00:00.39 | Dr-Linux | Skid: all i mean to ask, i wanna do same, like i wanna allocate a few numbers to my client, with IP and port |
00:00.46 | Dr-Linux | as my provider did |
00:00.57 | Skid | I don't know really.. |
00:01.01 | Skid | sorry |
00:01.07 | Skid | ask him? :) |
00:01.38 | Dr-Linux | he doesn't tell me |
00:01.58 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
00:02.23 | Dr-Linux | Skid: and i was not given any username or password :S |
00:02.42 | Skid | i really dont know to tell the truth |
00:03.26 | Dr-Linux | hhm... :S |
00:06.06 | Grizzy | I wish either asterisk regular expressions were unix filename or unix or perl regular expressions, OR that I could find a complete spec on them. |
00:07.52 | Skid | http://www.voip-info.org/wiki/view/Asterisk+Expressions ? |
00:07.54 | Skid | no good? |
00:08.39 | Grizzy | I don't see it on that page |
00:10.46 | Grizzy | there's a bit of a blurb on p 27 of the handbook, but I don't think it's complete (4.1.4) |
00:10.56 | jbalcomb | i've just set up Asterisk. when i pickup the phone i get a dial tone, i dial 500 for the demo and the line stays up but i don't hear anything. what should i be looking for? |
00:13.09 | Grizzy | Thanks for hints, Skid. |
00:17.17 | [av]bani | whats this thing cisco calls "ghost digits" ? |
00:19.45 | Jon335 | I have the same question as lunaphyte: What is considered a good rate for a Toll-Free DID? |
00:19.59 | *** join/#asterisk iq|mobile (n=iq@71-214-5-20.omah.qwest.net) |
00:20.15 | AlexCTI | There someone familiar with pri_net? |
00:27.36 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
00:33.23 | Primer | I simply cannot get this cisco ata to get its sip settings from the file it sets over tftp |
00:33.26 | Primer | any of its settings |
00:34.47 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
00:35.07 | *** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
00:37.21 | *** join/#asterisk malverian (n=malveria@adsl-065-005-207-210.sip.gnv.bellsouth.net) |
00:42.03 | *** join/#asterisk _Simon (n=IRC@i216-58-40-193.cybersurf.com) |
00:42.38 | *** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
00:42.57 | _Simon | hey gang. I'm using IAX2 softphone and hardphone. Even using echo test as a test case, I have to call 3-4 times for it to work. is this an IAX2 NAT issue? Why does it randomly pick up but many times not? Anyone know how to resolve this? |
00:43.29 | *** part/#asterisk Priam (i=mike@towerravens.com) |
00:46.17 | *** join/#asterisk rogercharlie (n=rogercha@c-69-181-20-122.hsd1.ca.comcast.net) |
00:50.28 | *** join/#asterisk pigpen2 (n=mark@66.118.8.74) |
00:50.28 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
00:51.43 | Jon335 | What is considered a good rate for a Toll-Free DID? |
00:51.52 | pigpen2 | Hi all. Quick question: Using asterisk 1.2.4 with Polycom 601's - How can I disable the notification "beep" when a second call comes in? This is kinda a pain when receiving credit card numbers.... |
00:52.40 | pigpen2 | Jon335, Nuphone was doing it for .02/min.... |
00:52.48 | pigpen2 | Last time I looked. |
00:53.28 | _Simon | can anyone help with my IAX2 issue please? just looking for some ideas. I thought it was my soft phone but the hard phone is doing the same |
00:53.35 | _Simon | would it be the PBX or the network itself? |
00:53.53 | *** join/#asterisk hfern (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net) |
00:54.13 | pigpen2 | I am quite busy, but I might be able to help. |
00:54.14 | Jon335 | pigpen2: What about a Canadian Toll-Free DID? |
00:54.23 | Qwell | Jon335: about 10x that much |
00:54.31 | pigpen2 | heh...I have heard Canada is much higher. |
00:55.04 | _Simon | pigpen2: were you speaking to me or Jon, sorry just don't wanna get confused lol |
00:55.13 | pigpen2 | _Simon, well, your resolution may be several things...but a hint of what is going wrong is a good idea.... |
00:55.19 | pigpen2 | you. |
00:55.27 | Jon335 | pigpen2: Unlimitel is 4c/min, I think that is the best |
00:55.32 | _Simon | sure, I mentioned it earlier, basically what is happening is: |
00:55.39 | _Simon | I'm using IAX2 softphone and hardphone. Even using echo test as a test case, I have to call 3-4 times for it to work. is this an IAX2 NAT issue? Why does it randomly pick up but many times not? Anyone know how to resolve this? |
00:56.00 | *** join/#asterisk angom_h (n=angom@red-corp-200.79.145.199.telnor.net) |
00:56.00 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
00:56.01 | _Simon | most of the time the call doesn't even register on the pbx console |
00:56.01 | pigpen2 | well, nat usually doesn't affect iax.... |
00:56.08 | _Simon | pigpen2: yeah thats what I thought |
00:56.17 | pigpen2 | having packet loss? |
00:56.20 | Heimidal | what codec is ideal? |
00:56.20 | _Simon | but if the asterisk console doesn't output anything |
00:56.21 | pigpen2 | high latency? |
00:56.22 | *** join/#asterisk jahani (n=k@adsl-240-241-192-81.adsl2.iam.net.ma) |
00:56.28 | _Simon | shouldn't be high latency |
00:56.35 | pigpen2 | Heimidal, depends...how much bandwidth. |
00:56.49 | pigpen2 | well, if it is so high it doesn't get there.... |
00:57.06 | Heimidal | I'd like to be able to have 4 channels on half a T1 |
00:57.10 | _Simon | it should get there no prob |
00:57.24 | _Simon | its on a server external of my lan, but the server is on the same ISP as me |
00:57.27 | _Simon | so should be pretty fast |
00:58.28 | _Simon | I also set the IAX2 port as high QoS on the router |
00:58.59 | pigpen2 | well, like you said, if the asterisk cli never shows a connection...then...well...no call... |
00:59.07 | pigpen2 | the packets may not be getting there. |
00:59.15 | _Simon | pigpen2: when the call *does* work the audio is fine with no interruption |
00:59.25 | _Simon | so I would assume the connection is working well if that is the case |
00:59.38 | pigpen2 | qualify set? |
00:59.48 | pigpen2 | qualify=yes or qualify=500 ? |
00:59.48 | _Simon | umm whats that? lol |
00:59.59 | pigpen2 | check it out in the wiki... |
01:00.04 | _Simon | k |
01:00.06 | pigpen2 | but just stick it in the iax config. |
01:00.28 | pigpen2 | this way your server keeps up with the device....with regular check in's (the way I understand it) |
01:00.31 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
01:00.57 | Skid | ho hum |
01:01.00 | Skid | any dev's around |
01:01.05 | Skid | I think i've found a bug in 1.2.5 |
01:01.06 | Qwell | Skid: always |
01:01.18 | pigpen2 | crap...it is Qwell.... |
01:02.28 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
01:02.59 | pigpen2 | Qwell, can I ask you a quickie? |
01:03.03 | _Simon | bandwidth=low |
01:03.03 | _Simon | tos=lowdelay |
01:03.06 | _Simon | are those okay? |
01:03.43 | Qwell | pigpen2: just ask, if I know the answer, I'll tell you |
01:03.55 | Qwell | (if it isn't too involved, that is) |
01:04.07 | pigpen2 | <PROTECTED> |
01:04.17 | pigpen2 | Other than that...everything is great. |
01:05.23 | Skid | it seems mixmontior has some issues already too |
01:05.25 | Skid | doh |
01:05.32 | Skid | i like that app :) |
01:05.38 | Qwell | pigpen: I didn't know the answer 10 minutes ago when you asked, and I still don't |
01:05.45 | Qwell | If somebody knows, they'll answer... |
01:05.50 | pigpen2 | thank...me too... |
01:05.57 | pigpen2 | sorry..I didn't know you were watching. |
01:06.05 | pigpen2 | thanks. |
01:06.06 | Qwell | I'm always watching |
01:06.16 | pigpen2 | good to know. |
01:06.21 | Grizzy | the eye of god. : o ) |
01:06.46 | Qwell | I lurk - it's what I do |
01:07.07 | Grizzy | What chat client do you use, Qwell? |
01:07.18 | Qwell | /ctcp version me |
01:07.34 | Skid | ho hum, so can anyone see a problem with exten => s,9,MixMonitor(${DATETIME}) |
01:07.41 | Skid | (basically kills the cLI) |
01:07.59 | Skid | call's been answered and is in a dialplan, basically |
01:08.22 | Skid | before i go submitting a bug which may not be a bug |
01:08.29 | Skid | doesn't do it if i just mixmonitor(recording.wav) say |
01:08.37 | Skid | but does with the datetime var |
01:08.39 | Grizzy | qwell - &*&%^& GAIM, I can't figure out how to CTCP Version. |
01:08.54 | Skid | 01:09 [freenode] CTCP VERSION reply from Qwell: xchat 2.4.5 Linux 2.6.15-gentoo-r4 [x86_64/2.22GHz/SMP] |
01:08.57 | Skid | :) |
01:09.05 | *** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br) |
01:10.07 | Grizzy | It beats 6 open chat processes (to go with my 3 sipphones), but it's pretty incomplete. |
01:10.50 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
01:12.40 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
01:16.14 | Splat | if I set a sipura/linksys spa3000 to use asterisk as an outbound proxy server does that mean it will ignore the dialplan settings it has and just go according to asterisk's dialplan? |
01:16.34 | tehdely | eric_hill: pridialplan=unknown isn't helping |
01:16.40 | tehdely | i set the span on intense debug |
01:16.44 | tehdely | and i'mseeing absolutely nothing |
01:16.48 | tehdely | when an inbound call is coming |
01:16.59 | tehdely | something must be misconfigured on the switch |
01:17.07 | tehdely | i have a feeling this ball is in the CLEC's court |
01:17.13 | tehdely | unless there's something else i could have overlooked |
01:17.30 | tehdely | he claims asterisk is returning an error on INVITE but why wouldn't i at least see evidence of the failed INVITE |
01:18.13 | Grizzy | I love the phone company so much. (NOT) |
01:18.56 | *** join/#asterisk hfern_ (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net) |
01:20.29 | tehdely | does anyone see any evidence of a failed incoming call in this debug output: http://pastebin.com/621008 |
01:20.39 | tehdely | note that outgoing calls over this span are working perfectly |
01:20.44 | tehdely | but incoming ones seem to go straight to /dev/null |
01:20.47 | tehdely | (caller hears a busy signal) |
01:27.29 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
01:30.18 | *** join/#asterisk hfern (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net) |
01:32.53 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net) |
01:33.04 | *** join/#asterisk juice (n=juice@mo-69-68-106-145.dyn.sprint-hsd.net) |
01:38.29 | rogercharlie | does the flash proxy work for all manager connections or just flash clients? |
01:39.23 | *** join/#asterisk riddlebox (n=james@24-207-158-49.dhcp.stls.mo.charter.com) |
01:40.21 | *** join/#asterisk websae2k (i=websae@CPE-24-167-204-30.wi.res.rr.com) |
01:40.37 | rogercharlie | so many fun things use the manager API |
01:41.31 | websae2k | anyone need carrier services :)? |
01:41.57 | Strom_C | websae2k, please stop spamming the channel |
01:42.17 | websae2k | just trying to help a fellow asterisk user(s) |
01:42.39 | Strom_C | put your ad on the voip-info wiki |
01:42.43 | Strom_C | don't spam the IRC channel |
01:43.21 | Heimidal | for what reason would 's' not match for a given number? I keep getting "1112223333@demo doesn't exist" when trying to use the demo context |
01:43.34 | Heimidal | (replace 1112223333 with my real number, of course |
01:43.36 | Qwell | Heimidal: When the number isn't s |
01:43.36 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
01:43.51 | Heimidal | Qwell: isn't s supposed to mean "start"? |
01:43.54 | Qwell | yes |
01:44.04 | Qwell | If you want to match your number, put in a pattern that matches it |
01:44.10 | Abydos313 | evening everyone |
01:44.15 | Heimidal | well, I don't want to match my number |
01:44.19 | Heimidal | it just keeps saying that |
01:44.27 | CrashHD | hey fella's. I have two * boxes setup for trunking, which always fail to natively bridge a call. is there a way to disable this functionality? |
01:44.28 | Qwell | You obviously do want to match your number |
01:44.29 | Heimidal | I'm just wanting to call in and have it go to the demo |
01:44.57 | *** join/#asterisk Lino` (n=Lino@i577BDE76.versanet.de) |
01:45.12 | Heimidal | ok, I don't want it to match my number explicityluy |
01:45.17 | Heimidal | *explicitly |
01:45.24 | Qwell | What type of line? |
01:45.55 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
01:46.13 | Heimidal | it's configured to register with Teliax, if that's what you mean |
01:46.17 | Heimidal | I'm a bit new to this :) |
01:49.05 | Lino` | ~seen Possible |
01:49.09 | jbot | possible <n=Babbel@23.255-136-217.adsl-fix.skynet.be> was last seen on IRC in channel #asterisk, 14d 13h 29m 48s ago, saying: 'I guess not'. |
01:49.15 | Lino` | hm |
01:49.29 | Lino` | cisco systems is the worst buerocracy ever. |
01:50.01 | Lino` | in order to buy a snt contract i need to send in an invoice, the license, the serial number and a photo of the hardware (!) wtf? |
01:50.29 | *** join/#asterisk Strom_C (n=strom@66.159.243.59) |
01:54.24 | CrashHD | what dial option do you send to disable native bridging? |
01:58.26 | rogercharlie | Does the FOP work as a manager proxy for programs other than FOP? |
01:58.55 | rogercharlie | I am having no luck sending things to the 4445 FOP port instead of the 5038 port |
02:00.27 | Lino` | hmmm |
02:00.39 | Lino` | rogercharlie: if it would work it would be pretty insecure huh? |
02:01.51 | rogercharlie | well locking down and configuring would be next |
02:03.20 | Lino` | no |
02:03.27 | Lino` | i mean like if you could use FOP server as a proxy |
02:03.44 | Lino` | then "almost everybody" could play the manager for you ;) |
02:03.58 | rogercharlie | well under manager proxies it is listed |
02:04.06 | Lino` | "hello i'm another dirty little hacker and i'll be your manager for today *transfers call*" |
02:04.26 | Lino` | quite a funny thought *g* |
02:04.37 | rogercharlie | yah it would be easier than getting a smarnet contract |
02:04.48 | Lino` | oh i hate cisco. |
02:05.21 | Lino` | im trying to give money to them. i asked several resellers if they could sell the contract |
02:05.46 | *** join/#asterisk trbldwine (i=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
02:05.58 | Lino` | result: one berlin based reseller laughs about me, they printed out my request email framed it and put it on the wall |
02:06.09 | rogercharlie | the contract is under 10 so no one cares |
02:06.10 | Lino` | just because i sent an email 3 oclock in the morning |
02:06.26 | Lino` | yeah but i have 50 phones |
02:06.26 | Lino` | :D |
02:06.45 | Lino` | and i have a customer who has another 100 and no contract |
02:07.13 | Lino` | what i did now is i ordered one single contract from an american reseller |
02:07.31 | Lino` | 6$ for the contract, 66$ for the shipping |
02:07.38 | *** join/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
02:07.50 | Lino` | if its the right contract i'll buy the amount needed, if not then i wont |
02:07.51 | Lino` | :-P |
02:08.13 | rogercharlie | at least the phones are nice |
02:08.41 | rogercharlie | besides no deny call button, and a hidden DND button, and no presence, they rock |
02:08.56 | Qwell | sorry, no presence? |
02:09.00 | blitzrage | Qwell: ! |
02:10.07 | Lino` | yeah |
02:10.14 | Lino` | if you dont have 7961 or 7970 |
02:10.28 | Lino` | the bare 7960 does not have those lit line buttons |
02:10.36 | Qwell | Lino`: nor do they need them |
02:10.44 | Lino` | but with my 7970ies and 7961s the hint stuff works |
02:11.00 | Lino` | maybe its because the 7960ies are SIP |
02:11.07 | Lino` | and the 7961 7970 are sccp |
02:11.09 | Lino` | :-P |
02:11.23 | rogercharlie | yahsers no SIP love |
02:11.37 | Strom_C | there's apparently now SIP firmware for the 7970 |
02:11.41 | Lino` | i prefer sccp, i have several 7961, 7970 and 7920 with it |
02:11.54 | Lino` | but sccp is better than sip, why downgrade? |
02:12.08 | Lino` | (ok you'll need an extra module because skinny sucks) |
02:12.17 | Qwell | Lino`: I'm working on that |
02:12.24 | *** join/#asterisk ravenpi (n=chatzill@londonderry-cuda3-24-51-50-156.lndnnh.adelphia.net) |
02:12.29 | Lino` | on what? @ Qwell |
02:12.47 | Qwell | chan_skinny |
02:12.58 | Qwell | today, I got it to actually like...do stuff :P |
02:13.42 | Strom_C | heh |
02:13.43 | Lino` | yeah, i hate it |
02:13.43 | rogercharlie | I miss critch on the mailing list |
02:13.43 | Lino` | when i try to connect a 7920 to chan_skinny |
02:13.43 | Qwell | well, Sergio is a prick, so...yeah. Fuck chan_sccp |
02:13.43 | Lino` | my asterisk crashes |
02:13.43 | Lino` | let him be a prick, let him be something different, as long as the software works ;) |
02:13.45 | Qwell | Lino`: http://bugs.digium.com/view.php?id=6772 |
02:14.31 | Lino` | thats what a prick has to look like |
02:14.39 | Strom_C | that would somehow imply that ballmer has style to begin with |
02:14.42 | Qwell | Lino`: That'll get * to not crash. It still won't like...work, but... |
02:15.06 | Qwell | Lino`: If somebody were to donate a 7920 to the cause, I'd be sure to get them working too. :P |
02:15.41 | Lino` | hey ;) |
02:15.52 | Qwell | and actually, I am serious... |
02:16.15 | Qwell | I need to get/borrow various sccp phones, to test them all out. See if there are any quirks |
02:16.23 | Lino` | where are you from? |
02:16.28 | Qwell | southern CA |
02:16.38 | Lino` | kk |
02:16.47 | Strom_C | Qwell, I'd donate, but all I've got are 7960s :) |
02:16.52 | Lino` | too far away to throw a 7920 |
02:16.55 | Qwell | Strom_C: yeah, I've got 7960's to test with |
02:17.03 | Qwell | That's all I've got though |
02:17.04 | X-Rob | I don't! |
02:17.15 | Lino` | lol |
02:17.20 | Qwell | 7941/7961 would be nice |
02:17.31 | Strom_C | I forget Qwell - where are you again? San Diego? |
02:17.32 | Nugget | it's all qwell's fault |
02:17.35 | Qwell | and some of the cheap stuff, like 1912 |
02:17.37 | Qwell | Nugget: ;) |
02:17.40 | Lino` | today i had to make a screenshot of my working asterisk test installation for a new firmware version... but how to make a screenshot of the console? |
02:17.51 | Qwell | Strom_C: about 15 miles from downtown LA |
02:17.51 | Lino` | 7912 |
02:17.57 | Qwell | right |
02:18.04 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
02:18.05 | Strom_C | Qwell, oh sweet. we should have lunch sometime ;) |
02:18.05 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
02:18.09 | Lino` | cheap stuff is the exact word for it |
02:18.22 | Qwell | and a 7985... |
02:18.26 | Lino` | huh |
02:18.29 | Lino` | that conference thingy? |
02:18.34 | Qwell | If somebody gives me a 7985, I'll get video working, for sure. :P |
02:18.39 | Lino` | no video |
02:18.47 | Qwell | yes video |
02:18.52 | Lino` | ah that video conference stuff |
02:19.01 | Lino` | its ugly |
02:19.01 | Qwell | no, just a video phone |
02:19.05 | Strom_C | Videphones are the wave of the future! </1971> |
02:19.08 | Lino` | i dont want it. |
02:19.40 | Lino` | oh i remember, when I signed a contract with "Deutsche Bundespost" back in 14 years ago for a single ISDN line (BRI) |
02:19.53 | Lino` | they were like buy a videophone, in 5 years everybody will have one |
02:20.16 | Nugget | heh |
02:20.26 | Lino` | today the ISDN standard has changed, the videophones sold back then are useless and they were fricken expensive. |
02:20.45 | Lino` | thank god i didnt buy those ;) |
02:21.11 | Strom_C | You know what's totally mad? I have three telephones on my desk and I'm considering moving stuff to a different desk so I can put more telephones on my desk |
02:21.29 | Qwell | Strom_C: You need at least 6 |
02:21.45 | Qwell | with no fewer than 2 in use at any given time |
02:21.52 | Strom_C | I already have fourteen line appearances within arm's reach |
02:22.20 | *** join/#asterisk trbldwine (i=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
02:23.49 | Lino` | lol |
02:24.26 | Lino` | i'm happy with 2*7960, 1*7970, 1*Ascom Eurit 40 and 2 non working 7920 on my desk |
02:24.42 | Lino` | and a cd burning robot which makes crummy noises |
02:25.04 | Qwell | Lino`: send me a 7920, I'll get it working. ;) |
02:25.15 | Qwell | ..eventually |
02:25.21 | Lino` | well, actually |
02:25.35 | Strom_C | ive got 2 7960s, a Nortel 9417CW, a turquoise Western Electric 2500, and a bright orange rotary Stromberg-Carlson Slenderet hanging around my desk |
02:25.43 | Lino` | i'll try to get a working call manager ;) |
02:25.52 | Lino` | nortel |
02:25.55 | Lino` | meridian? |
02:26.03 | Strom_C | Lino`, it's an analog desk set |
02:26.21 | Lino` | ah ok |
02:26.50 | Lino` | i spent new years eve in a datacenter of a big bank 1000km from here, they had those really huge nortel phones |
02:26.54 | Lino` | butt ugly |
02:27.04 | Strom_C | which "really huge nortel phones"? |
02:27.40 | Lino` | i am looking for the number |
02:27.53 | Lino` | http://paragonnt.com/products/MVC-017L.jpg <= those with a display extension right next to it |
02:27.56 | bugz | i like my ip601 |
02:27.59 | Lino` | like i said, butt ugly |
02:28.04 | bugz | i get to watch everyone |
02:28.33 | Strom_C | Lino`, that's not so bad |
02:28.34 | Lino` | ip601? |
02:28.46 | bugz | with an expansion module |
02:28.50 | Heimidal | can someone tell me why this causes the error "Mar 24 20:28:23 NOTICE[4193]: chan_iax2.c:7198 socket_read: Rejected connect attempt from 207.174.202.3, request '312546XXXX@incoming' does not exist"? http://rafb.net/paste/results/RceagI47.html |
02:29.19 | Lino` | well they are switching to cisco throwing away (or maybe selling) 2500 phones. |
02:29.32 | Lino` | every single one is this butt ugly phone with extension |
02:29.51 | bugz | haha i got some nasty phones off a russian oil rig |
02:29.56 | bugz | these things were like 30 years old |
02:29.57 | Lino` | huh? |
02:30.01 | Lino` | russian phones o_O |
02:30.06 | Lino` | 2 line interface? |
02:30.07 | bugz | yeah, get this |
02:30.13 | Lino` | 2 wire |
02:30.23 | bugz | they had this box marked KGB on the deck in the electrical closet |
02:30.31 | Lino` | lol |
02:30.56 | bugz | it was made of this WWII loking metal box with all these steel conduits and stainless steel buttons |
02:31.04 | Lino` | http://www.polycom.com/pw_files/SP_IP601_3ExpMod_Left.gif <= you mean this phone? |
02:31.11 | bugz | that thing could survive a direct hit |
02:31.17 | Lino` | http://www.lino-helms.com/kiste1.jpg <= boxes like those? |
02:31.31 | Lino` | (i use those to store cd / dvd media *lol*) |
02:31.35 | bugz | thats the ip601 |
02:31.40 | justinu | bugz, got any pics of those russian phones? |
02:31.53 | bugz | man no |
02:31.55 | bugz | but i can get some |
02:31.59 | bugz | now that i think about it |
02:32.02 | justinu | cool |
02:32.02 | bugz | the rig is still docked |
02:32.11 | justinu | i like funky electronics from soviet russia |
02:32.12 | Lino` | go and steal 'em |
02:32.23 | bugz | haha, they were like "hey you dont got in there!" |
02:32.34 | Lino` | in soviet russia phone calls you |
02:32.35 | justinu | bugs, you live in russia? |
02:32.37 | justinu | heh |
02:32.39 | bugz | "WOW DUDE LOOK AT THIS SHIT!" |
02:32.41 | Heimidal | no one? |
02:32.43 | bugz | ahaha.. |
02:32.43 | bugz | no |
02:32.46 | bugz | i live in Houston |
02:33.05 | Lino` | well houston is not that far away from soviet russia ;) |
02:33.14 | Lino` | i used to live in san antonio :p |
02:33.24 | bugz | yeah, in Houston the Hurricane does YOU! |
02:33.34 | justinu | lol |
02:33.36 | Lino` | lol |
02:33.46 | bugz | i put a 2 u server on it to drive a bunch of ip301's |
02:33.57 | justinu | cool |
02:34.02 | Lino` | i went to corpus christi once, twc announced a hurricane |
02:34.03 | bugz | theres like this regulation now where those rigs need a phone like every 10 feet |
02:34.04 | justinu | how they like the new phones? |
02:34.16 | bugz | they were tripping |
02:34.27 | bugz | the fax to email thing blew their mind |
02:34.29 | Lino` | as soon as i arrived they started the announcement |
02:34.33 | justinu | heh |
02:34.39 | Lino` | and all those rednecks started boarding up their windows etc. |
02:35.01 | bugz | lol |
02:35.05 | bugz | some of them anyway |
02:35.11 | _Simon | pigpen: are you still there? |
02:35.15 | bugz | the ones i know down there go fishing and surfing during hurricanes |
02:35.35 | Lino` | thats a once-in-a-lifetime experience - i'll never forget that... cheap houses made of wood which will be blown away anyway but lots of boards in front of the windows and duct tape (!!!) |
02:35.53 | _Simon | hey gang. I have to redial iax like 3-4 times to get a call to actually connect. even with echotest, this is on iax hard and soft phones. I am getting these when I do iax2 debug: http://pastebin.ca/ |
02:35.55 | _Simon | what does that mean? |
02:36.00 | bugz | i like the whole "im not going anywhere" attitude |
02:36.05 | Lino` | as if duct tape would protect them from anything... |
02:36.11 | Luke-Jr | Anyone here using SellVoIP and supporting a guest IAX account? |
02:36.32 | justinu | _Simon: that's not the right link for your paste |
02:36.39 | _Simon | ops |
02:36.41 | _Simon | sorry :) |
02:36.46 | bugz | we got a nice embedded system out now... |
02:36.51 | _Simon | http://pastebin.ca/46901 |
02:37.02 | _Simon | justinu: thanks ;) that line ^^ |
02:37.02 | bugz | it does all this freaky wanpipe stuff out of the box |
02:37.02 | _Simon | hehe |
02:37.11 | Lino` | i just thought about how it would be like if cisco had stores |
02:37.21 | Lino` | like a seven eleven |
02:37.24 | justinu | not sure... i'm not an iax expert |
02:37.28 | justinu | what does INVAL mean? |
02:37.34 | bugz | i would get drunk and do a router run |
02:37.40 | _Simon | not sure. its weird. if I redial and redial over and over it eventually works |
02:37.42 | Lino` | like in futurama with that conveyor belt on the ground |
02:37.54 | _Simon | if I reset asterisk, it works right away for the first 5 or so times, then starts intermittently do that |
02:37.59 | Lino` | "you know our policy is, if you are not 100% satisfied..." |
02:38.10 | X-Rob | _Simon, sounds like an * bug. |
02:38.10 | justinu | _Simon: you using qualify? |
02:38.14 | Lino` | "... i hate you. *pushs button* *customers are taken away by belt*" |
02:38.33 | _Simon | justinu: someone mentioned it before. I set qualify=yes in the iax accounts. i thought it fixed it,b ut really just resetting asterisk helped |
02:38.40 | justinu | _Simon: is there a nat involved/ |
02:38.40 | Lino` | hmmm |
02:38.42 | justinu | ? |
02:38.44 | _Simon | after 10 mins or so, all the problems again |
02:38.45 | _Simon | yes |
02:38.53 | _Simon | but IAX is supposed to be nat traversal? |
02:38.54 | justinu | is SPI (stateful packet inspection) turned on? |
02:39.06 | _Simon | umm not sure, its a linksys |
02:39.09 | justinu | some cheapy linksys/dlink/netgears offer "SPI", but it really just fucks things up |
02:39.10 | Lino` | looks like your firewall is trying to kill you. |
02:39.17 | justinu | i bet it's a firewall problem, simon |
02:39.20 | justinu | firewall/nat |
02:39.34 | _Simon | but those iax2 debug errors, that means its receiving something? |
02:39.47 | Lino` | yeah, at home i have a wrt54g. that little sonofabitch hates my 7960 |
02:39.50 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
02:39.50 | bugz | i like this asterisk bug: "when i check my voice mail 'the voice' stutters the date.." |
02:39.52 | justinu | turn on iax2 debug when you first start * |
02:39.54 | justinu | let it run |
02:40.03 | _Simon | thats what I did |
02:40.16 | _Simon | it works great for 5-6 calls or whatever |
02:40.18 | justinu | your paste didn't seem to have that much |
02:40.19 | _Simon | then all the sudden that junk shows up |
02:40.21 | _Simon | lol |
02:40.23 | bugz | anyone have any snort rules for sip over udp |
02:40.26 | _Simon | no I just pasted the errors |
02:40.34 | *** join/#asterisk ravenpi (n=chatzill@londonderry-cuda3-24-51-50-156.lndnnh.adelphia.net) |
02:40.35 | justinu | dude, looking thru boring logs is part of this job |
02:40.44 | bugz | hahaha |
02:40.47 | _Simon | hehe |
02:40.50 | Lino` | now seriously who pays money for a 7902 ??? |
02:41.04 | Lino` | i wouldnt event want it as a gift. |
02:41.04 | ravenpi | True 'nuff. And there are a lot of tools out there to help get rid of the "boring" parts of log reading. |
02:41.10 | justinu | yeah |
02:41.12 | Lino` | like grep |
02:41.21 | Lino` | or rm |
02:41.23 | Lino` | *gg* |
02:41.24 | bugz | hah, how bout some Luisa |
02:41.31 | bugz | a Luisa/Asterisk box |
02:41.32 | Sedorox | <PROTECTED> |
02:41.33 | justinu | grep is my tool; i understand there's a lot of neat new fangled dealies that sort stuff, etc. |
02:41.35 | _Simon | is qualify=yes okay? thats what I set |
02:41.48 | justinu | yeah, that sends a keepalive ever 3 seconds i think |
02:41.56 | justinu | that should be enough for even the most nazi nat |
02:42.07 | Strom_C | nazi nat! |
02:42.09 | Frogzoo | Lino`: swatch & bb |
02:42.10 | _Simon | haha |
02:42.21 | _Simon | hmm wonder what the issue is then |
02:42.25 | *** join/#asterisk Weezey (n=ohno@206.210.109.228) |
02:42.32 | justinu | check that SPI stuff |
02:42.38 | justinu | i bet your iax2 packets are getting mangled |
02:42.52 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
02:42.56 | _Simon | ya theres not much on a linksys to check though I don't think, its just a consumer router |
02:42.57 | justinu | since it's UDP, there's no guaranteed delivery |
02:43.05 | justinu | which explains your intermittant issue |
02:43.13 | Weezey | I'm getting no compatible codecs error when I try to dial SIP/exten@hostname.com, what do I need to change? |
02:43.24 | Lino` | well simon |
02:43.27 | justinu | iax2 packet INVAL is probably "your request is broken, try again" |
02:43.27 | Lino` | thats not 100% true |
02:43.32 | Lino` | linksys has some nasty functions |
02:43.36 | justinu | true |
02:43.37 | Lino` | like packet analyzing or qos |
02:43.45 | justinu | look very carefully thru all it's settings |
02:43.48 | _Simon | I set IAX2 port as high QoS |
02:43.50 | justinu | i bet you'll find it |
02:43.51 | _Simon | was that smart to do? |
02:43.52 | Lino` | which seems to improve things, but in fact it fucks things up |
02:44.08 | _Simon | oh so that was a bad thing to set? |
02:44.09 | Lino` | throw away your router and get a real one ;) |
02:44.14 | Lino` | well |
02:44.15 | bugz | minimize delay, not maxmize throughput |
02:44.34 | _Simon | well.. part of my issue is I'm trying to see if users will be able to use it on regular hardware |
02:44.38 | _Simon | linksys super popular in homes so |
02:44.53 | Strom_C | Weezey, is the extension you're dialing on your local box? |
02:44.54 | blitzrage | did you guys know its Friday night? :) |
02:45.02 | Weezey | no sirt |
02:45.03 | Weezey | sir |
02:45.04 | Lino` | yeah but to quote clayton bigsby "linksys stinks and i hate it" |
02:45.10 | Strom_C | where is it? |
02:45.10 | _Simon | lol |
02:45.15 | _Simon | so I should turn QoS off for iax port |
02:45.15 | _Simon | ? |
02:45.17 | Weezey | my other box. |
02:45.20 | Strom_C | ok |
02:45.23 | Lino` | well try turning off all the fancy stuff |
02:45.25 | Lino` | like qos |
02:45.27 | Lino` | firewall |
02:45.27 | Lino` | etc |
02:45.31 | Strom_C | what codecs are you allowing? |
02:45.35 | Lino` | (parental control *lol*) |
02:45.36 | Weezey | ulaw |
02:45.38 | Weezey | on both |
02:45.38 | _Simon | hehe |
02:45.47 | Strom_C | are you disallowing everything else? |
02:45.47 | Frogzoo | _Simon: QoS should be fine, so long as sip/iax gets priority |
02:46.18 | Weezey | Strom_C: the no codec thing actually might be right. |
02:46.23 | Weezey | one sec.. |
02:46.39 | justinu | what model linksys |
02:46.43 | CoffeeIV | I want to confirm my router is passing IAX2 to the right place. Is there a netcat command that should provoke a response from asterisk's IAX2 port ? |
02:46.47 | Weezey | Strom_C: nop |
02:46.48 | Weezey | e |
02:47.03 | bugz | time for some solaris action |
02:47.06 | Lino` | (just imagine the posts in the forums: "my mommy does not want me to use iax (whew its late i always want to write aix... im kinda ibm poisoned) how to kill linksys parental control" |
02:47.08 | Lino` | ) |
02:47.19 | Strom_C | weezey, in the appropriate section of your sip configuration, put "disallow=all" followed by "allow=ulaw" |
02:47.32 | Heimidal | can someone tell me why this causes the error "Mar 24 20:28:23 NOTICE[4193]: chan_iax2.c:7198 socket_read: Rejected connect attempt from 207.174.202.3, request '312546XXXX@incoming' does not exist"? http://rafb.net/paste/results/RceagI47.html |
02:47.42 | Weezey | Strom_C: all better, I'm using realtime |
02:47.48 | X-Rob | Heimidal, it means you don't have 315... in [incoming] in extensions.cofn |
02:47.54 | Weezey | and disallow=all comes after allow=ulaw |
02:48.01 | Weezey | so it ignored allow= |
02:48.03 | Strom_C | Weezey, well that will fuck it up |
02:48.08 | Weezey | all better |
02:48.12 | bugz | Heimidal: look in extensions.conf for 312546XXXX in the [incoming] context |
02:48.14 | Weezey | thanks for the inspiration |
02:48.15 | Heimidal | X-Rob: what if I want an extension that can take care of two incoming numbers? |
02:48.35 | justinu | whoever had the linksys, what model is it? |
02:48.46 | X-Rob | Heimidal, then put both numbers in [incoming] |
02:49.04 | Heimidal | with what syntax? |
02:49.10 | bugz | exten => 939222XXXX,1,Dial(Sip/100) |
02:49.14 | bugz | exten => 939555XXXX,1,Dial(Sip/100) |
02:49.19 | X-Rob | or use a wildcard -- exten _312X. |
02:49.24 | X-Rob | bugz, you forgot _ |
02:49.27 | Heimidal | ah |
02:49.54 | X-Rob | Heimidal, to correct bugz: exten => _312X.,1,Dial(SIP/123) |
02:50.02 | Heimidal | I guess I got confused by the asterisk book... it shows exactly what I put in that paste |
02:50.02 | X-Rob | _ means 'I have wildcard characters' |
02:50.41 | bugz | i have an asterisk 1.0.9 box up for almost a year |
02:51.00 | bugz | normally the drives fail long before that |
02:51.10 | bugz | sure have run into some crappy hardware |
02:51.41 | *** join/#asterisk maxx4life (n=max4life@71-35-210-12.slkc.qwest.net) |
02:52.21 | Heimidal | I find it strange that you can't pass something to a context and have it start with the "s" extension without having to explicitly declare the number |
02:52.53 | justinu | there's a lot of strange things about the asterisk dialplan |
02:53.06 | bugz | yeah, like how it works |
02:53.26 | justinu | it seems like every pbx was designed by someone from another solar system |
02:53.39 | justinu | i had this panasonic kxt that had the weirdest CLI |
02:54.01 | Heimidal | and if you have to explicitly declare everything, why does "s" even exist? |
02:54.23 | bugz | to confuse n00bs |
02:54.28 | justinu | heh |
02:54.43 | justinu | supposedly s exists when there is no DNIS info |
02:54.50 | justinu | for when... |
02:54.55 | Sedorox | "I see n00b$... they're everywhere.. and they don't even know they're n00b$" |
02:54.57 | *** join/#asterisk jroysdon (n=jroysdon@c-67-181-65-139.hsd1.ca.comcast.net) |
02:55.47 | bugz | snort keeps throwing these "(portscan) UDP Filtered Portscan" from my providers sip router |
02:56.25 | bugz | i scanned one of those boxes once because of this and found irc running on it but with a Cisco IOS fingerprint |
02:56.48 | bugz | that was a little unsettling |
02:57.20 | justinu | odd |
02:57.34 | justinu | did it actually speak ircd on 6667, or did 6667 just answer |
02:58.38 | bugz | when i tried to connect it i got connection refused, scanned it again and then it was closed |
02:58.56 | bugz | i called them and asked wtf was going on and they had no idea |
02:59.08 | justinu | hmm |
02:59.21 | bugz | that particular router was a peering system i believe with alot of latency on it |
02:59.22 | justinu | that is a bit unsettling |
03:00.07 | bugz | i see wierd stuff all the time on this network |
03:00.30 | bugz | i had the balls to apply for a job on monster.com that said "LINUX GURU NEEDED" |
03:00.44 | justinu | hah |
03:01.08 | bugz | its cool here, ive had the opportunity to put snort and iptables on every kind of network you can imagine |
03:01.08 | Nugget | no true unix guru considers himself one. |
03:01.22 | bugz | Nugget: ahh grasshoppaa.. |
03:01.29 | justinu | wax on, wax off |
03:01.55 | Nugget | "linux" guru makes it even more suspicious. |
03:02.49 | bugz | the IT departments i run in to are either a) MCSE/retardd, b) Cisco noobs, or c) pothead hippies who know everything |
03:02.59 | Nugget | I interviewed a guy at ud.com who was a self-described linux expert who, upon learning that we also used freebsd, asked if freebsd used "linux commands like 'tar' and 'ls'" |
03:03.35 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
03:03.42 | ravenpi | Which "matches" first: exten => _1600, or exten => 1600 -- in other words, does wildcard have higher, or lower (or same) matching priority. |
03:03.45 | jroysdon | Anyone ever used gparted? I'm using it now to resize a windows/ntfs install on a borrowed work laptop that I want to test fc5 on. |
03:03.49 | bugz | Nugget: i ran into a guy that told me he used iptraf once and that made him an expert on my firewall configuration |
03:03.53 | ravenpi | Nugget: man, now THAT'S a Linux guru. |
03:03.56 | justinu | know it all pothead hippies are ok |
03:04.15 | bugz | like my dad lol |
03:04.43 | bugz | "dad put down the pipe, the DHS is calling again..." |
03:04.45 | ravenpi | jroysdon: I've used parted, but never gparted -- and I never tried doing a resize (call me chicken). How's it working? |
03:04.46 | justinu | heh |
03:05.15 | Nugget | I don't think I'll ever trust partition resizing. call me superstitious. |
03:05.21 | *** join/#asterisk _Simon (n=IRC@i216-58-40-193.cybersurf.com) |
03:05.25 | _Simon | hmm sorry got disconnected |
03:05.29 | Nugget | especially on ntfs, which seems like negligent insanity to me |
03:05.31 | jroysdon | ravenpi, it's running.. it resized the first fat partition, and now it is doing the ntfs/restore partition. |
03:05.32 | _Simon | setting QoS or not setting it isn't making much difference |
03:05.41 | justinu | what can't it work right? |
03:05.43 | _Simon | there seems to be like 20 asterisk processes when I do a "ps" is that normal? |
03:05.58 | jroysdon | Partition Magic 9 does great with resizing... but I don't own a legal copy these days, and I'm trying to stay totally legal/free |
03:06.01 | justinu | _Simon: good question |
03:06.20 | _Simon | theres literally 20 asterisk -vvvvcd processes when I run asterisk in CLI |
03:06.34 | bugz | http://pastebin.com/621104 |
03:06.43 | bugz | here is a piece of some old QoS |
03:06.49 | justinu | if init runs asterisk from the startup script in /etc/init.d, i see a bunch of asterisk processes in ps |
03:06.57 | justinu | if I run it from the shell manually, i see one |
03:07.11 | _Simon | justinu: I'm running from shell, and I still see like 20 of them |
03:07.50 | _Simon | so is that normal? if not maybe thats my issue |
03:08.01 | bugz | http://pastebin.com/621108 |
03:08.29 | bugz | _Simon: i think all the modules make asterisk spawn processes |
03:08.32 | justinu | i'm not sure if it's normal |
03:08.41 | justinu | it doesn't seem to cause any problems, for sure |
03:08.48 | _Simon | ok, I'll assume its normal |
03:09.03 | Strom_C | _Simon, running asterisk in the background, I only have one asterisk process going |
03:09.06 | bugz | _Simon: i know when one of our bigger systems goes under a load there are like 50 processes going |
03:09.13 | _Simon | ah |
03:09.17 | bugz | _Simon: when its in trouble there are hundreds |
03:09.26 | _Simon | ok :) |
03:09.29 | _Simon | so now heres another thing |
03:09.42 | _Simon | as soon as I run asterisk |
03:09.44 | _Simon | I do a |
03:09.49 | _Simon | iax2 show channels |
03:09.59 | _Simon | and theres some IP connected to my server I don't know that IP |
03:10.01 | justinu | _Simon: you're the one with the linksys? what model is it |
03:10.10 | justinu | could be from the samples, simon |
03:10.13 | _Simon | its *ALWAYS* connected, even if I stop/start its there |
03:10.14 | justinu | they connect to digium, etc. |
03:10.19 | _Simon | ahh |
03:10.22 | bugz | lol |
03:10.36 | _Simon | where would that be defined? |
03:10.46 | bugz | id sure like to get my hands on about 100 of those wrtg's with 32 megs of ram |
03:10.47 | _Simon | I just want to start eliminating things that could be causing issues |
03:11.02 | bugz | i think its v4.0 of the firmware |
03:11.03 | _Simon | justinu: and to your question. WRT54G |
03:11.09 | justinu | ok, then it has SPI |
03:11.20 | *** join/#asterisk ravenpi (n=chatzill@londonderry-cuda3-24-51-50-156.lndnnh.adelphia.net) |
03:11.42 | _Simon | so where would I find that connection that might be going to digium? |
03:11.47 | _Simon | in which conf file? |
03:11.48 | justinu | in iax.conf |
03:12.26 | bugz | ive created a pretty extensive toolbox for using asterisk -rx stuff w/php and bash |
03:12.47 | bugz | it seems like everyone wants to build a gui for it but nobody can agree on exactly how |
03:12.50 | _Simon | hmm.. don't see anything in iax.conf except users |
03:13.05 | _Simon | any other file perhaps? |
03:13.17 | justinu | if it's iax, it's in iax.conf |
03:13.22 | justinu | check for includes, etc. |
03:13.25 | _Simon | thing thats weird, the IP is a comcast IP |
03:13.41 | justinu | how are they registered to you? |
03:13.45 | justinu | iax2 show registry |
03:13.45 | bugz | haha, who did you buy this from again? |
03:13.53 | justinu | iax2 show users |
03:13.56 | _Simon | k |
03:14.03 | justinu | iax2 show peers |
03:14.45 | justinu | and plain: show channels |
03:15.44 | Heimidal | according to the asterisk book, http://rafb.net/paste/results/4hmgMJ38.html should play the sound file and wait for user input. However, I've just tried it, and after playing the file, it immediately hangs up. |
03:15.49 | Heimidal | what am I doing wrong? |
03:15.51 | _Simon | ahh ok found who it is |
03:15.53 | _Simon | thats cool :) |
03:16.00 | justinu | Heimidal: check the error log |
03:16.00 | _Simon | its just an iax peer so I doubt its the problem |
03:16.03 | justinu | set debug 5 |
03:16.09 | justinu | i had that problem too |
03:16.20 | justinu | turns out i wasn't specifiying the file type, or something |
03:16.26 | _Simon | he just left it running over night |
03:16.26 | _Simon | lol |
03:16.52 | Heimidal | <PROTECTED> |
03:16.52 | Heimidal | <PROTECTED> |
03:16.56 | justinu | did you check the SPI setting in your router? |
03:17.00 | Heimidal | that's what happens after it plays the file. |
03:17.15 | justinu | Heimidal: check /var/log/asterisk/messages |
03:17.17 | _Simon | justinu: I'm not really sure which part of the web config is SPI |
03:17.20 | justinu | or whatever it's called |
03:17.26 | justinu | _Simon: get the manual from the net, and find it |
03:17.36 | _Simon | well I know theres port forwarding, port triggering |
03:17.45 | justinu | another guy had this similar problem as you, and that was the fix |
03:18.29 | bugz | _Simon: put this in a file called "/usr/local/bin/commands" |
03:18.31 | bugz | http://pastebin.com/621123 |
03:18.45 | bugz | then build some aliases in .bashrc |
03:19.06 | bugz | i want to rewrite the whole UNIX/Asterisk interface like that |
03:19.30 | _Simon | hmm cool I'll take a look at that after :) |
03:21.07 | Heimidal | justinu: there don't seem to be any errors with my current dialplan |
03:21.43 | CoffeeIV | my * seems to listen on port 4569 when I connect from the localhost, but not externally. No iptables or hosts.deny are blocking anything, bindaddr is 0.0.0.0. Any hints on what it might be ? |
03:22.21 | Qwell | CoffeeIV: check netstat |
03:23.59 | CoffeeIV | Qwell: no 4569 in the netstat output, I see the 5038 for the telnet api and few other things |
03:24.47 | Qwell | netstat -lna | grep 4569 |
03:25.01 | jroysdon | fyi, gparted just worked great - no problems |
03:25.48 | CoffeeIV | Qwell: udp 0 0 0.0.0.0:4569 0.0.0.0:* |
03:26.01 | Heimidal | bleh... this is frustrating |
03:26.16 | _Simon | ok this is the only thing I can find |
03:26.30 | _Simon | Firewall Protection: "enables or disables the SPI firewall" |
03:26.31 | _Simon | its enabled |
03:26.44 | justinu | disable that |
03:27.10 | _Simon | doesn't that take off the firewall? |
03:27.13 | justinu | no |
03:28.53 | X-Rob | Heimidal, re that enter-number thing, the book is wrong |
03:28.58 | X-Rob | there needs to be a 'WaitExten' there. |
03:28.59 | *** join/#asterisk fugitivo (n=user@201.255.182.148) |
03:29.07 | fugitivo | Hi |
03:29.11 | justinu | if you use 1.2, yes |
03:29.26 | justinu | either that or set autofallthrough=no |
03:29.59 | X-Rob | which is duplicating old behaviour, which again, is wrong. |
03:30.02 | X-Rob | 8) |
03:30.07 | justinu | heh |
03:30.28 | fugitivo | This nokia 770 is awesome |
03:30.57 | _Simon | ok actually the docs as far as I can see says "Firewall Protection" disabling this means turning off the firewall |
03:30.58 | _Simon | lol |
03:31.00 | _Simon | thats not good |
03:31.16 | justinu | it will not turn off the firewall, don't worry. |
03:31.24 | justinu | it turns off SPI only, which is bullshit |
03:31.31 | _Simon | so SPI is bad? |
03:32.18 | justinu | yes, it mangels IAX apparently |
03:34.38 | _Simon | ok I'll give it a shot for 5 mins or so and see if it helps :) |
03:34.49 | *** join/#asterisk usam (n=usam@203.156.42.218) |
03:35.08 | usam | inband-dtmf, only for alaw+ulaw ? |
03:35.19 | Heimidal | X-Rob: thanks! |
03:35.29 | _Simon | that sucks so bad that defaults for a router kills IAX though |
03:35.29 | _Simon | lol |
03:35.56 | _Simon | justinu: if the asterisk box is behind a router as well the same type. will it mangle the server too? |
03:36.00 | _Simon | or just clients? |
03:36.07 | justinu | possible both |
03:36.14 | justinu | i'd turn it off on both sides |
03:36.26 | _Simon | ok, ya I'm getting the problem still |
03:36.35 | _Simon | so I'll disable on the server end |
03:37.19 | *** join/#asterisk jero (n=jero@modemcable062.46-82-70.mc.videotron.ca) |
03:38.24 | _Simon | curious to see what other things are affected by SPI being off lol |
03:39.17 | _Simon | ah crap that didn't help |
03:39.20 | _Simon | still having the issues |
03:39.27 | _Simon | damn what a pain in the nuts |
03:40.09 | justinu | :( |
03:40.17 | justinu | sorry that's not it... |
03:40.37 | justinu | make a tethereal capture and see if you can figure out why the server may not like the stuff you send it |
03:40.42 | X-Rob | _Simon, has anyone asked you what version of * you're using at both ends? (What are they?) |
03:41.18 | _Simon | X-Rob: just a iax soft phone/iax hard phone on one end and like a week old SVN version of asterisk |
03:41.23 | justinu | _Simon: http://lists.digium.com/pipermail/asterisk-dev/2003-June/000938.html |
03:41.25 | X-Rob | *BAHAHAHA* |
03:41.26 | Supaplex | and do they stay crunchy in milk? |
03:41.31 | X-Rob | _Simon, IAX in SVN is _broken_ at the moment |
03:41.34 | justinu | lol |
03:41.35 | X-Rob | Use 1.2.5 |
03:41.37 | _Simon | omg really? |
03:41.38 | _Simon | lol |
03:41.44 | justinu | why do people run SVN? |
03:41.51 | Strom_C | X-Rob, how broken is it? |
03:42.02 | Supaplex | because ages ago we kept pushing cvs head |
03:42.03 | justinu | what's wrong with the release versions? |
03:42.03 | X-Rob | Strom_C, I don't know, but it plays up. There's been a couple of reports of it on -dev |
03:42.10 | _Simon | if it was public knowledge it would of been a simple answer for me |
03:42.11 | _Simon | lol |
03:42.25 | X-Rob | svn is for developers, and those with experience |
03:42.32 | X-Rob | it was different in cvs times |
03:42.51 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
03:42.59 | _Simon | ok lemme try getting 1.2.5 and test |
03:43.13 | X-Rob | simon, grab 1.2 svn |
03:43.27 | X-Rob | eg instead of /trunk, get /branches/1.2 |
03:43.39 | _Simon | k |
03:43.56 | X-Rob | Çommands to get the current snapshot from the release branch of SVN: |
03:43.56 | X-Rob | # svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
03:43.57 | X-Rob | # svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2 |
03:43.57 | X-Rob | # svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2 |
03:44.53 | _Simon | is it a problem if I don't use version 1.2 of zaptel? |
03:45.00 | X-Rob | trunk of zaptel? |
03:45.05 | _Simon | ya |
03:45.05 | X-Rob | trunk of zaptel is good. |
03:45.08 | X-Rob | much better echo cancellation |
03:45.09 | X-Rob | works fine |
03:45.11 | _Simon | k :) |
03:45.19 | _Simon | so I'll just leech asterisk and compile it |
03:45.24 | X-Rob | yup |
03:45.40 | _Simon | you know what might be helpful is some sort of asterisk advisory or something |
03:45.40 | _Simon | lol |
03:45.51 | X-Rob | 'don't use SVN unless you know what you're doing' |
03:45.52 | _Simon | like a news page or something lol |
03:46.00 | _Simon | true but even if someone knows what their doing |
03:46.07 | _Simon | for example. I'm mostly using svn because I need realtime support |
03:46.17 | _Simon | as realtime is still experimental |
03:46.28 | _Simon | but I've been 6 months working on "experimental" |
03:46.41 | X-Rob | realtime is in 1.2? |
03:46.57 | _Simon | I don't even think realtime is completed currently in svn |
03:47.02 | _Simon | many CLI commands don't support it etc |
03:51.35 | *** join/#asterisk bmg505 (n=leon@165.146.13.215) |
03:51.49 | *** join/#asterisk Splat (n=Splat@220-253-101-194.TAS.netspace.net.au) |
03:52.04 | _Simon | are they doing work on iax? making it better or something? |
03:53.33 | X-Rob | fiik |
03:53.45 | _Simon | fiik? |
03:54.00 | X-Rob | fucked if I know. |
03:54.04 | _Simon | oh.. haha |
03:57.03 | *** join/#asterisk coppice (n=chatzill@218.203.17.210.dyn.pacific.net.hk) |
03:58.48 | SkramX | If i come back later, could I get some help with configing a TDM card? |
03:58.49 | bugz | haha thats funny.. "Are they going to work on IAX?" |
03:58.54 | bugz | _Simon: good question dude |
03:58.59 | SkramX | I have done a lot of virtual asterisk stuff, just nothing physical yet |
03:59.32 | bugz | i have a new distro out |
03:59.35 | bugz | called "Ass-Linux" |
03:59.39 | bugz | for asterisk noobs |
03:59.51 | bugz | seriously |
03:59.55 | bugz | haha.. |
04:00.08 | mitcheloc | why not call it Asterisk Live? that's a good name, |
04:00.33 | SkramX | what is this you all are naming? |
04:00.47 | mitcheloc | an asterisk distro, like asterisk@home |
04:00.54 | SkramX | oh okay |
04:02.26 | X-Rob | I prefer 'assman' as the name-of-the-year. |
04:02.39 | Grizzy | Someone's code I was auditing collected segfaults and restarted the code in the main loop. : o ) |
04:02.53 | bugz | X-Rob: no, 'assman' is the GUI |
04:04.55 | X-Rob | that's pretty damn hard to search for on google, youknow. |
04:05.09 | SkramX | what do you call a TDM 4 port with 4 incoming/FXO ports? |
04:05.14 | SkramX | whats the model #, thatis? |
04:05.41 | X-Rob | I can never remember which digit's which. I call it 'A TDM 400 with 4 FXO ports, please' |
04:06.00 | Splat | do you need libpri if you don't have isdn? |
04:06.11 | _Simon | so far asterisk 1.2 is still working... I think its fixed but will give it another 10 mins |
04:06.39 | SkramX | meh, ill ask for config help when i get the hardware/ssh access :) |
04:08.13 | *** join/#asterisk tkprojects (n=chatzill@c-24-13-177-60.hsd1.il.comcast.net) |
04:08.40 | tkprojects | does anyone know wafro or how to get in touch with him? |
04:09.04 | bugz | this embedded system rocks |
04:09.14 | bugz | they put a gui on it for us to customize |
04:09.27 | X-Rob | ~seen wafro |
04:09.33 | jbot | wafro <i=matt@i216-58-44-251.cybersurf.com> was last seen on IRC in channel #asterisk, 189d 22h 4m 46s ago, saying: 'but i was looking around and people have made some progress with other firmwares by loading a non encrypted XML'. |
04:09.41 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
04:10.26 | SkramX | lol |
04:10.48 | bugz | i bow to no os |
04:10.53 | bugz | >=] |
04:16.02 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
04:16.17 | bugz | time for some ET |
04:16.47 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
04:19.12 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
04:19.24 | astra^^ | hi all |
04:20.12 | _Simon | wicked thanks guys, asterisk 1.2 did the trick :) you guys rock! |
04:20.21 | _Simon | looks like enabling SPI has no probs as well |
04:24.14 | viLeR | A little question: it is possible to make two steps at the same time ? for example: exten => 900,3,Dial(IAX2/john) & System("scipt.sh) |
04:26.23 | tsume | games are a waate of time |
04:26.27 | justinu | yep |
04:26.29 | tsume | *waste |
04:26.34 | justinu | i think they help out a bit too |
04:26.52 | tsume | I work for fun |
04:27.05 | justinu | the virtual worlds some of these people have created are pretty amazing |
04:27.08 | tsume | earn money for fun, so I can get a couple of akitas |
04:27.22 | _Simon | I love C# |
04:27.27 | _Simon | I did an iax wrapper in C# |
04:27.33 | tsume | justinu: real friends are better, especially when they protect you too |
04:27.44 | tsume | _Simon: yea? |
04:27.46 | justinu | uh, ok |
04:28.09 | _Simon | tsume: yeah I'm working on a jabber client which integrates iax |
04:28.18 | justinu | i'm an introvert because I like games? |
04:28.23 | justinu | that's an interesting jump of logic |
04:28.56 | tsume | justinu: ususaaly extroverts do something outside of the crackerbox |
04:29.08 | justinu | i have my real life hobbies too |
04:29.14 | tsume | cracerbox == apartment |
04:29.20 | justinu | i own a house, also :P |
04:30.20 | X-Rob | bugger apartmens |
04:30.32 | X-Rob | airoplanes hit them. |
04:30.42 | X-Rob | they'd have to be a damn good shot to hit my house. |
04:31.58 | coppice | we have a big hill just to protect our apartment from the planes :-) |
04:32.39 | Nugget | yay planes. |
04:32.39 | justinu | heh |
04:33.32 | justinu | any instrument rated pilots here? |
04:33.49 | Nugget | just vfr for me at the moment. |
04:33.53 | Nugget | and I haven't flown in ages |
04:33.59 | justinu | what's ages? |
04:34.00 | Nugget | too busy hacking on flightaware :) |
04:34.07 | Nugget | I haven't been PIC in 10 months. |
04:34.09 | justinu | flightaware? |
04:34.14 | justinu | 10 months isn't too bad |
04:34.18 | justinu | it's kinda like riding a bike |
04:34.33 | Nugget | yeah |
04:34.38 | tsume | stop chit chatting and go code |
04:34.47 | tsume | before I yank yer ears off and feek them to the dogs |
04:34.51 | justinu | cut him some slack, it's friday night |
04:34.55 | justinu | damn slave drivers |
04:35.05 | tsume | :D |
04:35.14 | tsume | I take classes from the Man |
04:35.14 | coppice | i just realised how many years it is since I last designed a piece of an aeroplane. its making me feel old :-( |
04:35.19 | Nugget | we're actually working on airport resources right now, but mostly I'm drinking guinness and complaining about dbaker's code |
04:35.45 | Nugget | we're getting all the approach plates and stuff sorted out, and adding user-maintained fbo/ground facility stuff. |
04:35.57 | justinu | you have online approach plates? |
04:36.00 | Nugget | yes |
04:36.04 | justinu | what about enroute charts? |
04:36.16 | Nugget | http://flightaware.com/resources/airport/KAUS |
04:36.54 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
04:37.50 | justinu | looks like a nice resource |
04:37.57 | Nugget | it's been a lot of fun to build. |
04:38.12 | justinu | wish the gubment would put the enroute charts online already.... |
04:39.24 | Nugget | we experimented with overlaying the flight tracking on top of the sectional charts, but in practice it was really awful. |
04:39.31 | justinu | i found a very nice Cirrus SR22 add-on for FS2004 |
04:39.40 | justinu | sectionals are too cluttered |
04:39.43 | Nugget | yeah |
04:39.49 | justinu | the enroute's are a lot cleaner |
04:40.01 | justinu | every instrument is faithfully reproduced in this sim |
04:40.08 | justinu | the glass panel, autopilot, gps |
04:40.13 | justinu | very impressive |
04:40.16 | justinu | great for training |
04:40.23 | Nugget | I'm working on getting all the DP/STAR coordinates into the database so we can plot the filed route alongside the flight track. |
04:40.27 | Nugget | that's my current project |
04:40.30 | justinu | cool |
04:41.08 | justinu | sr22 is a nice plane, i'd like to buy a share in one |
04:41.14 | justinu | fast, modern |
04:41.22 | Nugget | yeah, I don't think I'm a good enough pilote for one yet, though. |
04:41.27 | justinu | why not? |
04:41.33 | Nugget | I'm not a very good pilot. :) |
04:41.36 | justinu | oh |
04:41.40 | justinu | i'm competent |
04:41.44 | Nugget | I've never flown anything other than 172s |
04:41.51 | justinu | i still prefer flying with other pilots |
04:41.54 | justinu | makes life easier |
04:41.57 | Nugget | and I'm pretty far from being able to pursue my IFR rating |
04:42.14 | justinu | IFR in the sr22 is a whole different world than in a round guage 172 |
04:42.18 | Nugget | yeah |
04:42.20 | justinu | you'd be amazeds |
04:43.25 | justinu | we could probably shape you up as a pilot, anyways |
04:43.28 | justinu | it's not all that tough |
04:43.52 | justinu | i got a chance to fly the airbus A320 sim at united headquaters, in denver |
04:43.55 | justinu | that was fun |
04:44.06 | Nugget | I flew the 777 simulator at united in denver. :) |
04:44.09 | justinu | cool |
04:44.16 | Nugget | one of the d.net guys works there at the ual flight center |
04:44.17 | justinu | the 777 is a pig tho |
04:44.32 | justinu | one of my buddies (real person) is a united captain |
04:44.53 | *** join/#asterisk fjean (n=fjean@201009208229.user.veloxzone.com.br) |
04:44.54 | Nugget | I like that the simulator includes the "fasten seat belt" ding. |
04:45.01 | justinu | it's amazing |
04:45.08 | justinu | the gfx are kinda primitive |
04:45.14 | justinu | but the motion/sound effects is fantastic |
04:45.18 | Nugget | yeah |
04:45.46 | Nugget | http://cuckoo.com/daniel/pictures/albums/ga-ual/ding.mov |
04:46.24 | justinu | the a320 is a real cinch to fly |
04:46.31 | justinu | does all the tough work for you |
04:46.40 | justinu | even tells you when to retard the throttles in the flare |
04:47.06 | justinu | i had them simulate a v1 cut, and even that was a cinch |
04:47.17 | justinu | v1 cut is the trickiest thing multi pilots have to practice |
04:47.41 | justinu | one engine fails at v1, which is commited to take off speed |
04:48.08 | justinu | back to the topic, i guess |
04:49.15 | opc0de | hey what's it called when you have n phone lines and they're all tied to a single number? ie if one line is busy, someone can call the same number and it rolls over to the next available phone line? |
04:49.22 | justinu | a hunt group |
04:50.05 | opc0de | this is something that must be provided by your telco right? |
04:50.10 | justinu | yeah |
04:50.19 | justinu | if you don't control the switch that sends you calls |
04:50.44 | opc0de | I have 4 lines coming into an FXO interface |
04:51.58 | opc0de | where do I find out how to configure this in asterisk? |
04:52.11 | justinu | you'll have to talk to the telco, like you said |
04:52.47 | opc0de | and the asterisk configuration is just the same as it would be for 4 separate lines on 4 channels? |
04:52.53 | justinu | basically |
04:54.24 | *** join/#asterisk jgomata (n=jgomata@201.143.139.160) |
04:54.50 | fjean | justinu, hi, tell me, is it a real pain to configure the iptable for SIP and IAX on an asterisk box ? |
04:55.24 | fjean | by the way I started to install SER and it's going well... |
04:59.04 | justinu | heh, hey |
05:04.12 | Strom_C | I say.... |
05:04.15 | Strom_C | DEAD HOOKERS |
05:04.33 | Qwell | Strom_C: please stay out of my trunk |
05:04.52 | Strom_C | I didn't hide them in your trunk |
05:04.57 | Strom_C | I hid them in SVN trunk |
05:05.03 | Qwell | interesting |
05:05.18 | Strom_C | chan_deadhookers |
05:07.36 | *** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au) |
05:08.36 | *** join/#asterisk Heim|away (i=Heimidal@phpbb/styles/heimidal) |
05:09.12 | Qwell | Strom_C: gotta be res_deadhookers...every part of * can use it that way |
05:09.18 | joelsolanki | anybody can tell where to buy quad server? |
05:09.27 | joelsolanki | any website plz |
05:09.33 | Qwell | joelsolanki: google.com |
05:09.36 | Qwell | newegg.com |
05:09.39 | Qwell | dell.com |
05:09.46 | Qwell | microsoft.com |
05:09.47 | Heim|away | rackmountmicro.com |
05:09.48 | Qwell | oh wait |
05:10.03 | joelsolanki | hmm let me check out. thanks |
05:10.03 | Qwell | qwellsdiscountquadservers.pk |
05:10.08 | Strom_C | hahaha |
05:10.13 | joelsolanki | hmm ok |
05:11.10 | Qwell | that reminds me... |
05:11.16 | Qwell | Strom_C: You know what I haven't seen in a while? |
05:11.22 | Heim|away | joelsolanki: I'll admit that was a shameless plug, but we've been around for two years and have an incredibly good track record. |
05:11.24 | Strom_C | what? |
05:11.25 | Qwell | Crazy Gideon commercials |
05:11.29 | *** join/#asterisk hfern (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net) |
05:11.34 | Strom_C | who is Crazy Gideon? |
05:11.38 | Qwell | wtf |
05:11.47 | Qwell | How can you possibly live in CA, and ask that question? |
05:11.55 | Strom_C | because I don't own a television set? |
05:11.58 | Qwell | ... |
05:12.06 | Qwell | He's Crazy Gideon! |
05:12.21 | Qwell | he like...smashes shit, throws blenders, etc |
05:12.25 | Qwell | because he's crazy |
05:12.29 | Strom_C | you mean Gallagher? |
05:12.32 | justinu | maybe he's dead |
05:12.37 | Qwell | no, Crazy Gideon...it's a retailer, heh |
05:12.44 | Strom_C | *shrug* |
05:12.48 | ravenpi | How do I direct a call based on the Zap channel it came in on? I don't see an obvious variable (${CHANNEL} appears not to do the trick). |
05:12.49 | Qwell | home appliances and such |
05:12.56 | Qwell | justinu: surely you know what I'm talking about? |
05:13.01 | justinu | yeah |
05:13.36 | *** join/#asterisk maxx4life (n=maxx4lif@71-35-210-12.slkc.qwest.net) |
05:13.39 | justinu | he seemed kinda high strung |
05:13.47 | Qwell | heh, just a tad |
05:14.02 | *** join/#asterisk rjt69 (n=Administ@wsip-68-15-224-183.om.om.cox.net) |
05:14.29 | ravenpi | In NYC metro area, there used to be Crazy Eddie, doing much the same stuff for much the same store -- he kinda stopped when it turned out the owners of the chain were cooking the books. |
05:14.41 | justinu | yeah |
05:14.42 | Qwell | nice |
05:14.43 | ravenpi | "Our prices are INSANE!" |
05:14.59 | fjean | ravenip, you could throw on different context for each channel...or... |
05:15.08 | justinu | i wonder if anyone from LA remembers "Z Best carpet cleaning" |
05:15.11 | Qwell | "I stock themem deep, and sell them cheap!" |
05:15.12 | Qwell | :D |
05:16.25 | ravenpi | fjean: but how do I make that happen? Can I bind a channel to a context somehow? In zapata.conf, maybe? *goes and looks* Ahhhh... I guess so. Thanks! |
05:18.03 | litage | are there any disadvantages to enabling NAT options for an extension even if the extension isn't NAT'd? |
05:19.07 | Corydon76-home | extensions are natted? |
05:19.24 | Qwell | Corydon76-home: I have an extension firewall |
05:19.46 | Corydon76-home | ...with a spoon... |
05:19.56 | Qwell | yes, yes |
05:20.15 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
05:20.22 | Qwell | file[laptop]: home?! |
05:20.33 | Corydon76-home | Indeed he is |
05:20.39 | Qwell | yuck...sorry |
05:21.12 | justinu | does canada suck that much? |
05:21.44 | Qwell | justinu: some parts. |
05:21.45 | file[laptop] | home, home on the range |
05:21.48 | Qwell | other parts suck |
05:21.52 | justinu | i've never been |
05:22.01 | Corydon76-home | but especially New Brunswick |
05:22.15 | fjean | hey, can we adjust the volume in some way without any zaptel device (ztdummy) ? |
05:22.16 | justinu | sounds nice enough |
05:22.35 | Qwell | fjean: volume of what? |
05:22.48 | fjean | qwell: sound |
05:22.57 | Corydon76-home | Oddly enough, file escaped back to Canada with his innocence still intact. |
05:23.23 | Qwell | fjean: yeah, thanks, I'd have never been able to guess THAT |
05:23.48 | fjean | qwell, well that was a pleasure then ;-) |
05:24.25 | file[laptop] | I'm a lil' bit tired so I'll be... unconscious... for probably a day LOL |
05:24.46 | Qwell | file[laptop]: any trouble on the flight(s(s)) home? |
05:24.59 | file[laptop] | yes |
05:25.26 | Qwell | sounds...fun |
05:25.36 | file[laptop] | delays mostly |
05:26.04 | *** part/#asterisk fjean (n=fjean@201009208229.user.veloxzone.com.br) |
05:28.04 | Corydon76-home | Did they lose your luggage again? |
05:29.18 | file[laptop] | no |
05:30.21 | Corydon76-home | Well, there's always next time |
05:30.37 | Corydon76-home | The question is, will I be around next time to lend you a change of clothes? |
05:30.39 | Qwell | You should call and complain |
05:31.07 | Qwell | "I expect you to lose my luggage, and you can't get that right? I demand that my luggage get lost on all future flights." |
05:31.12 | Corydon76-home | BTW, I've been scraping that shirt you wore of DNA. I'm planning to clone you. |
05:31.21 | file[laptop] | yay clone |
05:31.29 | Qwell | oh boy.. |
05:46.43 | *** join/#asterisk blkremedy (n=ur3rdeye@142M28.oasis.mediatti.net) |
05:47.27 | *** join/#asterisk blkremedy (n=ur3rdeye@142M28.oasis.mediatti.net) |
05:50.08 | bugz | I HATE ASTERISK |
05:50.48 | bugz | anyone wanna work on a dial plan with me |
05:51.01 | bugz | i will give you temporary root |
05:51.09 | bugz | >;] |
05:51.31 | Strom_C | what about candy? |
05:56.41 | Corydon76-home | What about male strippers? |
05:57.29 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
05:57.37 | mitcheloc | whooo hooo! strippers! |
05:57.50 | Strom_C | bugz, I'll see what I can do for you :) |
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06:24.06 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
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06:25.15 | rogercharlie | can the FOP proxy be used for other manager connections? |
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06:35.02 | *** join/#asterisk HamYaI (n=HamYai@125.24.0.35) |
06:36.08 | HamYaI | anyone using soyo package with asterisk? |
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06:39.22 | konfuzed | hey does anyone use thinktel.ca? im curious about service quality or service levels. I suppose Call Quality is most important? |
06:46.43 | litage | are there any disadvantages to enabling NAT options for a sip device even if the device isn't NAT'd? |
06:51.27 | *** part/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com) |
06:51.52 | blitzrage | nope |
06:52.07 | blitzrage | no change really in the packetes |
06:52.36 | blitzrage | might even be a good idea incase you put the device behind NAT at certain points (i.e. mobile device) |
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06:59.29 | Altair256 | hello everyone |
07:00.38 | Strom_C | hi |
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07:13.07 | Fedoracore6 | hai all |
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07:23.04 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-135.claranet.co.uk) |
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07:36.11 | litage | thanks blitzrage |
07:36.32 | Strom_C | dogballs |
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07:53.46 | argos73 | grrrr.... dog just chewed up the photo album of our trip to NYC last year... wife's gonna love that when she wakes up... |
08:00.04 | *** part/#asterisk xbit` (n=xbit@frugalware.elte.hu) |
08:02.29 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
08:13.52 | Mavvie | argos73: very simple solution: don't let her wake up. |
08:18.49 | argos73 | Mavvie: hehe |
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08:45.15 | Fedoracore6 | hai all i try using agi code for delete but still cannot delete |
08:45.16 | Fedoracore6 | http://pastebin.com/621384 |
08:45.24 | Fedoracore6 | this my code ... |
08:45.44 | Fedoracore6 | hemm any code i must add org somethig code i forget |
08:55.10 | Fedoracore6 | can i delete data in databases.. using delee by field |
08:56.05 | Fedoracore6 | cos i do this code http://pastebin.com/621384 |
08:56.43 | Fedoracore6 | whole my data in my databases delete |
08:56.45 | Fedoracore6 | huhuhuhu |
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08:58.14 | Fedoracore6 | hehhehhee |
08:58.20 | Fedoracore6 | krisguy |
08:58.24 | Fedoracore6 | whyyy zzz |
09:01.31 | vattern | I am having trouble receiving faxes with spanDSP |
09:01.33 | vattern | http://pastebin.com/621398 |
09:01.59 | vattern | any pointers ? |
09:11.14 | tzafrir_laptop | Sorry, I'm not familiar with sandsp/rxfax |
09:12.28 | *** join/#asterisk voip_learner (n=learner_@210.18.40.222.sify.net) |
09:17.33 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
09:19.48 | voip_learner | can I install asterisk and just create a dummy enviorment so that I can test SIP based tools? or may be 2-3 compuers in LAN where asterisk is install in one and rest two will communicate. |
09:22.28 | vattern | that should work for sip based comms. |
09:24.08 | voip_learner | hi vattern , I want to test some toos like SiVuS, vomit... |
09:24.24 | voip_learner | and also some sip fuzzing tool |
09:25.03 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
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09:27.44 | voip_learner | what distro are you using? , vattern |
09:29.37 | voip_learner | ok |
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10:05.52 | rogercharlie | does the FOP proxy work for other manager connections |
10:07.08 | rogercharlie | I am unable to proxy other manager programs though the flash proxy |
10:07.17 | rogercharlie | anyone able to do this? |
10:07.23 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
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10:22.21 | jpk | sodele... FYI: Powerline steht. Netto 28Mbit. |
10:22.35 | jpk | Nicht überragend, aber ok. |
10:22.49 | jpk | sorry. wrong window |
10:24.51 | Fedoracore6 | where i can find code delete field in my database |
10:25.10 | Fedoracore6 | cos i do alot of code ... still cannot success |
10:31.25 | *** join/#asterisk |cleric| (n=dacleric@p5482AC41.dip0.t-ipconnect.de) |
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10:52.41 | lemmy | hi |
10:53.17 | Fedoracore6 | hi |
10:53.58 | lemmy | did anybody manage to use ztdummy in domU 2.6.11? each time i load the ztdummy module the domU dies. i tried zaptel-1.2.4 and head. |
11:09.14 | *** join/#asterisk Plnt (n=someone@goodspeed.vscht.cz) |
11:09.21 | lemmy | i just need ztdummy for meetme so i tried app_conference too. but it crashes the asterisk on * 1.2.1 and 1.2.4 if two people join a conference room and start speaking. |
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11:27.57 | tzafrir_laptop | you don't need ztdummy for app_conference |
11:30.53 | lemmy | tzafrir_laptop: i know, but app_conference isn't working for me |
11:31.05 | lemmy | so i tried to get meetme running |
11:32.17 | lemmy | i would be happy to use app_conference though |
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12:12.22 | tzafrir_laptop | lemmy, please provide more details: kernel version, linux distro |
12:12.35 | tzafrir_laptop | What do you mean by "ztdummy dies"? |
12:12.59 | tzafrir_laptop | Do you get timing working properly? try zttest |
12:13.20 | lemmy | tzafrir_laptop: the xen domain/kernel crashes. |
12:13.32 | *** join/#asterisk RoyK (n=roy@242.80-203-45.nextgentel.com) |
12:13.58 | lemmy | its debian/etch, kernel 2.6.11 on xen 2.0.7 |
12:14.23 | lemmy | what else do you need? |
12:16.30 | *** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-119-81.d-ip.magma.ca) |
12:19.50 | lemmy | tzafrir_laptop: some extra informations http://pastebin.com/621508 |
12:22.39 | *** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-119-81.d-ip.magma.ca) |
12:23.48 | tzafrir_laptop | lemmy, do you have any interesting panic statement? |
12:30.38 | lemmy | nope, the hole domU crashes right after i load ztdummy. zaptel loads fine though. |
12:33.51 | RoyK | domU? |
12:34.06 | lemmy | RoyK: a domain inside xen. |
12:35.09 | lemmy | tzafrir_laptop: i attached the last few lines i get here: http://pastebin.com/621518 |
12:43.42 | SplasPood | lemmy: maybe try xen 3.0?? |
12:44.39 | lemmy | SplasPood: you got ztdummy working with xen 3.0? |
12:45.31 | HamYaI | anyone using soyo as your SIP provider? |
12:45.47 | HamYaI | or ever used |
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12:52.52 | *** part/#asterisk RassBariaw (n=Rass@cpe-66-65-37-45.nyc.res.rr.com) |
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12:53.18 | usn | hi folx |
12:55.25 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
12:55.31 | x86 | hmm, i have a Grandstream BudgetTone 101 phone that no longer gets caller ID info... |
12:56.01 | x86 | i thought maybe it was something I did with my asterisk configuration, but my X-Lite softphone still gets caller ID just fine |
12:56.13 | x86 | anyone know what might be causing this? |
12:56.40 | x86 | it happened after i moved all of my SIP peers and users into MySQL and started using RealTime |
13:00.48 | x86 | the display on the grandstream just says "nu" |
13:01.28 | *** join/#asterisk coppice (n=chatzill@169.198.17.210.dyn.pacific.net.hk) |
13:05.47 | x86 | this is so bizarre |
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13:08.40 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
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13:12.12 | SparFux | I want to send digits on an existing bri channel. Dial(CAPI/ISDNdev/<digits>/o,4) does the job, but I am afraid it tries to open the capi device again. How can I avoid this and just send the <digits> ? |
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13:21.59 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
13:22.07 | wiseguy_ | hello |
13:22.24 | wiseguy_ | somehow musiconhold plays strange sound |
13:22.28 | wiseguy_ | what can it be? |
13:24.41 | Jon335 | This is probably the most noobish question ever, but here it goes: I have an outbound call that I want to transfer to another extension; How do I do that? |
13:25.03 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-209-202.telkomadsl.co.za) |
13:25.48 | tzafrir_laptop | wiseguy_, edo you use mpg123? |
13:26.06 | wiseguy_ | tzafrir_laptop: yes |
13:26.21 | tzafrir_laptop | wiseguy_, any chance that you really use mpg321? |
13:26.46 | tzafrir_laptop | ls -l `which mpg123` and see if it is a symlink to mpg321 |
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13:28.49 | wiseguy_ | tzafrir_laptop: really |
13:29.59 | wiseguy_ | /usr/bin/mpg123 -> /etc/alternatives/mpg123 |
13:30.21 | wiseguy_ | /etc/alternatives/mpg123 -> /usr/bin/mpg321 |
13:32.41 | wiseguy_ | any other ideas? |
13:33.16 | SparFux | How can I change the path or name asterisk uses for mpg123 in the config file so that I can use a script even streaming ogg vorbis? I don't want to rename mpg123 and replace it with this script. |
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13:51.51 | Gennaro | hi some one can help me with an asterisk@home? |
13:52.01 | Gennaro | i installed it with an tdm400 |
13:52.18 | Gennaro | and i cant recive incoming call.. |
13:52.21 | Gennaro | why?!? |
13:53.40 | Gennaro | i need id service from my provider?!? |
13:54.59 | Gennaro | some channel for asterisk @ home ? |
14:00.43 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
14:03.14 | Gennaro | someone is connected?!? |
14:08.52 | Gennaro | some one can speack?!? |
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14:17.23 | tzafrir_laptop | wiseguy_, still here? |
14:18.19 | tzafrir_laptop | wiseguy_, mpg321 is indeed your problem. Either replace it with mpg123 from nonfree or whatever, or (better) convert mp3s to wavs offline |
14:18.44 | tzafrir_laptop | look for rawplayer on voip-info |
14:19.35 | FuriousGeorge | is two cores with the expense for asterisk? |
14:19.43 | FuriousGeorge | *are two |
14:22.28 | FuriousGeorge | *are two cores WORTH the expense, is what i meant to say |
14:24.20 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
14:24.21 | tzafrir_laptop | FuriousGeorge, not much different than a dual-cpu computer |
14:24.23 | *** join/#asterisk Djow1 (n=gavioes@200.103.150.81) |
14:24.32 | tzafrir_laptop | that is: you won't get twice the speed |
14:25.00 | FuriousGeorge | tzafrir_laptop: sure, i know what you mean |
14:25.29 | tzafrir_laptop | My hunch is that Asterisk is reasonably-well parallelised, due to the nature of its task |
14:25.38 | FuriousGeorge | my main concern is that i have to page() 17 extensions at the same time reliably, and im not sure how big of a cpu i need |
14:25.49 | FuriousGeorge | this big? |-------------| or bigger? |
14:27.14 | FuriousGeorge | seriously though, im an amd guy so im deciding between bartons or maybe an athlon 64, but then i got to thinking i might need more |
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14:35.02 | TinoW | hrm. msgsm.h:573: warning: 'xmc[48]' is used uninitialized in this function |
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14:46.08 | ambriento | furiousgeorge, how good is a 64 processor when your applications isnt 64 bits? |
14:47.41 | *** join/#asterisk _DAW (n=bob@adsl-150-58-20.msy.bellsouth.net) |
14:48.11 | _DAW | good morning all |
14:48.49 | TinoW | ambriento: it can leave a whole 32 bits address space to your application ;) |
14:49.08 | coppice | for some things dual cores really fly with *. It depends what you are doing. Software echo cancellation occurs in the driver, and it doesn't use the cores well. Codecs, like G.729, do make very good use of the cores, as long as their caches are big enough |
14:49.32 | SplasPood | Does anyone know how I would get the name of the current sip user from within the dialplan? |
14:49.54 | FuriousGeorge | ambriento: are the fastest intel 32bit chips comparable to the athlon64s? i dont follow intel much at all |
14:50.06 | FuriousGeorge | my point is just that all the newer chips are 64bits anyway |
14:51.56 | ambriento | indeed they are |
14:52.15 | FuriousGeorge | is intel still making 32bit chips? |
14:52.44 | coppice | yes. all the laptop chips are still 32 bit |
14:52.55 | ambriento | even the p4 HT has EMT64 instructions, which will allow you to run 64btis OS on it |
14:53.42 | coppice | not all 64 bit chips are created equal. the AMDs really fly running my DSP code on 64 bit FC4. The Intels are considerably slower |
14:58.33 | ambriento | hmmmm |
14:59.26 | ambriento | coppice, there is no way to choose what to do with reversal polarity in Unicall, is there? |
14:59.48 | wiseguy_ | somehow musiconhold plays strange sound? |
15:00.00 | wiseguy_ | anyone has/had this problem? |
15:01.10 | ambriento | wiseguy_, your not that wise, are you? |
15:01.12 | ambriento | :) |
15:01.48 | ambriento | j/k... by strange sound you mean like "too fast" or "too slow" play? |
15:02.07 | wiseguy_ | yes, really ;-) |
15:02.21 | wiseguy_ | but i can't find anything about that in google |
15:02.28 | wiseguy_ | too slow |
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15:03.29 | wiseguy_ | ambriento: any suggestions? |
15:03.49 | ambriento | wiseguy_, I'm try to remember what I did. It happened to me once |
15:04.52 | ambriento | but this has to do with sample rate, like your file has 22K samples and you play it at 8K |
15:05.40 | ambriento | are you using mpg123? which asterisk version? |
15:06.00 | ambriento | did you try resample the file? |
15:06.01 | wiseguy_ | 1.2.5 |
15:06.14 | wiseguy_ | asterisk |
15:06.24 | ambriento | got it |
15:07.29 | ambriento | which mode in moh.conf? custom? |
15:08.04 | coppice | ambriento: currently people only use unicall for R2, and reverse polarity is not a concept which exists there |
15:08.20 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
15:08.38 | wiseguy_ | ambriento: i haven't tried resampling it |
15:10.22 | wiseguy_ | any recomendations? |
15:10.43 | tzafrir_laptop | wiseguy_, you're using mpg321 |
15:11.20 | tzafrir_laptop | Did you see my previous comments? |
15:11.29 | Katty | someone fed me egg today and didn't tell me :< |
15:11.38 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
15:12.01 | coppice | i fed me rice today and didn't tell me |
15:12.06 | ambriento | coppice, do you have like "to collect" calls in hk? like I call you but you will be charged for that? |
15:12.07 | wiseguy_ | tzafrir_laptop: no, please tell me again :( |
15:12.23 | Katty | coppice: vegans don't eat egg :< |
15:12.54 | Katty | and people who know this should know better. |
15:13.29 | coppice | ambriento: we only get charged for IDD. I'm not sure if collect calls are supported there, but I have no idea off hand just how they might be supported |
15:13.37 | ambriento | katty, how do you realize that fed you up with egg? Taste? |
15:13.52 | ambriento | IDD? |
15:14.02 | coppice | Katty: carnivores don't eat rice, and people who know this should know better |
15:14.07 | Katty | ambriento: i started feeling sick, so i asked them directly what they put in it |
15:14.24 | coppice | ambriento: international calls |
15:14.26 | ambriento | who are they? friends? |
15:14.42 | TinoW | hm. is asterisk source missing capi support? |
15:14.47 | Katty | relatives |
15:14.51 | wiseguy_ | tzafrir_laptop, ambriento any suggestions? |
15:14.55 | coppice | ambriento: what are you trying to do in HK? |
15:15.31 | ambriento | coppice, actually, I'm trying to figure out how PSTN works in there |
15:15.51 | ambriento | since its R2, I can know a little more about it |
15:16.03 | coppice | HK doesn't use R2 |
15:16.18 | cpm | http://pastebin.com/621656 |
15:16.19 | cpm | clues? |
15:16.35 | ambriento | and the "to collect" stuff I was asking you, its cause we have here in Brazil such thing |
15:16.37 | ambriento | oops |
15:16.55 | ambriento | really? Idk why I had that in my mind |
15:18.04 | coppice | i wonder why brazil needs to do that? In other countries R2 delivers collect calls with a special code |
15:18.06 | ambriento | may be I got confused with the unicall.conf.sample and your place :) they are near, rite? |
15:18.29 | tzafrir_laptop | wiseguy_, mpg321 is indeed your problem. Either replace it with mpg123 from nonfree or whatever, or (better) convert mp3s to wavs offline |
15:18.36 | cpm | found it, http://bugs.digium.com/view.php?id=6696 |
15:18.37 | cpm | shit! |
15:20.20 | wiseguy_ | tzafrir_laptop: you mean use sox instead or what? |
15:20.49 | ambriento | special code, like one of those bits? I'm not a telephony guy tbh. |
15:28.26 | *** join/#asterisk _deg_ (n=deg@201.22.28.245.adsl.gvt.net.br) |
15:30.23 | TinoW | very strange: http://pastebin.com/621681 |
15:32.28 | *** join/#asterisk hfern (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net) |
15:33.16 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
15:33.19 | ambriento | brb |
15:33.29 | ambriento | Royk is here |
15:33.39 | ambriento | :) |
15:33.42 | cpm | TinoW, no clue. |
15:34.25 | TinoW | nu sto... |
15:35.38 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj06.dialup.mindspring.com) |
15:36.25 | RoyK | ambriento: ? |
15:39.50 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfj06.dialup.mindspring.com) |
15:42.13 | *** join/#asterisk Fedoracore6 (n=deddd@60.50.132.131) |
15:49.40 | tzafrir_laptop | wiseguy_, basically: use the custom: method there with a wrapper script around sox |
15:52.26 | *** join/#asterisk stoffell (n=stoffell@d51A5808B.access.telenet.be) |
15:52.30 | RoyK | i use that as well |
15:52.47 | RoyK | but if i restart asterisk, a sox process remains spinning on 100% cpu |
15:56.24 | HamYaI | anyone using soyo as your sip proxy? |
15:57.08 | RoyK | soya :) |
15:57.34 | HamYaI | I'm having a problem where the system keeps silent while pronouncing "numbers" |
15:57.38 | Fedoracore6 | air soya ;D |
15:58.05 | HamYaI | RoyK: soya source? |
16:01.09 | wiseguy_ | HamYaI: seeya |
16:01.10 | wiseguy_ | :) |
16:01.50 | Fedoracore6 | :) |
16:02.07 | FuriousGeorge | anyone have experience using page or meetme with 17 people? |
16:02.16 | FuriousGeorge | meetme's would have 1 speaker 16 listeners |
16:07.07 | wiseguy_ | tzafrir_laptop, ambriento thanks, know it works perfect |
16:09.00 | FuriousGeorge | if you have experience with 12-infinity, that will do, also |
16:16.18 | *** join/#asterisk zekeonfir3 (n=zekeonfi@c-67-171-116-215.hsd1.ut.comcast.net) |
16:22.17 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
16:32.40 | *** join/#asterisk kd211 (n=danika@p54873DF7.dip0.t-ipconnect.de) |
16:34.33 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
16:34.40 | *** join/#asterisk dca (n=dca@c-67-166-21-138.hsd1.co.comcast.net) |
16:38.22 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
16:41.58 | *** join/#asterisk kd211 (n=danika@p54873DF7.dip0.t-ipconnect.de) |
16:42.45 | *** join/#asterisk Primer (n=vi@sh.nu) |
16:48.52 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
16:54.22 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
16:58.25 | *** join/#asterisk forao (n=fasfd@c-68-45-40-226.hsd1.nj.comcast.net) |
17:04.54 | *** join/#asterisk onsip (n=onsip@219.133.11.6) |
17:05.56 | *** join/#asterisk flashbac1 (i=flashbac@adsl-074-229-048-003.sip.bct.bellsouth.net) |
17:06.02 | flashbac1 | hey guys |
17:06.31 | flashbac1 | does anybody have any experience with voicemail ODBC storage? |
17:07.08 | flashbac1 | hello? |
17:07.17 | RoyK | Corydon-w: ding |
17:07.23 | *** join/#asterisk salviadud (n=ralfalfa@201.138.132.150) |
17:10.18 | onsip | hello every one. I am a newbie here, and also newbie on SIP. Is there anyone who could give me some sugestion on learning SIP and the archtecture of PBX. I'd be very happy for your kind hearted help. |
17:10.59 | salviadud | have you done your homework buddyboy? |
17:11.04 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
17:11.54 | salviadud | i'd suggest www.voip-info.org |
17:12.23 | onsip | Ok , thank you very much <salviadud>.:-) |
17:13.39 | justinu | onsip, i suggest starting with RFC 3261 |
17:15.10 | Katty | mew. |
17:15.20 | onsip | of course, rfc3261 and related rfcs is the best ones for newbies, but it's of course, tooo, boring. I think the useage of such rfcs is best for reference, not for study. I think. |
17:15.30 | justinu | i disagree |
17:15.41 | justinu | i think the RFCs are written very well |
17:15.50 | justinu | morning katty |
17:15.58 | Katty | hihi |
17:16.33 | justinu | it's saturday... what should we be doing? |
17:16.38 | Katty | napping. |
17:17.23 | TinoW | there are even some funny RFCs... |
17:17.38 | RoyK | Katty: http://static.flickr.com/19/116762025_b7a35a854a_o.jpg |
17:17.46 | Katty | justinu: you could meet me for lunch i suppose |
17:17.50 | justinu | napping... i just woke up |
17:17.55 | justinu | i couldn't sleep in this morning for some reason |
17:17.59 | justinu | lunch sounds good |
17:18.10 | justinu | let me get my scramjet warmed up |
17:18.38 | Katty | RoyK: :< |
17:18.42 | justinu | we either need to live in a smaller country, or need faster transportation |
17:19.00 | justinu | lol, that face has to be photoshopped |
17:19.04 | Katty | i'm all for a bullet train |
17:19.07 | Katty | 300mph. |
17:19.21 | justinu | they've got them up to 800 kph now, which is approaching jet speed |
17:19.27 | Katty | mew! |
17:19.31 | Katty | i'll take two, plskthx. |
17:19.35 | justinu | ditto |
17:22.22 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
17:36.08 | *** join/#asterisk SparFux (n=player@e182021137.adsl.alicedsl.de) |
17:42.22 | Lino` | hmmm |
17:42.56 | SparFux | What is a "native bridge"? |
17:43.13 | Lino` | lol |
17:43.33 | Lino` | thats when a connection is being established within a native trunk |
17:43.37 | iDunno | one made of wood. |
17:43.39 | Lino` | e.g. within one isdn card |
17:43.40 | iDunno | and string. |
17:43.40 | *** join/#asterisk angom_h (n=angom@red-corp-200.79.145.199.telnor.net) |
17:43.50 | Lino` | :D lol @ iDunno |
17:45.04 | SparFux | And what's a native trunk? |
17:45.31 | SparFux | Lino: Yes, that's what I am trying to do. But all of the sudden, the line is hung up! |
17:45.40 | iDunno | one that grows from the ground after a native has planted the seed? |
17:46.31 | SparFux | Look at this: http://pastebin.com/621885 |
17:46.57 | SparFux | I try to initiate an isdn conference with asterisk. |
17:47.17 | justinu | native bridge just means there's no protocol or codec translating involved, i thought |
17:48.42 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
17:50.03 | SparFux | justinu: sounds logical. |
17:50.05 | *** join/#asterisk Dovid (n=Dovid@89-138-126-99.bb.netvision.net.il) |
17:50.09 | a1fa | anybody experiencing issues with broadvoice ATM? |
17:50.17 | a1fa | i am unable to make outgoing phone calls |
17:50.24 | a1fa | god damn those stupid mother fuckers |
17:50.41 | salviadud | sue them |
17:50.45 | justinu | haha |
17:50.46 | a1fa | no |
17:50.49 | salviadud | that's what america is all about |
17:50.49 | a1fa | fuck sue them |
17:51.05 | a1fa | their support doesnt even answer phone |
17:51.14 | a1fa | i hate their onhold music |
17:51.19 | salviadud | well, call them up, and put mixmonitor on |
17:51.24 | salviadud | the judge might like it |
17:51.43 | salviadud | good evidence imho |
17:52.02 | Dovid | yea |
17:52.10 | Dovid | thjey have been going downt he shitter for a while |
17:55.19 | SparFux | Well, after attempting this native bridge I get and active hangup immediately. :-( -- Attempting native bridge of CAPI/ISDN1/27-5a and CAPI/ISDN1/008001071020-5b |
17:55.20 | SparFux | <PROTECTED> |
17:56.07 | TinoW | SparFux: just curious, did you compile asterisk + capi stuff yourself? |
17:56.19 | SparFux | No, it's the debian version I use. |
17:56.29 | TinoW | I see |
17:56.48 | SparFux | I will compile myself, but my current computer is too slow and it is no fun this way. |
17:57.07 | TinoW | SparFux: its quick, but the capi stuff does not compile for me |
17:57.20 | Lino` | lol |
17:57.22 | SparFux | TinoW: Well, I compiled chan-capi myself. |
17:57.23 | Lino` | well |
17:57.33 | TinoW | SparFux: which version? |
17:57.41 | SparFux | Tino: actual cvs. |
17:57.51 | a1fa | i've been on hold for 10 minutes |
17:57.53 | a1fa | damn it |
17:57.54 | a1fa | dude |
17:58.01 | TinoW | SparFux: I get http://pastebin.com/621681 |
17:59.27 | SparFux | Strange error... |
17:59.43 | TinoW | I rhink so |
18:00.54 | a1fa | "Thank you for calling Broadvoice" "Your call is important to us" |
18:01.01 | a1fa | Obviously not if I have to wait 15minutes |
18:01.13 | a1fa | god damn bastards |
18:01.20 | a1fa | freaking zipperheads |
18:01.23 | iDunno | mmhmm |
18:03.06 | Lino` | cisco is worse @ a1fa |
18:03.13 | Fedoracore6 | hai all i doing code for delete data in tables but when i using this code , DELETE FROM student WHERE kodsubjek1 = '$exten' |
18:03.33 | Fedoracore6 | still cant delete by field |
18:03.36 | Fedoracore6 | http://pastebin.com/621899 |
18:03.46 | a1fa | Lino` : no cisco is good.. I have an SLA with cisco |
18:03.52 | a1fa | Lino` : they answer call in 2s |
18:04.01 | a1fa | Lino` : i have a direct line to CISCO engineers |
18:04.09 | Fedoracore6 | its have code that,can delete by field |
18:04.42 | *** join/#asterisk Muecke77 (n=muecke77@p54A9F93D.dip.t-dialin.net) |
18:05.21 | a1fa | guys, is vonage's support this bad? |
18:07.33 | a1fa | ok |
18:07.37 | a1fa | my neck is killing me now |
18:07.37 | salviadud | guess so... |
18:07.39 | a1fa | i am suing them |
18:07.40 | a1fa | for pain |
18:07.40 | Dovid | people have mixed reports on vonage |
18:07.42 | a1fa | in my neck |
18:07.42 | salviadud | yeah! |
18:07.47 | salviadud | you sue those bitches |
18:07.50 | salviadud | but like i said |
18:07.55 | salviadud | use mixmonitor if you can |
18:08.02 | salviadud | good evidence, on audio |
18:08.09 | a1fa | i got a speakerphone on my uniden cordless phone |
18:08.13 | salviadud | very good leverage |
18:08.14 | a1fa | comes in handy |
18:08.26 | salviadud | are you using asterisk to call those bitches? |
18:08.34 | a1fa | yup |
18:08.35 | salviadud | you see, i got a sipura 3000 |
18:08.42 | a1fa | i got PAP2-NA |
18:08.44 | a1fa | :P |
18:08.47 | a1fa | i love me a PAP2 |
18:09.04 | salviadud | alright, just monitor the outgoing call |
18:09.10 | salviadud | i monitor all my calls... |
18:09.13 | salviadud | just for fun |
18:09.21 | salviadud | i think it's illegal |
18:09.25 | salviadud | yet, i don't care |
18:09.29 | salviadud | i am mexican |
18:10.59 | salviadud | viva la revolucion, viva la tarjeta tormenta y emiliano zapata, metanse un dedo por el culo, ARRIBAAAAAAAAAAA! |
18:11.32 | a1fa | :) |
18:11.47 | a1fa | my ast box is at a remote location |
18:11.54 | a1fa | its at a datacenter pumping 100mbits |
18:12.46 | salviadud | my ast box is an old sony vaio P3 |
18:12.53 | salviadud | with slackware 10.2 |
18:12.56 | salviadud | and custom kernel |
18:13.03 | salviadud | i hate sony |
18:13.20 | salviadud | yet linux is a miracle worker |
18:13.52 | Dovid | a1fa: what data center r u in ? |
18:14.51 | websae | anyone here like voipbuster.com? |
18:14.57 | websae | curious what your thoughts are |
18:17.20 | *** join/#asterisk Sponge_bob (i=None@cpe-66-27-171-121.socal.res.rr.com) |
18:17.29 | tehdely | websae: howdy |
18:17.32 | tehdely | did you get my email? |
18:17.53 | *** join/#asterisk _deg_ (n=deg@200.250.222.8) |
18:20.53 | TinoW | aha, when I remove /usr/lib/include/asterisk and put the build dir includes in the Makefile, chan capi 0.3.5 compiles for asterisk 1.0.x but not for 1.2.5 |
18:23.34 | *** join/#asterisk lorinc (n=ang@caracas-0638.adsl.interware.hu) |
18:30.30 | *** join/#asterisk SparFux (n=player@e182030071.adsl.alicedsl.de) |
18:34.21 | TinoW | http://www.junghanns.net/en/download.html is the right place? I dont find 0.4.0 there... |
18:34.33 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
18:39.59 | SkalTura | what's ATA & DID? |
18:41.19 | [av]bani | whats the command to hangup channels again? |
18:41.37 | TinoW | ++ath |
18:41.38 | TinoW | ;) |
18:41.42 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
18:42.09 | nitram | TinoW: go to chan-capi @sourceforge |
18:42.34 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
18:43.30 | Dovid | ~ata |
18:43.33 | jbot | well, ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
18:43.34 | Dovid | !ata |
18:43.43 | Dovid | ~did |
18:43.45 | jbot | extra, extra, read all about it, did is Direct Inward Dialing |
18:43.50 | Sponge_bob | if you want to setup a phone systems for 50 users, how do you gauge how many pstn lines to get? |
18:43.51 | [av]bani | no possible way to clear a channel? |
18:44.04 | [av]bani | :( |
18:44.14 | Dovid | Sponge_bob: depends on the placee, what they do etc. whats thier current volume ? |
18:45.01 | Sponge_bob | Dovid: thats hard to say. i guess i need those stats first eh? |
18:45.20 | [av]bani | this sucks. i have stuck channels and no way to clear them? |
18:45.29 | Dovid | yes |
18:45.37 | Dovid | u can force close them |
18:46.23 | [av]bani | how |
18:46.30 | Dovid | us the help cpmmand in cli |
18:46.31 | Dovid | brb |
18:46.34 | [av]bani | i did |
18:46.35 | [av]bani | nothing |
18:47.19 | Dovid | when u type help in the cli what do u get ? |
18:48.27 | *** part/#asterisk _deg_ (n=deg@200.250.222.8) |
18:48.48 | *** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-99-41.d-ip.magma.ca) |
18:49.09 | Dovid | [av]bani: what happens when u type in help in the cli ? |
18:50.06 | justinu | soft hangup didn't work? |
18:50.09 | [av]bani | nope |
18:50.14 | justinu | sip channels? |
18:50.15 | [av]bani | sip show channels |
18:50.16 | [av]bani | Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message |
18:50.19 | [av]bani | 65.100.14.254 1030 1ccd093a5e0 00102/00000 ulaw No Tx: ACK |
18:50.22 | [av]bani | 192.168.42.254 4000 34a37d1c7fa 00102/00000 ulaw No Tx: ACK |
18:50.25 | [av]bani | 65.100.14.254 4010 5f56b63e450 00102/00000 ulaw No Tx: ACK |
18:50.28 | [av]bani | 65.100.14.254 4011 079c278c13e 00102/00000 ulaw No Tx: ACK |
18:50.31 | [av]bani | 192.168.42.32 FXO1 475f0b39-8d 00101/00102 ulaw No Rx: ACK |
18:50.31 | justinu | how about setting rtptimeout? |
18:50.34 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140) |
18:50.34 | [av]bani | soft hangup 079c278c13e |
18:50.36 | [av]bani | 079c278c13e is not a known channel |
18:50.39 | [av]bani | asterisk bug? |
18:50.43 | justinu | soft hangup SIP/ |
18:50.50 | justinu | use tab completion to finish |
18:51.00 | justinu | use the channel ID you see in "show channels" |
18:51.02 | justinu | not sip show channels |
18:51.17 | [av]bani | why did the channel get hung? |
18:51.22 | [av]bani | er, why did these hang? |
18:51.31 | [av]bani | the phones were rebooted, the ata was rebooted |
18:51.32 | justinu | hard to say without a trace |
18:51.33 | *** join/#asterisk _deg_ (n=deg@200.250.222.8) |
18:51.38 | [av]bani | asterisk kept them open regardless |
18:51.40 | justinu | I use the rtptimeout to auto clear those things |
18:51.51 | justinu | well... if asterisk doesn't get a BYE it probably still thinks the call is alive |
18:52.13 | [av]bani | we came in in the morning and all the phones were stuck with "calls" |
18:52.25 | [av]bani | asterisk was completely borked |
18:52.42 | [av]bani | which is stupid |
18:52.45 | Dovid | using svn or release ? |
18:52.48 | [av]bani | release |
18:54.01 | Dovid | hmm |
18:54.06 | Dovid | ver. ? |
18:54.24 | [av]bani | 1.2.4 |
18:54.37 | *** join/#asterisk froguz (n=froguz@204-141-222-201.adsl.terra.cl) |
18:54.52 | [av]bani | do you know this exact specific bug was fixed in 1.2.5? |
18:55.13 | Dovid | dont know, never heard of this issue on the list |
18:55.23 | Dovid | could there have been a power failure ? |
18:55.48 | [av]bani | no, everything is on UPS, and we rebooted the phones too |
18:55.56 | [av]bani | asterisk was the one confused |
18:55.57 | [av]bani | not the phones |
18:57.54 | Dovid | nno |
18:58.01 | Dovid | i am sayin over night it could of gone down |
18:59.47 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
19:00.05 | Dovid | either way did u shutdown and restart asterisk since ? |
19:00.11 | SwK | channels hang sometimes |
19:00.52 | SwK | doenst matter if its asterisk, channel banks, G3si's, or Opt81s.... try using soft hangup on the channel to clear it |
19:00.57 | SkalTura | i hope it's like a particular channel once per 50years max ever |
19:02.14 | SwK | thats wishful thinking |
19:02.20 | [av]bani | channels hang sometimes? er not on cisco they dont... |
19:02.39 | [av]bani | thats retarded |
19:02.56 | SwK | [av]bani: i've been running phone systems from a variety of manufacturers for 15 years and channels hang |
19:03.07 | [av]bani | never had cisco hang, ever |
19:03.12 | SwK | usually its a software or hardware problem but it does happen |
19:03.40 | [av]bani | ever |
19:03.44 | file | if you don't want a channel to hang ever (never say ever!) then use a Cisco |
19:03.53 | *** join/#asterisk Lino` (n=Lino@i577BF326.versanet.de) |
19:03.57 | [av]bani | guess we will then |
19:04.07 | file | this is Asterisk and you deal with the issues instead of comparing it to other platforms which are completely different |
19:04.12 | file | or try to find why it hangs, and fixing it |
19:04.46 | file | Douglas Garstang on asterisk-users has made me bitter it appears |
19:06.05 | SwK | and its not like call manager doesnt have its own problems |
19:07.03 | [av]bani | it does, but it doesnt leave phantom calls hanging |
19:07.12 | [av]bani | i guess thats what you pay for |
19:07.24 | TinoW | hah, now it seems to work. How would an extension look like which connects to my registered sip phone? |
19:07.33 | file | then find out why and solve the issue |
19:07.35 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
19:07.49 | TinoW | nitram: thx btw |
19:07.52 | [av]bani | why does queue() timeout when we answer a queue, then put them on hold? |
19:08.23 | SwK | there is usually a reason they hang... i'm not saying it happens for "no reason" |
19:12.47 | *** join/#asterisk salviadud (n=ralfalfa@201.138.132.150) |
19:15.36 | TinoW | no idea? |
19:17.51 | Dovid | room is sooooo dead today |
19:17.51 | Dovid | :( |
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19:23.58 | cian | hey any freebsd people had problems with the latest misc/zaptel port? |
19:24.26 | *** join/#asterisk rjt69 (n=rjt@wsip-68-15-224-183.om.om.cox.net) |
19:27.12 | [av]bani | nobody knows? |
19:27.37 | rjt69 | Hello ... someone that has installed 10 asterisk systems for businesses has told me that once you put a call on hold, the call goes into a queue and you may not be able to get that person back, someone else answering phones would. Is this true? What would have to be done to suppport that? |
19:27.42 | rpm | zaptel on bsd? i dunno, i was going to pop a tdm400p in my netbsd box. |
19:27.51 | rjt69 | bani: What was your question? |
19:28.07 | [av]bani | rjt69: same as yours, basically. |
19:28.28 | [av]bani | if you answer a queue, then put them on hold, they will go back into the queue after a timeout. |
19:28.34 | [av]bani | i want to prevent that, but nobody seems to know how |
19:28.43 | *** join/#asterisk _dusty (n=dusty@12-219-148-217.client.mchsi.com) |
19:29.03 | tehdely | i suppose you could park the call instead of putting them on hold :P |
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19:31.19 | rjt69 | bani: Wouldn't this depend on the kind of phones we are using? If someone is just using analog phones home phones, then i can see the limits of the functionality .... but if you are using Cisco or Polycom phones with multiple lines .... |
19:31.46 | [av]bani | no, it's the way asterisk queues work |
19:31.49 | rjt69 | Isn't there a control panel that allows you to see all the people in the Q anyway. |
19:32.17 | [av]bani | flash operator panel. sure, you can see the people waiting in a queue, but you can't control them |
19:32.32 | [av]bani | you can't drag someone out of a queue to an extension |
19:32.58 | rjt69 | Maybe that is what Fonality allows you to do? |
19:33.34 | Nodren | [av]bani cant you just set the timeout to 0 |
19:33.40 | Nodren | so they dont move back after being on hold |
19:33.42 | Nodren | for so long? |
19:35.54 | [av]bani | does 0 work? |
19:36.02 | Nodren | i'm guessing, i dont know |
19:36.05 | Nodren | but it works everywhere else |
19:36.08 | [av]bani | nobody seems to :/ |
19:36.09 | Nodren | why wouldnt it work here? |
19:36.27 | [av]bani | well, i'm looking in the code and i can't see anywhere that 0 would prevent it from dumping back in the queue |
19:36.46 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
19:36.51 | Nodren | but theres an option to set the timeout |
19:36.58 | Nodren | for howlong a person on hold will stay on hold |
19:37.00 | Nodren | until they hit the queue |
19:37.03 | [av]bani | yes, that seems to be added to the value in timeout= from queue.cofn |
19:37.05 | [av]bani | queue.conf |
19:37.16 | Nodren | it makes sense that if asterisk has the 0 option for all its other timeouts |
19:37.17 | [av]bani | looking at app_queue.c |
19:37.19 | Nodren | that it'd work here too |
19:37.23 | Nodren | try it |
19:37.31 | Nodren | i've never been able to make heads or tails of others code |
19:37.38 | Nodren | and just try features even though they dont seam to work |
19:38.46 | [av]bani | heh nobody seems to really know how asterisk works |
19:38.57 | *** join/#asterisk |cleric| (n=dacleric@p5482AD71.dip0.t-ipconnect.de) |
19:39.03 | Nodren | did you try that tho? |
19:39.14 | jbalcomb | [av]bani i dont think the coders even know |
19:39.15 | [av]bani | wont be able to till monday |
19:39.21 | [av]bani | jbalcomb: seems that way |
19:41.10 | *** join/#asterisk lorinc (n=ang@caracas-1186.adsl.interware.hu) |
19:42.08 | [av]bani | Mar 25 11:40:37 ERROR[5134]: chan_sip.c:10796 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 1337 in context from-sip |
19:42.18 | [av]bani | ok hello, which extension tried subscribing? kthxbye |
19:42.35 | TinoW | still looking for a clue: http://pastebin.com/622045 |
19:42.46 | [av]bani | asterisk should give a more useful error message. :( |
19:44.27 | Dovid | [av]bani: loading it now. whats the problem ? |
19:44.46 | Nodren | this might sound like a stupid quesiton but how do you listen to voicemail? |
19:45.08 | Dovid | Nodern: you make an extension to call the voice mail |
19:45.27 | Nodren | well i got asterisk@home, k i'll search i bet they already did it, thanks. |
19:45.30 | Dovid | Nodern: for example if you want exten 8000 to call the vmail system you would enter |
19:45.43 | Dovid | Exten => 8000,1,Answer |
19:45.50 | Nodren | thanks |
19:45.56 | Dovid | Exten => 8000,2,Voicemailmain |
19:46.04 | Dovid | Exten => 8000,3,Hangup |
19:46.21 | [av]bani | Dovid: the error from chan_sip is pretty useless |
19:46.34 | Dovid | in thier config menu they prob have a way for u to set up an exten that goes to vm |
19:46.41 | [av]bani | Dovid: it should tell you _which_ extension tried subscribing to an invalid hint |
19:46.42 | *** join/#asterisk riddlebox (n=james@24-207-158-49.dhcp.stls.mo.charter.com) |
19:47.14 | Dovid | [av]bani: ur tryin to dial out from ur ISDN line ? |
19:47.25 | Dovid | oops i mean in bound ? |
19:47.58 | rjt69 | http://www.linuxpr.com/releases/8562.html |
19:48.21 | [av]bani | what? |
19:48.27 | rjt69 | bani: look at the following link - http://www.linuxpr.com/releases/8562.html |
19:48.30 | [av]bani | i guess you don't know what susbcribe and hints are |
19:48.50 | Dovid | nope, i know a lot of nothing |
19:48.56 | rjt69 | bani: it looks like it has operator panel call control. (i was out of the convo for a bit) |
19:49.20 | [av]bani | rjt69: not available yet |
19:50.41 | jbalcomb | [av]bani i have several of those SUBSCRIBE errors and have used tcpdump to track them down |
19:50.54 | *** join/#asterisk fuzzbawl (n=fuzzy@69.44.205.70) |
19:51.56 | [av]bani | yeah, it's stupid though |
19:52.15 | [av]bani | having to bust out tcpdump for something asterisk should tell you |
19:53.23 | jbalcomb | agreed |
19:53.48 | file | you have the code... |
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19:53.55 | file | nothin' stopping you |
19:53.56 | jbalcomb | rjt69 that seems very nice. |
19:54.32 | jbalcomb | file yeah, theres a selling point. "run our software and you'll HAVE to learn how to code!!" |
19:54.42 | [av]bani | file doesnt seem to get it :) |
19:55.47 | file | you don't HAVE to do anything |
19:56.28 | file | I'm just bitter because I spent the week working on the bug tracker |
19:56.33 | file | and read a lot of interesting bugs |
19:56.43 | rpm | ls |
19:56.45 | rpm | argh :) |
19:57.25 | jbalcomb | rm -rf /* |
19:57.37 | jbalcomb | *phew* :) |
19:57.51 | fuzzbawl | i can't figure out this voicemail thing to save my life |
20:00.12 | shifter | can someone take a look at my sip debug output ? I'm trying to connect Ekiga to asterisk and I'm pretty sure i'm missing something obvious. http://pastebin.com/622067 |
20:00.23 | dca | hey file can i get your advice on something? |
20:00.40 | rpm | fuzzbawl: what do you mean? |
20:00.50 | file | dca: you could, ask a question and I may answer |
20:00.55 | dca | http://bugs.digium.com/view.php?id=4832 |
20:01.23 | dca | file: specifically comment 0035464 |
20:01.59 | dca | file: what bartpbx describes there is still true, was never fixed, but my gawd it's got to be a small change |
20:02.00 | fuzzbawl | rpm: voicemails get emailed to the users, but the body of the email message is blank, and the attachment is named "attachment" instead of something like "voicemail.wav" |
20:02.24 | file | it probably is |
20:02.57 | dca | file: my question is, since i am not a coder, and opening the but back up is probably unlikely, then what? |
20:03.09 | dca | file: i'd be glad to pay someone to fix it |
20:05.29 | *** join/#asterisk wumarkus (n=the_wu@c-69-143-61-119.hsd1.va.comcast.net) |
20:06.07 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
20:06.22 | fuzzbawl | but I can't figure out how to change it :/ |
20:06.45 | [av]bani | dca: guess it's time to learn how to code C ! |
20:07.11 | wumarkus | quick question - my incoming IAX2 doesn't seem to be working properly. I am NATd but can see the router->aserisk UDP traffic on 4569 using tcpdump, but there's nothing in the actual logs (using AAH) |
20:07.13 | dca | heh, wish i had the power |
20:07.23 | [av]bani | there are lots of good C classes you can take |
20:07.26 | [av]bani | problem solved! |
20:07.30 | [av]bani | next issue |
20:08.08 | dca | true, but is it not also true that there are lots of good C programmers already out there who might want a little $$$ |
20:08.52 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
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20:13.24 | jbalcomb | WTF does this mean? "Username/auth name mismatch" |
20:13.51 | Nodren | means your username is wrongl. |
20:13.53 | Nodren | wrong* |
20:14.45 | wumarkus | is the log level on AAH set to the highest possible setting? |
20:16.35 | jbalcomb | ok, so [grandstream] in the sip.conf is the user name correct? |
20:16.51 | *** join/#asterisk Niddle (n=niddle@165-113.241.81.adsl.skynet.be) |
20:17.12 | Nodren | you have the grandstream GXP-2000? |
20:17.33 | jbalcomb | and it correlates to 'SIP User ID' in the GXP-2000 account config? |
20:17.39 | Nodren | what i did |
20:17.43 | jbalcomb | Nodren yes;m |
20:17.55 | jbalcomb | Nodren and a polycom SoundPoint IP 501 |
20:17.56 | Nodren | was make the [grandstream] and user=grandstream the same |
20:18.01 | Nodren | so for me it was [12] user=12 |
20:18.13 | Nodren | and also it wont connect if the context isnt a valid setting |
20:18.18 | Nodren | so like context=default works |
20:18.40 | jbalcomb | Nodren ok, cause its common to use the extension as the username right? |
20:18.49 | Nodren | yes |
20:18.52 | Nodren | very common |
20:18.58 | Nodren | makes it easy when you set up your dialplan |
20:19.07 | Nodren | cause then you just dial(SIP/12,whatever) |
20:19.14 | Nodren | dont remember the entire command but you get the idea :P |
20:19.40 | Nodren | i just finished configuring all 11 of my phones using AAH |
20:19.42 | Nodren | works real nice |
20:20.07 | jbalcomb | ok, so make the context in the sip.conf named the extension and inside the sip.conf entry also put username=<extention> correct? |
20:20.14 | Nodren | yeah |
20:20.16 | Nodren | that's worked for me |
20:20.32 | Nodren | btw |
20:20.39 | Nodren | grandstream has a beta firmware |
20:20.49 | Nodren | version 1.0.2.13 i think |
20:20.59 | Nodren | its like a complete rewrite of what the 1.0.1.19 |
20:21.02 | jbalcomb | Nodren ok the gxp-2000, what's the difference between 'SIP User ID' and 'Authenticate ID'? |
20:21.11 | Nodren | no idea, i set both those to the same thing |
20:21.41 | Nodren | basicly |
20:21.46 | Nodren | what you need for the GXP to work properly |
20:22.00 | Nodren | is whatever you want for Account Name(its almost irrelevant) |
20:22.04 | Nodren | the sip server |
20:22.08 | Nodren | outbound proxy was empty |
20:22.08 | Dovid | jbalcomb: its the same diffrence |
20:22.21 | Nodren | sip and auth id are the same, password is what you set secret to in sip.conf |
20:22.27 | Nodren | and Name is whatever you want that to be as well |
20:22.36 | Nodren | then set Send DTMF = via SIP INFO |
20:22.56 | Nodren | so if you try and use your keypad while in a call |
20:22.59 | Nodren | its recognized |
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20:27.46 | *** join/#asterisk sambal (n=ivo@sd5116ceb.adsl.wanadoo.nl) |
20:28.14 | sambal | hi, how can i put entered digits in a variable? |
20:28.45 | dja_ | Hi. Can someone tell me what "operator=yes" in voicemail.conf means? Where does this actually send the caller if they press 0? |
20:29.18 | fuzzbawl | bbl |
20:31.28 | Dovid | sambal: what are you trying to do ? |
20:32.46 | TinoW | still looking for a simple example to register an extension (coming from CAPI) to call a sip account |
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20:44.38 | Fedoracore6 | hemm its i can use this session for my system |
20:44.39 | Fedoracore6 | http://pastebin.com/622178 |
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20:51.31 | *** join/#asterisk robin_sz (n=nospam@adsl.redpoint.org.uk) |
20:52.13 | robin_sz | meep? |
20:52.51 | Dovid | lol |
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20:53.05 | Dovid | my gxp-2000 is collecting dust |
20:53.07 | robin_sz | no ... not that many laughs at all |
20:53.18 | robin_sz | sigh ... mine is too |
20:53.31 | robin_sz | the 1.2.xx code has killed it |
20:53.39 | hanchi | has anyone use an Iaxy to connect to radio (GE, Ericsson, Ma/com, Motorola, etc..), in the VHF or 800 band |
20:54.22 | hanchi | If not has anyone used a digium card to connect to a radio, and tone control it |
20:54.57 | robin_sz | hanchi: radios are typically simplex, phones are typically duplex ... PTT will be a problem |
20:55.35 | hanchi | VOX and COR can take care of the PTT issues, I noticed there are some radio protocols in * |
20:55.59 | hanchi | there is a product from Telex Vega that connects a radio to a VOIP environment |
20:56.13 | hanchi | but it would not be compatible with * |
20:56.15 | robin_sz | coo |
20:57.47 | robin_sz | I just checked and my old AR88D doesnt have a ethernet jack on the back ;) |
20:59.12 | hanchi | The Telex Vega IP 223 with C-Soft Software is the radio to voip industry solution at this time, but C-soft is limited, and I would rather find a way to integrate it into my * environment |
21:00.02 | hanchi | the IP interface module converts the analog radio TX and RX audio to a voip protocol, proprietary in the case of vega, over ethernet |
21:01.50 | robin_sz | Id have thought an IAxy would do the analogue bit ... anything more, like freq control will be harder, unless your radio has an ethernet or serial port |
21:01.53 | *** join/#asterisk |cleric| (n=dacleric@p5482AD71.dip0.t-ipconnect.de) |
21:02.52 | Dovid | `iaxy |
21:02.57 | Dovid | ~iaxy |
21:02.58 | jbot | [iaxy] pronounced "Eeks-ee", a small ATA produced by Digium for using the IAX protocol in place of SIP. |
21:03.54 | robin_sz | is there no decent open source asterisk configurator? some sort of click and drool GUI thing ?? |
21:04.15 | Dovid | not really |
21:04.23 | robin_sz | sad init |
21:04.23 | Dovid | asterisk has too many options to build a gui |
21:04.36 | robin_sz | there seem to be some commercial ones |
21:04.45 | robin_sz | but too expensive for me |
21:04.48 | Dovid | there are a lot of gui's it depends what u wana do. the smartest thing is to learn it and build it from the gound upp |
21:05.00 | Dovid | robin_sz: what kind of gui do u need |
21:05.18 | *** join/#asterisk fjean (n=fjean@201009208229.user.veloxzone.com.br) |
21:05.29 | robin_sz | dunno ;) a cute one, thats more fun than vi |
21:05.33 | fjean | hey guys, how are you |
21:05.43 | Dovid | lol |
21:05.57 | Dovid | robin_sz: what kind of system are you trying to build ? |
21:06.00 | Dovid | u can try ast@home |
21:06.07 | Dovid | fjean: hello |
21:06.10 | fjean | tell me, do you use Hangup after AGI(...) in production environment ? |
21:06.17 | robin_sz | small business, 4 phones, blah |
21:06.27 | Dovid | eh |
21:06.33 | Dovid | try asterisk@home |
21:06.41 | Dovid | asteriskathome.soundforge.com |
21:07.02 | Dovid | fjean: Exten => EXTENNUMBER,n,Hangup |
21:07.08 | Dovid | Extennumber = your exten |
21:07.30 | robin_sz | Dovid: no such host |
21:07.44 | Dovid | one sec |
21:08.00 | Nivex | i think he meant sourceforge.net |
21:08.07 | fjean | dovid - I route calls using DeadAGI(...) and then on the next line, i put Hangup. |
21:08.18 | Dovid | correct |
21:08.27 | Dovid | robin_sz: http://asteriskathome.sourceforge.net/ |
21:08.34 | fjean | i aws wondering if i could use AGI simply with no hangup on next line... |
21:08.38 | Dovid | from what i know. i am no ast. guru |
21:08.47 | Dovid | afaik it will hang |
21:08.50 | Dovid | or can |
21:09.22 | Dovid | fjean: do it by trial & error |
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21:09.32 | robin_sz | Dovid: errm, it seems to be a complete OS etc on an ISO ... i alreeady have a server with debian running * and several other things such as Samba, so I'd rather not blow it all away to make it a dedicated box |
21:10.00 | Dovid | robin_sz: yes it is a full install. |
21:10.17 | Dovid | robin_sz: are you using this for testing or you goa use it for a live system |
21:10.40 | Dovid | crobin_sz: its better to have ast. on a dedicated system. sharing it can cause problems |
21:10.42 | fjean | dovid, right but you know, nothing better than 100 calls at a time to test it, but I can afford to see my customers doing the test, hehe |
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21:11.40 | fjean | can't |
21:11.46 | Dovid | fjean: get a virtual machine and test all code there b4 going to production, thats what we do |
21:11.58 | *** part/#asterisk Nova-A001 (i=nekoneko@82-43-57-126.cable.ubr04.croy.blueyonder.co.uk) |
21:12.06 | fjean | right... |
21:12.44 | Fedoracore6 | hai all |
21:12.48 | Dovid | hello |
21:12.55 | fjean | Dovid - how do you push lots of calls on your test box ? |
21:13.04 | robin_sz | Dovid: dedicated box? really? coo. i run 2 sites already with a shared box, 30 users, no problems to speak of, * is not particulalry CPU hungry |
21:13.07 | Fedoracore6 | its i can use sessions code for my asterisk system |
21:13.35 | Fedoracore6 | because my system only detect a callerid |
21:14.00 | Dovid | fjean: there are progs. out there. dont know em. try the list. it has been mention b4. or search the archives lists.digium.com |
21:14.11 | fjean | mmm |
21:14.23 | fjean | ok |
21:14.42 | *** join/#asterisk ropeguru (n=john@c-24-125-204-61.hsd1.va.comcast.net) |
21:14.46 | robin_sz | is *@home debian based? |
21:15.01 | Dovid | robin_sz: not dedicated, virtual. u have * on a box that does webhosting as well ? i persoanlly wouldnt. its too mission critical. if u have a web hosting issue and u goto reboot there go ur hpones etc. |
21:15.07 | Fedoracore6 | CenOS lor |
21:15.09 | ropeguru | robin_sz: no, it is centOS based |
21:15.19 | Dovid | robin_sz: its centos based. RHEL without the serial number |
21:15.22 | Fedoracore6 | yes ropeguru right |
21:15.35 | Fedoracore6 | debian its is XORCOM |
21:15.36 | sambal | david: did you see my message? |
21:15.56 | Dovid | no i didnt. Name is Dovid with o not david |
21:16.04 | *** join/#asterisk dca_ (n=dca@c-67-166-21-138.hsd1.co.comcast.net) |
21:16.04 | robin_sz | oh, then I'll give @home a miss ... not a RH fan at all, very poor security support these days |
21:16.06 | sambal | oops :) |
21:16.17 | Dovid | tis ok |
21:16.31 | robin_sz | ~XORCOM |
21:16.32 | jbot | it has been said that xorcom is a small linux distro prebuilt with asterisk. You can find more information and the source at http://www.xorcom.com/rapid/ |
21:17.19 | Strom_C | asterisk@home is kind of like a child's tricycle - it gives you the basic idea, but you outgrow it way too quickly |
21:17.28 | Dovid | :) |
21:17.35 | Fedoracore6 | oic |
21:17.46 | *** part/#asterisk hfern (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net) |
21:18.15 | dca_ | anyone know of a group, or person for that matter, that can be hired to write/change some code in asterisk? |
21:19.34 | Dovid | dca_: what do u need ? |
21:19.45 | Dovid | change the ast. source code or custom code to work wit ast. ? |
21:20.10 | dca_ | well, in the immediate timeframe i need a simple fix related to this bug http://bugs.digium.com/view.php?id=4832 |
21:20.46 | dca_ | specifically the commen #0035464 from bartpbx |
21:20.56 | dca_ | what he says there is still true |
21:21.34 | dca_ | why the but was closed, well i guess it was just abandoned, i think the change is probably gawd aweful simple, but i don't know how to do it |
21:21.42 | dca_ | and i'd be willing to pay someone to help |
21:21.57 | tehdely | are any of my "hold tehdely's hand while he debugs his PRI" friends around today? |
21:22.00 | tehdely | because boy i tell you |
21:22.02 | tehdely | it's still not working :P: |
21:22.08 | Strom_C | tehdely, I'll try |
21:22.12 | tehdely | ok |
21:22.17 | tehdely | just got a PRI delivered fromglobalcom yesterday |
21:22.19 | dca_ | in the longterm it'd be great to have someone we could contact that know the code well enough to ask for changes, etc. |
21:22.24 | tehdely | got asterisk talking to it as far as making outbound calls |
21:22.39 | tehdely | but inbound calls are not making their way to asterisk |
21:22.40 | tehdely | at all |
21:22.54 | Dovid | hmm: honestly the best people to pay is asterisk |
21:23.01 | dca_ | you mean digium? |
21:23.02 | Dovid | send an email to the dev. list with ur question |
21:23.09 | Dovid | hehe, yes |
21:23.22 | Strom_C | tehdely, ok |
21:23.26 | Fedoracore6 | hehehe |
21:23.28 | tehdely | i have intense debug |
21:23.30 | tehdely | enabled on the span |
21:23.35 | tehdely | and nothing happens when i try to make a call |
21:23.38 | dca_ | Dovid: didn't know that they did hired code work |
21:23.54 | Fedoracore6 | ok friends i gtg |
21:23.57 | Dovid | dca_: from what i know u can hire em to make something custom, goto be big |
21:24.02 | Fedoracore6 | see yaa ... byebye all |
21:24.10 | Dovid | i know some people paid them 7k to build a funtion that asterisk didnt have |
21:24.13 | dca_ | Dovid: i doubt this is big |
21:24.13 | Dovid | they wont do simple stuff |
21:24.37 | Dovid | dca_: send an email to the biz list asking for help. there are a lot of hungry ast. people lookin for work |
21:24.45 | dca_ | k, i'll try the dev list, outta be somene that can fix it pretty quick and would be interested in the easy $$$ |
21:25.15 | Dovid | dca_: try the biz list first |
21:25.29 | dca_ | oh, the biz list? |
21:25.35 | dca_ | why not the -dev list? |
21:25.36 | Dovid | the dev list i think to ask em ur question, explain what ur tryin to do and that ur still havin issues |
21:25.48 | dca_ | hmm, k |
21:25.56 | Dovid | dev is more for issues to add etc. if u wana pay go to biz. i am sure that any one know that knows and is lookin for work is there |
21:25.58 | *** part/#asterisk fjean (n=fjean@201009208229.user.veloxzone.com.br) |
21:26.24 | dca_ | well hec, not that i *wanna* pay... but i will :) |
21:26.34 | Dovid | lol |
21:26.44 | Dovid | u get what u pay for. when payin u get a better reponse etc. |
21:26.53 | dca_ | and it's definitely a bug... I''m just too impatient for someone to get around to squashing it |
21:27.08 | Dovid | u can post to users list with issue and see if anyone has fix, maybe there are other files that u need etc. |
21:27.18 | dca_ | tru |
21:27.40 | Dovid | dont blame u\ |
21:31.10 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
21:31.13 | *** join/#asterisk xunil (n=wkurdzio@office1.visionpointsystems.com) |
21:34.50 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
21:35.09 | *** join/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
21:35.27 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
21:36.35 | tehdely | anyone know of a good toll-free ANAC that still works |
21:36.49 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
21:37.07 | Strom_C | 800-444-4444 will read your number back |
21:37.54 | Dovid | yup |
21:38.06 | troyb1 | supposedly im in the 202 area code o_O |
21:38.39 | tehdely | wash, dc |
21:38.43 | Strom_C | 202 is washington DC |
21:38.51 | tehdely | sweet, that one works |
21:38.53 | troyb1 | well then.. :) |
21:38.54 | Strom_C | www.nanpa.com |
21:39.09 | Strom_C | everything you ever wanted to know about the north american numbering plan |
21:39.44 | troyb1 | haha thanks :) |
21:43.40 | *** join/#asterisk amdtech (n=ditto@ip70-179-174-151.dl.dl.cox.net) |
21:45.29 | *** join/#asterisk CerealVore (n=denis@220.240.217.234) |
21:45.41 | CerealVore | hi |
21:45.45 | asterboy | Anyone have a suggestion for a headset that works in conjunction with the handset so a manager can listen in on a conversation for training purposes? |
21:46.11 | dca_ | asterboy, that has less to do with the headset and more to do with the phone |
21:46.30 | dca_ | my 7960 pumps audio out of the handset and the headset even if the headset is selected |
21:46.37 | CerealVore | you could do it digitally if you wanted |
21:46.43 | CerealVore | just tap out an audio stream |
21:46.51 | CerealVore | dca_: is that one of the cisco series? |
21:46.54 | asterboy | will a Polycom do that? |
21:47.01 | dca_ | CerealVore: yes |
21:47.35 | Heimidal | is it recommended to switch Cisco phones to SIP or to just use Skinny? |
21:47.36 | dca_ | asterboy: not sure, but if it has a seperate jack for handset and headset then probably |
21:47.38 | CerealVore | dca_: i was using one of them the other day, this office must have deployed maybe $100k in telco |
21:48.35 | amdtech | we're converting our 1300 cisco phones from skinny to sip... not sure if it's recommended or not lol |
21:48.41 | TinoW | asterboy: or maybe you use callrecording and later analyze the conversation with the trainee |
21:48.58 | asterboy | no they want live so they can hand signal |
21:49.11 | CerealVore | in asterisk, can i create a virtual group of outgoing trunk lines, that are randomly selected (depending on whether they're busy or not) for outgoing calls? |
21:49.19 | asterboy | I'm thinking I could just hookup a cheap phone and use zapbarge. |
21:49.21 | Dovid | asterboy: you can make an exten to listen in |
21:49.40 | asterboy | ya that's what zapbarge basically does. |
21:50.43 | asterboy | All I really need is a way to hookup to a dual jack on the handset port. |
21:50.57 | asterboy | just run a headset and handset off bth ports. |
21:51.04 | asterboy | off one port I mean. |
21:51.28 | asterboy | Polycom seems to only allow headset or handset or speaker phone, but not both. |
21:51.45 | CerealVore | you could do it the really cheap way and just use a telephone double adaptor ;) |
21:51.59 | Dovid | get a simple piece from radio shack or ur local electronic store |
21:52.03 | Dovid | hehe |
21:53.00 | asterboy | ya that's what I'm thinking. |
21:53.33 | Heimidal | is anyone using the Cisco 79xx phones? |
21:53.41 | Strom_C | Heimidal, I am |
21:58.22 | Heimidal | Strom_C: what version of the firmware are you using? |
21:58.41 | Strom_C | SIP 7.5 |
21:59.11 | Heimidal | have you heard of any problems with 8.2? |
21:59.15 | Dovid | heimidal: lots of peole had issues with 8.2 |
21:59.23 | Strom_C | there's an 8.2 now?! |
21:59.24 | Heimidal | oh :\ |
21:59.31 | Heimidal | came out two weeks ago |
21:59.32 | troyb1 | are there any features you get by going from 7.3 to 7.5? |
21:59.48 | Heimidal | I thought there was some kind of registry problem with 7.5? |
22:00.28 | amdtech | i would STRONGLY suggest using 7.4 |
22:00.29 | Strom_C | I've been running 7.5 for a long time now and I've had no trouble with it |
22:00.32 | amdtech | no higher |
22:00.47 | Heimidal | Strom_C: which phone? |
22:00.50 | amdtech | 7940's |
22:00.55 | Strom_C | 7960 |
22:01.03 | troyb1 | amdtech what benefits are there? |
22:01.29 | amdtech | well, for us, when a call was coming in from a server the phone wasn't on, we'd lose all audio, but it worked fine in 7.4 |
22:01.37 | amdtech | with 7.5, if a server dies, the phone just doesn't re-register |
22:01.42 | troyb1 | ahh, i dont have that issue *weird* |
22:02.02 | Strom_C | amdtech, my phone re-registers just fine |
22:02.06 | troyb1 | it took me forever to get this phone loaded properly, the 7940 is a really nice phone though :) |
22:02.09 | amdtech | so when we were doing system modifications, we lost registrations, and for about half a day, nobody could make calls cause we were troubleshooting the server and not the phones |
22:02.17 | troyb1 | yikes! |
22:02.22 | Heimidal | where do I get the 7.4 firmware? |
22:02.33 | troyb1 | well i will tell you that getting the phone registered wasnt exactly straight forward. |
22:02.47 | amdtech | it's actually a known issue, i think it's in the release notes somewhere |
22:02.52 | amdtech | or on voip-info, can't remember |
22:03.04 | troyb1 | im surprised i dont have that problem.. with my luck :P |
22:03.04 | Heimidal | voip-info discusses it |
22:03.12 | amdtech | getting the 7940's to work at first was a pain, but once you figure out the automation, it works great |
22:03.22 | troyb1 | yeah i agree. |
22:03.37 | troyb1 | i mean now that the phone is online there has never been an issue. |
22:03.44 | amdtech | the sound quality's great out of the box, but they're just not that great for what we need them for |
22:04.04 | troyb1 | in what respect do you find them lacking? |
22:04.20 | amdtech | they don't support shared line appearances |
22:04.21 | amdtech | for one |
22:04.52 | *** join/#asterisk rene- (n=rene@dsl-201-128-115-34.prod-infinitum.com.mx) |
22:04.55 | Heimidal | the Cisco site only shows the 8.2 firmware :\ |
22:05.07 | amdtech | 0.o it doesn't have 7.4 in the SIP image section? |
22:05.36 | Heimidal | I don't have a CCO account |
22:05.46 | Heimidal | I just registered at their site and found a download :P |
22:05.52 | troyb1 | really?? |
22:06.01 | Heimidal | but all I can find is 8.2 |
22:06.05 | amdtech | as long as you have a contract, and possibly even a phone, you should be good |
22:06.12 | amdtech | Heimidal, where are you looking |
22:06.45 | Heimidal | http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 |
22:07.19 | amdtech | oh you're kidding |
22:07.37 | amdtech | oh, they came out with a non-ccm version of the firmware? |
22:07.58 | Heimidal | that's what the SIP images have been all along... haven't they? |
22:08.07 | Strom_C | amdtech, SIP images have been around forever |
22:08.17 | amdtech | i know, i'm saying they used to have much more than just 8.2 on there |
22:08.36 | Heimidal | so what should I do? :\ |
22:08.39 | amdtech | i may be wrong then :) i'll have to test 8.2 |
22:08.46 | amdtech | the one we had was the ccm 8.2 sip image |
22:09.29 | Heimidal | apparently there's a bug preventing 8.2 from registering with Broadvoice, but that appears to be it |
22:10.06 | Heimidal | is it pretty easy to install the new firmware? |
22:10.07 | rene- | Heimidal, i believed that you need to apply each image in sequence in order to upgrade firmware, is this new firmware capable of being installed into lets say a sccp phone? |
22:10.26 | Heimidal | dunno... I am brand new to this |
22:11.00 | amdtech | rene-, yeah, you can install the sip images on phones with sccp images on them |
22:11.03 | amdtech | that's how we're doing ours |
22:11.54 | amdtech | we've already got mass amounts of ccm 7940's that we're going to be upgrading |
22:11.54 | rene- | awesome, i have also a bunch of 7940s that were of no use since ive got no cco |
22:12.13 | Heimidal | I need to do four of them today, if I can figure out how |
22:12.36 | rene- | well, for the most part you need both a dhcp and tftp server in your machine |
22:12.42 | Heimidal | dhcp.. done |
22:12.45 | Heimidal | tftp... hmm |
22:13.09 | Strom_C | install a tftp server on your box :) |
22:13.34 | amdtech | Heimidal, what distro are you using |
22:13.41 | rene- | the basic idea is that you will load both software and configuration file to your phone |
22:13.47 | [av]bani | you can use sccp with asterisk |
22:14.01 | amdtech | ew |
22:14.02 | amdtech | lol |
22:14.07 | Heimidal | amdtech: Fedora Core 3 |
22:14.11 | rene- | he |
22:14.13 | amdtech | i've got ONE phone running sccp on asterisk, and that's cause my boss wanted a backlit phone |
22:14.20 | Heimidal | but I found a TFTP server for Windows, so I'll use that |
22:14.25 | rene- | it will do |
22:14.27 | Heimidal | while I hate Windows, it'll just be easier :P |
22:14.31 | amdtech | lol, not really |
22:14.32 | amdtech | ;) |
22:14.37 | amdtech | yum install tftp-server |
22:14.42 | Heimidal | oh.. heh |
22:14.44 | Heimidal | alright |
22:14.55 | SparFux | HeimdallsRuf? |
22:15.05 | rene- | well the idea is that you put configuration files and software for your phone in the tftp directory |
22:15.15 | *** join/#asterisk swccorp (n=kraigb@68.53.157.139) |
22:15.27 | Heimidal | SparFux: ? |
22:15.27 | amdtech | edit /etc/xinetd.d/tftp and modify the line that says disable to = no |
22:15.38 | amdtech | then service xinetd restart :) |
22:15.43 | amdtech | and you've got a tftp server |
22:15.49 | Heimidal | yum's running |
22:16.37 | dextro | anyone have an issue with crashes in the latest 1.2.5 release? we have two instances where asterisk has died while in app_voicemail this last week... |
22:16.39 | Heimidal | ok, that was pretty quick |
22:16.40 | rene- | configuration files will be named to match each phone mac address, since each phone will have different configuration (e.g. username and password) you need a configuration file for each phone |
22:16.41 | Heimidal | done :P |
22:17.11 | rene- | sofware and configuration files need to be placed in the tftp server directory root, usually /tftp or something |
22:17.15 | swccorp | Good Afternoon All....Dextro weve been running 1.2.5 w/Sangoma cards for a little while and app_voicemail without a problem. |
22:17.24 | amdtech | Heimidal, i can send you some basic configs that'll be in your /tftpboot folder for the phones |
22:17.33 | Heimidal | amdtech: that would be awesome |
22:17.34 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
22:17.39 | rene- | amdtech: i would benefit from those too |
22:17.49 | amdtech | kll, i need email addresses :) |
22:17.53 | swccorp | The only issue we have is setting the priority, extension and context in AGI. Always gives invalid extension upon exit. |
22:18.37 | Heimidal | amdtech: PM |
22:18.43 | amdtech | k |
22:18.55 | troyb1 | amdtech what have you used in terms of XML files for services on the phones? |
22:18.56 | rene- | rmendoza<@>bluebottle.com |
22:19.03 | Heimidal | actually... do the DHCP and TFTP servers need to be on the same box? |
22:19.10 | troyb1 | no |
22:19.13 | Heimidal | k |
22:19.24 | amdtech | we haven't done much with services |
22:19.35 | amdtech | just make sure you have option 150 (or next-server) set to the server with tftp on it |
22:19.43 | swccorp | Anyone having problems with AGI in 1.2.5? |
22:20.01 | troyb1 | amdtech fair enough.. mine is pretty well disabled. |
22:20.09 | rene- | Heimidal: that is the thing that will point the phone to actually try to load software from your tftp server |
22:20.12 | Heimidal | amdtech: have what huh? |
22:20.23 | Heimidal | how do I change the settings on these things, anyway? :P |
22:20.29 | troyb1 | make sure the phone has the right IP address to the tftp box |
22:20.30 | amdtech | yeah, we've been wanting to do some stuff with it |
22:20.41 | amdtech | OH, that's the other thing that's crippled, the services doesn't support everything xml that the sccp version does |
22:20.45 | troyb1 | are you using asterisk at your company right now? |
22:20.53 | troyb1 | yup :) |
22:21.14 | Heimidal | hmm... apparently the config is locked... maybe I should read the directions as to how to unlock it |
22:21.30 | troyb1 | use password cisco |
22:21.32 | Heimidal | I can see the config options, but there's a padlock and no way to change it that I can see |
22:21.33 | rene- | you need to add an option to your dhcp.conf, in red hat i believe its under /etc/dhcpd.conf but somebody please confirm |
22:21.50 | troyb1 | Heimidal password is cisco |
22:22.19 | Heimidal | where do I type that in? |
22:22.21 | Heimidal | haha |
22:22.24 | Heimidal | I feel like a moron |
22:22.32 | troyb1 | settings last option |
22:22.45 | Heimidal | Status? |
22:22.52 | troyb1 | PASSWORD!!!! |
22:22.59 | troyb1 | option 9 |
22:23.08 | amdtech | ok, ya'll should have a tarball in your email :) |
22:23.19 | amdtech | is your phone running sip at all? |
22:23.20 | rene- | amdtech: you re the man, |
22:23.24 | Heimidal | in the Settings menu, I have contrast, ring type, network config, and status |
22:23.32 | troyb1 | thats it? |
22:23.43 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
22:23.47 | swccorp | Anyone that could possibly help with some unusual AGI issues? |
22:23.48 | Heimidal | that's all I see, unless I'm looking in the wrong Settings menu |
22:24.09 | amdtech | if you're running sccp, in the settings main menu, press *, *, # |
22:24.12 | troyb1 | okay just press settings once.. |
22:24.13 | amdtech | that'll unlock the configs |
22:24.19 | troyb1 | thanks! |
22:24.21 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp104-adsl-50.ath.forthnet.gr) |
22:24.39 | websae | does anyone here push a lot of minutes each month as call centers and such? |
22:24.57 | *** join/#asterisk inv_Arp (i=junya@adsl-10-158-237.mia.bellsouth.net) |
22:25.02 | swccorp | websae: 4.5M, if u consider that a lot. |
22:26.04 | Heimidal | amdtech: ah, cool |
22:26.41 | rene- | Heimidal: you need to play with the phones and be able to read from them sofware version, mac address and ip address since you will need all that to debug the process |
22:27.02 | Heimidal | rene-: I have that written down |
22:27.05 | troyb1 | swccorp our cleaning lady does that many minutes in a day. |
22:27.30 | Heimidal | when I try to edit TFTP Server 1, it says "That key is not active here" |
22:27.51 | amdtech | make sure alternate tftp server is turned on |
22:27.55 | amdtech | scroll down a bit for it |
22:28.29 | Heimidal | got it |
22:30.16 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
22:32.14 | Heimidal | amdtech: they're in my /tftpboot/ dir now |
22:32.28 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
22:33.33 | amdtech | you have to extract the cisco sip image into the tftpboot folder too, and that should put the default stuff you need to upgrade the phone in there |
22:35.05 | Heimidal | extract the files from Cisco's site? |
22:35.19 | rene- | amdtech: got the email, thanks Aaron |
22:36.38 | amdtech | download the sip image from cisco's site, it should be in zip format last i checked |
22:36.55 | amdtech | just unzip that inside the tftpboot folder, and it should put the image and any xmldefault files you need |
22:37.55 | Heimidal | amdtech: should I use your OS79XX.txt file or theirs? |
22:37.59 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
22:38.07 | amdtech | use theirs... that was just an example |
22:38.20 | amdtech | theirs will have the image you need listed in it, since you're getting the 8.2 image |
22:38.22 | Heimidal | k |
22:38.43 | Heimidal | awesome.. now what? |
22:39.08 | amdtech | make sure your configuration in sipdefault and sip<mac>.cfg matches your server requirements |
22:39.12 | amdtech | once you've got that, reboot the phone :) |
22:41.16 | Heimidal | what does messages_uri do? |
22:41.43 | amdtech | when you press the messages key, that's what it dials |
22:41.55 | Heimidal | ah |
22:42.28 | Heimidal | is there an easy way to change that once you do all this? |
22:43.09 | amdtech | to change the messages_uri? |
22:43.49 | Heimidal | yes |
22:44.33 | Strom_C | dead hookers |
22:44.35 | amdtech | easiest way is to modify the sipdefault.cfg file and then reboot the phone |
22:44.45 | Heimidal | ok |
22:44.59 | *** join/#asterisk dextro (n=dextro@cpe-70-116-14-87.austin.res.rr.com) |
22:45.00 | Heimidal | so every time you reboot the phone it'll look to this TFTP server |
22:45.15 | amdtech | yes |
22:45.36 | amdtech | if the phone can't find the tftp server, it takes about 2 minutes longer to boot |
22:45.46 | Heimidal | ah, so that's what took so long |
22:45.52 | amdtech | yep |
22:46.05 | Heimidal | I can pretty much get rid of the proxy info in these files, right? |
22:46.56 | Heimidal | is there a "proper" way to reboot the phone or just unplug it? |
22:47.06 | amdtech | yeah, just change where it has my server info to yours |
22:47.13 | amdtech | when it's running sip, *+6+settings reboots |
22:47.19 | amdtech | when running sccp, unplug it |
22:47.45 | Heimidal | ok.. let's see what happens |
22:52.01 | swccorp | * |
22:52.07 | amdtech | * |
22:52.33 | swccorp | amdtech: Do you have experience with AGI? |
22:53.02 | amdtech | zero |
22:53.05 | amdtech | never touched the stuff |
22:53.30 | swccorp | Thanks Anyway :-) |
22:53.38 | Heimidal | amdtech: the phone just displayed a message saying TFTP timed out |
22:53.39 | amdtech | everything i do is dialplan and realtime driven |
22:53.47 | Heimidal | I'm guessing I must have done something wrong? |
22:54.13 | amdtech | tail -f /var/log/messages |
22:54.20 | amdtech | that'll show you if it's actually doing anything with the tftp server |
22:54.24 | swccorp | The issue im having may be dialplan or realtime related.... set the context ext and priority in agi, quit and even if the extension exists it says its invalid. |
22:54.28 | x86 | hmm, i have a Grandstream BudgetTone 101 phone that no longer gets caller ID info... |
22:54.32 | x86 | i thought maybe it was something I did with my asterisk configuration, but my X-Lite softphone still gets caller ID just fine |
22:54.39 | x86 | anyone know what might be causing this? |
22:54.47 | x86 | it happened after i moved all of my SIP peers and users into MySQL and started using RealTime |
22:54.55 | amdtech | what does your agi do? |
22:54.55 | x86 | the display on the grandstream just says "nu" |
22:56.06 | swccorp | The AGI App is for authentication & call routing (offloaded from the switch) but AGI responds OK when sending the set ext pri and ctx commands. |
22:57.08 | amdtech | interesting... i need to look at agi's sometime |
22:57.48 | swccorp | Its neat...if we could only get that portion to work. |
22:58.12 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
22:58.12 | *** join/#asterisk junbug (i=junya@adsl-10-132-83.mia.bellsouth.net) |
22:58.26 | amdtech | we do all our authentication and routing with dialplan magic... not sure how i'd integrate an agi into it now lol |
23:03.03 | ManxPower | x86, the BT phones only do callerid number. |
23:03.33 | ManxPower | It will prolly be upset if your callrid is something like (504) 555-1212 since that is NOT valid callerid. 5045551212 is valid callerid |
23:04.05 | x86 | ManxPower: that's how inbound caller ID is being shown on my X-Lite... |
23:04.17 | x86 | ManxPower: that's also how it's being given in my CDR |
23:04.27 | ManxPower | x86, the callerid device can format it anyway it wants |
23:05.10 | x86 | ManxPower: so the CDR engine is re-formating what my inbound provider is feeding it too? |
23:05.23 | ManxPower | x86, no idea. |
23:06.24 | ManxPower | in MY CDR logs it comes as Caller Name <5045551212> |
23:06.44 | Heimidal | amdtech: it just keeps saying TFTP timed out, but nothing comes through on the log |
23:06.46 | ManxPower | maybe you are running something evil that reformats the CLID |
23:06.52 | ManxPower | like an AGI or AAH or whayever. |
23:06.56 | x86 | nope |
23:07.11 | x86 | ManxPower: it worked fine until i moved my SIP users to mysql and started using RealTime |
23:07.15 | ManxPower | is your incoming from PSTN or VoIP? |
23:07.22 | ManxPower | x86, ah. can't help you then |
23:07.25 | x86 | ManxPower: voip origination |
23:07.39 | x86 | ManxPower: CDR logs it as NPANXXEXTN |
23:07.42 | ManxPower | there are a lot of stupid VoIP companies out there. |
23:07.52 | amdtech | you probably try using the windows tftp server to see if it's the server |
23:07.56 | ManxPower | that would be the correct format. |
23:08.09 | Strom_C | my conclusion: realtime is still too fraught with problems to seriously consider at this point in time |
23:08.15 | ManxPower | i.e. all digits, no spaces, no bunctuation |
23:08.17 | amdtech | for what? |
23:08.29 | amdtech | sip users? |
23:08.36 | Qwell | ManxPower: That damn bunctuation always gets in the way |
23:08.39 | Qwell | ;) |
23:08.39 | x86 | ManxPower: right... thats what's coming in... |
23:08.46 | Strom_C | help help, I've been bunctuated |
23:15.35 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
23:17.43 | x86 | is there a way to record all inbound calls destined for a given extension, from the dialplan? |
23:18.55 | Strom_C | monitor() |
23:19.52 | amdtech | what he said :) |
23:23.23 | *** join/#asterisk vr_mex (n=vr_mex@dsl-201-129-240-63.prod-infinitum.com.mx) |
23:23.58 | x86 | how would i format the file name? |
23:24.33 | Qwell | however you like |
23:24.56 | x86 | for example, how can i incorporate the date, and some unique call ID? |
23:25.47 | x86 | also, Monitor is better than MixMonitor? |
23:25.52 | *** part/#asterisk vr_mex (n=vr_mex@dsl-201-129-240-63.prod-infinitum.com.mx) |
23:26.10 | Qwell | MixMonitor will combine the files on the fly |
23:26.18 | Qwell | so in many cases, it's better |
23:26.32 | troyb1 | hey Qwell |
23:26.44 | x86 | well how do i put in the unique ID and date? |
23:26.47 | x86 | is that possible? |
23:26.57 | Qwell | x86: channel variables |
23:28.02 | *** join/#asterisk vr_mex (n=vr_mex@dsl-201-129-240-63.prod-infinitum.com.mx) |
23:28.34 | vr_mex | I need help on setting up shorewall i am behind a dmz router |
23:28.42 | Qwell | vr_mex: #shorewall |
23:29.03 | x86 | Qwell: right, which ones? :) |
23:29.09 | x86 | Qwell: is there a list somewhere? |
23:29.10 | vr_mex | but I am setting it up for asterisk@home |
23:29.25 | Qwell | vr_mex: #asteriskathome |
23:29.26 | troyb1 | is there any reason to use asterisk@home instead of asterisk? |
23:29.39 | vr_mex | thanks |
23:29.42 | Qwell | troyb1: because you're too lazy and/or stupid to configure an OS |
23:29.48 | *** part/#asterisk vr_mex (n=vr_mex@dsl-201-129-240-63.prod-infinitum.com.mx) |
23:29.49 | Qwell | That's just my experience anyhow |
23:29.59 | troyb1 | well im happy to let you know my installs are 'regular' asterisk :P |
23:37.07 | brookshire | O RLY? |
23:37.16 | troyb1 | @_@ rlly :O |
23:37.40 | brookshire | SRLY? |
23:37.47 | troyb1 | nope.. im lying. |
23:38.18 | brookshire | orlyowl.com :( |
23:38.23 | brookshire | i'm addicted |
23:38.27 | Heimidal | this is torturous |
23:47.46 | *** join/#asterisk jebmpls (n=brandon@rrcs-67-53-20-211.west.biz.rr.com) |
23:48.26 | Strom_C | man |
23:48.32 | Strom_C | the channel is so dead today |
23:50.28 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp9-adsl-129.ath.forthnet.gr) |
23:52.49 | brookshire | myspace is slow too |
23:52.52 | brookshire | :( |
23:53.06 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp9-adsl-129.ath.forthnet.gr) |
23:54.45 | tsume | brookshire: no shit sherlock ;) |
23:57.31 | Heimidal | ugh.. tftp timeout over and over again |
23:58.09 | tsume | ftp > tftp for voip work |
23:58.22 | Strom_C | tsume, the cisco phones require tftp |
23:58.35 | tsume | Strom_C: how sad. |
23:58.46 | Heimidal | I don't understand what is going wrong |
23:59.07 | tsume | you mean.. cisco doesn't make perfect products *gasp* :) |
23:59.48 | Strom_C | Heimidal, is there anything between the phones and the tftp server? |