ever been less than 100% solid for me |
| 01:37.25 | tzanger | dial IAX2/echo@switch-1.nufone.net for their echo app |
| 01:37.38 | tzanger | now talk (or breathe, I find this gives me very easy to hear results) |
| 01:37.48 | tzanger | now od the same thing for switch-2.nufone.net |
| 01:38.02 | terrapen | my traceroutes to them are not pretty |
| 01:38.12 | tzanger | if it occurs on both it's very likely the problem is on your end or your link and not thiers |
| 01:38.16 | tzanger | er theirs |
| 01:38.58 | hardwire | enlighten me |
| 01:39.04 | tzanger | in fact this is such a great test I have set up echo@ for my own servers and request it for the people I work with |
| 01:39.13 | tzanger | makes it dead simple to do a quick test to see if the link is bad |
| 01:39.17 | terrapen | im getting to switch-2 ok |
| 01:39.31 | terrapen | actually both are ok now |
| 01:39.31 | tzanger | and I'm working on a little app that periodically dials these and plays a file and compares the result ot try and automatically determine link quality |
| 01:39.35 | terrapen | musta been a hiccup |
| 01:40.23 | terrapen | tzanger: interesting. |
| 01:41.00 | tzanger | terrapen: it shouldn't be too hard |
| 01:41.07 | tzanger | there are "fuzzy" comparison algos around |
| 01:41.17 | tzanger | look for a result that's > 95% similar and it should be good |
| 01:41.20 | dontmsgme | How does the nufone login work do you get a name/pw or just numbers |
| 01:41.28 | tzanger | dontmsgme: huh? |
| 01:42.09 | dontmsgme | I didnt get an email telling me what to put in extensions.conf |
| 01:42.38 | tzanger | nufone login is just like any other IAX2 provider |
| 01:42.41 | netsurfer | anyone know why all of a sudden my celeron 766 chip would be overheating? (fan is ok, heatsink is clean, and voltage is ok) |
| 01:42.47 | tzanger | IAX2/user@nufone/${EXTEN} |
| 01:42.50 | tzanger | and in iax.confr |
| 01:42.51 | tzanger | [nufone] |
| 01:42.54 | tzanger | type=peer |
| 01:42.56 | tzanger | context=NANPA |
| 01:42.59 | dontmsgme | Whats the address |
| 01:43.02 | tzanger | secret=yourpassword |
| 01:43.03 | dontmsgme | Its not just "nufone" |
| 01:43.10 | tzanger | switch-1.nufone.net and switch-2.nufone.net |
| 01:43.35 | tzanger | I have [nufone1] and [nufone2] and I acutally dial one of the other depending on a db var |
| 01:43.52 | *** join/#asterisk jerlique (jerlique@lnk2.adl.adsl.esc.net.au) |
| 01:43.53 | terrapen | ok, looks like i need to add both nufone servers |
| 01:43.55 | tzanger | so when link quality to one is temporarily shitty (usually a connection problem between me and htem) I flip the DB var and it goes out the other |
| 01:43.56 | terrapen | i currently only have one |
| 01:44.00 | jerlique | I get about 1 sec delay when talking to internal extensions over a lan. Any clues.... |
| 01:44.03 | harryvv | what record in asterisk records all incomming phone numbers |
| 01:44.06 | tzanger | jerlique: what codec |
| 01:44.12 | tzanger | harryvv: huh? |
| 01:44.14 | jerlique | ulaw |
| 01:44.21 | dontmsgme | I dont think you do user:password |
| 01:44.23 | harryvv | my asterisk phone rang without me heading the landlines ring. |
| 01:44.24 | tzanger | /var/log/asterisk/cdr-csv has all your call records |
| 01:44.28 | jerlique | actually both are on ulaw |
| 01:44.28 | dontmsgme | I think it gives you a string instead via email |
| 01:44.41 | harryvv | err xlite phone rang that is |
| 01:44.41 | tzanger | dontmsgme: you can do IAX2/user:pass@peer/exten@context if you like |
| 01:44.46 | tzanger | but then the passowrd shows up in all the logs |
| 01:45.05 | tzanger | jerlique: interesting |
| 01:45.09 | tzanger | connect them via a crossover cable and test |
| 01:45.10 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 01:45.21 | tzanger | you could just have really craptastical network cards or a shitty switch |
| 01:45.54 | tzanger | I try to get intel eepro cards in for everything I do, they just work, can saturate a 100mbit link without breaking a sweat and are very well supported in linux |
| 01:46.28 | *** join/#asterisk CCDAS (~spears_da@65.163.100.254) |
| 01:47.40 | dontmsgme | Can you get a DID with nufone |
| 01:47.41 | jerlique | what you mean ? they are both on a 100mb switch with virtually no data |
| 01:48.02 | harryvv | tzanger I have two master.csv and Master.csv.1 |
| 01:48.22 | tzanger | harryvv: ok so it's being rotated :-) |
| 01:48.44 | tzanger | jeez, you figure guitar tabs would be consolidated by now |
| 01:49.13 | harryvv | rotates? |
| 01:49.17 | harryvv | rotated? |
| 01:49.26 | NormAst | Hi all. |
| 01:49.52 | terrapen | i need to get my MWI working on my IP500 |
| 01:50.15 | tzanger | http://www.mixdown.ca/~andrew/dump/4058b5c229832.jpg oh my dear lord |
| 01:50.22 | harryvv | not sure which one it is. both end at the 5th. |
| 01:50.39 | Legend | tzafrir: 403 errors always scare me |
| 01:50.43 | tzanger | oh for fuck sakes |
| 01:50.58 | terrapen | dontmsgme: in michigan |
| 01:51.01 | tzanger | try again |
| 01:51.12 | tzanger | half the time the permissions are 644 and the other half they go to 600 |
| 01:51.13 | dontmsgme | Right.. |
| 01:51.15 | shido6 | u can get a 8xx number |
| 01:51.22 | dontmsgme | Cool |
| 01:51.48 | Legend | tzanger: i prefered the 403 |
| 01:51.51 | tzanger | hahahaha |
| 01:51.56 | Qwell | ...agreed |
| 01:52.00 | eipi | anyone is running asterisk on vmware? |
| 01:52.04 | tzanger | my ex sent that to me with "the real cause of the tsunami" |
| 01:52.08 | NormAst | eipi: yea.. |
| 01:52.17 | tzanger | NormAst: evening sir |
| 01:52.46 | eipi | normast: i have a problem with local sound |
| 01:52.51 | NormAst | Hay Tzanger.. How's are things going.. |
| 01:52.53 | harryvv | so a php grapical chart can be made of this log then. who is calling how often. |
| 01:53.59 | eipi | normast: the sound cuts every half second |
| 01:54.13 | harryvv | I need to get another asterisk box up and running to show phone activity. |
| 01:54.22 | eipi | normast: any idea? |
| 01:54.45 | harryvv | or better yet to just get another box with linux and php web pages to show activity. |
| 01:54.47 | NormAst | vmware not not really good for real time stuff. |
| 01:54.59 | eipi | :D i Imagine |
| 01:54.59 | NormAst | I just use it to play with.. you know if I crash it I restore from backup. |
| 01:55.12 | eipi | hehe |
| 01:55.28 | dizzydiffi | help pls |
| 01:55.36 | eipi | but there's any fix for sound? |
| 01:55.48 | dizzydiffi | i need help om openh323 |
| 01:56.08 | NormAst | eipi: No... are you getting scheduling in the past! |
| 01:56.08 | dizzydiffi | somebody |
| 01:57.14 | dizzydiffi | anybody |
| 02:00.34 | *** join/#asterisk ionix (ionix@MTL-HSE-ppp186113.qc.sympatico.ca) |
| 02:02.12 | dontmsgme | 'I dont see where you buy the DID at nufone, when you login it just goes to "activity" |
| 02:05.41 | jerlique | tzanger: I do have intel pro cards :) both pcs are p4's doing nothing atm. The * srv is running a p4 too and is sitting idle... |
| 02:06.18 | tzanger | jerlique: that does not seem right |
| 02:06.26 | tzanger | xconnect the two boxes (eliminate the switches) |
| 02:08.06 | jerlique | ok. The switch is a cisco 2950, and pumps out 100Mb quite easily during tests. |
| 02:08.12 | jerlique | I'll try it... |
| 02:19.08 | ariel_ | dontmsgme, nufone mainly does termination it only sells did in I guess one or two states only. |
| 02:20.55 | shido6 | go to did, dontmsgme |
| 02:21.00 | shido6 | then order did |
| 02:21.05 | shido6 | then choose tollfree |
| 02:21.11 | shido6 | and fill out the info and BOOM |
| 02:21.14 | shido6 | u have an 8xx number |
| 02:21.16 | shido6 | right NOW |
| 02:21.43 | *** join/#asterisk heath__ (~heath@12-215-34-6.client.mchsi.com) |
| 02:22.08 | *** join/#asterisk jterrero (~some@ool-43576e0d.dyn.optonline.net) |
| 02:22.53 | fafnir | d |
| 02:23.02 | jterrero | someone want to try and help me make sense of my issue? |
| 02:23.02 | jterrero | I am trying to flash a Cisco 7940 Phone to support SIP. Earlier today I had used the same config files on my notebook, the notebook acting as the TFTP server. When i get home though with the exception of my physical hardware. using the same config. 2 things happen |
| 02:23.02 | jterrero | in the tftp root there is a file call OS79XX.TXT, inside the file there is 1 line POS-06-0-00 (this is how the cisco phone knows what version of the firmware upgrade to download). after the phone pulls that file from TFTP its suppose to proceed to downloading POS-06-0-00.bin but instead it looks for POS-06-.bin |
| 02:23.03 | jterrero | so, I renamed POS-06-0-00.bin file to POS-06-0-.bin, the phone finds the file and it starts "upgrading firmware" but it hangs. When looking at my tftp logs, the image is sent but fails between 75% and 81% (never gets further then 81). |
| 02:23.07 | jterrero | Where should I start troubleshooting? I was thinking maybe I was dropping packets somewhere but then I crossed over the phone to my notebook and im still getting the same results. Also thought maybe it was the phone but I have 2 7940s and the same thing happens on both :( |
| 02:23.23 | jterrero | sorry for the flood |
| 02:24.28 | shido6 | what does the phone have on it already? |
| 02:24.39 | shido6 | what version of what firmware is on the phone already |
| 02:24.51 | shido6 | there is an upgrade path you need to follow |
| 02:25.43 | Evanrude | fyi: I think 7-03-01 or something is the newest firmware |
| 02:25.54 | Evanrude | POS-07-03-1-00 or something like that |
| 02:28.10 | jterrero | the phone has what ever firmware it ships with |
| 02:28.28 | jterrero | PC03C300 |
| 02:29.15 | *** join/#asterisk Rowter (~Drake@201.135.26.29) |
| 02:29.51 | Rowter | what's the best way to deal with a panasonic 12X32 integration with * |
| 02:30.06 | Evanrude | which TFTP server are you using? |
| 02:31.10 | jterrero | solarwinds |
| 02:31.26 | jterrero | same i used earlier at work to sucesfully flash the phones |
| 02:31.47 | jterrero | with the exception of my networking hardware at home, im using same machine as tftp, same tftp folder, same files, same configs |
| 02:32.03 | Rowter | jterrero, problem updating your cisco 7960? |
| 02:32.10 | eipi | what's best and complete web administration? |
| 02:32.48 | jterrero | Rowter: yeah, 7940 |
| 02:33.37 | Rowter | jterrero, reciving timeout ? |
| 02:34.31 | Sedorox | is it possible to have the X100P working in FreeBSD 4? |
| 02:37.31 | postel | jterrero: latest images are signed and previous ones aint, you cant just jump in and flash whatever, and NEVER as in NEVER rename image files, cisco.co has a guide on moving 79xx to SIP, you'll also find the upgrade path there |
| 02:38.16 | postel | that was cisco.com not co but you knew that already, right? |
| 02:40.40 | Nugget | if you download the firmware from cisco.co.uk then you'll need to upgrade all your rowters to rooters before the phone will work. |
| 02:47.00 | postel | if you download from co.uk you gonna end up with a pretty ugly looking teapot |
| 02:47.30 | Nugget | heh |
| 02:48.15 | postel | im in bloody London, cant believe i just said that |
| 02:48.17 | postel | :P |
| 02:49.37 | netsurfer | lmao Nugget u taking the piss outta us brits |
| 02:49.50 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 02:50.41 | postel | more brits alive? man its something like 3 in the morning |
| 02:51.34 | postel | at least come over, i need a beer buddy :p |
| 02:51.40 | postel | night is young |
| 02:51.44 | postel | (not) |
| 02:51.46 | netsurfer | lol |
| 02:52.07 | mikegrb | good idea, I should grab a beer and draw a hot bath |
| 02:53.25 | postel | Nugget: when you finish the Guinness keep the widgets for me, im making a mini snooker table :P |
| 02:53.32 | mikegrb | heh |
| 02:54.00 | *** join/#asterisk outsidefactor (barf@203-206-254-16.dyn.iinet.net.au) |
| 02:54.14 | Nugget | heh |
| 02:54.33 | Nugget | I've got one of each. the old style bottom-mount and the new style floating widget. |
| 02:54.48 | Nugget | no bottle-based rocket widget, though. |
| 02:55.20 | postel | the floating ones should do, if the snooker project goes tits up i can always switch to table tennis |
| 02:58.59 | tzanger | bah |
| 02:59.02 | tzanger | guinness |
| 02:59.06 | Nugget | heh |
| 02:59.07 | *** join/#asterisk calvinhp (~calvinhp@dhcp065-029-088-222.indy.rr.com) |
| 02:59.10 | CoaxD | thought i was gonna get some time to work on my voip stuff. heh. But NOOooOOoOoOo. Damn guineapig has to get hypocalcemia, forcing me to sit there and hand-feed the damn thing calcium all night. |
| 03:00.13 | bjohnson | Had a hobgoblin recently .. very good |
| 03:00.42 | ta[i]nted | tzanger i figured out what that mess was |
| 03:00.47 | postel | well, enough brainstorming for me, dont u DARE making the Royal Asterisk project without quoting me :p |
| 03:00.53 | postel | night people |
| 03:01.25 | ta[i]nted | tzanger windows was mangling the files with carriage return chars |
| 03:02.17 | tzanger | ta[i]nted: ahh |
| 03:02.26 | *** join/#asterisk toddf (yv7mwut0a3@default.fries.net) |
| 03:03.49 | *** join/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com) |
| 03:09.36 | NormAst | Tzanger: You were talking about Echo Canceller's.. Anyones you like? |
| 03:09.42 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 03:09.48 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 03:09.53 | MrEntropy | yo |
| 03:15.02 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfkh2.dialup.mindspring.com) |
| 03:18.37 | harryvv | cold there Boris? |
| 03:18.37 | harryvv | :) |
| 03:19.23 | *** join/#asterisk santiago (~santiago@63.245.86.104) |
| 03:19.27 | mgeorge | anyone here have experience with kphone? I'm having a weird rpoblem with it... |
| 03:19.39 | BoRiS | It was definately cold.... but not that I am back home, its just as cold here! :-p |
| 03:20.39 | netsurfer | is there any way I can get * to check if $var1 is an integer? |
| 03:21.08 | netsurfer | oops ${var1} |
| 03:25.09 | jsmith | netsurfer: Did you read README.variables? |
| 03:25.17 | jsmith | netsurfer: It might say. |
| 03:25.26 | netsurfer | yep im on it now |
| 03:27.39 | ph3nix | what are we paing brandy for |
| 03:27.41 | ph3nix | er |
| 03:27.58 | Beirdo | yay. my X100P clone arrived |
| 03:28.32 | Sedorox | I've had one... the one showed up at the other place.. trying to get it working.. |
| 03:28.50 | ta[i]nted | this is the tightest cup warmer i've ever seen |
| 03:28.53 | ta[i]nted | http://www.heise.de/ct/machflott/projekte/55631 |
| 03:28.59 | *** join/#asterisk DuraServer (~tech@64.95.36.4) |
| 03:30.44 | DuraServer | ?? Anyone Please?? Setting Callerid Name, outgoing calls, on PRI, ni-2 ?? |
| 03:31.48 | *** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de) |
| 03:32.01 | muesli | hi, i'm trying to setup a voicebox with asterisk |
| 03:32.17 | muesli | but the app_voicemail.so module fails loading with: "undefined symbol: adsi_available" |
| 03:32.21 | DuraServer | Set CallerID works fine for the numeric, but SetCIDName has no effect |
| 03:32.46 | jsmith | DuraServer: You might have to use SetCallingPres() first. |
| 03:33.00 | *** join/#asterisk PBXtech (~nik@216-160-234-124.slkc.qwest.net) |
| 03:33.05 | DuraServer | Isnt that for inbound calls only? |
| 03:33.39 | DuraServer | to preserver the callerid as the call passes through the incoming dial plan? |
| 03:34.16 | muesli | ah ok, got it. res_adsi.so |
| 03:34.34 | jsmith | DuraServer: No, it's for outbound. |
| 03:34.54 | PBXtech | hi jsmith long time no see |
| 03:35.32 | DuraServer | SetCallingPres(), in Zapata.conf?, not to be used in extensions conf anywhere, right/ |
| 03:35.32 | jsmith | PBXtech: Hey man, what's up? |
| 03:35.32 | jsmith | DuraServer: No, it's an application you call from extensions.conf |
| 03:35.41 | PBXtech | no much still doing the same old same old :) you still workin in orem? |
| 03:35.56 | DuraServer | ok, I will read up. Thanks very much |
| 03:36.18 | jsmith | PBXtech: Yup. You doing local termination yet? You still got all those DIDs? |
| 03:36.52 | PBXtech | i got DID in ogden slc and provo running through *. not really reselling or anything |
| 03:37.27 | jsmith | PBXtech: Maybe I'll have to "borrow" a piece of your block :-) |
| 03:37.36 | PBXtech | cool |
| 03:38.17 | PBXtech | i have 1 PRI incoming and sharing it with 3 (including my own) companies |
| 03:38.18 | tzanger | I love those Immodium commercials |
| 03:38.25 | PBXtech | cheep way to split it all up |
| 03:38.49 | jsmith | PBXtech: Nice... I want a (small) block of your DIDs, if you don't mind :-) |
| 03:39.06 | PBXtech | how many do you need, i can just get more. its like .15 cents each |
| 03:39.24 | PBXtech | got a T1 from ELI for network |
| 03:47.08 | Luke-Jr | http://pastebin.ca/5443 |
| 03:47.27 | Luke-Jr | Can anyone help me get the SIP requests from the Asterisk one to the working one? |
| 03:47.41 | Luke-Jr | (it's a diff between a failed registration and a working registration) |
| 03:48.33 | DuraServer | jsmith... when used in extensions.conf, error, No application 'SetCallingPres' |
| 03:49.14 | jsmith | DuraServer: Sorry, that was SetCallerPres() (Caller, not Calling) |
| 03:49.15 | Luke-Jr | mgeorge: what kphone prob? |
| 03:50.17 | DuraServer | I'm looking to Set the Outgoing CallerID Name. Any syntax tips on that? using a PRI with ni2 |
| 03:51.33 | Luke-Jr | DuraServer: SetCIDName? |
| 03:51.55 | Nugget | that would be too easy! |
| 03:52.03 | DuraServer | Yes, i use that, in the next priority after SetCallerID |
| 03:52.12 | dontmsgme | What is ni2? |
| 03:52.15 | dontmsgme | The PRI provider? |
| 03:52.29 | PBXtech | protocol for the PRI |
| 03:52.51 | DuraServer | I use SetCallerID(7325551234), then next I use SetCIDName(DURASERVER) |
| 03:53.10 | DuraServer | The PRI provider says they are passing everything through |
| 03:53.14 | DuraServer | that i send them |
| 03:53.34 | DuraServer | The numeric part of the caller id Sends ok, but not the name |
| 03:53.47 | DuraServer | the recipient gets the number and UNKNOWN |
| 03:53.50 | DuraServer | as the anme |
| 03:54.01 | DuraServer | PRI proto = ni2 |
| 03:54.51 | ManxPwr | DuraServer, Most terminating telcos toss away the NAME and replace it with whatever is associated with the NUMBER. |
| 03:54.55 | DuraServer | framing esf/bz2 |
| 03:55.13 | file[laptop] | SS7! |
| 03:55.25 | ManxPwr | So you basically cannot set the Caller*ID NAME. |
| 03:55.57 | file[laptop] | but sometimes, and I repeat SOMETIMES, if the number you're calling is within the same provider like... then SOMETIMES name is carried |
| 03:56.07 | DuraServer | That seems to be true, because if I send out for example, my own Home phone number as the caller id number, then the recipient sees the name associated with my home telephone number |
| 03:56.25 | ManxPwr | Don't get his hopes up, file[laptop] |
| 03:56.43 | file[laptop] | oh god timezones, I forgot |
| 03:56.49 | DuraServer | However, if no name is already associated in the recipient's telco, shouldn't the name from the PRI be utilized? |
| 03:56.56 | file[laptop] | DuraServer: no. |
| 03:57.07 | ManxPwr | DuraServer, That WOULD be the logical thing for them to do, huh? |
| 03:57.18 | file[laptop] | REALITY! |
| 03:57.38 | ManxPwr | file[laptop], Reality is for people that can't handle drugs. |
| 03:57.46 | DuraServer | Logic?, but then.... we're talking about the recipient being a Verizon customer. No Logic ... |
| 03:59.00 | DuraServer | Besides getting the PRI provider to be honest, is there any tips to debug, if they PRI provider is really passing it through? |
| 03:59.05 | *** join/#asterisk juice (~juice@mo-65-41-96-204.dyn.sprint-hsd.net) |
| 04:00.11 | NormAst | DuraServer: I was told you need to use Ni3 to get the CallerName to pass. |
| 04:00.20 | DuraServer | Test # 1 I have done... Placed a digium ffx card in the same server, and dialed through to a pots phone with callerid built in... The Name and Number Both go through to the pots phone. |
| 04:00.40 | DuraServer | typo... ffx=fxs |
| 04:02.51 | *** join/#asterisk This`That (No@dialup-ras2-216.pdx.or.uspops.net) |
| 04:02.55 | This`That | Hello all |
| 04:03.40 | DuraServer | Thanks Norm... I'll need to contact the PRI provider |
| 04:04.18 | NormAst | DuraServer: Not sure.. However.. Could you let me know if it works? |
| 04:04.38 | DuraServer | certainly |
| 04:05.41 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 04:06.23 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
| 04:07.30 | pcm | what's up? |
| 04:07.41 | zimdog | How do I refresh the list that shows up in sip show peers? |
| 04:08.04 | Nugget | you mean to load changes made to sip.conf? |
| 04:08.08 | Nugget | "reload" |
| 04:08.10 | Sedorox | reload> |
| 04:08.21 | labo | f5 :) |
| 04:08.21 | zimdog | I have done relaod and sip reload |
| 04:08.30 | *** part/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com) |
| 04:08.31 | Sedorox | lol labo |
| 04:08.32 | pcm | well then your device didn'g register ... |
| 04:08.38 | Nugget | then you've done what you need to do in order to refresh the list that shows up in sip show peers. |
| 04:08.54 | Nugget | even if it didn't register it will still show up. |
| 04:09.04 | pcm | yeah ... but no IP |
| 04:09.08 | pcm | unless you hardcode it |
| 04:09.11 | zimdog | It still shows the ip though |
| 04:09.18 | pcm | well maybe chan_sip.c has a bug... just do 'restart now' |
| 04:09.29 | pcm | it might be locked ... |
| 04:10.01 | zimdog | still there where would I hard code it ? sip.conf |
| 04:10.12 | pcm | nowehre else |
| 04:10.26 | Nugget | that would make more sense than, say, iax.conf. |
| 04:10.30 | pcm | [account]\nhost=1.2.3.4 |
| 04:11.09 | dontmsgme | When I got to create an account at Nufone it doesnt give me an option to buy a DID is that an email request only |
| 04:11.12 | zimdog | ok it is not hard coded |
| 04:11.26 | zimdog | is there an amount of time before they drop out? |
| 04:11.40 | pcm | qualify=amount_of_ms |
| 04:11.47 | pcm | I think |
| 04:12.35 | zimdog | I had qualify=no so I should change this and it will cause it to try and reregister? |
| 04:12.53 | pcm | no qualify is for heartbeat |
| 04:13.01 | pcm | it sends options sip message |
| 04:13.08 | pcm | and checks if a compatible sip device responds |
| 04:14.22 | pcm | if your sip device doesn't reregister you need to change the ;maxexpirey=3600 ; Max length of incoming registration we allow "defaultexpirey=120" |
| 04:14.41 | pcm | change both to 10 seconds to see if it works |
| 04:15.07 | liquidno2 | analog caller id is set at the carrier switch right? so I can't override the settings by setting my own callerid on my zap channel right? Even if I am just setting it to a caller id to one of the two numbers on the analog line? |
| 04:15.07 | pcm | and if you watch the sip messages there should a header "Expire: 10" .... |
| 04:15.09 | Primer | can a wrt54gp2a-at be ahcked to use asterisk? |
| 04:15.13 | Primer | a buddy of mine just bought one, and I figured I'd have him try to connect to my asterisk first |
| 04:15.21 | pcm | liquidno: right |
| 04:15.28 | liquidno2 | pcm: thought so |
| 04:15.54 | pcm | liquidno: but if you have two numbers on the line |
| 04:16.02 | pcm | then propably you always use one callerid |
| 04:16.02 | liquidno2 | I do |
| 04:16.08 | pcm | unless telco gives you some codes to change that |
| 04:16.15 | liquidno2 | I would like to force it to the business line |
| 04:16.15 | pcm | like dtmfs that you dial before you dial the number |
| 04:16.24 | liquidno2 | that is a good question |
| 04:16.33 | liquidno2 | if it was bellsouth I would say that they migth offer it... |
| 04:16.42 | Sedorox | Primer: I doubt it.. but if you find out.. let me know |
| 04:16.44 | pcm | maybe, call them in working hrs :) |
| 04:16.45 | pcm | heh |
| 04:16.50 | liquidno2 | but century tel... I doubt it |
| 04:16.59 | zimdog | pcm: thanks I will try that . I just commented it out and reloaded and it dropped it |
| 04:17.06 | Primer | Sedorox: yeah, seems that way |
| 04:17.09 | MrEntropy | is there a default entry i can make for asterisk to dial? this is slightly different than a default dialplan rule. An AGI script at one point does Dial(SIP/NNNNNNN). but if no user like that is registered, i'd like it to be forwarded/dialed to a certain user/sip-proxy |
| 04:17.48 | pcm | MrEntropy: just put some Goto(always-do-this,s,1) |
| 04:18.03 | pcm | and define [always-do-this] \n exten => s,1,Dial(SIP/always-go-here) |
| 04:18.21 | pcm | put it after the Dial ... if it fails then this wil lbe executed |
| 04:18.33 | MrEntropy | pcm: but the agi script does the dial |
| 04:18.41 | Luke-Jr | http://pastebin.ca/5443 <-- diff between failing SIP register and succeeding SIP register (diff sources/clients); any ideas on how to get Asterisk to succeed? :/ |
| 04:18.43 | liquidno2 | does * have the ability to change the ring method to stations? |
| 04:19.00 | pcm | Mrentropy ... hmmmm ... so anyways AGI script shouldn't hangup the channel |
| 04:19.04 | liquidno2 | like distcintive rings but internally? |
| 04:19.14 | pcm | then try to put it after AGI(whatever) |
| 04:19.37 | MrEntropy | pcm: this one does, it picks up and hangs up the channel, it's a DeadAGI to be more correct |
| 04:20.28 | pcm | luke-jr: you need asterisk to make the same contact field |
| 04:20.49 | pcm | luke-jr ... try to define account=<put_callerid_here> |
| 04:20.54 | *** join/#asterisk PeaCeKeePa (~free@d36-165-72.home1.cgocable.net) |
| 04:20.56 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 04:21.12 | file[laptop] | okay I've calmed down |
| 04:21.17 | PeaCeKeePa | ne1 know of a cheap DID for Toronto? |
| 04:21.19 | pcm | liquidno2: sure, just use some I think Dial(blah,,r2) |
| 04:21.24 | blitzrage | quitting this room, find me in #asterisk-doc (only stating for the sake of jbot if someone happens to ask :)) |
| 04:21.28 | pcm | liquindo: or read "show application dial" |
| 04:21.28 | *** part/#asterisk blitzrage (~blitzrage@d141-234-145.home.cgocable.net) |
| 04:21.49 | shido6 | 647 sound ok to u? |
| 04:21.55 | PeaCeKeePa | sure |
| 04:22.00 | PeaCeKeePa | how much? |
| 04:22.01 | shido6 | need it right now? |
| 04:22.11 | PeaCeKeePa | not tonight, but in the next week or so |
| 04:22.27 | shido6 | 8 bucks a month I think |
| 04:22.36 | liquidno2 | pcm: danke |
| 04:22.47 | PeaCeKeePa | with who? |
| 04:23.03 | shido6 | xetricom |
| 04:23.22 | NormAst | has anyone see this before "Don't know what to do if second ROSE component is of type 0x6" |
| 04:23.40 | pcm | like-jr: ir it's actually fromuser=<put_callerid_here> |
| 04:24.33 | pcm | normast: and it's what giving you that message ? |
| 04:24.38 | PeaCeKeePa | thx, had seen them b4, was looking for even cheaper ;) |
| 04:24.41 | Luke-Jr | pcm: where does that go? |
| 04:25.21 | pcm | in the definition of your peer .... |
| 04:26.01 | pcm | or try this register => alfa:beta@host/alfa |
| 04:26.18 | PeaCeKeePa | does anyone know the deal with vonage canada if they're giving away the linksys like they are in the states? |
| 04:27.09 | pcm | luke: and then have [host] be defined with type=peer or type=friend and have fromuser=alfa |
| 04:27.42 | harryvv | anyone here have voipjet and know what the proper way to asign a extention to one number? say assign 100 to my cell phone. This way anyone who dials in hears the irv can be given a option of calling my call without revealing the number. |
| 04:28.09 | pcm | luke: or read line 111 of configs/sip.conf.sample of cvs head |
| 04:28.16 | harryvv | I have something written out but voipjet rejected it. |
| 04:28.16 | pcm | luke: it's explained there |
| 04:28.49 | pcm | nope, no voipjet |
| 04:29.18 | zimdog | pcm: Are the ip addresses cached or something to show up in sip show peers. I removed the user and did a sip reload and the entry was gone. I then added it back and it immediatley showed up again with the ip. I also did the max and default expirey but this hasn'e changed anything. |
| 04:29.25 | *** part/#asterisk DuraServer (~tech@64.95.36.4) |
| 04:30.15 | Luke-Jr | nope :/ |
| 04:30.30 | *** join/#asterisk Nukemizer (Nuke@66.237.85.58.ptr.us.xo.net) |
| 04:30.49 | Luke-Jr | http://pastebin.ca/5447 <-- updated |
| 04:31.54 | pcm | pcm: yeah, they're cached in the db1 local database |
| 04:32.00 | pcm | ups: zimdog hehe |
| 04:32.02 | harryvv | pcm do you understand what im talking about? |
| 04:32.14 | Luke-Jr | Is there a ViewCVS so I don't need to fetch all of HEAD? |
| 04:32.25 | pcm | harryvv: I don't know ... don't have your service ... |
| 04:32.48 | pcm | Luke-jr: did you make contact header have your username instead of sip:s ? |
| 04:33.02 | Luke-Jr | my #, yes |
| 04:33.19 | harryvv | pcm, I think alot of voip carriers follow the same usernam password assignment. |
| 04:33.47 | zimdog | pcm: is there a way to flush the entries from the database? |
| 04:34.02 | pcm | zimdog: try to see 'database show' in the CLI |
| 04:34.15 | pcm | zimdog: and then database del ... |
| 04:34.35 | pcm | harryvv: why do you think so ? |
| 04:34.50 | pcm | Luke-Jr: and what do you receive as a response ? |
| 04:35.22 | pcm | Luke-Jr: did you use the same password as this box ? :) |
| 04:35.25 | harryvv | well, its not important what the username and password as much how its layed out in the extension context |
| 04:35.51 | Luke-Jr | pcm: http://pastebin.ca/5448 <-- response |
| 04:35.55 | Luke-Jr | pcm: same user/pass, yep |
| 04:36.23 | pcm | harryvv: well usually extension context stays the same and the username/password changes ... instead of giving every user a diffrent context |
| 04:36.35 | Luke-Jr | Actually... |
| 04:36.50 | Luke-Jr | The one that worked never asked for the # |
| 04:38.22 | harryvv | my password is 32 charicters long. so in a case where i wanted somone to dial my cell phone it would be exten => 100,1,Dial,iax2/2323@voipjet/604####### with 2323 a made up username |
| 04:38.45 | harryvv | or does the 32 charicter password belong in there to? |
| 04:39.45 | pcm | harryvv: well if you don't have a RSA key then you should have the pass there ... Dial(IAX2/user:pass@box/number) |
| 04:40.30 | pcm | luke-jr: wierd .... can you post 4 msgs from the first register through the 401 ? |
| 04:40.51 | *** join/#asterisk This`That (No@dialup-ras1-238.pdx.or.uspops.net) |
| 04:42.40 | harryvv | yea |
| 04:42.42 | harryvv | okay |
| 04:45.06 | Luke-Jr | pcm: http://pastebin.ca/5449 |
| 04:45.42 | zimdog | pcm: What is the family and tree. I looked in the wiki but did not see an example of how to use the commands. I figured the family is the first entry the key the second but this don't work. The syntax says Usage: database del <family> <key> |
| 04:47.01 | Corydon76-home | What's the entry that you want to delete? |
| 04:47.49 | pcm | zimdog: when you display all entries with database show the family/key => value |
| 04:48.35 | Corydon76-home | <PROTECTED> |
| 04:49.22 | zimdog | How come I can't paste it do I need to escape since their are slashes |
| 04:49.39 | Corydon76-home | Initial space |
| 04:49.48 | Corydon76-home | Prefix it with a space |
| 04:49.59 | zimdog | <PROTECTED> |
| 04:50.07 | zimdog | that is what I want to remove |
| 04:50.29 | Corydon76-home | So 'database del SIP/Registry 6400' |
| 04:51.16 | zimdog | cool that worked |
| 04:52.56 | zimdog | thanks Corydon76-home I would not have guessed that /SIP/Registry/ = family |
| 04:53.20 | Corydon76-home | You could also have done 'database del SIP Registry/6400' |
| 04:53.46 | Corydon76-home | But the first way was how it was intended to be |
| 04:54.09 | zimdog | Thanks. Though it does not seem to have changed anything when I do sip show peers |
| 04:55.06 | zimdog | Never mind I restarted * and now it is gone |
| 04:56.25 | pcm | luke: it seems right to me |
| 04:57.17 | pcm | luke: however maybe their implementation is screwed ... that box that yo uhave doesn't send the "algorithm=MD5" |
| 04:57.43 | *** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
| 04:57.43 | pcm | luke: you could try to edit chan_sip.c and put everything in order like with this other REGISTER request ... I mean the registartion data ...etc |
| 04:57.51 | Luke-Jr | what about the rpath thing? |
| 04:58.08 | pcm | luke: rpath ? |
| 04:58.17 | Luke-Jr | the client that works lacks it in Via |
| 04:58.33 | Luke-Jr | oops... rport |
| 04:58.45 | Luke-Jr | can I spoof the UA? |
| 05:05.33 | Luke-Jr | tried those... no help :/ |
| 05:05.35 | zimdog | Why do my local phones and the sip provider I registered with show the port as 5060 with sip show peers. But the external phone I try have register picks some random port. I only opened port 5060 udp on the firewall SO I know it comes in on that |
| 05:06.32 | *** join/#asterisk talkwebhosts (~admin@c-24-130-183-95.we.client2.attbi.com) |
| 05:06.58 | dontmsgme | Where can I get a DID that will work just like Voicepulse Connect |
| 05:10.29 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 05:13.15 | Nukemizer | once an IAXY device has been setup do you have to do a hard "reset" to the device and get it reconfigured at the Asterisk box ? |
| 05:14.12 | *** join/#asterisk alakdan (~alakdan@210.213.185.59) |
| 05:14.49 | alakdan | hello, whats the command in the extension.conf to say 'ring' twice |
| 05:19.25 | harryvv | alda, depends what device you want to ring. sip iax or zap |
| 05:20.03 | alakdan | harryvv: sip or iax |
| 05:20.17 | harryvv | btw, Is there a way to reduce the echo between extention 100 to voipjet i LA to my cell phone up here? Or is that still a function of tweeking tx rx in asterisk? |
| 05:21.17 | netsurfer | Feb 8 05:20:09 ERROR[12509]: acl.c:308 ast_netsock_bindaddr: Unable to bind to 0.0.0.0 port 4569: Address already in use <-- why would I be getting this? |
| 05:21.38 | *** join/#asterisk elkcsr (~elkcsr@CPE-67-48-91-54.neb.rr.com) |
| 05:22.00 | netsurfer | harryvv - did u add all the tweaks suggested on voipjet site ? |
| 05:22.07 | harryvv | yea |
| 05:22.11 | harryvv | did that days ago |
| 05:22.11 | *** join/#asterisk Schism (~schism@clt74-101-230.carolina.rr.com) |
| 05:22.17 | JerJer | tweaks? |
| 05:22.29 | netsurfer | just a few codec things |
| 05:22.31 | netsurfer | and jitter |
| 05:22.34 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 05:22.37 | harryvv | it is great on my local network but that is local. long distance is where echo is really prevelent. |
| 05:22.46 | JerJer | why should customers have to tweak their boxes just to use voipjet? |
| 05:22.59 | netsurfer | they arent voipjet specific |
| 05:23.29 | harryvv | Even central offices have echo cancel boards. |
| 05:23.50 | harryvv | but thay are very very fast to pickup on it and automaticly adjust. |
| 05:24.44 | implicit | JerJer: what do you mean/ |
| 05:24.53 | netsurfer | I had echo on my SIP connection once, but it just vanished on its own |
| 05:25.51 | netsurfer | harryvv - as a matter of interest, what pingtime are u getting to the voipjet server |
| 05:26.29 | harryvv | well, let me see |
| 05:26.39 | Luke-Jr | pcm: still around? |
| 05:27.00 | harryvv | 64 bytes from voipjet.com (64.28.107.46): icmp_seq=5 ttl=48 time=168 ms |
| 05:27.00 | harryvv | 64 bytes from voipjet.com (64.28.107.46): icmp_seq=6 ttl=48 time=167 ms |
| 05:27.00 | harryvv | 64 bytes from voipjet.com (64.28.107.46): icmp_seq=7 ttl=48 time=171 ms |
| 05:27.00 | harryvv | 64 bytes from voipjet.com (64.28.107.46): icmp_seq=8 ttl=48 time=213 ms |
| 05:27.31 | harryvv | whats considered substantial? |
| 05:27.38 | netsurfer | :o\ |
| 05:27.43 | netsurfer | thats kinda high |
| 05:28.12 | netsurfer | and sporadic |
| 05:28.13 | harryvv | mmm alot of people are probebly calling other people on there network |
| 05:28.14 | harryvv | :) |
| 05:28.23 | netsurfer | rtt min/avg/max/mdev = 159.006/160.346/161.540/0.834 ms |
| 05:28.32 | Qwell | I get a fairly steading 60ms to voipjet.com |
| 05:28.39 | netsurfer | im gettng less than .8ms difference |
| 05:28.43 | Qwell | steady, rather |
| 05:29.18 | harryvv | its 143 to 180 |
| 05:29.21 | netsurfer | harryvv - i'd say ur either a) using quite a bit of bandwidth locally or b) ur provider is contended quite a bit |
| 05:29.21 | terrapen | jesus, debian is so fucked |
| 05:29.32 | netsurfer | lmao terrapen i know the feeling |
| 05:29.34 | terrapen | the exim configuration is so completely ridiculous |
| 05:29.40 | terrapen | i've used exim since 1997 |
| 05:29.41 | Qwell | Do they have a second server or anything? |
| 05:29.44 | terrapen | i know it like the back of my hand |
| 05:29.57 | terrapen | but i can't figure out, for the life of me, how to configure it under debian |
| 05:30.01 | netsurfer | not at the moment Qwell |
| 05:30.09 | harryvv | netsurfer, its not my network its two nodes |
| 05:30.10 | harryvv | :) |
| 05:30.27 | netsurfer | did u do a tracert ? |
| 05:30.31 | harryvv | yes |
| 05:30.56 | netsurfer | outside ur providers network ? |
| 05:30.57 | harryvv | okay no thought i did |
| 05:31.43 | netsurfer | hmm is the 216.118.117.1 server still active |
| 05:31.44 | harryvv | what is it apt-get something other then tracert? |
| 05:31.54 | netsurfer | its kinda lower from here, ping time wise |
| 05:31.56 | terrapen | root@asterisk [322] 05:27:34 [/etc/exim4]-> ls -lR|wc -l |
| 05:31.57 | Qwell | traceroute |
| 05:31.57 | terrapen | 73 |
| 05:31.59 | terrapen | that's RIDICULOUS |
| 05:32.01 | harryvv | yea |
| 05:32.04 | terrapen | there should be ONE FUCKING FILE |
| 05:32.07 | terrapen | the configuration file! |
| 05:32.30 | techie | ! |
| 05:33.16 | harryvv | okay terrapen chill out |
| 05:33.17 | harryvv | ;) |
| 05:33.28 | harryvv | 26 hops to voipjet |
| 05:33.36 | harryvv | one to many routers |
| 05:33.37 | harryvv | :) |
| 05:33.50 | netsurfer | 26? lmao |
| 05:34.12 | harryvv | hop 22 is the bottle neck |
| 05:34.15 | harryvv | 22 66.194.86.14 (66.194.86.14) 226.572 ms 317.264 ms 227.067 ms |
| 05:34.27 | techie | 226ms holy fook |
| 05:34.31 | harryvv | hehhe |
| 05:34.38 | harryvv | mmm lets see who that is |
| 05:34.39 | harryvv | ;) |
| 05:34.39 | Qwell | heh, I get that same one, and its pretty slow too |
| 05:34.40 | netsurfer | 317 wow |
| 05:34.44 | Qwell | 15 66.194.86.14 (66.194.86.14) 97.102 ms 217.966 ms 220.157 ms |
| 05:34.47 | techie | more like a traffic light |
| 05:34.53 | netsurfer | im making it in 24 |
| 05:34.56 | netsurfer | 24 voipjet.com (64.28.107.46) 160.114 ms 161.514 ms 161.984 ms |
| 05:35.14 | harryvv | what are you guys getting? |
| 05:35.16 | harryvv | for 22? |
| 05:35.18 | Qwell | Somehow, I'm getting fairly decent pings to voipjet.com |
| 05:35.24 | Qwell | and its 19 hops |
| 05:35.30 | harryvv | or that is 66.194.86.14? |
| 05:35.33 | *** join/#asterisk three55ml (~who@cs662589-157.satx.rr.com) |
| 05:35.39 | Qwell | harryvv: Thats my 15 |
| 05:35.41 | netsurfer | 20 66.194.86.14 (66.194.86.14) 160.067 ms 161.773 ms 157.956 ms |
| 05:35.44 | three55ml | ANyone have experience with Realtime? |
| 05:35.59 | netsurfer | no, not me, three55ml |
| 05:36.04 | pcm | three: realtime streaming ? |
| 05:36.08 | three55ml | I'm getting a crash with no debug info while dialing an extension |
| 05:36.16 | three55ml | No, Asterisk Realtime (realtime config from a database) |
| 05:36.29 | netsurfer | ooooooooh |
| 05:36.35 | Luke-Jr | "1) Upgrade Asterisk to newest CVS image; Asterisk CVS-v1-0-12/23/04-22:36:11 (**** this is most important 401 errors; AUTH can only work on this release ****) " |
| 05:36.40 | netsurfer | i'd check u got the fields set up right, three55ml |
| 05:36.41 | Luke-Jr | 1.0.5 should work with this, right? |
| 05:36.42 | harryvv | 23 207.200.37.89 (207.200.37.89) 274.159 ms 184.015 ms 208.500 ms |
| 05:37.00 | harryvv | I must have hit a peak when i did the laste traceroute |
| 05:37.04 | netsurfer | im not hitting that router, harryvv |
| 05:37.14 | Qwell | weird, me neither. I'm hitting the .85 |
| 05:37.19 | three55ml | netsurfer: I do, most extensions work fine. When I hit one dialing out on an IAX2 connection it crashes |
| 05:37.22 | netsurfer | me too, Qwell |
| 05:37.31 | Qwell | its sitting at like 55ms |
| 05:37.32 | techie | 23 voipjet.com (64.28.107.46) 62.432 ms 62.226 ms 62.249 ms |
| 05:37.36 | pcm | three: post the gdb backtrace |
| 05:37.43 | Qwell | Thats about what I get too, around 60 |
| 05:37.43 | harryvv | Im going to have some fun and download neotrace ..if its still free. |
| 05:37.51 | three55ml | pcm: Good thinking, haven't even tried that yet |
| 05:37.54 | pcm | three: does it always dump on that one thing ? |
| 05:38.02 | *** join/#asterisk santiago (~santiago@63.245.86.104) |
| 05:38.07 | three55ml | pcm: As far as I can tell. |
| 05:38.27 | three55ml | pcm: Other extensions work OK |
| 05:38.30 | netsurfer | three - make sure the syntax is correct, all I can suggest |
| 05:38.54 | harryvv | Are there any neotrace like graphical traceroute guis for linux? |
| 05:39.40 | three55ml | It crashes on: Dial(IAX2/USER:PASS@66.234.228.170/${EXTEN}) |
| 05:39.40 | pcm | mrtg |
| 05:39.58 | netsurfer | Dial, |
| 05:40.00 | netsurfer | not Dial( |
| 05:40.08 | netsurfer | afiacr |
| 05:40.14 | three55ml | That's what I meant, sorry. It's in the database |
| 05:40.15 | pcm | it doesn't matter |
| 05:40.16 | netsurfer | afaicr * |
| 05:40.44 | pcm | three: do the gdb bt or let me :) |
| 05:40.51 | three55ml | pcm: Doing it now |
| 05:43.56 | pcm | oki |
| 05:45.03 | netsurfer | <silence> |
| 05:45.13 | pcm | <noise> |
| 05:45.20 | netsurfer | </silence> |
| 05:45.22 | netsurfer | :oP |
| 05:45.39 | pcm | syntax error |
| 05:46.00 | three55ml | http://pastebin.ca/5452 |
| 05:47.29 | three55ml | INSERT INTO `asterisk_extensions` VALUES (9, 'CONTEXT', '_1NXXNXXXXXX', 1, 'Dial', 'IAX2/USER:PAS@66.234.228.170/${EXTEN}'); |
| 05:47.55 | JerJer | grr |
| 05:47.56 | three55ml | That's the record that it's crashing on. I know the context it correct. CONTEXT, USER, and PASS have been removed. |
| 05:48.04 | JerJer | don't use that crap then |
| 05:48.06 | pcm | res_config_mysql.so is this a standard app ? |
| 05:48.32 | pcm | the category in ast_variable_append is NULL |
| 05:48.33 | three55ml | asterisk-addons, in CVS |
| 05:49.25 | three55ml | oops, there's a little more I missed, want me to paste it? |
| 05:49.39 | *** join/#asterisk satlink (satlink@66.178.97.50) |
| 05:50.27 | three55ml | http://pastebin.ca/5453 |
| 05:50.45 | JerJer | why is everyone insistant on forcing Asterisk to depend on a database to operate? |
| 05:51.24 | ManxPwr | JerJer, Because they think it will make their life easier. |
| 05:51.36 | three55ml | JerJer: Because I'm building a virtual PBX where users can dynamically build extensions and don't want to reload 1,000 times a day. |
| 05:51.59 | three55ml | http://www.spirestudios.com/misc/pbx_screenshot.jpg |
| 05:53.30 | satlink | three55ml: what kind of software is this? |
| 05:53.37 | JerJer | what is wrong with reloading? |
| 05:54.08 | *** join/#asterisk Chuji (Chuji@pcp09929633pcs.tulipgrove.tn.nash.comcast.net) |
| 05:54.20 | terrapen | i just broke the learning curve tonight |
| 05:54.24 | three55ml | If the realtime functionality is there, why not use it? Cuts down on a lot of resource usage across the board. |
| 05:54.27 | terrapen | * is so fucking cool |
| 05:54.35 | JerJer | realtime is not the answer |
| 05:54.47 | terrapen | three: what makes you think that using a DB will cut down on resources? |
| 05:55.03 | terrapen | it's a rare application that uses less resources when tied to a DB |
| 05:55.19 | three55ml | JerJer: To be honest, I don't know exactly what happens when Asterisk reloads. Is it unavailable at all during the reload? |
| 05:55.28 | JerJer | then u should learn |
| 05:55.33 | JerJer | before bloating it with bullshit |
| 05:55.42 | three55ml | No need to get hostile, I'm just asking a question. |
| 05:55.47 | netsurfer | three55ml - is that your own app ? |
| 05:55.51 | three55ml | Yeah |
| 05:55.55 | netsurfer | nice looking |
| 05:55.58 | three55ml | Thanks |
| 05:56.07 | netsurfer | open source or commercial ? |
| 05:56.17 | three55ml | Commercial, unfortunately :) |
| 05:56.20 | netsurfer | blaah |
| 05:56.22 | JerJer | i already see major problems with flexibility |
| 05:56.30 | JerJer | on that screen shot |
| 05:56.34 | datareactor | i get this error ERROR[2568]: Unknown signalling method 'bri_cpe_ptmp' |
| 05:56.36 | ManxPwr | When CVS-HEAD becomes a released version I'll prolly try doing a web interface for my users as well. |
| 05:56.46 | three55ml | I might do an open source version, haven't decided yet |
| 05:57.24 | three55ml | JerJer: That's an old screen shot. Feel free to chime in with any recommendations. I intentionally am limiting functionality to the basics. |
| 05:57.25 | terrapen | three, what OS is that running on? |
| 05:57.31 | three55ml | terrapen: Linux |
| 05:57.55 | JerJer | i don't chime in for free |
| 05:58.06 | terrapen | it does look nice. |
| 05:58.14 | terrapen | but i still don't see why you need a DB |
| 05:58.26 | JerJer | you can store the data in db, sure |
| 05:58.35 | JerJer | just don't force asterisk to depend on it |
| 05:58.35 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:58.40 | terrapen | three: what GUI toolkit is that? |
| 05:58.47 | three55ml | terrapen: All custom |
| 05:58.52 | Chuji | ok, now I'm interested |
| 05:58.58 | Chuji | Let me see the screenshot? |
| 05:59.03 | Chuji | I just /j'd |
| 05:59.05 | JerJer | looks like a website to me |
| 05:59.08 | pcm | three: what's the category anyways |
| 05:59.09 | terrapen | custom widgets? |
| 05:59.18 | pcm | three: I found a line where it segfaults .... most likely |
| 05:59.20 | terrapen | you wrote your own toolkit. |
| 05:59.28 | terrapen | no Qt, no GTK... |
| 05:59.34 | three55ml | JerJer: Thats exactly what I was doing at first. It's all in a database and config files that are included from the relevant Asterisk files are generated when changed. |
| 05:59.39 | Qwell | I'm with JerJer |
| 05:59.41 | three55ml | terrapen: It's a web program. |
| 05:59.49 | *** join/#asterisk dizzydiffi (dizzydiffi@adsl-70-240-241-42.dsl.hstntx.swbell.net) |
| 05:59.50 | terrapen | ok, that's what i thought. |
| 05:59.59 | dizzydiffi | yo peps |
| 06:00.05 | terrapen | then it's ________ (insert tool kit used by your browser) |
| 06:00.41 | three55ml | I'm not asking for "free advice", if doing it through config files is easier, I can go back to that method. |
| 06:00.45 | JerJer | thefallen: so why are you bloating it? |
| 06:00.47 | three55ml | pcm: Which line is it? |
| 06:00.56 | netsurfer | 5 hours to compile a kernel lmao |
| 06:01.08 | dizzydiffi | people anyone done anything on Openh323 |
| 06:01.16 | three55ml | Chuji: http://www.spirestudios.com/misc/pbx_screenshot.jpg |
| 06:01.16 | JerJer | a reload is only required when a change is made |
| 06:01.30 | pcm | three: what's the category ... need to understand to avoid the dump |
| 06:01.52 | three55ml | JerJer: I know, and that's how I was doing it at first. I was assuming that it * became temporarily unavilable during a reload (even if for a few seconds). |
| 06:02.00 | Chuji | three55ml : Hmm, seems like I saw this already. Did you post something to mailing list recently? |
| 06:02.01 | JerJer | why are you assuming? |
| 06:02.03 | three55ml | pcm: /msg me |
| 06:02.09 | JerJer | test |
| 06:02.12 | terrapen | hmmmm...austin... |
| 06:02.16 | JerJer | then test some more |
| 06:02.22 | JerJer | then have someone else test |
| 06:02.36 | dizzydiffi | lalalala |
| 06:02.40 | JerJer | then make assumptions based on imperial data |
| 06:02.42 | three55ml | Chuji: No, I was on here sometime last week |
| 06:02.59 | Chuji | three55ml : Ah, IC. looks good though. |
| 06:03.52 | three55ml | JerJer: Look, I'm not asking for you to solve the worlds problems. I was asking for an opinion. I've developed apps that've been deployed to hundres of thousands, I don't need a lesson in design. You don't have to be an elitist about a few questions in a DEVELOPER channel. |
| 06:04.22 | terrapen | developer channel? |
| 06:04.33 | three55ml | hehe, oops :) #asterisk-dev? |
| 06:04.37 | Juggie | three55ml, what are your questions? |
| 06:04.48 | dizzydiffi | juggie can u help |
| 06:05.01 | dizzydiffi | have u done anything with Openh323 |
| 06:05.03 | terrapen | three, where are you in SA? |
| 06:05.06 | terrapen | who do you work for? |
| 06:05.16 | netsurfer | anyone: what binds to port 4569 in * ? |
| 06:05.17 | JerJer | thefallen: so you can write code? big deal |
| 06:05.19 | three55ml | Juggie: I was having some issues with Realtime support. |
| 06:05.29 | JerJer | this is irc, deal with it |
| 06:05.43 | Juggie | realtime is not stable |
| 06:05.48 | JerJer | nor sane |
| 06:05.54 | three55ml | terrapen: Actually I live in Austin, just visiting down here right now. I own my own company, doing lots of things. |
| 06:05.56 | Chuji | nor necessary |
| 06:06.02 | Juggie | thats why its not in 1.0.x |
| 06:06.11 | terrapen | ah. |
| 06:06.21 | three55ml | Juggie: When Asterisk is reload through the CLI, are there any adverse affects in terms of temporarily causing an outage at all? |
| 06:06.25 | three55ml | reloaded |
| 06:06.26 | terrapen | Austinite visiting SA. heh. |
| 06:06.41 | three55ml | terrapen: You in TX? |
| 06:06.46 | Juggie | three55ml, not that i know of |
| 06:06.48 | JerJer | how about trying it for yourself and knowing what happens? |
| 06:07.07 | redder86 | JerJer: who do you use for credit card processing? |
| 06:07.09 | terrapen | i am in san antonio. |
| 06:07.11 | JerJer | pick up a phone, make a call and type reload in the asterisk cli |
| 06:07.11 | Juggie | i am not sure how it affects calls in progress |
| 06:07.40 | NormAst | three55ml: yea... sometimes you system will stop for a few seconds and calls won't go though. |
| 06:07.49 | JerJer | NormAst: um no |
| 06:07.58 | NormAst | jerjer: um yea. |
| 06:08.12 | NormAst | It effects sip phones. |
| 06:08.14 | JerJer | with 10,000 sip enties |
| 06:08.35 | JerJer | no single asterisk box needs even half that many |
| 06:09.01 | JerJer | it is all about management |
| 06:09.23 | NormAst | okay.. I am just saying... I have seen * pause when you type reload, and nothing happens for a few seconds. |
| 06:09.24 | pcm | reload can affect calls! |
| 06:09.36 | Juggie | NormAst, me too |
| 06:09.41 | pcm | it does to most of the modules: unreload and reload |
| 06:09.47 | JerJer | ok pausing on the cli means you have a dns entry that cannot be resolved |
| 06:09.51 | Juggie | k |
| 06:10.09 | *** join/#asterisk Inv_arp (junya@adsl-8-230-175.mia.bellsouth.net) |
| 06:10.16 | JerJer | but it does not effect call processing, unless you have a ton of user/peers |
| 06:10.21 | NormAst | jerjer... where would this dns entry be in ??? IAX.CONF? |
| 06:10.24 | pcm | it's not about pausing |
| 06:10.37 | Luke-Jr | pcm: I decided to email the person at sipmedia (the remote PBX) who wrote some * instructions and ask him for advice :) |
| 06:10.38 | JerJer | any VoIP channel |
| 06:10.46 | JerJer | ....that needs to lookup a DNS name |
| 06:10.50 | pcm | if you have MANY calls over IAX then guess what ... all those packets come through networking_thread |
| 06:11.02 | JerJer | pcm: and? |
| 06:11.02 | Luke-Jr | pcm: Hopefully they can figure it out, if they don't want me to ask for my within-14-days-refund :) |
| 06:11.03 | harryvv | I am looking for another wholsale voip carrier other then voipjet that would be closer to BC canada or located in canada? |
| 06:11.17 | harryvv | need to reduce the echo |
| 06:11.21 | pcm | luke: the best way ... they might do authorization some other way ... |
| 06:11.30 | Chuji | harryvv : iax.cc |
| 06:11.39 | harryvv | where is that located at? |
| 06:11.39 | Luke-Jr | pcm: Does * support other ways? |
| 06:11.41 | Chuji | ~iax.cc |
| 06:12.15 | NormAst | Anyways.. I am off to bed... |
| 06:12.21 | Chuji | Me too |
| 06:12.27 | Chuji | thsis is way too late for me |
| 06:12.28 | JerJer | i see how it is |
| 06:12.42 | JerJer | make statements and can't back them up |
| 06:12.46 | harryvv | wow thats not bad at all 15 hops and only 80 ms delay. |
| 06:12.50 | NormAst | no... |
| 06:13.01 | NormAst | You had 3 people that had the same problems.. That's backup. |
| 06:13.12 | JerJer | bleh |
| 06:13.21 | JerJer | i abuse the hell out of asterisk |
| 06:13.35 | JerJer | and have seen not one single issue with reloading |
| 06:13.46 | JerJer | even with 200 active channels |
| 06:14.01 | JerJer | so go away |
| 06:14.47 | NormAst | harryvv: try sip.tollfreetelecom.ca |
| 06:14.53 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 06:15.29 | redder86 | JerJer: who do you use for on-line credit card processing? |
| 06:15.47 | JerJer | authorize.net |
| 06:15.56 | redder86 | JerJer: have they been good to you? |
| 06:16.01 | JerJer | yes |
| 06:16.15 | JerJer | go thru charge.com and tell em i sent you :) |
| 06:16.23 | redder86 | JerJer: ;-) |
| 06:16.31 | JerJer | just force charge.com to use authorize.net instead of linkpoint |
| 06:17.03 | BoRiS | how? |
| 06:17.26 | harryvv | http://sip.tollfreetelecom.ca/ site does not exist |
| 06:18.31 | NormAst | Site's not up yet.. |
| 06:19.16 | NormAst | harryvv: What kind of ping times from BC? |
| 06:19.44 | harryvv | iax.cc is 20 ms faster then sip.tollfreetelecom.ca |
| 06:20.18 | NormAst | your getting 100ms... |
| 06:20.31 | harryvv | iax.cc goes to this site |
| 06:20.33 | harryvv | <PROTECTED> |
| 06:20.58 | Juggie | harryvv, one is also in canada |
| 06:21.01 | harryvv | siptollfree 10 h66-38-197-135.gtconnect.net (66.38.197.135) 76.942 ms 75.745 ms 77.177 ms |
| 06:21.01 | pcm | JerJer: ok, didn't look in the chan_iax2 recently |
| 06:21.08 | Juggie | and one is in usa |
| 06:21.35 | JerJer | ugg group telecom |
| 06:21.45 | NormAst | thanks harryvv.. |
| 06:21.58 | BoRiS | GT...lol |
| 06:22.01 | pcm | jerjer: but reload calls relaod_config and it prunes peers and users ... it locks userl.lock and peerl.lock ... and when there are registrations requests it goes and wants to find the peer (find_peer) or use ... and it locks same locks |
| 06:22.15 | harryvv | What is peak phone traffic time for these services anyway? 8 pm? |
| 06:22.19 | pcm | jerer: of course that shouldn't take that long but you might skip a few frames for a few calls ... |
| 06:22.23 | Juggie | gt runs canada along with rogers and bellnexia |
| 06:22.52 | JerJer | so one registration attempt it lost? the problem is? |
| 06:23.08 | pcm | no, the whole networking thread is stopped for some time ... |
| 06:23.19 | pcm | so you don't send the frames that are scheduled and you don't receive anything ... |
| 06:23.37 | JerJer | i have not seen that to be a problem |
| 06:23.41 | Juggie | pcm, patch it |
| 06:23.45 | pcm | and registrations come pretty often so the propability that this will happen when you do reload is more then 0 |
| 06:24.03 | pcm | JerJer: cause you have to hear that ... not see |
| 06:24.07 | JerJer | we reload highly loaded boxes without issue |
| 06:24.26 | JerJer | doing lots of IAX and some SIP |
| 06:24.26 | pcm | JerJer: well there is no other issue then loosing a few voice frames now and then |
| 06:25.50 | shido6 | ive seen asterisk pause but since I was ussing the box for a router as well my switch lights remained highly active as I was still downloading iso images |
| 06:25.55 | shido6 | on a celeron |
| 06:26.01 | shido6 | with 8k cache |
| 06:26.19 | shido6 | but no calls dropped |
| 06:26.24 | JerJer | don't run celeron |
| 06:26.27 | pcm | calls don't drop hehe |
| 06:27.00 | JerJer | i have asterisk running on a wrap board |
| 06:27.04 | shido6 | now my box is overkill for the 2 calls I place |
| 06:27.17 | JerJer | which is my edge device as well |
| 06:27.36 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
| 06:27.43 | dontmsgme | How do you buy a DID from Nufone |
| 06:27.50 | dontmsgme | Do you have to email a request? |
| 06:27.53 | shido6 | click on let me in sign into your account |
| 06:27.56 | shido6 | click on DID |
| 06:27.59 | shido6 | order new did |
| 06:28.05 | dontmsgme | There is no DID link |
| 06:28.06 | shido6 | and fill out the short form and BOOM check your email |
| 06:28.09 | shido6 | u have a new did |
| 06:28.26 | dontmsgme | Hah show me that link |
| 06:28.29 | shido6 | DIDs |
| 06:28.35 | shido6 | next to history |
| 06:28.50 | shido6 | before before make payment |
| 06:29.02 | JerJer | below the logo |
| 06:29.07 | shido6 | Account Summary History DIDs Make Payment Preferences Logout |
| 06:29.23 | shido6 | click on DIDs |
| 06:29.31 | shido6 | then click on the box to the right that says order new did |
| 06:29.38 | dontmsgme | "This is not implemented yet, sorry" |
| 06:29.52 | *** join/#asterisk freat[laptop] (~freat[lap@c-67-173-11-177.client.comcast.net) |
| 06:29.58 | shido6 | tollfree ? |
| 06:30.18 | shido6 | u have a weird web cache server running over there? |
| 06:30.19 | shido6 | refresh |
| 06:30.47 | shido6 | u can provision the toll free's |
| 06:31.04 | shido6 | no charge for the number just for your usage if its a regular 8xx and not a vanity number |
| 06:31.13 | shido6 | at this time |
| 06:31.27 | shido6 | the michigan dids, we're working on. |
| 06:31.47 | shido6 | got any digium gear in that box of yours? |
| 06:33.41 | shido6 | thow some digium gear in there for a reliable timing source and to support the cause, we need Digium to flourish, buy a card or two... stop messin with softphones get a device your lady can gossip on |
| 06:33.45 | shido6 | +r |
| 06:37.51 | *** join/#asterisk expressfone1 (~expressfo@62-15-97-163.inversas.jazztel.es) |
| 06:37.58 | expressfone1 | morning |
| 06:38.11 | JerJer | moo |
| 06:38.46 | expressfone1 | jerjer is cha_h323 working ok?? |
| 06:38.59 | expressfone1 | in 1.0.5 |
| 06:40.03 | expressfone1 | las time test it was no tinclude in stable * and have a lot of problems |
| 06:41.31 | JerJer | stable is not supported |
| 06:41.39 | JerJer | if u must use H.323 you must use head |
| 06:42.00 | xkev | who does kpfleming irc as? |
| 06:43.32 | *** join/#asterisk eipi (~eipi@OL128-44.fibertel.com.ar) |
| 06:46.00 | brc_ | hola jerjer |
| 06:46.16 | brc_ | ~seen kpfleming |
| 06:46.17 | jbot | kpfleming <~chatzilla@ip68-3-230-141.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 3d 2h 59m 54s ago, saying: 'nahh, got plenty to do this weekend to'. |
| 06:46.24 | Rowter | its possible to make a link between a panasonic and a digium card between E1, and work with panasonic channel bank? am testing this tomorrow, has anyone work this out? |
| 06:46.25 | xkev | ah, as kpfleming |
| 06:46.29 | Corydon76-home | ~seen mogorman |
| 06:46.30 | jbot | mogorman <~mogorman@dhcp-162.digium.com> was last seen on IRC in channel #asterisk, 11h 31m 7s ago, saying: 'yes'. |
| 06:46.30 | xkev | I checked whowas |
| 06:46.42 | brc_ | woah |
| 06:46.51 | brc_ | looks like he's from phoenix |
| 06:47.24 | xkev | #2615 was closed saying he was taking over the new message stuff, but haven't seen anything for app_queue. |
| 06:47.50 | xkev | not that I'm complaining, but I do want to roll somethin like that |
| 06:48.20 | brc_ | whatchya talking about? |
| 06:48.27 | brc_ | I haven't been in there for days |
| 06:49.10 | Corydon76-home | Elephants remember... |
| 06:49.33 | brc_ | no... |
| 06:49.37 | brc_ | I don't remember |
| 06:49.48 | eipi | hi |
| 06:49.53 | eipi | who mantains voip wiki? |
| 06:51.21 | terrapen | any ideas why this doesn't work as expected: |
| 06:51.22 | terrapen | exten => 8668177667,1,SetCIDName(eBay ${CALLERID}) |
| 06:51.23 | terrapen | ? |
| 06:52.27 | *** join/#asterisk netsurfer (netsurfer@82.133.103.189) |
| 06:52.37 | alakdan | is it possible to use gsm codec on iaxy? or just ulaw or adpcm? |
| 06:52.45 | JerJer | no gsm |
| 06:52.56 | alakdan | JerJer: thanks :) |
| 06:53.33 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 06:53.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:54.48 | brc_ | denon, |
| 06:54.50 | brc_ | hello! |
| 06:58.51 | JerJer | moo |
| 06:59.13 | *** join/#asterisk usam (~usam@203.156.41.110) |
| 06:59.19 | WilliamK | hey Jer |
| 07:01.34 | wasim | m00 |
| 07:02.12 | wasim | did you guys have a super bowl on the weekend? |
| 07:02.24 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 07:03.53 | WilliamK | yup |
| 07:08.41 | Corydon76-home | For once, the game was good and the commercials sucked. |
| 07:09.25 | wasim | close game? |
| 07:13.46 | Corydon76-home | It was, until the last 5 minutes |
| 07:14.26 | Corydon76-home | It was all over, when the interception occurred |
| 07:15.10 | wasim | circa 1986, me thinks ... mr simms |
| 07:15.24 | Corydon76-home | Philadelphia came back to score another touchdown, but they simply ran out of time... |
| 07:15.56 | Corydon76-home | But, yes, very good game |
| 07:16.23 | wasim | thats good, the 5 i saw between 1991-1996 were quite one sided |
| 07:17.06 | terrapen | wasim, where the heck are you? |
| 07:17.16 | wasim | terrapen: east of afghanistan |
| 07:17.24 | terrapen | pakistan? |
| 07:17.29 | terrapen | india? |
| 07:18.10 | wasim | the land of the pure! |
| 07:18.36 | terrapen | hmmmmmmm |
| 07:18.39 | wasim | i think i can hear the good ol' quaid turning in his grave of the travesty pakistan has become |
| 07:18.51 | terrapen | ok, so you're indian :) |
| 07:19.21 | wasim | lol ... no, my grandparents were, but then they migrated in 1947 west |
| 07:19.46 | terrapen | Nigeria? |
| 07:19.53 | Corydon76-home | Kashmir? |
| 07:20.35 | wasim | nigeria is higher that us on the top ten most corrupt countries of the world |
| 07:20.48 | *** join/#asterisk pelotas (~alek@host-ip130-216.crowley.pl) |
| 07:21.34 | wasim | Corydon76-home: Kashmir is really beautiful, i'm glad we've got somewhat peaceful overtures between the two countries, its a lovely place indeed |
| 07:22.36 | dontmsgme | In nufone when you choose DID it says "Svc not ready" is that something you can do now or is it upcoming |
| 07:22.40 | Corydon76-home | Yah, I have family that grew up somewhere in that general region |
| 07:24.17 | wasim | Corydon76-home! |
| 07:24.32 | Corydon76-home | My great aunt spoke Marathi fluently |
| 07:25.33 | terrapen | i wonder how well (or not) IAX2 will run over OpenVPN |
| 07:25.39 | terrapen | anybody tried? |
| 07:25.45 | Corydon76-home | terrapen: runs fine |
| 07:26.12 | Corydon76-home | As long as you set up your routes correctly (using iproute2, with a src parameter) |
| 07:26.13 | terrapen | i want to put one of these Polycoms at home |
| 07:26.23 | terrapen | iproute2? what is that? |
| 07:26.47 | Corydon76-home | i.e. the 'ip route' command, not the 'route' command |
| 07:27.06 | wasim | terrapen: your polycom talks IAX2? |
| 07:27.07 | terrapen | uh... |
| 07:27.13 | terrapen | wasim, oh, doh |
| 07:27.15 | terrapen | SIP |
| 07:27.32 | terrapen | cory, the phone will attempt to reach the * server's IP |
| 07:27.32 | Corydon76-home | i.e. ip route 10.1.1.1 via 10.1.1.2 src 192.168.1.1 |
| 07:27.55 | terrapen | i've never heard of an 'ip' command |
| 07:27.58 | terrapen | is this an * thing? |
| 07:28.13 | terrapen | my router handles the routing. |
| 07:28.16 | terrapen | via a static route |
| 07:28.18 | Corydon76-home | Assuming you're using 192.168 for your internal addresses and 10.* for your openvpn addresses |
| 07:28.38 | Corydon76-home | Nope, it's the command structure that's been around since kernel version 2.2 |
| 07:28.39 | terrapen | dude, i know how to do the routing :) |
| 07:28.47 | Corydon76-home | The route command is now deprecated |
| 07:28.47 | terrapen | i don't run linux anywhere except on the * server |
| 07:29.00 | Corydon76-home | Have you openvpn on the * server? |
| 07:29.50 | terrapen | nope, it runs on the OpenVPN server...but when the * server attempts to contact 10.40.0.0/24 (my home network), it will do so via the OpenVPN server |
| 07:29.52 | Corydon76-home | If your iax server is a vpn endpoint, you have to do the special routing command |
| 07:30.05 | terrapen | route add -net 10.40.0.0/24 10.5.0.1 |
| 07:30.17 | Corydon76-home | Ugh, stop that... Old, ugly command... |
| 07:30.25 | terrapen | not on BSD, it isn't :) |
| 07:30.34 | Corydon76-home | Oh |
| 07:30.35 | terrapen | and route(1) is beautiful :) |
| 07:30.43 | Corydon76-home | BSD doesn't support setting the src address |
| 07:30.54 | Corydon76-home | Thankfully, Asterisk doesn't run all that well on BSD |
| 07:31.06 | terrapen | tsk |
| 07:31.25 | terrapen | why in the world would you want to set a source address in a static route? |
| 07:31.27 | Corydon76-home | No, really, it doesn't support setting the src address |
| 07:32.05 | Corydon76-home | Consider if you have a PTP connection that uses Martian addresses, such that the connection doesn't waste real IPs |
| 07:32.16 | Corydon76-home | like over a frame relay |
| 07:32.23 | terrapen | if you don't want a machine using your route, the packet filter should prevent it from doing so |
| 07:32.39 | terrapen | we use 10.254.254.0/24 for that, corydon |
| 07:32.45 | Corydon76-home | You want packets originating on the endpoint, going over the frame relay to have the public IP, not the tunnel IP |
| 07:33.03 | terrapen | and they do have the public IP. |
| 07:33.12 | Corydon76-home | Using what command to route it? |
| 07:33.41 | *** join/#asterisk freat[laptop] (~freat[lap@c-67-173-11-177.client.comcast.net) |
| 07:33.44 | Corydon76-home | Now, machines that aren't on the endpoint are no problem... they already have the real IP and don't know anything about the tunnel |
| 07:34.03 | terrapen | route add -net 10.40.0.0/24 10.254.254.2 |
| 07:34.06 | terrapen | and on the other end |
| 07:34.11 | ManxPwr | Voice over Frame Relay has....issues. |
| 07:34.15 | terrapen | route add -net 10.0.0.0/24 10.254.254.1 |
| 07:34.27 | Corydon76-home | ManxPwr: we're not talking about voice, we're talking about packets |
| 07:34.43 | terrapen | so, on your client machine, 10.40.0.28, you just do this: |
| 07:34.49 | Corydon76-home | terrapen: but your packets will then still have a src address on the 10.* network |
| 07:34.51 | terrapen | route add -net 10.0.0.0/24 10.40.0.1 |
| 07:34.57 | terrapen | (if 10.40.0.1 is the openvpn server) |
| 07:35.01 | Corydon76-home | Not a real IP |
| 07:35.10 | terrapen | all i use is 10/8 |
| 07:35.20 | Corydon76-home | Hence, not routable on the Internet |
| 07:35.45 | Corydon76-home | Ah, you don't have any real IPs to deal with? |
| 07:35.52 | terrapen | no, each endpoint handles its internet-bound traffic |
| 07:35.56 | terrapen | and it works just fine |
| 07:36.02 | terrapen | both of my endpoints have public IPs |
| 07:36.22 | terrapen | trust me, it works fine :) |
| 07:36.30 | terrapen | you don't need a src address in the static route |
| 07:36.34 | Corydon76-home | Okay, and how does the endpoint with the real public IP going across the tunnel have the real IP? |
| 07:36.48 | terrapen | ? |
| 07:37.00 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 07:37.04 | terrapen | what real IP? |
| 07:37.17 | Corydon76-home | What's your real IP? |
| 07:38.07 | terrapen | { 10.0.0.0/24 } --- [OPENVPN SERVER] ---- { internet } --- [OPENVPN SERVER] --- { 10.40.0.0/24 } |
| 07:38.30 | terrapen | works just fine |
| 07:38.30 | terrapen | anything on 10.0.0.0/24 can ping anything on 10.40.0.0/24 just by adding a static route |
| 07:38.30 | Corydon76-home | I'm not talking about openvpn here |
| 07:38.35 | terrapen | route add -net 10.40.0.0/24 10.0.0.1 |
| 07:38.46 | Corydon76-home | What's your real IP across your frame relay? |
| 07:39.02 | terrapen | i have public and private network connections, corydon |
| 07:39.11 | terrapen | my network is very complex |
| 07:39.12 | Corydon76-home | Ah... |
| 07:39.12 | *** join/#asterisk Yoda-BZH (~yoda-bzh@84.4.50.146) |
| 07:39.34 | Corydon76-home | So you're probably using Cisco to connect, right? |
| 07:39.39 | terrapen | we have 3 Ciscos |
| 07:39.49 | terrapen | and several OpenBSD firewalls |
| 07:39.59 | Corydon76-home | Nevermind then. You're not using either Linux or BSD to do the tunnelling |
| 07:39.59 | terrapen | a few Soekris boxes running m0n0wall... |
| 07:40.14 | terrapen | no, each network is NAT'ed to the public internet |
| 07:40.40 | Corydon76-home | Our connection, until recently, was going over a T100P |
| 07:40.53 | terrapen | but any machine on any one of my networks (incl. home) can ping any other machine |
| 07:40.58 | Corydon76-home | Now it's going over two ports on a TE405P |
| 07:41.04 | terrapen | we have 3 T1's |
| 07:41.24 | terrapen | our stores get their internet access from a private point-to-point T1 to our co-lo cabinets |
| 07:41.31 | Corydon76-home | I can also ping any machine on my home or work networks |
| 07:41.46 | terrapen | my home machines are on cablemodem |
| 07:41.52 | Corydon76-home | Ditto |
| 07:43.16 | terrapen | actually... |
| 07:43.18 | Corydon76-home | The thing is, we need the source address to be correct, because our iax machine has 5 different addresses... |
| 07:43.19 | alakdan | for provisioning iaxy s it is possible to just put both |
| 07:43.20 | alakdan | codec: ulaw |
| 07:43.20 | alakdan | codec: adpcm |
| 07:43.20 | alakdan | or just only one? |
| 07:43.27 | terrapen | machines on the work network don't even have to add static routes to reach my home network |
| 07:43.35 | terrapen | they just send the packets to their router |
| 07:43.41 | terrapen | err gateway |
| 07:43.43 | Corydon76-home | terrapen: yeah, same here |
| 07:43.46 | eipi | I added an IAX user into iax.conf, how i can assign an extesion number to that iax client? |
| 07:43.49 | terrapen | which has the static route to the openvpn tunnel |
| 07:43.59 | terrapen | eipi, in extensions.conf |
| 07:44.06 | eipi | ok |
| 07:44.07 | Corydon76-home | terrapen: it's just basic networking that everything goes through the routers |
| 07:44.26 | terrapen | wierd....m0n0wall's WoL does not seem to be waking up my Apple |
| 07:45.01 | three55ml | eipi: Add an extension in whatever context you want |
| 07:45.26 | three55ml | Whatever your default is if you want it accessible from everything |
| 07:45.49 | Corydon76-home | terrapen: but the problem with having openvpn on your iax machine is that iax needs all packets to come to and from the same local address... |
| 07:46.13 | Corydon76-home | which is why the src parameter to the ip route command solves the issue |
| 07:46.18 | terrapen | yep. |
| 07:46.20 | terrapen | i hear ya |
| 07:46.34 | terrapen | but its silly to run OpenVPN on your PBX machine |
| 07:46.53 | Corydon76-home | No, not really |
| 07:47.06 | terrapen | OpenVPN, in my opinion, is something that should be run on a firewall, which should be kept seperate from a PBX for security reasons |
| 07:47.10 | Corydon76-home | Our PBX needs a public IP anyway, for client registrations |
| 07:47.22 | terrapen | i never run anything on the firewall that could hurt me if comprimised |
| 07:47.40 | terrapen | i don't want anyone on the puiblic internet reaching my PBX |
| 07:47.44 | terrapen | no reason for them to. |
| 07:47.52 | Corydon76-home | You have different priorities |
| 07:47.55 | terrapen | maybe someday, i will open up an IAX port |
| 07:48.04 | terrapen | if i become a road warrior again |
| 07:48.29 | Corydon76-home | Our customers save on long distance by calling us directly over voip |
| 07:48.37 | terrapen | gotchya. |
| 07:49.09 | terrapen | our situation is different i guess....we have unique machines for almost every function here |
| 07:49.14 | Corydon76-home | International long distance is expensive... 3 digit dialling for help is less so |
| 07:49.20 | terrapen | yep |
| 07:50.17 | terrapen | i wonder how * would run on my G5 (Mac OS X) |
| 07:50.22 | terrapen | i don't need any Zap stuff |
| 07:50.25 | terrapen | just SIP/IAX2 |
| 07:50.30 | Corydon76-home | Actually, we're probably about ready to remove the internal IP from our IAX server... |
| 07:50.43 | Corydon76-home | and just run it completely in the DMZ |
| 07:52.17 | Corydon76-home | Time for bed... |
| 07:55.41 | *** join/#asterisk godsmoke (~godsmoke@66-108-159-216.nyc.rr.com) |
| 07:56.00 | godsmoke | someone have voicepulse connect here? |
| 07:56.26 | three55ml | godsmoke: Yeah, what's up? |
| 07:58.11 | alakdan | anyone using iaxy? |
| 08:00.47 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 08:01.37 | eipi | anyone know any free chine iax server? :D |
| 08:01.45 | eipi | chinese |
| 08:05.03 | harryvv | free server ohh man now is labor over there being forced to work for free now? |
| 08:06.06 | sskyles | What do I need to do to compile Asterisk on the Mac? |
| 08:06.36 | sskyles | Do I need a new bison? |
| 08:06.39 | harryvv | www.voip-info.org |
| 08:10.18 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 08:10.19 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:18.31 | harryvv | give netsurfer a hand he just made a redundent asterisk server work when one failed. |
| 08:18.40 | harryvv | the other one took over. |
| 08:19.17 | netsurfer | heh.. not exactly.. apache yes.. not tried asterisk yet :oP |
| 08:20.11 | *** join/#asterisk freat[laptop] (~freat[lap@c-67-173-11-177.client.comcast.net) |
| 08:20.50 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 08:21.34 | *** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62) |
| 08:21.46 | zoa | damn out of office replies on the mailinglists |
| 08:22.00 | zoa | there is also someone sending me virus warnings |
| 08:22.17 | harryvv | usually end users who make the same mistakes |
| 08:22.18 | harryvv | :) |
| 08:22.18 | zoa | because it things my gpg key is an pasword protected executable |
| 08:22.37 | Mike_TK | zoa: I think it's not only your problem ;) |
| 08:23.03 | Mike_TK | zoa: Someone just put your address into 'From' and sending a spam. |
| 08:23.03 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 08:24.14 | *** join/#asterisk ltostain (~ltostain@i01v-30-28.d4.club-internet.fr) |
| 08:25.10 | *** join/#asterisk Mike_TK (~Mike_@213.180.245.62) |
| 08:25.13 | *** join/#asterisk ACiDV (~joel@122-68-181.dr.cgocable.ca) |
| 08:25.50 | zoa | no its not |
| 08:25.59 | zoa | it each time i send something to the asterisk-users |
| 08:26.00 | Makenshi | morning |
| 08:26.14 | zoa | some guys antivirus server things that my .asc file is a virus |
| 08:26.18 | zoa | because he cant unpack it |
| 08:31.16 | Mike_TK | zoa: Can be... sorry. |
| 08:31.59 | Mike_TK | So, any news in asterisk community? |
| 08:35.40 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:37.51 | wasim | Mike_TK: we fired one of the maids, and are waiting for the driver to come back from dubai |
| 08:38.32 | wasim | Mike_TK: she was causing deadlocks and had to go |
| 08:38.51 | *** join/#asterisk netsurfer (netsurfer@82.133.103.189) |
| 08:38.57 | Mike_TK | wasim: :) |
| 08:38.59 | ACiDV | I have a TE410 card set to pri_cpe (bchan=1-23, dchan=24) and each time I plug in a cable (cross over) Asterisk halt with a Segmentation Fault... kernel 2.6.10, latest cvs (zaptel, libpri, asterisk*) :( not try yet with "non-cvs" version |
| 08:39.26 | wasim | ACiDV: what does ztcfg show? |
| 08:39.27 | Mike_TK | ACiDV: Start with -g and check core dump. |
| 08:39.38 | ACiDV | I try immediatly |
| 08:41.01 | *** part/#asterisk ipso (~ipso@d207-81-249-35.bchsia.telus.net) |
| 08:41.23 | ACiDV | wasim: ztcfg -v show : SPAN1..4: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) ... 100 channels configured. (TE410+TDM22B) |
| 08:42.22 | ACiDV | doh, I'm back in Quebec and the server is in Toronto... I will call the guy tomorrow to plug it and I will log all information and come back here :| |
| 08:42.33 | wasim | ACiDV: what alarms do you have in zttool, without asterisk up |
| 08:43.19 | ACiDV | IIRC (10 hours ago), OK alarm for the Span 1 |
| 08:43.57 | eipi | what ports i need to be opened on my firewall to run SIP and IAX2? |
| 08:44.29 | ACiDV | eipi check wiki but 5060, 4569 and RTP port |
| 08:44.36 | ACiDV | ports |
| 08:44.46 | eipi | ok |
| 08:45.36 | ACiDV | send url in private |
| 08:47.32 | *** join/#asterisk twisted[work] (~twisted@twisted.active.supporter.pdpc) |
| 08:47.32 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 08:48.39 | *** part/#asterisk ltostain (~ltostain@i01v-30-28.d4.club-internet.fr) |
| 08:50.08 | libpcp | anyone has a config manual of sipura 3000? |
| 08:50.28 | libpcp | sipura website doesnt have a manual. |
| 08:50.44 | *** join/#asterisk Marouani (~ltostain@i01v-30-28.d4.club-internet.fr) |
| 08:51.12 | wasim | we really should have a neat farfon display at CEBIT ... |
| 08:54.05 | Marouani | Hi, When and where is CEBIT this Year ? |
| 08:57.48 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 08:57.52 | christo | morning guys |
| 08:58.41 | jerlique | I get about 0.5-1.0 sec delay when talking to internal extensions over a lan (cisco 2950 switch). I'm using Xten-Xten ulaw. Any clues.... |
| 09:02.05 | usam | hello, could some1 explian me what ip inspect really does? |
| 09:03.11 | satlink | jerlique: Had similar problem wit xpro. I tried to connect to asterisk through our external IP instead of internal, and de delay dissapeard.. no clue why.. |
| 09:04.02 | jerlique | not that definately using internal ips. there is no default route in this * box. |
| 09:13.09 | *** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz) |
| 09:13.29 | *** join/#asterisk sskyles (~sskyles@155.205.205.68.cfl.rr.com) |
| 09:16.37 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 09:18.07 | christo | woah |
| 09:18.39 | christo | I was a bit worried about that :) |
| 09:20.17 | djin | Well, asterisk-addons is next? |
| 09:20.26 | *** join/#asterisk adnans (~adnans@noterik2.demon.nl) |
| 09:21.46 | RoyK | christo: why not 1.0.5? |
| 09:21.48 | christo | djin - haha yeah - well remembered :) |
| 09:22.00 | christo | RoyK - I just took asterisk_stable from cvs |
| 09:22.18 | christo | coffee time... brb |
| 09:22.23 | djin | that's 1.0.5, christo. |
| 09:22.37 | RoyK | christo: don't do that |
| 09:22.39 | *** join/#asterisk Tornad (~Tornad@81.255.65.249) |
| 09:22.57 | RoyK | christo: asterisk cvs "stable" will become "stable" after released |
| 09:25.32 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 09:25.32 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:27.16 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 09:29.18 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 09:29.57 | christo | RoyK - don't get it |
| 09:30.05 | christo | isn't that the latest stable version? |
| 09:30.41 | RoyK | christo: yes |
| 09:30.43 | RoyK | christo: but |
| 09:30.49 | christo | I mean I took asterisk_stable from cvs and when I run it, it shows Asterisk 1.0.3 |
| 09:31.04 | RoyK | christo: fixes get put into cvs, then thogoughly tested, then RELEASED |
| 09:31.10 | RoyK | so get a release instead |
| 09:31.11 | RoyK | tar ball |
| 09:31.23 | christo | mmm I see |
| 09:31.41 | christo | so should I now do 'make uninstall' before I put the new RELEASED version in? |
| 09:32.00 | jerlique | who here has deployed agent logout? |
| 09:32.18 | RoyK | christo: just install the other |
| 09:32.23 | RoyK | it'll replace all the old stuff |
| 09:32.34 | RoyK | perhaps rm -rf /usr/lib/asterisk/modules by hand first |
| 09:32.39 | RoyK | to be sure no old shite is left |
| 09:33.55 | christo | okay RoyK thanks.. I'll go and look for the latest release tarball and do it that way - will I have to do the same for the libpri and zaptel stuff too? |
| 09:34.13 | RoyK | yes |
| 09:34.22 | christo | okay - back in a bit :) |
| 09:34.23 | djin | christo, stable cvs is working for me as well. Why change now? |
| 09:34.34 | christo | :/ er |
| 09:34.42 | christo | cos Roy said it might be a bit ropey.. |
| 09:34.45 | RoyK | djin: don't use CVS in production |
| 09:34.47 | RoyK | never |
| 09:34.48 | RoyK | ever |
| 09:34.54 | RoyK | I did that |
| 09:34.59 | djin | Well, I live dangerously. |
| 09:35.01 | RoyK | and suddenly my main SIP gateway was down |
| 09:35.03 | RoyK | oops |
| 09:35.12 | RoyK | a few thousand angry customers |
| 09:35.35 | djin | Well, that's a personal experience not always linked to cvs |
| 09:35.48 | christo | now now |
| 09:35.50 | christo | :) |
| 09:35.54 | djin | pre-latest stable (1.0.4) wasn't stable either. |
| 09:36.39 | djin | my advise, see if it works and know what to do if it doesn't. |
| 09:37.06 | RoyK | djin: that lasted a few hours before replaced by 1.0.5 |
| 09:37.16 | RoyK | djin: I'm still running 1.0.3 |
| 09:37.20 | christo | okay - I'll try with what I've got now, but I'lil be ready to replace it all in half an hour if I have probs |
| 09:38.04 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:38.18 | Delvar | morning |
| 09:38.30 | djin | RoyK, I agree on not using the latest technology, but apperently it worked for Christo and me. |
| 09:38.44 | RoyK | appearently |
| 09:40.02 | djin | RoyK, do you agree or are you spellchecking ;) |
| 09:40.48 | RoyK | :) |
| 09:41.22 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:41.26 | RoyK | I meant, appearently it works for you, and it might as well work for me, but the chances for something to fail if using CVS is far higher than compared to using a release |
| 09:41.30 | RoyK | that's what I meant |
| 09:41.32 | RoyK | "Cogito nullos ergo americano est" |
| 09:42.07 | talkwebhosts | Heading to vegas |
| 09:42.12 | talkwebhosts | see you all later |
| 09:43.07 | postel | djin: if he does spell check he needs to work a bit on it, the word he's looking for is "apparently" |
| 09:43.13 | christo | chaps, where can I find the asterisk-addons release which will go with asterisk-1.0.3 ? |
| 09:43.28 | djin | RoyK, I agree with you, but christo was quite happy his upgrade worked for him and I suggested not to risk failure by downgrading again. |
| 09:43.54 | djin | postel, lol |
| 09:43.55 | christo | aaaw - you guys still disagreeing about whether or not to use bleeding-edge ? :S |
| 09:44.15 | djin | yes, and you started it! |
| 09:44.25 | christo | sowie |
| 09:44.40 | postel | ets ow kei |
| 09:45.06 | djin | not much else to do here. |
| 09:45.11 | christo | heh |
| 09:45.26 | christo | so what about the right version if asterisk-addons? |
| 09:45.28 | djin | btw, did you 'checkout v1-0'? |
| 09:45.40 | christo | v1-0 of *? |
| 09:45.55 | djin | in the checkout command. |
| 09:45.57 | christo | yea I did ' cvs checkout asterisk-1.0_stable' |
| 09:46.26 | djin | cvs checkout -r v1-0 ?? |
| 09:46.28 | christo | then restarted asterisk with 'asterisk -rv' and lo! a new version :) |
| 09:46.34 | christo | I didn't use -r |
| 09:46.48 | christo | but it created all the folders anyway |
| 09:46.49 | RoyK | christo: ftp.digium.com |
| 09:46.50 | *** join/#asterisk heka (~fasada@82.114.68.126) |
| 09:51.41 | *** join/#asterisk Madkiss (madkiss@madkiss.staff.freenode) |
| 09:51.52 | Madkiss | Hi folks |
| 09:52.21 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:52.32 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 09:54.58 | Zeeek | hey now |
| 09:57.23 | djin | hi Zeeek. |
| 10:04.46 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 10:06.57 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
| 10:07.15 | shaZwaz | Hello room |
| 10:08.14 | *** join/#asterisk IronHelix (~irc@ool-182c8f9f.dyn.optonline.net) |
| 10:11.54 | Zeeek | . |
| 10:12.25 | *** join/#asterisk m4rc (m4rc@urtica.linuxnews.pl) |
| 10:12.32 | shaZwaz | hi Zeeek |
| 10:13.18 | Zeeek | GOOD ${VARIABLE} all |
| 10:16.26 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
| 10:17.43 | Madkiss | Could somebody point me to an asterisk how to, please? |
| 10:17.59 | netsurfer | ${greeting} Zeeek |
| 10:18.25 | netsurfer | Madkiss - www.voip-info.org |
| 10:20.21 | *** join/#asterisk mak_ (~mak@privat.ua-online.net) |
| 10:20.25 | mak_ | hi |
| 10:20.37 | Madkiss | netsurfer: Thanks |
| 10:22.23 | Zeeek | quiet night ans quiet stars |
| 10:25.26 | Madkiss | netsurfer: Let's say I just installed asterisk from the source -- what would be a good point to start now? |
| 10:26.47 | *** join/#asterisk lohelle (~post@213.184.212.218) |
| 10:27.43 | *** join/#asterisk mrproper_ (~hehe@CPE-203-45-98-127.nsw.bigpond.net.au) |
| 10:27.47 | datareactor | ~docs |
| 10:27.48 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 10:28.17 | datareactor | Madkiss try the following urls |
| 10:28.21 | mrproper_ | anyone have an idea as to why i get one way voice using an iax client with 2 nats (people at asterisk end can hear my iax softphone but i cant hear them) |
| 10:29.04 | Madkiss | datareactor: yes? |
| 10:29.26 | shaZwaz | ~seen omer |
| 10:29.28 | jbot | omer <~omer@202.147.174.177> was last seen on IRC in channel #asterisk, 43d 56m 35s ago, saying: 'ok..'. |
| 10:29.42 | datareactor | the links are pasted above my message |
| 10:29.46 | shaZwaz | ~seen omer_ |
| 10:29.47 | jbot | i haven't seen 'omer_', shaZwaz |
| 10:29.50 | Madkiss | You said "following" ;) |
| 10:30.02 | Zeeek | Check this OT Flash: http://www.adcritic.com/interactive/view.php?id=5927 scary and funny |
| 10:30.13 | Zeeek | Starter tutorial: |
| 10:30.13 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 10:30.13 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 10:30.13 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 10:30.13 | Zeeek | THE reference of the moment: |
| 10:30.13 | Zeeek | http://www.asteriskdocs.org |
| 10:30.32 | Zeeek | wow I'm very late in my answer |
| 10:30.33 | Madkiss | ah, great |
| 10:30.36 | Madkiss | thanks |
| 10:30.59 | Zeeek | Madkiss if you're not too mad... ^^^^^^^^^^^^^^^ |
| 10:31.28 | shaZwaz | ~seen n3o |
| 10:31.29 | jbot | n3o <~neo@202.142.189.86> was last seen on IRC in channel #asterisk, 19d 1h 12m 35s ago, saying: 'u understart my problem st ?'. |
| 10:32.23 | Zeeek | I'm having weird connection issues again with freenode |
| 10:33.09 | datareactor | Zeeek are you connected though dialup ? |
| 10:33.10 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 10:33.27 | Zeeek | no but the IRC server connection is weird for two days |
| 10:33.44 | Zeeek | big bursts of posts then nothing and then I'm suddenly not connected to it |
| 10:33.55 | Zeeek | so I can't enjoy all the wisdom by lurking |
| 10:34.06 | Madkiss | That's what you typically call "lag" |
| 10:34.30 | Zeeek | doesn't seem to affect the rest of the connections though |
| 10:35.02 | Delvar | try a different irc server? |
| 10:35.14 | *** join/#asterisk CMike (~a_mike@c-304171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 10:35.48 | Zeeek | but then I'll be naked!!!!! shocking! |
| 10:36.36 | *** join/#asterisk speakman (~speakman@c-38aa71d5.07-39-6f73641.cust.bredbandsbolaget.se) |
| 10:36.43 | speakman | Hi! anyone awake? |
| 10:36.47 | Zeeek | a few |
| 10:37.03 | shaZwaz | u mean in the real sense ? |
| 10:37.07 | djin | Zeeek, indeed a cool Flash,. |
| 10:37.18 | mrproper_ | anyone have an idea as to why i get one way voice using an iax client with 2 nats (people at asterisk end can hear my iax softphone but i cant hear them) |
| 10:37.28 | Zeeek | great piece of work, eh? :) |
| 10:37.44 | Zeeek | http://www.adcritic.com/interactive/view.php?id=5927 scary and funny |
| 10:38.43 | djin | Really cool and so true. |
| 10:38.55 | Zeeek | mrproper that's unusual with IAX and very usual with SIP |
| 10:39.05 | Zeeek | heh |
| 10:39.09 | speakman | Do anyone know if the DTMF CLIP has been fixed correctly? |
| 10:39.17 | mrproper_ | zeeek: tell me about it |
| 10:39.22 | Zeeek | the mailing list would know |
| 10:39.26 | CMike | hi all |
| 10:39.36 | mrproper_ | zeeek: not to mention the call hangs up after 30 seconds |
| 10:39.38 | Zeeek | remind me what DTMF CLIP is for? |
| 10:39.51 | christo | right - I'm in ftp.digium.com and I need the version of asterisk-addons which will go with asterisk 1.0.3. does that have to be asterisk-addons 1.0.3, or can I use 1.0.4 which is up there on the ftp site? |
| 10:40.07 | Zeeek | mrproper do you have a service like FWD you can test with using echo() ? |
| 10:40.10 | CMike | Swedish Callerid probably ... The CID come before the first ring on the line as DTMF |
| 10:40.32 | speakman | Zeeek: as CMike just said ;) |
| 10:40.44 | speakman | CMike: and with the polarity reversed.. |
| 10:40.50 | Zeeek | k |
| 10:41.27 | djin | christo, I don't think so. |
| 10:41.34 | Zeeek | I have my own CID issue with y newer siemens DECT phone. No CID connected to asterisk TDM400 FXS |
| 10:41.56 | christo | djin - to which one? |
| 10:42.13 | CMike | speakman: wel... almost .. a few more parameters.. |
| 10:43.25 | mrproper_ | Zeeek: i think i know the problem, im testing it via iax softphone--->nat--->internet--->nat--->asterisk---->oh323--->oh323 gateway--->isdn---->to my mobile |
| 10:43.39 | mrproper_ | zeeek: and its breaking down on the oh323 part |
| 10:43.39 | Zeeek | that's asking a lot ! |
| 10:44.41 | Madkiss | Zeeek: The first how to you pointed me to is nice indeed. Now I need just to know how to extend it for using a SIP-provider |
| 10:46.41 | CMike | speakman: it's actually this order for DTMF CID in sweden: polarityrevrsal then after 180-1000ms - D-(0+trunkcode+subscribernumber)-C |
| 10:46.42 | Zeeek | Madkiss what exactly do you need to know? |
| 10:46.52 | CMike | then polarityreversal + ring... |
| 10:47.06 | Zeeek | Madkiss - the author of the article gives his config files |
| 10:49.59 | *** join/#asterisk MacDeath (david@at.work.web.za) |
| 10:49.59 | christo | so anybody know what this means? : cdr_addon_mysql.c:109: `mysql_lock' undeclared (first use in this function) |
| 10:50.26 | shaZwaz | means u dont have mysql header files may be |
| 10:50.37 | christo | where should they be? |
| 10:50.44 | MacDeath | christo : i think you have the wrong mysqlclient libs |
| 10:50.45 | christo | is that the mysql-devel thing? |
| 10:50.55 | MacDeath | mysqlclient-devel |
| 10:51.20 | christo | oh |
| 10:51.41 | MacDeath | has anyone here used voipsupply.com? |
| 10:51.41 | shaZwaz | christo: u have the mysql-devel installed ? |
| 10:51.52 | Zeeek | . |
| 10:52.01 | christo | which: no mysql-devel in (/usr/local/sbin:/usr/sbin:/sbin: .... |
| 10:52.02 | christo | I guess not |
| 10:52.23 | MacDeath | christo : what os? |
| 10:52.29 | shaZwaz | install the mysql-devel package |
| 10:52.34 | christo | slackware - |
| 10:52.41 | christo | not sure where I can find the slackware package |
| 10:53.11 | shaZwaz | christo: u know how to install ? |
| 10:53.27 | christo | I know how to install slack packages.. I just can't find one for mysql-devel.. |
| 10:53.52 | shaZwaz | google it |
| 10:54.34 | shaZwaz | stfw |
| 10:54.38 | shaZwaz | ~stfw |
| 10:54.39 | jbot | extra, extra, read all about it, stfw is Search the F|_|cking Web: http://www.google.com/ |
| 10:54.48 | MacDeath | LOL |
| 10:54.51 | MacDeath | has anyone here used voipsupply.com? |
| 11:00.25 | *** join/#asterisk Tommmo (~tps@203.62.181.52) |
| 11:00.42 | Zeeek | how do I search the web? Where is this google? |
| 11:00.44 | Tommmo | is it possible to control silence suppression from within asterisk? I'm getting the beginning and end of sentences clipped and i want to turn it off? |
| 11:00.46 | Zeeek | ~google |
| 11:00.47 | jbot | it has been said that google is a search engine found at http://www.google.com/ |
| 11:00.55 | Tommmo | polycom phone -> asterisk -> cisco gateway -> PRI |
| 11:01.12 | Zeeek | ~google google |
| 11:02.08 | shaZwaz | Tommmo: u have echo canceler enabled ? |
| 11:02.13 | Zeeek | Tommmo VAD settings ? |
| 11:03.54 | Tommmo | yeah can i disable vad, would it be done on the gateway? |
| 11:04.50 | MrEntropy | has asterisk got any support for E1 cards that have DSPs onboard? |
| 11:05.13 | Tommmo | i've put "no vad" under my voice-port and dial-peers on the gateway but it's still clipping the first fraction of a second |
| 11:06.37 | djin | MrEntropy, define what boards and what you're looking for. |
| 11:07.15 | MrEntropy | djin: anything really, but i'd like to take the load off asterisk in terms of sound processing, what about Audiocodes TP240 for example? |
| 11:08.12 | christo | I've searched slackware.com and linuxpackages.net. I've also googled and searched on mysql.com - can anybody please suggest where I can get the mysql-devel package? |
| 11:08.14 | Zeeek | . |
| 11:08.42 | VoIPMasta | christo: it's in www.mysql.com |
| 11:08.57 | VoIPMasta | christo: when you download the source code the "devel package" is included |
| 11:09.25 | VoIPMasta | christo: it's only missing when you install mysql from .rpm/.deb and other kinds of precompiled bynaries |
| 11:09.43 | djin | As for as I know, there is support for Dialogic boards. But personally I prefer to invest in CPU. It's cheaper and I feel more confident using Digium boards (and support them financially ;) |
| 11:10.38 | christo | VoiPMasta - so I have to download the mysql sources - is there anything to compilebuild, or is it just some libraries that need to be there? |
| 11:11.39 | *** join/#asterisk godsmoke|laptop (~godsmoke@pool-151-202-70-31.ny325.east.verizon.net) |
| 11:12.03 | godsmoke|laptop | I'd like to find someone who can answer some questions about CPN and Caller ID and how information is handled in Asterisk |
| 11:12.10 | godsmoke|laptop | anyone confident in that area? |
| 11:18.52 | MrEntropy | djin: dialogic is a brand? |
| 11:20.12 | MrEntropy | djin: how many E1's can a single asterisk box do before you get desync problems? |
| 11:24.25 | Zeeek | what is CPN? |
| 11:24.35 | Madkiss | Is there anything that speaks against installing asterisk into /usr/asterisk? |
| 11:25.28 | kajtzu | Madkiss: no |
| 11:25.40 | kajtzu | Madkiss: your system, you can install * wherever you want :) |
| 11:26.11 | Madkiss | Well, you know -- some strange programs out there do odd things if you don't do what they want you to do |
| 11:26.45 | godsmoke|laptop | Zeeek: "Calling Party Number" |
| 11:27.05 | godsmoke|laptop | it's not the same as "Caller ID" |
| 11:27.20 | godsmoke|laptop | and yet again removed from "ANI" :) |
| 11:27.33 | MrEntropy | Madkiss: if you're talking about putting the binary in /usr/asterisk you'll have to either add it to your path or make a link. |
| 11:28.15 | Madkiss | MrEntropy: Why, what needs it? |
| 11:28.40 | MrEntropy | Madkiss: your shell if you want to run it without specifying the path |
| 11:28.52 | *** join/#asterisk RjT (blowme@dsl-204.215.240.220.dsl.comindico.com.au) |
| 11:30.37 | *** join/#asterisk Martohtar (Martohtar@82.196.218.130) |
| 11:30.49 | RjT | do i need to use special hardware or can i use a modem card and a normal phone ? |
| 11:31.34 | godsmoke|laptop | RjT: to do what? |
| 11:31.53 | RjT | make phonecalls through broadband |
| 11:32.09 | RjT | i dont understand how asterisk exactly |
| 11:32.10 | christo | VoiPMasta - do I have to compile all of mysql just to get the devel stuff? |
| 11:32.13 | RjT | just found out about it |
| 11:32.32 | RjT | asterisk works exactly |
| 11:33.07 | godsmoke|laptop | ok ... well, if you're just looking at "basic" voip -- asterisk is not going to come into play |
| 11:33.24 | godsmoke|laptop | if you're interested in managing more complicated features, or multiple lines through a PBX -- asterisk is a good option |
| 11:33.46 | RjT | oh ok what would u recomend > |
| 11:33.57 | godsmoke|laptop | well, you're being really vague |
| 11:35.00 | Madkiss | MrEntropy: I want to put _everything_ into /usr/asterisk |
| 11:35.10 | RjT | ok is it posible to use voip with normal phone system ? |
| 11:35.32 | godsmoke|laptop | RjT: sort-of ... yes |
| 11:35.42 | Zeeek | ok. |
| 11:36.12 | godsmoke|laptop | what you'll do, is contact a VoIP provider ... who will probably provide you with a "VoIP adapter" which will manage your phone calls through the internet, and allow you to plug in normal phones |
| 11:36.15 | RjT | ok thats what im interest in lol , ive only just read about voip so i dont really understand that much |
| 11:36.53 | godsmoke|laptop | Vonage is probably the most well-known residential provider ... they can answer questions like that ... considering this channel is supposed to be about Asterisk |
| 11:36.56 | MrEntropy | Madkiss: go right ahead, i don't think anything terrible will happen, it'll just be hell annoying =) |
| 11:37.54 | Madkiss | erm ... wait, there are no configure-scripts? |
| 11:38.01 | Madkiss | Then how the heck do I tell it where to go? |
| 11:38.11 | *** join/#asterisk waddy (waddy@66.90.92.190) |
| 11:42.43 | *** join/#asterisk speakman (~speakman@c-38aa71d5.07-39-6f73641.cust.bredbandsbolaget.se) |
| 11:43.01 | speakman | CMike: Still here? |
| 11:43.08 | speakman | Lost connection to my IRC client |
| 11:45.12 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 11:45.14 | bowman | kme |
| 11:45.17 | bowman | morning |
| 11:45.50 | bowman | how do I automatically park inbound calls? |
| 11:47.14 | shaZwaz | bowman: that can be done in the dialplan |
| 11:47.42 | bowman | shaZwaz: yeah, but if I use "Park" the callers is told his parking position, which I don't want ;) |
| 11:47.50 | bowman | s/callers/caller/g |
| 11:49.03 | shaZwaz | see ParkandAnnounce |
| 11:51.21 | shaZwaz | ~adn |
| 11:51.22 | jbot | extra, extra, read all about it, adn is the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS |
| 11:51.38 | bowman | ok, I can certainly just specify an empty string for "console", but that gives me warnings :-) |
| 11:58.36 | bowman | is there anything I can specify for "dial" in order to avoid warnings? |
| 12:04.31 | *** join/#asterisk godsmoke|laptop (~godsmoke@66-108-201-1.nyc.rr.com) |
| 12:11.34 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 12:12.07 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 12:12.16 | mAsH` | hi all |
| 12:12.28 | shaZwaz | howdy |
| 12:14.00 | bowman | damnit... |
| 12:14.51 | shaZwaz | bowman: create a dummy ext |
| 12:15.10 | bowman | shaZwaz: I have a different problem now :) |
| 12:15.26 | shaZwaz | Murphys knew that |
| 12:15.57 | bowman | shaZwaz: I specified parkbyebye|s|0 as the timeout fallback. context is all set up and extensions are reloaded, yet I get "starting SIP/phone3blabla at parkbyebye,s,0 failed, falling back.." etc. >:( |
| 12:16.17 | Luke-Jr | RjT: Note, though, that Vonage's TOS suck |
| 12:16.31 | bowman | ouh |
| 12:16.34 | bowman | fsck, my error :D |
| 12:16.51 | shaZwaz | another of murphy's law |
| 12:20.01 | *** part/#asterisk santiago (~santiago@63.245.86.104) |
| 12:22.29 | MacDeath | has anyone here used voipsupply.com? |
| 12:23.34 | *** join/#asterisk satlink (satlink@66.178.97.50) |
| 12:23.49 | *** part/#asterisk satlink (satlink@66.178.97.50) |
| 12:23.56 | *** join/#asterisk satlink (satlink@66.178.97.50) |
| 12:27.54 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 12:44.59 | *** join/#asterisk TheEmperor (TheEmperor@218.111.48.122) |
| 12:52.12 | Madkiss | Uhm, what do I have to configure asterisk to if I want it to redirect incoming calls from FreeWorldDial to my workstation? |
| 12:52.14 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 12:55.17 | shido6 | a register line /your extension |
| 12:55.30 | shido6 | have your extension ring your workstation |
| 12:55.57 | Madkiss | aharm, and can it register with more than one registrar at the same time? e.g. I have FWD and sipgate accounts |
| 12:56.40 | bowman | does it make sense to retrieve a parked call with ParkedCall(..) or should I rather Dial(${PARKEXTEN}), given that I've set that variable before I dial? |
| 12:57.10 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 12:59.55 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 13:06.27 | mak_ | how can I dial with asterisk and say something after when it has been answered ? I mean asterisk shold tell something after answer :) |
| 13:08.58 | Zeeek | . |
| 13:09.03 | vaewyn | If you are dialing with .call files you tell it the extension/context to drop into... make your first entry in that either a "Playback" or a "Background" if you want them to be able to dial an extension |
| 13:09.31 | vaewyn | * only does what you tell it to :} |
| 13:10.10 | Zeeek | and then, not always |
| 13:10.26 | mak_ | so asterisk dial, and then play some file ? :) |
| 13:10.30 | Zeeek | exten => 999,1,SelfDestruct(45,r) |
| 13:10.40 | vaewyn | hahaha |
| 13:10.48 | vaewyn | get rid of the ,r though ;P |
| 13:10.54 | Zeeek | such a great audience |
| 13:10.57 | vaewyn | don't want to alert the klingons! |
| 13:11.06 | Zeeek | NEVER use a t though! |
| 13:11.13 | vaewyn | bwahahaha |
| 13:11.17 | bowman | was that the FBI evidence killer extension, Zeeek? :) |
| 13:11.22 | *** join/#asterisk darkflib (darkflib@dialup019.ts006.bmt.esat.net) |
| 13:11.25 | Zeeek | no |
| 13:11.42 | Zeeek | that's exten =>998,Shred() |
| 13:11.53 | Zeeek | that's exten =>998,1,Shred() |
| 13:11.56 | Zeeek | sorry |
| 13:12.04 | bowman | exten => 1337,1,IgniteTNT() :) |
| 13:12.26 | vaewyn | nonnono... exten => 1337,1,KillFBIServers() |
| 13:12.30 | vaewyn | :} |
| 13:12.40 | datareactor | why the asterisk ring the first ring as feak ? |
| 13:12.50 | Luke-Jr | wtf? |
| 13:13.00 | vaewyn | datareactor: you using a FXO card? |
| 13:13.30 | vaewyn | (aka X100P) |
| 13:13.38 | datareactor | vaewyn no i have bri |
| 13:13.48 | bowman | out |
| 13:13.48 | *** part/#asterisk bowman (~bowman@snert3.tal.de) |
| 13:13.51 | vaewyn | umm... not sure then... FXO does it for the callerid |
| 13:14.00 | vaewyn | no idea why BRI would |
| 13:14.18 | datareactor | vaewyn ok no prob |
| 13:14.31 | Madkiss | shido6: What does the entry in extensions.conf for the FWD incoming calls look like? |
| 13:14.59 | Luke-Jr | Madkiss: as a guess: [fwd-incoming] exten => yourfwdnumber,... |
| 13:15.18 | RjT | do u need a provider if u have all the neccasary hardware and using aterisk ? |
| 13:15.23 | datareactor | anyone use go2call successfully with * it has been many days i am trying but asterisk never register |
| 13:15.25 | Madkiss | May I paste four lines? |
| 13:15.49 | Luke-Jr | Madkiss: pastebin.ca |
| 13:16.22 | Madkiss | a, hrm, wait, I slowly begin to realize ... |
| 13:16.37 | datareactor | RjT for international dialing ? |
| 13:17.02 | RjT | either |
| 13:17.13 | RjT | probably withing same country |
| 13:17.20 | RjT | within |
| 13:17.21 | Luke-Jr | RjT: for incoming DID calls? |
| 13:18.02 | datareactor | do you Telco charge same $ for all cities ? |
| 13:18.58 | Madkiss | What will "context=from-sip" lead to in sip.conf for a certain user-entry? |
| 13:19.20 | Luke-Jr | Madkiss: [from-sip] in extensions.conf |
| 13:19.51 | Madkiss | Okay |
| 13:20.07 | Madkiss | What if I want outgoing calls from [8888] in sip.conf to be led through FWD? |
| 13:20.38 | Luke-Jr | What context is 8888 configured for? |
| 13:20.48 | Madkiss | from-sip ... |
| 13:21.02 | Luke-Jr | and what name is the FWD connection? |
| 13:21.20 | Madkiss | Do you mean its context? The one from the register-line? |
| 13:21.47 | Luke-Jr | I mean its heading |
| 13:21.53 | Luke-Jr | like [8888], but for the FWD conn |
| 13:22.04 | Madkiss | That ought to be fwd-out |
| 13:22.09 | Luke-Jr | [from-sip] exten => _.,1,Dial(SIP/fwd-out) |
| 13:22.11 | Luke-Jr | try that |
| 13:22.19 | Luke-Jr | in extensions.conf |
| 13:22.53 | Luke-Jr | Disclaimer: I just started using * yesterday |
| 13:22.55 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:23.31 | Madkiss | Oh, then you are better of than me, I started four hours ago ... |
| 13:23.36 | Luke-Jr | heh |
| 13:23.49 | Madkiss | But from what I saw so far, asterisk rocks the rolling stones for sure |
| 13:23.59 | Luke-Jr | Just thought I should mention I'm not an nearly an expert yet |
| 13:26.05 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 13:26.32 | Zeeek | I Luke-Jr you put your finger on one irritating thing - the lack of a word for the peer/friend/user entry in sip and iax .conf files |
| 13:26.56 | Zeeek | so that it would be easier to say "context? Extension? frammus?" |
| 13:27.10 | Madkiss | yeah, or "Bill Clinton?" |
| 13:27.16 | Luke-Jr | Zeeek: hmm? |
| 13:27.17 | Zeeek | it's like an ID block |
| 13:27.34 | Zeeek | or an account - SIP account, IAX2 account |
| 13:27.38 | Luke-Jr | that works |
| 13:27.49 | Zeeek | depends on what you mean by "sex" |
| 13:27.56 | Luke-Jr | ... |
| 13:28.24 | Luke-Jr | where does 'sex' come into the picture? |
| 13:29.24 | bjohnson | ask Bill |
| 13:29.36 | Zeeek | depends on what you mean by 'come' |
| 13:29.41 | Zeeek | muahahaha |
| 13:30.35 | tih | Zeeek: be nice -- Clinton was the best president those poor bastards have had since... |
| 13:30.38 | tih | ...hmm... |
| 13:30.46 | Zeeek | Washington? |
| 13:30.50 | tih | ...have they had any good ones since the first few? |
| 13:30.50 | Luke-Jr | tih: I somewhat doubt that |
| 13:30.55 | Zeeek | Personally I loved Clinton |
| 13:30.55 | bjohnson | hahaha |
| 13:31.01 | bjohnson | Nixon |
| 13:31.04 | Luke-Jr | tih: Reagan, at least, was better than Clinton |
| 13:31.17 | bjohnson | maybe best since Kennedy |
| 13:31.20 | Zeeek | Reagan was like H323 |
| 13:31.42 | bjohnson | but when comparing morons .. debating who is the least moronic is somewhat futile |
| 13:31.49 | tih | bjohnson: yup. |
| 13:31.51 | Luke-Jr | i suppose |
| 13:31.53 | Zeeek | none of them are morons |
| 13:32.00 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 13:32.11 | tih | Zeeek: Reagan was pretty close, but only because he was senile. |
| 13:32.13 | *** join/#asterisk ToyMan (~konversat@204.8.82.238) |
| 13:32.48 | Zeeek | that was later - he was clever when young - not that I liked him btw |
| 13:33.08 | Zeeek | but BUsh is in no way stupid or even a Texan as most people know |
| 13:33.24 | Luke-Jr | but he did push that evil Patriot Act |
| 13:33.27 | Zeeek | he tries to be a rancher only to make the lower classes vote for him |
| 13:33.51 | tzafrir | while we're in philosophic mode, I'd like to write my thoughts about a web interface for asterisk |
| 13:33.56 | Zeeek | I voted against him and think he's bad and his whole admin abunch of thugs and crooks |
| 13:34.12 | Zeeek | NO tzafrir this is only for asterisk talk! Please! the nerve |
| 13:34.24 | tih | Zeeek: we're in for some interesting times, for sure. |
| 13:34.29 | bjohnson | I voted against him twice |
| 13:34.34 | bjohnson | in the same election |
| 13:34.40 | bjohnson | and I'm not American |
| 13:34.43 | bjohnson | hahahaha |
| 13:34.48 | Luke-Jr | bjohnson: LOL |
| 13:35.00 | bjohnson | well .. I AM American .. just not USA |
| 13:35.15 | bjohnson | North American |
| 13:35.18 | tih | Zeeek: over the next decade or two, we're going to see a new "iron curtain" going up, and the most interesting bit is whether it will be to the west or the east of England. |
| 13:35.25 | Luke-Jr | too bad Keyes wasn't the Rep candidate in '00 |
| 13:36.03 | Zeeek | tih I thought it was at the Turkish border :) ? |
| 13:36.32 | bjohnson | I find it how ironic quickly the people in the "Land of the Free" gave up their freedoms/rights after 9/11 |
| 13:36.37 | tzafrir | sorry, I had a phone |
| 13:36.41 | tzafrir | call |
| 13:36.41 | Zeeek | anyone seeing any problems with nufone lag today? |
| 13:37.05 | tzafrir | anyway, a web interface is not good enough for status reports |
| 13:37.29 | Zeeek | but great fopr selling pr0n videos |
| 13:37.50 | tzafrir | because you need something that can push information to the client and can do selective updates |
| 13:38.00 | Zeeek | so who is going to be the first to order that new IAX phone from CHina? |
| 13:38.21 | tih | Zeeek: what IAX phone? Where? |
| 13:38.34 | Zeeek | tzfrir it's like a chat - so it'd be better in Java or something, no? |
| 13:38.40 | vaewyn | Zeeek: I am at my standard 30-35ms to Nufone |
| 13:38.49 | Zeeek | lemmie see if I can remember the URL |
| 13:38.55 | tzafrir | Zeeek, can you assume a browser today supports java? |
| 13:39.02 | Delvar | yes |
| 13:39.03 | Zeeek | vaewyn have you been watching the console? |
| 13:39.16 | bjohnson | Zeeek: a few people already have them if they're what I think you are refering to |
| 13:39.20 | Zeeek | tzafrir if they need they will get it |
| 13:39.29 | vaewyn | Zeeek: nope... but I have been watching the log and my rrd graphs |
| 13:39.36 | tzafrir | why not use X then? |
| 13:39.46 | Zeeek | As in Active? NAH! |
| 13:39.57 | tzafrir | An X server is quite small. Smaller than a JVM |
| 13:39.58 | Zeeek | http://www.iaxtalk.com/product_info.php?products_id=29&osCsid=ad71e835182026991786a19799ffaa22 |
| 13:40.16 | Zeeek | IAX phone frm China^^^^^^^^^^^^^^^^ |
| 13:40.22 | tzafrir | http://tzafrir.freeshell.org/cyg-x/ |
| 13:40.27 | Madkiss | What happened to the "addmailbox"-thing? |
| 13:40.28 | *** join/#asterisk rene- (~rene-@201.137.86.219) |
| 13:40.40 | rene- | Hi there |
| 13:41.16 | Zeeek | tzafrir how does this connect to asterisk? What are you proposing to do? |
| 13:41.35 | tzafrir | a local X app running on the * server |
| 13:41.50 | Zeeek | X as in Xwin? |
| 13:42.10 | Zeeek | more people have Java than X IMO |
| 13:42.11 | tzafrir | local app on the * server. yes, X as in X11 |
| 13:42.21 | Zeeek | I don't do X on mine |
| 13:42.31 | rene- | we ve been asked to deliver redundant * service, i understand that a way to do this is using a mirror * server and that using DNS SVR records could help me in the case of primary server failure... does this way of doing things depends on any way on the ip telephones, as in do they need to implement DNS SVR in a way? |
| 13:42.59 | Zeeek | rene FWIW the Grandstream BT100 series does look for a SRV first |
| 13:43.03 | tzafrir | Zeeek, it is the X client, not the server. The server will be on the user's machine |
| 13:43.11 | Zeeek | Eeeeewwwww |
| 13:43.27 | Zeeek | but a good idea even if I don't like it :) |
| 13:43.43 | rene- | Zeeek: what if it didnt? wouldnt DNS server take care of a non DNS SVR aware client? |
| 13:44.15 | Zeeek | rene the phone will look first for SRV then simply ask for the domain ip |
| 13:44.22 | tzafrir | Zeeek, the point is that I don't like the laternatives. Name an alternative and I tell you what I dislike about it |
| 13:44.26 | bjohnson | Zeeek: I don't recognize that one but a couple of people on here do have iax phones |
| 13:44.33 | rene- | allright |
| 13:44.35 | Zeeek | Java talking to a simple process of any kind |
| 13:45.01 | vaewyn | The IAX phones most of the people have on here are wasim's farfon |
| 13:45.04 | tzafrir | a. requires opening a separate hole for that java client |
| 13:45.12 | Zeeek | the client asks for a checksum or something every 15 secs if no change, no dialog? |
| 13:45.31 | bjohnson | tzafrir: what about some graphic system that would work on both windows and linux? |
| 13:45.33 | tzafrir | b. this is an ugly java 1.1 GUI (this is what browsers support) |
| 13:45.35 | Zeeek | vaewyn I have one on the way but they have barely shipped any so far |
| 13:45.42 | bjohnson | like GTK |
| 13:45.57 | vaewyn | If you are in Europe they are fairly easy to get... in the US... good luck... |
| 13:45.57 | Zeeek | GTK might be ok. Even I have that |
| 13:46.16 | tzafrir | bjohnson, a. we need to install that app on each computer the user is at |
| 13:46.16 | bjohnson | just read yesterday that Trolltech made GTK GPL for windoes |
| 13:46.26 | Zeeek | Trolltech ? |
| 13:46.29 | vaewyn | that was Qt... not GTK |
| 13:46.30 | Madkiss | I might be ready to go, how do I start asterisk? |
| 13:46.32 | tzafrir | bjohnson, a. b. We need to develop cross-platform |
| 13:46.43 | bjohnson | Zeeek: check slashdot .. maybe I misread it |
| 13:46.46 | Zeeek | asterisk -cvvvvvvv |
| 13:46.49 | tzafrir | bjohnson, c. We still need to punch a hole for the app |
| 13:46.56 | vaewyn | bjohnson: Qt... not gtk |
| 13:47.03 | bjohnson | you'll have to do c anyway |
| 13:47.05 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
| 13:47.06 | bjohnson | oops |
| 13:47.07 | RoyK | how Qt... |
| 13:47.16 | Zeeek | Flash |
| 13:47.19 | tzafrir | qt, gtk, wxwidgets, fltk, whatever |
| 13:47.22 | Zeeek | on port 80 |
| 13:47.32 | vaewyn | hehehe... I use fltk on my pda... :} |
| 13:48.15 | Madkiss | uhm. |
| 13:48.20 | tzafrir | flash is non-free in my book. no decent free viewer. I generally avoid it |
| 13:48.25 | Madkiss | Feb 8 14:47:39 WARNING[31289]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x3c89081c (len 374) to 69.90.155.70 returned -1: Bad file descriptor |
| 13:48.26 | bjohnson | tzafrir: would be nice (but I acknowledge .. not necessary) if the server component was contained within * itself |
| 13:48.31 | vaewyn | flash sux... use the XML/javascript voodoo that google uses... way powerful and no "client" necessary beyond a web browser |
| 13:48.42 | bjohnson | rather than relying on another packahe and configuration |
| 13:48.49 | vaewyn | just look at google suggest and google maps |
| 13:49.03 | Madkiss | What the hell is that? |
| 13:49.05 | Zeeek | Flash viewers are all Free? |
| 13:49.14 | bjohnson | he wants to push from the server .. so xml/web browser are out |
| 13:49.46 | tzafrir | vaewyn, google has two separate interfaces: a. html interface for users/browsers, and b. xml-rpc for programs |
| 13:50.11 | tzafrir | It doesn't require punching a hole in the firewall, but it does require punching a hole |
| 13:50.35 | vaewyn | tzafrir: not the API stuff... the honest to goodness pages for google suggest and google maps... have you guys actually looked at how they do that? |
| 13:50.36 | bjohnson | Zeeek: I think I saw that phone before but it wasn't actually available .. can you confirm it is shipping? |
| 13:50.48 | tzafrir | Zeeek, then how come none of them was yet ported to s390 or mips? |
| 13:51.01 | *** join/#asterisk Memme (~Memme@i3ED65FD9.versanet.de) |
| 13:51.10 | Zeeek | tzafrir you meant open source, that's a different matter |
| 13:51.35 | Zeeek | bjohnson I have no idea - there is no address on the site and shipping is almost as much as the phone itself |
| 13:51.46 | Zeeek | like $40 to eu |
| 13:51.51 | Madkiss | err, nobody has an idea of what that "Bad file descriptor"-thing is? |
| 13:52.04 | vaewyn | Madkiss: sorry... not I |
| 13:54.15 | Madkiss | hm, let's see |
| 13:54.42 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 13:55.02 | Zeeek | anyone know p10-0.core02.dca01.atlas.cogentco.com |
| 13:55.26 | tzafrir | The thing is that with an X server I don't need extra software on the client machine except an X server, which is a standard component |
| 13:55.54 | christo | okay - does anybody know how to install mysql-devel? I have extracted the mysql tarball and repointed my asterisk-addons makefile to various lib directories, but the build still dies with `mysql_lock' undeclared |
| 13:56.03 | vaewyn | umm... hello... go with 0 extra and just make it work on the browser :} |
| 13:56.22 | tzafrir | And the app is in a very good place to do whatever it wants. I don't need to worry about session management, hijacking and the llot. |
| 13:56.28 | tzafrir | ssh does it for me |
| 13:57.04 | tzafrir | christo, distro? |
| 13:57.21 | christo | slackware |
| 13:57.59 | tzafrir | doesn't slackware have a mysql package? I don't think slackawre has those devil packages |
| 13:58.16 | christo | there's no mysql-devel package listed anywhere |
| 13:58.35 | christo | also, I have a mysql server already running on this machine, but for some reason I keep getting breakage when comiling *-addons |
| 13:58.40 | tzanger | morning everyone |
| 13:58.55 | vaewyn | morning tzanger |
| 13:59.07 | Zeeek | tzafrir someof us use all windows boxes except for the asterisk servers |
| 13:59.22 | oej | ~seen royk |
| 13:59.23 | jbot | royk is currently on #asterisk (4h 42m 46s). Has said a total of 29 messages. Is idling for 12m 16s |
| 13:59.36 | tzafrir | I realise that. This is why I bothered packaging the X server in a simple package |
| 13:59.42 | Madkiss | Feb 8 14:58:26 NOTICE[23625]: chan_sip.c:7654 handle_request: Registration from '<sip:madkiss@192.168.0.4>;tag=176595830' failed for '192.168.0.1' -- but the password is correct for sure |
| 14:00.06 | oej | Madkiss: There's always a failure in SIP auth |
| 14:00.13 | Zeeek | tzafrir so what is needed to get the X server to work? cyngus ? |
| 14:00.36 | tzafrir | again, see http://tzafrir.freeshell.org/cyg-x/ |
| 14:00.45 | Madkiss | oej: What? |
| 14:00.58 | bjohnson | quick offtopic question .. how do you check the md5 of a file/package? |
| 14:01.05 | Madkiss | md5sum |
| 14:01.15 | vaewyn | bjohnson: md5sum filename |
| 14:01.27 | Zeeek | tzafrir ok, so suppose I run this X server now? What is happening on the * box? |
| 14:01.29 | tzafrir | bjohnson, rpm -V ? |
| 14:01.58 | Zeeek | I run a client of yours? |
| 14:02.12 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 14:02.14 | tzafrir | you ssh to connect to the windows box and execute a certain single command |
| 14:02.23 | bowman | are there ways to react on DTMF tones during calls? |
| 14:02.28 | Zeeek | which runs the client |
| 14:02.31 | Madkiss | oej: I mean, I am just trying to REGISTER with my asterisk |
| 14:02.39 | tzafrir | yes (with X forwarding enabled) |
| 14:02.44 | Zeeek | bowman Playback is listening |
| 14:03.09 | Zeeek | oops during calls.... sorry |
| 14:03.15 | bowman | :) |
| 14:03.54 | Zeeek | it would be nice to be able to interrupt a call but I don't think you can |
| 14:03.59 | Zeeek | other than # |
| 14:04.06 | Zeeek | maybe * to hang up |
| 14:06.21 | vaewyn | Background listens for DTMF if you use it instead of Playback |
| 14:06.37 | bowman | I still mean "during CALLS" ;) |
| 14:06.55 | tzafrir | The problem is that I'm very convinced in my points, but there seems to be no other who took that route. This surely means something |
| 14:06.59 | bowman | while two people speak, one of them should be able to control the call or initiate a conference by pushing stuff like *#713812836 |
| 14:07.10 | vaewyn | hmm... nah... I think you can only do that with transfer turned on and make "trick" extensions for transfer |
| 14:07.15 | bjohnson | bowman: some dtmf are listened to .. look at T and t options in dial command |
| 14:07.20 | Zeeek | . |
| 14:08.20 | bowman | anyone got info on configuration of SIP-enabled addresses, e.g. foo@bar.com ? |
| 14:08.34 | bjohnson | tzafrir: I think there will be many who don't like idea of running X on a server .. that doesn't mean there is a better way and I can't think of one |
| 14:09.10 | Zeeek | Flash |
| 14:09.29 | vaewyn | FLASH SUCKS MONKEY BALLS! Get over it |
| 14:09.34 | Zeeek | why do you need the source code when Flash is available for all platforms |
| 14:09.35 | vaewyn | :} |
| 14:09.48 | Zeeek | I'm over it - but it's perfect for this one thing |
| 14:09.54 | Zeeek | I hate Flash |
| 14:09.57 | tzafrir | such as mips and s309 I mentioned |
| 14:10.02 | Zeeek | except for stuff like this |
| 14:10.07 | Madkiss | The funny thing is: It doesn't even ask for the password |
| 14:10.12 | Madkiss | It just tells me -- go away |
| 14:10.24 | vaewyn | No it isn't... cause good luck supporting your app in the future... flash changed so often ist is laughable |
| 14:10.50 | Zeeek | http://www.adcritic.com/interactive/view.php?id=5927 |
| 14:11.05 | Zeeek | Look at this vaewyn - you'll laugh then cry |
| 14:11.06 | tzanger | Vexi |
| 14:11.08 | tzanger | www.vexi.org |
| 14:11.24 | tzanger | open source engine, has java, win32, linux and macosx engine binaries |
| 14:11.28 | tzanger | we use it for a lot of things here |
| 14:11.52 | tzanger | veyr nice... xmlrpc or soap interfaces, svg and multimedia are in the works... very cool |
| 14:11.55 | Zeeek | tzafrir tell us what your going to put in the monitor data? |
| 14:12.09 | Zeeek | like watching calls? |
| 14:12.12 | Zeeek | lines? |
| 14:12.42 | tzafrir | I need a general console for managing and monitoring * |
| 14:12.57 | bjohnson | that's 2 separate functions |
| 14:13.10 | tzafrir | Currently the user interface for Rapid is terminal-based |
| 14:13.11 | Zeeek | and you want to share this with TheCommunity (tm) |
| 14:13.20 | tzafrir | (we use a menu program called pdmenu) |
| 14:13.20 | Madkiss | oej: Can you please tell me what the hell you meant by "There's always a failure in SIP auth"? |
| 14:13.28 | Zeeek | [tinc] |
| 14:13.30 | bjohnson | monitoring needs the real time data .. but managing does not |
| 14:13.46 | tzafrir | no, I want to as the collective opinion of TheCommunity |
| 14:13.51 | Zeeek | Madkiss in most auth, the first challenge is like "huh? You gotta authenticate?" |
| 14:13.55 | *** join/#asterisk santiago (~santiago@63.245.86.104) |
| 14:14.02 | tzafrir | (why won't tab complete "TheCommunity"?) |
| 14:14.04 | Zeeek | so there is always |
| 14:14.23 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
| 14:14.26 | Zeeek | tzafrir you mean we can vote for features :) |
| 14:14.50 | vaewyn | Zeeek: read this... be barely amazed... think about it for a while... and when the light comes on you will be VERY amazed http://www-106.ibm.com/developerworks/web/library/wa-httpget/ |
| 14:15.16 | Madkiss | Zeeek: aharm. Well, the problem persists, even after 35 conection attemps. "Feb 8 15:12:48 NOTICE[20841]: chan_sip.c:7654 handle_request: Registration from '"Martin Loschwitz" <sip:madkiss@192.168.0.4>' failed for '192.168.0.1'" again and again, all the time. |
| 14:16.24 | Zeeek | vaewyn look interesting |
| 14:16.46 | Zeeek | Microsoft has always bragged about such things in their asp stuff btw |
| 14:16.58 | vaewyn | Zeeek: that is the basic trick underlying google suggest and google maps... :} |
| 14:17.07 | Zeeek | we all know how bad it sucks to keep refreshing whole pages for no change |
| 14:17.31 | vaewyn | yep... and this works in 99.9% of browsers out of the box |
| 14:17.32 | Zeeek | never used google suggest yet |
| 14:18.01 | Zeeek | but hey, tzafrir is the one looking! |
| 14:18.19 | Zeeek | maybe I should try this with my own info page |
| 14:18.19 | pointer-gaim | anyone using a diameter implementation for billing/accounting? |
| 14:18.23 | tzafrir | vaewyn, looks like a browser-specific feature |
| 14:18.42 | vaewyn | tzafrir: works in IE, Gecko based and Opera |
| 14:18.48 | vaewyn | no modifications |
| 14:18.50 | Zeeek | they mention activeX which is not 99% |
| 14:19.26 | vaewyn | It uses ActiveX when available... and XMLHttpRequest when it is not (which works in all gecko browsers and several others) |
| 14:19.39 | Zeeek | very interesting indeed |
| 14:19.48 | Zeeek | where is there a test demo page? |
| 14:20.06 | vaewyn | just save "listing 1" locally and use it |
| 14:20.13 | vaewyn | calls itself by default |
| 14:20.25 | vaewyn | which is a stupid trick but... it shows the potential |
| 14:20.50 | vaewyn | I am starting to use it heavily for realtime form validation and status watching and... etc.. etc... |
| 14:21.06 | Zeeek | what is background.html? |
| 14:21.28 | vaewyn | that is the file with theexample |
| 14:21.41 | vaewyn | it loads itself into it's own textarea :} |
| 14:22.22 | *** join/#asterisk mflorell (~mattf@rrcs-24-173-158-34.se.biz.rr.com) |
| 14:22.37 | vaewyn | The only problem with this is that people must have javascript active in order to use it |
| 14:22.44 | Zeeek | that's acceptable |
| 14:22.51 | *** join/#asterisk Druken (Druken@67.69.139.226) |
| 14:22.54 | vaewyn | yep |
| 14:22.55 | Druken | morning everyone, |
| 14:22.59 | vaewyn | morning |
| 14:23.01 | Zeeek | heh it loads the 404 file in |
| 14:23.03 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 14:23.06 | Druken | does anyone know how to fix a broken odbc ?? :) |
| 14:23.30 | Druken | Feb 8 09:17:25 WARNING[147465]: res_config_odbc.c:68 realtime_odbc: SQL Alloc Handle failed with -1 on pgsql1! |
| 14:23.40 | Druken | keep gettin that on all odbc requests.... |
| 14:23.43 | vaewyn | Zeeek: 404? |
| 14:23.56 | Druken | my phones won't register to the server, because i'm using realtime with odbc |
| 14:24.03 | Zeeek | it wants to load background.html and not fining one loads the 404 file |
| 14:24.22 | rene- | any clues on the whereabouts of chan_spy? |
| 14:24.26 | vaewyn | Zeeek: did you save it as background.html? |
| 14:24.37 | vaewyn | shouldn't be a 404 since it is just trying to load itself |
| 14:24.42 | Zeeek | nope |
| 14:24.52 | tzafrir | vaewyn, it still requires constant polling |
| 14:24.58 | *** join/#asterisk toddf (f7e0ws5q7u@default.fries.net) |
| 14:25.15 | bjohnson | this e164.org concept is great. Is there a way to find out who's using it? |
| 14:25.42 | vaewyn | tzafrir: yes... but you can't do true push in a broadly distributed application because NAT will trash it every time... consider the "polling" as "registering" :} |
| 14:25.53 | Zeeek | bjohnson - try their numbers? |
| 14:26.06 | bjohnson | are their numbers listed somewhere? |
| 14:26.13 | Zeeek | I hope NOT! |
| 14:26.28 | tzafrir | vaewyn, with a real program I can. I just have to keep a connection open |
| 14:26.33 | bjohnson | hope not? |
| 14:26.36 | bjohnson | why? |
| 14:26.38 | Zeeek | the idea is to try thenumber there first |
| 14:26.48 | Zeeek | I don't want my numbers listed |
| 14:26.57 | bjohnson | right .. so is there a list of name and numbers that is browsable? |
| 14:27.02 | vaewyn | tzafrir: ummm... which is the same as polling and leaving an open connection every 5 minutes waiting for data |
| 14:27.11 | Zeeek | I don't think so - write duane and ask him |
| 14:27.43 | vaewyn | tzafrir: make a request... and wait for data... the server only sends data when it has it... or a "retry" request after 5 minnutes |
| 14:27.44 | tzafrir | vaewyn, but you don't have too many clients together. |
| 14:28.04 | vaewyn | ?? |
| 14:28.11 | vaewyn | I don't get your meaning |
| 14:28.30 | tzafrir | vaewyn, how often should the display be freshed about the status of extensions? |
| 14:28.39 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 14:28.46 | vaewyn | tzafrir: when they change... |
| 14:28.59 | vaewyn | which is EASILY doable with pull faking push |
| 14:29.09 | *** part/#asterisk alakdan (~alakdan@210.213.185.59) |
| 14:29.38 | vaewyn | ok... now I have to make a proof of concept :} |
| 14:29.44 | tzafrir | wait 5 minutes for an answer? this is generally a timeout |
| 14:30.19 | vaewyn | yeah.. wait 5 minutes... if you have gotten no data then timeout and make another request.... repeat as necessary |
| 14:30.40 | vaewyn | server pushes data to you when it has it |
| 14:30.50 | Zeeek | so if I had a page that did a sip show peers and asked for it this way what is the diff? |
| 14:32.18 | vaewyn | Zeeek: You can have a CGI (or prefereably mod_perl) on the server watch and send data when changes occure... instead of hving to flood the server with client requests every couple seconds to check state |
| 14:32.41 | vaewyn | plus... no refresh or such involved.. only the data changes |
| 14:32.50 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 14:33.49 | vaewyn | dude! I am going to make a thing that shows all my jabber conacts status live... that will be a good proof |
| 14:33.53 | vaewyn | of concept |
| 14:34.53 | vaewyn | (if I can type well enough to code) |
| 14:34.54 | vaewyn | :} |
| 14:35.01 | netsurfer | :oP |
| 14:35.31 | *** join/#asterisk ClayReiche123 (fwuser@acxexch1.accxx.com) |
| 14:35.57 | Zeeek | (waiting for the proof of concept) |
| 14:36.25 | vaewyn | it's gonna take a little bit of time... but I am going to try and implement a web version of "watch" :} |
| 14:36.58 | *** join/#asterisk jgaviria (~jgaviria@63.245.86.116) |
| 14:37.25 | ClayReiche123 | Is there any realtime code in the latest stable branch? |
| 14:37.42 | tzafrir | yes, through odbc |
| 14:37.49 | Zeeek | I still don't see how anything can happen without clicking on load? |
| 14:37.49 | *** join/#asterisk lohelle (~post@213.184.212.218) |
| 14:38.08 | ClayReiche123 | odbc and MySql? |
| 14:38.17 | netsurfer | Zeeek - * reloads the db every so often |
| 14:38.21 | vaewyn | Zeeek: with that code it can't... but you can start a chain of events with "onLoad" for the page |
| 14:38.41 | tzafrir | if you want mysql you need to configure unixodbc to access to it |
| 14:38.41 | Zeeek | so you mean calling load periodically |
| 14:39.16 | netsurfer | baah my 2nd asterisk box wont accept connections from xlite client |
| 14:39.33 | vaewyn | Zeeek: yep... but the secret is the server holding the connection and not sending data till a real change takes place... then you don't spank your web server trying to keep track of changes |
| 14:39.34 | jgaviria | i need to put my asterisk behind a firewall, i just need to do sip connections, somebody knows what ports i need to permit in the firewall? |
| 14:39.44 | Zeeek | 5060 |
| 14:39.51 | tzafrir | and rtp |
| 14:39.57 | Zeeek | and 10000-10100 |
| 14:40.02 | tzafrir | which is configurable in rtp.conf |
| 14:40.02 | Zeeek | as an example |
| 14:40.16 | jgaviria | tzafir, Zeek, thanks, all tcp? |
| 14:40.17 | ClayReiche123 | I tried PostGreSQL about a month ago with no success(on HEAD), has anything changed that might give me a different result? |
| 14:40.22 | tzafrir | udp |
| 14:40.33 | *** join/#asterisk jsolares (~jsolares@mail.epa.com.gt) |
| 14:40.45 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 14:41.46 | jgaviria | tzafrir: all udp ? |
| 14:43.23 | Zeeek | I see some wackiness in this whole thing. A lot of the data on asterisk itself is already polled |
| 14:43.40 | Zeeek | sometimes people say you need 5060 in both |
| 14:45.46 | tzafrir | jgaviria, yes, all udp |
| 14:50.46 | bprice20 | what are you guys talking about... I missed the beginning of this conversation |
| 14:50.53 | bprice20 | iax2, iax.conf? |
| 14:52.38 | *** join/#asterisk firstsword (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 14:52.52 | firstsword | does anyone has a at-323? |
| 15:01.10 | *** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net) |
| 15:01.50 | jgaviria | somebody using r2 signalling with asterisk? |
| 15:02.03 | *** join/#asterisk jcims (~rrich@cpe-69-135-121-57.columbus.rr.com) |
| 15:02.46 | jcims | is there a 'right' way to ask asterisk to dump it's sip registry externally? 'echo sip show registry | asterisk -rc' seems crude |
| 15:02.47 | *** join/#asterisk mishehu (mishehu@cshells.shavedgoats.net) |
| 15:03.17 | *** join/#asterisk CBAsteriskUK (~cblunt@217.206.164.219) |
| 15:04.14 | tzafrir | asterisk -rx "sip show registry" |
| 15:04.37 | CBAsteriskUK | Hi, any one know about polarity reversal hangup? |
| 15:04.56 | jcims | tzafrir: thanks! |
| 15:06.14 | tzafrir | btw: we have a script that runs: asterisk -rx "$cmd" </dev/null |
| 15:06.37 | tzafrir | because otherwise it makes pdmenu's "display" unhappy |
| 15:07.22 | bjohnson | I keep discovering new ways to make free calls using voip |
| 15:07.37 | *** join/#asterisk forkqueue (~sam@spc1-ward2-5-0-cust207.bagu.broadband.ntl.com) |
| 15:11.34 | netsurfer | quick warning guys, anyone buying x100p cards from ebay be careful of seller "nkans" |
| 15:12.29 | Qwell | netsurfer: his card works fine for me... |
| 15:12.54 | netsurfer | Qwell - I have had NO contact from him, no emails no card no nothing |
| 15:13.26 | netsurfer | should have been here 10 days ago |
| 15:13.34 | Qwell | odd, no usps tracking number? I got mine within like 2 days |
| 15:14.16 | netsurfer | no, not even had an email from him |
| 15:14.38 | netsurfer | and no answer to the so called business number |
| 15:14.40 | Qwell | hmm |
| 15:14.45 | bjohnson | has anyone used a free service to get around double nat issue? |
| 15:15.11 | forkqueue | bjohnson: I've used free software to get around it :) |
| 15:15.16 | forkqueue | bjohnson: sipproxd |
| 15:15.32 | bjohnson | can that run behind a nat router? |
| 15:15.36 | Qwell | netsurfer: sorry to hear it. I hope it works out for you. I assume you've left negative feedback? |
| 15:15.48 | Liquide | what's needed to use fax or sms on voip? |
| 15:15.48 | netsurfer | Qwell not yet |
| 15:15.49 | bjohnson | things I've looked at all need to run on a public ip |
| 15:15.51 | forkqueue | bjohnson: I actually run it *on* my router (linksys wrt54g) |
| 15:16.02 | Qwell | I would. I bet he jumps right on it. |
| 15:16.21 | bjohnson | hmm .. I use ipcop .. I wonder if I could run something on it |
| 15:16.25 | netsurfer | yyeah, by leaving ME negative feedback |
| 15:16.36 | bjohnson | Liquide: check the wiki |
| 15:16.36 | netsurfer | ebay sux for that |
| 15:18.29 | Qwell | off to work, good luck netsurfer |
| 15:18.40 | netsurfer | Qwell - thx |
| 15:18.55 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:22.37 | JunK-Y | why im doin' it: ast_indicate(chan, AST_CONTROL_RINGING); and im getting a fast busy, wtf? |
| 15:24.06 | netsurfer | indications.conf set up properly ? |
| 15:24.37 | JunK-Y | brb, coffee break :) |
| 15:24.40 | *** join/#asterisk Memme (~Memme@i3ED61920.versanet.de) |
| 15:25.00 | *** join/#asterisk Astinus_ (~abba@213.167.111.138) |
| 15:27.15 | eKo1 | Hmm...why the heck is the call cutting out.... |
| 15:29.22 | Madkiss | How do I check if asterisk registered with FWD? |
| 15:30.26 | netsurfer | Madkiss - on sip? sip show registry |
| 15:30.55 | netsurfer | it seems to be down, at least I cant get registered on it OR sipgate.co.uk |
| 15:31.20 | netsurfer | oh.. its registered now :) |
| 15:31.28 | Madkiss | Host Username Refresh State |
| 15:31.30 | Madkiss | fwd.pulver.com:5060 610077 105 Registered |
| 15:31.35 | Madkiss | ha, great. |
| 15:31.37 | netsurfer | thats it! :) |
| 15:32.37 | JunK-Y | netsurfer: i guess its oky, a simple Busy(); is fine into my dialplan |
| 15:32.40 | vaewyn | ~Zeeek |
| 15:32.53 | vaewyn | ~seen Zeeek |
| 15:32.58 | jbot | zeeek <~icechat5@Zeeek.sustaining.supporter.pdpc> was last seen on IRC in channel #asterisk, 49m 18s ago, saying: 'sometimes people say you need 5060 in both'. |
| 15:33.11 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:33.11 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:33.49 | vaewyn | tzafrir: proof of concept... not a really good one... but it is one: http://www.andrews.edu/wwwrogue/Watch/index.html |
| 15:33.59 | speakman | Anyone got X100P working properly in Europe?+ |
| 15:34.56 | Madkiss | In that case, the only thing I still need to find out is why asterisk refuses to accept my kphone's connection |
| 15:36.12 | tzafrir | vaewyn, but this refreshes everything every second. Not acceptable for a monitor. even top(1) and watch(1) don't do that |
| 15:36.13 | netsurfer | Madkiss - sip debug ip <kphoneIP> |
| 15:36.21 | *** join/#asterisk fenlander (~irc@82.152.81.57) |
| 15:36.24 | tzafrir | they do differential updates |
| 15:36.27 | fenlander | hi |
| 15:36.42 | tzafrir | anyway, it's a start, but I have to go now |
| 15:36.51 | eKo1 | Hmm...the call seems to cut right during the bridging. |
| 15:36.59 | vaewyn | tzafrir: it only refreshes when the server senses a difference... at the moment that is every second... js here is every 10 seconds |
| 15:37.47 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfkh1.dialup.mindspring.com) |
| 15:37.48 | vaewyn | tzafrir: notice you didn't have to reload and I controlled the timing :P |
| 15:38.13 | Madkiss | netsurfer: ha, well, now how would that help me? ;) |
| 15:38.37 | Madkiss | Transmitting (no NAT): |
| 15:38.37 | Madkiss | SIP/2.0 403 Forbidden |
| 15:38.59 | pcm | put secret= in your account |
| 15:39.07 | Madkiss | I have |
| 15:39.24 | pcm | host=dynamic ? |
| 15:39.33 | Madkiss | 12345 -- but my kphone does not even get a request for the passwd |
| 15:40.01 | pcm | put 'secret=' (no pass) |
| 15:40.32 | Madkiss | host=dynamic, yes |
| 15:41.24 | *** join/#asterisk ball (~ball@216.176.80.183) |
| 15:41.27 | tzafrir | kphone? |
| 15:41.38 | Madkiss | yes |
| 15:41.42 | tzafrir | make sure you restart it to apply configuration changes |
| 15:41.50 | ball | Is there a general telecomms channel on Freenode, that is not *-specific? |
| 15:41.50 | Madkiss | I did |
| 15:42.26 | Madkiss | harm, wait, looks like I made it work |
| 15:43.03 | Madkiss | indeed. |
| 15:43.08 | Madkiss | Well, I have no idea how, but ... |
| 15:43.25 | netsurfer | lmao.. my cheap chinese sip phone did that too |
| 15:43.33 | netsurfer | got it working, dont know how |
| 15:43.55 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 15:44.58 | ball | hello _Brian |
| 15:45.09 | _Brian | hi Ball... |
| 15:45.30 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 15:45.32 | _Brian | hwo goes it? |
| 15:46.29 | *** join/#asterisk lespiggot (~les@217.206.141.131) |
| 15:46.30 | ball | _Brian: it goes :-) |
| 15:46.30 | ball | hello calvinhp, lespiggot |
| 15:46.34 | _Brian | i hear ya |
| 15:47.20 | Madkiss | Some FWD-user here? |
| 15:47.27 | ball | Oh bugger, I can't seem to download NCSA Mosaic any more |
| 15:48.20 | eKo1 | g729tolin_framein: Received a G.729 frame that was 12 bytes from RTP <--- What size are these frames supposed to be? |
| 15:48.29 | netsurfer | Madkiss - yup |
| 15:51.01 | Madkiss | Can asterisk handle more than one registrar at the same time? |
| 15:51.43 | netsurfer | yup |
| 15:54.19 | pcm | ekol: 10 or 20 bytes |
| 15:55.50 | *** part/#asterisk lespiggot (~les@217.206.141.131) |
| 16:02.08 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
| 16:02.13 | *** join/#asterisk twisted[work] (~twisted@twisted.active.supporter.pdpc) |
| 16:02.13 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 16:06.29 | eKo1 | pcm: Why then would it send 12 bytes?! |
| 16:09.19 | *** part/#asterisk ball (~ball@216.176.80.183) |
| 16:12.59 | Luke-Jr | stupid company... they don't use normal SIP pass |
| 16:15.08 | pcm | luke: you found out ? :) |
| 16:15.16 | pcm | eko1: maybe voice supporession |
| 16:17.09 | *** join/#asterisk samueltc (~samuel@levinux.UQAR.UQUEBEC.CA) |
| 16:17.12 | pcm | luke: so you found out ? |
| 16:17.24 | pcm | luke: finally that they don't use MD5 ? |
| 16:17.42 | samueltc | hi |
| 16:17.47 | pcm | hi |
| 16:18.08 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 16:18.13 | *** join/#asterisk sabre (~urfos@69.149.209.83) |
| 16:19.03 | CCDAS | hello everyone. Has anyone had any experience with the H.323 driver that is included with Asterisk? |
| 16:19.11 | shido6 | yes |
| 16:19.19 | samueltc | hey, when i'm calling someone and I talk over his voice cut, there is any tweaking I can do to fix that? |
| 16:19.33 | CCDAS | shido6: are you responding to my question? |
| 16:19.34 | shido6 | jitter or stop your pron downloads |
| 16:19.37 | shido6 | yes CCDAS |
| 16:19.55 | samueltc | shi |
| 16:19.57 | pcm | anyone using charge.com ? |
| 16:20.02 | shido6 | yes, pcm |
| 16:20.12 | pcm | not as a customer :) |
| 16:20.21 | shido6 | ? |
| 16:20.23 | pcm | I mean not for paying but for receiving :) |
| 16:20.29 | shido6 | yes |
| 16:20.37 | CCDAS | Im getting PWLib Assertion fail: Jutter buffer thread did not terminate, file jitter.cxx, line 279 error=9. Hare you seen this before? |
| 16:20.40 | samueltc | shido6: you talking to me? there is no pr0n download.. network link is unused |
| 16:20.59 | shido6 | samueltc what are you running asterisk on |
| 16:21.07 | shido6 | hope its not a celeron |
| 16:21.13 | CCDAS | Rh9 |
| 16:21.16 | shido6 | dont have Xwdinwos running do you? |
| 16:21.21 | CCDAS | no |
| 16:21.37 | Luke-Jr | pcm: yea; their mySoftphone app doesn't do normal auth |
| 16:21.52 | cervajs | where i can see changes in cvs stable? (after fixing iax2 floating point error, i cannot see callerid on sip phones) |
| 16:21.53 | samueltc | shido6: the box is dedicated to asterisk (at this time only 1 link (sip-->iax(voipjet))) |
| 16:21.56 | CCDAS | Its a P4 2.8 512M |
| 16:21.59 | Luke-Jr | pcm: I need to fax them images of my credit card to change to normal SIP |
| 16:22.02 | shido6 | ahh voipjet |
| 16:22.07 | CCDAS | ? |
| 16:22.13 | pcm | that's sad ... maybe you should 'extract the hashing' code :) |
| 16:22.18 | shido6 | good luck with them, not sure why they are having so many problems lately |
| 16:22.28 | samueltc | shido6: I have the same problem with other |
| 16:22.38 | shido6 | what "other" ? |
| 16:22.50 | samueltc | nufone |
| 16:22.53 | Luke-Jr | pcm: No idea how :( |
| 16:23.08 | shido6 | what codec did u use with NuFone? |
| 16:23.16 | samueltc | shido6: ulaw |
| 16:23.22 | shido6 | why? |
| 16:23.26 | samueltc | ulaw with voipjet |
| 16:23.31 | shido6 | are you sending faxes? |
| 16:23.38 | CCDAS | I am also running Ulaw. No faxes. |
| 16:23.39 | samueltc | shido6: are you talking to me? |
| 16:23.45 | shido6 | then u dont need to use ulaw |
| 16:23.53 | shido6 | use gsm or ilbc or speex or g729 |
| 16:24.03 | samueltc | shido6: ok |
| 16:24.04 | pcm | luke: can you force your softphone to register every second ? |
| 16:24.13 | samueltc | shido6: thanks i'll try that |
| 16:24.19 | shido6 | CCDAS what error are you getting again? |
| 16:24.21 | *** join/#asterisk datareactor (datareacto@203.81.195.76) |
| 16:24.44 | CCDAS | Assertion fail: Jitter buffer thread did not terminate, file jutter.cxx. |
| 16:24.48 | _Brian | Has anyone been able to get Directed pickup working as described in http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup |
| 16:24.48 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.6) |
| 16:24.50 | CCDAS | Line 279, Error=9 |
| 16:25.24 | pcm | Brian: sure |
| 16:25.55 | CCDAS | The calls will go from SIP to H323 clients and back fine. But when the H323 client uses hold, transfer or something like that the audio is lost and then this error appears. |
| 16:25.56 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 16:26.01 | _Brian | pcm: can you provide any pointers? as you can see the web page does not provide much insight :( |
| 16:26.19 | pcm | Brian: but after I read about 'direct' then no ... I didn't try 'YET' :) |
| 16:26.35 | shido6 | CCDAS do you have any digium gear in your box? |
| 16:26.42 | CCDAS | not in this one. |
| 16:26.43 | Madkiss | <PROTECTED> |
| 16:26.48 | Madkiss | What does that mean exactly? |
| 16:26.54 | shido6 | then turn off jitter |
| 16:27.04 | _Brian | pcm: hehe. no prob..i will keep plugging away at it. |
| 16:27.12 | Luke-Jr | pcm: No; I can't even runi t |
| 16:27.22 | Luke-Jr | pcm: It requires Windoze |
| 16:27.30 | CCDAS | Sorry I am not sure how, can you elaborate? |
| 16:27.53 | _Brian | Madkiss: that means that your sipcall to freenet-out (id 684f) returned a circuit-busy |
| 16:28.36 | Madkiss | and what is that? |
| 16:28.37 | eKo1 | Hmm...I have a U.S. number which when called is redirected by my provider to my * server. I'm looking at the debug and it says: SIP/2.0 407 Proxy Authentication Required |
| 16:28.50 | *** join/#asterisk hmmhesays (~hmmhesays@66.173.103.108) |
| 16:29.26 | eKo1 | But I set insecure=yes on my provider's sip entry in sip.conf. Why is it still doing authentication? |
| 16:29.28 | _Brian | Madkiss: what? |
| 16:29.29 | hmmhesays | is there a short simple way to collect digits and play them back to the caller? because the way i'm doing it seems messy |
| 16:29.39 | Madkiss | _Brian: circuit-busy |
| 16:30.13 | CCDAS | shido6: would you mind hand holding. I am unsure how to turn this off. |
| 16:30.20 | netsurfer | eKo1 - unless u set insecure=very it will want the provider to authenticate before putting a call thru |
| 16:30.31 | eKo1 | Oops, I meant insecure=very |
| 16:30.33 | RoyK | s/thru/through/ :P |
| 16:30.37 | netsurfer | eKo1 http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20insecure |
| 16:30.53 | *** join/#asterisk eipi (~eipi@OL128-44.fibertel.com.ar) |
| 16:31.00 | _Brian | hmmhesays: yes, just do a read and then do a Saydigits |
| 16:31.03 | eKo1 | netsurfer: I did set that already. |
| 16:31.22 | hmmhesays | thanks |
| 16:32.19 | _Brian | hmmhesays: sure....np |
| 16:32.36 | _Brian | Madkiss: you want to know what a circuit-busy means? |
| 16:33.01 | Madkiss | yes |
| 16:33.22 | *** join/#asterisk hellmasterbr (~root@201.12.132.171) |
| 16:33.33 | hellmasterbr | hohoh |
| 16:33.43 | hellmasterbr | lots of ppl, |
| 16:33.51 | CCDAS | shido6: Are you still here? |
| 16:33.52 | *** join/#asterisk zno (~zeno@ip-160-79-174-99.autorev.intellispace.net) |
| 16:34.14 | zno | any plans for application Math to be merged into stable? |
| 16:34.37 | hellmasterbr | i need a little help about voicemail user/pass |
| 16:34.47 | hellmasterbr | anyone could help this old monkey ? |
| 16:36.11 | hellmasterbr | i cant access the voicemail system, i just try to send user/pass, but it keeps saying it is not valid |
| 16:36.17 | hellmasterbr | it sounds so simple |
| 16:36.20 | hellmasterbr | but i cant solve it |
| 16:36.34 | hellmasterbr | just tryied with default 1234 example account |
| 16:36.41 | Madkiss | _Brian: Well, infact, what I generally want is: I have three providers. I want calls to FWD go out via FWD, I want calls to SIPgate to go out via SIPgate and I want the rest to go out via Freenet |
| 16:37.06 | shido6 | DATMF problems hellmasterbr |
| 16:37.08 | shido6 | err |
| 16:37.09 | shido6 | DTMF |
| 16:37.14 | shido6 | sounds like DTMF problems, hellmasterbr |
| 16:37.17 | hellmasterbr | hummm |
| 16:37.19 | shido6 | what codec are you using? |
| 16:37.29 | bjohnson | who wants to make free calls via voip? |
| 16:37.29 | hellmasterbr | gsm |
| 16:37.31 | shido6 | and what protocol |
| 16:37.34 | hellmasterbr | sip |
| 16:37.42 | hellmasterbr | i can left the mail |
| 16:37.44 | shido6 | then set it as rfc2833 or info |
| 16:37.44 | hellmasterbr | i can dial |
| 16:37.49 | shido6 | and also set that in your phone |
| 16:37.50 | VoIPMasta | bjohnson: free calls... to where? |
| 16:37.55 | shido6 | then do a sip reload |
| 16:37.55 | hellmasterbr | hummm |
| 16:37.57 | shido6 | at the cli |
| 16:38.02 | shido6 | and try to get your vmail again |
| 16:38.05 | hellmasterbr | ok, will try it now |
| 16:38.13 | shido6 | asterisk needs to hear your dtmf which it isnt getting right now because you dont have |
| 16:38.13 | hellmasterbr | thanx by now |
| 16:38.20 | hellmasterbr | hummm |
| 16:38.24 | shido6 | dtmfmode=rfc2833 set |
| 16:38.28 | shido6 | in [general] stanza of your sip.conf |
| 16:38.37 | bjohnson | VoIPMasta: start a wiki page for Free Voip Calling and I'll feed you some info .. different routes to different numbers |
| 16:38.37 | shido6 | get it? |
| 16:38.41 | hellmasterbr | was what i was thinking, but i could not figure how to "it cant be sending..." |
| 16:38.49 | hellmasterbr | sure |
| 16:38.55 | bjohnson | VoIPMasta: but I only want to go through it once and have it available for everyone |
| 16:39.02 | hellmasterbr | just a minute, will change and retest |
| 16:39.10 | shido6 | im not going anywhere |
| 16:39.17 | xkev | ~seen kpfleming |
| 16:39.22 | jbot | kpfleming <~chatzilla@ip68-3-230-141.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 3d 12h 52m 59s ago, saying: 'nahh, got plenty to do this weekend to'. |
| 16:39.23 | VoIPMasta | bjohnson: .ok, I can start that wiki |
| 16:39.24 | tzanger | shido6: did you get the support@ ticket? |
| 16:39.34 | tzanger | shido6: #6115 |
| 16:39.36 | shido6 | err |
| 16:39.40 | shido6 | RT? |
| 16:39.47 | _Brian | maddkiss: i dont use FWD.....:( but I did see someone doing LCR at http://www.voip-info.org/tiki-index.php?page=Find%20next%20available%20least%20cost%20outbound%20channel |
| 16:39.50 | tzanger | shido6: RT? |
| 16:39.54 | shido6 | stdby |
| 16:39.54 | *** join/#asterisk purplebob (~don@206-230-187-185.sugardog.com) |
| 16:39.58 | bjohnson | VoIPMasta: it's nothing that the collective knowledge here doesn't already know about .. but it needs to be documented somewhere |
| 16:40.28 | hellmasterbr | at sip.info, im using dtmfmode=info |
| 16:40.30 | bjohnson | VoIPMasta: make it obvious so a wiki search on "free" will pick it up |
| 16:40.31 | VoIPMasta | bjohnson: I do agree |
| 16:40.57 | shido6 | can your phone be set to use rfc2833, info, or inband? |
| 16:41.04 | shido6 | just make that agree to whatever you put in sip.conf |
| 16:41.07 | bjohnson | I'll dictate here so others can input with info |
| 16:41.09 | hellmasterbr | well |
| 16:41.11 | shido6 | and issue a sip reload at the asterisk CLI |
| 16:41.12 | hellmasterbr | i use xlite |
| 16:41.16 | purplebob | need a little hint on modifying my dial plan. I want an extension to send a call into Queue X, if its not answered in 3 minutes, send the call to an outside #, if thats busy, go to voicemail. |
| 16:41.24 | bjohnson | ~docs |
| 16:41.25 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 16:41.26 | hellmasterbr | and asterisk keeps talking about rf83... not complete |
| 16:41.35 | purplebob | got the first two parts working. just the busy part is baffling me. |
| 16:41.37 | bjohnson | VoIPMasta: http://www.voip-info.org/wiki-Asterisk |
| 16:41.52 | *** join/#asterisk Slainte (Slainte@207.228.155.26) |
| 16:41.58 | xkev | purplebob, queues.conf has a queue timeout config var, so it can return back to dthe dialplan when it expires |
| 16:42.14 | Madkiss | _Brian: So based on the number to call, there is no way to determine on where the call will go to? |
| 16:42.28 | _Brian | Madkiss: of course there is... |
| 16:42.30 | *** join/#asterisk freat (~freat@node-40242662.mdw.onnet.us.uu.net) |
| 16:42.37 | xkev | hellmasterbr, disable silence suppression. it stops the media stream |
| 16:42.38 | freat | g'morning |
| 16:42.41 | purplebob | yeah, thanks, got that part working. just trying to get the if the second # is busy part, go to voicemail. |
| 16:43.06 | _Brian | Madkiss: create a lcr table, and place in the numbers and who you want to deliver them with |
| 16:43.07 | xkev | there are samples on wiki for testing cause of return from Dial() |
| 16:43.26 | Madkiss | _Brian: erm. Well, erm ... |
| 16:43.27 | freat | can I configure * to use g.729 (I've got 2 licenses currently) first, then gsm if it runs out of g.729 licenses? |
| 16:43.38 | Madkiss | _Brian: After all, it's the whole FWD, Freenet and Sipgate net ;) |
| 16:43.45 | freat | not sure how to set the preferred codec |
| 16:43.50 | xkev | purplebob, e.g. http://www.voip-info.org/tiki-index.php?page=Stdexten%20macro |
| 16:44.03 | RoyK | freat: using iax? sip? h323? |
| 16:44.05 | RoyK | what? |
| 16:44.09 | bjohnson | Madkiss: smaller installs just use the dialplan instead of using a lcr table |
| 16:44.21 | Madkiss | What the hell is a dialplan? |
| 16:44.27 | freat | RoyK: got sip phones, but connect to external VoIP provider via IAX2 |
| 16:44.31 | Madkiss | You folks keep telling me about a dial plan ... -sigh- |
| 16:44.32 | RoyK | ~lart Madkiss |
| 16:44.39 | Madkiss | bah |
| 16:44.39 | freat | RoyK: the provider supports g.729, |
| 16:44.48 | RoyK | freat: then you don't need g.729 licenses |
| 16:44.52 | xkev | madkiss dialplan == extensions.conf |
| 16:44.56 | RoyK | freat: as asterisk can run in passthrough mode |
| 16:45.02 | Zeeek | Madkiss |
| 16:45.04 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 16:45.05 | xkev | *CLI> show dialplan |
| 16:45.10 | RoyK | freat: disallow=all and allow=g729 in sip.conf |
| 16:45.22 | freat | RoyK: but what if they check their voicemail? |
| 16:45.31 | freat | RoyK: which is on the * server? |
| 16:45.31 | *** join/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 16:45.32 | _Brian | Why lart someone, when you can have the bot do it for you :) |
| 16:45.51 | pcm | Madkiss: not necessary .. if you first answer the line and you have incoming call in g729 and you want to playback/background you'll need g729 unless you have a prompt in g729 |
| 16:45.57 | shido6 | NO |
| 16:46.02 | Nugget | oh, kickass. x-lite added their acoustic echo cancellation stuff to x-pro. |
| 16:46.05 | Madkiss | pcm: err? |
| 16:46.12 | purplebob | thanks |
| 16:46.18 | shido6 | promprts cannot be heard if you do not have a g729 license and your endpoint is set to use g729 |
| 16:46.27 | freat | RoyK: the phones (Polycom IP 500s) do support g.729, but I have to change dtmfmode (in sip.conf for that phone) if I change codec |
| 16:46.32 | pcm | shido6: they can if they're recorded in G729 |
| 16:46.46 | shido6 | yes but the ivrs that come with asterisk arent in g729 |
| 16:46.55 | pcm | shido6: then transcode :) hehe |
| 16:46.56 | bjohnson | VoIPMasta: ready? |
| 16:47.04 | VoIPMasta | yup |
| 16:47.16 | pcm | shido6: they're not so that the codec would be selling |
| 16:47.17 | shido6 | u can change each ivr into that codec using a hacked soundeditor |
| 16:47.17 | xkev | g729 gobbles up cpu anyway |
| 16:47.22 | shido6 | transcoding requires licenses |
| 16:47.28 | shido6 | to and from g729 |
| 16:47.40 | pcm | shido6: i mean you can convert it to g279 |
| 16:47.44 | xkev | ..even on the polycom, they've had bugs with conferencing and g729 due to cpu load |
| 16:47.45 | bjohnson | Madkiss: the dialplan is the extensions.conf file .. it tells asterisk how to route calls (both in AND out) |
| 16:48.19 | *** join/#asterisk zpn (~xpn@dhcp-166.digium.com) |
| 16:48.20 | CCDAS | shido6: let me know when you are available for one more question. |
| 16:48.34 | hellmasterbr | sure asterisk is a great piece of software |
| 16:48.39 | hellmasterbr | but you guys, are best |
| 16:48.59 | freat | hmmm |
| 16:49.09 | shido6 | CCDAS just ask but thanks for the heads up |
| 16:49.16 | CCDAS | shido6: jitter problem resolved. New message: H.323 CONTROL PROTOCOL ERROR (Logical Channel : Timeout). Any thoughts? |
| 16:49.20 | freat | back to my original question... can I make g.729 preferred and gsm second? |
| 16:49.47 | shido6 | sometimes I walk away and turn on the speech agent so it says whatever is posted to the channel but i dont always catch the question as its scrolled up too high and Im too lazy to go through them all |
| 16:49.56 | bjohnson | VoIPMasta: first is method for free calls is that everyone you want to call sets up a voip system to send and recieve calls. You dial them directly using fully qualified domain names (fqdn) or IP addresses |
| 16:49.57 | *** join/#asterisk TheEmperor (TheEmperor@218.111.49.115) |
| 16:50.14 | Madkiss | Why the hell does asterisk continue to dial a number even if I _shut down_ the client? |
| 16:50.29 | Madkiss | my "dialplan" must be total screwed |
| 16:50.49 | xkev | define client |
| 16:50.56 | pcm | madkiss: the hangup must not be received |
| 16:50.59 | Madkiss | linphone, a soft client |
| 16:50.59 | _Brian | define hell |
| 16:51.07 | _Brian | :) |
| 16:51.09 | xkev | if you just disappear off the net (say sip/h323) how is it supposed to know you've hung up? |
| 16:51.21 | Madkiss | Hell is definetely everywhere where asterisk is involved. -duck- |
| 16:51.35 | xkev | then use something else |
| 16:51.38 | pcm | madkiss: if you have analog fxs port and you are setup to recognize the flash (transfer=yes and something else) then your hangup is gonig to be delayed by 3-4 sec |
| 16:51.44 | xkev | I recommend LignUp sip pbx |
| 16:51.49 | forkqueue | Madkiss: It sends the SIP/invite, you disappear, it has no way of knowing you're no longer there.. |
| 16:52.03 | forkqueue | Madkiss: Pebkac |
| 16:52.12 | xkev | if you want a gui and a 4GB oracle-based pbx that can't do half of what * can |
| 16:52.12 | Madkiss | yeah, but I clicked several times on "release or refuse" in linphone |
| 16:52.14 | hellmasterbr | have a great time ppl |
| 16:52.14 | bjohnson | VoIPMasta: second method (since the first method is not practical for most end users) is to use some kind of registry service so that users "register" where they can be found and you dial their account at the registry service. Probably the most well known is Free world Dialup (often referred to as FWD) which is of course free .. and also allows you to dial some toll free numbers in a variety of countries as well as connect to people connected |
| 16:52.15 | bjohnson | to voip through certain other voip providers |
| 16:52.21 | Madkiss | It should _then_ already have stopped to dial |
| 16:52.30 | Beirdo | has anyone ever tried to use mplayer for MOH? |
| 16:52.43 | xkev | madkiss are you behind nat? |
| 16:52.49 | xkev | you using sip or iax? |
| 16:53.02 | Madkiss | xkev: No, erm, asterisk runs on the computer that has net access. |
| 16:53.11 | Madkiss | xkev: sip |
| 16:53.14 | bjohnson | Beirdo: thought about it .. decided it should be possible .. didn't actually try it |
| 16:53.19 | Beirdo | hehe |
| 16:53.23 | xkev | linphone btw, is a piece of crap |
| 16:53.23 | hmmhesays | asterisk runs on the computer that has ESP access |
| 16:53.36 | Madkiss | xkev: Okay, what to use instead? |
| 16:53.37 | bjohnson | VoIPMasta: still with me? |
| 16:53.43 | VoIPMasta | yes |
| 16:54.01 | xkev | kphone sucks about half the time, and the best thing I've found is x-lite inside wine |
| 16:54.04 | Beirdo | OK, so I might end up having to do it.. I want to use an online stream (RealAudio) for the MOH |
| 16:54.13 | hmmhesays | sjphone works pretty well |
| 16:54.19 | xkev | yeah, and sjphone is gud |
| 16:54.44 | xkev | madkiss, try some other UAs before you blame asterisk :) |
| 16:54.56 | Nivex | I've had good luck with iaxcomm connected to either my asterisk box or FWD's IAX offering |
| 16:54.59 | hmmhesays | in linux it's much better than windows |
| 16:55.46 | *** join/#asterisk p0lar (~p0lar@64.254.254.4) |
| 16:55.59 | Nivex | kphone is great for voice but tries (and fails) to do DTMF inband |
| 16:56.06 | xkev | beirdo, if you can get mplayer or something to feed raw (or whatever it is /dev/dsp takes, I just used mpg123) to stdout, you can use it as moh |
| 16:56.18 | Beirdo | cool |
| 16:56.24 | xkev | I have five streaming mp3 feeds going for mine |
| 16:56.24 | bjohnson | ok. fourth method is to use e164.org and it's sister site http://www.like2fone.com. With this system, you can register your phone number and others can do a lookup for that phone number and dial to the voip destination you have specified instead of using the pstn. Like2phone is a system that allows you to easily change the voip destination specified at e164.org. These lookups are very easy with asterisk using enumlookup (example configs on |
| 16:56.25 | bjohnson | e164.org site) |
| 16:56.33 | *** join/#asterisk nicolasg (~nicolasg@host-52.6.60.66-ta.adsl.netizen.com.ar) |
| 16:56.40 | Beirdo | at 8ksample/8 bit/mono? |
| 16:56.44 | xkev | yes |
| 16:56.49 | Beirdo | gotcha |
| 16:57.01 | tih | Any of you tried to run firmware 1.0.5.22 on Grandstream BT100? |
| 16:57.17 | p0lar | Has anyone had any luck with SuSE 9.2 for pwlib compilation? |
| 16:57.18 | xkev | be aware that when someone is not listening to the audio, it will block until stdout buffer fills. this will usually break the stream |
| 16:57.36 | shido6 | p0lar |
| 16:57.38 | VoIPMasta | bjohnson: I guess you missed the 3rd method |
| 16:57.38 | shido6 | still at it, eh? |
| 16:57.42 | xkev | ariel_, I can dial one of my MOH and hear yesterday's news for 10 seconds before it rebuffers :) |
| 16:57.45 | p0lar | :D |
| 16:57.49 | xkev | s/ariel_,/ie:/ |
| 16:57.52 | Zeeek | tih yes |
| 16:58.01 | Beirdo | cool. |
| 16:58.09 | bjohnson | VoIPMasta: ahh yes .. renumber as approipriate .. they aren't necessarily in any order anyway |
| 16:58.12 | tih | Zeeek: It works for me on their HT286, but not on the BT100... |
| 16:58.17 | Zeeek | tih doesn't work for me |
| 16:58.21 | p0lar | I had 9.1 running.. 9.2 breaks things. :( |
| 16:58.35 | Zeeek | tih but then no fw over .11 works on mine :( |
| 16:58.40 | Beirdo | I guess the next thing to figure out would be how to get mplayer to buffer up less so the delay won't be as great |
| 16:58.41 | tih | Zeeek: damn thing doesn't try to call out -- it's arping for the name server, but that's all it does. |
| 16:58.42 | bprice20 | any experience with voipong |
| 16:58.49 | Madkiss | okay, the problem continues ... |
| 16:58.50 | Zeeek | tih - hey that's my prob too |
| 16:58.58 | Zeeek | tih mine doesn't do any DNS at all |
| 16:59.00 | bprice20 | supposed to be a sniffer that can parse voip packets into wav's |
| 16:59.02 | Madkiss | I called my mobile phone, I rejected the call at the mobile phone |
| 16:59.10 | tih | Zeeek: OK, thanks - guess it's back to .11 on the BT100, then. |
| 16:59.13 | Madkiss | and asterisk continued to dial |
| 16:59.19 | stickynomore | could anybody tell me what i need to do get callerid working right again on my sip phones? I updated to the newest stable as of yesterday and now it only will show name and asterisk or asterisk/asterisk as the caller |
| 16:59.23 | tih | Zeeek: weren't you the only person in the world this happened to? |
| 16:59.27 | Zeeek | tih I've tried ever version since 11 and wrote GS |
| 16:59.30 | Zeeek | ya |
| 16:59.34 | hmmhesays | callerid is broken now isn't it? |
| 16:59.42 | Zeeek | and I keep trying new version s :) |
| 16:59.46 | stickynomore | in stable too? |
| 16:59.52 | Zeeek | nah |
| 17:00.00 | tih | Zeeek: here's a data point: my BT100 was working on a higher firmware release than .11, at work, on a /24 net, but it's not working here, on a /27 net. |
| 17:00.27 | tih | Zeeek: I'll experiment, and let you know what I find. |
| 17:00.29 | Zeeek | hmmmmm but I'm on a 24 |
| 17:00.34 | bjohnson | VoIPMasta: next. there are a few voip providers that will give you a little time for a sample. voipjet.com is one (they give you $0.25 which roughly equals 15 minutes to ANY North American phone number) |
| 17:00.37 | tih | Zeeek: ah. Hmpf. |
| 17:00.40 | Zeeek | ya please because there are new features |
| 17:00.54 | Zeeek | <PROTECTED> |
| 17:01.00 | tih | Zeeek: yup. |
| 17:01.04 | VoIPMasta | bjohnson: listing that method would be like telling the users to cheat |
| 17:01.15 | Zeeek | tih they told me "use STUN" |
| 17:01.18 | Zeeek | ya right |
| 17:01.24 | tih | Oh, well -- it's a good excuse to put up the gratissip.dk tftp server, so I can differentiate between units. |
| 17:01.37 | tih | ...but *after* dinner, I think. |
| 17:01.45 | Zeeek | might has well have said "reinstall windows" |
| 17:01.49 | VoIPMasta | bjohnson: the termination providers that offer a free test account are providing it for test purposes, so that you can try their services before sending a payment. They do so because they know that the service will convince the customer and will return the 0.25 in profits. |
| 17:01.52 | xkev | bjohnson, in case of voipjet or similar, how do they feel about being the LD gateway for a wannabe voip provider? :) do they override caller-id/ani presentation? |
| 17:01.58 | bjohnson | VoIPMasta: it's a valid offer provided by the voip provider |
| 17:02.03 | Slainte | hmmm, Anyone have it when only certain numbers dialled will cause echo? I have tried everything from setting taps to 256, echotraining, etc |
| 17:02.08 | Zeeek | tih eats early - must be in holland :) |
| 17:02.13 | bjohnson | xkev: I have no idea |
| 17:02.21 | Zeeek | where all restaurans are closed at like 8PM! |
| 17:02.23 | VoIPMasta | bjohnson: however telling them to use it for free PSTN calling would make the provider to stop offering such accounts |
| 17:02.29 | xkev | k, just fishin. light went off in my head :) |
| 17:02.54 | tih | Zeeek: I'm in Norway, actually... |
| 17:02.54 | VoIPMasta | bjohnson: As I doubt that VoIPJet would be able to stay in business if they gave 5,000 0.25 test accounts |
| 17:03.01 | bjohnson | VoIPMasta: we're just listing methods available. The services offered may change from time to time |
| 17:03.17 | Zeeek | tih GO EAT! |
| 17:03.32 | coppice | Slainte: only some numbers echoing badly is normal when you don't have a decent echo canceller |
| 17:03.39 | bjohnson | VoIPMasta: and yes .. using a free sample MAY be enough to convince someone to spend $10 |
| 17:04.03 | VoIPMasta | bjohnson: but someone that is looking for "free voip" is because he/she is not willing to pay |
| 17:04.19 | bjohnson | VoIPMasta: I was thinking that this may be most suitable for people just getting into voip and looking for a low/no cost method to start |
| 17:04.31 | Beirdo | VoIPMasta: or because they are starting out and want to try stuff out. |
| 17:04.44 | VoIPMasta | bjohnson: then the right category would be "voip newbies" and not "free voip" |
| 17:05.03 | Beirdo | I'm not willing to pay for VOIP when I'm not sure how well it will all work (for instance) |
| 17:05.04 | Nugget | <vaewyn> "free VoIP" == "Punch the monkey for a free ipod" |
| 17:05.18 | VoIPMasta | .LoL |
| 17:05.19 | bjohnson | VoIPMasta: in the case of voipjet sample .. they advertise it on their main web page .. I don't think they will mind if we direct more potential customers their way |
| 17:05.24 | vaewyn | bwahaha.. shoot now I have been quoted |
| 17:05.38 | Beirdo | and certainly not planning on doing a locked-in contract like Vonage's "no contract but have to pay to disconnect" crap |
| 17:05.53 | Slainte | coppice: I have echocancel=256 in my zapata.conf, I have echocancelwhenbridged=yes and echotraining=yes |
| 17:05.55 | Zeeek | I though all voIP was Free? |
| 17:05.59 | VoIPMasta | bjohnson: I'm sure they won't mind if we send potential customers... but you are talking about sending them kiddies that will sign up more than once for a free test account |
| 17:06.10 | Slainte | coppice: however only certain nu7mbers will have an echo, and it is always callside |
| 17:06.58 | VoIPMasta | bjohnson: The "regular" traffic that hits VoIPJet's website is of ppl looking for "termination providers" and not for "free voip" |
| 17:07.04 | xkev | anyone have CNAM (caller name) on their PRI? |
| 17:07.04 | bjohnson | Zeeek: free after you pay for internet connection, hardware, and time to learn |
| 17:07.11 | VoIPMasta | bjohnson: That's why they can advertise it on their website |
| 17:07.19 | Slainte | coppice: I have read every signle snippit on the Wiki about it, but dont have a clear direction. People with similar experiences and solutions is my only hope I think. |
| 17:07.20 | bjohnson | Zeeek: it's the pstn<->voip gateways that are more restrictive |
| 17:07.23 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:07.35 | Beirdo | bjohnson: that's priceless |
| 17:07.41 | Beirdo | "Free $0.25 use" |
| 17:07.42 | Beirdo | hehe |
| 17:07.42 | Zeeek | well, $600 + $25/month is almost free? |
| 17:07.47 | coppice | Slainte: sad, isn't it, but quite normal without a good echo canceller. * doesn't have a good echo canceller |
| 17:08.01 | Beirdo | someone needs to teach them that Free = $0.00 |
| 17:08.09 | VoIPMasta | Zeeek: VoIP is designed to "lower long distance costs" not as a "worldwide free phone system" |
| 17:08.11 | *** join/#asterisk ionix (~ioni@66.38.219.151) |
| 17:08.12 | ionix | Hello |
| 17:08.33 | Zeeek | I think voIP ans asterisk are just an excuse to spend hours on IRC |
| 17:08.34 | ionix | How do I make asterisk queue calls even when there are nobody able to answer the queue ? |
| 17:08.35 | bjohnson | VoIPMasta: you decide whether it should be included or not .. the next person (Beirdo?) can edit as they see fit |
| 17:08.58 | Zeeek | but seriously, the "free worldwide call" question comes up daily on FWD |
| 17:09.03 | xkev | ionix, there's a flag in queues.conf |
| 17:09.05 | VoIPMasta | Zeeek: Then you've found your reason to stay on IRC, most of us are here to help and to research |
| 17:09.10 | ionix | xkev: yeh joinempty |
| 17:09.12 | ionix | it's set yo yes |
| 17:09.15 | Slainte | coppice: are you saying there is nothing I can do, and that * is not a production quality system? With all the people who use it certainly someone has been able to minimize the impact. |
| 17:09.18 | xkev | joinempty = yes |
| 17:09.23 | ionix | I use AgentCallbackLogin, does that makes a difference |
| 17:09.25 | xkev | then it should work |
| 17:09.29 | coppice | VoIP is just a plot by DSP silicon vendors to get their parts into otherwise cheap and simple phones :-) |
| 17:09.36 | VoIPMasta | Zeeek: FWD is a nice network, but you can't expect that the entire world would be registered at FWD, and I do also think that FWDs server wouldn't be able to handle the load |
| 17:09.44 | vaewyn | coppice: hehehehe |
| 17:09.49 | Zeeek | VoIPMasta I've been helped and I've helped in about equal amounts |
| 17:10.07 | ionix | http://pastebin.com/239217 |
| 17:10.09 | ionix | look at that |
| 17:10.36 | Zeeek | I mentioned FWD because it is lietrally a gateway to voIP in that geeks like me got on FWD and three days later had asterisk running :) |
| 17:10.39 | pcm | coppice: sounds like truth :) |
| 17:10.53 | ionix | queue settings: |
| 17:10.53 | ionix | exten => 6612,1,Answer |
| 17:10.53 | ionix | exten => 6612,2,Queue(tech) |
| 17:10.53 | ionix | exten => 6612,3,Hangup |
| 17:10.55 | redder86 | http://bugs.digium.com/bug_view_page.php?bug_id=0002556 does anyone know what the "earlier" bug report is? I'd like to read it and mark didn't referece it. |
| 17:11.05 | bjohnson | VoIPMasta: we're getting off track here .. people looking for unlimited, free, world wide calling are likely going to lose patience with the time involvement and restrictions inherent in these systems |
| 17:11.10 | bjohnson | VoIPMasta: next. |
| 17:11.14 | Zeeek | so all the new people come through FWD looking for free PSTN but many are just curious about the technology - and they end up with * |
| 17:11.29 | xkev | ionix what is your retry, timeout, and strategy? |
| 17:11.37 | ionix | let me paste on pastebin |
| 17:11.41 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 17:11.44 | mAsH` | hi all |
| 17:11.49 | bjohnson | Madkiss: look at the superdial macro on the wiki .. it is easiest way to add lcr to your extensions.conf |
| 17:11.51 | Zeeek | bjohnson - absolutely and screw them anyway - if I wanna make a call, I can do it on PSTN too |
| 17:11.56 | VoIPMasta | Zeeek: I registered at FWD and found that it was of no use to me, none of my contacts uses FWD |
| 17:12.30 | Beirdo | bjohnson: nice of them to put in a bogus address in whois too |
| 17:12.35 | Zeeek | VoIPMasta same here - it can be useful for testing though and there are a FEW knowledgeble folks who hang there |
| 17:12.37 | vaewyn | Dundi and e164 are what everyone assumes FWD is |
| 17:12.38 | ionix | http://pastebin.com/239219 |
| 17:12.39 | bjohnson | VoIPMasta: you may find other uses for it .. could be used for your own SIP devices to deal with NAT (SIP to FWD and IAX to your *) |
| 17:12.40 | Beirdo | and a bogus phone number |
| 17:12.50 | Zeeek | e164 is a great idea |
| 17:13.02 | VoIPMasta | bjohnson: why not just plug them directly into my asterisk? |
| 17:13.14 | Zeeek | Call "tellme" at FWD - it's a good demo |
| 17:13.24 | bjohnson | VoIPMasta: doesn't work in a double nat system .. may not apply to you but DOES apply to others |
| 17:13.47 | egrat | is it possible to use asterisk with a normal voice modem and a standard pots line? |
| 17:13.50 | hmmhesays | there's no good reason to be using a double natted system |
| 17:13.51 | VoIPMasta | bjohnson: I have my own STUN server, and I have used SIP phones on double nat without a problem |
| 17:13.59 | VoIPMasta | egrat: no |
| 17:14.09 | xkev | ionix try setting a retry= value |
| 17:14.20 | shido6 | egrat get some digium gear and an IP Phone |
| 17:14.22 | shido6 | or IAXy |
| 17:14.30 | ionix | ok |
| 17:14.30 | egrat | so i can't run my own asterisk server from home using my normal phone line to make calls through that line while away from the house using a sip phone? |
| 17:14.34 | VoIPMasta | egrat: you need a FXO card to connect your phone line into your asterisk server |
| 17:14.35 | bjohnson | VoIPMasta: please note that I was just describing an alternate use of FWD |
| 17:14.36 | mAsH` | can i modify the language of the message that voicemail send me via email ? |
| 17:15.02 | *** join/#asterisk ezabi (~ezabi@82.201.194.37) |
| 17:15.02 | shido6 | yes |
| 17:15.21 | xkev | egrt, sipura spa-3000 is nice for home use too |
| 17:15.23 | bjohnson | VoIPMasta: next .. there's that sister site to FWD .. whaterever they call it today (was Bellster). you subscribe and in turn for sharing your pstn connection, you get to use other people's pstn connection |
| 17:15.23 | vaewyn | egrat: you can... need a TDM400P with 1 FXO interface... 1FXS also if you want to drive your internal phones from it also |
| 17:15.24 | VoIPMasta | mAsH`: you can modify almost everything, asterisk is open source |
| 17:15.25 | shido6 | u can modify the email that goes out |
| 17:15.37 | bjohnson | xkev: definitely .. got one at home |
| 17:15.50 | egrat | thanks guys. |
| 17:15.58 | vaewyn | egrat: or a spa-3000 with a * server on the network |
| 17:16.00 | xkev | no caller-id on my clone MD3200 card :( shoulda got the sipura-3k instead of 2k |
| 17:16.23 | bjohnson | hmmhesays: sure there is .. EASE (ease for everything BUT SIP) |
| 17:16.37 | VoIPMasta | bjohnson: do you have the URL to that site? |
| 17:16.45 | hmmhesays | since when does double nat make anything easier? |
| 17:16.51 | mAsH` | VoIPMasta: i known ;) |
| 17:16.56 | ezabi | guys does anybody know where does the variable progzone in the zaptel.conf get the definition for the zones from? |
| 17:17.17 | egrat | when they say 2.4 kernel, do they mean 2.4 only or 2.4+ (can I use 2.6?) |
| 17:17.28 | Zeeek | doubleNAT can be done - it's a little like sex with aliens - I do it every day. DoubleNAT I mean |
| 17:17.35 | bjohnson | VoIPMasta: let me see if I can find one |
| 17:17.43 | tzanger | sweet my blackfin dev board arrived today |
| 17:17.46 | xkev | ionix, did that help? |
| 17:17.51 | ezabi | egrat: i'm working with 2.6 here and it's fine |
| 17:17.55 | VoIPMasta | mAsH`: so why do you ask? I mean you asked if it was possible to translate the message sent in the email but you say you did already know that... |
| 17:17.56 | egrat | ok. thanks. |
| 17:18.01 | VoIPMasta | mAsH` :) |
| 17:18.02 | xkev | it looks lik you were only getting one queue run |
| 17:18.12 | hmmhesays | it would rock if you could compile asterisk as a kernel module |
| 17:18.31 | xkev | uh why? |
| 17:18.32 | VoIPMasta | egrat: I have it running on 2.6 without a problem |
| 17:18.43 | hmmhesays | just to say my kernel has asterisk support |
| 17:18.52 | xkev | there's a place for userland, and there's a place for kernel space |
| 17:18.59 | ezabi | LOL @ hmmhesays |
| 17:18.59 | xkev | this ain't windows :) |
| 17:19.04 | bjohnson | VoIPMasta: just thought of another possible use for FWD .. connecting to someone else that uses another voip provider (like vonage) |
| 17:19.21 | hmmhesays | i guess I really don't have any good reasons other than that |
| 17:19.40 | xkev | you could "say" it has asterisk support by compiling zaptel :) |
| 17:19.50 | ezabi | anybody around here into this zaptel thing that can help me? |
| 17:20.10 | pcm | ezabi: what zaptel thing ? |
| 17:20.13 | shido6 | ezabi whats wrong? |
| 17:20.17 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 17:20.21 | bjohnson | VoIPMasta: http://www.fwdout.net/web/ |
| 17:20.23 | VoIPMasta | ezabi: what do you need |
| 17:20.34 | hmmhesays | yeah xkev, i suppose you could |
| 17:20.49 | Zeeek | bjohnson I've never tried that but FWD performance is already a little sketchy with asterisk (most users have just a phone) so I wonder how well it'd work to try a call to one of their peers |
| 17:21.00 | bjohnson | VoIPMasta: I think that's all I can think of now |
| 17:21.15 | ezabi | my problem is as stated earlier is that i cannot get my zaptel fxo to detect my local pstn call progress |
| 17:21.25 | Zeeek | calling the FWD echo or time ext are helpful tests though |
| 17:21.31 | shido6 | where are you geographically? |
| 17:21.34 | shido6 | ezabi ? |
| 17:22.07 | ezabi | shido6: in egypt |
| 17:22.11 | hmmhesays | a tiny little planet called earf |
| 17:22.16 | VoIPMasta | bjohnson: .ok, hold on, I'm writing everything down |
| 17:22.18 | bjohnson | Zeeek: I don't know .. I don't use it .. but it is supposed to be possible |
| 17:22.18 | shido6 | hrmmm |
| 17:22.18 | kram | ezabi: ahlin wesahlin |
| 17:22.28 | *** join/#asterisk darkflib (darkflib@dialup019.ts006.bmt.esat.net) |
| 17:22.35 | Zeeek | kram is there a language you don't speak? |
| 17:22.36 | ezabi | kram: hala feek |
| 17:22.37 | bjohnson | hmmhesays: wow .. the rest of use are one earth |
| 17:22.42 | bjohnson | err .. on earth |
| 17:22.56 | Zeeek | [solder] hahaha |
| 17:23.14 | kram | ezabi: oomy musraya |
| 17:23.19 | ezabi | so i just called the |
| 17:23.21 | ManxPower | happy mardi gras |
| 17:23.34 | Zeeek | and even Manx speaks French! |
| 17:23.46 | ezabi | kram and ur other half? |
| 17:23.57 | kram | american mutt :) |
| 17:24.05 | kram | mom is from iskanderaya |
| 17:24.07 | ManxPower | Zeeek, "mardi gras" is actually Bulgarian. The french stole it from us. |
| 17:24.17 | Zeeek | who's us? |
| 17:24.37 | ezabi | kram: cool nice to meet u, i just returned from there |
| 17:24.43 | kram | cool ;) |
| 17:24.45 | Zeeek | kram ah that's why she speaks French |
| 17:24.50 | kram | i haven't been since december 2001 |
| 17:24.54 | ManxPower | Me new /away message seems to be doing it's job. |
| 17:25.09 | ezabi | kram: it changed a lot |
| 17:25.23 | kram | hopefully for the better |
| 17:25.27 | bjohnson | Zeeek: "us" are the people being pursued by "them" |
| 17:25.30 | ezabi | kram: really a beauty now |
| 17:25.31 | xkev | manx, you rollin in the dough yet? :) |
| 17:25.31 | kram | mom always talks about how it was back in the 50's and 60's |
| 17:25.37 | kram | awesome |
| 17:25.42 | kram | hopefully one day i can return |
| 17:25.57 | CCDAS | Anyone familiar with one-way audio with * H.323? |
| 17:25.59 | vaewyn | kram: you've got the airmiles ;P |
| 17:26.06 | ManxPower | xkev, no, it was to keep people from /msg'ing me. |
| 17:26.11 | kram | hehe, yah a lot of it is time |
| 17:26.15 | outtolunc | kram's gonna make a chan_transporter .. yeehaw |
| 17:26.15 | ezabi | kram: ok gimme abuzz on my FWD if u come ;) |
| 17:26.18 | kram | but maybe one day i can come to Egypt and talk about Asterisk hehe |
| 17:26.36 | ManxPower | CCDAS, one-way audio is almost always 1) a codec problem or 2) a NAT problem. |
| 17:26.41 | shido6 | CCDAS I am sending you messages |
| 17:26.45 | Zeeek | ManxPower http://www.eastjeffersonparish.com/culture/MARDIGRA/HISTORY/history.htm |
| 17:26.51 | Zeeek | nothing mentioned here |
| 17:26.55 | xkev | manx, yes I was being facetious :) |
| 17:27.09 | CCDAS | what if is not nat'd and if both are using Ulaw? |
| 17:27.14 | ezabi | kram: we're already doin that, in fact i was thinkin of demoing it in our next linux install fest |
| 17:27.26 | ManxPower | Zeeek, Of course. They never actually admit they stole it. |
| 17:27.29 | vaewyn | kram: You gonna be hanging out for most of VON or just there for your speaking? |
| 17:27.33 | shido6 | are you running cvs head ccdas? |
| 17:27.46 | Zeeek | what is theis they and us bullshit? |
| 17:27.47 | ezabi | ManxPower: NAT |
| 17:27.52 | ManxPower | CCDAS, If you NATing H323 you have some REALLY serious problems. |
| 17:28.04 | CCDAS | Asterisk CVS-v1-0-01/08/05-20:21:08 built by root@MSLS on a i686 running Linux |
| 17:28.06 | Zeeek | TINT TINU |
| 17:28.39 | CCDAS | No nat, private network access only. |
| 17:28.54 | ManxPower | CCDAS, As far as I know there is NO support for NAT with either of the H323 channel drivers available for Asterisk. |
| 17:29.22 | ManxPower | CCDAS, Then you may have a codec issue. disallow=all and allow=ulaw on BOTH sides. |
| 17:29.30 | ManxPower | CCDAS, Other than that I cannot help you further. |
| 17:29.37 | CCDAS | ok... I will see. |
| 17:29.44 | ezabi | H323 isn't that nice anyway |
| 17:30.40 | bjohnson | Zeeek: you use externip= for double nat correct? |
| 17:31.00 | Zeeek | yes sir, and no domain name cause that didn't work |
| 17:31.18 | Zeeek | I changed it to =domain.com and when the ip changed it BROKE! |
| 17:31.23 | *** join/#asterisk Pantanero (~Pantanero@a213-22-84-10.netcabo.pt) |
| 17:31.32 | Zeeek | whereas using an ip and restarting wokred |
| 17:31.33 | bjohnson | ManxPower: kind of like US Whitehouse tours never mention the war of 1812 |
| 17:31.54 | ta[i]nted | have you guys seen http://maps.google.com ? |
| 17:31.58 | ta[i]nted | that is AMAZING |
| 17:32.04 | hmmhesays | google has maps? |
| 17:32.06 | hmmhesays | wahoo! |
| 17:32.22 | bjohnson | Zeeek: I'm playing with ways to try to find one that doesn't require knowing the remote IP address or remote LAN subnet |
| 17:32.42 | ta[i]nted | hmmhesays not just maps.. the GUI is really fantastic |
| 17:33.09 | hmmhesays | i see that |
| 17:33.11 | Zeeek | bjohnson what d'ya mean? One what? |
| 17:33.15 | hmmhesays | that's farking awesome |
| 17:33.29 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 17:33.32 | bjohnson | Zeeek: a method |
| 17:33.40 | bjohnson | Zeeek: I'm playing with ways to try to find a method that doesn't require knowing the remote IP address or remote LAN subnet |
| 17:34.07 | Zeeek | I still don't follow |
| 17:34.32 | Zeeek | I have a phone at home that connects to the office asterisk - which is on dynamic ip |
| 17:34.53 | Zeeek | the phone happens to be on a fixed one and so it does't register |
| 17:35.07 | vaewyn | ta[i]nted: yeah... google and their httpxmlrequest wonders rock |
| 17:35.12 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 17:35.24 | ezabi | so seemingly nobody has a thing to say about this zap detect |
| 17:35.25 | Zeeek | vaewyn this is what you were talking about earlier |
| 17:35.32 | vaewyn | Zeeek: yep |
| 17:35.33 | Zeeek | good demo of the concept |
| 17:35.49 | bjohnson | tzanger: they even have your neighbourhood on google maps |
| 17:35.57 | Zeeek | indeed "fucking amazing" is a decent adjective |
| 17:36.03 | ezabi | anyway later everbody, bye |
| 17:36.19 | vaewyn | Zeeek: http://www.andrews.edu/wwwrogue/Watch/index.html <- is the demo I came up with |
| 17:36.34 | vaewyn | Zeeek: google's are way better demos :P |
| 17:36.45 | Zeeek | ...looking... |
| 17:37.14 | *** part/#asterisk ezabi (~ezabi@82.201.194.37) |
| 17:37.20 | vaewyn | Mine just shows the dynamic data a little more forcefully :P |
| 17:37.37 | Zeeek | vaewyn so tell me how you are generating the 'data' |
| 17:37.50 | Madkiss | err. My asterisk is logged into the Freenet server, and when I call the number, I get a strange tone (beep-beep-beep without interrupts) but asterisk does absolutely _nothing. It doesn't even write something in the log. |
| 17:38.03 | vaewyn | Zeeek: mod_perl script that is watching every 2 seconds for changes in 'top' |
| 17:38.29 | Zeeek | if no change, no write so no GET? |
| 17:38.34 | vaewyn | Zeeek: could be longer... or better done... but this is just a proof |
| 17:38.37 | ManxPower | Madkiss, You are watching the Asterisk CLI? |
| 17:38.41 | bjohnson | vaewyn: was that for tzafrir ? |
| 17:38.45 | Zeeek | I'm still missing a basic element |
| 17:38.53 | Madkiss | ManxPower: Yes. |
| 17:39.09 | ManxPower | Madkiss, then the call is not getting to asterisk and the problem is somewhere else. |
| 17:39.53 | vaewyn | Zeeek: It does a get right off the bat... I don't return data and allow another GET until a change has taken place... like if I turned this up to every 3 minutes instead of every 2 seconds there would be one GET every 3 minutes instead of once every 2 seconds |
| 17:40.23 | vaewyn | Zeeek: basically I only send changes when they happen... if they didn't happen then I would timeout and the browser would try again |
| 17:40.46 | vaewyn | Zeeek: thus making a poor mans 'push' with 'pull' technology only |
| 17:41.07 | Zeeek | I see the value but not the mechanism server side |
| 17:41.23 | vaewyn | can you read perl? |
| 17:41.24 | Zeeek | the client code excutes onLoad |
| 17:41.46 | Zeeek | decipher, yes? |
| 17:42.03 | Zeeek | yiou game to pastebin to the world? |
| 17:42.09 | Zeeek | or rather msg? |
| 17:42.12 | Madkiss | ManxPower: Well, without asterisk, it works. |
| 17:42.18 | *** join/#asterisk Dalion (anon@Toronto-HSE-ppp3771411.sympatico.ca) |
| 17:42.23 | vaewyn | Zeeek : is there http://pastebin.ca/5476 |
| 17:42.33 | VoIPMasta | what would you say is the easiest way to add LCR for PSTN calls? |
| 17:42.49 | bjohnson | VoIPMasta: superdial macro on wiki |
| 17:42.51 | vaewyn | Zeeek: is mod_perl just cause I am into that... could be a simple CGI or whatever |
| 17:42.52 | Zeeek | thanks vaewyn I think this will be interesting |
| 17:43.04 | Dalion | why is it so har dto restart * |
| 17:43.05 | Dalion | lol |
| 17:43.10 | Zeeek | I can easily install that if it were advantageous but we mostly use PHP |
| 17:43.11 | Dalion | so many users on system |
| 17:43.16 | bjohnson | VoIPMasta: just list one route after the next in your extensions.conf |
| 17:43.26 | vaewyn | techincally that code will watch any text producing command and when it differs sends an update |
| 17:43.39 | vaewyn | and right now it polls every 2 seconds |
| 17:43.41 | bjohnson | VoIPMasta: and of course, pattern matching |
| 17:43.46 | Dalion | guess we need a reload realtime |
| 17:43.48 | Dalion | instead |
| 17:43.53 | Zeeek | I'm trying to see how I could make this into a show zap channels type interface |
| 17:44.07 | Zeeek | So I know who they(re talking to at the office :) |
| 17:44.37 | ManxPower | Madkiss, If the call isn't getting to asterisk.... |
| 17:45.05 | bjohnson | VoIPMasta: for instance, for my long distance I check e164.org (thanks to Nugget), then try one voip provider, then the next, then the third, then try my 3 pstn lines in sequence until I get a way out |
| 17:45.08 | vaewyn | Zeeek: replace "top -bn 1" with "asterisk -rx 'zap show channels'" and probably up the wait time a bit so you don't pound * |
| 17:45.08 | VoIPMasta | bjohnson: that's what I need, pattern matching, and I don't see it in superdial |
| 17:45.47 | Zeeek | vaewyn I'll have to play around because as I said, I have php installed but not mod_perl atm |
| 17:46.08 | vaewyn | Zeeek: is doable in PHP... so have fun :P |
| 17:46.21 | Zeeek | thanks this is an interesting direction! |
| 17:46.34 | Zeeek | (as if I needed more new projects!) |
| 17:46.58 | vaewyn | If you really want to get creative... return javascript encoded data structures and then parse them to set images and forms and such to have graphical status indicators ;P |
| 17:47.10 | vaewyn | but that is getting fancy :P |
| 17:47.25 | vaewyn | yeah... new timewaisters :P |
| 17:47.30 | vaewyn | wasters even |
| 17:47.32 | Zeeek | actually, why not then just generate the graphics iserver side int he first place |
| 17:48.01 | Madkiss | ManxPower: Negative, it works when connecting with the Windows software |
| 17:48.09 | Zeeek | anyway it still comes down to "sampling" server side, just avoids the pull |
| 17:48.14 | vaewyn | You could... but transfering a big graphic blob is harder then letting a javascript toggle between 2 piddly images for each channel |
| 17:48.37 | Zeeek | big graphics blob? one bw gif with a number written in it |
| 17:48.38 | bjohnson | VoIPMasta: http://pastebin.ca/5479 |
| 17:48.48 | Zeeek | 800 bytes! |
| 17:48.59 | Zeeek | with css, 400 and no graphic |
| 17:49.13 | Zeeek | and use a local css! |
| 17:49.36 | vaewyn | Zeeek: ohh but i was thinking more than that... I was thinking an interface that would have red/green etc indication... and in use indicators and boxes that keep track of "time in use" for each channel... and...etc... etc... etc |
| 17:49.47 | Zeeek | I know, I know |
| 17:49.50 | VoIPMasta | bjohnson: thanks :) |
| 17:49.54 | vaewyn | hehehe |
| 17:49.59 | bjohnson | VoIPMasta: that's my LD LCR routing .. tries 4 voip routes and then pstn .. the order they are tried is my lcr. I just include that context where I want users to be able to dial out |
| 17:50.58 | bjohnson | VoIPMasta: easy to add or changing routing since the macro doesn't change .. it is just one line that calls the macro that gets changed, added, or deleted |
| 17:51.42 | bjohnson | VoIPMasta: you should sign up att e164 if you intend to keep that enumlookup stuff |
| 17:53.00 | Zeeek | e164 accepts contributions in case anyone is interested btw |
| 17:53.24 | Zeeek | I fronted a couple of bucks because I think it's a cool idea |
| 17:54.07 | Madkiss | Is somebody from Germany here using Asterisk and Freenode? |
| 17:54.47 | *** join/#asterisk eipi (~eipi@OL128-44.fibertel.com.ar) |
| 17:55.09 | bjohnson | Zeeek: turns out you can fill in a white page listing on e164.org for each phone number you register. You can only browse them if you have a phone number registered |
| 17:55.13 | Madkiss | Or can somebody provide me with an extensions.conf-file that is ready for use with one provider? |
| 17:55.26 | shido6 | "/usr/src/asterisk/configs/*" |
| 17:55.29 | shido6 | look there |
| 17:55.29 | shido6 | first |
| 17:55.38 | Zeeek | IIRC you can also register blocks of 700 numbers |
| 17:56.02 | Madkiss | shido6: These files are _that overloaded that you can't understand just a single bit from it |
| 17:56.08 | Madkiss | If you are a starter, like I am. |
| 17:57.24 | zno | providers like voicepulse gives you an example, albeit a bare bones one |
| 17:58.00 | hmmhesays | Madkiss.... play around with it |
| 17:58.07 | Zeeek | another goo duse of FWD - getting it working with asterisk you'd have to make a dialplan for it |
| 17:58.30 | Zeeek | understand of the dialplan is key to doing anything with asterisk |
| 17:58.47 | pcm | Zeeek: i'd say it's the first step |
| 17:59.09 | Zeeek | no, the first step is knowing you can't live without asterisk |
| 17:59.15 | Zeeek | like alcoholism |
| 17:59.29 | Zeeek | *then* you need to study the dialplan :) |
| 17:59.30 | pcm | Zeeek: are you an asterisk addict ? |
| 17:59.40 | pcm | hehe; |
| 17:59.42 | Madkiss | Feb 8 18:58:36 WARNING[31397]: chan_sip.c:6811 handle_response: Forbidden - wrong password on authentication for INVITE to '"Martin Loschwitz" <sip:madkiss@freenet.de>;tag=as6ad6b7ca'-- although it's using exactly the same data it's using for registering with the server ... |
| 17:59.44 | Zeeek | I am an addict of everything I ever try and like |
| 17:59.57 | *** join/#asterisk Casper_UA (~casper@as-2-22.ar43-2x.kharkov.ukrtel.net) |
| 18:00.12 | VoIPMasta | do you think that FoIP will become a reality some day? |
| 18:00.27 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 18:00.28 | Casper_UA | hi |
| 18:00.33 | Zeeek | Madkiss do you have canreinvite set in sip.conf ? |
| 18:00.42 | Zeeek | Fuck over IP? |
| 18:00.49 | Dalion | whats FOIP ?\ |
| 18:00.50 | Zeeek | what is foip? |
| 18:00.51 | techie | that would be nice |
| 18:00.54 | Dalion | fod over ip |
| 18:00.57 | Dalion | food |
| 18:00.57 | VoIPMasta | Fax over IP |
| 18:00.59 | Zeeek | bjoIP |
| 18:01.06 | Zeeek | OSoIP |
| 18:01.06 | Madkiss | Zeeek: No |
| 18:01.19 | Zeeek | Madkiss are you behind NAT? |
| 18:01.20 | Dalion | why not $oIp |
| 18:01.20 | harryvv | zeek, can i see your rules for poptray? mine not working the way it should. |
| 18:01.25 | Madkiss | Zeeek: No. |
| 18:01.37 | Dalion | harryvv: what poptray |
| 18:01.41 | Zeeek | harryvv I never did any but I'm sure it works fine |
| 18:01.48 | Casper_UA | anybody knows how to set dtmf interdigit timeout when dialing out through fxo x100p? |
| 18:01.59 | harryvv | zeekm okay you only made that mailbox for vm only? |
| 18:02.05 | VoIPMasta | http://www.webopedia.com/TERM/F/FoIP.html |
| 18:02.08 | Zeeek | Casper I think it's on your phone, not * |
| 18:02.23 | pcm | Zeeek: he's talking about x100p! |
| 18:02.27 | Zeeek | harryvv I found the software for you but I don't use it |
| 18:02.34 | harryvv | ohh okay |
| 18:02.58 | Zeeek | Casper dialing on what phone? |
| 18:03.17 | pcm | Zeeek: asterisk is dialing over x100p hehe |
| 18:03.23 | Zeeek | the timeout won't be in asterisk |
| 18:03.26 | Madkiss | Zeeek: What's next? |
| 18:03.31 | harryvv | mmm well the rules dont have the option to ignore all other mail. So thats what I was looking for to msg me when a rule equals my asterisk hostname email address but ignore all others. |
| 18:03.36 | Casper_UA | Zeek: i use e.g. dial(zap/1/123) |
| 18:03.58 | Zeeek | and your dialing is too fast for the network? |
| 18:04.03 | Zeeek | PSTN ? |
| 18:04.06 | Casper_UA | Zeek: yes |
| 18:04.21 | Casper_UA | it gets confused |
| 18:04.26 | Zeeek | did you look at zapata.conf options? |
| 18:04.38 | Zeeek | asterisk can even dial using rotary dials! |
| 18:04.38 | Casper_UA | no appropriate opts there :( |
| 18:04.40 | harryvv | stagering the amount of mail I get from asterisk users mailing list ;) |
| 18:04.54 | Zeeek | harryvv get another address and deal with it! |
| 18:04.59 | harryvv | :) |
| 18:05.19 | Zeeek | Casper you don't mean the interdigit delay you mean the dialing speed |
| 18:05.26 | Casper_UA | Zeek: pulse dialing is done using on/off hooks, it's simple |
| 18:05.47 | Casper_UA | Zeek: i mean interdigit delay |
| 18:05.50 | tzanger | bjohnson: my neighbourhood? |
| 18:05.52 | Zeeek | anyway, if there's an option it would be in zapata.conf IMO - or maybe in the source |
| 18:06.01 | harryvv | Of course one idea is to setup a email voicemail server |
| 18:06.07 | Casper_UA | no opts :( |
| 18:06.10 | Zeeek | SOURCE |
| 18:07.45 | Zeeek | Casper what about switchtype? |
| 18:08.00 | Zeeek | where are you and what are you connected to? |
| 18:08.16 | Casper_UA | it's fxo |
| 18:08.23 | Zeeek | yes? and? |
| 18:08.36 | Zeeek | country? |
| 18:08.39 | Zeeek | telco? |
| 18:08.45 | Casper_UA | so no swithtype is appropriate there |
| 18:08.57 | Casper_UA | swithtype is for digital lines |
| 18:09.05 | Zeeek | ok |
| 18:09.14 | Zeeek | country? |
| 18:09.21 | Zeeek | why is the timing wrong? |
| 18:09.36 | Casper_UA | Zeek: thnx, i've got the answer... |
| 18:09.50 | Zeeek | good! |
| 18:09.57 | Zeeek | tell us what is was? |
| 18:10.19 | Casper_UA | SOURCE ;)ppp |
| 18:10.23 | tih | Zeeek: well, it turns out I was running .11 on the BT100 previously, and upgrading to .22 made it fail in the same way as yours. Downgrading back to .11 makes it work, with no config changes. |
| 18:10.29 | christo | well I sorted it in the end - with the latest tarballs of asterisk and asterisk-addons, rather than the cvs versions, everything was okay |
| 18:10.34 | christo | better go now |
| 18:10.35 | christo | bye |
| 18:10.44 | CCDAS | Asterisk CVS-v1-0-01/08/05-20:21:08 built by root@MSLS on a i686 running Linux |
| 18:10.45 | Zeeek | tih - you MUST tell me when you figure out why!!!! |
| 18:10.45 | tih | Zeeek: guess we've got BT100s from the same batch. :-) |
| 18:10.52 | *** part/#asterisk christo (~chris@office.enovi.com) |
| 18:10.55 | Zeeek | what the fsck would do this? |
| 18:11.28 | freat | hey there... is it possible to patch stable with the packet loss concealment stuff? |
| 18:11.42 | tih | Zeeek: yeah, I'll be experimenting with it, now I've got it happening. Just need a hub next to it, to be able to get a good tcpdump. |
| 18:11.49 | freat | I noticed there's a lot of work going on with it now... http://bugs.digium.com/bug_view_page.php?bug_id=0002532 |
| 18:11.59 | *** join/#asterisk hardwire (~hardwire@209.112.194.45) |
| 18:12.04 | freat | would sure love to get PLC working on ulaw |
| 18:12.11 | Zeeek | tih I did the dumps for Grandstream - I can see that there is NO DNS happening |
| 18:12.33 | Zeeek | I have a 101 - builtin hub - very handy |
| 18:12.48 | hmmhesays | http://local.google.com |
| 18:13.41 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 18:14.12 | vaewyn | ok... so where is docs on setting up Ts on *? (signalling etc...) I have mine set up but am trying to help a coworker that isn't using NI2 |
| 18:15.05 | Zeeek | tih I hope maybe you kept my email address? If not please do share anything you find out about the GS problem |
| 18:15.14 | Zeeek | I'll usually be here in the morning |
| 18:16.15 | tih | Yes, of course... |
| 18:16.23 | *** join/#asterisk p0lar (~p0lar@64.254.254.4) |
| 18:16.42 | Zeeek | in the meantime, the phone works just fine. It's just too bad not to be able to take advantage of the newer features |
| 18:17.08 | hardwire | good people of VoIP happyness |
| 18:17.11 | hardwire | how are you all today |
| 18:17.17 | Zeeek | we are well |
| 18:17.19 | Madkiss | erm, how do I realise STUN in asterisk? |
| 18:17.20 | Zeeek | and you? |
| 18:17.39 | pcm | hardwire: what's the problem ? |
| 18:17.40 | Zeeek | you don't - just use a STUN server for the client |
| 18:18.01 | ManxPower | Madkiss, very few people use STUN with Asterisk. Generally only very large installations have any need for STUN. |
| 18:18.08 | ta[i]nted | how do I get my own DID? |
| 18:18.12 | ta[i]nted | w/o using voip provider |
| 18:18.23 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 18:18.24 | ManxPower | ta[i]nted, You buy a PRI line from your local LEC |
| 18:18.27 | vaewyn | ta[i]nted: talk to your local LECs |
| 18:18.35 | tzafrir_laptop | hi folks |
| 18:18.47 | *** join/#asterisk sskyles (~sskyles@155.205.205.68.cfl.rr.com) |
| 18:19.44 | ta[i]nted | are DID's cheap? |
| 18:19.44 | Madkiss | Destroying call 'ECBF3A6C-1DD1-11B2-836B-E131BD7502B4@192.168.0.1' |
| 18:19.44 | Madkiss | <PROTECTED> |
| 18:19.47 | Madkiss | <PROTECTED> |
| 18:19.50 | Madkiss | I get this |
| 18:19.57 | vaewyn | ta[i]nted: depends how many you purchase |
| 18:20.15 | vaewyn | and what they come in on |
| 18:20.16 | Zeeek | Madkiss you have a very polite provider! |
| 18:20.24 | Luke-Jr | What's an LEC? |
| 18:20.27 | vaewyn | (ie frac T1 full T1...etc |
| 18:20.39 | Zeeek | Mine says "FOAD or get it right!" |
| 18:20.45 | freat | Local Exchange Carrier |
| 18:20.47 | ta[i]nted | vaewyn what if i wanted 10 over T1 |
| 18:20.55 | Madkiss | Zeeek: Something is terribly wrong. It should use exactly the same data as it does for registering with the registrar. |
| 18:20.57 | Luke-Jr | freat: Where do they get DIDs from? |
| 18:21.07 | Zeeek | Madkiss maybe not |
| 18:21.18 | ManxPower | Madkiss, you don't have codecs when registering. try disallow=all and allow=ulaw only |
| 18:21.24 | Zeeek | I have one provider that needs two different servers to be used |
| 18:21.29 | vaewyn | ta[i]nted: around here... about 400$/month... varies GREATLY on local though... we are on the middle->high side |
| 18:21.40 | *** join/#asterisk darkflib (darkflib@dialup090.ts008.bmt.esat.net) |
| 18:21.57 | vaewyn | locale even |
| 18:21.59 | darkflib | does nufone still accept paypal for credit? |
| 18:22.14 | Luke-Jr | darkflib: From what I hear, only for initial signup |
| 18:22.23 | darkflib | k... |
| 18:22.46 | Zeeek | then what? Beads? |
| 18:22.57 | Zeeek | jewels? |
| 18:23.02 | darkflib | no.. firewater... |
| 18:23.04 | vaewyn | credit card |
| 18:23.06 | Zeeek | he=aha |
| 18:23.13 | Zeeek | yes I remember now |
| 18:23.17 | Zeeek | VISA |
| 18:23.29 | vaewyn | any of the 4 majors |
| 18:23.39 | hardwire | pcm: no problems at all |
| 18:23.40 | hardwire | heh |
| 18:23.42 | darkflib | pity... I have money on my paypal account but not my visa card |
| 18:23.49 | Zeeek | my local supermarket suddenly started asking for ID when you pay by AmEx |
| 18:23.50 | hardwire | unless you include a mild rash of insanity. |
| 18:24.13 | Zeeek | turns out someone with a stolen card went in that same store last weekend 4 times in one day |
| 18:24.23 | harryvv | haahah |
| 18:24.38 | hardwire | couldn't find something that fitted them? |
| 18:24.39 | vaewyn | Zeeek: Walmart you have to type in your pin for mastercard now... debit or credit |
| 18:24.47 | PakiPenguin | Zeeek: sip allows listen in? |
| 18:24.47 | vaewyn | which is nice |
| 18:24.54 | Zeeek | got no PIN for AmEx unless you pay extra |
| 18:24.56 | Luke-Jr | PakiPenguin: yes |
| 18:25.04 | PakiPenguin | Luke-Jr : how? |
| 18:25.10 | Zeeek | you know AmEx you can literally buy a Rolls on that card? |
| 18:25.11 | Luke-Jr | PakiPenguin: sniff the packets? |
| 18:25.14 | PakiPenguin | Zeeek: my amex has pin pin:p |
| 18:25.16 | Zeeek | no lilmit |
| 18:25.23 | harryvv | well, if the owner of the card relized it was stolen would have reported it and the thief would have been caught. We had a case where it was ID theft and vendor called RCMP he waited for the guy to arive again then grabed him by the through and sqeeezed :) |
| 18:25.25 | vaewyn | *nods* that is what is scary |
| 18:25.57 | Zeeek | "shoulder surfing" in airports works well |
| 18:26.07 | Zeeek | so be careful! |
| 18:26.15 | harryvv | I know about that |
| 18:26.20 | PakiPenguin | anyone tried maps.google.com? |
| 18:26.23 | Zeeek | imagine checking your email on one of those free terminals |
| 18:26.27 | vaewyn | PakiPenguin: yep... it rocks |
| 18:26.35 | PakiPenguin | it sure does! |
| 18:26.40 | harryvv | zeek, some people in the world just dont have common sence and get there things stolen ;) |
| 18:26.46 | PakiPenguin | Zeeek: never , they are filled with keyloggers! |
| 18:26.49 | Zeeek | vaewyn gave aus a technology preview |
| 18:26.54 | Luke-Jr | Zeeek: Um... yay for passwords based on time of day? |
| 18:27.10 | PakiPenguin | Luke-Jr: maybe RSA digital key? |
| 18:27.16 | Zeeek | I used to use expiring addresses |
| 18:27.18 | Luke-Jr | PakiPenguin: RSA on a public terminal? |
| 18:27.22 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 18:27.38 | PakiPenguin | ah , here they have open terminals , you can install anything :p |
| 18:27.42 | Zeeek | the newest technology uses not only what you type but *how* you type it |
| 18:27.45 | Madkiss | I don't get it. I just don't get it. |
| 18:28.22 | Zeeek | if your watch is off yer screwed :) |
| 18:28.27 | Luke-Jr | Nah |
| 18:28.34 | Luke-Jr | Just have the time part work for 2 hours ea |
| 18:28.36 | Luke-Jr | and overlap one hour |
| 18:28.46 | Zeeek | shit in 2 hrs they'd already own your home |
| 18:28.56 | Luke-Jr | Fine |
| 18:28.59 | Luke-Jr | Include sequence # |
| 18:29.00 | Luke-Jr | =p |
| 18:29.37 | freat | Zeek, the point is you change the key every 2 hours so they only have 2 hours to try to crack your authentication, then the key changes |
| 18:29.46 | sskyles | That technology already exists and is old. I remember those little cards with the display changing every minute. |
| 18:29.51 | Nugget | Luke-Jr: I think PakiPenguin was talking about RSA's SecurID keychain dealie. |
| 18:29.52 | *** join/#asterisk planetWayne (~wayne@cpc1-lich2-5-0-cust62.brhm.cable.ntl.com) |
| 18:29.55 | freat | RSA works like this, but changes the key very frequently |
| 18:30.03 | Nugget | those update the key every 60 seconds |
| 18:30.10 | freat | yep |
| 18:30.30 | vaewyn | any physical form of security... cards, biometrics, rfid.. etc... can be foiled... the only truly unbeatable system is the individuals brain and what it holds... then you at least have to torture them to get in as them :P |
| 18:30.38 | Luke-Jr | Well, I could probably do just about anything in theory... just have my handheld computer calculate whatever |
| 18:30.47 | sskyles | I think they even called them smart cards. Or maybe they doubled as smart cards, I don;t remember. |
| 18:30.55 | Luke-Jr | vaewyn: combine both |
| 18:30.59 | freat | vaewyn: combine physical (something you have) with something you know |
| 18:31.00 | Zeeek | you'd need a brain for that to work |
| 18:31.09 | hmmhesays | anyone compiled a 2.6 kernel lately? |
| 18:31.15 | Luke-Jr | hmmhesays: sure |
| 18:31.16 | planetWayne | hiyall |
| 18:31.18 | Zeeek | that's why we use the what you tyope and how you type it |
| 18:31.18 | freat | those USB fingerprint dongles are cool |
| 18:31.25 | Nugget | Luke-Jr: http://www.rsasecurity.com/node.asp?id=1157 |
| 18:31.30 | Luke-Jr | freat: fingerprint for password is very very bad |
| 18:31.30 | vaewyn | Luke-Jr: freat: correct... |
| 18:31.33 | Zeeek | none of my passwords will work unless dyslexia is applied :) |
| 18:31.37 | hmmhesays | i can't find where the hell to disable the gamma drivers |
| 18:31.39 | hmmhesays | *driver |
| 18:31.42 | vaewyn | freat: and can be defeated with a gummy bear |
| 18:31.45 | freat | nahh you have to use combo of fingerprint PLUS a password |
| 18:31.47 | Luke-Jr | hmmhesays: gamma? |
| 18:31.53 | harryvv | vaewyn, rfid may be harder to crack then you know of. I regularly enter the US with nexas card and it has rfid in it. While everyone else waits up to 45 min at the border mine is less then 5 min at nexas ;) |
| 18:31.58 | hmmhesays | yeah, it seems to be broken |
| 18:32.01 | Nugget | biometrics are inappropriate for passwords because they aren't secret and they can't be revoked. |
| 18:32.02 | hmmhesays | i'm using xconfig |
| 18:32.03 | Luke-Jr | freat: Why not all 3? =p |
| 18:32.07 | freat | so you physically must have the dongle, and still you must know the password |
| 18:32.22 | Luke-Jr | freat: anyone can buy a dongle |
| 18:32.31 | vaewyn | harryvv: they are defeatable... trust me |
| 18:32.38 | Luke-Jr | freat: read the fingerprint, ask a password, and have a prog calculate the key from those |
| 18:32.39 | freat | Luke-Jr: the dongle has encrypted info on it |
| 18:32.51 | harryvv | vaewyn, from open souce software? |
| 18:33.10 | vaewyn | harryvv: without software at all... ;P |
| 18:33.36 | Nugget | the other problem with using fingerprints is that you have, at most, only ten of them. I have a lot more than ten passwords. |
| 18:33.45 | harryvv | btw, the inventor of the rfid chip in the early 70s was a hamradio operator who is getting almost nothing for his invention. |
| 18:33.47 | freat | Luke-Jr: you have to set them up. the usb dongle is protected by fingerprint + password, then it contains the authentication info for your system once authenticated |
| 18:34.09 | Nugget | Do you really want the "key" you use to buy groceries to be the same key as the one you use to start your car or open your front door? |
| 18:34.20 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 18:34.22 | harryvv | scarry |
| 18:34.23 | harryvv | :) |
| 18:34.36 | vaewyn | none of your keys should ever be the same... if a site can verify it they can replicate it |
| 18:34.39 | freat | Nugget: no... I want my own RFID reader so I can run my house with it |
| 18:34.41 | Nugget | exactly |
| 18:34.52 | _Brian | Has anyone been able to get Directed pickup working as described in http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup |
| 18:35.10 | harryvv | even though the car idea is not new been around for a while. One store uses biometrics to allow a shopper to purchace groshries in seattle. |
| 18:35.13 | freat | Nugget: have my own RFID card in my wallet. walk up to the front door... it opens (pending retinal scan) |
| 18:35.14 | vaewyn | freat: about 20$ to build a close range one... about 200$ to buy one |
| 18:35.21 | _Brian | and this would be by specifying an extensions to pick up the call, not just putting all phones into the same group |
| 18:35.33 | Luke-Jr | Is there any software to read a fingerprint from a touchscreen? |
| 18:35.38 | freat | the retinal scan thing seems kinda scary to build yourself tho... |
| 18:35.59 | vaewyn | Luke-Jr: nope... touchscreens are the wrong technolgy.... not near high enough resolution on individual points |
| 18:36.09 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 18:36.15 | vaewyn | freat: was talking rfid :} |
| 18:36.23 | Luke-Jr | vaewyn: it certainly seems to be detailed resolution o.o |
| 18:36.24 | harryvv | luke, its not software limites its material limited. it cannot be done..as of yet. |
| 18:36.52 | Nugget | http://www.schneier.com/crypto-gram-9808.html#biometrics |
| 18:37.03 | vaewyn | Luke-Jr: of single point only though... it can't distinguish between 2 pencils touching it .5cm from each other |
| 18:37.24 | Luke-Jr | vaewyn: I think it can... |
| 18:37.25 | vaewyn | and you need WAY better resolution to read fingerprint |
| 18:38.08 | vaewyn | The best touchpads/screens out there now are just starting to tell when 2 individual fingers are touching versus 1 finger |
| 18:38.14 | freat | you can buy fingerprint readers pretty cheap... the ones I looked into though, the dev kit was like $2000 |
| 18:39.00 | freat | I wanted to make a biometric access control for a beer fridge... oh wlel |
| 18:39.02 | hardwire | finger finger finger |
| 18:39.14 | hardwire | freat: maybe you should just put a breathalizer access on it |
| 18:39.23 | freat | hmmm |
| 18:39.59 | harryvv | 2k thats all? |
| 18:40.01 | *** join/#asterisk zetho (~zetho___@dkj211.neoplus.adsl.tpnet.pl) |
| 18:40.19 | vaewyn | The absolute best security that actually works is still just a mag/smartcard and a 6+digit pin |
| 18:40.22 | harryvv | I guess its important for access security :) |
| 18:40.35 | vaewyn | (usb dongle also) |
| 18:40.38 | JerJer | palm scan + pin |
| 18:41.15 | vaewyn | nope... biometrics is still not valid |
| 18:41.20 | harryvv | mike, what do the lines on jerjers hand say? |
| 18:41.22 | mikegrb | we had retinal scanners + smart card + pin at my last place of employment |
| 18:41.26 | vaewyn | "I sliced my hand open" oops |
| 18:41.46 | mikegrb | harryvv: he has a very long wealth line developing in the future |
| 18:41.58 | mikegrb | harryvv: I think nufone will do well from his palm |
| 18:42.03 | vaewyn | changing your contacts throws off most retinal scanners and you have to reprogram them |
| 18:42.07 | harryvv | :) |
| 18:42.18 | zetho | please could you point me a good HOWTO for asterisk to set up SIP proxy ? |
| 18:42.37 | *** join/#asterisk Bile_One (~TomSawyer@pcp03281999pcs.gillst01.ar.comcast.net) |
| 18:42.40 | hardwire | apt-get isntall siproxd |
| 18:42.58 | vaewyn | Give your people something that tells you who they should be.. (mag/smart card, usb key, rfid) and then make them prove it with a 6+digit pin |
| 18:43.02 | harryvv | obviosly the one I like the most is when the front attendent knows me by face ;) |
| 18:43.05 | zetho | no plz. although i am using debian, i don't want siproxd |
| 18:43.26 | zetho | any good asterisk howto would be appreciated |
| 18:43.30 | Bile_One | Bprice20, you here today? |
| 18:43.40 | harryvv | www.voip-info.org |
| 18:44.09 | zetho | is that all ? |
| 18:44.35 | mikegrb | zetho: asterisk doesn't work as a sip proxy |
| 18:44.41 | Madkiss | ah! I got that problem fixed for now. Right, then the next question that is unanswered is -- why the hell do phones calling my VoIP only hear "peep-peep-peep"? |
| 18:44.43 | Bile_One | Anyone use call files and have problems with the messages getting played before the call is answered by the intended user? |
| 18:45.21 | vaewyn | Bile_One: your on PRI and not FXO correct? |
| 18:45.26 | Corydon-w | Nope. Are you using VOIP channels? |
| 18:45.46 | Luke-Jr | Madkiss: you're ahead of me now ;) |
| 18:45.49 | hardwire | it would be suwheat if asterisk supported sip video |
| 18:45.56 | Bile_One | Vaewyn, SIP calls are working fine, it is when they go out on a ZAP channel it happens. |
| 18:46.04 | zetho | I see mikegrb... so i want to plug asterisk to my MultiVoip 3010, and what is the SIP synonime for h323 gatekeeper (that i was using for some time now) ? |
| 18:46.10 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 18:46.11 | Madkiss | Luke-Jr: ;P |
| 18:46.15 | vaewyn | Bile_One: is the ZAP a T interface.. or analog? |
| 18:46.23 | Bile_One | Analog X100P |
| 18:46.40 | vaewyn | Bile_One: that is why... call progress on FXO is a crapshoot at best |
| 18:46.46 | mikegrb | zetho: I don't know h323 so I don't know exactly the role of gatekeeper |
| 18:47.02 | vaewyn | wb kram |
| 18:47.06 | Bile_One | From the CLI I see that asterisk says the call is answered and begins the play. |
| 18:47.16 | zetho | oh.. strange mikegrb ;] i thought it's one tousand times easier that SIP :) |
| 18:47.37 | Corydon-w | Yep, X100P always sees connection when there's an available channel. |
| 18:47.52 | zetho | everything worked fine until I've plugged Cisco Callmanager to my MultiVoip3010... |
| 18:47.53 | Corydon-w | That's the problem with analog channels |
| 18:48.11 | vaewyn | zetho: Crisco call mangler! no wonder :P |
| 18:48.16 | Bile_One | What is the best way to forecast around this problem then? |
| 18:48.20 | vaewyn | analog bites for call progress |
| 18:48.39 | Corydon-w | There's no best... just various compromises |
| 18:48.41 | zetho | u c.. CM works fine for.... (let's guess).... CISCO IP phones:).. but no other |
| 18:49.05 | vaewyn | zetho: doesn't even work for those after certain sizes of installs |
| 18:49.07 | darkflib | to dial an international number from within the us is a 011 prefix isn't it? |
| 18:49.16 | Bile_One | Vaewyn, what happens when I have my sip exten forwarded to another sip, does asterisk auto assume an answer on the forward? |
| 18:49.36 | *** join/#asterisk Tili (~Tili@202-133-65-23-dialup.sat.net.pk) |
| 18:49.37 | zetho | hmm.. have you tried Cisco 1770 Callmanager ? it's simple and works just fine for me |
| 18:49.40 | vaewyn | Bile_One: SIP it knows exactly when answer takes place... analog it has to guess... a lot |
| 18:49.57 | santiago | Hi, what is zttest and ztspeed for? |
| 18:50.13 | Corydon-w | Even with SIP sometimes the first few seconds of audio get cut off |
| 18:50.18 | vaewyn | zetho: yes... and we broke it... just like every other one the cisco reps keep trying to sell us... CCM should NEVER be used in medium->large installs |
| 18:50.50 | Bile_One | Corydon-w, it hasn't happened yet to me. |
| 18:51.08 | Corydon-w | Bile_One: there's also some very alpha quality call-progress detection code in zapata that you can turn on |
| 18:51.24 | Corydon-w | It's not very reliable, but it may work for your purposes |
| 18:51.27 | zetho | you're right vaewyn |
| 18:51.29 | Bile_One | Corydon-w, where is the info located? |
| 18:51.47 | Corydon-w | Bile_One: See configs/zapata.conf.sample |
| 18:51.56 | Bile_One | Okay |
| 18:52.32 | Corydon-w | Bile_One: It's called callprogress |
| 18:52.44 | zetho | I;ve got one more question. I've got Multivoip (an E1[or T1 in US] to IP converter) and want to set up something like SIP gatekeeper ... is that possible on asterisk ? |
| 18:52.45 | Bile_One | Corydon-w, On the Wiki? |
| 18:52.54 | Corydon-w | Bile_One: no, on your install |
| 18:52.58 | vaewyn | callprogress does help... still consider it a crapshoot |
| 18:53.20 | Corydon-w | JunK-Y: what? |
| 18:53.33 | JunK-Y | Corydon-w: ya got my mail last week? |
| 18:53.41 | JunK-Y | about the bchan restart |
| 18:53.47 | Corydon-w | I guess |
| 18:53.48 | *** join/#asterisk festr_ (~festr@ns.regnet.cz) |
| 18:53.49 | festr_ | hello |
| 18:53.51 | Bile_One | Vaewyn, okay, it is just for call outs using a call file. |
| 18:53.58 | festr_ | there is bug in new commit to sip_chan |
| 18:54.02 | festr_ | stable |
| 18:54.05 | JunK-Y | it seeems to work fine if ur remote end locks up some channels |
| 18:54.13 | festr_ | but i cannot reopen bug in mantis |
| 18:54.20 | JunK-Y | if ya do a pri restart, its hangups all the actives channels too. |
| 18:54.20 | festr_ | even i'm logged as regular user |
| 18:54.23 | Corydon-w | festr_: so open a new bug |
| 18:54.35 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
| 18:55.10 | Corydon-w | JunK-Y: yeah, we had problems of all channels locking up, so that wasn't an issue |
| 18:55.32 | Corydon-w | Those issues seem to have largely resolved themselves (knock on wood) |
| 18:55.51 | pcm | yunki: that's why there is an optino to delay it .... |
| 18:56.01 | pcm | junky: or turn it off (maybe) |
| 18:56.01 | JunK-Y | dont ya want the ability to restart only a bhan? |
| 18:56.06 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 18:56.13 | JunK-Y | pcm: turn off what? |
| 18:56.19 | pcm | junky: restarts |
| 18:56.23 | *** join/#asterisk toddf (k62q75tw4t@default.fries.net) |
| 18:56.30 | JunK-Y | pcm: i dont get ur point, sorry. |
| 18:57.16 | Corydon-w | pcm: it's a patchset that allows a forced restart |
| 18:57.20 | pcm | junky typed: "if ya do a pri restart, its hangups all the actives channels too." |
| 18:57.47 | JunK-Y | so whats ur solution exactly ??? |
| 18:58.04 | JunK-Y | i do a pri restart bchan <span> <chan_num> |
| 18:58.28 | Corydon-w | pcm: the problem is, some channels get locked up, and other channels are being used at the time |
| 18:58.34 | pcm | junky: ;resetinterval = 3600 |
| 18:58.45 | pcm | junky in zapata.conf |
| 18:58.56 | Corydon-w | pcm: I think you misunderstand the discussion |
| 18:59.08 | pcm | corydon: so you want to restart only idle channels (from asterisk point of view) ? |
| 18:59.09 | *** join/#asterisk jterrero (~some@66.28.34.162) |
| 18:59.11 | Corydon-w | pcm: the point is that we WANT a restart |
| 18:59.34 | Corydon-w | pcm: not that we want to DELAY a restart |
| 18:59.44 | jterrero | anyone ever get an sip_xmit error (192.168.10.10 returned -1: Bad file descriptio) |
| 19:00.06 | pcm | corydon: so you want to restart only idle channels (from asterisk point of view) ? |
| 19:00.21 | Corydon-w | pcm: yeah, and there's a patch in the bugtracker for that |
| 19:00.37 | Bile_One | Corydon-w, when making a change to zapata.conf does Asterisk require a reboot of the server? |
| 19:00.37 | JunK-Y | only a channel |
| 19:00.39 | pcm | corydon: I see |
| 19:00.43 | JunK-Y | cory: i didnt submit it yet. |
| 19:00.51 | JunK-Y | Bile_One: yes. |
| 19:00.56 | redder86 | Bile_One: not a reboot, but asterisk needs to be restarted |
| 19:00.58 | Bile_One | Thanks Juggie |
| 19:00.59 | Corydon-w | Bile_One: not a restart of the server, just a restart of the Asterisk process |
| 19:01.27 | JunK-Y | i hope it's what he means. (server probably means * server) |
| 19:01.42 | Corydon-w | Restart the daemon, not the machine... |
| 19:01.51 | JunK-Y | exactly. |
| 19:01.58 | Corydon-w | "server" is unclear |
| 19:02.10 | JunK-Y | Corydon-w: gonna submit it soon. |
| 19:02.20 | Madkiss | handle_request: Unable to create/find channel |
| 19:02.21 | Madkiss | uhm |
| 19:02.21 | tessier_ | http://pastebin.ca/5482 |
| 19:02.25 | tessier_ | Anyone know what could cause that? |
| 19:02.25 | Madkiss | What's that? |
| 19:02.29 | JunK-Y | i'll re-open ur bug to submit my patch, all right? |
| 19:02.39 | Corydon-w | Sounds fine to me |
| 19:02.40 | tessier_ | I dial one of my DID's and a zillion calls get opened to my SIP gateway. |
| 19:02.44 | pcm | corydon: the patch should be just one line for the stuff unless you add some config stuff |
| 19:02.44 | tessier_ | I don't see an calls coming back in. |
| 19:02.48 | tessier_ | Very weird. |
| 19:03.05 | santiago | does anyone knows what zttest and ztspeed are for? |
| 19:03.05 | Corydon-w | pcm: WTF are you talking about? |
| 19:03.16 | JunK-Y | Corydon-w: if ya've the chance to test it, it would be appreciated too. |
| 19:03.29 | Corydon-w | JunK-Y: my channels haven't locked up recently |
| 19:03.41 | *** join/#asterisk zainasterisk (~zain@202.147.160.245) |
| 19:03.48 | pcm | santiago: they're for finding out how fast is your machine (in zt*** score) |
| 19:04.12 | jterrero | anyone ever get an sip_xmit error (192.168.10.10 returned -1: Bad file descriptio) |
| 19:04.35 | Corydon-w | JunK-Y: pcm is in Huntsville; he's obviously on crack... ;-) |
| 19:04.44 | santiago | ./ztspeed |
| 19:04.45 | santiago | Count: 1426887 |
| 19:05.10 | pcm | Corydon: well you must be right cause you're corydon ... heh |
| 19:05.28 | Corydon-w | That's right. ;-) |
| 19:05.33 | zainasterisk | does context in zaptel.conf is used for only outgoing calls from ZAp Channel? |
| 19:05.39 | jterrero | how can i find earlier versions of asterisk ? |
| 19:05.51 | Nugget | jterrero: cvs |
| 19:07.29 | jterrero | anyone ever get an sip_xmit error (192.168.10.10 returned -1: Bad file descriptio) |
| 19:07.31 | santiago | jeje |
| 19:07.33 | zainasterisk | Does [context] in zaptel.conf is used for only outgoing calls from ZAp Channel? |
| 19:07.49 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 19:08.03 | ctooley | Is someone from TX Link here by any chance? Please PM me. |
| 19:08.05 | pcm | santiago: so that when you overload it and the number drops to < 1000 then it's bad |
| 19:08.21 | oej | ~seen anthm |
| 19:08.26 | jbot | anthm is currently on #asterisk (3h 35m 15s) |
| 19:08.28 | pcm | zainasterisk: only for incoming zap calls ... |
| 19:09.05 | zainasterisk | pcm: u mean context in zaptel.conf is primarily used for INCOMMING calls???? |
| 19:09.26 | vaewyn | zainasterisk: only used for incoming |
| 19:09.27 | Corydon-w | zainasterisk: It's used for both |
| 19:09.39 | zainasterisk | because i m using FXO Line |
| 19:09.54 | Corydon-w | It's the initial context for both types of channels |
| 19:09.55 | pcm | corydon: right, for both , hehe |
| 19:09.59 | zainasterisk | right |
| 19:10.00 | tzafrir_laptop | zainasterisk, context? in /etc/zaptel.conf? |
| 19:10.05 | zainasterisk | right |
| 19:10.06 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 19:10.07 | santiago | pcm, thanks, and zttest? |
| 19:10.25 | santiago | ./zttest |
| 19:10.26 | santiago | Opened pseudo zap interface, measuring accuracy... |
| 19:10.26 | santiago | 99.975586% 99.975586% ... |
| 19:10.27 | zainasterisk | infact I want my Zap Channel (FXO) to answer calls |
| 19:10.33 | Corydon-w | Oh, and context is only in /etc/asterisk/zapata.conf, not /etc/zaptel.conf |
| 19:10.33 | santiago | --- Results after 130 passes --- |
| 19:10.34 | santiago | Best: 100.000000 -- Worst: 81.079102 |
| 19:10.48 | pcm | zttest is the same but it tries to open the zap devices |
| 19:10.52 | zainasterisk | but it seems like it is not picking up the calls |
| 19:11.02 | Corydon-w | zainasterisk: then you need exten => s in whatever context you've placed the FXO channel into |
| 19:11.07 | firestrm | anyone here used REL3 for asterisk? |
| 19:11.16 | zainasterisk | Corydon-w: Did it already! |
| 19:11.36 | santiago | pcm, ok, thanks |
| 19:11.49 | JunK-Y | Corydon-w: i need to makes HTTP request in an app, i'll can put in prod ur app_curl. |
| 19:12.05 | firestrm | the solaris experiment failed btw.. asterisk compiled fine.. zaptel drivers are a problem |
| 19:12.06 | PakiPenguin | can * cdr record the ip address of the calling device? |
| 19:12.15 | Corydon-w | JunK-Y: try that again, in English this time |
| 19:12.20 | tzafrir_laptop | zainasterisk, in the CLI run 'show zap channel NNN' to make sure you got the right context |
| 19:12.26 | godsmoke | anyone in here knowledgeable about CPN and Caller ID in Asterisk IAX peering, and how everything is handled from the PSTN network? |
| 19:12.34 | ctooley | Is anyone using TXLink termination or origination services here? |
| 19:12.50 | zainasterisk | tzafrir_laptop: it is showing everything |
| 19:12.54 | zainasterisk | perfectly! |
| 19:12.54 | Corydon-w | tzafrir_laptop: flip show and zap... i.e. zap show channel NNN |
| 19:13.13 | tzafrir_laptop | right |
| 19:13.27 | zainasterisk | signalling=fxs_ks |
| 19:13.27 | zainasterisk | language=en |
| 19:13.27 | zainasterisk | context=sales |
| 19:13.27 | zainasterisk | channel => 4 |
| 19:13.44 | zainasterisk | under extensions.conf |
| 19:13.44 | zainasterisk | [sales] |
| 19:13.44 | zainasterisk | exten => s, 1, Answer |
| 19:13.44 | zainasterisk | exten => s, 2, AGI(outbox.agi) |
| 19:13.55 | tzafrir_laptop | zainasterisk, use pastebin |
| 19:13.58 | tzafrir_laptop | ~pastebin |
| 19:13.59 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 19:14.24 | zainasterisk | back to the point |
| 19:14.45 | tzafrir_laptop | zainasterisk, IIRC there are no contexts in zapata.conf itself |
| 19:14.45 | zainasterisk | under extensions.conf is there any mistake? |
| 19:15.21 | p0lar | ~beer |
| 19:15.22 | jbot | well, beer is ummm, ummm good!, or good for you! |
| 19:15.31 | zainasterisk | tzafrir_laptop: right, I knew |
| 19:15.47 | Madkiss | How do I allow any SIP traffic in? |
| 19:16.46 | tzafrir_laptop | zainasterisk, so next step would be to do some tracing. |
| 19:16.54 | tzafrir_laptop | set verbose 3 |
| 19:16.59 | tzafrir_laptop | set debug |
| 19:17.08 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
| 19:17.21 | tzafrir_laptop | and see what happens in the CLI when you call |
| 19:17.55 | sezuan | Hi! |
| 19:17.56 | WifiFred | Hi! |
| 19:18.40 | zainasterisk | tzafrir_laptop: console shows that, it forwarded the call to the line on Zap Channel and it is answering |
| 19:18.55 | zainasterisk | but the bell rings normally, rather than executing the AGI |
| 19:19.17 | zainasterisk | it seems that, if it is not going through CONTEXT in first place |
| 19:19.28 | Bile_One | Corydon-w, I tried it, and it failed terribly. How is this done when using like a T-1 and placing the calls fromt the .call file? |
| 19:20.24 | Corydon-w | Bile_One: like we said, it's a terrible hack. You're just going to have to play around with it to get it to work the way you want |
| 19:20.40 | sezuan | Can asterisk register on pstn gateway services and redirect forwarded calls to that service with some regex magic? I tried to do that with partysip, but it can't authenticate. |
| 19:20.58 | Bile_One | Corydon-w I can't believe no one has tried this before? |
| 19:21.43 | Corydon-w | Bile_One: many of us who have tried it are happier on T1 circuits |
| 19:22.24 | zainasterisk | OK, how to assign an extension to Zap_Channel? |
| 19:22.37 | Bile_One | Corydon-w, Can you describe how it work via the T-1 circuits, please? |
| 19:23.22 | Corydon-w | There's no need to use callprogress on digital circuits. Call Progress is already builtin. |
| 19:23.30 | Bile_One | Zainasterisk, zap channel is used it isn't an extension |
| 19:23.45 | Corydon-w | Bile_One: callprogress is a hack for analog channels |
| 19:24.34 | zainasterisk | Bile_One: I know, but even if I want to assign how can I??? |
| 19:24.45 | Bile_One | Corydon-w Okay so if I have a T-1 I wouldn't be having this trouble right since the PSTN would be handling the answer, etc..? |
| 19:25.14 | Bile_One | Zainasterisk, do it as a varible, and rename it. |
| 19:25.39 | *** part/#asterisk darkflib (darkflib@dialup090.ts008.bmt.esat.net) |
| 19:25.42 | zainasterisk | ok |
| 19:26.43 | zainasterisk | how can we identify a [context] to handle "incomming calls" under extensions.conf??? |
| 19:26.43 | Bile_One | Zainasterisk, it is a lame answer I know. |
| 19:26.43 | zainasterisk | Bile_One: its ok |
| 19:26.52 | Bile_One | You add the context [incoming-calls] |
| 19:27.12 | zainasterisk | nope |
| 19:27.41 | Bile_One | Well start there. |
| 19:27.44 | zainasterisk | does this text differs? I guess all the context are nothing more than tags, rather than commond |
| 19:28.25 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 19:29.04 | Bile_One | Zainasterisk, contexts are very powerful. You can call one from within another, and veven from within other files. So the are actually objects. |
| 19:29.07 | buddah | is there something i have to setup to get 4 digit dialing from sip phone to sip phone? |
| 19:29.20 | Nugget | buddah: yes. |
| 19:30.16 | buddah | ok |
| 19:30.23 | buddah | Nugget: what might it be? |
| 19:30.35 | buddah | Nugget: i can show the extensions/sip entries for the phone if needed |
| 19:30.41 | Nugget | buddah: you need to set up 4 digit dialing. |
| 19:30.47 | buddah | Nugget: how? |
| 19:30.50 | Mother_ | bwaaahahaha |
| 19:30.54 | Nugget | learn how to configure your dialplan. |
| 19:31.09 | Nugget | you've reached the point in asterisk where it's time to actually read the documentation. |
| 19:31.49 | buddah | wiki the best source? |
| 19:31.55 | Nugget | your question can be rephrased as "why can't you guys just tell me what to type?" |
| 19:32.10 | buddah | no, RTFM was a good enough answer |
| 19:32.20 | Nugget | search the wiki for "dialplan" |
| 19:32.21 | buddah | the main part is that there is somthing i need to set up |
| 19:32.24 | buddah | k |
| 19:32.26 | buddah | thx |
| 19:33.46 | zainasterisk | :D |
| 19:36.50 | *** join/#asterisk DaveCanoe (~Dave@H6.C30.B96.tor.eicat.ca) |
| 19:37.24 | zainasterisk | Nugget: I want Asterisk to answer my calls on my telephone number, I have attached the telephone line with Zap FXO Channel |
| 19:37.41 | zainasterisk | when I call to my telephone number asterisk is doing nothing |
| 19:37.47 | zainasterisk | can you put some light? |
| 19:38.18 | Nugget | the key is the default context you have defined in zap.conf and what's in that context. |
| 19:38.25 | *** part/#asterisk zetho (~zetho___@dkj211.neoplus.adsl.tpnet.pl) |
| 19:38.27 | ta[i]nted | what type of latency (ping times) should i expect from my voip providers to have good sound quality? |
| 19:38.56 | Luke-Jr | Can someone call my FWD #? |
| 19:38.59 | [Sim] | under 100ms is nice :) |
| 19:39.10 | zainasterisk | :D |
| 19:39.35 | zainasterisk | Nugget: oki doki |
| 19:39.39 | ta[i]nted | Luke-Jr what is it |
| 19:39.41 | Luke-Jr | 521542 |
| 19:41.12 | zainasterisk | Nugget: I have put the Dialplan under default, but no help :( |
| 19:41.18 | tessier_ | grr |
| 19:41.22 | tessier_ | Someone sent me flowers. At work. |
| 19:41.33 | Madkiss | This is a real, real strange effect. If I log in via kphone directly to freenet, I can call the number, if I log in via asterisk, I can't call the corresponding number |
| 19:41.34 | tessier_ | I appreciate the thought and all but I kinda wish chicks wouldn't do that sometimes. |
| 19:42.00 | Nugget | zainasterisk: now on to the second half of my comment. |
| 19:42.19 | Luke-Jr | Madkiss: disallow=all allow=ulaw ? |
| 19:42.27 | Mother_ | zainasterisk: read http://www.loligo.com/asterisk/current/ you have context examples there |
| 19:42.29 | Nugget | just putting it "in default" doesn't really address the issue. |
| 19:42.56 | zainasterisk | Nugget: I am just asking in dialplan to execute an AGI script :( |
| 19:43.03 | Nugget | this isn't a matter of changing some setting. you just need to sit down and program asterisk to do what you want it to do, whatever that happens to be. |
| 19:43.04 | zainasterisk | Mother_: thanks |
| 19:43.24 | ManxPower | Luke-Jr, I already told Madkiss that, but I don't think he listened. |
| 19:43.31 | Mother_ | make sure you understand the relationship between context in zapata.conf and the other conf such as SIP etc. |
| 19:43.40 | Luke-Jr | ManxPower: oh |
| 19:43.50 | Madkiss | ManxPower, Luke-Jr: I have this in the General-section: |
| 19:43.56 | zainasterisk | Mother_: thank you |
| 19:44.03 | Madkiss | disallow=all -- |
| 19:44.04 | Madkiss | allow=gsm |
| 19:44.04 | Madkiss | allow=ilbc |
| 19:44.04 | Madkiss | allow=ulaw |
| 19:44.28 | ManxPower | Madkiss, Well that is |
| 19:44.28 | Luke-Jr | Madkiss: that goes under the SIP connection |
| 19:44.28 | ManxPower | Madkiss, Well that is NOT disallow=all and allow=ulaw only, now is it? |
| 19:44.32 | Luke-Jr | Madkiss: and only have ulaw |
| 19:45.01 | Luke-Jr | Madkiss: When unsure, drop everything but ulaw :) |
| 19:45.08 | ManxPower | Madkiss, shido is very good at asterisk, but sometimes overestimates users. |
| 19:45.29 | ta[i]nted | lol |
| 19:45.49 | Madkiss | Feb 8 20:45:37 NOTICE[14797]: chan_sip.c:7399 handle_request: Unable to create/find channel |
| 19:46.32 | ta[i]nted | Madkiss put the keyboard down and step away from the asterisk box |
| 19:47.04 | *** join/#asterisk jcollie (~jcollie@161.210.6.51) |
| 19:47.12 | ta[i]nted | Madkiss post your extensions.conf and sip.conf |
| 19:47.16 | jcollie | hello all |
| 19:47.21 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 19:47.23 | Madkiss | ta[i]nted: right, wait |
| 19:47.45 | ManxPower | POST THEM TO PASTEBIN.CA NOT THE CHANNEL! |
| 19:47.45 | markit | anyone able to reach this url? I was yesterday, but not today... http://ipphone.eezeephone.com/index_files/Page700.htm |
| 19:48.21 | markit | it's a 4 port ata... would like to know if it work with asterisk... |
| 19:48.35 | *** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net) |
| 19:48.53 | Derkommissar | can someone take a look at these sip messages and tell me whats wrong |
| 19:49.03 | Derkommissar | why cant i send a call from one asterisk box to an other |
| 19:49.11 | Derkommissar | http://www.pastebin.com/239285 |
| 19:49.25 | Luke-Jr | Can someone call my FWD #? 521542 |
| 19:49.36 | jcollie | markit: i get dns timeout |
| 19:50.06 | jcollie | markit: looks the their one and only DNS server is down |
| 19:50.06 | ManxPower | Derkommissar, do you have a G729 license? |
| 19:50.51 | Madkiss | http://pastebin.ca/5488 |
| 19:51.02 | markit | jcollie: ok, thanks a lot |
| 19:51.03 | Madkiss | ManxPower: I would not have posted them to channel for sure ... |
| 19:51.11 | Madkiss | ta[i]nted: There it is ... |
| 19:51.38 | Luke-Jr | Madkiss: You should generally remove passwords when you post things |
| 19:51.50 | Madkiss | erm, I did. |
| 19:52.03 | Luke-Jr | ah, so 'nuts' isn't your password... XD |
| 19:52.09 | Madkiss | I hope you don't think I use "nuts" as my standardpassword ... |
| 19:52.15 | Derkommissar | yes |
| 19:52.17 | Luke-Jr | LOL |
| 19:52.23 | Luke-Jr | Well, I would hope not, but... |
| 19:52.25 | Luke-Jr | you never know |
| 19:53.02 | ManxPower | Madkiss, You've not be around here for very long, have you? |
| 19:53.02 | Luke-Jr | Madkiss: Under [freenet-out] put disallow=all allow=ulaw |
| 19:53.14 | ManxPower | Derkommissar, looks to me like a password/username issue, but I'm not an expert. |
| 19:53.22 | Madkiss | ManxPower: Not in this channel, no |
| 19:53.25 | PakiPenguin | Luke-Jr: your number's busy |
| 19:54.01 | Luke-Jr | PakiPenguin: hrm |
| 19:54.03 | Luke-Jr | Feb 8 19:53:22 [asterisk] NOTICE[11588]: chan_sip.c:7271 in handle_request: Failed to authenticate user "BNC - Systems" <sip:272713@65.39.205.121>;tag=as38acd2e0_ |
| 19:54.06 | Luke-Jr | is that it, I wonder? |
| 19:54.08 | zainasterisk | maa chudai |
| 19:54.30 | *** join/#asterisk VoIPster (~VoIPster@pD9ECB1FD.dip.t-dialin.net) |
| 19:54.30 | PakiPenguin | yes Luke-Jr |
| 19:54.31 | Madkiss | ManxPower: I exactly started today, but actually, some of the howtos and manuals you see have paragraphs that clearly conflict with each other. |
| 19:54.47 | *** join/#asterisk Alric (~nbowyer@masq.hyperusa.com) |
| 19:55.25 | Madkiss | Luke-Jr: Still the same problem |
| 19:55.28 | Luke-Jr | PakiPenguin: any idea why it would do that? |
| 19:56.06 | PakiPenguin | Luke-Jr: no idea , fwd , works okay for me |
| 19:56.15 | PakiPenguin | call my number if you can |
| 19:56.23 | PakiPenguin | 272713 |
| 19:56.26 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 19:56.50 | dan2 | kram ping |
| 19:56.55 | Luke-Jr | Call Failed: Forbidden :( |
| 19:57.09 | PakiPenguin | Luke-Jr: it went in ... |
| 19:57.20 | harryvv | unable to register extention errors are usually associated with the lack of register => statments at the top of iax and sip.conf? |
| 19:57.45 | ManxPower | hardwire, no, it's caused by incorrect register statementws |
| 19:58.14 | netsurfer | harryvv |
| 19:58.17 | harryvv | :) |
| 19:58.18 | jcollie | does anyone know of an external ringer that can talk SIP to asterisk? |
| 19:58.25 | netsurfer | they can also be due to duplicate lines |
| 19:58.40 | harryvv | you mean the ones in the top of iax and sip |
| 19:58.45 | netsurfer | for example two s,1,whatever in the one context |
| 19:59.05 | netsurfer | anywhere in the conf |
| 19:59.12 | harryvv | sure |
| 19:59.30 | netsurfer | it should tell you which extension it cant register |
| 19:59.42 | harryvv | iax show registery whould show something |
| 19:59.57 | netsurfer | not that |
| 20:00.35 | harryvv | iax2 show registry |
| 20:00.45 | netsurfer | Unable to register extension 's', priority 25 in 'submenu9', already in use <-- example |
| 20:00.54 | harryvv | yea getting that error |
| 20:01.12 | ManxPower | hardwire, that is NOT a registry problem, that is a duplicate exten line problem. |
| 20:01.16 | netsurfer | then go into [submenu9] in this case, go then to s,25, |
| 20:01.17 | VoIPster | could I ask for a test call to FWD 607837? |
| 20:01.22 | Madkiss | Luke-Jr: So, any other idea? |
| 20:01.33 | harryvv | duplicate extention line okay |
| 20:01.36 | Alric | VoIPster: You can ask... :) |
| 20:01.40 | VoIPster | :) |
| 20:01.41 | netsurfer | right, harryvv |
| 20:02.02 | Alric | Don't ask to ask, just ask. If that makes any sense... |
| 20:02.21 | PakiPenguin | VoIPster: that was me :p |
| 20:02.22 | VoIPster | Mmmh. An echo terminal has just called me :) |
| 20:02.42 | ManxPower | I thought FWD let you call yourself. Personally I think FWD and IAXTel are pretty useless services and I don't use them. |
| 20:02.48 | PakiPenguin | VoIPster: on a laptop :) |
| 20:03.00 | VoIPster | Could you hear me Paki? |
| 20:03.10 | PakiPenguin | yeah , lound n clear |
| 20:03.15 | VoIPster | MaxPower: it does |
| 20:03.20 | harryvv | no echo? |
| 20:03.34 | harryvv | :) |
| 20:03.36 | ManxPower | VoIPMasta, then what's the point of hving someone else call you? |
| 20:03.45 | VoIPster | PakiPenguin: but you didn't say anything! :) |
| 20:04.11 | PakiPenguin | VoIPster: i did ! |
| 20:04.51 | VoIPster | PakiPenguin: really? |
| 20:04.56 | PakiPenguin | yeah |
| 20:05.05 | PakiPenguin | fwd does g729? or gsm? |
| 20:05.07 | VoIPster | PakiPenguin: mmmh. I didn't hear you |
| 20:05.18 | VoIPster | PakiPenguin g711ulaw |
| 20:06.00 | hardwire | hmmphm |
| 20:07.27 | harryvv | netsurfer thanks fairly easy to fix. Thought I dumped that line last night. |
| 20:07.40 | netsurfer | ;) yw |
| 20:07.40 | *** join/#asterisk loick (~loick@APuteaux-151-1-37-106.w82-124.abo.wanadoo.fr) |
| 20:07.41 | hardwire | hmmphm |
| 20:07.57 | PakiPenguin | holdon a sec :p problem on my side :p |
| 20:08.13 | *** join/#asterisk numbone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
| 20:09.11 | *** join/#asterisk jets (jetsn@omega.pmt.org) |
| 20:09.21 | PakiPenguin | VoIPster: Pakistan |
| 20:09.27 | modulus_ | Asterisk::Manager sucks |
| 20:09.53 | VoIPster | Aha |
| 20:09.56 | ta[i]nted | modulus_ |
| 20:10.07 | ta[i]nted | how much to colo a tower at your work |
| 20:10.24 | eKo1 | If I have insecure=very on a sip friend in sip.conf, any call I make to it will bypass authentication right? |
| 20:10.35 | tessier_ | eKo1: Yes. |
| 20:10.47 | tessier_ | eKo1: Only if you don't provide a username/pass probably |
| 20:11.12 | tzafrir_laptop | what exactly is vm/ in the sounds/ directory? |
| 20:11.14 | tessier_ | asterisk is still changing fast. I upgraded my asterisk and my sip.conf entries that don't authenticate broke. |
| 20:11.17 | eKo1 | You mean, if there is a username= and password= in the sip entry? |
| 20:11.21 | tessier_ | tzafrir: The sounds for the vm? |
| 20:11.26 | tzafrir_laptop | Why is there a separate vm and voicemail? |
| 20:11.48 | tessier_ | eKo1: I suspect if you provide a username and password it will authenticate with them. If you don't you have to add insecure=very to tell it you really know what you are doing and you intend to do no authentication. |
| 20:11.49 | tzafrir_laptop | should they be the same dir? |
| 20:11.55 | tessier_ | tzafrir: one is a symlink., |
| 20:12.03 | tessier_ | tzafrir: It is for historical backwards compatibility reasons |
| 20:12.09 | ManxPower | tessier_, What did you upgrade from and what did you upgrade to? |
| 20:12.11 | tessier_ | tzafrir: I suspect they will do away with the symlnk someday |
| 20:12.12 | modulus_ | anways |
| 20:12.19 | modulus_ | Asterisk::Manager SUCKS |
| 20:12.24 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 20:12.31 | jets | Does anyone know what O1 is? |
| 20:12.35 | VoIPster | I use gnugk, have an H.323 softphone reqistered and can't seem to make asterisk place a caller on hold. Any hints? |
| 20:12.57 | tzanger | hmm I think peer1's pulling some shennanigans |
| 20:12.58 | VoIPster | (<--newbie) |
| 20:13.03 | eKo1 | I'm asking because I'm having trouble recieving calls from my provider. |
| 20:13.10 | tzanger | my connections through peer1 are spotty today |
| 20:13.15 | tzafrir_laptop | Anyway, as I have the sounds under /usr/share/asterisk/sounds . voicemail is a symlink to /var/spool/asterisk/voicemail |
| 20:13.25 | tzafrir_laptop | So vm can be a symlink to voicemail? |
| 20:14.53 | *** join/#asterisk adrianhensler (~chatzilla@blk-222-123-93.eastlink.ca) |
| 20:15.28 | redder86 | http://bugs.digium.com/bug_view_page.php?bug_id=0003518 - the workaround was a simple little AGI. I don't understand why this "not supported" feature of SIP transfers of Queue-answered calls is so difficult. |
| 20:15.34 | ManxPower | tzafrir, You must be running some non-standard Asterisk distro or non-Linux. The detault place for sounds in Asterisk is /var/lib/asterisk/sounds |
| 20:15.57 | eKo1 | The call I'm making goes like this: Asterisk One ---> Provider 1 ---> PSTN ---> Provider 2 --> Asterisk 2. |
| 20:16.03 | harryvv | netsurfer got one more warning need to rid of ;) |
| 20:16.20 | netsurfer | k? |
| 20:16.37 | harryvv | <PROTECTED> |
| 20:16.50 | harryvv | it cannot find it |
| 20:18.06 | netsurfer | go into modules.conf.. noload => res_odbc.so |
| 20:18.13 | netsurfer | think thats the module... |
| 20:18.15 | harryvv | k |
| 20:19.22 | *** join/#asterisk FuRR_ ([tweKgeJGG@bko29.chapman.edu) |
| 20:19.33 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 20:20.09 | tzanger | kram: werd |
| 20:21.01 | Derkommissar | Im going to test that |
| 20:21.29 | ta[i]nted | Derkommissar wasn't that an 80s band? |
| 20:21.31 | AgiNamu | Does anyone know where to get the G.729 "external" DLL for FireFly? |
| 20:21.40 | Derkommissar | Yea |
| 20:21.50 | tzanger | "I'm going ot test that" -- I don't remember hearing of an 80's band like that |
| 20:22.08 | ta[i]nted | tzanger |
| 20:22.19 | ta[i]nted | "DerKommissar" = 80's band |
| 20:22.24 | tzanger | :-) |
| 20:22.43 | Nugget | Der Kommissar is a song, not a band. The band was named After The Fire |
| 20:23.03 | ta[i]nted | oh |
| 20:23.27 | ta[i]nted | it comes up in my mp3 playlist every few days |
| 20:23.56 | Nugget | and since you stole it off some ghetto pirate2pirate network, you have shitty and incomplete mp3 tags. |
| 20:23.59 | Nugget | way to go. |
| 20:24.10 | ta[i]nted | actually.. |
| 20:24.22 | egrat | exit |
| 20:24.23 | ta[i]nted | i have this billboard top 100 collection since 1967 |
| 20:24.42 | ta[i]nted | it's around 3200 mp3s.. with full mp3 tags |
| 20:24.49 | Nugget | My copy is from TimeLife: Sounds of the Eighties -- The Early '80s |
| 20:25.00 | eKo1 | Any SIP experts out here? |
| 20:25.09 | jets | A handful. |
| 20:25.18 | Nugget | eKo1: do you really need an expert, or just some guy who knows the answer to the question you're avoiding asking? |
| 20:25.23 | eKo1 | I mean experts with the protocol. |
| 20:25.30 | ta[i]nted | mine fits in my 40GB ipod |
| 20:25.33 | FuRR_ | eKo1: "out here?" wouldnt that be "in here" or "out there"... |
| 20:25.41 | jets | Yes. |
| 20:26.25 | eKo1 | OK, my providers proxy server is sending my * an INVITE but * seems to respond with a 407. |
| 20:26.54 | eKo1 | Looking at the from and to fields in the header, i notice that they have the same ip. |
| 20:26.58 | AgiNamu | Has anyone ever bought licenses from VoiceAge? |
| 20:27.30 | ManxPower | AgiNamu, not since the dawn of time. Digium buys the license and sells licenses to end users. |
| 20:27.31 | *** join/#asterisk Wireless (~bad@220.233.77.87) |
| 20:27.34 | eKo1 | I also notice that the from field has the extension number of the phone I dialed on the other * server I made the call from. |
| 20:27.38 | AgiNamu | Manx, no, for Softphone |
| 20:27.40 | AgiNamu | FireFly |
| 20:27.48 | ManxPower | AgiNamu, HAHAHAHA! |
| 20:28.05 | Wireless | Need help with using firefly (freshtel) as incoming gateway. |
| 20:28.16 | Wireless | calls can be bridged to the Zap channel but not elsewhere. |
| 20:28.18 | Wireless | anyone? |
| 20:28.44 | freat | afternoon |
| 20:28.46 | AgiNamu | Manx... what's so funny? |
| 20:28.49 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 20:28.54 | AgiNamu | How the hell am I supposed to get G729 with Firefly? |
| 20:29.12 | freat | anyone using the Packet Loss Concealment / jitterbuffer patches on stable? |
| 20:29.16 | ManxPower | AgiNamu, Isn't that a question for the Firfly people? |
| 20:29.33 | AgiNamu | yea, I suppose. but they dont seem to answer |
| 20:29.43 | AgiNamu | Probably they are pissed cause I figured out that they are selling $20 USB phones for $100 |
| 20:29.49 | Wireless | aginamu, do you mean the firefly service or the softphone? |
| 20:30.24 | AgiNamu | softphone |
| 20:30.29 | AgiNamu | actually, any IAX softphone |
| 20:30.29 | Wireless | oh ok. |
| 20:30.46 | AgiNamu | OK,... I see Sipro only starts licensing with a minimum quantity of 50K... so fuck them |
| 20:30.56 | eKo1 | Hmm...maybe if I change the realm in the sip entry of my provider... |
| 20:31.00 | FuRR_ | anyone know of a good primer for IAX2. I am thinking about building my own ATA and need to wrap my head around the protocol itself. |
| 20:31.30 | freat | FuRR_ there's documentation for it on digiums website |
| 20:31.30 | Wireless | building your own? for mass production? |
| 20:31.36 | oej | FuRR: There's a draft document (PDF) with a specification out there. |
| 20:31.37 | freat | http://www.digium.com/ |
| 20:31.41 | ManxPower | AgiNamu, patented codecs are an issue for open source software. |
| 20:31.51 | oej | FurRR_: YOu will find it if you find the IAX2 page on the wiki |
| 20:31.53 | FuRR_ | Wireless: eh more so of a project, to bridge an interesting problem right now i have in my lab |
| 20:32.07 | oej | FuRR_: That doc and the source is all there is |
| 20:32.09 | AgiNamu | Manx, even if it's not open source... how do I buy a single freaking license? I guess I can't. |
| 20:32.16 | Wireless | furr, maybe cheaper/easier to just buy an iaxy? |
| 20:32.18 | AgiNamu | Well, I'll just tell our customers to build their own |
| 20:32.23 | AgiNamu | and they can worry about licenses. |
| 20:32.38 | FuRR_ | Wireless: the current IAXy doesnt have the port density I need for my application |
| 20:33.08 | FuRR_ | and the only ones that come close are way out of my price range |
| 20:33.10 | dan2 | oej!!! |
| 20:33.16 | Wireless | furr, i see... |
| 20:33.22 | FuRR_ | ...and are based on SIP |
| 20:33.41 | Wireless | how many ports you need? |
| 20:33.50 | FuRR_ | 24 |
| 20:33.54 | tzafrir_laptop | is there's any free (copyright-wise, not patent-wise) G729 support for * available? |
| 20:34.10 | Nugget | no. |
| 20:34.12 | ManxPower | AgiNamu, Licenseing terms are up to the patent holders. If they don't want to sell individuals there's nothing you can do about it. |
| 20:34.38 | ManxPower | FuRR_, It's called a Channel Bank + T100P + Asterisk |
| 20:34.43 | FuRR_ | tzafrir_laptop: if that were the case alot of people wouldve switched over to it by now |
| 20:34.53 | FuRR_ | ManxPower: doesnt help in my situation |
| 20:34.54 | Wireless | furr, how about an asterisk box with 6 4-port FXS cards? |
| 20:35.09 | ManxPower | Wireless, Are you TRYING to set him up to fail?? |
| 20:35.19 | ManxPower | "6 4-port cards" indeed! |
| 20:35.20 | Wireless | nono... but what did i do wrong? |
| 20:35.33 | AgiNamu | tza: VoiceAge has an 'open implementation' for win32 |
| 20:35.55 | *** join/#asterisk lohelle (~post@213.161.252.253) |
| 20:36.00 | FuRR_ | ManxPower: i need to be able to slide asterisk in -- presumably into an already deployed area and basically be given a network drop and thats it |
| 20:36.23 | ManxPower | FuRR_, Good luck with that. |
| 20:36.30 | Wireless | manx, pardon my stupidity, but what did i do wrong? |
| 20:36.34 | FuRR_ | ManxPower: thats what i said |
| 20:36.42 | ManxPower | Wireless, Have you ever tried to put 6 Digium cards in one machine?? |
| 20:36.50 | ManxPower | You'll run out of interrupts |
| 20:36.51 | Wireless | okok... i get your point. :-) |
| 20:37.22 | ManxPower | and that's not even counting the 6,000 interrupts per second that have to be serviced and completed fast. |
| 20:37.32 | FuRR_ | Wireless: show me a current mobo with 6pci slots |
| 20:37.36 | bjohnson | FuRR_: wouldn't a * server fall into that description? |
| 20:37.39 | harryvv | Guess there never will be more then a 6 slot motherboard any time soon. |
| 20:37.45 | Wireless | furr - i have one with five. |
| 20:37.51 | lohelle | Is there any mainboards out there where you REALLY can choose which IRQ for EACH PCI slot + integrated devices? |
| 20:37.54 | FuRR_ | Wireless: does 5 = 6 ? |
| 20:38.04 | Wireless | furr, hey be nice! :-) |
| 20:38.06 | AgiNamu | I guess I can buy the $10 ones from Digium |
| 20:38.07 | ManxPower | number of slots really isn't the issue. Each Digium card needs to have it's own (not shared) IRQ. |
| 20:38.28 | ManxPower | AgiNamu, the Digium G729 licenses are only for Asterisk |
| 20:38.55 | ManxPower | ~docs |
| 20:38.56 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:39.00 | bjohnson | I don't understand why the * + channel bank can't be put in |
| 20:39.49 | PakiPenguin | <PROTECTED> |
| 20:39.53 | FuRR_ | bjohnson: sometimes we're pressed into a situation where the client doesnt want to rip out phones and install ip handsets and sometimes the cost in getting enough digium cards, or even enough physical ports to handle their number of handsets is too great. So, we're looking at building our own SuperATA |
| 20:40.10 | ManxPower | FuRR_, Um, this would be a superATA. |
| 20:40.23 | FuRR_ | ManxPower: SuperATA/66 hehe |
| 20:40.24 | ManxPower | The combination would convert IAX to analog ports. |
| 20:40.34 | FuRR_ | ManxPower: yes |
| 20:40.35 | hardwire | ok |
| 20:40.35 | ManxPower | And that's what you want, right? |
| 20:40.35 | *** join/#asterisk cp5 (~samy@h-66-167-231-76.lsanca54.dynamic.covad.net) |
| 20:40.37 | cp5 | hola |
| 20:40.40 | AgiNamu | hmm... so for softphones, your fucked with G729 |
| 20:40.40 | hardwire | anybody here familiar with Vanu Inc? |
| 20:40.52 | AgiNamu | unless someone throws down $70K+ in licenses to start |
| 20:40.54 | PakiPenguin | FuRR_: http://farfon.com/ff110.pdf |
| 20:41.06 | lohelle | anyone using Avaya 4630 + asterisk? can get one (6 months old) for 100$. |
| 20:41.15 | cp5 | anyone know how i can access the 'agents' struct that was initiated from chan_agent.c, but access it from app_queue.c |
| 20:41.25 | bjohnson | FuRR_: for less than $30 USD per port? |
| 20:41.26 | FuRR_ | yahtze |
| 20:41.37 | hardwire | yay |
| 20:41.39 | FuRR_ | thanks PakiPenguin |
| 20:41.41 | PakiPenguin | there's a super ata coming too |
| 20:41.52 | FuRR_ | bjohnson: yes |
| 20:42.06 | FuRR_ | bjohnson: think of a situation where you have 500 or so ports in a building |
| 20:42.06 | ManxPower | FuRR_, talk to wasim before you think about building hardware. |
| 20:42.13 | FuRR_ | 30$ /pt is a considerable cost |
| 20:42.25 | hardwire | a lifetime liscence.. right? |
| 20:42.32 | ManxPower | FuRR_, you're not going to get it cheaper than $30 - $50 per port. |
| 20:43.01 | FuRR_ | ManxPower: I already have the resources to build the cabinet |
| 20:43.04 | ManxPower | FuRR_, You want something that neither Asterisk nor Digium nor any other company can deliver at the current time. |
| 20:43.15 | FuRR_ | ManxPower: I just need to build the low level software to drive it |
| 20:43.18 | AgiNamu | Furr, learn mandarin and fly to shanghai or shenzhen |
| 20:43.22 | tzanger | what's the general status of asterisk on uclibc? |
| 20:43.27 | AgiNamu | but $30 a port is really low |
| 20:43.38 | ManxPower | FuRR_, To drive *what*? |
| 20:44.00 | FuRR_ | ManxPower: basically its just going to be a cabinet with line cards in it |
| 20:44.07 | ManxPower | FuRR_, from who? |
| 20:44.34 | hardwire | have you guys seen the PCI device with a few strongarms on it |
| 20:44.36 | FuRR_ | ManxPower: FXS ports |
| 20:44.36 | ManxPower | Oh! You want to write a replacement operating system for your PBX. |
| 20:44.51 | hardwire | that are bootstrapped to basically just encode from their t1 port? |
| 20:44.57 | tzanger | how did you arrive at $30/port? Is this an actual physical device with a BOM you've detailed? |
| 20:44.59 | hardwire | g.729 |
| 20:45.18 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 20:45.23 | tzanger | hardwire: yeah but you have the same basic issue that any DSP card has -- multiple PCI bus traversals to get the data |
| 20:45.29 | FuRR_ | ManxPower: All the cabinet will be is a dumb device acting as a giant ATA speaking IAX2 to the asterisk box |
| 20:45.30 | hardwire | ah |
| 20:45.45 | ManxPower | FuRR_, When you are done with that project can you build me a hovercraft out of tin cans and cat hair? |
| 20:45.48 | hardwire | tzanger.. then use USB!! |
| 20:45.56 | tzanger | ha |
| 20:46.07 | FuRR_ | ManxPower: sure, if i dont electricute myself first |
| 20:46.31 | ManxPower | FuRR_, nothing personal, but I put crazy people on /ignore. |
| 20:46.39 | FuRR_ | I know I saw on VOIP Supply they had a 16pt SIP gateway |
| 20:46.47 | hardwire | http://www.aculab.com/products/dna/ip_telephony_card.htm |
| 20:46.50 | hardwire | weird stuff |
| 20:46.57 | FuRR_ | ManxPower: and what makes you think this idea is crazy |
| 20:47.02 | hardwire | tzanger: what are you trying to do ATM? |
| 20:47.23 | tzanger | hardwire: what do you mean |
| 20:47.29 | hardwire | at the moment |
| 20:47.31 | hardwire | whats your issue? |
| 20:47.31 | AgiNamu | Furr_ wtf are you trying to do? |
| 20:47.45 | tzanger | hardwire: what do you mean, what am I trying to do? |
| 20:47.54 | hardwire | I am just butting in |
| 20:47.56 | hardwire | whats the topic :) |
| 20:48.15 | tzanger | hardwire: FuRR_'s got some kind of SuperATA and claiming $30/port, I just wanted more details |
| 20:48.27 | ManxPower | hardwire, What does the card COST? |
| 20:48.31 | FuRR_ | i never claimed $30/pt |
| 20:48.33 | FuRR_ | i rebuffed |
| 20:48.35 | hardwire | alotofmoney |
| 20:48.44 | FuRR_ | i figured i could probably do it cheaper than 30$ a port |
| 20:48.51 | tzanger | just curious as to what it's got compared to a T100P+AB1 for 24 FXS |
| 20:49.00 | tzanger | FuRR_: FXS only or FXS+FXO? |
| 20:49.00 | hardwire | ah |
| 20:49.01 | hardwire | gotcha |
| 20:49.09 | FuRR_ | Only FXO |
| 20:49.12 | FuRR_ | err |
| 20:49.14 | FuRR_ | FXS |
| 20:49.17 | FuRR_ | hehe |
| 20:49.18 | hardwire | you are crazy |
| 20:49.23 | hardwire | its settled |
| 20:49.25 | hardwire | end of discussion |
| 20:49.37 | FuRR_ | i dont see anything crazy about it, only the fact it hasnt been done yet |
| 20:49.55 | AgiNamu | well hey, let us know when it's built--im sure you'll sell a lot |
| 20:49.56 | FuRR_ | it seems like a logical progression for companys that dont want to give up their vested infrastructer |
| 20:50.00 | hardwire | err |
| 20:50.01 | hardwire | so |
| 20:50.02 | AgiNamu | at $30 a port or less... that's damn good. |
| 20:50.06 | tzanger | FuRR_: as I said, your development time & debugging time is going to eat you alive to get under the $750 that a 24 port FXS AB1+T100P in a cheapass system will get you |
| 20:50.08 | hardwire | SuperATA is what? |
| 20:50.15 | hardwire | 24 ports of FXS? |
| 20:50.16 | hardwire | more? |
| 20:50.17 | ManxPower | hardwire, The world needs people that push the limits, but I wish they would do it in private and wash their hands after. |
| 20:50.18 | tzanger | hardwire: what he wants to build |
| 20:50.30 | tzanger | ManxPower: hahaha |
| 20:50.32 | hardwire | ManxPower: I push my limits all the time.. |
| 20:50.36 | hardwire | and I have to clean up the mess quite often |
| 20:50.42 | hardwire | VHFoIP is no fun |
| 20:51.03 | AgiNamu | $30 is a lot acutally. it should have an FXO and FXS for that much money |
| 20:51.04 | ctooley | I don't mean to be repetitive but is there anyone from TX Link here? |
| 20:51.16 | FuRR_ | AgiNamu: FXO is not the issue |
| 20:51.35 | hardwire | FuRR_: the only way you are going to push that is if you make your own g.729a codec outside of digiums codebase |
| 20:51.44 | AgiNamu | Wait, it does G729 too?? |
| 20:51.55 | FuRR_ | AgiNamu: from what ive seen, most people are scared to upgrade for fear of losing their vested phone sets |
| 20:51.58 | tzanger | $30 a port means you can get a 24-port version for $720 |
| 20:52.21 | FuRR_ | how in the heck did G729 get put into this |
| 20:52.22 | tzanger | which is my cost for a T100P+AB1 ... well add $75 for the P3/750 with 128M RAM and 6.4G HDD |
| 20:52.22 | *** join/#asterisk rene- (~rene@dsl-200-67-149-252.prod-empresarial.com.mx) |
| 20:52.25 | rene- | hey |
| 20:52.26 | Luke-Jr | AgiNamu: Where are you seeing prices < $55 ??? |
| 20:52.29 | hardwire | FuRR_: heh |
| 20:52.33 | AgiNamu | Like - China |
| 20:52.42 | tzanger | so say $900 I can supply a system that does that at 24 ports for $37/port |
| 20:52.51 | tzanger | without any design work |
| 20:52.54 | hardwire | isn't there an fxs to 24 port t1? |
| 20:52.54 | AgiNamu | So Furr_, you're manufacturing a device with 24 FXO ports, G729 transcoding, and a single 100Mbps ethernet to speak IAX2 and it's gonna be $700? |
| 20:53.01 | hardwire | all digital no transcoding |
| 20:53.03 | hardwire | out there somewhere |
| 20:53.04 | tzanger | hardwire: you have no idea what you're saying. :-) |
| 20:53.06 | rene- | i wanted to know if there was anything special i need to write in my extensions.conf to dial to a pbx that requires 9 to dial outside |
| 20:53.06 | hardwire | like a happy little pbx |
| 20:53.07 | Luke-Jr | AgiNamu: anywhere online? |
| 20:53.10 | FuRR_ | AgiNamu: aparently |
| 20:53.11 | hardwire | tzanger: I probably don't |
| 20:53.14 | hardwire | cause I am quite insane. |
| 20:53.23 | rene- | im doing 9{$EXTEN} but i dont know if its enough |
| 20:53.38 | hardwire | tzafrir: but what brought you to that conclusion? |
| 20:53.39 | AgiNamu | Luke, no. We're working on putting up the phones. (They do 5 protocols, PA168 based) |
| 20:53.47 | rene- | probably there is some expression to make * wait between the 9 and the ${EXTEN} |
| 20:53.49 | AgiNamu | But after shipping, taxes, etc. it's a lot more |
| 20:54.00 | AgiNamu | If you need the best price, it's better to find someone in china to build for you |
| 20:54.05 | ManxPower | rene-, "w" = .5 second wait on ANALOG ports |
| 20:54.20 | Wireless | rene: it's 9${EXTEN} not 9{$EXTEN} |
| 20:54.25 | Luke-Jr | AgiNamu: I just want a one or two port FXS... not mass-quantity enough for something like that o.o |
| 20:54.31 | rene- | Wireless im aware that was my typo |
| 20:54.33 | AgiNamu | oh |
| 20:54.35 | rene- | thank Manx |
| 20:54.44 | *** join/#asterisk Syrus_ (~pascal@tahiti.mpl.rullier.net) |
| 20:54.48 | AgiNamu | Well, I have some, 1 AND 2 port IAX2 devices (do GSM, G729 too) |
| 20:54.53 | AgiNamu | but they're in guatemala |
| 20:55.06 | AgiNamu | $104 retail |
| 20:55.10 | *** part/#asterisk jcollie (~jcollie@161.210.6.51) |
| 20:55.10 | Luke-Jr | ouch |
| 20:55.12 | AgiNamu | yea |
| 20:55.16 | AgiNamu | plus shipping :( |
| 20:55.22 | AgiNamu | We can go down a lot in volume |
| 20:55.29 | AgiNamu | jsut that w'ere not a hardware company so its not worth it |
| 20:55.37 | AgiNamu | to sell one by one |
| 20:55.43 | AgiNamu | ariavoice is selling them too |
| 20:55.45 | AgiNamu | similar |
| 20:55.49 | AgiNamu | like $80 or so? |
| 20:56.00 | *** join/#asterisk BBRodriguez (~alex@pD9563916.dip.t-dialin.net) |
| 20:56.04 | Madkiss | I found this in the debug messages when an incoming call comes: |
| 20:56.05 | Madkiss | Found peer 'freenet-out' |
| 20:56.05 | Madkiss | Reliably Transmitting (no NAT): |
| 20:56.05 | Madkiss | SIP/2.0 407 Proxy Authentication Required |
| 20:56.06 | AgiNamu | and some guy in germany, but his units are $100 euros or more i think |
| 20:56.17 | Madkiss | Am _I asking my _Phone-provider to authenticate? |
| 20:56.21 | Luke-Jr | $55 is expensive for me |
| 20:56.32 | AgiNamu | luke, for an IAX2 device? |
| 20:56.35 | AgiNamu | Then... good luck. |
| 20:56.41 | Luke-Jr | AgiNamu: SIP works too |
| 20:56.48 | AgiNamu | oh then im sure there are discount sellers |
| 20:56.52 | AgiNamu | i MUST have IAX2 |
| 20:56.52 | firestrm | can anyone tell me the name of the magical script that automagicly adds the proper files to /etc/init.d for zaptel startup? I did it once before, but im brain fartting on how to do it again.. |
| 20:56.54 | ManxPower | I consider $50 to "get a phone on someone's desk" to be CHEAP. |
| 20:56.55 | AgiNamu | SIP is a piece of shit |
| 20:56.58 | rene- | Manx sorry to bother is w an option for the Dial command or an app by itself? |
| 20:57.04 | Luke-Jr | AgiNamu: why? |
| 20:57.06 | AgiNamu | manx, damn straight |
| 20:57.10 | FuRR_ | firestrm: ntsysv ? |
| 20:57.12 | AgiNamu | look up SIP, RPT and NAT |
| 20:57.21 | AgiNamu | pretty much SIP + NAT doesnt work. |
| 20:57.25 | ManxPower | rene-, Dial(Zap/g1/9w${EXTEN}) |
| 20:57.27 | AgiNamu | it *might* work |
| 20:57.29 | Luke-Jr | But it'd be on a LAN |
| 20:57.31 | AgiNamu | in *some* conditions |
| 20:57.36 | AgiNamu | yea, fine, you've got a different need :) |
| 20:57.37 | firestrm | FuRR_, no it was part of the makefile.. if i remember correctly.. but im not sure |
| 20:57.38 | Luke-Jr | going thru * to get anywhere |
| 20:57.53 | AgiNamu | i need to drop ship phones and not worry |
| 20:58.01 | ManxPower | Generally SIP + NAT works fine for corporate enviroments, but not for service providers. |
| 20:58.01 | AgiNamu | IAX2 does it for me. SIP does not. |
| 20:58.12 | AgiNamu | yea, in a corp. env. where you can setup the firewall, fix the nat |
| 20:58.13 | AgiNamu | etc. |
| 20:58.25 | sskyles | hmm, I got Asterisk running on my PowerMac Last night... |
| 20:58.28 | AgiNamu | but not for letting people take them around 3rd world countries and expecting it to work |
| 20:58.30 | rene- | gracias Manx :-) |
| 20:58.32 | ManxPower | i.e. If you accept the few limitations I outlined in my recent message to the mailing list, then SIP + NAT works just peachy. |
| 20:58.35 | firestrm | it was somthing like make startup.. |
| 20:58.44 | Luke-Jr | sskyles: Well, the problem is that KPhone's audio is screwed up |
| 20:58.45 | AgiNamu | its the "accept the few limitations" part that sucks :) |
| 20:58.58 | AgiNamu | at any rate, I'm getting good prices for the IAX devices so |
| 20:59.01 | AgiNamu | works fine for me |
| 20:59.02 | FuRR_ | it kinda scares me how the current crop of service providers offer really nothing in the way of encrypting SIP packets, at least here at work we have a VPN setup for our remote phones |
| 20:59.17 | ManxPower | AgiNamu, Basically: No reinvites, and the Asterisk server can't be behind NAT. |
| 20:59.19 | AgiNamu | furr.... almost no one does any encryption on the internet today |
| 20:59.27 | AgiNamu | Manx, it's more than that |
| 20:59.29 | AgiNamu | a lot more |
| 20:59.35 | AgiNamu | double NAT for instance |
| 20:59.39 | ManxPower | And honestly I don't think IAX2 transfers work for many NAT routers anyway. |
| 20:59.42 | Luke-Jr | AgiNamu: I encrypt most everything |
| 20:59.44 | AgiNamu | can fuck it up. STUN sucks |
| 20:59.49 | AgiNamu | Luke, I mean service providers |
| 20:59.51 | ManxPower | AgiNamu, "double nat"? |
| 20:59.53 | AgiNamu | sure |
| 20:59.56 | FuRR_ | AgiNamu: i think your a bit off there |
| 20:59.58 | AgiNamu | ISP NATs your Router |
| 20:59.59 | ManxPower | I just said you can't have Asterisk behind NAT. |
| 21:00.04 | AgiNamu | no, the client |
| 21:00.21 | ManxPower | AgiNamu, Oh! You mean "SIP + NAT does not work if you have the worlds shittiest ISP" |
| 21:00.26 | AgiNamu | lol yep |
| 21:00.34 | ManxPower | My only answer to that is "Duh!" |
| 21:00.44 | AgiNamu | Furr_ really? Then why are we still using SMTP with no encryption for most email? |
| 21:00.45 | Luke-Jr | If they NAT the customers, I would dispute their usage of the term "ISP" |
| 21:00.53 | AgiNamu | Why is it that a lot of email programs dont even know what S/MIME is? |
| 21:00.54 | ManxPower | Luke-Jr, I would as well. |
| 21:01.08 | AgiNamu | Yea, go ahead and dispute it. meanwhile... I gotta make it work with them :) |
| 21:01.16 | Luke-Jr | AgiNamu: Most I know recognize S/MIME |
| 21:01.33 | AgiNamu | i dont believe any of the web-based ones do |
| 21:01.41 | firestrm | FuRR_, i remembered (actually i went through the makefile line by line..) make config .. funny that they dont have that on the asteriskdocs page.. |
| 21:01.43 | AgiNamu | and i dont know of many ISPs that allow you to do SSL SMTP |
| 21:01.46 | Luke-Jr | SquirrelMail seems to |
| 21:01.51 | Luke-Jr | AgiNamu: SSL SMTP is pointless |
| 21:01.52 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 21:01.56 | AgiNamu | why is that so? |
| 21:02.02 | AgiNamu | i mean, SSL POP3 |
| 21:02.06 | Luke-Jr | AgiNamu: If it's PGP encrypted like it should anyway... |
| 21:02.20 | AgiNamu | PGP only solves part |
| 21:02.21 | Luke-Jr | SSL POP3 might make more sense |
| 21:02.21 | Nivex | SSL SMTP wouldn't be pointless if *everyone* did it |
| 21:02.28 | AgiNamu | nivex - bingo |
| 21:02.58 | AgiNamu | unfortunately, its simply never going to happen since its impossible for a worldwide PKI |
| 21:03.21 | tessier_ | ManxPower: SMTP+AUTH is godly |
| 21:03.32 | AgiNamu | As far as voice being encrypted |
| 21:03.37 | AgiNamu | you dont get that from your current phoen system |
| 21:03.41 | tessier_ | ManxPower: Finally, no more having to explain to lusers why they have to change their smtp server to whatever server is local to their network. |
| 21:03.47 | *** part/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 21:03.52 | AgiNamu | so why would you (being a provider) want to go thru the hassle for voip? |
| 21:03.53 | Madkiss | ah, a little step forward again |
| 21:03.57 | ManxPower | tessier_, Exactly. Especially lap top users. |
| 21:04.03 | Madkiss | Feb 8 22:03:25 WARNING[14621]: app_dial.c:701 dial_exec: Dial argument takes format (technology1/[device:]number1&technology2/[device:]number2...|optional timeout) |
| 21:05.00 | ManxPower | Madkiss, Well fix it. |
| 21:05.22 | FuRR_ | AgiNamu: its harder for someone to tap your phone line, than it is for someone to sniff your data |
| 21:05.39 | FuRR_ | look at cable modem users |
| 21:05.45 | FuRR_ | cable modems are basically hubs |
| 21:05.57 | FuRR_ | you can see all traffic for your segment (block? neighborhood even) |
| 21:06.45 | *** join/#asterisk outsidefactor (barf@203-206-254-16.dyn.iinet.net.au) |
| 21:06.59 | Wireless | i'm having an issue with bridging calls from an IAX provider. |
| 21:07.25 | Wireless | Someones calls a internal number I have on an IAX provider, my * will answer and then ask for extension |
| 21:07.35 | Madkiss | ManxPower: What is "Unable to create channel of type SIP"? |
| 21:07.37 | Beirdo | FuRR_: I don't think that's technically accurate any more |
| 21:07.40 | Wireless | caller punches in ext and * calls. |
| 21:07.43 | Madkiss | The client is registered there |
| 21:08.03 | ManxPower | Wireless, notransfer=yes in iax.conf |
| 21:08.08 | Wireless | i can get it to work on Zap channels but not on other IAX/SIP. |
| 21:08.11 | crash3m | FuRR_: gimme a $3 phone with an 'in use' indicator, a pair of wire strippers, and a 9/16" thinwall socket...I'll tap your phone :P |
| 21:08.18 | Wireless | manx, thanks. tried that - doesn't work... |
| 21:08.35 | FuRR_ | crash3m: yes, and i can see you doing it |
| 21:08.41 | crash3m | very trye |
| 21:08.43 | crash3m | true even |
| 21:08.52 | FuRR_ | i guess it just comes down to knowing your being tapped |
| 21:09.22 | FuRR_ | id feel much better if you were standing out there in some coveralls with a Mitel patch on (even though Mitel doesnt do residential) then if someone was secretly sniffing my data |
| 21:09.25 | FuRR_ | wierd how that is |
| 21:09.33 | AgiNamu | Furr_.... well.... easier tapping data versus phone? perhaps in some scenarios. |
| 21:10.06 | Beave | Depends on where you live as well, for example.. appartments are usually easy to tap |
| 21:10.09 | harryvv | btw, what is used to bridge digital phones with asterisk? I see alot of mital phones here |
| 21:10.24 | FuRR_ | Beave: true |
| 21:10.29 | Wireless | manx, any more ideas? It happens even when dialing out using another IAX provider using the same codec, or even when calling locally registered SIP phones. |
| 21:10.51 | AgiNamu | What is the security on IAX2 when dialing a context? Am I correct in that you can do IAX2/whatever<some syntax>context? |
| 21:11.25 | *** part/#asterisk outsidefactor (barf@203-206-254-16.dyn.iinet.net.au) |
| 21:12.49 | Wireless | the called phone would ring. |
| 21:13.08 | Wireless | and i the callee can answer, but on the other end the caller's phone just keeps ringing non stop. |
| 21:13.20 | Wireless | the only exception is when using Zap channels, when everything works. |
| 21:14.37 | Wireless | i mean, the callers just hear the ringing tone non stop, even when the other side has answered. |
| 21:15.12 | ManxPower | Wireless, good thing you are discovering this before deployment, huh? |
| 21:15.55 | Wireless | manx: eh? I'm still in the testing stage, when i'm the only tester. |
| 21:16.15 | Wireless | manx: so just keep calling myself using a variety of phones. |
| 21:16.53 | ManxPower | Wireless, Some people, blinded by the vision of Asterisk, deploy an entire office using and not actually do any real testing beforehand. These people are commonly called "soon to be unemployed" or sometimes "idiots" |
| 21:17.19 | *** join/#asterisk rowters (~Drake@201.133.209.110) |
| 21:17.30 | *** join/#asterisk Corvin (~zbysio@chello084010031149.chello.pl) |
| 21:17.38 | harryvv | Manx, and thay come here? :) |
| 21:17.58 | harryvv | I agree with the testing part. |
| 21:18.04 | godsmoke | someone here must know something about how asterisk handles CPN and Caller ID -- someone :) |
| 21:18.30 | Bile_One | Manxpower, how much about the licensing do you know? |
| 21:18.37 | ManxPower | godsmoke, Asterisk CID guts, no. General CID stuff, etc. |
| 21:18.56 | harryvv | Test each and every connection that lives up to whats its job is. Test zap you name it. I even told netsurfer to make another astersk box be the CO and have test scripts to send out calls to the tested asterisk box and put a simulted load on it. |
| 21:19.06 | godsmoke | well ... I'll ask you, and we'll see if you know enough to help |
| 21:19.12 | ManxPower | Bile_One, not much. I got the $30 license for my personal use. |
| 21:19.36 | godsmoke | ManxPower: ok -- so, from what little I know about the telephone network, as a trunked pbx, it's asterisk's job to decide if caller id should be displayed to the end-user |
| 21:19.38 | Bile_One | Okay so we would need a better license but we could do the personal for or testing right? |
| 21:19.46 | Bile_One | Or = our |
| 21:19.49 | *** join/#asterisk vondax (~dax@host81-139-83-249.in-addr.btopenworld.com) |
| 21:19.58 | Wireless | manx: I'm just a home user and asterisk is a hobby. I hope u're not trying to imply anything about me. |
| 21:20.21 | godsmoke | ManxPower: and in a PSTN network, this is controlled by the CLID switch in the SS7 signalling ... and the PBX decides if the CPN should be displayed on the box, or whatever |
| 21:20.23 | Bile_One | ManxPower, I will get that to start, but we like it. |
| 21:20.31 | harryvv | This is how microsoft worked. Ask every employee willing to test windows 2000 over night by placing debuggers on the network and load testers. All these combined boxes would put the windows2000 servers to there paces over night |
| 21:20.39 | ManxPower | Bile_One, Check with Cepstral people, but it seemed like you only needed an expensive license if you wanted to use the Cepstral API. I've only used the command line application to generate TTS .WAV files, so no direct API usage. |
| 21:21.09 | ManxPower | godsmoke, Something like that, but that only applies to PRI connectens between the PSTN and Asterisk |
| 21:21.28 | godsmoke | ManxPower: ok ... so what's the equivalent of that switch in the VoIP world? |
| 21:21.41 | ManxPower | godsmoke, Huh? |
| 21:21.44 | Bile_One | ManxPower, their website has a list but their terms (words) are not the same as what we think they should mean. |
| 21:21.46 | godsmoke | or is there no such thing as caller-id block in VoIP? (detined for a VoIP pbx) |
| 21:22.02 | godsmoke | destined* |
| 21:22.07 | ManxPower | godsmoke, that is vendor specifc |
| 21:22.13 | godsmoke | uh ... |
| 21:22.21 | godsmoke | I don't think we're understanding each other |
| 21:22.52 | ManxPower | SS7 protocol has options for blocking CallerID Display. So does the PRI protocol. SIP, IAX, method of doing that would be vendor dependent. |
| 21:22.54 | godsmoke | in a PSTN network ... the CPN is the caller's number -- blocked or not -- and then the CLID switch tells the CLEC's PBX whether or not to display the CPN on the CID box ... what's the equivalent of that system in VoIP? |
| 21:23.20 | ManxPower | godsmoke, there is no such system in voip. |
| 21:23.35 | godsmoke | hrm |
| 21:23.36 | *** join/#asterisk slak- (~alex@dsl093-236-218.hfd1.dsl.speakeasy.net) |
| 21:23.40 | godsmoke | so there is no way to block a number? |
| 21:23.47 | slak- | hey can someone help me get mysql setup for a cdr table |
| 21:23.50 | slak- | so i can use asterisk-stat |
| 21:23.54 | slak- | damnit |
| 21:24.05 | ManxPower | godsmoke, Sure. dial a code that tells your local pbx not to transmit the number when the call happens |
| 21:24.19 | ManxPower | that code would be vendor specific. |
| 21:24.34 | godsmoke | hrm |
| 21:24.48 | godsmoke | so what happens to the ANI? |
| 21:25.11 | ManxPower | godsmoke, as far as I've been able to determine the concept of ANI does not exist in any VoIP protocol. |
| 21:25.27 | godsmoke | well ... it's in IAX |
| 21:25.52 | ManxPower | you either accept anonymous calls or all call are authenticated and you know who it is regardless of their CLID. |
| 21:26.09 | godsmoke | CPN and ANI are in the IAX protocol ... |
| 21:26.25 | ManxPower | godsmoke, Has anyone used them? 8-) |
| 21:26.30 | godsmoke | yes .... |
| 21:26.37 | godsmoke | they HAVE to be used in the PSTN network |
| 21:26.47 | *** join/#asterisk adnans (~adnans@linuxgoeroe.demon.nl) |
| 21:26.51 | godsmoke | it's illegal not to use one |
| 21:26.58 | ManxPower | godsmoke, a *LEC will set the ANI if they want. |
| 21:27.20 | godsmoke | alright |
| 21:27.23 | vondax | godsmoke: American PSTN you mean ...... |
| 21:27.33 | godsmoke | vondax: the only one I have experience with, yes |
| 21:27.39 | vondax | no such thing as ANI in most of the world ;-) |
| 21:27.57 | rene- | what is one to do with a sipura that gives no dialtone, does not respond to ping yet it says it has an IP valid in my network |
| 21:28.02 | godsmoke | ok -- well, I'm not meaning to address the world's population, vondax |
| 21:28.30 | vondax | but we are listening :-) |
| 21:28.39 | godsmoke | then continue to listen :) |
| 21:29.07 | Madkiss | Now calling in works, but when trying to call from the inside out, I again get "Feb 8 22:27:35 NOTICE[31346]: app_dial.c:749 dial_exec: Unable to create channel of type 'SIP'" -- this is getting me, really |
| 21:29.25 | ManxPower | Madkiss, PASTE the Dial line. |
| 21:30.12 | firestrm | yikes.. went to mpg123.de to download source.. this was in the news.. Yes! The project is not maintained at the moment and there are some serious security problems in the latest player versions. It is highly recommended to not use the source code you can download from this site. Check Debian Advisory or Gentoo Advisory for more information. Thanx. |
| 21:30.16 | Madkiss | -- Executing Dial("SIP/2000-6851", "SIP/0123456789@freenet-out|30|r") in new stack |
| 21:30.22 | firestrm | can i use mpg321 instead? |
| 21:30.25 | slak- | ManxPower, Any experience setting up mysql for asterisk cdr? |
| 21:30.34 | ManxPower | slak-, NEVER! |
| 21:30.48 | ManxPower | Madkiss, uh, the Dial line from extensions.conf. |
| 21:30.51 | slak- | lamer ;) |
| 21:31.16 | Madkiss | That would be "exten => _.,1,Dial(SIP/${EXTEN}@fwd-out,30,r)" |
| 21:31.32 | ManxPower | firestrm, "make mpg123" in the asterisk source firectory. |
| 21:31.46 | ManxPower | Madkiss, no it would be something @freenet-out |
| 21:31.57 | firestrm | ManxPower, thanks.. |
| 21:32.03 | Madkiss | oh, excuse me, I scrolled the wrong line |
| 21:32.03 | Madkiss | Right |
| 21:32.06 | *** join/#asterisk outsidefactor (barf@203-206-254-16.dyn.iinet.net.au) |
| 21:32.08 | Madkiss | exten => _.,1,Dial(SIP/${EXTEN}@freenet-out,30,r) |
| 21:32.12 | Madkiss | That's it |
| 21:32.24 | ManxPower | Madkiss, and you have a [freenet-out] in sip.conf? |
| 21:32.26 | tzanger | ugh |
| 21:32.31 | tzanger | my path to switch-1 today is sucking goatass |
| 21:32.36 | tessier_ | Does include => contextname recurse? |
| 21:32.40 | tzanger | usually it's switch-2 that's less good fo rme |
| 21:32.42 | ManxPower | Madkiss, never ever ever ever use _. |
| 21:32.44 | tessier_ | It looks like ti does which is causing me real problems. |
| 21:32.48 | Madkiss | [freenet-out] \ type=peer \ etc. pp. |
| 21:32.58 | Madkiss | ManxPower: What instead? |
| 21:33.01 | tessier_ | ManxPower: But what if you do want to match everything? |
| 21:33.08 | ManxPower | Madkiss, put the entry on pastebin.ca sans the password |
| 21:33.16 | ManxPower | tessier_, Then you are crazy as a loon. |
| 21:33.31 | tessier_ | ManxPower: For example I have a ton of DID's pointing at me, not all of which are in use. I would like to put a message saying the number is not in service. |
| 21:33.34 | ManxPower | Madkiss, How about a real pattern matching what you want to send out? |
| 21:33.40 | tessier_ | I could put in an entry for each and every DID... but that's 30,000. |
| 21:34.08 | firestrm | ManxPower, that just downloaded mpg123 from the very same site that had the caution/warning/danger in the news |
| 21:34.09 | ManxPower | tessier_, I use something like exten => _5047245XXX,1,Playback(go-away-lamer) |
| 21:34.11 | Madkiss | http://pastebin.ca/5495 |
| 21:34.32 | tessier_ | ManxPower: I have a bunch of different area codes coming in. |
| 21:34.55 | ManxPower | tessier_, I guess it still sucks to be you, but that still doesn't mean that _. won't cause problems. |
| 21:34.59 | firestrm | ManxPower, do i need to be worried about this security breach they talk about? |
| 21:35.07 | *** join/#asterisk LUTOR_ASI (~newbie@icable-docsis2-108.ibw.com.ni) |
| 21:35.18 | ManxPower | firestrm, I don't believe so. |
| 21:35.29 | ManxPower | Madkiss, I don't see anything specifically wrong. |
| 21:35.39 | Madkiss | ManxPower: Neither do I |
| 21:35.41 | harryvv | hi firestrm :) |
| 21:35.48 | ManxPower | Madkiss, other than the host is NOT dynamic. |
| 21:35.48 | BBRodriguez | Madkiss: you need to have "194.97.54.97 freenet.de" in your /etc/hosts |
| 21:35.51 | firestrm | hi harryvv |
| 21:35.51 | ManxPower | And that is your problem |
| 21:35.58 | Madkiss | BBRodriguez: I do |
| 21:36.05 | harryvv | firestrm what is the total round trip cost for avgas to von? |
| 21:36.10 | Madkiss | Well, if I change that entry, outgoing calls will stop to work |
| 21:36.16 | Madkiss | aeh, incoming calls, I mean. |
| 21:37.11 | BBRodriguez | Madkiss, change type to friend |
| 21:37.44 | Madkiss | I tried that, no effect |
| 21:37.53 | BBRodriguez | Madkiss, host = freenet.de in your sip.conf |
| 21:38.43 | firestrm | harryvv, im going to have to flight plan that one.. im going to guess at 300 gal |
| 21:39.05 | firestrm | harryvv, im guessing at 5 hours flight time |
| 21:40.25 | ionix | Hey guys |
| 21:40.41 | ionix | I have DIDs pointing to my asterisk box but when someone calls me, he doesn't hear the ringtone |
| 21:40.57 | ionix | any idea why asterisk wouldn't be signaling back the status? |
| 21:41.11 | ionix | DIDs are coming through a cisco switch to a nextone switch then to asterisk |
| 21:41.19 | ionix | it's not a zaptel device connected to t1. it's sip only |
| 21:42.16 | Mw3 | how can i 'link' a lots of analog extensions (about 100) to a pc (i'd like to use them with asterisk)? i mean with what kind of hardware |
| 21:42.20 | ionix | <PROTECTED> |
| 21:42.20 | ionix | <PROTECTED> |
| 21:43.01 | ManxPower | Mw3, Analog as in you have plugged them into an analog line from the phone company and used them? |
| 21:43.26 | *** join/#asterisk crazyhick (~crazyhick@cbn-24.182.134.130.cbnstl.net) |
| 21:43.26 | outtolunc | (4) 24port fxs channel banks = 96 analog ports |
| 21:43.43 | outtolunc | and a 4 port t1 digium card |
| 21:44.07 | Mw3 | ManxPower: yes, this is an existing analog setup but i'd like to integrate to asterisk :) |
| 21:44.27 | ManxPower | Mw3, then listen to outtolunc |
| 21:44.29 | Mw3 | outtolunc: hm, where can i find documentation about this 24 port fxs channel bank ? |
| 21:44.51 | ManxPower | Mw3, channel banks are a standard telephone device. |
| 21:45.07 | outtolunc | they are plentiful, just google 'fxo channel bank' |
| 21:45.07 | *** join/#asterisk devious (intrin@c68.112.146.203.stc.mn.charter.com) |
| 21:45.12 | outtolunc | er fxs |
| 21:45.15 | devious | will fbsd 5.3 run * ok? |
| 21:45.21 | devious | with zaptel and what not |
| 21:45.28 | slak- | mysql> CREATE TABLE cdr; |
| 21:45.28 | slak- | ERROR 1113: A table must have at least 1 column |
| 21:45.28 | slak- | ' |
| 21:45.29 | slak- | heh |
| 21:45.31 | ManxPower | devel, no |
| 21:45.33 | slak- | anyone know wtf |
| 21:45.42 | slak- | im connecting to an empty db cdrdb |
| 21:45.42 | tzanger | ugh |
| 21:45.45 | tzanger | peer1's sucking ASS today |
| 21:45.47 | devious | what do u suggest using? |
| 21:45.48 | slak- | i wanna create a new table |
| 21:45.52 | slak- | for cdr |
| 21:45.57 | Wireless | Manx, I find that I can forward all the incoming calls from the IAX provider to a fixed number (be it a local client or to an external service), but when the dialplan answers the calls first, the forwards to sip/iax won't work. |
| 21:46.03 | ManxPower | devel, Linux is the only supported OS for Asterisk |
| 21:46.56 | *** join/#asterisk p0lar (~p0lar@64.254.254.4) |
| 21:46.57 | Wireless | notransfers=yes doesn't help |
| 21:47.23 | Mw3 | oh, this is fucking expensive :) |
| 21:47.41 | outtolunc | hehe |
| 21:47.48 | ManxPower | Mw3, It's telephony. NOTHING is cheap. |
| 21:47.49 | outtolunc | 100 anything isn't cheap |
| 21:48.17 | Mw3 | i should consider changing the whole thing to voip :) |
| 21:48.19 | Mw3 | with softphones |
| 21:48.40 | ManxPower | ok. That's enough. I'm outta here. I'll be on #asterisk-stable |
| 21:48.42 | *** part/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 21:49.15 | *** join/#asterisk dslx (~jay@network-operations-center.dslx.net) |
| 21:50.49 | dslx | I have not been on IRC for many years so excuse me if I screw up protocol, can I just a topic? |
| 21:51.10 | firestrm | when i checkout asterisk from the cvs, am i grabbing stable or unstable? |
| 21:51.15 | crazyhick | has anyone had problems with retreiving a parked call? i placed include => parkedcall in my extension.conf but i can not retreive the call |
| 21:51.24 | crazyhick | i mean parkedcalls |
| 21:51.54 | dslx | listen on the transfer, the parking starts on the next extension |
| 21:52.36 | crazyhick | i did. lets say it announced 701. i try calling 701 from another extension i get a quick busy. |
| 21:53.07 | terrapen | ok. stupid question. i want to pass the extension that is calling to VoiceMailMain |
| 21:53.23 | terrapen | so that I don't have to authenticate when dialing voicemail from my own phone |
| 21:53.39 | terrapen | but i'm not seeing which variable i should be using |
| 21:53.51 | terrapen | VoiceMailMain(s${SOMETHING}) |
| 21:54.58 | *** join/#asterisk v-office (~acidtrux@amako.team-anomaly.org) |
| 21:55.18 | mrempire | Guys: FYI:: isdn /dev/ttyI* -- these devices are depriciated in the new kernel 2.6, i think capi will be the standard! |
| 21:56.07 | LUTOR_ASI | I have a problem, when i made a call from iax (nicaragua) to another iax(panama) server trying to dial to pstn my whole system crashes., anybody had have this problem before..?? |
| 21:56.59 | v-office | hello everyone |
| 21:57.12 | v-office | Can anyone tell me if Asterisk supports answer confirmation on non-Zap channels, like SIP? |
| 21:57.20 | v-office | I've searched the wiki and google to no avail |
| 21:57.40 | dslx | I am looking to support hundreds of companies using a single *. Is anyone doing this now? |
| 21:58.07 | terrapen | dslx, if you have the hardware, you probably could |
| 21:58.46 | v-office | dslx: what do you mean by 'support' ? |
| 21:58.48 | Wireless | can someone help with a iax bridging (i assume) issue? |
| 21:59.02 | dslx | I am running our office now and it has worked perfectly - I am looking to add VOIP services to our DSL service |
| 21:59.22 | v-office | dslx: you mean serve VoIP to customers? |
| 21:59.51 | terrapen | dslx, there are pages in the wiki about this |
| 22:00.47 | dslx | crazyhick, have you added the parkedcalls context to your extension.conf? |
| 22:01.37 | dslx | Yes, I provide DSL, and I would like to add a VOIP offering. I would need some type of GUI interface for each customer |
| 22:03.23 | FuRR_ | dslx: AMP ? |
| 22:03.27 | shido6 | no |
| 22:03.29 | shido6 | AMP is crap |
| 22:03.49 | shido6 | good idea but it has a long way to go, I will rephrase |
| 22:04.00 | dslx | I have looked at the wiki including redundancy. I do know know if I can move from a pbx setup to a VOIP provider using * |
| 22:04.45 | shido6 | moving to asterisk or using existing pbx with asterisk? |
| 22:06.00 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 22:06.59 | dslx | No. Our inhouse telephone system is *. I am very comfortable with it. I am concerned about offering VOIP service with * |
| 22:07.06 | brc_ | why |
| 22:07.22 | brc_ | what are you asking exactly? |
| 22:08.01 | dslx | The ability for users to make their own changes seem limited. WHen I want somthing special doen, I go into the configs and modify them. Has anyone come out with a good user GUI? |
| 22:08.14 | brc_ | ~seen kpflemming |
| 22:08.16 | jbot | brc_: i haven't seen 'kpflemming' |
| 22:08.25 | brc_ | dslx, nope, you'll have to write one |
| 22:08.31 | three55ml | dslx: I'm working on one if you want to see some screenshots |
| 22:08.47 | dslx | three, I do |
| 22:08.52 | brc_ | three55ml, I'd like to see them also ;) |
| 22:09.16 | brc_ | three55ml, webbased? |
| 22:09.22 | three55ml | Let me upload a new one, still work in progress |
| 22:09.23 | three55ml | Yeah |
| 22:09.52 | v-office | the beginnings of support proper support for Asterisk + database I think will enable or at least greatly ease what you're looking for |
| 22:10.18 | v-office | dslx: I've implemented similar configurations in the past where my customers were able to modify their sections of the dialplan, but it wasn't the prettiest solution ever |
| 22:10.26 | Luke-Jr | I suggest avoiding MyPhoneCompany if you can |
| 22:10.37 | dslx | ? |
| 22:10.59 | Luke-Jr | the unneccesary hassle to get it setup |
| 22:11.07 | dslx | ok |
| 22:11.09 | FuRR_ | three55ml: i would like to see it as well |
| 22:11.17 | Luke-Jr | sending them copies of the credit card |
| 22:11.19 | Luke-Jr | signing a form |
| 22:11.23 | Luke-Jr | etc |
| 22:12.27 | dslx | I guess I would like to assign a context to each company and give them the ability to setup extensions & such |
| 22:12.38 | v-office | dslx: I've done such a thing before, about a year ago |
| 22:12.50 | three55ml | Latest screenshot - http://www.spirestudios.com/misc/pbx_screenshot_02082004.jpg |
| 22:12.54 | v-office | dslx: but it involved a cron job to dump extensions.conf out of a mysql table and reload * |
| 22:12.59 | v-office | dslx: ugly, but it worked |
| 22:13.00 | tessier_ | ugh...I really need to find some good info on how the dialplan is parsed. |
| 22:13.16 | Luke-Jr | v-office: What is * SQL support for if not to bypass that? |
| 22:13.16 | tessier_ | Does it go in order, do context includes recurse, etc. |
| 22:13.29 | three55ml | Going to be a commercial service at first, then I'm going to release an open-source version. Currently the features are limited to avoid end-user confusion. |
| 22:13.41 | v-office | Luke-Jr: thats exactly what it's there for...but its only in CVS Head, and as far as I know, incomplete |
| 22:14.01 | *** join/#asterisk buddah (~hnic@67.110.253.129) |
| 22:14.04 | Luke-Jr | v-office: it seems to be in 1.0.5... |
| 22:14.12 | buddah | anyone know what would cause this message? using a cisco ata-186 |
| 22:14.13 | buddah | NOTICE[5968]: chan_sip.c:7807 handle_request: Unable to create/find channel |
| 22:14.14 | brc_ | three55ml, that looks very nice |
| 22:14.20 | three55ml | brc_: Thanks |
| 22:14.20 | brc_ | three55ml, what are you doing it in? |
| 22:14.26 | brc_ | language/framework |
| 22:14.34 | v-office | Luke-Jr: well perhaps I need to check that out. I'm talking from my experiences a year ago plus reading the wiki |
| 22:14.45 | three55ml | brc_: Mostly PHP. Kind of a custom framework we use on all of our web apps. |
| 22:14.49 | v-office | Luke-Jr: is it the Realtime Asterisk thats in 1.0.5? |
| 22:14.58 | Luke-Jr | v-office: no idea |
| 22:15.04 | brc_ | three55ml, awesome |
| 22:15.05 | Luke-Jr | v-office: I hate databases |
| 22:15.07 | dslx | three55ml, looks good! |
| 22:15.10 | v-office | Luke-Jr: where are you seeing references to database in 1.0.5? |
| 22:15.18 | Luke-Jr | v-office: ebuild USE flags, IIRC |
| 22:15.20 | eKo1 | 1.0.5 stable has realtime |
| 22:15.22 | three55ml | dslx: Thanks |
| 22:15.23 | brc_ | three55ml, are any of the other sidebar tabs functional yet? |
| 22:15.23 | eKo1 | ? |
| 22:15.30 | three55ml | brc_: All of them are |
| 22:15.40 | v-office | I think you are refering to the applications that do SQL calls |
| 22:15.41 | three55ml | brc_: There's an admin panel as well |
| 22:15.41 | v-office | not realtime |
| 22:15.50 | v-office | I really don't think realtime is in stable, it's too new |
| 22:15.53 | Luke-Jr | <eKo1> 1.0.5 stable has realtime |
| 22:16.06 | v-office | eKo1: are you stating or asking? |
| 22:16.18 | eKo1 | Hmm...didn't see any realtime stuff. |
| 22:16.31 | three55ml | eKo1: It's in asterisk-addons in CVS |
| 22:17.00 | dslx | three55ml - how to you interface with * |
| 22:17.29 | three55ml | dslx: Right now there's two methods, still determing the best. Either Realtime support in a database or dynamically generated config files. |
| 22:17.33 | three55ml | determining |
| 22:17.37 | buddah | on the sip debug i keep getting a SIP/2.0 407 Proxy Authentication Required, but we have no outbound proxy being used |
| 22:17.56 | eKo1 | buddah: I get that too. |
| 22:18.05 | buddah | yeah, 2 sec silence then fast busy |
| 22:18.09 | buddah | all i'm getting |
| 22:18.13 | eKo1 | Of course, I do have an outbound proxy. |
| 22:18.28 | buddah | then at the console i get the unable to create channel message |
| 22:18.43 | buddah | but same exact configuration works at the one here in our office |
| 22:18.59 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 22:19.07 | buddah | just this one in the field doesnt work |
| 22:19.15 | buddah | is that 407 stopping it? |
| 22:19.31 | terrapen | i wish the polycom didn't take so freakin long to reboot. |
| 22:19.35 | terrapen | this waiting sucks |
| 22:19.55 | *** join/#asterisk pablo1999 (~rcrane1@68.208.111.104) |
| 22:20.01 | *** part/#asterisk Bile_One (~TomSawyer@pcp03281999pcs.gillst01.ar.comcast.net) |
| 22:21.30 | pablo1999 | When I try to contact somebody at the office using the extension the person hears the phone ring but when they answer the phone they cant hear my voice. |
| 22:21.43 | pablo1999 | Any inputs? |
| 22:22.20 | three55ml | pablo1999: SIP or NAT? |
| 22:22.25 | three55ml | SIP or IAX, sorry |
| 22:22.36 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 22:22.50 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 22:22.54 | Sedorox | pablo1999: check the codecs being used |
| 22:22.55 | pablo1999 | SIP |
| 22:23.06 | BBRodriguez | hi everyone, is there a way to catch "wrong password on INVITE" in dialplan ? or more acurately, what would be the error code returned, when a sip channel fails to dial ? |
| 22:23.18 | *** join/#asterisk coolschooluk (~coolschoo@100.213.adsl.brightview.com) |
| 22:23.23 | three55ml | pablo1999: You behind a router or firewall? |
| 22:23.40 | *** part/#asterisk coolschooluk (~coolschoo@100.213.adsl.brightview.com) |
| 22:23.43 | pablo1999 | I'm behind a linksys router Wireless G |
| 22:23.44 | shido6 | u may have incorrect syntax in your dialplan |
| 22:24.00 | three55ml | pablo1999: Check to make sure port 5060 for SIP and 10,000-20,000 for RTP are forwarded. All UDP. And if you're using NAT, make sure "nat=yes" is set in your sip.conf. |
| 22:24.09 | v-office | Can anyone tell me if Asterisk supports answer confirmation on non-Zap channels, like SIP? |
| 22:25.52 | three55ml | pablo1999: That's probably it, make sure you have those settings I listed above |
| 22:26.01 | brc_ | you mean where you know when the channel's been answered? |
| 22:26.05 | *** join/#asterisk zetho (~zetho___@dkj211.neoplus.adsl.tpnet.pl) |
| 22:26.06 | v-office | brc_: yeah |
| 22:26.10 | brc_ | yes it does |
| 22:26.11 | v-office | but more than that |
| 22:26.21 | v-office | make it so the user can determine whether or not to take the call after answering |
| 22:26.28 | brc_ | ahh |
| 22:26.30 | v-office | IOW, hit # to accept |
| 22:26.31 | pablo1999 | the rtp ports that i'm showing on the conf for RTP are 16384 -> 16482 |
| 22:26.35 | brc_ | I can't recall |
| 22:26.40 | v-office | like zap does with Dial(Zap/c/101) |
| 22:26.41 | FuRR_ | v-office: wouldnt the user picking up the call negate the descion to answer it |
| 22:26.45 | v-office | but thats channel specific |
| 22:26.53 | v-office | FuRR_: not necessarily |
| 22:27.08 | v-office | the idea is to dodge voicemails from unwittingly accepting an incoming call (ie a cell phone) |
| 22:27.30 | FuRR_ | v-office: unless you do an IVR for the calling party to state their name, and then ring the phone, have an IVR to present to the calee the callers info and then a simple 1 to accept 2 to decline |
| 22:27.56 | v-office | FuRR_: hmm that might work, thanks |
| 22:28.06 | three55ml | pablo1999: http://www.voip-info.org/tiki-index.php?page=Asterisk%20firewall%20rules |
| 22:28.34 | three55ml | You could probably set it to just those ports, but 10-20,000 is pretty standard for RTP I believe |
| 22:29.32 | FuRR_ | v-office: always happy to promote laziness in the office |
| 22:29.54 | FuRR_ | anybody know where i can get different "voices" for festival |
| 22:30.49 | pablo1999 | ok, thanks |
| 22:31.21 | brc_ | FuRR_, use http://www.cepstral.com/ instead... |
| 22:31.31 | brc_ | sounds much better then festival |
| 22:31.36 | Beirdo | Urgent handler |
| 22:31.39 | *** join/#asterisk gtigene (~chatzilla@208.239.206.195) |
| 22:31.43 | brc_ | still sounds poor compared to ATT natural voices though |
| 22:31.45 | Beirdo | what is that supposed to be telling me? |
| 22:31.51 | three55ml | AT&T and amzing |
| 22:31.53 | brc_ | Beirdo, normal |
| 22:31.53 | three55ml | is amazing |
| 22:31.53 | FuRR_ | brc_: i miss "MacInTalk" it had some of the coolest voices |
| 22:32.02 | Beirdo | that's normal? :) |
| 22:32.05 | brc_ | Beirdo, do not run with -d |
| 22:32.08 | Beirdo | OK :) |
| 22:32.37 | Beirdo | well, I just unblocked IAX in my firewall, so I just wanted to make sure it wasn't something horrid |
| 22:32.45 | brc_ | yup |
| 22:32.46 | Beirdo | first tine, etc |
| 22:32.57 | Beirdo | first time rather... |
| 22:33.35 | gtigene | When I call Dial from my dial plan, can I give it an agent queue? (I looked at the wiki for Dial but it did not specify) |
| 22:33.42 | brc_ | eh? |
| 22:33.56 | brc_ | what are you trying to do |
| 22:34.11 | eKo1 | I just did a cvs checkout on asterisk-addons and I don't see any realtime stuff in there. |
| 22:34.21 | three55ml | eKo1: It's in asterisk-addons |
| 22:34.33 | three55ml | eKo1: "cvs co asterisk-addons" once you're logged in |
| 22:34.37 | Beirdo | my X100P clone arrived last night, so I have my work cut out for me now :) |
| 22:34.40 | eKo1 | Yeah, I just did that. |
| 22:34.50 | gtigene | I will have hunt groups (queues) and agents and distribute calles to them like in an old fashioned pbx. So I don't want to route the call to a specific channel but rather to a group of agents |
| 22:34.53 | three55ml | eKo1: It's res_config_mysql.c |
| 22:35.03 | harryvv | anyone here use iax.cc? |
| 22:35.05 | three55ml | Just do a make, then make install and it should be setup |
| 22:35.07 | brc_ | gtigene, use queues |
| 22:35.21 | gtigene | brc, I am listttening |
| 22:35.23 | eKo1 | Why is it called res_config_mysql? Is it only for mysql? |
| 22:35.46 | three55ml | eKo1: There's one with a standard ODBC driver, I think res_odbc_config.c or something |
| 22:35.50 | brc_ | gtigene, with roundrobin mode |
| 22:36.33 | eipi | brc_ did you tried ATT Natural Voices? |
| 22:36.48 | eKo1 | three55ml: Where? I don't see it. |
| 22:37.34 | three55ml | eKo1: Let me look real quick |
| 22:37.53 | brc_ | eipi, no, it's not available for asterisk |
| 22:38.09 | brc_ | eipi, you'd have to work out a (large) deal with att |
| 22:38.11 | brc_ | heh...heh... |
| 22:38.17 | gtigene | brc, thanks |
| 22:38.20 | brc_ | gtigene, make sense? |
| 22:38.33 | eipi | no, but did you tried the demo of ATT Natural Voice? it's perfect |
| 22:38.42 | brc_ | yup |
| 22:38.46 | three55ml | eKo1: Hmm, I don't see it either...look around the Wiki |
| 22:38.47 | brc_ | that's why I mentioned it |
| 22:38.51 | brc_ | it's very nice |
| 22:39.09 | gtigene | Brc, I am still in doubt, maybe I need to read the info again |
| 22:39.43 | brc_ | k...leme know if you don't get it working |
| 22:40.36 | gtigene | brc, what I am thinking about is the line I put in extensions.conf that says if the caller pressed 2 (for example) then I have to dial the first available agent in sales queue (for example) |
| 22:40.42 | slePP | this is a bad day. |
| 22:41.11 | brc_ | slePP, |
| 22:43.20 | tessier_ | How do people usually contract their contexts such that incoming calls do not have access to the outgoing lines? |
| 22:43.28 | FuRR_ | AT&T Natural Voices demo (http://www.research.att.com/projects/tts/demo.html) |
| 22:43.33 | dontmsgme | How much is a T1/Pri in/around los angeles |
| 22:44.03 | *** join/#asterisk inv_arp (~snara003@havana.aul.fiu.edu) |
| 22:44.11 | jets | tessier_: I usually make a [corporate] which has all of the extensions |
| 22:44.20 | jets | a [local] and an [ld] and a [incoming] |
| 22:44.26 | jets | I set my PRI's and linecards to incoming |
| 22:44.33 | tessier_ | incoming is where you put your DID's etc? |
| 22:44.43 | tessier_ | And then you put your phones into the local context? |
| 22:44.45 | jets | put the phones in corporate, and corporate includes local and ld. |
| 22:44.47 | terrapen | this damn polycom phone...had everything working fine, then i go to add a second line to it and break everything |
| 22:44.56 | terrapen | one of these days.... |
| 22:44.59 | inv_arp | hmm... need a cheap miami DID (just incoming) what sites offer good rates? |
| 22:45.07 | terrapen | nufone.net |
| 22:45.10 | terrapen | oh |
| 22:45.12 | terrapen | miami |
| 22:45.13 | terrapen | nm |
| 22:45.16 | terrapen | try voicepulse connect |
| 22:45.19 | shido6 | nice |
| 22:45.33 | jets | So DID's go in to incoming i make an alias for exten => _XXX,goto(corporate,1,${EXTEN}) |
| 22:45.33 | FuRR_ | anyone know if ViaVoice does TTS |
| 22:45.39 | gtigene | brc, thanks again |
| 22:45.47 | inv_arp | terrapen, ahh k thx |
| 22:45.50 | *** part/#asterisk gtigene (~chatzilla@208.239.206.195) |
| 22:46.02 | jets | in incoming if someone dials a three digit extension it sends them to the phones, otherwise [incoming] has no access to local/ld/corporate |
| 22:46.36 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 22:47.07 | eipi | furr_: naturalvoices.att.com/demos |
| 22:47.31 | Luke-Jr | Is there a chart w/ providers and the DIDs they suport? |
| 22:47.34 | FuRR_ | eipi: i know i just posted that up |
| 22:47.44 | FuRR_ | eipi: cept ATT wants some cash for their software |
| 22:47.50 | eipi | :) |
| 22:48.01 | FuRR_ | eipi: and i dont think it runs under linux |
| 22:48.13 | FuRR_ | ViaVoice however is essentailly free and has native linux support |
| 22:48.27 | FuRR_ | just wondering if it does TTS, as Ive only used it for STT |
| 22:49.09 | djin | How does ViaVoice STT perform? |
| 22:49.10 | *** join/#asterisk tull (~sdk@wwwcache2.livjm.ac.uk) |
| 22:49.23 | tull | Hello. |
| 22:49.27 | eipi | viavoice? 30k software? |
| 22:49.28 | eipi | :D |
| 22:49.42 | tull | does asterisk support sipura's SPA-3000? |
| 22:49.48 | *** join/#asterisk mrverizone (~mrverizon@pa-robinson1b-88.pit.adelphia.net) |
| 22:50.17 | three55ml | tull: Yes |
| 22:51.04 | mrverizone | hello |
| 22:51.18 | tull | three55ml thanks. |
| 22:51.35 | eipi | first time i see: http://freetts.sourceforge.net/docs/index.php |
| 22:51.53 | eipi | that was a free java tts |
| 22:51.55 | FuRR_ | eipi: just want something where i can change the voice for different applications |
| 22:52.14 | FuRR_ | eipi: if i use java i have to use something like FastAGI as java is pretty heavy on system resources |
| 22:52.28 | FuRR_ | FastAGI would allow me to put it on a diff box |
| 22:53.15 | jets | Is there a commercial voice recognition like sphinx that supports asterisk and works well? |
| 22:53.55 | *** join/#asterisk figfig (~figfig@h-68-164-90-141.snvacaid.dynamic.covad.net) |
| 22:55.05 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
| 22:55.56 | figfig | I am trying to set up my asterisk server so that when a call comes in from broadvoice, it forwards it to my cell phone. I have the line exten => s,1,Dial(SIP/<cell phone number>@broadvoice, 20) in the dialplan, but when I call in, I get a response 408, temporarily unavailable from broadvoice. Any idea what I am doing incorrectly? |
| 22:56.32 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-1056.nb.aliant.net) |
| 22:57.33 | Sedorox | so your trying to send it back out via broadvoice also? |
| 22:58.15 | figfig | yes |
| 22:58.17 | figfig | I am |
| 22:58.40 | figfig | Via other tests, I can originate multiple calls through broadvoice |
| 22:58.46 | figfig | so I doubt this is the issue. |
| 22:59.19 | Sedorox | I'm not farmilar with broadvoice.. so bare with me.. but I'm assuming you can dial nay number through them? |
| 22:59.35 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 22:59.48 | ta[i]nted | FuRR_ what is fastAGI? |
| 23:00.33 | figfig | sedorox: I can dial the same number elsewhere in the dialplan, just not in that line. |
| 23:01.25 | Sedorox | using the same form of <number>@broadvoice? |
| 23:02.27 | figfig | I beleive so (only slightly different- the exact line is exten => _1NXXNXXXXXX,1,dial(sip/${EXTEN}@broadvoice,60) |
| 23:02.40 | figfig | but ${EXTEN} is the same phone number. |
| 23:02.49 | Sedorox | yea |
| 23:02.52 | Sedorox | hmmm |
| 23:03.13 | Sedorox | maybe give it a wait time first? |
| 23:03.33 | figfig | I put a wait of 10 secs in right away, and still the same result. |
| 23:03.40 | Sedorox | hmmm |
| 23:03.46 | Sedorox | weird.. beats me... |
| 23:03.52 | figfig | I too am confused. |
| 23:04.16 | figfig | I double checked that I could have 2 simultaneous calls from the lan, and it worked. |
| 23:04.18 | Sedorox | does the box need to answer the line before it dials out again? |
| 23:04.43 | figfig | I don't think so |
| 23:04.46 | djin | any disadvantages with using SCCP instead of SIP (7920 doesn't support SIP)? |
| 23:05.05 | mrverizone | figfig do you have your call waiting working on hour server |
| 23:05.12 | figfig | I viewed with ethereal, and I don't think it responded to the first call until I had failed with the redirect. |
| 23:05.24 | denon | man |
| 23:05.28 | denon | someone tell me a stable build of * |
| 23:05.37 | denon | shit breakin on every build Ive tried |
| 23:05.45 | figfig | djin: I don't know what scpp is. |
| 23:05.57 | figfig | mrverizone: I don't understand? |
| 23:06.35 | mrverizone | figfig sorry,, so you have the call waiting feature working on you asterisk box with broad voice |
| 23:06.37 | terrapen | FINALLY.... |
| 23:06.47 | terrapen | i think i figured out Polycom provisioning |
| 23:06.50 | terrapen | what a bitch |
| 23:07.01 | figfig | I wonder if I put an answer in the dialplan. |
| 23:07.05 | mrverizone | what you are trying to do, could lines |
| 23:07.14 | terrapen | the config files that i found via the wiki did not steer me in the right direction |
| 23:07.35 | terrapen | i should add a wiki entry with instructions for how to properly and easily provision multiple polycom phones |
| 23:07.47 | mrverizone | what you are trying to do, could take up tow lines, or are you ports open, and not block with the incomming call, that you can not call back out |
| 23:08.17 | mrverizone | I am having a hard time typing on my knee with my laptop |
| 23:08.19 | mrverizone | sorry |
| 23:08.23 | figfig | mrverizone: how to I get call waiting working (why is it even pertinent?) |
| 23:08.25 | mrverizone | no 1 |
| 23:08.46 | mrverizone | call waiting show the ability to have two calls working on your line at one time |
| 23:08.57 | mrverizone | they are both incomming calls, |
| 23:09.13 | figfig | mrverizone: I can call two lines from two different phones. |
| 23:09.16 | mrverizone | you are trying to have an incomming call, and an out going call at the same time |
| 23:09.23 | figfig | mrverizone: one line is coming in, one is going out. |
| 23:09.41 | mrverizone | yeah, that is right, we have done up to 4 out going call at one time, for got about that |
| 23:09.49 | mrverizone | on one line |
| 23:09.58 | mrverizone | broad voice does not know how to stop that |
| 23:10.09 | figfig | I put the Answer line in, and it worked. |
| 23:10.12 | figfig | That was it. |
| 23:10.14 | djin | figfig, SCCP, better known as Skinny, is a native Cisco competitor of SIP. |
| 23:10.43 | harryvv | interesting |
| 23:10.52 | terrapen | http://www.jwz.org/gruntle/phones.html |
| 23:11.07 | figfig | thank you all for your info. |
| 23:11.08 | terrapen | i want to implement his ideal voicemail system |
| 23:11.28 | terrapen | press one putton and your voicemail messages just start playing, automatically deleting after they are played |
| 23:12.00 | *** join/#asterisk heath__ (~heath@12-215-32-191.client.mchsi.com) |
| 23:12.27 | *** part/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 23:12.42 | modulus_ | boy Asterisk::Manager sure does suck |
| 23:13.33 | modulus_ | "Odd number of elements in hash assignment at Asterisk/Manager.pm line 257, <GENO> line 7. |
| 23:15.28 | *** join/#asterisk mrproper_ (~mrproper_@61.95.55.242) |
| 23:16.16 | *** join/#asterisk abcbooze (abcbooze@marooned.us) |
| 23:16.19 | jets | <modulus_> Asterisk::Manager sucks |
| 23:16.23 | jets | Heh write your own :P |
| 23:17.32 | *** join/#asterisk Emrah (~root@hosting.eknw.com) |
| 23:17.39 | Emrah | Hello |
| 23:17.46 | Emrah | I think that I have an important bug |
| 23:17.58 | modulus_ | jets, i'm just fixing the existing one |
| 23:18.37 | jets | I just had to laugh because I saw you annoyed about an hour ago too |
| 23:18.44 | tzanger | stevekstevek: are you around? |
| 23:18.49 | Sedorox | Emrah: eh? |
| 23:18.53 | *** join/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
| 23:19.08 | modulus_ | jet's, yeah i thought i was calling the methods incorrectly |
| 23:19.16 | modulus_ | jets, but as it turns out, the module just sucks |
| 23:20.00 | Emrah | bout the security, or maybe not. I have an IAX connexion between server1 and server2. In server2, I have an extension like that exten => _97./30000,1,Dial(H323/${EXTEN:2}). With server1, I'm registered with the account 1000 over IAX2. But when I make in server1 SetCallerId(30000), in the extensions.conf, It seems that I'm the owner of the 30000 account. Is that normal? |
| 23:20.55 | Emrah | So, I can call extension 97888@server2... |
| 23:22.30 | Emrah | Is that normal? |
| 23:22.40 | inv_arp | hmm livevoip.com looks promising |
| 23:23.02 | three55ml | inv_arp: I have an account with them |
| 23:23.04 | Madkiss | haha, this is so goddamn crazy ... "host=freenet.de" in sip.conf makes me able to call to the outside but once I try to call asterisk, I get "The number is not available", if I turn it into "host=dynamic", I can dial in but not dial out |
| 23:23.13 | inv_arp | three55ml, works well? |
| 23:23.20 | three55ml | inv_arp: Only downside is you have to load each new DID with $29.95 worth of credits |
| 23:23.25 | three55ml | They're support can be a bit "snappy" too |
| 23:23.28 | three55ml | Their |
| 23:23.37 | inv_arp | three55ml, im getting the 6.99 plan |
| 23:24.17 | Luke-Jr | Anyone used Teliax? |
| 23:24.24 | inv_arp | three55ml, looks like 6.99 per month pay as you go... one call at a time tho |
| 23:24.30 | three55ml | Ah |
| 23:24.47 | three55ml | Where on their site is that? |
| 23:25.16 | Sedorox | By that: we mean no more than 1 call will be allowed in progress at one time. |
| 23:25.19 | inv_arp | three55ml, http://www.livevoip.com/index.php?subject=1&content=DIDSales |
| 23:25.22 | three55ml | Oh I see it |
| 23:25.29 | Dalion | Luke |
| 23:25.31 | Dalion | yep |
| 23:25.32 | Dalion | ;) |
| 23:26.22 | harryvv | If talking on line one of xlite and a second call comes in on line two and I hit line two will line one be on hold hearing moh ? i have it enabled. |
| 23:26.42 | inv_arp | three55ml, 6.99 is the cheapest i can find for a did in my area miami |
| 23:27.06 | toddf | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/c1573.html |
| 23:27.38 | toddf | uuugh |
| 23:27.38 | toddf | musiconhold requires a timing source |
| 23:27.38 | three55ml | inv_arp: They work great for me, but I'm using the IAX interconnect |
| 23:27.38 | harryvv | i have x100p installed and working. |
| 23:29.06 | inv_arp | three55ml, yea i prefer an iax provider b/c i run * behind nat |
| 23:29.10 | Dalion | .. |
| 23:30.47 | three55ml | inv_arp: Yeah. They work great so far |
| 23:30.57 | three55ml | You might want to check out VoicePulse Connect too |
| 23:31.42 | *** part/#asterisk Marouani (~ltostain@i01v-30-28.d4.club-internet.fr) |
| 23:31.55 | inv_arp | three55ml, ahh k |
| 23:32.45 | Sedorox | hmmmm |
| 23:32.46 | *** join/#asterisk pdracevich (~Administr@smtp.aucklandtax.co.nz) |
| 23:32.50 | pdracevich | moring all |
| 23:33.00 | godsmoke | good evening |
| 23:34.02 | Godsey | is there a way to take a polycom ip 300 back to MGCP after loading SIP firmware? |
| 23:34.13 | Godsey | do I just need to find mgcp firmware? :) |
| 23:34.23 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:34.35 | pdracevich | i need major help, I have a Asterisk unit on ip 218.101.54.x.x and another on 210.54.x.x and I want to able to connect them together please help :) |
| 23:34.59 | three55ml | Use IAX, go to http://www.voip-info.org and do a search |
| 23:36.28 | labo | glue |
| 23:36.35 | jets | IAX trunk |
| 23:36.39 | pdracevich | pdracevich: takes the rope and hangs himself |
| 23:37.01 | *** join/#asterisk r0d3nt|m (RatMan@64.60.114.35) |
| 23:37.07 | *** part/#asterisk r0d3nt|m (RatMan@64.60.114.35) |
| 23:37.20 | Godsey | I'm switching our hold queue over to shoretel unit |
| 23:40.22 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 23:43.33 | Madkiss | Just the last attempt today. When calling out, I see this: |
| 23:43.37 | Madkiss | Reliably Transmitting (no NAT): |
| 23:43.37 | Madkiss | SIP/2.0 404 Not Found |
| 23:44.02 | Madkiss | From: "Martin Loschwitz"<sip:2000@192.168.0.4>;tag=4099578350520438744 |
| 23:44.02 | Madkiss | To: <sip:0123456789@192.168.0.4>;tag=as088e32a8 |
| 23:44.07 | Madkiss | THat's apparently nonsense |
| 23:44.19 | Madkiss | or rather, obviously ... How do I fix that? |
| 23:45.33 | *** join/#asterisk wm501 (~wm501@ip70-177-80-170.ok.ok.cox.net) |
| 23:45.33 | harryvv | Are there any voipphone to bluetooth products on the market yet? |
| 23:45.38 | harryvv | bluetooth earpiece |
| 23:46.16 | godsmoke | you mean, a stationary voip phone ..? |
| 23:46.21 | godsmoke | with a bluetooth host adapter in it? |
| 23:46.30 | godsmoke | there's not much use for that |
| 23:46.35 | harryvv | wel, bluetooth i think is limited to 10 feet at the most. |
| 23:46.41 | wm501 | I have a wisip phone from pulver innovations, On my local wireless network i can connect to my PBX fine, make an recieve calls, but on other wireless network, i change the server ips in my phone to my ip address at home, and i forwarded port 5060 (UDP) through to my pbx server but It won't register |
| 23:46.46 | godsmoke | 10 meters -- but that's not my question |
| 23:46.47 | wm501 | do i need to forward 5060 tcp/ip through also? |
| 23:47.28 | godsmoke | harryvv: my point was -- why do you want to use a bluetooth headset with a desk phone? -- I can't imagine many people want to do that |
| 23:47.39 | mrproper_ | wm501: your going to have alot of trouble with sip behind a nat |
| 23:48.00 | *** join/#asterisk Blake0PS (BlakeOPS@c-24-245-24-84.mn.client2.attbi.com) |
| 23:48.02 | greg_work | any simply way to play .gsm files on the sound card? |
| 23:48.07 | greg_work | simple* |
| 23:48.13 | harryvv | bluetooth earpiece would rid of the cord between the wifivoip phone and earpiece is self. |
| 23:48.29 | harryvv | gred, yea see if you get a responce by entering the command play |
| 23:48.29 | godsmoke | see ... you didn't answer my question |
| 23:48.33 | drumkilla | you can get bluetooth adapters that plug into the headset jack |
| 23:48.35 | godsmoke | you're talking about a WIFI voip phone |
| 23:48.36 | harryvv | greg_work that is |
| 23:48.39 | drumkilla | and then use a headset to that |
| 23:48.40 | tzanger | applied patches in 2532 and 3400... let's see if this works better |
| 23:48.48 | Sedorox | my school's tech support has a few of these |
| 23:48.49 | Sedorox | http://plantronics.com/north_america/en_US/products/cat2420498/cat2420498/prod5210004 |
| 23:48.51 | drumkilla | but all of that will bring you up to about $180 |
| 23:48.52 | greg_work | harryvv: in * console? nope |
| 23:48.53 | Sedorox | can be used with any normal phone |
| 23:49.01 | Sedorox | the hting.. all are over $300 |
| 23:49.12 | *** join/#asterisk florz (nobody@odnb-d9baa5e6.pool.mediaWays.net) |
| 23:49.31 | harryvv | greg then you dont have sox installed |
| 23:49.45 | harryvv | apt-get install sox if your running debian. |
| 23:49.53 | greg_work | i do have sox installed |
| 23:49.57 | harryvv | mmm |
| 23:50.05 | harryvv | man sox then |
| 23:50.06 | harryvv | ;) |
| 23:51.19 | greg_work | oh, its in a shell. |
| 23:51.34 | harryvv | of course |
| 23:51.56 | harryvv | play file.gsm -r 8000 |
| 23:52.03 | Sedorox | http://staples.com/Catalog/Browse/sku.asp?PageType=1&Sku=577467&bcFlag=True&bcSCatId=3&bcSCatName=Technology&bcCatId=14&bcCatName=Telephones+%26+2%2DWay+Radios&bcClassId=142842&bcClassName=Internet+Phone |
| 23:52.09 | Sedorox | wish that wasn't vonage locked... |
| 23:52.29 | harryvv | btw, how do vonage and other lock there ATAs? |
| 23:52.47 | labo | the pap2 ? |
| 23:52.52 | FuRR_ | is there currently any support in * for speech recognition |
| 23:52.58 | Blake0PS | what does the 's' mean here? exten => 700,1,Goto(default,s,1) |
| 23:53.16 | Sedorox | si |
| 23:53.24 | FuRR_ | harryvv: Cisco offered in the 2.1 series of the ATA186 an option to lock the ATA so it couldnt be reset to ward off the ebay factor |
| 23:53.43 | harryvv | furr, what is involved in locking one? |
| 23:53.44 | labo | same priority |
| 23:53.47 | *** part/#asterisk zetho (~zetho___@dkj211.neoplus.adsl.tpnet.pl) |
| 23:53.50 | Sedorox | Blake0PS: goto the s extention in the default context |
| 23:53.56 | labo | the pap2N isn't locked. |
| 23:54.13 | FuRR_ | harryvv: its a software issue |
| 23:54.17 | labo | its only for isps, but you can get it on ebay, cheap. |
| 23:54.22 | *** join/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net) |
| 23:54.31 | Sedorox | the pap2N? |
| 23:54.33 | harryvv | furr, so it requires a password to unlock it then. |
| 23:54.37 | jdv79 | are there any docs on how to program an IAXY? |
| 23:54.44 | FuRR_ | harryvv: no, once its locked its locked |
| 23:54.51 | jdv79 | i got iaxyprov but no docs |
| 23:55.05 | jdv79 | i setup a config file...nothing |
| 23:55.23 | harryvv | interesting. |
| 23:55.23 | Sedorox | I can't find one on ebay that doesn't seem to be locked |
| 23:55.54 | harryvv | I would like to know how thay are locked ;) |
| 23:56.01 | jdv79 | there's got to be a doc somewhere |
| 23:56.18 | labo | buy a spa, same thing. |
| 23:56.27 | Sedorox | call up cisco and tell them your a VOIP provider and want to lock your ata's down :-p |
| 23:56.29 | Sedorox | labo: $$ :( |
| 23:56.39 | FuRR_ | harryvv: knowing cisco, if you crack one open there should be a battery inside -- or a reset pin |
| 23:56.54 | harryvv | to reset the eeprom |
| 23:57.07 | FuRR_ | harryvv: yeah, or a JTAG header that you can solder on a db9 to |
| 23:57.07 | terrapen | anybody want to buy a Cisco 7960, new in box? |
| 23:57.19 | harryvv | why whats wrong with it? |
| 23:57.30 | terrapen | nothing. i just decided that i want a polycom |
| 23:57.38 | terrapen | the cisco firmware upgrade thing kind of scares me |
| 23:57.42 | terrapen | im not really sure what to do |
| 23:57.47 | Sedorox | how much? |
| 23:58.11 | terrapen | i paid $299 a week ago...i'll sell it for $299 w/ free fedex |
| 23:58.17 | labo | they kinda screwed it up when packaging the latest POS.XXX |
| 23:58.42 | Sedorox | hmmm |
| 23:58.57 | terrapen | i *should* figure out how to get the latest SIP firmware on there |
| 23:59.04 | terrapen | but im not really sure where to start |
| 23:59.09 | terrapen | and i got my IP500 working so nicely |
| 23:59.11 | *** join/#asterisk MrClean (~seabrook@store-fw.porchlight.ca) |
| 23:59.14 | labo | i can help you |
| 23:59.20 | terrapen | ok |
