| 00:00.13 | kusznir | hmm...I think I might have found it...my default context does not have any way to get to the local context. |
| 00:01.05 | akrall_ | anybody gotten unicall to compile under cvs-head? |
| 00:01.38 | jskcr | kusznir: did you do a init keys in the console? |
| 00:02.42 | *** join/#asterisk _cleric_ (n=dacleric@p5482B56A.dip0.t-ipconnect.de) |
| 00:02.54 | kusznir | Umm...I don't think so because i don't know what you're talking about :) |
| 00:03.36 | jskcr | paste you [iaxtel-outbound] from iax.conf without usernames and passwords on pastebin |
| 00:03.49 | Hmmhesays | there is a tornado north of here |
| 00:03.51 | tugalone | does asterisk have nat traversal support? |
| 00:04.47 | jskcr | kusznir: try setting it up as a peer in iax.conf ie http://www.voip-info.org/tiki-index.php?page=Asterisk+config+iax.conf |
| 00:05.26 | jskcr | then use exten => _1800XXXXXX,1,Dial(IAX2/iaxtel-outbound/${EXTEN}) |
| 00:05.40 | *** join/#asterisk spootnick (n=irc@CPE-147-10-168-100.nsw.bigpond.net.au) |
| 00:06.15 | tugalone | can't see it here - http://www.asterisk.org/features. i guess not. |
| 00:06.20 | spootnick | does Dial() still jumps to priority x+102 when busy in Asterisk 1.2* ? |
| 00:06.37 | kusznir | brb...got to go watch the stove for a minute... |
| 00:10.27 | tugalone | can v1.2.0-beta1be compiled on Mac OS X? |
| 00:10.59 | tugalone | the front page mentions some support for it, but has anyone tried it here? |
| 00:11.36 | hugo-v6 | wtf is zapatellar |
| 00:11.42 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 00:11.45 | hugo-v6 | ller |
| 00:12.46 | Agrajag- | gday. im trying to find out how to go about just using asterisk as a sort of auditing tool for my normal home phone. i can't find any info on what hardware would be required or anything. any pointers? |
| 00:12.47 | jskcr | kusznir: make sure your iax.conf has register => user:password@iaxtel.com also |
| 00:13.01 | kusznir | it does. |
| 00:13.24 | kusznir | come to think of it, though, I don't know if my registration is still valid. |
| 00:13.46 | jskcr | Agrajag-: voip-info.org |
| 00:13.59 | tugalone | "include/asterisk/poll-compat.h:92: error: conflicting types for 'poll' /usr/include/sys/poll.h:109: error: previous declaration of 'poll' was here" - doesn't. |
| 00:14.13 | kusznir | I have 2 register statements in there (one for iaxtel and one for asterlink (which I'm not using yet in the dialplan)), and only the asterlink has a "registered with host <ip> who sees me as <ip>". |
| 00:14.33 | jskcr | remove asterlink for now |
| 00:14.48 | Agrajag- | jskcr: yeah i've looked there, all i could find was http://www.voip-info.org/tiki-index.php?page=Analog+Telephone+Information which doesn't discuss what hardware is required? |
| 00:15.02 | kusznir | I did, and did a reload. |
| 00:15.08 | jskcr | Agrajag-: how fast is your computer your using right now |
| 00:15.27 | jskcr | kusznir: log into iaxtel and make sure your accounts valid |
| 00:15.29 | Agrajag- | jskcr: err 2.8ghz |
| 00:16.37 | jskcr | Agrajag-: well on celerons you can run quite a few calls like a t1's worth |
| 00:16.41 | kusznir | ok. |
| 00:17.14 | jskcr | also search you extensions.conf and make sure you dont have a rule overiding your 800 rule |
| 00:17.18 | jskcr | hya kram |
| 00:17.43 | Agrajag- | jskcr: im talking about what i need to be able to plug a single phone line/phone into my computer so i can audit it |
| 00:17.52 | jskcr | x100p |
| 00:18.15 | jskcr | single line zaptel device or a sipura fx0 woul work okay too |
| 00:18.31 | kusznir | Thanks for all the help. Unfortunately, "real-work" calls now...I'll be back at it tonnight. (btw: my iaxtel accout is still good) |
| 00:18.35 | Agrajag- | ok, thanks, i'll check them out |
| 00:18.36 | kram | or a TDM01B, i'd like to add |
| 00:18.53 | Agrajag- | whatever's cheapest :) |
| 00:19.42 | jskcr | TDM01B would probable cause you the least headache |
| 00:20.43 | Hmmhesays | FRONTIER AND REILE'S ACRES AT 735 PM OXBOW AT 740 PM MOORHEAD AT 745 PM IF YOU ARE IN THE PATH OF THIS TORNADO ABANDON CARS AND MOBILE HOMES FOR A STURDY BUILDING. |
| 00:20.45 | Agrajag- | hmm they seem to be a bit more expensive that i thought they might be |
| 00:21.15 | Hmmhesays | ABANDON CARS lol, yeah right... drive the fsck away |
| 00:21.37 | blitzrage | ...and people said this Ferrari was stupid - later suckah's! |
| 00:22.02 | Hmmhesays | "i've got room for one hot chick" |
| 00:22.06 | Hmmhesays | "maybe two" |
| 00:22.18 | file | I hate waiting for calls |
| 00:22.23 | blitzrage | 3 if you strap one to the roof |
| 00:22.38 | *** join/#asterisk veteran (n=veteran@cpe-66-25-30-132.houston.res.rr.com) |
| 00:22.40 | Hmmhesays | its nasty, I just took the bike out and went riding towards where it was supposed to start |
| 00:22.57 | blitzrage | Hmmhesays: isn't that the exact thing you're NOT supposed to do? :) |
| 00:23.13 | Hmmhesays | it was all nice and toasty warm, then suddenly I got blasted with cold air in about a 30mph side gust |
| 00:23.25 | Hmmhesays | blitzrage it hasn't started yet here |
| 00:23.32 | Hmmhesays | they were thinking it was going to form about 15 miles north of here |
| 00:23.49 | blitzrage | Hmmhesays: ahhh - luckily I live in a place where natural disasters don't happen |
| 00:23.52 | Hmmhesays | after that I turned around and hauled ass back home |
| 00:24.17 | Hmmhesays | like the cops are going to pull a guy over doing 80 in a 40 on a bike riding away from a storm |
| 00:24.26 | blitzrage | Hmmhesays: exactly :) |
| 00:24.42 | Hmmhesays | hey file |
| 00:24.49 | *** join/#asterisk psyoptix (i=psywar@rasterburn.org) |
| 00:25.15 | psyoptix | is this the right place for user questions, or is this chan for developers? |
| 00:25.42 | fugitivo | user questions |
| 00:25.46 | psyoptix | cool, ty |
| 00:26.42 | psyoptix | Basically I upgraded * and zap* and now when I try to enter voicemail (via a sip line), it doesn't recognize my password. I set the debug level high and verified that it isn't recognizing the digits reliably |
| 00:27.16 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 00:27.35 | psyoptix | anyone got an idea as to what coudl be wrong? I sending DTMF inband, because with the default (rfc2833?) I couldn't access remote DTMF-operated systems reliably |
| 00:27.46 | psyoptix | which is to say, at all. |
| 00:27.52 | akrall_ | anybody gotten unicall to compile under cvs-head? |
| 00:31.27 | psyoptix | oh yeah. just got this msg on console: app_voicemail.c:3389 vm_execmain: Unable to read password |
| 00:32.12 | psyoptix | the password is in the config file, which is owned by asterisk like everything else, why can't it read it? |
| 00:33.13 | Hmmhesays | does cmd set still work in stable? |
| 00:33.20 | Hmmhesays | or, not still work, but does it work at all |
| 00:33.21 | hugo-v6 | psyoptix: dunno why, but maybe strange chars? missing "" or '' or no \n at the end of the file |
| 00:34.41 | *** join/#asterisk IOscanner (n=IOscanne@c-67-166-249-43.hsd1.tx.comcast.net) |
| 00:34.55 | IOscanner | Anyone try the T.38 patch? |
| 00:35.09 | psyoptix | it was working before the upgrade... same format |
| 00:35.11 | *** join/#asterisk liberie_ (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net) |
| 00:35.21 | akrall_ | Anybody knows how to implement chanspy on 1.0.9? chanspy is on cvs-head but could it be ported to 1.0.9? |
| 00:35.25 | psyoptix | something different occured, I think with the voicemail upgrade |
| 00:35.59 | psyoptix | BTW, what does a leading - mean on the vm password? I've forgotten and none of the examples say |
| 00:37.25 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
| 00:38.36 | CaNaBiS_ | bkw_, know how to convert from mgcp to sccp on a 7940? |
| 00:40.04 | jskcr | IOscanner: Im gonna test out the t.38 patch this weekend |
| 00:41.03 | *** part/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
| 00:43.06 | Hmmhesays | can mailbox numbers have letters in them? |
| 00:43.51 | psyoptix | what use would that be? |
| 00:44.18 | Hmmhesays | cause the mailbox numbers I have are alphanumeric |
| 00:44.24 | Hmmhesays | er.. the sip users |
| 00:44.24 | psyoptix | for all the alphanumeric phones out there? |
| 00:44.27 | Hmmhesays | no |
| 00:44.54 | Hmmhesays | would be like cell phones where you don't have to put in your mailbox number |
| 00:45.28 | *** part/#asterisk akrall_ (n=akrall@201.144.58.186) |
| 00:46.41 | *** join/#asterisk liberie_ (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net) |
| 00:47.14 | psyoptix | I'm not sure I understand... putting in your mailbox number is only required under certain conditions, depending on the way you call VoicemailMain |
| 00:50.56 | spootnick | does Dial() still jumps to priority 102 when busy in Asterisk 1.2* ? |
| 00:51.17 | *** join/#asterisk brc_ (n=brc@ip70-176-64-134.ph.ph.cox.net) |
| 00:53.51 | Hmmhesays | according to the wiki <shrug> |
| 00:54.07 | Hmmhesays | you'd think it could just jump to the next priority |
| 00:54.38 | Hmmhesays | or it should just jump to the next priority, then you can just send it to whatever s- context you want |
| 00:54.39 | spootnick | Hmmhesays: probably. but there's also mention that 102 is the next priority for busy |
| 00:54.54 | Hmmhesays | its actually n+101 |
| 00:55.01 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 00:55.16 | hugo-v6 | damn shit. 3am and gf said she drives to me know (will arrive in about an hour) and i have to work at 8am |
| 00:55.30 | spootnick | s- doesn't suit me. I need to specify the extensions since i'm setting up fallthrough on busy for numbers on my isp |
| 00:55.33 | Hmmhesays | you know what to do right hugo-v6? |
| 00:55.44 | psyoptix | Hmmhesays: IF NOT DIAL GOTO 102 |
| 00:55.47 | hugo-v6 | Hmmhesays: well i guess ;) |
| 00:55.52 | Hmmhesays | s- could so you fine spootnick |
| 00:56.11 | psyoptix | what asterisk really needs is less GOTOs and more COME FROMs |
| 00:56.18 | Hmmhesays | s-BUSY,1,goto(<my next number>,1) |
| 00:56.21 | hugo-v6 | fsck through the night and go to work :> |
| 00:56.29 | Hmmhesays | hugo-v6 you got it man |
| 00:56.41 | spootnick | Hmmhesays: s-BUSY... yeah, i saw it there. thing is, if I don't specify my numbers on this context i'm dealing with (external incoming calls), i don't get a ring |
| 00:56.43 | Hmmhesays | gotta have your priorities |
| 00:56.45 | spootnick | or anything |
| 00:56.59 | Hmmhesays | answer the call first? |
| 00:57.27 | spootnick | you mean, answer using the number, then "s" from that point and on? |
| 00:57.33 | Hmmhesays | aye |
| 00:57.37 | spootnick | ummm |
| 00:57.42 | Hmmhesays | i got around some stupid ringing problems doing that |
| 00:58.06 | Hmmhesays | if you use the if you use c on cmd dial you won't fark up your cdrs either |
| 00:58.37 | *** join/#asterisk JunK-Y (n=junky@67.71.158.118) |
| 00:58.38 | spootnick | i'm trying it right now |
| 00:59.13 | Hmmhesays | answer and set some variables if you need to, send it onto a different-context,s,1 |
| 00:59.28 | Hmmhesays | exten => s,1,Dial(${user},30) |
| 00:59.28 | Hmmhesays | exten => s,2,Goto(s-${DIALSTATUS},1) |
| 00:59.28 | Hmmhesays | exten => s-NOANSWER,1,Voicemail(u${user}) |
| 00:59.28 | Hmmhesays | exten => s-CHANUNAVAIL,1,Voicemail(u${user}) |
| 00:59.52 | Hmmhesays | i defined the ${user} variable earlier |
| 01:00.05 | spootnick | weird... it hanged up after answer() |
| 01:00.18 | Hmmhesays | pastebin your dialplan |
| 01:00.19 | spootnick | i mean, i answered using the number, then 's' from that point on |
| 01:00.49 | Hmmhesays | i fix it for you if you send me a case of budweiser |
| 01:02.07 | Hmmhesays | hey, in dial if you do get a busy and n+101 doesn't it exist, it continues to the next priority in the current context doesn it? |
| 01:02.21 | tzanger | Hmmhesays: at least ask for real beer |
| 01:02.56 | spootnick | Hmmhesays: deal. pastebinned already |
| 01:03.15 | Hmmhesays | ahah link me |
| 01:03.25 | spootnick | http://pastebin.ca/22200 |
| 01:03.30 | Hmmhesays | tzanger: sorry that is my favorite middle class beer |
| 01:03.50 | *** join/#asterisk simprix (n=simprix@24-231-248-225.static.aldl.mi.charter.com) |
| 01:04.17 | simprix | Does the TE110P support PRI in the united states ? |
| 01:04.26 | spootnick | Hmmhesays: quick update, http://pastebin.ca/22201 |
| 01:04.54 | Hmmhesays | and you are getting no ringing? |
| 01:05.29 | Hmmhesays | or what exactly is the problem? |
| 01:05.30 | spootnick | nope, it's ringing |
| 01:05.37 | spootnick | now it just hangs up after answer |
| 01:06.00 | Hmmhesays | what does it say on your console? |
| 01:06.04 | spootnick | if i use the number, if answers fine. the only thing is i can't handle an "extension busy then jump to a different one" situation |
| 01:06.39 | Hmmhesays | i think if n+101 doesn't it exist it will continue in the on to the next priority |
| 01:06.48 | Hmmhesays | can someone say for sure? |
| 01:07.13 | spootnick | <PROTECTED> |
| 01:07.35 | spootnick | then executing hangup... |
| 01:08.01 | Hmmhesays | why are you using n? priority? |
| 01:08.13 | spootnick | n |
| 01:08.40 | spootnick | if you check the comments on the pastebin, that's the way it was working fine before |
| 01:09.05 | spootnick | i was using the full number until the point where an user hits a value for an extension |
| 01:09.47 | Hmmhesays | you need something to tell it to go to the s extension |
| 01:09.59 | Hmmhesays | exten => did,n,goto(s,1) |
| 01:10.47 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 01:10.47 | Hmmhesays | know what i'm saying? |
| 01:11.11 | spootnick | yeah, i'm giving it a shot now |
| 01:11.33 | Hmmhesays | i need to go find some booze |
| 01:11.34 | Hmmhesays | brb |
| 01:12.45 | gek | i just signed up with nufone and then i realized their web interface isn't even DONE YET |
| 01:12.53 | gek | i'm sick of giving all these companies a free $10 |
| 01:12.53 | *** join/#asterisk bsd3 (n=bsd@203.134.193.178) |
| 01:13.05 | gek | can someone tell me a termination provider that doesn't suck? i don't even care how muhc money it costs now |
| 01:13.14 | hugo-v6 | Hmmhesays: i got vodka in my freezer for such cases ;) |
| 01:13.51 | Hmmhesays | yeah i got some rumple mints |
| 01:13.57 | *** join/#asterisk kshumard_home (n=ksh@pcp08979908pcs.huntsv01.al.comcast.net) |
| 01:15.06 | psyoptix | okay I have new information - if I bypass the voicemail password using s in the call to VoiceMailMain, I can't do anything... DTMF codes unrecognized. Anyone got any clues? |
| 01:15.18 | psyoptix | This is a SIP connection, used to work fine |
| 01:15.38 | psyoptix | O_o |
| 01:15.43 | *** join/#asterisk simprix (n=simprix@24-231-248-225.static.aldl.mi.charter.com) |
| 01:15.54 | simprix | Do all the grandstream phones work with asterisk ? |
| 01:16.09 | spootnick | psyoptix: dtmfmode=rfc2833 ? |
| 01:16.20 | Hmmhesays | so file, can you answer my dial question? |
| 01:16.30 | Hmmhesays | since i'm just to lazy to look |
| 01:17.15 | psyoptix | spootnick: no, inband |
| 01:17.35 | psyoptix | last time I tried rfc2833, I coudn't access remote systesms that wanted DTMF |
| 01:17.52 | psyoptix | did something changed to break inband? |
| 01:17.52 | Hmmhesays | spootnick: did that work for you? |
| 01:18.19 | Hmmhesays | heh, you need to read up on dtmf transporting there buddy |
| 01:18.41 | spootnick | Hmmhesays: well, i now narrowed my dialplan down a little bit by using "s". but i still can't get the redirect on busy thing to work. http://pastebin.ca/22203 |
| 01:18.44 | *** join/#asterisk roulduke_ (i=ru62bk4y@p508D246B.dip0.t-ipconnect.de) |
| 01:19.09 | psyoptix | where do I read about DTMF transporting? |
| 01:19.10 | Hmmhesays | is asterisk actually returning a busy though? |
| 01:19.52 | psyoptix | I don't recall anything about DTMF transporting in the manual... |
| 01:19.57 | spootnick | Hmmhesays: yep |
| 01:20.00 | psyoptix | but I'm not a telco guy |
| 01:20.03 | Hmmhesays | http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode |
| 01:20.20 | hugo-v6 | how often have i told that the elmeg/snom phones rock |
| 01:20.20 | Hmmhesays | spootnick rock |
| 01:20.33 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 01:20.44 | spootnick | Hmmhesays: i'm dialing a busy extension. then, instead of redirecting to a second ext, i get a busy reply |
| 01:21.00 | Hmmhesays | look where you are using dial |
| 01:21.02 | hugo-v6 | i got the elmeg ip 290 which is the same as the snom ip 190. and its lovely. looks clean and works fine :) |
| 01:21.02 | *** join/#asterisk file (n=jcolp@mctnnbsa30w-156034035106.nb.aliant.net) |
| 01:21.02 | Nivex | Hmmhesays: and I thought *I* liked my tech. |
| 01:21.04 | MikeJ[Laptop] | Hmmhesays, this is a family channel! |
| 01:21.12 | Hmmhesays | MikeJ[Laptop], since when? |
| 01:21.27 | Nivex | I always thought this channel was more PG-13 |
| 01:21.36 | hugo-v6 | Hmmhesays: since a few got kids and pets :) |
| 01:22.12 | simprix | Do all the grandstream phones work with asterisk ? |
| 01:22.19 | Hmmhesays | so spootnick you just want me to tell you what to do? or should I bait you |
| 01:22.24 | Hmmhesays | and make you figure it out |
| 01:22.37 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 01:22.43 | hugo-v6 | mafia? |
| 01:22.48 | hugo-v6 | *fear* |
| 01:22.57 | newsmafia | Newsmafia! |
| 01:23.05 | spootnick | Hmmhesays: nah, bait me. i like to struggle |
| 01:23.29 | Hmmhesays | ok, where in that dialplan are you sending a busy back to wherever you are sending the call to |
| 01:23.30 | hugo-v6 | tell me the news from teh family |
| 01:23.36 | Hmmhesays | *wherever the call is coming from |
| 01:23.51 | spootnick | exten => 1,102,Busy() |
| 01:24.06 | spootnick | but that doens't even get executed, as far as i can tell from the console |
| 01:24.24 | gek | i need a termination provider with did's that doesn't suck!!! |
| 01:24.38 | psyoptix | Hmmhesays: that link doesn't say anything about DTMF "transporting", although is does suggest * may have changed default codecs and thus broke my DTMF |
| 01:24.50 | Hmmhesays | where is the call before you want it to be at exten => 1,102,BUSY |
| 01:24.55 | h3x | hugo-v6: is the elmeg firmware the same |
| 01:24.59 | MikeJ[Laptop] | gek, they all suck... they are the phone company ;) |
| 01:25.05 | *** join/#asterisk Gronker__ (n=Gronker2@70.152.166.108) |
| 01:25.15 | *** part/#asterisk Gronker__ (n=Gronker2@70.152.166.108) |
| 01:25.36 | spootnick | Hmmhesays: at exten => 1,n,Dial(${EXTENSION2},25,rmtT) (actually, noop, and that's getting executed, but for debugging only) |
| 01:26.05 | *** join/#asterisk liberie_ (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net) |
| 01:26.07 | spootnick | k, so next would be exten=> 1,n,GoTo(...) ? |
| 01:26.27 | hugo-v6 | h3x: yep. its only labeld elmeg. the elmeg guys make the case and the snom ppl make the stuff that makes it work |
| 01:26.43 | Hmmhesays | well... what do you think would make the call jump from priority 3 to 102? |
| 01:26.45 | simprix | is 512 ddr ram good for a pri configuration with 20 extensions ? |
| 01:26.47 | h3x | are they that much cheaper? |
| 01:26.55 | spootnick | Hmmhesays: a busy dialstatus |
| 01:27.23 | spootnick | Hmmhesays: i even tried to use that GotoIf to check ${DIALSTATUS}, but it didn't work as well |
| 01:27.27 | Hmmhesays | spootnick, not quite |
| 01:27.28 | hugo-v6 | h3x: for me and in .de yes. i get a elmeg ip 290 for about 100euros and the same snom for about 125euros |
| 01:27.35 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 01:27.35 | psyoptix | okay, rfc2833 fixed my local DTMF xmission |
| 01:27.39 | Hmmhesays | dialstatus is the status when asterisk is dialing the call |
| 01:28.23 | psyoptix | I still think the config files need a comefrom command, just for symmetry |
| 01:28.27 | h3x | they dont have the equivalent of the snom 360 huh |
| 01:28.29 | Hmmhesays | in that case anyway, you have your busy at the wrong priority |
| 01:28.41 | hugo-v6 | h3x: dunno. give me a second |
| 01:28.58 | gek | MikeJ[Laptop]: but there must be one that actually works. sixtel can't even activate my number or turn on international calling and nufone local did thing says "not implemented" on the website. just need another one to waste $10 on. |
| 01:29.21 | Hmmhesays | dial jumps to n+101 on busy |
| 01:29.22 | spootnick | Hmmhesays: ok, it's on set(CALLERID(name) = ...) |
| 01:29.45 | Hmmhesays | your dial line is already on priority 2 |
| 01:29.47 | Hmmhesays | you do the math |
| 01:29.59 | spootnick | n+102 |
| 01:30.02 | spootnick | ? |
| 01:30.03 | *** join/#asterisk CaNaBiS (i=canabis@ip68-111-51-215.br.no.cox.net) |
| 01:30.15 | Hmmhesays | no.... next priority plus 101? |
| 01:30.37 | spootnick | n+101 =) |
| 01:30.47 | hugo-v6 | h3x: no they dont |
| 01:30.49 | hugo-v6 | :( |
| 01:31.01 | hugo-v6 | only the ip 290 which is a snom ip 190 |
| 01:31.17 | Hmmhesays | well current priority +101, n increments the priority by one |
| 01:31.41 | hugo-v6 | Hmmhesays: isnt it ie 1 + 101 not 2 + 101 if 101 is bussy or on phne? |
| 01:31.42 | spootnick | Hmmhesays: so what's the value for current priority? |
| 01:31.46 | l1nux | whoooooooo :D |
| 01:31.54 | l1nux | work fine |
| 01:31.56 | hugo-v6 | s/if 101/if 1/ |
| 01:31.58 | Vco | gek |
| 01:32.05 | Hmmhesays | http://www.voip-info.org/tiki-index.php?page=Asterisk+priorities |
| 01:32.07 | Vco | cdn/us or international? |
| 01:32.08 | *** join/#asterisk CaNaBiS_ (i=canabis@ip68-111-51-215.br.no.cox.net) |
| 01:32.12 | Hmmhesays | there you go spootnick I have baited you |
| 01:32.17 | l1nux | night all |
| 01:32.22 | *** part/#asterisk l1nux (n=l1nux@lns-vlq-5-tou-82-64-211-239.adsl.proxad.net) |
| 01:34.31 | spootnick | Hmmhesays: does autofallthrough has to be set to no? |
| 01:34.39 | Hmmhesays | negatory |
| 01:35.06 | spootnick | so that's autofallthrough=yes ? |
| 01:35.08 | spootnick | =) |
| 01:35.09 | Hmmhesays | you need to create a priority 103 it looks like |
| 01:35.40 | spootnick | create priority 103? that's news for me |
| 01:35.49 | Hmmhesays | just try it |
| 01:36.00 | Hmmhesays | tell me if it works, its been awhile since i've had to do this |
| 01:37.42 | psyoptix | 03:31 -!- l1nux [n=l1nux@lns-vlq-5-tou-82-64-211-239.adsl.proxad.net] has left #asterisk ["Konversation terminated!"] |
| 01:37.53 | *** join/#asterisk CaNaBiS (i=canabis@ip68-111-51-215.br.no.cox.net) |
| 01:37.54 | psyoptix | haha... he's the Kompressor or something |
| 01:38.23 | *** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
| 01:38.28 | Vco | fucking kde apps |
| 01:38.35 | Vco | thats so gay |
| 01:38.40 | spootnick | Hmmhesays: ok, thing is, it never reaches exten => 1,n+102,Goto(incoming-bigair,2,1) or whatever i put after that. i get the busy reply on the console, then when it timeouts, the "t" extension catches it |
| 01:39.10 | spootnick | spootnick: so apparently it's holding on the line when busy, instead of doind something else right away |
| 01:39.12 | *** join/#asterisk Moc_ (n=mochouin@207-199-0-72-ppp.3menatwork.com) |
| 01:39.20 | spootnick | (i'm talking to myself already) |
| 01:39.51 | Hmmhesays | do I have an old version of your dialplan or what? |
| 01:40.51 | file | okay I'm back |
| 01:41.04 | spootnick | Hmmhesays: here's the full latest one. http://pastebin.ca/22204 |
| 01:42.03 | Hmmhesays | exten => 1,n,Dial(${EXTENSION2},25,rmtT) |
| 01:42.08 | Hmmhesays | <PROTECTED> |
| 01:42.11 | spootnick | yep |
| 01:42.27 | flewid | <PROTECTED> |
| 01:42.31 | Hmmhesays | does dial still return n+101 on busy? |
| 01:42.43 | flewid | i would have to make the agents login to local + remote queues on all of them right? |
| 01:42.55 | flewid | and everything duplicated on each server basically |
| 01:43.05 | spootnick | Hmmhesays: i can see it says busy on the console. but it holds there instead of moving on to n+101 or whatever |
| 01:43.19 | spootnick | spootnick: i ain't sure about n+101. it doesn't seem to be executed at all |
| 01:44.02 | Hmmhesays | is ${EXTENSION2} a valid user? |
| 01:44.04 | flewid | http://www.nastybits.ca/ourmap.jpg is what i mean of remote + main server |
| 01:44.28 | spootnick | Hmmhesays: yes it is. i can dial it fine |
| 01:44.47 | spootnick | Hmmhesays: internally or coming from outside |
| 01:44.51 | spootnick | all works |
| 01:45.04 | *** join/#asterisk Snake-Eyes (n=blog@203.201.98.84) |
| 01:45.24 | spootnick | exten => 1,n+101,Goto(incoming-bigair,2,1) <==== never gets executed |
| 01:45.40 | *** part/#asterisk bsd3 (n=bsd@203.134.193.178) |
| 01:46.17 | Hmmhesays | your goto is not valid |
| 01:46.38 | Hmmhesays | you are telling the call to goto incoming-bigair extension 2 priority 1 |
| 01:46.41 | spootnick | how come? it's the same goto i'm using to handle the "i" and "t" extensions |
| 01:46.46 | Hmmhesays | try extension s |
| 01:46.49 | Hmmhesays | ;) |
| 01:46.54 | *** part/#asterisk veteran (n=veteran@cpe-66-25-30-132.houston.res.rr.com) |
| 01:47.16 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
| 01:49.34 | spootnick | <PROTECTED> |
| 01:49.35 | spootnick | <PROTECTED> |
| 01:49.40 | spootnick | Hmmhesays: i tried. same thing |
| 01:49.50 | *** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
| 01:50.11 | spootnick | Hmmhesays: you meant exten => 1,n+101,Goto(incoming-bigair,s,1), right? didn't work |
| 01:50.38 | spootnick | it doesn't surprise me actually. what i need is get the user diverted to number 2, priority 1 |
| 01:50.39 | Hmmhesays | exten => 1,103,Goto(incoming-bigair,2,1) |
| 01:50.45 | clyrrad | can anyone point me to a document that shows how to connect a software sip phone to an * box on a different network? |
| 01:50.49 | *** join/#asterisk hat (n=hat@bb220-255-134-33.singnet.com.sg) |
| 01:51.03 | Hmmhesays | in the same context? |
| 01:51.13 | Hmmhesays | goto(2,1) |
| 01:51.22 | Hmmhesays | right now you are sending it to an invalid extension |
| 01:51.58 | hat | hello, where is the output of NoOp dialplan application call? |
| 01:52.21 | clyrrad | on the CLI |
| 01:52.50 | hat | thanks. let me check |
| 01:53.12 | spootnick | Hmmhesays: ok, changed to exten => 1,103,Goto(2,1). but still, it doesn't get executed. it says "SIP/1235-df71 is busy", no busy tone to the calling party |
| 01:53.32 | Hmmhesays | does it auto fallthrough then? |
| 01:53.41 | spootnick | it seems it's on hold you know... then the timeout arrives, and it send to the "t" extension, which means "operator" in my dialplan |
| 01:53.42 | tzafrir_laptop | hi, where can I get the current asterisk tarballs from? both mirrors linked from http://www.asterisk.org/download give an empty dir on http and don't answer ftp |
| 01:54.05 | Hmmhesays | ok, someone else, if you have dial in priority 2, it will exit at 103 on busy right? |
| 01:54.36 | tzafrir_laptop | Alternatively: where can I get the zaptel 1.2 beta tarbal from? Can anybody upload it for me somewhere, please? |
| 01:55.16 | tzafrir_laptop | or mail tzafrir@cohens.org.il |
| 01:55.46 | spootnick | Hmmhesays: weird. i was even expecting i did something wrong on Dial(), kind of saying it should hold until the other side picks up |
| 01:56.49 | tzafrir_laptop | tzafrir_laptop, email email already appears in tons of places |
| 01:57.06 | *** join/#asterisk Beccara (n=Beccara@222-152-27-117.jetstream.xtra.co.nz) |
| 01:57.20 | spootnick | Hmmhesays: dunno if it's worth mentioning, but this is asterisk 1.2 beta |
| 01:58.00 | clyrrad | can anyone point me to a document that shows how to connect a software sip phone to an * box on a different network? |
| 01:58.23 | Vco | oh i'm sure a few of them will have attatchments.... |
| 01:58.25 | Vco | ;) |
| 01:58.26 | Hmmhesays | shrug, haven't played with beta, |
| 01:58.28 | Hmmhesays | much |
| 01:58.42 | Hmmhesays | but change that goto to goto(2,1) |
| 01:58.58 | spootnick | Hmmhesays: be amazed. take a look |
| 01:59.00 | spootnick | exten => 1,n,Dial(${EXTENSION2},25,rmtT) |
| 01:59.00 | spootnick | exten => 1,n,Goto(2,1) |
| 01:59.04 | spootnick | that works |
| 01:59.18 | Hmmhesays | cool, they must have removed that n+101 crap |
| 01:59.47 | Hmmhesays | being you can differentiate dialstatus now |
| 01:59.56 | *** join/#asterisk zotz_ (n=zotz@24.231.36.100) |
| 01:59.56 | spootnick | but it just doesn't sound right... i don't really feel safe when there's no sort of "query" to check if it's really busy |
| 01:59.57 | tzafrir_laptop | Hmmhesays, you never goto x+101 if that priority doesn't exist |
| 02:00.20 | Hmmhesays | tzafrir_laptop I know |
| 02:00.32 | Corydon76-home | You can get the old +101 behavior if you add the option j to the Dial |
| 02:00.55 | spootnick | oh can i ? |
| 02:01.02 | tzafrir_laptop | I use a Goto(S-${DIALSTATUS) anyway even now |
| 02:01.43 | Corydon76-home | You mean Goto(S-${DIALSTATUS},1) |
| 02:01.44 | spootnick | Hmmhesays: Corydon76-home is right. if you put j on dial, you can use 103. tested and it worked |
| 02:01.56 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
| 02:02.11 | tzafrir_laptop | Corydon76-home, right. |
| 02:02.13 | spootnick | i just don't wanna stick with the "old" syntax |
| 02:02.15 | Corydon76-home | spootnick: If you read UPGRADE.txt, you can find out all sorts of things |
| 02:03.00 | spootnick | Corydon76-home: good. tks a lot. it always gets down to RTFM, as i should have imagined |
| 02:03.41 | spootnick | i never quite understood how to use s-busy, s-noanswre |
| 02:03.43 | tzafrir_laptop | Anyway, can anybody help me with my upgrade effort by sending me a zaptel tarball? |
| 02:03.51 | Corydon76-home | It's not the manual, it's a listing of behavior changes in 1.2 |
| 02:04.07 | spootnick | Corydon76-home: yeah, i meant the concept of RTFM |
| 02:04.48 | tzafrir_laptop | spootnick, you put whatever you want there. Possibly a Goto to another place. |
| 02:04.55 | *** join/#asterisk _maydayjay_ (n=maydayja@ip101109.101.nas.net) |
| 02:05.19 | Vco | good lord, there are so many things wrong with this page i don't know where to start... http://www.fordvoice.org/ |
| 02:05.23 | Vco | template or not.. |
| 02:05.56 | spootnick | tzafrir_laptop: ok, but i mean, if I do need to say e.g.: "dial 3 for sales", then i think i need to use sth like exten => 3,1,... |
| 02:06.18 | spootnick | i just never figured out how s-busy fits in this scenario |
| 02:06.33 | spootnick | seems 3-busy would be reasonable, but i think that doesn't exist |
| 02:07.56 | tzafrir_laptop | spootnick, but then you wouldn't have used Dial in the first place |
| 02:08.35 | spootnick | well, ok. i'll be going. Hmmhesays, i'll send that crap czech beer you asked for. Corydon76-home, tks |
| 02:08.42 | spootnick | =) |
| 02:08.58 | Hmmhesays | lol |
| 02:09.09 | *** join/#asterisk asteriskph (n=yakumo@203.87.204.126) |
| 02:09.17 | tzafrir_laptop | Or rather, in 3,1 you put Goto(phones-context,sales3,1) where 'sales3' may be replaced by the actual number. |
| 02:09.31 | *** join/#asterisk santiago (n=santiago@63.245.86.254) |
| 02:09.34 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 02:12.21 | clyrrad | how do you connect a remote sip phone to your * box? Do you need any other port forwards orther than 5060 as definined in sip.conf? |
| 02:12.40 | clyrrad | keep getting connection timeout :/ |
| 02:13.41 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-13-64.cybersurf.com) |
| 02:14.06 | tzafrir_laptop | ftp2.digium.com now seems to have some contents. |
| 02:14.30 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
| 02:15.09 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 02:18.09 | *** join/#asterisk largezhang (n=largezha@nusnet-229-213.dynip.nus.edu.sg) |
| 02:19.30 | *** join/#asterisk tengulre (n=tengulre@61.185.238.166) |
| 02:20.16 | *** part/#asterisk largezhang (n=largezha@nusnet-229-213.dynip.nus.edu.sg) |
| 02:24.48 | *** join/#asterisk litage (n=nick@203.201.97.143) |
| 02:24.49 | *** join/#asterisk Snake-Eyes (n=blog@203.201.97.143) |
| 02:25.29 | *** join/#asterisk roulduke_ (i=2cpjtovy@p508D246B.dip0.t-ipconnect.de) |
| 02:25.35 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 02:26.11 | Vco | well... |
| 02:26.21 | Vco | i guess that wasn't the power cord i thought it was.... |
| 02:28.56 | *** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer) |
| 02:33.09 | Corydon76-home | Odd. With the functionality provided by bug 5055, I would have thought somebody else would have tried it by now. |
| 02:34.29 | MikeJ[Laptop] | Corydon, heh.. yeah right |
| 02:34.50 | file | Corydon76-home: hahahahahahahahahahahaha |
| 02:37.45 | Corydon76-home | This is indeed a disturbing universe. |
| 02:37.58 | file | because you're here? |
| 02:38.20 | Corydon76-home | ~google "this is indeed a disturbing universe" |
| 02:38.54 | MikeJ[Laptop] | ummmmmm |
| 02:38.56 | MikeJ[Laptop] | NO! |
| 02:39.07 | MikeJ[Laptop] | I'd rather go test that bug... |
| 02:39.13 | MikeJ[Laptop] | just kidding!!! |
| 02:39.29 | Corydon76-home | Hmm, and jbot is dead... |
| 02:39.34 | MikeJ[Laptop] | what's new |
| 02:39.41 | MikeJ[Laptop] | file killed him |
| 02:39.49 | *** join/#asterisk valence (n=valence@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 02:39.50 | MikeJ[Laptop] | with poisoned toast |
| 02:40.21 | file | yay death |
| 02:40.21 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfndm.dialup.mindspring.com) |
| 02:40.23 | MikeJ[Laptop] | file, I hear that stuff is asidic |
| 02:40.28 | file | very |
| 02:40.32 | Corydon76-home | acidic |
| 02:40.34 | MikeJ[Laptop] | wash, quick |
| 02:40.47 | MikeJ[Laptop] | Corydon, it's late.. bugger off :P |
| 02:41.02 | Corydon76-home | MikeJ[Laptop]: acetylsalicylic acid? |
| 02:41.22 | Corydon76-home | Well, okay |
| 02:41.29 | MikeJ[Laptop] | lysergic acid diethylamide? |
| 02:41.37 | MikeJ[Laptop] | Corydon, you wish! |
| 02:41.47 | MikeJ[Laptop] | back boy, back!! |
| 02:41.57 | Corydon76-home | I don't know if I wish or not. I've never seen you. |
| 02:42.03 | Corydon76-home | Have a pic? |
| 02:42.08 | MikeJ[Laptop] | yes I do. |
| 02:42.37 | Corydon76-home | I have, however, seen file. |
| 02:42.43 | MikeJ[Laptop] | me too.. |
| 02:42.45 | MikeJ[Laptop] | ummm |
| 02:42.59 | file | never seen me in real life! |
| 02:43.01 | *** join/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg) |
| 02:43.01 | MikeJ[Laptop] | Corydon, he is but a young boy... |
| 02:43.05 | MikeJ[Laptop] | I have! |
| 02:43.08 | *** part/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg) |
| 02:43.10 | Corydon76-home | MikeJ[Laptop]: he's legal, though |
| 02:43.11 | file | yes, MikeJ has |
| 02:43.23 | MikeJ[Laptop] | but he was grumpy the whole time |
| 02:43.28 | Corydon76-home | 18 is old enough! |
| 02:43.32 | file | yup grumpy |
| 02:43.37 | file | Corydon76-home: I'm 19 in October :P |
| 02:43.44 | MikeJ[Laptop] | somthing about getting stuck on a plane, and another plane blowing up or somthing like that |
| 02:43.47 | Corydon76-home | Where's Sleepy and Dopey? |
| 02:43.47 | PakiPenguin | http://www.sounerd.com.br/index.php?option=com_content&task=view&id=237&Itemid=43 |
| 02:43.55 | PakiPenguin | check this out :) |
| 02:44.17 | MikeJ[Laptop] | PakiPenguin, NO! |
| 02:44.24 | Corydon76-home | It's been Freenoded |
| 02:44.27 | PakiPenguin | MikeJ[Laptop], okay :) |
| 02:44.30 | MikeJ[Laptop] | stop telling people what to do? |
| 02:44.30 | PakiPenguin | haha |
| 02:44.37 | file[laptop] | QaF is on! |
| 02:44.53 | MikeJ[Laptop] | do you know hom many times I had to tell my kid today to stop telling people what to do... |
| 02:44.55 | MikeJ[Laptop] | it's rude |
| 02:45.44 | MikeJ[Laptop] | see, watch this... |
| 02:45.48 | MikeJ[Laptop] | DO MY LAUNDRY! |
| 02:45.51 | MikeJ[Laptop] | see... |
| 02:45.52 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
| 02:45.53 | MikeJ[Laptop] | rude |
| 02:45.53 | file[laptop] | okay! |
| 02:45.57 | file[laptop] | buy me a plane ticket. |
| 02:46.05 | file[laptop] | and then I can do it |
| 02:46.06 | MikeJ[Laptop] | to where? |
| 02:46.06 | Corydon76-home | PakiPenguin: I disagree. Spaces having semantic expression is EEEEEEEEvil |
| 02:46.23 | file[laptop] | to wherever laundry needs doing |
| 02:46.24 | PakiPenguin | :) hahah |
| 02:46.30 | Agrajag- | if i have a tdm400p card with an fxo and fxs, with asterisk, would i be able to run a custom external command, say, when someone picks up the the receiver on their analogue phone? |
| 02:46.30 | Corydon76-home | You should have chosen something better, like Ada |
| 02:46.51 | MikeJ[Laptop] | Agrajag-, yes |
| 02:46.58 | Agrajag- | awesome, ta |
| 02:47.10 | Corydon76-home | Yes, as long as you have the channel in immediate=yes mode |
| 02:47.29 | MikeJ[Laptop] | Corydon, he didn't ask how...;) |
| 02:47.45 | Corydon76-home | MikeJ[Laptop]: he might be surprised otherwise... ;-) |
| 02:47.50 | MikeJ[Laptop] | Corydon, are you coming up to von? |
| 02:47.59 | Agrajag- | we have an office speaker system which plays music through one of those funky apple wireless devices, it'd be cool to mute it when someone picks up the phone :P |
| 02:48.00 | Corydon76-home | Nope |
| 02:48.41 | MikeJ[Laptop] | Agrajag-, there is already some stuff that lets you crank down the muzak when you are on the phone. |
| 02:48.45 | Corydon76-home | If you have an Asterisk convention in Huntsville, Atlanta, Nashville, Bowling Green, Knoxville, Memphis, etc., I'll go, though |
| 02:48.46 | MikeJ[Laptop] | muted.c ?? |
| 02:49.05 | drumkilla | we've talked a bunch about having a true dev meeting in Huntsville |
| 02:49.12 | MikeJ[Laptop] | Corydon, it's too soggy down there right now |
| 02:49.13 | drumkilla | where we get a bunch of people together and write code :) |
| 02:49.24 | Corydon76-home | drumkilla: a true dev meeting? |
| 02:49.34 | drumkilla | yeah, rent a big room and code |
| 02:49.44 | Agrajag- | hmm but this is rather specialised, the box is streaming it to an itunes thing running in a wireless router. it wouldn't be hard to do as long as asterisk can run an external command |
| 02:49.44 | Corydon76-home | As opposed to that con in chicago masquerading as a developer's con? |
| 02:49.45 | drumkilla | with whiteboards |
| 02:49.46 | MikeJ[Laptop] | drumkilla. we can't even get anyone to show up to a dev call... why bother trying |
| 02:49.57 | drumkilla | Corydon76-home: I never said anything about that :-p |
| 02:50.09 | Corydon76-home | drumkilla: and beer? :-) |
| 02:50.15 | drumkilla | Corydon76-home: heck yeah! |
| 02:50.21 | drumkilla | MikeJ[Laptop]: don't crush my dreams :) |
| 02:50.39 | Corydon76-home | That's how OpenBSD does a con... lots of coders and lots of beer... |
| 02:50.44 | drumkilla | I have a stupid lab that starts at exactly the same time as the call |
| 02:50.49 | MikeJ[Laptop] | Corydon, and all the woman you want... |
| 02:50.50 | drumkilla | that sounds pretty awesome |
| 02:51.05 | Corydon76-home | MikeJ[Laptop]: woman, singular |
| 02:51.06 | MikeJ[Laptop] | :D |
| 02:51.07 | drumkilla | MikeJ[Laptop]: remind me to call mark about von ... I'm going to see if I can come up for a couple days |
| 02:51.11 | MikeJ[Laptop] | no.. |
| 02:51.16 | drumkilla | haha |
| 02:51.16 | MikeJ[Laptop] | I can't type tonight |
| 02:51.32 | MikeJ[Laptop] | drumkilla, I still have not gotten my tickets |
| 02:51.36 | drumkilla | I figured I could put the burdon on you to make me remember |
| 02:51.38 | drumkilla | oh well :) |
| 02:51.52 | Corydon76-home | MikeJ[Laptop]: you coming to Phreaknic? |
| 02:51.53 | MikeJ[Laptop] | had to get kevin to hound donna to even return my call about when people were going |
| 02:51.53 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 02:52.05 | MikeJ[Laptop] | Corydon, Probably no |
| 02:52.09 | Corydon76-home | It's so cheap, you won't believe that we're actually holding a con... |
| 02:52.21 | MikeJ[Laptop] | I am using up most of my go away points with my wife already |
| 02:52.32 | Corydon76-home | Generally, most of the Digium crew makes it to Phreaknic |
| 02:52.34 | file[laptop] | drumkilla: you can't catch me!!! |
| 02:53.13 | drumkilla | g'night folks |
| 02:53.26 | file[laptop] | night night |
| 02:53.56 | *** join/#asterisk hellagony (n=egutierr@200.121.208.202) |
| 02:54.07 | file[laptop] | I wonder what Holly is doing in Asia... |
| 02:54.14 | clyrrad | how do you connect a remote sip phone to your * box? Do you need any other port forwards orther than 5060 as definined in sip.conf? |
| 02:55.36 | JerJer | if you do it right, you don't port forward |
| 02:55.51 | t3t | hi JerJer |
| 02:55.56 | JerJer | moo |
| 02:56.01 | t3t | unless your * box is behind the nat :) |
| 02:56.23 | clyrrad | Yes the box is behind the nat.... we just can not get the phone to connect to the Asterisk box remotely not sure why |
| 02:57.03 | clyrrad | We have a port forward 5060 to the * box, but can not seem to establish a connection... Is there anything extra we need to add in sip.conf to allow remote soft phones to connect? |
| 02:57.21 | t3t | clyrrad, it's easiest to just put the * box in front of the nat |
| 02:58.22 | clyrrad | we have even disabled the firewall to debug with the same results.... Is there a different configuration needed to connect a remote phone as opposed to an internal phone? |
| 02:58.37 | JerJer | perhaps a 0.0.0.0 in bindaddr |
| 02:58.44 | clyrrad | already done that |
| 02:58.45 | fugitivo | clyrrad: iax is easier for that |
| 02:58.51 | file[laptop] | you need localnet and externip set if behind NAT, and if it has a LAN IP... |
| 02:58.55 | file[laptop] | so the SIP messages contain the right IP |
| 02:59.00 | clyrrad | iax to connect external soft phones? |
| 02:59.04 | fugitivo | yes |
| 02:59.27 | fugitivo | i said easier, not best option |
| 03:00.42 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 03:00.55 | clyrrad | We will be using X-Lite for the softphones, they are supposed to connect to * on a non internal network, we can get them working internally, just not externally over another network |
| 03:01.04 | clyrrad | using SIP |
| 03:01.33 | t3t | clyrrad, can you see the packets from the outside hitting the * box? |
| 03:01.36 | fugitivo | i know, did you set what file said? |
| 03:01.57 | spackle | clyrrad: do you have NAT on both ends or just your Asterisk server end? |
| 03:01.58 | clyrrad | yes, we have those set already |
| 03:02.03 | clyrrad | both ends |
| 03:02.08 | fugitivo | nat=yes ? |
| 03:02.13 | clyrrad | yes thats done |
| 03:02.20 | clyrrad | on server side |
| 03:02.40 | fugitivo | qualify=yes ? |
| 03:02.50 | t3t | clyrrad, can you see the packets from the outside hitting the * box?? |
| 03:02.52 | clyrrad | thats we dont have... |
| 03:03.25 | fugitivo | does the softphone register to the * ? |
| 03:03.48 | clyrrad | no, thats what we cant get them to do, they time out |
| 03:04.00 | fugitivo | then it's another problem |
| 03:04.02 | clyrrad | we can get them to connect internally just not externally |
| 03:04.07 | fugitivo | check if you see the packets with tcpdump |
| 03:04.13 | clyrrad | checking |
| 03:05.19 | fugitivo | clyrrad: did you open 5060-5063 ? |
| 03:05.20 | *** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
| 03:05.20 | *** mode/#asterisk [+o bkw__] by ChanServ |
| 03:05.47 | fugitivo | clyrrad: you'll need 10000-20000 also |
| 03:05.52 | clyrrad | we did 5060 |
| 03:06.11 | fugitivo | do you see the packets? |
| 03:06.22 | clyrrad | yes we saw them |
| 03:06.53 | fugitivo | you get nothing on the cli? |
| 03:06.59 | fugitivo | sip debug |
| 03:07.04 | clyrrad | thast whats strange |
| 03:07.06 | clyrrad | we did a sip debug |
| 03:07.16 | spackle | what about a log on the firewall? |
| 03:07.18 | clyrrad | and see nothing hitting the CLI |
| 03:07.25 | fugitivo | the packets are reaching the * box? |
| 03:08.05 | clyrrad | thats what we are trying to determine, we are opening those other ports you mentioned |
| 03:10.15 | clyrrad | we finished the TCP forwards 5060-5063, and 10000 to 20000 are all being forwared to the * box, and we still get timeout |
| 03:10.24 | clyrrad | do we need UDP as well? |
| 03:10.32 | fugitivo | you NEED udp |
| 03:10.44 | clyrrad | on the same ports? |
| 03:10.47 | fugitivo | yes |
| 03:10.54 | fugitivo | 10000-20000 is only udp |
| 03:11.24 | clyrrad | ok we are removing TCP on those ports changing to UDP... is that why we cant see anyting on the CLI? |
| 03:11.30 | *** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net) |
| 03:11.39 | fugitivo | you had only tcp 5060 opened? |
| 03:11.49 | clyrrad | yes |
| 03:13.26 | Vco | ...uncomfortable silence...... |
| 03:16.16 | fugitivo | so... |
| 03:17.18 | spackle | buttons on your underwear |
| 03:17.32 | fugitivo | how do you know?? |
| 03:17.51 | spackle | sew, buttons on your underwear - get it? sorry. |
| 03:18.53 | spackle | the antici...... |
| 03:19.00 | spackle | .....pation is killing me. |
| 03:19.22 | Vco | indeed |
| 03:19.45 | *** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
| 03:20.35 | clyrrad | fugitivo... we have confirmed that 5060-5063 TCP are forwared to *, we have also confirmed 10000-20000 UDP are also forwared to * and we still get a timeout on the soft phones, and see nothing hitting the CLI |
| 03:20.41 | Agrajag- | if i just have a single fxo in my asterisk box, can i make calls from other networked computers just using their soundcard devices? |
| 03:21.00 | fugitivo | clyrrad: what about the packets? |
| 03:21.04 | file | you need 5060 UDP forwarded...d |
| 03:21.26 | fugitivo | clyrrad: udp! udp! |
| 03:21.48 | Qwell | yeah, make sure you do udp. :D |
| 03:21.50 | clyrrad | file... we are making that change |
| 03:21.54 | Qwell | Don't pull a Qwell |
| 03:21.58 | spackle | arghh, the wait for LOTR:ROTK was shorter. |
| 03:22.06 | spackle | ;-) |
| 03:22.16 | clyrrad | fugitivo... .sip debug shows no packets, and tcpdump shows packets from what we belive is the softphone trying to connect |
| 03:22.23 | Qwell | Agrajag-: sure |
| 03:22.36 | fugitivo | clyrrad: first you need 5060 udp forwarded |
| 03:22.47 | clyrrad | doing that now |
| 03:23.13 | Agrajag- | Qwell: what software do the client boxes need? |
| 03:23.25 | Qwell | Agrajag-: a softphone |
| 03:23.52 | clyrrad | 5060 is now UDP, still getting timeout and nothing hitting the CLI |
| 03:24.18 | fugitivo | clyrrad: do you see the packets? |
| 03:24.35 | bkw__ | set verbose 100000000000000000000 |
| 03:24.48 | Agrajag- | Qwell: ta |
| 03:25.13 | clyrrad | which packets should i look for? If you mean on sip debug we see nothing, if you mean on TCPDUMP we see what we belive is the remote soft phone tryign to connect, the IP address is that of the client machine running the remote softphone, is this the packets you are refering to? |
| 03:26.13 | fugitivo | clyrrad: tcpdump, just to see if forwarding is working |
| 03:26.38 | fugitivo | clyrrad: in whick port and protocol do you see the packets? |
| 03:28.00 | *** join/#asterisk KaBewM (n=DA-MAN@24-180-28-208.dhcp.psdn.ca.charter.com) |
| 03:31.36 | *** join/#asterisk synapseattac (n=safy@ool-43539023.dyn.optonline.net) |
| 03:32.08 | clyrrad | fugitivo.... we have been watching and can not see the packets from anything other than SSH |
| 03:32.41 | fugitivo | then that's the problem |
| 03:33.49 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 03:34.11 | fugitivo | 2 options |
| 03:34.15 | fugitivo | 1- forwarding is not working |
| 03:34.21 | fugitivo | 2- the softphone is not configured correctly |
| 03:34.51 | clyrrad | the softphone worked with this configuration when connected internally so cant be that |
| 03:35.17 | fugitivo | but internally will use another ip address, or no? |
| 03:35.41 | clyrrad | yes, internally we use a 192... externaly we use a WAN IP |
| 03:36.01 | fugitivo | and is the correct ip entered in the softphone? |
| 03:36.04 | hat | hi, how can i see the output of NoOP() ? |
| 03:36.21 | clyrrad | fugitivo, yes we have trippled checked that |
| 03:36.33 | fugitivo | then forwarding is not working |
| 03:37.19 | clyrrad | hrm.... trying it again we have made all the forwarding changes you mentioned, but will double check again |
| 03:37.38 | fugitivo | what are you using as firewall/router? |
| 03:37.56 | clyrrad | openbsd/pf |
| 03:38.11 | clyrrad | with a VERY lean config |
| 03:38.13 | DarthClue | hat: NoOP outputs to the cli |
| 03:38.25 | clyrrad | DarthClue... yes |
| 03:38.28 | fugitivo | check the openbsd logs |
| 03:39.19 | clyrrad | what are we looking for in the logs? |
| 03:39.45 | fugitivo | dropped packets or something like that |
| 03:39.50 | clyrrad | we have dropped all block rules, we have got rid of everyting except those port forwards |
| 03:40.47 | clyrrad | are you running * behind a openbsd FW? |
| 03:40.59 | fugitivo | no |
| 03:41.18 | fugitivo | use tcpdump in the openbsd |
| 03:41.30 | *** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
| 03:41.30 | fugitivo | and check if you can see the packets |
| 03:41.56 | clyrrad | we have been doing that... we can not see the packets you refer to |
| 03:42.10 | fugitivo | on the openbsd? |
| 03:42.20 | fugitivo | or * ? |
| 03:42.29 | clyrrad | yup, and on * as well with sip debug ON |
| 03:42.50 | fugitivo | you sure the softphone is configured correctly? |
| 03:43.10 | *** join/#asterisk transgress (n=transgre@71.14.20.160) |
| 03:43.26 | clyrrad | I am going to say Yes, based on the fact that it works connected to an internal * box.... |
| 03:43.29 | Vco | can you telnet any other port number to the asterisk server from outside? |
| 03:43.37 | clyrrad | is there something different i need to add or change to connect to a remote * box? |
| 03:43.49 | fugitivo | clyrrad: just the ip address |
| 03:44.24 | clyrrad | we made the change to the WAN IP and tripple checked its the right one |
| 03:44.26 | Vco | god dotnetnuke is a pig........ |
| 03:45.09 | clyrrad | antying about a NAT ip or something like outgoing proxy we need to add in XLITE? that would be different from an internal * box? |
| 03:45.14 | fugitivo | clyrrad: can you put * with no fw for 2 min to test it? |
| 03:45.59 | clyrrad | no becase we are remote into the box... we have stripped out EVERYTHING all block rules you name it they are all gone, we are at a bare bones config with the port forwards you told us enabled |
| 03:47.41 | clyrrad | here are our firewall rules again |
| 03:48.50 | clyrrad | 5060-5063 tcp are open, udp 5060 is open, udp 10000-20000 are all open |
| 03:49.06 | clyrrad | here is the pastebin http://pastebin.ca/22212 |
| 03:51.05 | *** join/#asterisk bmg505 (n=leon@rndf-146-9-42.telkomadsl.co.za) |
| 03:52.16 | *** join/#asterisk spootnick (n=irc@CPE-147-10-168-100.nsw.bigpond.net.au) |
| 03:52.39 | fugitivo | clyrrad: ok |
| 03:52.59 | fugitivo | clyrrad: try a pass out for $int too |
| 03:53.53 | fugitivo | pass out on $int inet proto udp from any to 172.16.0.3 port $voip_udp keep state |
| 03:54.17 | *** join/#asterisk pussfeller (n=todd@12.150.129.171) |
| 03:54.20 | fugitivo | packets in from $ext and out to $int |
| 03:56.02 | fugitivo | clyrrad: maybe you want to use pass in quick for forwarded packets |
| 03:56.05 | clyrrad | ok those changes have been made, try to connect |
| 03:56.07 | *** join/#asterisk STUN (n=maxgluck@200.109.166.83) |
| 03:57.04 | fugitivo | pass in quick on $ext proto udp from any to 172.16.0.3 port $voip_udp keep state |
| 03:58.03 | STUN | any good source on configuring STUN with Asterisk? |
| 03:58.31 | STUN | I have already set up the server, but still can't register... |
| 03:58.32 | clyrrad | ok thats done |
| 03:58.44 | fugitivo | if it doesn't work i give up |
| 03:58.56 | clyrrad | LOL |
| 03:59.01 | gambolputty | I have a * box with 1.2.0 beta 1 using realtime. Incoming calls via SIP URL are not being processed by some test instructions I put in the default context. Any ideas? |
| 03:59.59 | clyrrad | are you able to connect to us? |
| 04:00.13 | fugitivo | you didn't give me the ip |
| 04:00.44 | clyrrad | <PROTECTED> |
| 04:03.02 | fugitivo | [ok... |
| 04:03.12 | fugitivo | do you see any packet? |
| 04:03.29 | clyrrad | no we can not see anyting on the CLI |
| 04:03.35 | fugitivo | tcpdump |
| 04:03.49 | clyrrad | from the * box? |
| 04:03.54 | fugitivo | yes |
| 04:04.00 | fugitivo | or bsd |
| 04:04.12 | fugitivo | no, better bsd |
| 04:04.22 | fugitivo | we know that packets aren't reaching * box |
| 04:05.15 | clyrrad | ok we are going to tcpdump on the bsd try to connect again |
| 04:05.58 | clyrrad | we see sip on port 449 |
| 04:06.41 | clyrrad | we are looking for your connection attempts to see if you are hitting any of the boxes |
| 04:06.41 | fugitivo | replace any with your ip address on the rdr rules |
| 04:06.48 | hat | hi, how to see the output of NoOp function? I cannot find it in the CLI console and /var/log/asterisk/full etc |
| 04:07.10 | fugitivo | rdr on $ext proto udp from any to 69.194.84.116 port 5060 -> 172.16.0.3 |
| 04:07.36 | *** join/#asterisk tq (n=tq@200.117.234.254) |
| 04:07.39 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 04:08.48 | spackle | hat, what is your verbose setting? |
| 04:09.33 | hat | where ? |
| 04:09.46 | hat | /usr/sbin/asterisk -vr |
| 04:09.59 | clyrrad | fugitivo... ok thats done |
| 04:09.59 | hat | actually, i also set the /etc/asterisk/logger.conf file |
| 04:10.27 | clyrrad | hat.... when you start asterisk start it as asterisk -vvvvvgc then you should see the NoOp output |
| 04:10.30 | spackle | use a few more v's or 'set verbose 10' at the console |
| 04:10.59 | hat | funny :( clyrrad, even i set ;debug => debug |
| 04:11.00 | hat | ;console => notice,warning,error |
| 04:11.00 | hat | console => notice,warning,error,debug |
| 04:11.00 | hat | ;messages => notice,warning,error |
| 04:11.00 | hat | full => notice,warning,error,debug,verbose |
| 04:11.00 | clyrrad | fugitivo..... mrse.com.ar is that you? We can see packets from this |
| 04:11.03 | hat | doesn't work |
| 04:11.36 | fugitivo | clyrrad: yes |
| 04:11.40 | hat | thanks. let me try |
| 04:11.46 | fugitivo | clyrrad: where do you see it? |
| 04:12.03 | clyrrad | ok we can see you on UDP 449 |
| 04:12.12 | clyrrad | and we can see our own connections as well |
| 04:12.16 | fugitivo | what about now? |
| 04:12.17 | clyrrad | there you are again |
| 04:12.19 | clyrrad | yup |
| 04:12.20 | *** join/#asterisk WilliamK (n=wkeller@c-67-172-202-228.hsd1.tx.comcast.net) |
| 04:12.21 | clyrrad | we can see you |
| 04:13.02 | WilliamK | clyrrad, I can tell, is that a bad thing? |
| 04:13.02 | WilliamK | =) |
| 04:13.33 | fugitivo | clyrrad: forwarding is not working, let me check your pf rules again |
| 04:13.45 | WilliamK | last CVS build is broken |
| 04:13.49 | clyrrad | we are making another pastebin |
| 04:13.49 | WilliamK | of * |
| 04:13.55 | Ariel_ | has anyone here worked with a wellgate/welltech 3804 before for use with asterisk? |
| 04:14.00 | JerJer | WilliamK: define broken |
| 04:14.30 | *** join/#asterisk opus_ (n=opus@dahphish.org) |
| 04:14.30 | WilliamK | Makefile is goofed |
| 04:14.36 | Qwell | JerJer: non working |
| 04:14.40 | WilliamK | make[1]: Entering directory `/usr/src/asterisk/apps' |
| 04:14.40 | WilliamK | Makefile:99: *** missing separator. Stop. |
| 04:14.40 | WilliamK | make[1]: Leaving directory `/usr/src/asterisk/apps' |
| 04:14.40 | WilliamK | make: *** [clean] Error 1 |
| 04:14.47 | WilliamK | that's the error |
| 04:14.51 | clyrrad | http://pastebin.ca/22214 |
| 04:14.58 | JerJer | smells like you had a confict |
| 04:15.02 | JerJer | what's on line 99 / |
| 04:15.03 | JerJer | ? |
| 04:15.23 | opus_ | rm -rf / |
| 04:15.25 | opus_ | whats up |
| 04:15.34 | Qwell | WilliamK: works here |
| 04:15.47 | WilliamK | Qwell, interesting |
| 04:15.50 | opus_ | try rm -rf Makefile && cvs co Makefile |
| 04:15.53 | WilliamK | you just pull from CVS? |
| 04:15.58 | Qwell | WilliamK: 2 seconds ago |
| 04:16.03 | Qwell | after you said something |
| 04:16.19 | *** join/#asterisk alexis101 (n=alexis@toronto-HSE-ppp4327833.sympatico.ca) |
| 04:16.27 | JerJer | my job here is done |
| 04:16.28 | WilliamK | and I pulled about 2 mins prior to that, wacky |
| 04:16.28 | opus_ | williamK I am installing from CVS onto a new machine which should be done in the next 10 minutes, lets see if it happeneds there. |
| 04:16.45 | clyrrad | fugitivo.... how does that pastebin look? |
| 04:16.47 | alexis101 | hi there ... i have a question about extconfig.conf |
| 04:17.01 | fugitivo | clyrrad: is that all your pf.conf? |
| 04:17.02 | Qwell | WilliamK: actually, Makefile hasn't changed since at least last night |
| 04:17.07 | WilliamK | the only thing I have installed besides * that would have touched that makefile is spanDSP |
| 04:17.13 | Qwell | codecs/Makefile has |
| 04:18.10 | WilliamK | <<<<<<< Makefile |
| 04:18.11 | alexis101 | i am wondering where do i write the password to connect at the mysql database ??? |
| 04:18.14 | WilliamK | that's line 99 |
| 04:18.20 | WilliamK | the patching mess it up? |
| 04:18.20 | opus_ | res_mysql.conf |
| 04:18.26 | alexis101 | thx |
| 04:18.31 | clyrrad | fugitivo.... yes thats all of it |
| 04:18.56 | Qwell | WilliamK: yeah, looks like its a conflict |
| 04:19.10 | Qwell | there was* |
| 04:19.29 | Qwell | WilliamK: You'll probably also see a >>>> line |
| 04:20.57 | WilliamK | ok, removed the comments from the file and it builds |
| 04:21.08 | WilliamK | cvs goofed it apparently |
| 04:23.09 | fugitivo | clyrrad: ok, try this http://pastebin.ca/22215 |
| 04:24.01 | fugitivo | clyrrad: that must work, if not, i give up, really |
| 04:27.20 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
| 04:27.38 | clyrrad | ok done.... try again |
| 04:30.07 | hat | hi, clyrrad, can i create an extension to dial two numbers and then connect this two calls together? |
| 04:32.15 | *** join/#asterisk liberie (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net) |
| 04:32.19 | *** join/#asterisk Exstatica (i=Exstatic@static-71-116-196-11.lsanca.dsl-w.verizon.net) |
| 04:34.16 | opus_ | SIP/EXTEN1&SIP/EXTEN2 |
| 04:34.17 | opus_ | then |
| 04:34.30 | opus_ | forward each call to a conference room |
| 04:35.16 | Qwell | Won't that only connect whichever one answers? |
| 04:35.27 | Ariel_ | Qwell, yes |
| 04:35.32 | Ariel_ | use a call file |
| 04:35.47 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
| 04:36.08 | Qwell | "Please hold while the other party connects." |
| 04:36.48 | Ariel_ | have a macro call each line then transfer it to the meetme is another way of doing it. |
| 04:37.01 | Ariel_ | or to each other. Hummm |
| 04:37.49 | hat | hj, How do I dialout using an extensions.conf and connect to an outside number? |
| 04:38.27 | hat | i have two numbers needed to be called by asterisk and to establish a call path between these two numbers |
| 04:38.45 | hat | Basically, i think this is a web based call back. |
| 04:38.49 | hat | please help |
| 04:38.52 | Ariel_ | exten => NXXNXXX,1,Dial(Zap/g1) |
| 04:39.30 | Ariel_ | hat, look at the wiki for settings on call files |
| 04:39.32 | hat | i think it doesn't work. |
| 04:39.48 | opus_ | how many people TOTAL in the conversation? |
| 04:40.02 | hat | Ariel_, i put a test.call file which is modified from sample.call to /var/spool/asterisk/output |
| 04:40.41 | hat | to trigger one number and then an extension(callback extension) is invoked, within that extension, i try to Dial the second number |
| 04:41.50 | Ariel_ | hat, there is a program out there called Vicidial which does that for you. It's used as a preditive dialer. |
| 04:42.21 | hat | Ariel_, what is Vicidial, an application of asterisk? |
| 04:42.36 | hat | Ariel_, does my way work? |
| 04:43.06 | Ariel_ | an add-on to asterisk from the people who bring you astguiclient |
| 04:43.36 | Ariel_ | hat, you need to do it via the a call fine or the manager api |
| 04:43.53 | hat | First, i put one file test.call to the outgoing directory and it does call the first numbre. |
| 04:44.07 | Ariel_ | boy it's taking along time to download the manual from Welltech for there gateway. |
| 04:44.44 | hat | once the first phone is pickup, the asterisk callback extension is invoked. Within |
| 04:45.30 | hat | this extension, i try to Dial the second number, It does work but only one person can hear the voice |
| 04:45.53 | hat | I don;t know whether it is due to firewall problem or not |
| 04:45.58 | opus_ | how many people TOTAL in the conversation? |
| 04:46.09 | hat | two |
| 04:46.21 | opus_ | then you have firewall problems |
| 04:46.25 | Ariel_ | only one person hears the voice |
| 04:46.28 | Ariel_ | hummm |
| 04:46.48 | Ariel_ | what type of devices are you using for the dialing? sip /zap /???? |
| 04:47.06 | hat | actually, there is no firewall, the asterisk server and two x-lite sip phones within the intranet |
| 04:47.15 | opus_ | then you have nat problems |
| 04:47.30 | hat | where nat come from, within the same network |
| 04:47.53 | hat | opus_, do you mean my x-lite configuration wrong |
| 04:47.53 | hat | ? |
| 04:47.58 | fugitivo | my desktop icons are huge |
| 04:48.18 | hat | yes, xlite is strange, it always try to discover some NAT ... |
| 04:48.34 | opus_ | hat just use the steps of elimination... if one x-lite works to the other, yay no network or conf problem |
| 04:49.14 | bkw__ | FYI folks that get the Double Ringback on sipuras.. look for Sticky 183 and set it to YES |
| 04:49.29 | file[laptop] | S.O.S! |
| 04:49.42 | Qwell | hmm |
| 04:50.02 | Qwell | anybody happen to know if ManxPower is still going to Astricon? |
| 04:50.12 | hat | opus_, the problem is that when i set verbose 10 from CLI, there is another error, let me login into that machine to get the error message |
| 04:50.21 | *** join/#asterisk root (n=root@202.171.49.33) |
| 04:50.29 | Qwell | ROOT HAS LANDED!! |
| 04:51.31 | Ariel_ | bkw_, sticky 183??? |
| 04:52.26 | Ariel_ | file[laptop], what can we assist you with? |
| 04:53.16 | file[laptop] | absolutely nothing |
| 04:53.28 | opus_ | absolute asterisk |
| 04:53.35 | Qwell | absolut vodka |
| 04:53.41 | file[laptop] | ANYWAY about the sticky 183 |
| 04:53.57 | Ariel_ | yes what about it? |
| 04:54.02 | file[laptop] | ariel_: a situation can occur where you get a 183 Session Progress with inband progress, and then a 180 Ringing... the Sipura mixes the audio together so you get two ringbacks |
| 04:54.28 | file[laptop] | sticky 183 causes the Sipura to ignore the 180 Ringing and use the inband progress |
| 04:54.44 | Ariel_ | ok so where is the sticky 183 |
| 04:54.51 | file[laptop] | ask bkw :) |
| 04:55.35 | JerJer | (00:49:22) bkw__: FYI folks that get the Double Ringback on sipuras.. look for Sticky 183 and set it to YES |
| 04:55.45 | hat | opus_, the error is Sep 7 12:15:50 WARNING[4880]: channel.c:2646 |
| 04:55.45 | hat | ast_channel_bridge: Private bridge between |
| 04:55.45 | hat | SIP/gwen-77f1 and SIP/yang-7864 failed |
| 04:55.49 | JerJer | i would presume in the advanced config methods |
| 04:55.59 | hat | i don't know what is that |
| 04:56.01 | JerJer | ok i am really outta here now |
| 04:56.13 | opus_ | hat your codec don't match |
| 04:56.21 | hat | how, |
| 04:56.22 | hat | ? |
| 04:56.37 | hat | opus_, do you know how i trigger the two calls? |
| 04:57.29 | hat | i put a call file into outgoing directory to trigger the first number, |
| 04:57.34 | opus_ | private bridge between - make sure they both have the same codecs |
| 04:58.08 | Ariel_ | hat use canreinvite=no on both setups for the sip phones |
| 04:58.24 | hat | and once the first phone is pickup, the callback extension is invoked by asterisk. Within callback extension, i call the second number. Ariel_, let me try |
| 04:58.26 | opus_ | Ariel - whoah yeah |
| 04:58.33 | hat | what that? |
| 04:58.52 | hat | Ariel_, what is the mechanism of noinvite |
| 04:59.20 | Ariel_ | canreinvite=no will keep asterisk in the path for translation of codec's |
| 04:59.49 | hat | let me try. wait. |
| 04:59.49 | Ariel_ | canreinvite=yes it will try to get your device connect to each others instead of going through asterisk. |
| 05:00.42 | hat | Ariel_, if i configure x-lite sip phone to use the same codes? |
| 05:00.51 | hat | anyway, let me try your suggestion first |
| 05:01.53 | hat | [gwen] |
| 05:01.53 | hat | type=friend |
| 05:01.53 | hat | username=gwen |
| 05:01.53 | hat | secret=gwen |
| 05:01.53 | hat | host=dynamic |
| 05:01.54 | hat | context=tutorial |
| 05:01.56 | hat | canreinvite=no |
| 05:02.00 | hat | right? |
| 05:02.09 | fugitivo | nat=yes |
| 05:02.15 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 05:02.20 | hat | why nat=yes sir |
| 05:02.21 | hat | ? |
| 05:02.22 | opus_ | qualify=yes |
| 05:02.26 | fugitivo | qualify=yes |
| 05:02.34 | fugitivo | oops |
| 05:02.37 | fugitivo | :) |
| 05:02.50 | hat | Even all components are within the same network? |
| 05:04.35 | *** join/#asterisk blake (n=blake@131.93.21.11) |
| 05:10.31 | *** join/#asterisk jeh (n=jeh@ext122.almare.com) |
| 05:10.36 | *** join/#asterisk Deedubb (n=Deedubb@S010600055d22c57f.vf.shawcable.net) |
| 05:10.55 | alexis101 | anyone here make realtime work with voicemail ?? |
| 05:11.22 | mog_home | yeah |
| 05:11.25 | *** join/#asterisk Stephnie (i=st@203.215.180.250) |
| 05:11.27 | mog_home | couple times |
| 05:11.46 | Stephnie | hi |
| 05:11.48 | Deedubb | Hello. Can I buy a TDM10B and purchase voip phones to have a voip system internally but keep my analog system externally? |
| 05:11.49 | alexis101 | because i had no problem with sip but voicemail driving me crazy |
| 05:12.27 | alexis101 | WARNING[25443]: app_voicemail.c:2602 leave_voicemail: No entry in voicemail config file for ********** |
| 05:12.32 | Ariel_ | Deedubb, with the rules you create you can do anything |
| 05:12.53 | mog_home | you have it really connecting over mysql alexis101? |
| 05:13.05 | Ariel_ | alexis101, check the context for voicemail you have setup |
| 05:13.11 | fugitivo | mysql is evil |
| 05:13.13 | Deedubb | Ariel_: but that card will enable me to connect my analog signal to my computer then I just use ether to talk to the phones right? |
| 05:13.33 | Ariel_ | Deedubb, well yes |
| 05:13.41 | Deedubb | Ariel_: the problem being I guess only one external call based on my current phone service |
| 05:13.45 | Ariel_ | mysql is good |
| 05:13.54 | Stephnie | how to get the length of incoming digits? any function in asterisk? |
| 05:13.59 | fugitivo | Ariel_: is a friend of sco, it's evil |
| 05:14.05 | fugitivo | did you read the news? |
| 05:14.15 | Deedubb | oh another slashdot monkey |
| 05:14.28 | Ariel_ | fugitivo, so if I did that then I would think that the world is coming to an end |
| 05:14.33 | alexis101 | yeah well in my extconf i have these line voicemail => mysql,asterisk,voicemail_users |
| 05:14.51 | Deedubb | fugitivo: for your own sanity's safety: stop reading slashdot comments |
| 05:15.14 | fugitivo | i didn't read slashdot comments |
| 05:15.20 | Ariel_ | fugitivo, I actually (don't let it be known) like MS SQL |
| 05:15.27 | fugitivo | i like ms sql too |
| 05:15.30 | Deedubb | OMG! |
| 05:15.43 | fugitivo | :) |
| 05:16.14 | Ariel_ | mog_home, are you working? |
| 05:16.29 | Deedubb | I've called slashdot, they said they're sending people in here right away to get the story so asterisk can be poop-listed for this abomination! |
| 05:16.43 | mog_home | always |
| 05:16.45 | mog_home | ariel |
| 05:16.54 | fugitivo | http://www.mysql.com/news-and-events/news/article_948.html |
| 05:17.34 | Ariel_ | mog_home, yes just was wondering. Manxpower was needing a te410p for some setup in Baton Rouge. |
| 05:17.44 | *** join/#asterisk liberie (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net) |
| 05:17.45 | *** join/#asterisk exes (i=1000@c-67-187-108-171.hsd1.tx.comcast.net) |
| 05:17.48 | mog_home | whats he need done |
| 05:17.57 | mog_home | i can pop on and hook him up |
| 05:18.02 | Ariel_ | needs a card for the morning |
| 05:18.07 | exes | where could I find information about the best possible phone to purchase |
| 05:18.17 | Ariel_ | polycome |
| 05:18.26 | Ariel_ | polycom polycom polycom |
| 05:18.31 | mog_home | bah |
| 05:18.33 | mog_home | cisco |
| 05:18.35 | Ariel_ | IP-600 |
| 05:18.36 | mog_home | 7960 |
| 05:18.50 | Ariel_ | cisco has sip firmware lisence issues |
| 05:18.51 | exes | I hear that the Ciscos can't utilize all the features of asterisk, this was from someone who has a cisco |
| 05:19.15 | DarthClue | polycom if you don't want to spend an arm and a leg, cisco if you don't mind sacrificing your first born |
| 05:19.19 | mog_home | well cisco wants you to use a call manager |
| 05:19.36 | mog_home | but cisco7960 is in my opinion classiest sip phone |
| 05:19.38 | Ariel_ | call manager sucks |
| 05:19.42 | mog_home | if you have an arm or leg to spare |
| 05:19.44 | mog_home | yeah |
| 05:19.45 | mog_home | dugh |
| 05:19.52 | mog_home | we are in #asterisk |
| 05:20.21 | spackle | polycom |
| 05:21.46 | Deedubb | Who has the cheapest VoIP phones in North America? |
| 05:21.48 | *** join/#asterisk drooth (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
| 05:22.09 | DarthClue | price or quality? |
| 05:22.14 | Deedubb | price |
| 05:22.17 | mog_home | budgetone |
| 05:22.27 | DarthClue | budgetone has it on both |
| 05:22.31 | mog_home | yeah |
| 05:22.35 | mog_home | they pretty junky |
| 05:22.40 | mog_home | but 40 bucks is 40 bucks |
| 05:23.23 | Deedubb | I'm just playing/testing |
| 05:23.32 | mog_home | your better off |
| 05:23.35 | mog_home | with a good ata |
| 05:23.39 | mog_home | also onloy a few bucks |
| 05:23.41 | shido6 | pap2-na |
| 05:23.44 | shido6 | $70 |
| 05:23.47 | fugitivo | yeah |
| 05:23.54 | spackle | agreed, sipura 2000 or whatever it is these days |
| 05:23.55 | fugitivo | pap2-na |
| 05:23.57 | shido6 | fuck it $65 |
| 05:24.09 | fugitivo | shido6: you lowered the price? |
| 05:24.13 | shido6 | just now |
| 05:24.26 | shido6 | Im putting up a page |
| 05:24.26 | fugitivo | tell me when it's $50 |
| 05:24.27 | mog_home | sipuras are like 50 used on ebay |
| 05:24.29 | mog_home | they own |
| 05:24.38 | mog_home | esp with a wireless phone |
| 05:24.44 | shido6 | well mine have 2 ports and are new in the box |
| 05:25.03 | spackle | shido6: are they unlocked? |
| 05:25.05 | fugitivo | shido6: i get new ones for that price in argentina |
| 05:26.17 | shido6 | PAP2-NA "NA" unlocked |
| 05:26.17 | shido6 | yes |
| 05:26.29 | shido6 | im in canada |
| 05:26.39 | shido6 | and can ship from the us if necessary |
| 05:26.52 | Deedubb | 2 phones can't talk directly to eachother eh? can 2 asterisk systems talk directly to eachother? like for branch offices? |
| 05:26.54 | exes | any opinions on the Polycom SoundPoint VoIP IP 300 |
| 05:27.03 | mog_home | yes deedubb |
| 05:27.05 | mog_home | they can |
| 05:27.06 | fugitivo | Deedubb: sure |
| 05:27.09 | Deedubb | sweet! |
| 05:27.16 | mog_home | and 2 sip phones can talk directly to each other |
| 05:27.22 | mog_home | but its not worth doing |
| 05:27.29 | spackle | deedubb, you can dial by IP address, pain in the ass. |
| 05:27.39 | mog_home | and they can have the media stream pass directly to each other |
| 05:27.42 | mog_home | via reinvite |
| 05:28.05 | Deedubb | but I assume with asterisk I could setup a 3 digit number to point to that IP? |
| 05:28.20 | Deedubb | and these phones don't have speed dial? |
| 05:28.36 | spackle | some have programmable buttons. |
| 05:28.40 | mog_home | yes you could |
| 05:28.44 | mog_home | but you wouldnt want to |
| 05:28.56 | mog_home | go read sip.conf in asterisk |
| 05:29.05 | Deedubb | as soon as I get a phone |
| 05:29.09 | mog_home | you dial sip/PHONE_NAME |
| 05:29.16 | *** part/#asterisk spackle (n=spackle@209.234.83.19) |
| 05:29.19 | mog_home | and phonename is how you labled the phone in asterisk |
| 05:29.29 | mog_home | but you can give it any extension |
| 05:29.47 | shido6 | Deedubb yes, all day long |
| 05:29.57 | mog_home | up to 2000 or something digits long |
| 05:32.32 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 05:33.49 | *** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it) |
| 05:37.52 | hat | opus_ and Ariel_ |
| 05:38.02 | Ariel_ | yes |
| 05:38.12 | hat | hi, i found the problem, it is due to one of my new microphone have problem :) |
| 05:38.26 | santoshr | help ... please.. if a call has been sent through a h323 channel.. how does the called party transfer the line.. u know flash kinda thing.. how is tht done |
| 05:38.26 | hat | it is brand new. But it just cannot work. Really sorry |
| 05:39.07 | opus_ | huh |
| 05:39.23 | opus_ | hat that one gets me like once every three months |
| 05:39.31 | Ariel_ | hat, ok glad you resolved the problem |
| 05:39.53 | hat | hehe. i just bought two microphone two days ago. and try to test this monring |
| 05:40.08 | opus_ | what type of microphones? |
| 05:40.19 | opus_ | i'm looking for a killer microphone for a speaker phone mic |
| 05:40.20 | hat | let me see the brand |
| 05:40.29 | opus_ | i want to have the mother-lode-of-god speaker phone system |
| 05:41.13 | opus_ | Polycom(R) SoundStation VTX 1000 is only $1800 |
| 05:41.15 | Ariel_ | opus_, wow |
| 05:41.19 | opus_ | fuck that |
| 05:41.26 | hat | Altec Lansing |
| 05:41.37 | opus_ | I want to spend $30 and have like dual channel ultra wide band speaker system using a soundcard |
| 05:41.40 | Ariel_ | IP-501 and IP-600 work great for speaker phones |
| 05:41.41 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 05:41.46 | fugitivo | why is so expensive a speaker with microfone? :) |
| 05:42.06 | Stephnie | http://pastebin.ca/22218 <----- any help please? |
| 05:42.07 | opus_ | they hold the patents that unlock the secret of all spreaker phones or something |
| 05:42.08 | Ariel_ | because he can..... |
| 05:42.22 | opus_ | oh |
| 05:42.23 | hat | by the way, how to check whether the voice stream is go though my asterisk server or not? |
| 05:42.49 | santoshr | guys.. |
| 05:42.51 | Ariel_ | voicemail system going through your sservice??? |
| 05:43.00 | Ariel_ | what does the cli say when you go to voicemail |
| 05:43.02 | santoshr | if a call has been sent through a h323 channel.. how does the called party transfer the line.. u know flash kinda thing.. how is tht done |
| 05:43.11 | hat | i didn't try voiceemail. |
| 05:43.36 | Ariel_ | santoshr, don't know don't use anything to do with h323 |
| 05:43.37 | oej | Stephnie: There's no encryption in the Asterisk dial plan, you have to use external commands with system() |
| 05:43.53 | fugitivo | system() is evil |
| 05:44.03 | hat | Ariel_, in my callback service, i call two legs. I am wondering whether voice stream is going through asterisk server or not |
| 05:44.04 | fugitivo | like mysql and sco |
| 05:44.15 | Stephnie | oej: I am not good on external commands.....any help on that or idea? |
| 05:44.21 | santoshr | Ariel_: i have a voip system .. in place.. wokring on h323. i was to setup an pbx on h323 |
| 05:44.30 | mog_home | why is system() evil? |
| 05:44.34 | mog_home | and mysql? |
| 05:44.35 | opus_ | Stephine - google for 'poker' 'agi' 'example' from i think pbx free ware, its a perl script |
| 05:44.37 | mog_home | i like mysql |
| 05:44.38 | oej | Stephnie: Man OpenSSL or "man crypt" |
| 05:45.12 | Ariel_ | I'm with mog_home |
| 05:45.14 | Deedubb | thanks for the info guys, good night |
| 05:45.22 | Stephnie | thanks opus_ , and oej ...I check these things... |
| 05:45.23 | mog_home | i mean mysql is a company |
| 05:45.25 | Ariel_ | Deedubb, good night |
| 05:45.33 | mog_home | they are gonna sell things to customers |
| 05:45.40 | *** join/#asterisk clive- (n=pirch@rndf-146-47-230.telkomadsl.co.za) |
| 05:46.04 | opus_ | "i'm sco certified" haha |
| 05:46.08 | fugitivo | system() is evil because it can do nasty things |
| 05:46.13 | mog_home | umm yeah |
| 05:46.21 | fugitivo | and mysql is nasty |
| 05:46.23 | opus_ | MYSQL advertises OLAP technology, however they don't know what the fuck it means haha |
| 05:46.23 | mog_home | any thing worth its salt can be used for great good or evil |
| 05:46.53 | mog_home | i mean asterisk can be used for telemarketing and sex hotlines, or to help get information to flood victimes |
| 05:46.54 | Ariel_ | well good and evil is part of everything. |
| 05:46.57 | opus_ | system("dial dirty 900 numbers and eat chicken after midnight") |
| 05:47.13 | fugitivo | hehe |
| 05:47.16 | Ariel_ | teast like chicken |
| 05:47.16 | opus_ | mog_home exactly |
| 05:47.17 | Stephnie | fugitivo: if system() is evil then any other solution ?? http://pastebin.ca/22218 ?? |
| 05:47.31 | shido6 | Asterisk can be used to detonate car bombs |
| 05:47.36 | mog_home | and if you want safe system |
| 05:47.47 | mog_home | just run asterisk as a user who has no ability to do anything |
| 05:47.49 | mog_home | but run asterisk |
| 05:47.58 | Ariel_ | shorewall |
| 05:48.05 | opus_ | Stephnie AGI |
| 05:48.14 | fugitivo | mog_home: that's dangerous |
| 05:48.20 | opus_ | cracking DTMF codes isn't that hard |
| 05:48.24 | mog_home | very |
| 05:48.24 | fugitivo | mog_home: asterisk user has privilegies over asterisk, evil! |
| 05:48.30 | mog_home | but so is system apperantly.... |
| 05:48.33 | Stephnie | ok |
| 05:48.56 | opus_ | run asterisk as root with strict selinux policy and turn on the NX bit :) |
| 05:49.04 | Ariel_ | argh it's late I need sleep, sleep I tell you. Sleep I need... |
| 05:49.26 | mog_home | ew /me thinks selinux is over rated |
| 05:49.28 | opus_ | later Ariel |
| 05:49.39 | Ariel_ | opus_, working on a system can't go yet |
| 05:49.40 | Ariel_ | argh |
| 05:49.53 | Ariel_ | that bed is sure nice a warm calling my name.... |
| 05:49.53 | opus_ | huhu? |
| 05:50.05 | mog_home | machine not doing your bidding ariel? |
| 05:50.09 | fugitivo | Stephnie: what you need to do, is too complicated for me to understand at this hours |
| 05:50.15 | opus_ | make clean && make bed |
| 05:50.24 | Ariel_ | mog_home, it's in NZ and there having issues with a wellgate |
| 05:50.34 | mog_home | wellgate? |
| 05:50.46 | Ariel_ | wellgate/welltech FXO gateway |
| 05:50.54 | mog_home | ahh |
| 05:50.59 | Ariel_ | I told them to get a TDM04b instead |
| 05:50.59 | mog_home | sip based i take it |
| 05:51.01 | Stephnie | fugitivo: I just want to change the incoming digits to something and then send them to another machine.. |
| 05:51.03 | mog_home | yay |
| 05:51.05 | mog_home | go digium |
| 05:51.16 | fugitivo | Stephnie: why you want to do that? |
| 05:51.21 | opus_ | AGI script will do it. |
| 05:51.23 | mog_home | stephnie i can do it |
| 05:51.35 | mog_home | but you will have to go through digium for me to do it |
| 05:51.40 | drooth | opus: the zyxel works! I'm stoked. |
| 05:52.05 | Stephnie | because of a very strange problem I have stucked in :S |
| 05:52.10 | opus_ | drooth cool, you gotta go war voip driving now |
| 05:52.11 | Stephnie | mog_home: how? |
| 05:52.23 | drooth | opus: i know! I should go now |
| 05:52.26 | mog_home | i would just open a tcp socket have shared key |
| 05:52.28 | mog_home | and send it over that |
| 05:52.37 | mog_home | or i would do it over my new spiffy jabber interface |
| 05:52.46 | mog_home | or tack it on iax or sip packet |
| 05:52.47 | mog_home | etc |
| 05:52.52 | Ariel_ | vpn |
| 05:53.00 | drooth | opus: Put an extension for yourself in my config and I will call you from it at my local coffee shop (free wifi 24 hrs.) |
| 05:53.05 | dudes | drooth - you get some work done today |
| 05:53.17 | Stephnie | mog_home: is that for me ? :o |
| 05:53.23 | santoshr | asteriskph: dude u around |
| 05:53.29 | mog_home | yeah stephnie |
| 05:53.36 | mog_home | its not to complicated |
| 05:54.02 | Stephnie | ok what you want me to do then? |
| 05:54.09 | santoshr | how do i transfer a line |
| 05:54.13 | mog_home | call or email digium a dev spec |
| 05:54.23 | mog_home | and tell them matt o'gorman was gonna do it |
| 05:54.28 | *** join/#asterisk grimse (n=grimse@p5481C30A.dip.t-dialin.net) |
| 05:54.28 | mog_home | they will get to you in the morning |
| 05:54.34 | mog_home | as it is 1 am out here |
| 05:55.14 | Stephnie | long procedure :( |
| 05:55.19 | mog_home | <PROTECTED> |
| 05:55.21 | mog_home | sorry |
| 05:55.27 | Stephnie | :) |
| 05:55.42 | mog_home | but as soon as you pay i could have it done in a few hours i think |
| 05:55.50 | mog_home | so can you take me through the flow? |
| 05:56.54 | Stephnie | how much I have to pay? |
| 05:57.01 | Ariel_ | mog_home, have you played with echo issue's and clicking problem with the tdm04b's rev I? |
| 05:57.02 | Stephnie | :) |
| 05:57.02 | mog_home | im not in sales.... |
| 05:57.03 | mog_home | sorry |
| 05:57.07 | mog_home | yeah |
| 05:57.13 | mog_home | its usually interrupt related |
| 05:57.15 | Ariel_ | rev I seems to have more problems then my E/H |
| 05:57.24 | mog_home | also there is a new kick ass echo can |
| 05:57.28 | mog_home | in zaptel |
| 05:57.33 | mog_home | but you have to define it |
| 05:57.38 | Ariel_ | yes but the fxotune gives me errors |
| 05:57.43 | mog_home | really |
| 05:57.48 | Ariel_ | unable to fill buffer |
| 05:57.49 | santoshr | how to lfash a line and transfer to a diff exten |
| 05:57.55 | mog_home | what version of code? |
| 05:58.05 | *** part/#asterisk outsidefactor (n=blah@203-206-247-109.dyn.iinet.net.au) |
| 05:58.06 | Ariel_ | about 4 hours ago |
| 05:58.12 | mog_home | hrm |
| 05:58.54 | Ariel_ | well it's late going to keep working on it in the morning. Got the well gate customer to wait till the morning as well. (bed is really calling) |
| 05:59.14 | mog_home | night |
| 05:59.20 | mog_home | if it is still happening tommorrow |
| 05:59.26 | mog_home | msg me |
| 05:59.27 | Ariel_ | good night folks. See you all in testing land in the morning. |
| 05:59.32 | mog_home | im sure we could get it fixed for you |
| 05:59.33 | fugitivo | good night ariel |
| 05:59.35 | Ariel_ | mog_home, sure will thanks |
| 05:59.40 | mog_home | no problem |
| 06:01.51 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
| 06:03.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:03.28 | Netgeeks | Hrm, can you nest Macros in the dialplan? |
| 06:03.42 | mog_home | i heard there was a bug with that at the moment |
| 06:03.46 | mog_home | but i havent tested it |
| 06:04.31 | Netgeeks | okay, I guess I can give it a test |
| 06:05.24 | mog_home | if it doesnt work it segfaults |
| 06:06.06 | Netgeeks | The dialplan so very much needs LIST or ARRAY type variables... |
| 06:07.39 | mog_home | you can do some nifty stuff with ael, but i havent played with it |
| 06:09.12 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 06:09.55 | *** join/#asterisk spootnick (n=irc@CPE-147-10-168-100.nsw.bigpond.net.au) |
| 06:10.43 | spootnick | i looked through the AGI API, it doesn't seem to have any function to retrieve the number of ongoing calls. does anybody know how to retrieve it? |
| 06:17.14 | *** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com) |
| 06:17.37 | opus_ | beer time! |
| 06:19.59 | dudes | it's not beer thirty ... it's whiskey is your friend time |
| 06:20.41 | Netgeeks | ouch, nope, nested macros are a no-no |
| 06:20.46 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
| 06:21.17 | mog_home | yeah thats what i thought |
| 06:22.54 | Netgeeks | Hey dudes, hows it coming along? |
| 06:23.04 | dudes | Netgeeks - it goes |
| 06:23.10 | *** join/#asterisk Russ (i=user@ip70-190-169-162.ph.ph.cox.net) |
| 06:23.38 | mog_home | well |
| 06:23.41 | Netgeeks | I'm getting REAL real close to a release of the project I've been working on |
| 06:23.46 | dudes | Wish things would get more organized, but we managed 34 agents on a SC opteron |
| 06:23.48 | mog_home | nice |
| 06:24.12 | dudes | with MOH and no issues. We'll see how many it will take soon once more leads coem in |
| 06:24.16 | Netgeeks | got the clustering all fixed, and am finishing up the trunk grouping, and it should be done |
| 06:24.45 | Netgeeks | The billing module and ITSP module are on track for end of the month completion as well |
| 06:24.49 | fugitivo | dudes: are you using opteron servers? |
| 06:25.15 | dudes | Netgeeks - we've been doing some mods to the core to cut code by 40% ... but I wish we'd finish the current core more |
| 06:25.23 | dudes | but the new core will kick a lot of ass |
| 06:25.44 | Netgeeks | hehehe, ah the woes of branched code |
| 06:25.44 | *** join/#asterisk nick125_lappy (n=nick@unaffiliated/nick125) |
| 06:26.27 | dudes | we just finished redesigning the code to work around the asterisk deadlock issues ... and now we have so much more stress testing to do soon |
| 06:26.33 | dudes | pain in the ass |
| 06:26.58 | nick125_lappy | hey, anyone here know of some configs for asterisk 1.2 beta that are small, and dont have alll the comments and such, like those one configs that have comments but are small...cant remember the name |
| 06:28.11 | Netgeeks | nick125, have you been drinking? |
| 06:28.35 | nick125_lappy | no lol |
| 06:28.42 | dudes | hehe |
| 06:29.19 | Netgeeks | you are looking for 1.2 beta configs that are small and don't have all the comments like those one configs that do have comments but are small? |
| 06:29.35 | Netgeeks | *boggle* |
| 06:29.38 | nick125_lappy | im not paying attention |
| 06:29.44 | Netgeeks | lol |
| 06:29.47 | dudes | Well pay attention |
| 06:29.58 | nick125_lappy | configs that have small comments, not like the big ol huge ones in the sample configs |
| 06:30.03 | Russ | whats a good way to get started with DID and outbound calls with DIY hardware? |
| 06:30.06 | Russ | sipphone? |
| 06:30.12 | Russ | (in the US) |
| 06:30.44 | Netgeeks | nick125: ah, no I don't know of any off the top of my head |
| 06:30.58 | mog_home | russ you can get dids from lots of people |
| 06:31.02 | mog_home | asterlink |
| 06:31.11 | mog_home | nuphone |
| 06:31.13 | Netgeeks | you could just kind of go through the samples and delete all the comments, then you would have small files with no comments |
| 06:31.15 | mog_home | etc |
| 06:31.24 | Russ | I know, there are tons of choices |
| 06:31.32 | Russ | but I don't have enough info to sort through them all |
| 06:31.51 | mog_home | nufone |
| 06:32.12 | *** join/#asterisk IgorG (n=gia@195.162.32.126) |
| 06:32.18 | Russ | whats the pricing on nufone? |
| 06:32.21 | nick125_lappy | i wish i could remember the name of those configs |
| 06:32.48 | Russ | I know, their site doesn't have pricing information... |
| 06:32.53 | DarthClue | asterlink, 2 cents per minute, toll free did |
| 06:33.01 | *** part/#asterisk IgorG (n=gia@195.162.32.126) |
| 06:33.07 | Russ | I don't want a toll free did |
| 06:33.09 | Russ | I want a local one |
| 06:33.20 | mog_home | lol |
| 06:33.25 | Assid | umm.. toll free means they charge you for incoming right? |
| 06:33.28 | mog_home | sorry i didnt know term rates |
| 06:33.49 | fugitivo | troll free |
| 06:33.54 | DarthClue | picky, picky. you almost always get charged for did, whether local or toll free |
| 06:34.25 | Russ | I expect a monthly fee and per minute change |
| 06:34.28 | Russ | er, charge |
| 06:34.29 | *** part/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 06:34.47 | DarthClue | Russ, asterlink may not have local, but they can get you per minute with no monthly commitment |
| 06:34.52 | opus_ | dudes what are you working on curious |
| 06:34.55 | hat | hi, who is using TE411P digium card? Which computer (hardware) is better? |
| 06:35.02 | Assid | DarthClue: for the DID yes.. but what about per minute charges of the call itself? |
| 06:35.03 | mog_home | have tested it hat |
| 06:35.25 | *** join/#asterisk mmmToop (n=chatzill@196.209.43.6) |
| 06:35.33 | mog_home | it works great in dl360 |
| 06:35.34 | hat | mog_home, what computer are you using? |
| 06:35.39 | mog_home | and dell 2850 |
| 06:35.41 | hat | dell360? |
| 06:35.43 | DarthClue | um, ok, maybe you guys need some coffee ... 2 CENTS PER MINUTE ... scroll up |
| 06:35.54 | hat | two cpu or 1 cpu? |
| 06:36.03 | mog_home | no compaq dl 360 g4 |
| 06:36.28 | *** join/#asterisk SaltY_ (n=altraide@61.68.220.96) |
| 06:36.39 | hat | i never use compaq before. HOw is IBM server? |
| 06:36.39 | opus_ | <dudes> Netgeeks - we've been doing some mods to the core to cut code by 40% ... but I wish we'd finish the current core more |
| 06:36.45 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:36.46 | opus_ | asterisk codebase? |
| 06:36.49 | *** part/#asterisk SaltY_ (n=altraide@61.68.220.96) |
| 06:36.56 | mog_home | havent played with it |
| 06:37.02 | mog_home | havent heard anything bad either |
| 06:37.05 | dudes | opus_ - gnudialer code |
| 06:37.12 | Russ | hawe you heard anything about sipPhone? |
| 06:37.18 | mog_home | yes |
| 06:37.20 | mog_home | they are fine |
| 06:37.26 | Assid | hrmm.. would a p4 2.4 be able to handle 20-30 simultanous calls in pass through mode? |
| 06:37.28 | hat | dell 2850? 1 or 2cpu? |
| 06:37.30 | opus_ | oh, yeah drooth sent me a link about that |
| 06:37.30 | mog_home | almost all pro. voip providers are the same |
| 06:37.33 | drooth | opus: dudes is the guy who made the dialler I showed you be4 |
| 06:37.36 | mog_home | yes assid |
| 06:37.41 | mog_home | if no transcoding |
| 06:37.42 | opus_ | cool |
| 06:37.48 | mog_home | i think we have one that is 1 one that is 2 |
| 06:37.54 | drooth | we are talking about it; it's awesome! |
| 06:37.59 | opus_ | cool dude:) |
| 06:38.04 | drooth | cool dudes |
| 06:38.12 | hat | mog_home, how about the performance? cpu usage? |
| 06:38.18 | Assid | mog_home: upto how many do you think it would be possible.. only pass through.. no transcoding? |
| 06:38.22 | opus_ | hey Netgeeks whens your super asterisk build coming out:) |
| 06:38.27 | drooth | opus I put in the TOS bit and reset |
| 06:38.29 | mog_home | 2.4 ghz |
| 06:38.34 | mog_home | what type |
| 06:38.37 | mog_home | sip zap iax? |
| 06:38.44 | mog_home | hat its a beast |
| 06:38.49 | opus_ | drooth - the screen -x feature is fucking awesome never knew about it until today |
| 06:38.50 | mog_home | you can do loads of calls on it |
| 06:38.51 | hat | EuroISDN |
| 06:38.54 | Assid | iax |
| 06:38.55 | drooth | opus: serious??? |
| 06:38.59 | opus_ | yeah |
| 06:39.03 | drooth | cool at least i can offer something |
| 06:39.20 | hat | mog_home, how many current call does dell 2850/2cpu server can handle? any idea? |
| 06:39.29 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 06:39.33 | mog_home | hmm well two te411s |
| 06:39.35 | hat | IBM is not cheap compared to DELL |
| 06:39.38 | drooth | opus u want me to try your ext. real quick |
| 06:39.41 | mog_home | so 250 or so zap channels |
| 06:39.43 | Netgeeks | opus_ I'm finishing up the trunking section tonight, and probably will have it tested by the end of the week. It will be ready then, but I've got two HUGE projects one due on the 15th and the other on the 19th, so I will probably not do much about it til the week of the 26th |
| 06:39.46 | mog_home | maybe 500 or so sip |
| 06:39.46 | opus_ | lemme check the exten |
| 06:40.03 | Assid | 500 sip ?? on a 2.4 Ghz? |
| 06:40.16 | hat | mog_home, not bad. do you mean that all voice stream passing throuhg Dell server? |
| 06:40.25 | mog_home | i believe so |
| 06:40.29 | fugitivo | mog_home: doing transcoding? |
| 06:40.29 | mog_home | but no transcoding |
| 06:40.33 | mog_home | actually i know so |
| 06:40.35 | fugitivo | ok |
| 06:40.49 | mog_home | it can do a full quad card of g729 |
| 06:40.50 | mog_home | i think |
| 06:40.54 | hat | so powerful? I really don't believe :) |
| 06:40.56 | *** join/#asterisk dolson (n=dana@toronto-HSE-ppp4302073.sympatico.ca) |
| 06:40.57 | mog_home | i believe |
| 06:41.03 | mog_home | id have to get specs for you |
| 06:41.08 | mog_home | i think we have em on website |
| 06:41.12 | mog_home | somewwhere... |
| 06:41.17 | hat | please let me know. |
| 06:41.18 | mog_home | i dont work in that lab |
| 06:41.18 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 06:41.26 | mog_home | yeah no problem hat |
| 06:41.27 | Assid | yep no transcoding.. its only there to route the calls.. and incase the other asterisk box isnt available.. then it will do voicemail recording.. and email it .. but till then.. it will only just route it |
| 06:41.47 | mog_home | what assid? |
| 06:41.51 | hat | i have one IBM server with 2CPU already. And i need to get another one |
| 06:42.02 | hat | either dell or IBM. I am choosing |
| 06:42.06 | mog_home | terminating a lot of calls? |
| 06:42.12 | hat | callback service |
| 06:42.13 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 06:42.18 | hat | two number callback |
| 06:42.42 | Assid | yeah.. but the other asterisk box (which eventually gets the calls, the net keeps going down) so during that time.. i need to put it to voicemail etc. |
| 06:43.03 | hat | In addition, is it possible to plug two te411p cards into the same machine? |
| 06:43.07 | mog_home | well i know there are some people doing watchdog stuff |
| 06:43.10 | mog_home | like that |
| 06:43.13 | mog_home | yes it is hat |
| 06:44.06 | Assid | mog_home: what about a 3.0ghz HT, using agents/queue and agent recording ? |
| 06:44.10 | hat | thanks. mog_home. If you can pass me some specs about dell 2850 and loading testing result, i am very appreciate |
| 06:44.18 | mog_home | muxing your recordings? |
| 06:44.21 | Assid | yep |
| 06:44.27 | Assid | need to sound like a conversation |
| 06:44.33 | mog_home | yeah hat, i think we have em publicly available |
| 06:44.47 | mog_home | yeah, assid, well what format are the calls comming in |
| 06:44.51 | Assid | gsm |
| 06:44.58 | mog_home | recording in gsm? |
| 06:45.00 | Assid | ilbc is heavier |
| 06:45.06 | Assid | yes.. recording in gsm as well |
| 06:45.17 | Assid | but.. this has to do transcoding as well.. |
| 06:45.21 | dudes | ilbc isn't a bad codec ... |
| 06:45.30 | Assid | ulaw->gsm |
| 06:45.36 | dudes | lpc10/20 ... ecks |
| 06:45.45 | Assid | dudes: ilbc is more resource hungry |
| 06:45.53 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
| 06:46.32 | dudes | Assid - that it maybe. But it's a good sounding codec and pretty good on b/w |
| 06:46.54 | Assid | dudes: gsm and ilbc are pretty comparable to one another |
| 06:46.58 | fugitivo | gsm is good too |
| 06:47.19 | *** join/#asterisk Osurac (n=joe@ip68-227-215-210.dc.dc.cox.net) |
| 06:47.27 | dudes | Assid - I like ilbc more than gsm myself |
| 06:47.30 | Assid | mog_home: any clue ? |
| 06:47.43 | Assid | dudes: i used to as well, now recently.. i just use gsm |
| 06:48.03 | Assid | mog_home? |
| 06:48.08 | *** part/#asterisk Osurac (n=joe@ip68-227-215-210.dc.dc.cox.net) |
| 06:48.16 | dudes | I haven't tried GSM on the box we have asterisk on yet |
| 06:48.26 | dudes | But on the old box ilbc was better |
| 06:49.25 | hat | mog_home, from where i can get the load testing result? thanks |
| 06:49.30 | Assid | mog_home: you still around? |
| 06:49.36 | clive- | I like gsm...its very easy on the cpu |
| 06:49.50 | Assid | yeah.. easier on CPU = more agents.. |
| 06:49.54 | mog_home | im back |
| 06:50.12 | Assid | specially since your talking about transcoding + recording |
| 06:50.12 | mog_home | so your recording to gsm from gsm? |
| 06:50.23 | Assid | yes.. PLUS.. transcoding.. |
| 06:50.28 | Assid | ulaw->gsm |
| 06:50.34 | mog_home | hrm |
| 06:50.59 | mog_home | i would think at least a full quad span but i am not sure |
| 06:51.00 | Assid | they bought this stupid voip router device (8 port) |
| 06:51.05 | razu | is there any way to control the volume on any sip phones ? |
| 06:51.12 | razu | on asterisk ? |
| 06:51.18 | mog_home | you can tast it assid real easily |
| 06:51.21 | Assid | which doesnt have gsm codec mentioned in there |
| 06:51.23 | *** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) |
| 06:51.26 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 06:51.26 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 06:51.44 | mog_home | i mean you get another pc |
| 06:51.49 | mog_home | that just puts in iax calls in ulaw |
| 06:51.58 | mog_home | have your other end answer in gsm |
| 06:52.08 | mog_home | and record it |
| 06:52.27 | mog_home | and barge the channel or listen till the recordings till they sound bad |
| 06:53.00 | Assid | mog_homethe recording+transcoding must happen on the same pc |
| 06:53.15 | mog_home | yeah i know |
| 06:53.20 | *** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net) |
| 06:53.24 | mog_home | you have the second pc seperate |
| 06:53.24 | Assid | as of right now.. they are gonna be starting with 20 agents |
| 06:53.30 | mog_home | taht isnt getting over loaded |
| 06:53.37 | mog_home | to test the loading on the other box |
| 06:54.26 | Assid | well.. not in production.. so i could use the same box for now.. |
| 06:54.37 | Assid | i gotta yet "start" it sometime this week |
| 06:54.55 | mog_home | indeed |
| 06:55.31 | Assid | how many do you think i should be able to handle? |
| 06:55.32 | Assid | just a guess |
| 06:55.47 | mog_home | i think maybe 96 channels |
| 06:55.50 | mog_home | but im not sure |
| 06:55.56 | mog_home | you said it was single proc? |
| 06:56.21 | Assid | single proc physically.. HT .. so it catches it as a SMP (logical) |
| 06:56.38 | Manipura | I have a question, what linux distro is it easiest to install * on |
| 06:56.45 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 06:56.47 | Manipura | so far, RH9 for me. |
| 06:56.51 | mog_home | debian |
| 06:56.58 | mog_home | but fedora and redhat are good |
| 06:57.05 | Assid | Manipura: doesnt make a difference.. but debian is prolly the easist |
| 06:57.11 | Assid | apt-get install asterisk |
| 06:57.16 | mog_home | ewww |
| 06:57.16 | Assid | unless you want the later versions |
| 06:57.18 | Manipura | fedora needed some weird kernel config or something I couldn't understand |
| 06:57.18 | mog_home | build asterisk |
| 06:57.22 | mog_home | from source |
| 06:57.23 | mog_home | always |
| 06:57.24 | Assid | yeah |
| 06:57.27 | Assid | i use HEAD |
| 06:57.34 | mog_home | even if you use stable |
| 06:57.48 | mog_home | i trust digium cvs much more than debian redhat etc 's stuff |
| 06:58.03 | Assid | somehow .. the first and ever time i tried stable.. it crashed on me |
| 06:58.08 | mog_home | lol |
| 06:58.12 | Manipura | Yep, me too.. after I failed my friend tried installing it with yast on his suse system... lol... never got it started |
| 06:58.12 | mog_home | it happens |
| 06:58.14 | Assid | so i started following cvs |
| 06:58.14 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 06:58.54 | santoshr | how to flash a line and listen to a exten tht the party dials.. suppose i sent a call to call@ip over h323. and he want to send it to some other extension.. something tht happens a normal physical epbax call transfer.. how to do tht in asterisk |
| 06:58.56 | *** join/#asterisk abel (n=abel@tor/session/x-93ba141b5cd28b82) |
| 06:59.22 | wasim | santoshr: with zap channels you can use Flash(), with others you can use # transfer |
| 07:00.32 | nick125_lappy | i knew i should have done these configs earlier |
| 07:00.35 | Assid | hrmm.. is there a way to set a "transfer" while a call is ringing.. |
| 07:00.44 | santoshr | wht would the syntax be for # transfer |
| 07:00.49 | Manipura | nick125_lappy is in the same time zone as me |
| 07:01.05 | wasim | santoshr: you press #, asterisk says transfer, you dial the extension and hangup |
| 07:01.31 | nick125_lappy | Manipura, neat |
| 07:01.32 | Assid | like if im calling exten 1234 .. and during the rining of the call.. i want to cut away and go to another extension instead |
| 07:01.42 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
| 07:01.56 | Manipura | Although I'm canadian...... *evil grin* |
| 07:02.00 | mog_home | you want to drop the call / or give it to someone else? |
| 07:02.13 | wasim | santoshr: you ahve to have T or t or both in your Dial options |
| 07:02.31 | nick125_lappy | this box should be fun, pretty much putting two different things on the same asterisk box....contexts should make this easy...hopefully |
| 07:02.38 | Assid | well.. i wnt it to allow the user to change extenions BEFORE the call is picked up |
| 07:03.28 | santoshr | wasim: T or t ? |
| 07:03.50 | nick125_lappy | yay more fun, have to setup sip clients...fun fun *cough* |
| 07:04.34 | *** join/#asterisk Dybdahl (n=Lars@cpe.atm2-0-7138.0x50a6f736.odnxx10.customer.tele.dk) |
| 07:04.54 | hat | mog_home, just confirm with you that dell 2850 cpu works with te4110P card well, right? I just follow your experience and don't want to challenge myself :) |
| 07:05.08 | nick125_lappy | i hope this works with 1.2 lol |
| 07:05.22 | mog_home | yes it will |
| 07:05.27 | mog_home | i dont have em |
| 07:05.28 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-13-64.cybersurf.com) |
| 07:05.35 | mog_home | but its what we use to test at digium |
| 07:06.02 | hat | sorry, what is "em"? |
| 07:06.14 | mog_home | we use 2850s to test asterisk be |
| 07:06.22 | mog_home | and we have tested 410 and 411 on it |
| 07:06.35 | hat | 2850? |
| 07:06.43 | mog_home | dell 2850 |
| 07:06.51 | mog_home | as well as the compaq dl360 |
| 07:07.12 | Russ | are there any DID services that let you keep your existing number? |
| 07:07.13 | hat | i am scrared because digium te411p doesn't work on some hardware. |
| 07:07.56 | nick125_lappy | Russ, you could port it to somewhere, you might be able to do that with someone like voicepulse or broadvoice |
| 07:08.28 | nick125_lappy | porting the number usually takes about 24 hours or so |
| 07:08.52 | hat | mog_home, what os you are using on 2850? |
| 07:08.54 | Russ | I heard bad things about broadvoice, but I also heard good things |
| 07:09.04 | mog_home | we tested with fedora core 3 |
| 07:09.10 | mog_home | and redhat enterprise |
| 07:09.31 | hat | good. i have redhat enterprise |
| 07:09.54 | Assid | mog_home: does asterisk use/require any special flags to use the full processing of a box? |
| 07:10.03 | mog_home | no |
| 07:10.03 | Assid | like if i have SMP |
| 07:10.06 | mog_home | but it runs better as root |
| 07:10.13 | Assid | yeah.. it would run as root |
| 07:10.13 | hat | mog_home, is this : Linux mobmeee 2.4.21-4.ELsmp #1 SMP Fri Oct 3 17:52:56 EDT 2003 i686 i686 i386 GNU/Linux |
| 07:10.18 | mog_home | as root has slightly more importance than regular users |
| 07:10.33 | mog_home | it wont matter it will execute threads on both procs |
| 07:10.38 | mog_home | if you didnt edit the code |
| 07:11.11 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 07:11.14 | Assid | okay.. thought maybe i have to compile with certain flags or something to get the little extra juice outta it |
| 07:11.20 | mog_home | nope |
| 07:11.32 | mog_home | we try to be hardcore from the start |
| 07:11.51 | Assid | u in the dev team? |
| 07:12.15 | Assid | you even |
| 07:12.22 | Assid | bad habbit :( |
| 07:12.30 | mog_home | heh |
| 07:12.52 | hat | dell 2850 with 2cpu and 2G memory is about USD$3000. |
| 07:12.54 | mog_home | i am a padawon in the dev team |
| 07:12.58 | mog_home | they are jedis... |
| 07:13.22 | hat | is dell server stable? i have no experience of using dell server before |
| 07:13.30 | mog_home | yeah its quality |
| 07:13.33 | Assid | hat: yep |
| 07:13.52 | Assid | memory isnt that much of a requirement in asterisk |
| 07:13.54 | Assid | from what i heard |
| 07:13.59 | mog_home | yeah |
| 07:14.00 | Assid | ijts more just abt the cpu processing |
| 07:14.02 | mog_home | its all about proc |
| 07:14.10 | hat | Assid, thanks. i will get one. IBM is much more expensive... |
| 07:14.21 | mog_home | we could care less about mem |
| 07:14.30 | hat | Assid, i will run another application server in the same server. |
| 07:14.41 | Assid | yep..thats why im just sticking to 512 |
| 07:14.49 | mog_home | yeah thats fine |
| 07:15.00 | Assid | hat: so am i.. mail server (for sending out the voicemails) and ftp.. to login and download the agent recordings |
| 07:15.12 | Assid | and oh yeah.. postgres for CDR |
| 07:15.23 | Assid | or mysql.. not sure which to chose yet |
| 07:15.48 | hat | Assid, cool. solved my problem. I cannot make decision before discussion due to the uncertainty of compatibility issue |
| 07:15.50 | Assid | and ofcourse apache.. for logs |
| 07:16.24 | hat | Assid, i also use postgresql for cdr . is it possible to use postgresql to store extension definition also? |
| 07:16.38 | Assid | mog_home: if the muxing/recording server gets full.. how do i load balance with another server |
| 07:16.48 | hat | I am using JBoss application server to do the frontend work. |
| 07:17.04 | Assid | hat: realtime application .. new thing in CVS .. check it out |
| 07:17.23 | Russ | hmm..thats pointless, the places that take your old number don't serve my area code |
| 07:17.29 | hat | hm.. |
| 07:17.33 | mog_home | well |
| 07:17.35 | mog_home | there is a patcvh |
| 07:17.40 | mog_home | on pbxfreeware.org |
| 07:17.50 | mog_home | i believe that starts recording as one |
| 07:17.52 | mog_home | so there is no muxing |
| 07:17.59 | Assid | this is what im thinking |
| 07:18.10 | mog_home | that would be best way to record that much |
| 07:18.13 | Assid | the server which does nothing but route the calls.. |
| 07:18.31 | Assid | should ring alternatively on alternate servers |
| 07:18.56 | Assid | routing server ->server A/B |
| 07:19.06 | Assid | just gotta figure out how to do that |
| 07:19.41 | Assid | routing server should have a pretty high limit since no transcoding.. and works with GSM |
| 07:21.02 | Assid | i know queue/manager/agent you can specify how to ring which extension.. but i cant figure out how to do it in the basic level |
| 07:21.22 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 07:21.46 | hat | hi, what is ETSI ? |
| 07:21.54 | hat | sorry, i am new to telephony |
| 07:22.13 | Assid | brb... gotta get ready |
| 07:24.32 | *** join/#asterisk secure75 (n=mic@p549A0A51.dip0.t-ipconnect.de) |
| 07:26.18 | hat | Hi, Assid, what is ETSI E1 ? |
| 07:26.42 | nick125_lappy | whats the new syntax for asterisk 1.2 for caller id? |
| 07:27.00 | nick125_lappy | (setting callerid) |
| 07:27.13 | *** join/#asterisk syle2 (n=blag@wnpgmb06dc1-44-164.dynamic.mts.net) |
| 07:27.28 | nick125_lappy | aint it like Set(CALLERID=())? |
| 07:28.38 | X-Rob | <PROTECTED> |
| 07:28.38 | X-Rob | <PROTECTED> |
| 07:28.38 | X-Rob | <PROTECTED> |
| 07:28.42 | X-Rob | (from 'UPGRADE.txt'.) |
| 07:29.10 | nick125_lappy | ah, i knew it was something weird like that lol |
| 07:30.06 | *** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it) |
| 07:30.16 | *** join/#asterisk Lars (n=Lars@cpe.atm2-0-7138.0x50a6f736.odnxx10.customer.tele.dk) |
| 07:33.19 | drooth | hey all, i noticed my TOS bit is being stripped on ougoing calls, any ideas why? |
| 07:33.33 | hat | Assid, what signaling used by your E1 line? |
| 07:33.39 | drooth | it is happening on different phones |
| 07:39.03 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 07:41.25 | *** join/#asterisk Lars (n=Lars@cpe.atm2-0-7138.0x50a6f736.odnxx10.customer.tele.dk) |
| 07:41.41 | *** join/#asterisk manojoswal (n=shree@59.95.2.154) |
| 07:41.54 | manojoswal | hi |
| 07:42.12 | manojoswal | i need paid support for asterisk |
| 07:42.16 | manojoswal | anyone who can help |
| 07:42.17 | manojoswal | ? |
| 07:42.29 | wasim | manojoswal: what areas? |
| 07:42.39 | Zeeek | be sure to ask on the biz or user mailing list as well |
| 07:43.00 | wasim | bonjour monsieur zeeek |
| 07:43.14 | manojoswal | i have an asterisk server set, i need to make automatic calls with a fixed message |
| 07:43.14 | Zeeek | hi wasim! |
| 07:43.45 | manojoswal | the server was running, now its facing problems, message not being played |
| 07:44.29 | wasim | manojoswal: whats the problems/ |
| 07:44.46 | manojoswal | not sure, i am a total non techie |
| 07:44.54 | wasim | manojoswal: where is your techie? |
| 07:45.06 | manojoswal | if you can provide support we can chat on msn of something |
| 07:45.13 | manojoswal | he is not responing |
| 07:45.40 | manojoswal | he is not responding back he has left th porject in between |
| 07:46.07 | wasim | don't you hate it when that happens |
| 07:46.17 | oej | manojoswal: Where are you located? |
| 07:46.23 | jalsot | hi |
| 07:46.24 | manojoswal | pune, india |
| 07:46.33 | wasim | wheee ... nearby too |
| 07:46.33 | manojoswal | hate it badly |
| 07:46.43 | manojoswal | where are u located? |
| 07:46.52 | oej | I am in Stockholm, Sweden |
| 07:46.56 | jalsot | is here any iax guru? having some problems with iaxcomm bulk unregistration, UDP checksum mismatch, etc. |
| 07:47.09 | *** join/#asterisk Bonzai090 (n=pirch@wbs-146-190-212.telkomadsl.co.za) |
| 07:47.29 | oej | wasim: The IAX guru you are. Help friends need. Ahead you step. :-) |
| 07:47.48 | jalsot | :) |
| 07:47.50 | oej | Bad Yoda-imitation... |
| 07:48.07 | Bonzai090 | hi all.. i have a default extention 200 that all calls gets forwarded to how can i get any of the other extentions to pick up that call if no one is at that extenention?. |
| 07:48.29 | *** join/#asterisk teapot (n=tandrews@mail.grok.org.za) |
| 07:48.30 | oej | Bonzai090: Check call gruoups and pickup groups |
| 07:48.38 | Bonzai090 | well all incomming calls en at extention 200 |
| 07:48.51 | Bonzai090 | oej ok thanks will google on that |
| 07:48.58 | teapot | morning |
| 07:49.01 | oej | bonzai090: np |
| 07:49.18 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:49.33 | Bonzai090 | asterisk has been a steep learning curve i am short alot of hair lol |
| 07:49.40 | nick125_lappy | i think i just got my asterisk back up without going back to amp :o |
| 07:49.51 | teapot | can someone tell me what a "native bridge" is |
| 07:49.55 | *** join/#asterisk meppl (n=mephisto@p54AADB0A.dip.t-dialin.net) |
| 07:49.56 | Bonzai090 | hreh nick125_lappy i tiik me 3 days to get that right hehe |
| 07:49.57 | *** join/#asterisk juice_ (n=juice@mo-67-77-188-181.dyn.sprint-hsd.net) |
| 07:49.58 | jalsot | I use asterisk cvs head [-08/17/05] and iaxcomm clients connected to * [on the same switch]. in 1st round they were registered [not all for 1st step - typo in pwd :)], but after some time they got unreacheable |
| 07:50.09 | X-Rob | indians and aboriginies make it when they want to cross a body of water without getting wet. |
| 07:50.16 | manojoswal | wasim : can you pls chat with me on msn |
| 07:50.24 | teapot | :) |
| 07:50.26 | Bonzai090 | lol X-Rob |
| 07:50.27 | nick125_lappy | im soo tired im like shaking |
| 07:50.32 | teapot | and africans X-Rob |
| 07:50.34 | jalsot | turned on qualify, and somehow they got unregistered because they were not available |
| 07:50.40 | Bonzai090 | nick125_lappy get some red bull hehe |
| 07:50.44 | jalsot | any idea what can be wrong? |
| 07:50.44 | oej | teapot: A bridge is when Asterisk connects an incoming and outgoing call. Native bridge is used when both call legs are on the same technology, like SIP, IAX2, ZAP - then the channel can do briding, which means that we can do local tricks in the channel |
| 07:51.01 | oej | teapot: Like the RTP native bridge where we re-invites calls |
| 07:51.27 | teapot | oej: so on a zaptel (digium) card you would connect 2 channels directly ? |
| 07:51.42 | teapot | (without going via PCI... ?) |
| 07:51.50 | oej | teapot: I don't know much about zaptel, but that's possible |
| 07:52.11 | oej | teapot: I don't think it's hardware only, but maybe zaptel kernel-driver only, not involving Asterisk |
| 07:52.29 | teapot | oej: ah, ok |
| 07:53.43 | *** join/#asterisk shree (n=shree@61.3.176.15) |
| 07:53.50 | shree | hi |
| 07:53.51 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:54.07 | jalsot | is it normal that I see checksum incorrect in UDP header for IAX2? tetherel showed that on asterisk box |
| 07:54.23 | Netgeeks | oej: you know much about the realtime architecture internally? |
| 07:54.42 | oej | Netgeeks: I can't claim to know that. A bit, but not all of it. |
| 07:55.02 | shree | hi |
| 07:55.20 | shree | anyone here who can offer paid professional support on troubleshooting astrerisk |
| 07:55.45 | oej | shree: Would be easier if you told us the problem area. Everyone has different expertise |
| 07:55.52 | Netgeeks | just wondering how difficult it would be to add a "system-id" field. Where an asterisk system would only fetch lines from teh db where the system id field matched a value set maybe extconfig.conf or such |
| 07:56.12 | mog_home | ^_^ |
| 07:56.18 | oej | Netgeeks: Good question. We need a global system-id in Asterisk for sure, that is a good use of it |
| 07:56.20 | mog_home | or meerly cocky |
| 07:56.25 | mog_home | little bit of both |
| 07:56.47 | mog_home | hey oej |
| 07:56.53 | shree | ok, i have a a system that makes automated calls |
| 07:56.53 | *** join/#asterisk Zeeek_ (n=icechat5@62-240-244-9.adsl.claranet.fr) |
| 07:56.59 | oej | Hey, good morning mog_home |
| 07:57.06 | mog_home | is there an easy system for automating the parsing of config files |
| 07:57.19 | mog_home | like if i have [user] with info below |
| 07:57.22 | oej | mog_home: It's called asterisk |
| 07:57.22 | shree | its simple, we upload a file in a particualal area |
| 07:57.27 | mog_home | and i just need to make nodes of nodes |
| 07:57.36 | mog_home | i see how ast_variable_retrieve works |
| 07:57.45 | mog_home | just dont know how to grab all my labels |
| 07:57.47 | mog_home | oh wait |
| 07:57.50 | shree | it parses the file and makes a call and plays a message on connect |
| 07:57.55 | mog_home | pbx_dundi's way makes sence |
| 07:58.00 | mog_home | ill just do it that way |
| 07:58.12 | oej | The asterisk expert answers himself quickly ;-) |
| 07:58.34 | mog_home | who says the code isnt self documenting.... |
| 07:59.02 | shree | anyone who can help on autodialer for a fee or suggest someone who provides support for a fee |
| 07:59.20 | mog_home | as we can do it |
| 07:59.26 | mog_home | ^_^ |
| 07:59.27 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 07:59.29 | mog_home | and i need the money... |
| 08:00.16 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 08:00.58 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 08:00.58 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 08:01.03 | *** join/#asterisk SwK[Work1 (n=SwK@border0hsv.asterisksgi.com) |
| 08:01.09 | Akelavlk | Does anybody know, how to make 3-way calling? |
| 08:01.15 | Netgeeks | Twisted! |
| 08:01.16 | Assid | heya mog |
| 08:01.23 | mog_home | Hi |
| 08:01.29 | Assid | does GSM in VBR mode help? |
| 08:01.42 | mog_home | couldnt tell you firmly, but i assume so |
| 08:01.48 | mog_home | no wait |
| 08:01.49 | mog_home | opposite |
| 08:01.55 | mog_home | no wait |
| 08:02.00 | mog_home | yes it should help |
| 08:02.01 | Assid | confusing isnt it |
| 08:02.07 | mog_home | man its 3 in the morning |
| 08:02.23 | Assid | okay.. how do you enable VBR for *<->* ? |
| 08:02.26 | Assid | redbull? |
| 08:02.35 | mog_home | liquid energy |
| 08:02.35 | Assid | hrmm i used to use it during my workouts |
| 08:03.22 | mog_home | heh its not quite like gaterade |
| 08:03.25 | Assid | i found coffee and coke to be much more energising |
| 08:03.27 | mog_home | more like liquid sleep |
| 08:03.33 | Agrajag- | gday. im a bit confused. if i have an fxo card, do i need an fxs card as well to be able to use an analogue phone? or can i use phones on the same line that's plugged into the fxo? |
| 08:03.59 | *** join/#asterisk weazul (n=weazul@82-169-62-42-mx.xdsl.tiscali.nl) |
| 08:04.00 | Assid | Agrajag-: fxo is for connecting to your pots |
| 08:04.17 | Assid | however.. if you want to connect to a phone.. then you would need fxs |
| 08:04.33 | Assid | one gives dialtone.. the other uses it |
| 08:04.51 | Agrajag- | yeah - but you can have multiple devices using the same dialtone right? |
| 08:04.57 | Agrajag- | like i can have 3 phones on the same line |
| 08:05.15 | Assid | your pstn exchange is FXS so you would need FXO.. but if you want to run it on your own pbx exchange.. you would need fxs |
| 08:05.30 | Assid | IF YOU WANT TO GIVE DIALTONE.. that is |
| 08:06.01 | *** join/#asterisk juice__ (n=juice@mo-67-77-178-112.dyn.sprint-hsd.net) |
| 08:06.27 | Agrajag- | ok. so i can't have the asterisk box with an fxo card, so it can 'listen' on the line, and do things like auditing - and have the phones on the same line, but not plugged into the box? |
| 08:06.50 | Assid | not sure i follow |
| 08:06.55 | Zeeek_ | Agrajag- you can, but there are a few problems with this |
| 08:06.58 | weazul | good morning all, is somebody awake who is familiar with asterisk and radius? |
| 08:07.05 | *** join/#asterisk _m_ (n=m@nat-ph3-1.rz.uni-karlsruhe.de) |
| 08:07.08 | Zeeek_ | if it's ONLY for auditing, no problems at all |
| 08:07.32 | Agrajag- | ok, so what are the problems? what would i lose out on? |
| 08:07.33 | Zeeek_ | if the phone si the be used as a phone, asterisk doesn't know about it |
| 08:07.56 | Agrajag- | it can't tell that i've picked up the phone somewhere else and the line is being used? |
| 08:08.14 | Zeeek_ | if someone is on the phone asterisk will pick it up like an obnoxious mother in law |
| 08:08.22 | Zeeek_ | No it doesn't know |
| 08:08.43 | Zeeek_ | asterisk doesn't detect dialtone in fact |
| 08:08.50 | *** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-48-89.dialin.kamp-dsl.de) |
| 08:08.59 | Agrajag- | ok. so i really need an fxo and a couple of fxs's to go with it |
| 08:09.14 | Zeeek_ | yes or SIP or IAX phones |
| 08:09.29 | Zeeek_ | or an ATA (analog to SIP/IAX) |
| 08:09.42 | Agrajag- | yeah. which i don't have - im looking for the least expensive option :) |
| 08:10.00 | Zeeek_ | the ATA is around $80 ? |
| 08:10.11 | Agrajag- | hmm ok |
| 08:10.14 | Zeeek_ | the phones (Grandstream) are less |
| 08:10.35 | Zeeek_ | I have used 3 BT101 for over a year, and they work fine for those who are ona budget |
| 08:10.35 | nick125_lappy | whats the format for sending asterlink calls over sip? |
| 08:10.41 | weazul | but grandstream phones sucks bigtime firmware is full of bugs |
| 08:10.59 | Zeeek_ | not the ones I use |
| 08:11.16 | Zeeek_ | but if you need a good phone, it costs |
| 08:11.19 | weazul | or you do not know it is a bug ;-) |
| 08:11.20 | Agrajag- | the other thing im not sure about is what devices can be used - at http://www.asterisk.org/hardware it lists "Generic X100P Intel IA92 WinModem compatible with X100P" - i dont suppose any modem can be used as an fxs though? |
| 08:11.32 | Zeeek_ | weazul I like a couple of those bugs |
| 08:12.00 | Zeeek_ | but I'm on a very old but stable firmware |
| 08:12.06 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 08:12.49 | Zeeek_ | Agrajag- there are multi channel ATA |
| 08:12.57 | opus_ | i got some of those generic FXO cards |
| 08:12.58 | opus_ | works great |
| 08:12.59 | Zeeek_ | you can plug 2-4 phones into some of them |
| 08:13.01 | opus_ | paid $3 on ebay |
| 08:13.11 | Agrajag- | yeah i've found cheap fxo cards |
| 08:13.31 | Agrajag- | but then i need something (fxs) to plug the phone in to |
| 08:13.32 | opus_ | +dont suppose any modem can be used as an fxs though? |
| 08:13.38 | opus_ | Nope you need special card |
| 08:13.40 | mog_home | those are junky... |
| 08:13.42 | Agrajag- | ok |
| 08:13.55 | opus_ | FXS has to supply like 48 AC voltage |
| 08:14.33 | Assid | is there a decent ip phone which supports GSM/ilbc codec? |
| 08:14.43 | opus_ | no because GSM/ilbc sucks |
| 08:14.58 | Assid | hrmm.. well. not enough bandwtih to use ulaw |
| 08:15.14 | opus_ | g729 then? |
| 08:15.32 | Agrajag- | ok i guess my cheapest/best option would be a TDM400P with 1 fxs and 1 fxo |
| 08:15.49 | opus_ | even better: don't do any analog at all |
| 08:15.59 | Agrajag- | but i only have analog phones |
| 08:16.00 | Assid | nah.. i gotta test with tons of servers.. would cost quite a bit to buy that many licenses |
| 08:16.15 | *** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 08:16.15 | opus_ | sellem before they become obselete!!! |
| 08:16.20 | Agrajag- | hehe |
| 08:16.35 | opus_ | it will be like the analog cellphone one day |
| 08:16.50 | opus_ | APOIP. analog phone over IP |
| 08:16.57 | STUN | what will happen to ATAs? |
| 08:17.08 | Agrajag- | hmm well i think that day isn't going to be very soon |
| 08:17.20 | opus_ | STUN they will be used to hold down paper when the wind blows |
| 08:17.23 | Akelavlk | Does anybody know, how to make 3-way calling? |
| 08:17.26 | X-Rob | Assid - PA1668 phones and GXP-2000' sboth support ilbc and GSM |
| 08:17.35 | STUN | jejeje |
| 08:17.36 | nick125_lappy | anyone here use asterlink? if so, can you give me your sip line to make calls out? |
| 08:18.27 | Assid | X-Rob: how much are they? |
| 08:19.20 | *** join/#asterisk Moc_ (n=mochouin@207-199-0-72-ppp.3menatwork.com) |
| 08:20.27 | *** join/#asterisk optimator (n=kvirc@70-58-53-58.eugn.qwest.net) |
| 08:20.29 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 08:20.51 | nick125_lappy | exten => s,4,Dial(SIP/<asterlink account number>@asterlink_host41/${ARG1}) ; arg1 is the phone number (i.e 1800-321-5432) |
| 08:21.03 | nick125_lappy | ^^ whats wrong with that? |
| 08:21.15 | nick125_lappy | asterlink_host41 is the connection in my sip.conf config file |
| 08:23.13 | nick125_lappy | any ideas? |
| 08:23.42 | weazul | X-Rob: GXP-2000 is really a nice phone but still has verry buggy firmware!!! |
| 08:24.15 | Assid | i dont see GSM/ilbc in GXP-2000 |
| 08:24.35 | weazul | Assid: be prepared for that |
| 08:25.13 | weazul | GXP-2000 has supports GSM |
| 08:25.17 | Assid | weazul: my friend said hes gonna get me the SPA-841 .. but i dont know/think hes gonan really get it |
| 08:25.22 | Assid | Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model 102D). Dynamic negotiation of codec and voice payload length |
| 08:25.26 | Assid | dont see GSM |
| 08:25.50 | weazul | you've the phone right now? |
| 08:25.54 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 08:26.10 | Assid | nah.. seeing it on voipsupply |
| 08:26.11 | weazul | it's located on the account 1 tab . on the bottom |
| 08:26.23 | *** join/#asterisk wuwu (n=wolfgang@81.223.6.242) |
| 08:26.35 | weazul | where you can select "Preferred Vocoder:" |
| 08:26.42 | wuwu | hi all, i have a question regarding call groups - specially sip callgroups |
| 08:26.49 | Zeeek_ | even thge BT101 had gsm now |
| 08:26.58 | weazul | GSM is one of the codecs |
| 08:27.00 | Zeeek_ | all the chinese IAX/SIP phones have it |
| 08:27.06 | wuwu | withing ZAP it is possible to dial a complete group with ZAP/g1 - is this also possible within SIP channels ? |
| 08:27.10 | X-Rob | Zeeek_ - that's the PA1688 chip |
| 08:31.46 | Zeeek_ | yup |
| 08:31.46 | X-Rob | it roxx0rz. |
| 08:31.46 | Assid | weazul: dont have it right now.. |
| 08:31.46 | Assid | hrmm.. maybe i should ask my cousin to get it from hongkong/china |
| 08:31.46 | Zeeek_ | they're about $60 |
| 08:31.46 | *** join/#asterisk meppl (n=mephisto@p54AAEEDE.dip.t-dialin.net) |
| 08:31.46 | Assid | Zeeek_: which ones? |
| 08:31.46 | X-Rob | Wups. No ILBC on GXP-2000, but definately does have GSM |
| 08:31.46 | Zeeek_ | the PA1688 phones |
| 08:31.46 | X-Rob | Dinner time. |
| 08:31.46 | X-Rob | Sorry for doubting you weazul 8) |
| 08:31.46 | oej | X-rob: Have you tested the latest firmware for gxp? Does it have subscriptions? |
| 08:31.46 | Assid | Zeeek_: are they good? |
| 08:31.47 | Zeeek_ | they are quirky but perfect for experimentation |
| 08:31.47 | Zeeek_ | I have three of them |
| 08:31.47 | Zeeek_ | all on IAX2 for the moment |
| 08:31.47 | Assid | well.. i am gonna use them more than just experiments |
| 08:31.47 | Zeeek_ | iLBC and gsm |
| 08:31.47 | X-Rob | oej - it definately does _not_ have subscriptions |
| 08:31.47 | Zeeek_ | nothing is good for $60! |
| 08:31.47 | X-Rob | it's actually significantly broken |
| 08:31.47 | oej | X-rob: Well, the marketing guys told us about the subscriptions at Von in Santa Clara this year... |
| 08:31.47 | X-Rob | oej, yeah. I've been emailing them sip traces for about 6 months now. |
| 08:31.53 | Assid | is the support supposed to say gsm ? or some kinda g.xxx ? |
| 08:32.24 | X-Rob | they're going to be implementing subscriptions the saem as the snoms (eg, dialog+xml) |
| 08:33.28 | X-Rob | I have to go cook dinner (sigh, pregnant wife has spoken), will be back later hopefully. |
| 08:34.37 | *** join/#asterisk fenlander (n=neils@82.152.81.57) |
| 08:35.40 | *** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
| 08:36.25 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
| 08:36.52 | weazul | does anyone has experiance with the new beta version of asterisk? is it stable enough or .... |
| 08:39.07 | *** part/#asterisk secure75 (n=mic@p549A0A51.dip0.t-ipconnect.de) |
| 08:39.38 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:39.40 | oej | weazul: Please test it. When we release it, we have considered it ready for production use, not before |
| 08:43.05 | nick125_lappy | im using the beta on a stable box, the beta itself is fine, its just my configs that are kinda...yeah.. |
| 08:43.36 | Ahrimanes | hehe |
| 08:43.54 | nick125_lappy | but someone here could help me fix that :) |
| 08:44.07 | nick125_lappy | <nick125_lappy> exten => s,4,Dial(SIP/<asterlink account number>@asterlink_host41/${ARG1}) ; arg1 is the phone number (i.e 1800-321-5432) |
| 08:44.12 | nick125_lappy | whats wrong with that dial line? |
| 08:44.13 | Assid | actually.. im thinking of cleaning up my samples conf's |
| 08:44.27 | Assid | like just taking the configs i want.. and removing the other contexts |
| 08:44.34 | Assid | can make the loading/parsing a whole lot easier |
| 08:45.11 | *** join/#asterisk ogun (n=ogun@h236n2fls34o865.telia.com) |
| 08:45.33 | nick125_lappy | any ideas while that dial line doesnt work? |
| 08:46.10 | nick125_lappy | it seems like its trying to dial account number@asterlink_host41/${ARG1} with asterlink_host41/${ARG1} as the host |
| 08:46.25 | shido6 | brb |
| 08:47.32 | *** join/#asterisk Beccara (n=Beccara@222-152-27-117.jetstream.xtra.co.nz) |
| 08:47.44 | syle | http://www.ebgames.com/ebx/categories/systems/xbox360/ |
| 08:48.29 | ScaredyCat | nick125_lappy: are you registered with that server? |
| 08:49.00 | nick125_lappy | yep |
| 08:49.13 | ScaredyCat | in that case just do: |
| 08:49.39 | nick125_lappy | Sep 6 01:25:17 WARNING[15325]: chan_sip.c:1863 create_addr: No such host: asterlink_host41/1<phonenumber> |
| 08:49.41 | ScaredyCat | s,4,Dial(SIP/${ARG1}@asterlink_host41) |
| 08:49.56 | *** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net) |
| 08:50.08 | nick125_lappy | ok |
| 08:51.00 | ScaredyCat | iax and sip have slightly different dial strings... you were trying what looks like an iax2 dialstring on sip |
| 08:51.26 | nick125_lappy | yeah i know |
| 08:51.35 | Zeeek_ | ? |
| 08:51.56 | *** join/#asterisk _meppl_ (n=mephisto@p54AAD75A.dip.t-dialin.net) |
| 08:51.58 | Zeeek_ | ScaredyCat longtime no C |
| 08:52.09 | *** join/#asterisk RoyK (n=roy@80.239.107.80) |
| 08:52.12 | ScaredyCat | :D |
| 08:52.18 | ScaredyCat | C++ now |
| 08:52.40 | Zeeek_ | yeah, I've had enough of you! :) |
| 08:52.49 | ScaredyCat | wuss |
| 08:52.52 | Zeeek_ | 'sup ? |
| 08:53.05 | ScaredyCat | busy... too busy :( |
| 08:53.13 | Zeeek_ | me too |
| 08:53.39 | RoyK | arg. i want astmm :( |
| 08:53.44 | Zeeek_ | having too much work is like discovering girls. Nothing else gets done |
| 08:53.58 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 08:54.14 | *** join/#asterisk simong (n=simong@h100n1fls34o884.telia.com) |
| 08:54.35 | nick125_lappy | hem.. |
| 08:54.38 | nick125_lappy | Sep 6 01:53:13 NOTICE[15625]: chan_sip.c:8985 handle_response: Failed to authenticate on INVITE to '"Host41 Communications" <sip:1800*******@**.**.**.***>;' |
| 08:54.46 | ScaredyCat | 18 hours days... I'm going to explode |
| 08:54.55 | nick125_lappy | i think that might be the problem... |
| 08:54.57 | *** join/#asterisk yaboo (n=jsirucka@220.245.131.131) |
| 08:55.02 | RoyK | Zeeek_: heh |
| 08:55.18 | ScaredyCat | nick125_lappy: you have a space |
| 08:55.41 | nick125_lappy | btw, this is 1.2 |
| 08:56.14 | ScaredyCat | you have [asterlink_host41] in your sip.conf / |
| 08:56.15 | ScaredyCat | ? |
| 08:56.22 | nick125_lappy | yup |
| 08:56.30 | ScaredyCat | and it has your auth details? |
| 08:56.37 | Zeeek_ | ScaredyCat you use asterlink? I have a quest. |
| 08:56.41 | nick125_lappy | yep |
| 08:56.45 | ScaredyCat | I don;t Zeeek_ |
| 08:56.55 | Zeeek_ | ok then I don't have a question. |
| 08:58.14 | *** join/#asterisk newlook_ (n=newlook@203-59-168-152.perm.iinet.net.au) |
| 08:58.21 | *** join/#asterisk loick (n=loick@81.255.80.161) |
| 08:58.22 | nick125_lappy | it shows the asterlink server in sip show registry |
| 08:58.27 | nick125_lappy | as registered |
| 08:59.20 | ScaredyCat | ok, so try replacing asterlink_host41 with it's FQDN or ip |
| 08:59.27 | ScaredyCat | in the dial |
| 08:59.39 | nick125_lappy | ok |
| 09:00.03 | nick125_lappy | how would i put the username in that too? |
| 09:00.10 | ScaredyCat | no need |
| 09:00.12 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:00.15 | nick125_lappy | ok |
| 09:00.41 | ScaredyCat | unless bkw_ has it config'd like that, but since ur reg'd we can try this first... |
| 09:00.57 | ScaredyCat | if it dowsn't work you may nedd to add fromuser= etc.. |
| 09:01.02 | ScaredyCat | but lets try this one fisrt |
| 09:01.08 | nick125_lappy | nope |
| 09:01.09 | nick125_lappy | same thing |
| 09:01.38 | ScaredyCat | ok, do you have a fromuser= in sip.conf entry for asterlink_host41 ? |
| 09:01.56 | nick125_lappy | ooo |
| 09:02.03 | nick125_lappy | thats what i forgot |
| 09:02.24 | nick125_lappy | but it seems to do the samething |
| 09:02.28 | nick125_lappy | with the fromuser |
| 09:02.32 | *** part/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 09:02.58 | ScaredyCat | did you change back the ip to asterlink_host41? |
| 09:03.01 | nick125_lappy | yes |
| 09:03.02 | ScaredyCat | in the dial |
| 09:03.08 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 09:03.09 | *** join/#asterisk Asylum (n=Asylum@dsl-58-6-126-60.qld.westnet.com.au) |
| 09:03.23 | ScaredyCat | did he give a fromdomain= too ? |
| 09:03.40 | nick125_lappy | i dont think so |
| 09:03.56 | nick125_lappy | when i talked to him, he pretty much said all i needed to add was the fromuser= IIRC |
| 09:04.24 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
| 09:04.35 | Mimmus | can someone explain me the Asterisk mechanism of call accounting? |
| 09:04.42 | Romik | any body can advice which unix tooll can convert mp3 to .g729 ? |
| 09:05.05 | Asylum | Just wondering if anyone can help, I have the TE110P card cable connected to the ISDN connection... but no such luck. there isn't even a LED light up on the back of the card... I'm suspecting it's either drivers.. I have tried updating the drivers via CVS but after I follow all instuctions and reboot centos boot but asterisk doesn't... |
| 09:05.07 | Ahrimanes | Romik: sox |
| 09:05.12 | Ahrimanes | ? |
| 09:05.23 | Ahrimanes | oh.. g.729.. will have a look |
| 09:05.32 | mog_home | dont think there is one |
| 09:05.35 | ScaredyCat | mmmm.... |
| 09:05.38 | mog_home | well legal one |
| 09:06.12 | ScaredyCat | Asylum: did you modprobe the card |
| 09:06.44 | ScaredyCat | you need to modprobe the driver each time you boot... |
| 09:06.48 | Asylum | Yes I did. Still didn't work... |
| 09:06.57 | Asylum | every single time? |
| 09:07.06 | ScaredyCat | yes |
| 09:07.08 | ScaredyCat | type: ztcfg -vv |
| 09:07.23 | ScaredyCat | that should error if the card isn't modprobed |
| 09:07.34 | Asylum | 0 channels configured |
| 09:07.43 | Asylum | So that means it is picking up the card? |
| 09:07.44 | ScaredyCat | nick125_lappy: can you wait for bkw_ to appear.. might be easier |
| 09:07.48 | ogun | Is there any way to check the number of agents logged on to a queue in the dialplan? |
| 09:07.54 | Asylum | Sorry for my ignorence, this is my first setup.. |
| 09:07.59 | Romik | Ahrimanes: how with to convert .mp3 to .g729 to save CPU cycles... |
| 09:07.59 | nick125_lappy | ScaredyCat, sure |
| 09:08.02 | ogun | So as to hinder the last agent to log out? |
| 09:08.03 | ScaredyCat | ogun show queue |
| 09:08.10 | Ahrimanes | Romik: sox.sourceforge.net mentions: PCM, U-law, A-law, G7xx ADPCM files |
| 09:08.42 | ogun | scaredycat: That I know, but I was looking to do it in the dialplan. Specifically in my "agent logout" extension |
| 09:08.42 | ScaredyCat | Asylum: when you boot, before starting * you need to modprobe the card. but you need to have the config files set up too |
| 09:08.48 | ScaredyCat | ahhh |
| 09:08.52 | nick125_lappy | hrm... |
| 09:08.55 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 09:08.56 | nick125_lappy | asterlink_host41 66.250.69.17 255.255.255.255 5060 Unmonitored |
| 09:09.00 | nick125_lappy | ^^ in sip show peers |
| 09:09.02 | Romik | Ahrimanes: there some web service, http://www.asteriskguru.com/audio_conversion.php but i need to convert huge quantity .....i want command line utility |
| 09:09.11 | Akelavlk | Hello, it's possible use two softphones from one IP? |
| 09:09.26 | Romik | Akelavlk: yes |
| 09:09.28 | Akelavlk | I run two DIAX from one PC and it's does not work.. |
| 09:09.33 | Asylum | Ok i'll look into it.. gives me some kind of idea |
| 09:09.43 | Akelavlk | Romik, how should I set up AIX clients? |
| 09:10.06 | Romik | Akelavlk: run different programs but from same IP |
| 09:10.55 | Akelavlk | I made a DIAX copy, and change configuration. but I can connect just from one. Second, show me error:the server could not be contacted. |
| 09:11.24 | ScaredyCat | ogun: you could use setgroup (pre 1.2 /not cvs head) and getgroup to determin that.. |
| 09:11.35 | Akelavlk | What is BEST AIX free softphone? |
| 09:12.40 | Zeeek_ | my personal favorite is iaxphone version 1 |
| 09:13.02 | Zeeek_ | but I don't use any of them extensively |
| 09:13.19 | ScaredyCat | did you change the PORT for one of the phones Akelavlk? |
| 09:13.23 | ogun | scaredycat: Thanks for the tip. Looks like just the thing for me. |
| 09:13.40 | nick125_lappy | hrm.. |
| 09:13.41 | nick125_lappy | SIP/2.0 407 Proxy Authentication Required |
| 09:14.10 | ScaredyCat | ogun: bear in mind that they screwed around with the setgroup etc in 1.2 and cvs head... |
| 09:14.15 | ScaredyCat | (god knows why) |
| 09:14.20 | Akelavlk | No, I didn't change a port.. |
| 09:14.30 | ScaredyCat | so if you upgrade you'll need to change stuff |
| 09:14.52 | Akelavlk | Should I use two ports, when I want use two softphones from one IP? |
| 09:14.57 | ScaredyCat | Akelavlk: well, 2 apps wont be able to open the same port... |
| 09:15.03 | ogun | scaredycat: Aye, noted. Probably part of the "change commands into Set()" big thingie |
| 09:15.04 | ScaredyCat | actually, changing it wont help either |
| 09:15.09 | ScaredyCat | not on the same pc |
| 09:15.12 | Kumbang | hi everyone, do you guys run audiocodes with * cvs-head |
| 09:15.24 | Ahrimanes | Romik: http://www.germanixsoft.de/index.php |
| 09:15.25 | ScaredyCat | ogun: ya... |
| 09:15.35 | Akelavlk | And how can I setup IAX conf to run on two ports? |
| 09:15.37 | ScaredyCat | regular breakage if you ask me ogun |
| 09:15.40 | nick125_lappy | in my asterlink_host41 section, type, secret, host, fromuser, and nat |
| 09:15.43 | Romik | Ahrimanes: this is unix command line? |
| 09:15.45 | nick125_lappy | thats right, right? |
| 09:15.53 | Zeeek_ | hi Ahrimanes |
| 09:16.17 | Ahrimanes | hey Zeeek_ :-) |
| 09:16.23 | ScaredyCat | username= |
| 09:16.26 | ScaredyCat | too |
| 09:16.27 | ScaredyCat | ? |
| 09:16.32 | Ahrimanes | Romik: oh sorry no.. windows application but capable of batch conversion |
| 09:17.06 | nick125_lappy | ScaredyCat, grr! i forgot to add that...now its added |
| 09:17.19 | ScaredyCat | but is it working :? |
| 09:17.38 | nick125_lappy | it says the circuit is busy |
| 09:17.50 | ScaredyCat | well that's a start :) |
| 09:19.45 | ScaredyCat | so it's now challenging your auth, but letting you in... |
| 09:19.53 | Mimmus | why I'm getting always "h" as destination in my CDRs? |
| 09:19.54 | jalsot | does anybody know where can I find IAX2 new RFC? |
| 09:20.12 | ScaredyCat | h = hangup extension |
| 09:20.20 | mog_home | hey is any dev still up? |
| 09:20.27 | ScaredyCat | there's an RFC for IAX2? |
| 09:20.37 | X-Rob | No |
| 09:20.41 | mog_home | there is a work in progress |
| 09:20.42 | Mimmus | ScaredyCat: ok, why? |
| 09:21.02 | jalsot | I have seen some new text files about IAX2 |
| 09:21.18 | jalsot | not just the classical iax.pdf from Mark.. |
| 09:21.22 | Mimmus | all outgoing calls are directed to 'h' ext? |
| 09:21.25 | jalsot | anybody know where to find? |
| 09:21.35 | nick125_lappy | xlite says '404 not found' |
| 09:21.55 | ScaredyCat | were you dialing a valid number? |
| 09:22.04 | nick125_lappy | 91<my number> |
| 09:22.30 | X-Rob | Well. |
| 09:22.31 | ScaredyCat | 91 being the country... no 00 first? |
| 09:22.39 | nick125_lappy | 9 is the prefix, 1 is the country |
| 09:23.47 | ScaredyCat | ok, but are you passing the 9 to asterlink too ? |
| 09:23.57 | nick125_lappy | the 9 is cutoff by the macro |
| 09:24.01 | ScaredyCat | ok, good |
| 09:24.05 | nick125_lappy | so it sends 1<number here> |
| 09:24.08 | nick125_lappy | to asterlink |
| 09:24.09 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 09:24.17 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 09:24.41 | nick125_lappy | i think it has to do with asterlink not authing the invite |
| 09:24.49 | ScaredyCat | since it's in the us do you need to dial the 1 ? |
| 09:25.03 | nick125_lappy | thats how i was told to dial it |
| 09:25.10 | ScaredyCat | ie, can't you just dial 212 5551212 |
| 09:25.13 | nick125_lappy | it worked before too, with my old iax2 setup |
| 09:25.14 | ScaredyCat | ahh ok |
| 09:26.22 | ScaredyCat | yeas, and I guess you'r swithcing to sip for quality... |
| 09:26.30 | nick125_lappy | yeah |
| 09:27.16 | Assid | quality? |
| 09:27.28 | X-Rob | *blink* |
| 09:27.31 | Assid | i didnt know there is any quality difference in sip/iax |
| 09:27.35 | X-Rob | There isn't. |
| 09:27.36 | ScaredyCat | there is |
| 09:27.39 | Assid | nope |
| 09:27.44 | nick125_lappy | yeah, there is |
| 09:27.46 | Assid | iax just has reduced headers. |
| 09:27.47 | ScaredyCat | try it. |
| 09:28.01 | X-Rob | You're insane. |
| 09:28.06 | Assid | means effectively it should be clearer since it has less bandwith requirements |
| 09:28.08 | ScaredyCat | setup a miliwat and connect via sip and aix |
| 09:28.16 | ScaredyCat | iax |
| 09:28.58 | nick125_lappy | well, if tehre wasnt issues, why would asterlink suggest everyone to move to sip? |
| 09:29.38 | ScaredyCat | bkw_'s already done some tests on it... |
| 09:29.42 | Assid | i hink they mean move to sip over traditional pstns |
| 09:29.51 | ScaredyCat | not sure how 'laboratory conditions' it was |
| 09:29.56 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:29.59 | *** join/#asterisk loick (n=loick@81.255.80.161) |
| 09:30.06 | X-Rob | if there _is_ a difference, it's a bug, and it'll get fixed. |
| 09:30.18 | ScaredyCat | eventually |
| 09:30.23 | Assid | yeah.. there cant be a difference |
| 09:30.26 | Assid | its all on the codec |
| 09:30.32 | Assid | thats where quality differs |
| 09:30.51 | Assid | so long as the bandwith remains the same |
| 09:30.56 | X-Rob | and ulaw is ulaw, no matter what layer 3 protocol you're using. |
| 09:31.12 | X-Rob | (or for !americans, alaw) |
| 09:31.27 | Assid | whats the difference between alaw and ulaw? |
| 09:31.38 | ScaredyCat | 1 byte |
| 09:31.48 | Zeeek_ | I took the change to mean asterlink themselves found issues in their IAX2 stuff |
| 09:31.51 | Assid | bah |
| 09:31.59 | oej | bkw_ found out that the IAX2 implementation in Asterisk doesn't scale very well when you have a large amount of traffic. SIP + RTP does. |
| 09:32.02 | ScaredyCat | they did Zeeek_ |
| 09:32.14 | *** join/#asterisk Hallski (n=micke@c-15c072d5.07-93-73746f29.cust.bredbandsbolaget.se) |
| 09:32.23 | Zeeek_ | oej thx for that clarification |
| 09:32.42 | Zeeek_ | because I've never had good luck with asterlink but all the others I use work fine |
| 09:32.47 | oej | SIP + RTP handle audio in a separate thread, if I understand bkw_ correctly |
| 09:33.03 | Assid | hrmm |
| 09:33.10 | Assid | so i gotta use sip now onwards? |
| 09:33.12 | Zeeek_ | the problem I have is that the lag between us varies a lot and they refuse calls if the lag is too high |
| 09:33.30 | nick125_lappy | hrmm...i get that invite error, and this: -- Got SIP response 488 "Not Acceptable Here (codec error)" back from 66.250.69.14 |
| 09:33.37 | Zeeek_ | I have no problem with voipjet, voicepulse IAX2 |
| 09:33.49 | ScaredyCat | roflmao |
| 09:33.57 | Assid | iax is supposed to be better than sip |
| 09:34.01 | oej | ...at least until we fix the scalability issue in chan_iax2 |
| 09:34.12 | oej | Better in what sense? |
| 09:34.18 | Assid | overheads |
| 09:34.24 | JamesDotCom | sip > iax |
| 09:34.27 | Assid | and effectively quality |
| 09:34.42 | JamesDotCom | the problem is |
| 09:34.47 | JamesDotCom | so few people ever understand sip |
| 09:34.51 | Zeeek_ | the IAX2 argument was put forth in earlier asterisk docs, yes |
| 09:34.51 | ScaredyCat | nick125_lappy: what codec are you trying to use? |
| 09:34.54 | nick125_lappy | gsm |
| 09:34.55 | JamesDotCom | or take a look at the rfc |
| 09:35.25 | ScaredyCat | did you sip debug and see the codec compatablilty |
| 09:35.26 | ScaredyCat | ? |
| 09:35.54 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 09:35.59 | nick125_lappy | where does it say the codec compatability? |
| 09:36.25 | ScaredyCat | it'll give you a list of your codecs and the far end, then say witch you both have |
| 09:36.34 | ScaredyCat | which |
| 09:36.39 | X-Rob | oej - wanna talk about 4877 if you've got a sec? |
| 09:36.53 | ScaredyCat | and then pick from those |
| 09:37.07 | oej | Which one is that, X-rob? |
| 09:37.10 | Assid | we really should have a general server where everyone can just login their asterisk boxes |
| 09:37.15 | Assid | for this channel |
| 09:37.18 | X-Rob | snom transfers crash asterisk |
| 09:37.25 | nick125_lappy | ok |
| 09:37.31 | Assid | and keep some kinda meetme rooms |
| 09:37.35 | ScaredyCat | snom, don't talk to me about snom... yukc |
| 09:37.43 | oej | X-rob: That's an evil issue report, with several different issues. Any news? |
| 09:37.51 | ScaredyCat | assid, try iaxtel |
| 09:38.04 | jalsot | is this the latest IAX2 RFC proposal? http://splurge.peoples-wireless.com/iax/iax.txt |
| 09:38.05 | X-Rob | Was hoping to ask for your assitance in debugging. Possibly in #asterisk-dev? |
| 09:38.24 | Assid | okay bottom line.. what would be better? iax/sip for around 10/15 simultanous calls? |
| 09:38.36 | Mimmus | does _. matches also 'h', 't', etc? |
| 09:38.38 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 09:38.50 | X-Rob | Mimmus - yes. That's why you shouldn't use it. |
| 09:38.57 | ScaredyCat | jalsot: is that Iax or iax2? |
| 09:39.10 | Mimmus | X-Rob: thanks, my dialplan is full of these! |
| 09:39.18 | *** join/#asterisk pr0m (n=pr0methe@2002:184b:c446:13:0:0:0:50) |
| 09:39.18 | jalsot | ScaredyCat: I guess, iax2 [April 26, 2005] |
| 09:39.31 | ScaredyCat | Expires: October 28, 2005 yea |
| 09:39.34 | ScaredyCat | :) |
| 09:39.35 | oej | X-rob: right |
| 09:39.39 | X-Rob | Mimmus - you want to use _X. |
| 09:40.03 | Mimmus | X-Rob: I will rewrite some contexts, thanks |
| 09:41.28 | Assid | ScaredyCat: is the 1700 number pstn accesible number? |
| 09:42.23 | ScaredyCat | via 3rd parties yes... |
| 09:42.35 | Mimmus | X-Rob: THANKS, your suggestion seems to solve my problem with all CDRs having 'h' has destination! |
| 09:42.39 | ScaredyCat | iaxtel dialin numbers (2 stage dialing) |
| 09:43.04 | ScaredyCat | youu just need to find one near you... |
| 09:44.23 | *** join/#asterisk SrFr (n=blaat@cust.12.229.adsl.cistron.nl) |
| 09:45.27 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
| 09:46.35 | Mimmus | thanks again to X-Rob. I'm leaving |
| 09:46.39 | fulgas | with a h extension shouldn't all hangups be forward to that extension ? |
| 09:46.53 | Assid | looks like i gotta shift my connectiosn from iax-> sip |
| 09:46.57 | X-Rob | Yay |
| 09:47.02 | X-Rob | I'm the fucking GEEK. |
| 09:47.09 | X-Rob | I fix L33T KERNEL PROBLEMS |
| 09:47.28 | X-Rob | ...if only I could fix SIP transfers... sigh. |
| 09:47.54 | Assid | rfc's |
| 09:47.56 | ScaredyCat | so, should I encrypt just the rtp or the whole protocol? |
| 09:48.19 | Assid | u wanna hav a encrypted transfer? |
| 09:48.27 | ScaredyCat | yes |
| 09:48.38 | Assid | cool |
| 09:48.48 | Assid | what swphone u gonna use? |
| 09:48.56 | ScaredyCat | my own |
| 09:48.58 | ScaredyCat | :) |
| 09:49.28 | Assid | is gnophone available for windows? |
| 09:49.42 | ScaredyCat | dunno... |
| 09:49.44 | SrFr | I have a strange problem: in extensions.conf, I read the variable $agi_callerid with a shell script to use it. It works perfect. the variable contains the originating telephone number. But when I pick up a ringing phone with another phone using *8# , the $agi_callerid contains MY sip number (the sip number of the phone I pick it up with). very strange |
| 09:51.42 | mog_home | gnite |
| 09:51.44 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 09:52.23 | h3x | diax phone does url pops |
| 09:52.24 | h3x | heh |
| 09:54.22 | weazul | has somebody tested Asterisk 1.2.0-beta1 already????? |
| 09:54.56 | weazul | anybody? |
| 09:55.02 | h3x | yes |
| 09:55.11 | h3x | just dont get it from cvs |
| 09:55.33 | weazul | ok and are there big improvements in your opinion? |
| 09:55.59 | h3x | from what |
| 09:56.19 | weazul | from latest 1.0x head... |
| 09:56.22 | Asylum | ScaredyCat: I've configured the zaptel drivers and i have 24 channels configured no erros.. Still can't dial in.. just get engaged signel any suggestions? |
| 09:56.35 | h3x | you mean v1-0 stable? |
| 09:56.37 | weazul | yes |
| 09:56.46 | h3x | well it has asterisk realtime |
| 09:56.49 | h3x | so if you need that |
| 09:56.59 | Asylum | there is now a flashing red light at the back of the card.. |
| 09:57.09 | weazul | hmkay nice nice... |
| 09:57.28 | h3x | i upgraded for the sql modules from the dialplan to work right |
| 09:58.47 | *** join/#asterisk transgress (n=transgre@71.14.20.160) |
| 09:58.58 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 10:01.14 | ScaredyCat | Asylum: did you also configure /etc/asterisk/zapata.conf ? |
| 10:02.23 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 10:02.44 | zobia | Hello every one. who can help me with the dtmf digits problem |
| 10:03.26 | *** join/#asterisk juice_ (n=juice@mo-67-77-189-118.dyn.sprint-hsd.net) |
| 10:03.42 | zobia | i dial from xlite to the asterisk server, when i input the digits to go to IVR system. my digits is gone |
| 10:03.56 | zobia | i mean the system did not receive any digits from my xlite. |
| 10:04.14 | zobia | i press the number for instance 1 then press # |
| 10:04.32 | zobia | my dtmfmode set to inband |
| 10:04.53 | zobia | thanx in advanced |
| 10:06.37 | zobia | Hello. |
| 10:09.00 | zobia | thank , i already solve the problem. |
| 10:09.18 | RoyK | dtmfmode? |
| 10:10.41 | Asylum | ScaredyCat: Yes i added switchtype channel and soo on |
| 10:10.55 | *** part/#asterisk SrFr (n=blaat@cust.12.229.adsl.cistron.nl) |
| 10:11.37 | *** join/#asterisk JessicaX^ (i=Jessie@86.112.145.198) |
| 10:11.45 | JessicaX^ | Hurray! |
| 10:11.50 | JessicaX^ | Hmm i <3 asterisk |
| 10:12.11 | JessicaX^ | Except some comedian made it keep saying people could lose weight when they phoned up.. lol |
| 10:12.26 | Asylum | ScaredyCat: also done /etc/zaptel.conf with loadzone span bchan dchan |
| 10:13.06 | ScaredyCat | did you restart * after doing that? |
| 10:13.21 | Asylum | nope. i'll do that now |
| 10:13.37 | ScaredyCat | take a look at zttool too.. it'll show the state of the card |
| 10:13.58 | Asylum | unloading zaptel hardware drivers failed |
| 10:14.15 | Asylum | error while rebooting.. ok i'll try that once it's rebooted |
| 10:16.52 | *** join/#asterisk Bonzai090 (n=pirch@wbs-146-190-83.telkomadsl.co.za) |
| 10:18.00 | Asylum | ok rebooted.. |
| 10:18.10 | Asylum | zttool current alarms "Red Alarm" |
| 10:19.46 | YoYo | *YAWN* |
| 10:20.55 | X-Rob | Asylum - so you have broken zaptel. Congrats. |
| 10:21.13 | YoYo | eh? what kind of broken zaptel? |
| 10:21.21 | *** part/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 10:21.22 | X-Rob | <Asylum> zttool current alarms "Red Alarm" |
| 10:21.24 | YoYo | red alarm could indicate that the line is broken |
| 10:21.36 | X-Rob | nono |
| 10:21.43 | X-Rob | All I'm saying is that zaptel, the concept, is borked. |
| 10:21.54 | X-Rob | there are many reasons for red alarm. |
| 10:21.58 | Asylum | So that means that there is no connection between the box and the wall? |
| 10:22.01 | YoYo | dun think so hoss |
| 10:22.18 | Asylum | Or could it be just the setup of the zaptel drivers? |
| 10:22.26 | X-Rob | Asylum - that'll be the problem. |
| 10:22.32 | Asylum | It's a brand new TE110P Digium card |
| 10:22.34 | X-Rob | (unless it was working before) |
| 10:22.37 | *** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl) |
| 10:22.47 | X-Rob | Woo |
| 10:22.53 | Asylum | Well, it's a new setup.. never done this before? so it could be my fault? |
| 10:22.55 | X-Rob | Asylum - I like you. |
| 10:22.58 | YoYo | Asylum, do you know for a fact that this line is good? |
| 10:23.01 | X-Rob | You're in qld _and_ you're on westnet. |
| 10:23.03 | Asylum | no i don't.. |
| 10:23.07 | Asylum | Yes X-Rob :P |
| 10:23.16 | X-Rob | Standard onramp? |
| 10:23.18 | YoYo | hrrm... that doesn't help things |
| 10:23.23 | Asylum | onramp 10 |
| 10:23.25 | X-Rob | ok |
| 10:23.39 | X-Rob | stick this in your /etc/zaptel.conf: |
| 10:23.43 | Asylum | mmk |
| 10:23.44 | X-Rob | span=1,1,0,ccs,hdb3,crc4 |
| 10:23.44 | X-Rob | bchan=1-10 |
| 10:23.44 | X-Rob | unused=11-15,17-31 |
| 10:23.44 | X-Rob | dchan=16 |
| 10:23.44 | X-Rob | loadzone = au |
| 10:23.44 | X-Rob | defaultzone = au |
| 10:23.45 | YoYo | iirc, the te cards need to be jumpered or T1/E1 operation... do you have it set correctly? |
| 10:23.46 | X-Rob | (sorry for the dump) |
| 10:24.04 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 10:24.05 | RoyK | ~pastebin? |
| 10:24.07 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 10:24.08 | X-Rob | yes yes |
| 10:24.10 | X-Rob | hush 8) |
| 10:24.26 | RoyK | :{ |
| 10:24.34 | X-Rob | Pr0n mustace! |
| 10:24.37 | X-Rob | mustache even |
| 10:24.41 | X-Rob | :{) |
| 10:24.46 | RoyK | lol |
| 10:24.57 | Baph | can anyone recommend a decent but inexpensive (read: cheap) PCI card with min 1xFXO 2xFXS max 2xFXO 4xFXS (or combination of cards)... so far, I'm looking at the TDM series (eg TDM22B)? |
| 10:25.52 | YoYo | Baph: if you need to keep things open for future expansion, get a T1 card and a channel bank |
| 10:26.02 | Asylum | lol |
| 10:26.04 | Asylum | well |
| 10:26.23 | Asylum | 11 channels configured |
| 10:26.27 | Asylum | still a red light |
| 10:26.32 | YoYo | red lights suck |
| 10:26.39 | X-Rob | How's your Onramp delivered? |
| 10:26.42 | Asylum | Do i have to modprobe it after changing config? reboot? |
| 10:26.53 | X-Rob | Asylum - uh. Yes. I think |
| 10:26.56 | X-Rob | rmmod te110xp |
| 10:26.58 | X-Rob | modprobe te110xp |
| 10:26.59 | Baph | YoYo, I'm looking at a SOHO setup, the max expansion I'll need will be probably 8xFXS 2xFXO... I'm still not even sure what connection (physical size) a T1 is lol |
| 10:27.16 | X-Rob | ..(I think that ztcfg can fix it, but, it's easier to just unload and reload) |
| 10:27.23 | Asylum | i've been using modprobe wcte11xp ? |
| 10:27.34 | X-Rob | yes |
| 10:27.38 | X-Rob | I just realised I had a brainfart |
| 10:27.42 | Asylum | lol |
| 10:27.53 | Asylum | mo use rmmod wcte11xp ? |
| 10:27.58 | X-Rob | yup |
| 10:28.11 | Asylum | device / resource busy |
| 10:28.15 | X-Rob | shut down asterisk |
| 10:28.24 | X-Rob | (or whatever's trying to use it) |
| 10:29.07 | *** join/#asterisk sigterm (i=sigterm@devious.info) |
| 10:29.12 | Asylum | reboot lol |
| 10:29.25 | X-Rob | no |
| 10:29.27 | X-Rob | don't reboot |
| 10:29.31 | Asylum | oh |
| 10:29.33 | Asylum | 2 late? |
| 10:29.45 | X-Rob | This is linux. You really _never_ have to reboot it |
| 10:29.51 | JessicaX^ | Haha |
| 10:29.52 | Asylum | lol |
| 10:30.04 | Asylum | So I should have just "shutdown.. asterisk.. then restart asterisk.. |
| 10:30.09 | JessicaX^ | I'm adding that to my "<3" list |
| 10:30.10 | X-Rob | you should have typed: |
| 10:30.13 | Asylum | instead of the os.. |
| 10:30.13 | X-Rob | 'asterisk -r' |
| 10:30.15 | X-Rob | 'stop now' |
| 10:30.21 | Asylum | Noice! |
| 10:30.25 | Asylum | i'll remember that. |
| 10:30.32 | X-Rob | JessicaX^ - your <3 list? |
| 10:30.35 | X-Rob | Less than three things? |
| 10:30.59 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 10:31.29 | X-Rob | Asylum - you'll need to. 'asterisk -r' will become your friend. |
| 10:31.35 | X-Rob | Where abouts in qld are you? |
| 10:31.44 | Asylum | Brisbane.. you? |
| 10:31.47 | X-Rob | Gladstone |
| 10:31.50 | Asylum | oh |
| 10:31.56 | Asylum | So you can just come here and help me lmao :P |
| 10:32.07 | X-Rob | $80/hour + expenses, no problem 8) |
| 10:32.26 | Asylum | I'm setting it up for a friend who has just opened a real estate.. and it's starting to annoy me that i can't get it working lol |
| 10:32.28 | Asylum | ok |
| 10:32.30 | Asylum | still a red light |
| 10:32.35 | X-Rob | JessicaX^ - you're meant to say 'Turn it 90 degrees!' |
| 10:32.55 | X-Rob | ..but it's not funny when I prompt you. |
| 10:32.57 | ScaredyCat | <PROTECTED> |
| 10:33.23 | X-Rob | ScaredyCat - I'm a geek. I can't catch for shit. |
| 10:33.36 | X-Rob | I have just discovered something INCREDIBLY ANNOYING. |
| 10:33.37 | ScaredyCat | neiher can ur team :P |
| 10:33.52 | ScaredyCat | bada boom boom |
| 10:34.06 | X-Rob | CentOS 4.1 does _NOT_ have a perl-Net-Telnet RPM! Argh. Here I've been saying how nice it is, and it DOESN'T HAVE NET::TELNET! |
| 10:34.09 | X-Rob | ScaredyCat - hah. |
| 10:34.12 | X-Rob | *phtttbt* |
| 10:34.20 | ScaredyCat | use cpan |
| 10:34.24 | X-Rob | yeah yeah |
| 10:34.35 | X-Rob | but I'm trying to shrink the AMP dependancies down |
| 10:34.43 | X-Rob | Asylum - Wups. Sorry. OK |
| 10:34.47 | ScaredyCat | and telnet ... *shudder* |
| 10:34.59 | Asylum | X-Rob: I stoped it then started it.. is says span configured for esf/b8zs ? is this wrong |
| 10:35.03 | JessicaX^ | :D |
| 10:35.18 | X-Rob | ScaredyCat - pray, how exactly are you meant to connect to the management interface w/o net::telnet? |
| 10:35.23 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:35.29 | ScaredyCat | you're not... |
| 10:35.34 | YoYo | Asylum, then why did you specify that in your config? |
| 10:35.40 | ScaredyCat | it should be ssh |
| 10:35.54 | ScaredyCat | it's a crime |
| 10:36.03 | Asylum | YoYo: Just been reading from online forums YoYo.. |
| 10:36.11 | ScaredyCat | and it should NEVER be open to the outside world |
| 10:36.12 | YoYo | eh? |
| 10:36.14 | JessicaX^ | It |
| 10:36.18 | JessicaX^ | It's a crime? |
| 10:36.20 | JessicaX^ | moider! |
| 10:36.25 | X-Rob | Asylum - that's for US stuff. |
| 10:36.32 | Asylum | Well, I've been trying to get it working from stuff i've read on the internet! |
| 10:36.33 | X-Rob | OnRamps use the configuration I gave you |
| 10:36.36 | YoYo | Asylum, esf/b8zs is US/T1 stuff |
| 10:36.44 | YoYo | you need AU config |
| 10:36.49 | X-Rob | Next question |
| 10:36.58 | X-Rob | on the TE110, did you change the jumper to 'E1'? |
| 10:36.59 | puzzled | morning all |
| 10:37.01 | YoYo | and I dunno what that might be... you guys are upside down there |
| 10:37.06 | JessicaX^ | Ooh, can i offer a big hug to an Asterisk Developer? ^_^ |
| 10:37.11 | X-Rob | YoYo - I've given him the config 8) |
| 10:37.20 | X-Rob | span=1,1,0,ccs,hdb3,crc4 |
| 10:37.25 | YoYo | Jessica: there's none around here... will I do? |
| 10:37.41 | Asylum | X-Rob i have changed that span=1,1,0,ccs,hdb3,crc4 in the /etc/zaptel.conf |
| 10:37.42 | JessicaX^ | Sure :) |
| 10:37.49 | X-Rob | Asylum - see your /msgs |
| 10:37.53 | JessicaX^ | YoYo :D |
| 10:37.53 | Asylum | but it still says te110p span configure for esf |
| 10:37.55 | X-Rob | note how I've messaged you |
| 10:38.27 | YoYo | Asylum, rmmod, the insmod/modprobe again |
| 10:38.27 | ScaredyCat | mmmm.... |
| 10:38.29 | YoYo | then ztcfg -vvv |
| 10:39.07 | ScaredyCat | 5 bacon butties, gone! |
| 10:39.08 | YoYo | also, make sure you channel assignments are correct (but don't ask me, I dun rmeember the funky layout of the e1) |
| 10:39.20 | YoYo | so, Jessica, what's got you in such a good mood? |
| 10:39.34 | JessicaX^ | Well, nothing much in particular |
| 10:39.38 | JessicaX^ | I'm usually like this |
| 10:39.58 | JessicaX^ | Oh, maybe it's this film - Unleashed - such a good film, it is :) |
| 10:40.02 | ScaredyCat | bchan=1-15,17-31 |
| 10:40.02 | ScaredyCat | dchan=16 |
| 10:40.06 | YoYo | lol |
| 10:40.28 | X-Rob | ScaredyCat - he's got an onramp 10, which is bchan=1-10 then unused=11-15,17-31 |
| 10:40.39 | *** join/#asterisk tandem1 (n=tandem@misp.misp.tuiasi.ro) |
| 10:40.45 | ScaredyCat | onramp a provider? |
| 10:40.56 | ScaredyCat | Dchan? |
| 10:41.05 | X-Rob | Nah. We only have one telco that provides PRI here |
| 10:41.10 | X-Rob | guess where the dchan goes 8) |
| 10:41.16 | X-Rob | and 'OnRamp' is the name of the PRI |
| 10:41.21 | X-Rob | uh |
| 10:41.24 | ScaredyCat | ahhh |
| 10:41.28 | X-Rob | We also have 'OnRamp 2' for a BRI |
| 10:41.32 | ScaredyCat | wtf |
| 10:41.36 | X-Rob | and then OnRamp 10/20/30 for PRI |
| 10:41.48 | ScaredyCat | why don;t they call it what it is |
| 10:41.51 | YoYo | ScaredyCat, fractional PRI |
| 10:41.52 | X-Rob | coz they're telstra |
| 10:41.58 | X-Rob | and they can do whatever the hell they want. |
| 10:42.03 | YoYo | service is called Onramp |
| 10:42.14 | ScaredyCat | sounds offroad |
| 10:42.34 | X-Rob | it was named when 'on ramp to the information superhighway' didn't sound lame. |
| 10:42.38 | ScaredyCat | too much Kangaroo stew methinks |
| 10:42.44 | X-Rob | in those 45 minutes. |
| 10:42.51 | ScaredyCat | 4.5 ;) |
| 10:43.09 | ScaredyCat | o, but your dchan is still 16 ? |
| 10:43.12 | ScaredyCat | ok |
| 10:43.14 | YoYo | information superhighway |
| 10:43.20 | X-Rob | ScaredyCat - yep. |
| 10:43.42 | Asylum | check your msgs |
| 10:43.48 | X-Rob | me? |
| 10:43.50 | Asylum | :P |
| 10:43.52 | Asylum | lol |
| 10:43.52 | X-Rob | I'm not getting any messages |
| 10:43.53 | ScaredyCat | you? |
| 10:43.56 | Asylum | your not? |
| 10:43.59 | X-Rob | Nope |
| 10:44.21 | Asylum | did you get that one? |
| 10:44.24 | X-Rob | nope |
| 10:44.26 | YoYo | well, he's getting msg's from me |
| 10:44.26 | ScaredyCat | stop sending me messages |
| 10:44.29 | ScaredyCat | :P |
| 10:44.31 | X-Rob | hehehe |
| 10:44.32 | Asylum | Private messages from unregistered users are currently blocked due to spam problems, |
| 10:44.36 | X-Rob | Aaaaah |
| 10:44.41 | X-Rob | that makes sence. |
| 10:44.42 | ScaredyCat | register |
| 10:44.44 | YoYo | haha |
| 10:44.56 | ScaredyCat | /msg nickserv register pick-a-password |
| 10:45.08 | ScaredyCat | then |
| 10:45.15 | YoYo | /msg nickserv register this-password-sucks |
| 10:45.22 | ScaredyCat | /msg nickserv identify the-password-you-picked |
| 10:45.39 | Asylum | X-Rob ? working now? |
| 10:45.41 | Asylum | cheers gjys |
| 10:45.42 | X-Rob | yp |
| 10:45.42 | Asylum | guys even |
| 10:45.44 | YoYo | /msg nickserv identify your-sucky-password |
| 10:46.30 | ScaredyCat | you can now use the Asylum nick every time you come back, just remember to identify when you join the server |
| 10:46.40 | Asylum | :) |
| 10:47.17 | YoYo | and if someone else steals your nick, you can recover it |
| 10:47.28 | ScaredyCat | yes... |
| 10:47.35 | Zeeek_ | oh? |
| 10:47.36 | X-Rob | That was me! |
| 10:47.38 | X-Rob | D'oh |
| 10:47.43 | ScaredyCat | no it was you YoYo :P |
| 10:48.08 | YoYo | that's ok... 'cause I'm +q on #FreeBSD for some fucked off reason |
| 10:48.09 | ScaredyCat | ghost em |
| 10:48.18 | YoYo | but only on this nick |
| 10:48.22 | YoYo | I think I needs a new one |
| 10:48.36 | ScaredyCat | ^o^o |
| 10:49.08 | ScaredyCat | right... |
| 10:49.18 | YoYo | oh cool! someone lifted the +q :) |
| 10:49.27 | ScaredyCat | bkw_: when you appear I have an idea |
| 10:49.39 | YoYo | omg, another idea? |
| 10:49.41 | YoYo | what is it this time? |
| 10:49.58 | sigterm | heh you piss off jer lately? |
| 10:50.13 | *** join/#asterisk Spacebar_ (n=stingray@stingr.net) |
| 10:50.14 | sigterm | he's been a madman with that this past week |
| 10:50.22 | YoYo | who what? |
| 10:50.29 | ScaredyCat | you only have to speak to piss him off |
| 10:50.34 | sigterm | jer.. #freebsd. |
| 10:50.38 | YoYo | oh |
| 10:50.46 | YoYo | no... it was some bitch... |
| 10:50.47 | sigterm | ScaredyCat: lately, yea |
| 10:50.54 | sigterm | quite testy |
| 10:51.07 | YoYo | someone made a comment about how cool "make buildworld" was... then I asked if he's ever tried Gentoo |
| 10:51.11 | YoYo | then BAM! +q |
| 10:51.16 | ScaredyCat | lol |
| 10:51.24 | sigterm | ahahah |
| 10:51.29 | YoYo | like, I was there to badmouth FreeBSD and promote linux |
| 10:51.42 | ScaredyCat | :P |
| 10:52.17 | YoYo | but, I use linux because nobody will get serious about getting Axtrix fully functional on FBSD |
| 10:52.24 | *** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 10:52.32 | Zeeek_ | you could |
| 10:52.48 | YoYo | I do not have the technical knowledge to do so |
| 10:54.41 | YoYo | but, I'm reasonably happy with my Gentoo box |
| 10:55.06 | Zeeek_ | the way to go then is to stimulate interest on #freebsd and lists - get a few competant people interested |
| 10:55.08 | YoYo | Linux wilson 2.6.10-gentoo-r6 #3 Thu Mar 17 17:32:24 EST 2005 i686 Intel(R) Pentium(R) 4 CPU 2.40GHz GenuineIntel GNU/Linux |
| 10:55.16 | YoYo | <PROTECTED> |
| 10:55.22 | Zeeek_ | it would be good forasterisk, too |
| 10:55.32 | YoYo | why dislike gentoo? |
| 10:55.33 | ScaredyCat | booo1!! |
| 10:55.39 | ScaredyCat | shit kept breaking |
| 10:55.45 | YoYo | Zeeek, there are lots of FBSD users who are interested in Asterisk |
| 10:55.59 | Zeeek_ | can't any of them make improvements then? |
| 10:56.13 | YoYo | but for some reason, we've not gotten the attention of anyone capable of getting zaptel fully ported |
| 10:56.16 | ScaredyCat | you could ask jerjer YoYo |
| 10:56.18 | ScaredyCat | ;) |
| 10:56.24 | Zeeek_ | I tried * on FreeBSD 4.7, still there in fact (on a rented server, no ZAP) |
| 10:56.50 | Zeeek_ | it seemd to work ok other than ZAP and moh |
| 10:57.00 | ScaredyCat | moo |
| 10:58.16 | JessicaX^ | Hey hey! |
| 10:58.22 | YoYo | yeah, I had it working on 5.something, even with zap (x100), but what I need, is the tor2 and wct1xxp drivers |
| 10:58.22 | JessicaX^ | turn those frowns around :) |
| 10:58.36 | X-Rob | :-( |
| 10:58.39 | Zeeek_ | how were the cookies? |
| 10:58.39 | X-Rob | )-: |
| 10:59.00 | JessicaX^ | :D |
| 10:59.01 | JessicaX^ | They roxed |
| 10:59.02 | JessicaX^ | =) |
| 11:02.05 | X-Rob | Did you give her _special_ cookies? |
| 11:02.29 | *** join/#asterisk Mother_ (n=Mother@53.Red-217-126-93.pooles.rima-tde.net) |
| 11:03.50 | *** join/#asterisk casterman (n=casterma@83.214.16.191) |
| 11:04.50 | *** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg) |
| 11:05.41 | *** join/#asterisk jannnn (i=foobar@outpost.zedz.net) |
| 11:05.49 | jannnn | hi |
| 11:06.04 | *** part/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 11:06.06 | casterman | hi |
| 11:06.38 | *** join/#asterisk Tili (i=Tili@202-133-65-212-dialup.sat.net.pk) |
| 11:06.41 | JessicaX^ | Yes, they were disco biscuits |
| 11:07.27 | ScaredyCat | with lights? |
| 11:08.10 | ScaredyCat | right, that's it - I'm off to break iax2 |
| 11:08.11 | JessicaX^ | Nope ;-; |
| 11:08.35 | jannnn | I did setup an asterisk server a few month ago. connections between isdn and sip (snom,sipura adapter, grandstream adapter)and betwen sip-sip (grandstream and sipura) works. if I try to call the sipura from snom or |
| 11:09.06 | jannnn | greandstream the sipura is ringing but after the connect you can hear nothinh |
| 11:09.16 | jannnn | nothing.any hints where to search ? |
| 11:09.24 | Zeeek_ | nat problems? |
| 11:09.58 | Zeeek_ | jannnn descripe the network where asterisk is and the one where the client (phone) is |
| 11:10.25 | jannnn | I disable nat in the sip.conf. all phones are connectet via vpn (ipsec) |
| 11:10.46 | Zeeek_ | that doesn't answer my quest |
| 11:10.51 | ScaredyCat | ion |
| 11:10.59 | Zeeek_ | hi ian |
| 11:12.39 | Tili | hi ScaredyCat. |
| 11:13.09 | ScaredyCat | lo Tili |
| 11:13.13 | ScaredyCat | how's you? |
| 11:13.25 | Tili | ScaredyCat: got a few minutes? |
| 11:13.31 | Tili | i am good. how are you? |
| 11:13.35 | jannnn | asterisk with connection to pbx is behind sdsl 2.3 mbit phone is behind adsl 2mbit. the two networks are connectet via ipsec, they can connect via privat class with the server. the two phone (grandstream and snom) are on the same subnet, the sipura is on another. there is no "connection" from both client networks to each other, only to the server. |
| 11:13.38 | ScaredyCat | this iax stuff is fun :) |
| 11:13.42 | ScaredyCat | good here too |
| 11:15.11 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:18.15 | RoyK | hmmmmmmmmm |
| 11:18.20 | Tili | ScaredyCat: what iax stuff? |
| 11:18.23 | Tili | hi RoyK |
| 11:19.19 | RoyK | using asterisk behind a welltech sip proxy, i only get audio from the calling client. client->welltech proxy->asterisk SIP call starts, but asterisk never sends any RTP traffic back to the client |
| 11:19.30 | jannnn | Zeeek_: do I need a "direct" connection between the sip clients ? I thought asterisk handels the connection so I need only the connection to asterisk ? |
| 11:19.44 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 11:20.40 | RoyK | jannnn: asterisk will bridge calls between sip clients, yes, asterisk cannot do native transfer/proxying on SIP |
| 11:20.43 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 11:20.58 | Bonzai090 | hey guys my callerid for incomming calles show up as unknown how can i get the phone number from my pstn ?. |
| 11:21.17 | Zeeek_ | subscribe to the paid service? |
| 11:21.35 | jannnn | Roky: so I need to enable sip proxy at the clients ? |
| 11:22.18 | Tili | jannnn: RTP packets are sent between 2 SIP clients. so they just need a way to do that directly. |
| 11:23.07 | jannnn | thanks, I think I'll find now a solution |
| 11:23.23 | Bonzai090 | any ideas any one i do have callerid enabled on my phone lines |
| 11:23.28 | ScaredyCat | muwhahahahahahaahhahahahahahahahaahahahahahhaha |
| 11:23.39 | ScaredyCat | fingerprint call authentication! |
| 11:24.17 | Tili | jannnn: may be you can use an RTP proxy. |
| 11:24.21 | darkskiez | ScaredyCat: what brought that on ? |
| 11:24.40 | ScaredyCat | I found my fingerprint reader |
| 11:24.51 | Zeeek_ | Bonzai090 sometimes people have theirs turned off. You called yourself to see if it works? |
| 11:24.53 | darkskiez | build it onto the dial button on the phone |
| 11:25.10 | ScaredyCat | roflmao |
| 11:25.27 | Zeeek_ | he gets this way from time to time... |
| 11:25.39 | Zeeek_ | from the peroid he spent in the jungle with fever |
| 11:25.42 | ScaredyCat | fingers.conf :: deadfingerdetect=yes |
| 11:26.39 | Bonzai090 | Zeeek_ caller id internal works i see the extentions but if some one from outside calls me it does not show up. my mobile phones use to show up on my old pbx but it shows unknown now |
| 11:26.45 | Bonzai090 | on my asterisk |
| 11:27.31 | Bonzai090 | all i really want is the callerid to show up on the switchboard |
| 11:29.20 | darkskiez | what country u in |
| 11:29.27 | Hallski | anyone knows what's the minimum set of config files you need, opening /etc/asterisk is pretty daunting for a new user and it would be nice to be able to remove what's not needed and add things as we go along |
| 11:30.11 | Bonzai090 | darkskiez in south africa but i know we use the german exchange system here. |
| 11:31.57 | *** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 11:32.34 | DrukenHME | i thought voip was illegeal in south africa ? |
| 11:33.28 | memic | i always get -- Extension 's' in context 'pbx-trunk' from '597609' does not exist. Rejecting call on channel 0/2, span 1 |
| 11:33.35 | memic | when i take up the phone |
| 11:33.42 | Tili | Hallski: read the manual for asterisk. Decide then what you need and what not. |
| 11:33.43 | memic | connectet via hfc in nt mode |
| 11:33.56 | Zeeek_ | maybe there's no 's' in pbx-trunk ? |
| 11:33.57 | memic | what means this Extension s |
| 11:34.02 | Hallski | Tili: yes but is there a list of "these must exist for it to start"? |
| 11:34.02 | memic | hm |
| 11:34.05 | memic | maybe read man |
| 11:34.06 | Bonzai090 | DrukenHME nope not any more its legal finaly |
| 11:34.16 | Hallski | Tili: or is the answer no one except what you decide you need? |
| 11:35.00 | memic | :) |
| 11:36.19 | Tili | Hallski: it really depends on what you want to do. you need extensions logger modules asterisk sip iax musiconhold voicemail zapata. Asterisk will complain for files it doesn't find. so you can add them if you are using that particular module/feature of asterisk. |
| 11:36.32 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
| 11:36.50 | Hallski | Tili: ah ok thanks, I think my problem was that the init.d-script on ubuntu didn't seem to work properly |
| 11:36.53 | Hallski | till try it out, thanks |
| 11:36.58 | memic | s = start |
| 11:36.59 | memic | well |
| 11:37.20 | memic | sounds good |
| 11:37.22 | *** join/#asterisk Morex (n=blah@host81-157-123-58.range81-157.btcentralplus.com) |
| 11:37.25 | Morex | Hello all |
| 11:37.33 | Morex | Question for you. |
| 11:37.43 | Morex | I'm going to need to take 10,000 simultaneous calls. |
| 11:37.50 | Morex | All to a single number. |
| 11:38.04 | Morex | So I'm going to need to load balance the calls between several * servers |
| 11:38.09 | Morex | Anybody know how to do that? |
| 11:38.32 | Bonzai090 | finaly i can chuck out a old panasonic pbx :> |
| 11:38.53 | Bonzai090 | any one want a free panasonic analogue pbx?.. lol |
| 11:39.12 | Bonzai090 | i'll even chuck in some phones |
| 11:39.17 | ScaredyCat | :) |
| 11:39.23 | ScaredyCat | ebay! |
| 11:39.47 | Bonzai090 | lol ScaredyCat i wont wish this POS on my worste enemy |
| 11:39.56 | ScaredyCat | but Tili wants it |
| 11:40.03 | ScaredyCat | lol |
| 11:40.04 | X-Rob | Morex - I suggest you spend some money at digium. |
| 11:40.07 | X-Rob | tell them that |
| 11:40.16 | X-Rob | and ask 'em how much it's going to cost for them to spec it out. |
| 11:40.17 | Bonzai090 | ScaredyCat hmm as tempting as it is |
| 11:40.18 | Tili | I can take any crap as long as it has anything to do with telephony |
| 11:40.23 | ScaredyCat | *cough* sangoma *cough* |
| 11:40.36 | Morex | x-rob: It's all VOIP (IAX or SIP) |
| 11:40.52 | Morex | For this one anyway |
| 11:41.05 | ScaredyCat | what does it terminate to? |
| 11:41.23 | Morex | We're forwarding 100 of the calls to another number |
| 11:41.29 | ScaredyCat | for just SIP calls use SER |
| 11:41.43 | ScaredyCat | that'll handle bucket loads with a tiny machine |
| 11:42.20 | Morex | ScaredyCat: Where do I get it? |
| 11:42.28 | ScaredyCat | 1 sec |
| 11:42.45 | ScaredyCat | http://www.iptel.org/ser/download/ |
| 11:42.57 | *** join/#asterisk Lars (n=Lars@cpe.atm2-0-7138.0x50a6f736.odnxx10.customer.tele.dk) |
| 11:43.01 | ScaredyCat | brb |
| 11:43.14 | Morex | ScaredyCat: THANK YOU!!! |
| 11:43.16 | DrukenHME | bonez39: sure, but free means i don't pay shipping either :) |
| 11:43.36 | DrukenHME | Bonzai090: that was ment for you :) |
| 11:43.58 | Bonzai090 | DrukenHME naa shipping is your baby lol |
| 11:44.13 | Bonzai090 | any one got a idea on caller id on my incomming lines?.. |
| 11:44.22 | Bonzai090 | where can i look for a solution on that |
| 11:44.29 | DrukenHME | hey now, free is free... don't be chincing out like publishers clearing house |
| 11:44.56 | Bonzai090 | want to be able to "black list" people i dont want to speak to can only do that if there caller id comes up |
| 11:45.21 | *** join/#asterisk asr__ (i=asr@pimpbox.latency.net) |
| 11:46.37 | Zeeek_ | Bonzai090 you should search the mailing list for people in your country or with your hardware |
| 11:46.46 | Zeeek_ | if not in the US |
| 11:47.20 | Zeeek_ | internal CID will always work in a properly configured asterisk. Externalm depends on a lot of shit |
| 11:47.21 | *** join/#asterisk jaike (n=jaike@203.131.137.76) |
| 11:47.25 | Bonzai090 | Zeeek_ hmmm i'll give a guy a call that i know in the area was hoping to get a quick answer right now i need to go and sort out some 7000 series cisco routers. |
| 11:47.32 | Bonzai090 | thanks for the help guys |
| 11:47.43 | jaike | anyone know how to disable call waiting on polycoms? |
| 11:47.44 | Zeeek_ | k |
| 11:48.02 | oej | jaike: I would like to know that as well... Haven't found a way to do it. |
| 11:48.24 | ScaredyCat | np Morex |
| 11:48.29 | jaike | oej: yup...messes up queueing |
| 11:49.04 | Morex | OK next question |
| 11:49.07 | Morex | Let |
| 11:49.25 | jaike | the polycom is getting more calls cause its putting the other calls in the queue in its waiting line |
| 11:49.28 | Morex | Let's say we get 20,000 people calling our 10,000 call system |
| 11:50.00 | Morex | We want to answer the first 10,000 calls, and not accept the other 10,000 calls. |
| 11:50.28 | Morex | How do we do that? |
| 11:50.51 | DrukenHME | exactly what do you mean 10,000 call system ? |
| 11:51.05 | darkskiez | morex: if you only have 10000 lines, that'll be easy. |
| 11:51.17 | Morex | Drunken: A system of load balanced asterisk servers handling 10,000 simultaneous calls |
| 11:51.36 | Morex | Darkskiez: We have effectively unlimited lines, but limited servers |
| 11:51.58 | darkskiez | unlimited lines? |
| 11:52.07 | Morex | (or rather, our VOIP termination provider has a lot of lines, but they're pooled between all their customers) |
| 11:52.25 | darkskiez | how much bandwidth do you have to your voip provider? |
| 11:52.27 | Morex | ...And we don't want to take more than we can handle otherwise the call quality will suffer. |
| 11:52.31 | Morex | Coloed. |
| 11:52.39 | Tili | while dialing out, how can i transfer the calledparty on answer to another extension. We can only transfer both parties using G^context^exten^prio |
| 11:52.51 | Tili | I dont want user to send any DTMG |
| 11:53.14 | Tili | s/DTMG/DTMF |
| 11:53.23 | DrukenHME | Tili: go research .call files |
| 11:53.32 | darkskiez | Morex: sounds like a fun project. when you figure it out, post about it on the wiki |
| 11:54.20 | Tili | DrunkenHME: already using outgoing calls. |
| 11:54.27 | DrukenHME | Morex: i highly doubt it's possible... you would have to be able to track all the channels from all the servers... |
| 11:54.30 | puzzled | Morex: if you have the cash talk to the people at Signate |
| 11:55.16 | DrukenHME | well, what would the purpose of transfer on answer be ? |
| 11:55.51 | ScaredyCat | app_universal |
| 11:56.00 | ScaredyCat | share vars between servers |
| 11:56.01 | Tili | transfer the called party so that astcc can let calledparty dial a number. this way i want to bill outgoing call by astcc and then number dialer by calledparty also through astcc. |
| 11:56.07 | Morex | How about if we told each server to hang up the call immediately if it's dealing with too many? |
| 11:56.39 | Morex | But then my * servers would be dealing with the hang up |
| 11:56.49 | Morex | Maybe SER can be configured to do it. |
| 11:57.17 | DrukenHME | Tili: are you talking about a callingcard callback ? |
| 11:57.30 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 11:57.31 | Tili | DrunkenHME: yes |
| 11:57.49 | Bonzai090 | the Playback command can i add more than one file to play back in a row cause i have 3 files to play but there is a gap between them |
| 11:57.57 | DrukenHME | Tili: ok, so you use the CID information from the incoming call, to create the call file |
| 11:58.01 | Tili | DrunkenHME: something that bills both callback call and call made by called party. seems like i am stuck |
| 11:58.11 | ScaredyCat | Morex: take a look at http://www.automated.it/files/ start from (New - An implementation of shared registrations - 14th Feb 2005) and read upwards... SetUnivar etc ) |
| 11:58.28 | Zeeek_ | ScaredyCat you updataed something on the site?????? |
| 11:58.53 | Bonzai090 | ahh ScaredyCat that your site?.. |
| 11:59.00 | ScaredyCat | yes, |
| 11:59.03 | ScaredyCat | brb cat feeding |
| 11:59.06 | Zeeek_ | yes what? |
| 11:59.08 | Tili | DrunkenHME: not exactly. i take number from incoming call and then call them back |
| 11:59.22 | Bonzai090 | ScaredyCat thanks for the site man was a nice learning ground 4 me :> |
| 12:00.24 | *** join/#asterisk voipguy (n=voipguy@196.200.26.42) |
| 12:00.30 | *** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg) |
| 12:01.29 | ScaredyCat | np Bonzai090 |
| 12:01.54 | ScaredyCat | yes I updated the sie Zeeek_ and yes it's my site |
| 12:01.57 | ScaredyCat | so yes |
| 12:02.00 | ScaredyCat | to both |
| 12:02.34 | Zeeek_ | OMG! you *updated* it? |
| 12:02.41 | ScaredyCat | bleh |
| 12:02.42 | Morex | Scaredy: Thanks! |
| 12:02.56 | memic | hm my hfc card dont want to call out |
| 12:02.58 | Zeeek_ | last time I looked, it said something aboput there being FXO modules for the TDM400 someday |
| 12:03.27 | Zeeek_ | bravo, it's the first site I've always recommended |
| 12:03.44 | DrukenHME | i must say it's the first i've ever heard of it |
| 12:03.46 | memic | i can call in but nothing goes out |
| 12:10.56 | DrukenHME | ScaredyCat: i'm curious, do you get spam in the websites email ? |
| 12:11.16 | *** join/#asterisk littleball (n=littleba@cm157.epsilon173.maxonline.com.sg) |
| 12:11.28 | ScaredyCat | yes |
| 12:11.37 | ScaredyCat | lots |
| 12:11.43 | ScaredyCat | but I use filtering.. |
| 12:11.48 | ScaredyCat | agressive |
| 12:11.49 | DrukenHME | :) |
| 12:11.53 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 12:11.53 | DrukenHME | spamassassin |
| 12:12.14 | Mimmus | what's the exact meaning of ResetCDR? |
| 12:12.18 | JessicaX^ | CanIt is also good |
| 12:12.26 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 12:12.34 | ScaredyCat | on the server yes, but I also grab it from the server to a local server that applies even more agressive filtering |
| 12:13.28 | littleball | hi,sorry, trouble someone to explain to me one English sentence due to i cannot understand my friend's email(my English is bad). "The job is quite lobo." What does it mean? |
| 12:13.51 | DrukenHME | ScaredyCat: ahh... i have mine configured like hotmail, got the whole junkmail folder and mail gets deletted after 10 days |
| 12:14.03 | *** join/#asterisk visik7 (n=mierda@unaffiliated/visik7) |
| 12:14.04 | JessicaX^ | low paying |
| 12:14.55 | ScaredyCat | must be non-uk english... |
| 12:15.21 | littleball | ScaredyCat, i also think so. I cannot understand |
| 12:15.22 | littleball | :) |
| 12:15.29 | DrukenHME | since when do they speak "ENGLISH" in the uk ? :) |
| 12:16.20 | littleball | ScaredyCat, i can trigger call back already, thanks for your help ! |
| 12:16.39 | *** join/#asterisk RoyK (n=roy@213.160.242.93) |
| 12:17.13 | voipguy | hi |
| 12:17.13 | Tili | ScaredyCat: how SRI solves problem of load balancing |
| 12:17.14 | ScaredyCat | DrukenHME: since we sent you lot over there ;) |
| 12:17.29 | ScaredyCat | littleball: good glad it helped |
| 12:17.37 | voipguy | any body know of a good billing solution for asterisk? |
| 12:17.41 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 12:17.56 | ScaredyCat | Tili: it's not about load balancing, more about uptime for client side |
| 12:17.59 | littleball | it is nice and i am happy. i will notify you when i donate to your site :) |
| 12:18.00 | ScaredyCat | but |
| 12:18.17 | Faithful | Hey guys if I am running ilbc to my service provider is there any advantange in running ilbc locally on the phones too? |
| 12:18.18 | ScaredyCat | set/get univar can help |
| 12:18.33 | ScaredyCat | since you can tell other servers call counts etc.. |
| 12:19.03 | ScaredyCat | Faithful: sure, there's not codec translation |
| 12:19.04 | Tili | voipguy: me |
| 12:19.04 | Tili | haha |
| 12:19.10 | *** join/#asterisk xheliox (n=jeff@user-0c6se1v.cable.mindspring.com) |
| 12:19.31 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:19.40 | littleball | yes, it works. Now i can use callback to trigger the call between two SIP phones. I think it should not a problem to trigger to E1 line numbers. (I am getting the digium card) |
| 12:19.59 | ScaredyCat | yes, you only need to change the channel... |
| 12:20.07 | ScaredyCat | and it'll work with your E1 |
| 12:20.15 | ScaredyCat | s |
| 12:20.35 | JessicaX^ | And in the blue corner, weighing in at just under 100mb! It has been named as the best PBX ever, and is also known as the "southern Dandy." it's Asterisk! The Reigning Champion! |
| 12:20.35 | littleball | great. |
| 12:20.57 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 12:21.09 | voipguy | TIli: what do you recommend? |
| 12:21.10 | ScaredyCat | 100mb JessicaX^? less than that... |
| 12:21.18 | JessicaX^ | And, in the red corner, weighing in at over 2 GB, It has been named a rip-off of asterisk, and is also known as the "Drunken hobo", it's Microsoft Ampersand! |
| 12:21.40 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 12:21.42 | ScaredyCat | abaccus? |
| 12:21.49 | JessicaX^ | lol |
| 12:22.04 | DrukenHME | ampersand? |
| 12:22.16 | JessicaX^ | Yep, their rip off |
| 12:22.24 | Uther_P | I haven't heard of that! |
| 12:22.33 | Uther_P | oh well... they'll fuck it up anyway |
| 12:22.36 | JessicaX^ | lol |
| 12:22.44 | JessicaX^ | And it's a jab from Asterisk, it's all over! |
| 12:22.48 | JessicaX^ | Asterisk wins! |
| 12:22.49 | Uther_P | they probably stole a bunch of asterisk's source too |
| 12:23.04 | Uther_P | that would be microsoft's mo |
| 12:27.38 | JessicaX^ | Uther_P, this reminds me of Family guy |
| 12:27.39 | JessicaX^ | lol |
| 12:27.39 | visik7 | does zaphfc support all kind of hfc chipset or only hfc-s ? |
| 12:27.39 | DrukenHME | i dunno, just voip and windows seems wrong... |
| 12:27.39 | JessicaX^ | "Alright, so we attack the rice krispie place at dawn, assuming judd hersh delivers the goods!" |
| 12:27.39 | Uther_P | heh, family guy rocks.. I have the first 3 seasons on dvd and all the rest on my pvr |
| 12:27.39 | Uther_P | haha |
| 12:27.39 | JessicaX^ | I gained my love of Journey through Family guy |
| 12:27.39 | JessicaX^ | lol |
| 12:27.39 | Uther_P | "you saved my ass back there", "you saved my ass!"... ."here's to snap"... "to snap!" |
| 12:27.39 | JessicaX^ | lol |
| 12:27.39 | Uther_P | heh, hopefully not because of how they were singing it |
| 12:27.39 | JessicaX^ | "OH MY GOD! I love this song, and i LOVE IT when amateurs sing the lyrics! BUT I HATE BASEBALL CARDS!!!!" |
| 12:27.39 | JessicaX^ | the funniest bit EVER |
| 12:27.39 | JessicaX^ | was |
| 12:27.40 | JessicaX^ | "Nobody messes with adam we" |
| 12:27.40 | JessicaX^ | XD |
| 12:27.40 | Uther_P | (drops casket)... (old woman) "oh my god.... hey, that is journey" |
| 12:27.40 | JessicaX^ | lol |
| 12:27.40 | JessicaX^ | Quagmire is hillarious |
| 12:27.40 | JessicaX^ | "What the hell is CPR?!?" |
| 12:27.40 | JessicaX^ | ahahaha |
| 12:27.40 | Uther_P | you know mother... this might have passed for a perishable banana pudding.. but with out the ;nilla waffers, its just another one of your retched culiary abortions! |
| 12:27.40 | *** join/#asterisk osource (n=opensour@203.99.58.250) |
| 12:27.40 | JessicaX^ | NOW CLEAN IT UP! |
| 12:27.40 | JessicaX^ | :D |
| 12:27.40 | JessicaX^ | <3 Pheonix Nights too |
| 12:27.40 | Uther_P | heh, "will somebody get patches the hell out of here before he decides to bend a fresh biscuit on the conveyor belt |
| 12:27.40 | JessicaX^ | It's hilarious >_> |
| 12:27.41 | JessicaX^ | ahahaha yeah |
| 12:27.41 | Uther_P | haven't heard of it |
| 12:27.41 | JessicaX^ | Download it |
| 12:27.41 | Uther_P | okie dokie |
| 12:27.42 | JessicaX^ | Or i'll send it to you, it's so funny |
| 12:27.42 | JessicaX^ | "OOh! Look at them, she could breast feed a creche" |
| 12:27.47 | Uther_P | you got the bandwidth? |
| 12:27.50 | JessicaX^ | I'm usually not into that kinda thing, lol |
| 12:27.54 | JessicaX^ | YHep, i think so |
| 12:27.57 | JessicaX^ | Yep* |
| 12:27.57 | Uther_P | cool |
| 12:28.45 | JessicaX^ | It's so funny though |
| 12:28.45 | Tili | voipguy: what do u want. pre-paid or post paid. |
| 12:29.17 | *** join/#asterisk jeffik (n=Jeff@toronto-HSE-ppp3985354.sympatico.ca) |
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| 12:36.36 | *** mode/#asterisk [+ooo kram denon bkw_] by irc.freenode.net |
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| 12:36.37 | Zeeek_ | ²whoa |
| 12:36.38 | Zeeek_ | bloodyell |
| 12:36.38 | *** join/#asterisk docelm0 (n=docelm0@66.237.242.41.ptr.us.xo.net) |
| 12:36.38 | Tili | voipguy: you can use astcc for pre-paid/calling card billing, although nobody really likes it |
| 12:36.38 | JessicaX^ | holy |
| 12:36.38 | ScaredyCat | wtf! |
| 12:36.38 | JessicaX^ | it was those haxors :( |
| 12:36.38 | ScaredyCat | that filled my scrollback :( |
| 12:36.38 | Zeeek_ | painfull, that |
| 12:36.39 | Zeeek_ | a full scrollback in the middle of the night and you have to get up |
| 12:36.39 | Zeeek_ | and empty it |
| 12:36.39 | Mimmus | what's the exact meaning of ResetCDR? I'm reading "Causes the Call Data Record to be reset". Reset? |
| 12:36.39 | ScaredyCat | it means 'pretend the call started from now' |
| 12:37.11 | Mimmus | but I have it in a macron called 'hangupcall'! |
| 12:37.11 | memic | anybody knows howto forward calls between sipphones? |
| 12:37.12 | ScaredyCat | the question is why Mimmus |
| 12:37.30 | ScaredyCat | do they have transter buttons memic |
| 12:37.46 | ScaredyCat | transfer |
| 12:37.46 | memic | ok |
| 12:38.04 | Mimmus | ScaredyCat: I'm using AsteriskAtHome, I noticed ResetCDR in this macro |
| 12:38.04 | memic | any they need to have internal numbers? |
| 12:38.05 | *** join/#asterisk rbd (n=robbyd@24-181-194-064.dhcp.hckr.nc.charter.com) |
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| 12:41.43 | Mimmus | memic: as last resort, you can try with '#' button |
| 12:42.05 | *** part/#asterisk jeremi (n=netnameu@204.116.189.12) |
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| 12:43.08 | memic | ok |
| 12:43.10 | memic | will test |
| 12:43.43 | *** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-48-89.dialin.kamp-dsl.de) |
| 12:43.49 | littleball | "The job is quite lobo" means "You can get lots of personal time". It is Singapore English :) |
| 12:43.54 | netnameus | I'm using a polycom phone, and when I dial a number that requires an extension, and I dial that extension, nothing happens. It's like i dont even dial the extension |
| 12:44.22 | netnameus | i've tried multiple places that i know people's extensions, but none work, so i think it's either the phone, or my asterisk setup |
| 12:44.35 | netnameus | any ideas? |
| 12:45.53 | *** join/#asterisk newl (n=newlook@203-59-184-194.dyn.iinet.net.au) |
| 12:47.04 | Bonzai090 | hmm i need a voice telling a customer that his call has ben forwarded via hong kong at premium rates of $14.50 per minute |
| 12:47.14 | Bonzai090 | think that wil deter some anoying customers lol |
| 12:47.17 | Hallski | hmm, will you need some hardware to do conferences? |
| 12:49.04 | ScaredyCat | I presume nagios will let me send a heartbeat to it and alert if the heartbeat doesn;t arrive |
| 12:49.38 | visik7 | ScaredyCat are u trying to clustering * ? |
| 12:50.16 | ScaredyCat | working on it |
| 12:51.04 | littleball | ScaredyCat, what is your strategy of clustering asterisk? One which level? OS, or Database.... etc? |
| 12:51.25 | netnameus | <PROTECTED> |
| 12:51.32 | netnameus | <PROTECTED> |
| 12:51.41 | ScaredyCat | os, * and db's |
| 12:51.58 | ScaredyCat | try different dtmf setting s netnameus |
| 12:52.12 | Tili | bye ScaredyCat. your pbx monitor is cool. |
| 12:52.15 | ScaredyCat | sounds like * doesn;t hear your dtmf |
| 12:52.20 | ScaredyCat | l8r tili |
| 12:52.22 | ScaredyCat | pah! |
| 12:52.25 | ScaredyCat | too slow |
| 12:52.34 | netnameus | where can i change those ScaredyCat? |
| 12:52.44 | ScaredyCat | in sip.conf or on the polycom |
| 12:52.59 | ScaredyCat | it si a sip polycom, right? |
| 12:53.07 | netnameus | yes |
| 12:53.09 | ScaredyCat | k |
| 12:53.14 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 12:53.17 | ScaredyCat | in sip.conf or on the polycom itself then |
| 12:53.29 | littleball | I also plan to use cluster in the future. Is it possible to centralize all sip request to one big server and then distribute the real voice stream etc to satelite servers? |
| 12:53.34 | ScaredyCat | you need to find a match... have you looked at the wikki/ |
| 12:53.35 | ScaredyCat | ? |
| 12:53.49 | ScaredyCat | littleball: I'd use SER for that |
| 12:53.51 | crash3m | littleball: using ser |
| 12:54.14 | littleball | Sorry, what is "SER"? google it |
| 12:54.35 | ScaredyCat | http://www.iptel.org/ser/download/ |
| 12:54.41 | Bonzai090 | ScaredyCat u recon i can get asterisk on a 4gb SD card?... |
| 12:54.45 | netnameus | dtmfmode=inband what else can/should I try ScaredyCat? |
| 12:54.53 | ScaredyCat | yes Bonzai090 - easily |
| 12:55.11 | ScaredyCat | look at my CF edition... |
| 12:55.17 | ScaredyCat | it's tiny |
| 12:55.29 | Bonzai090 | ScaredyCat hmmm will you be here in a hour or so's time?.. |
| 12:55.33 | ScaredyCat | though I run it all from ram.. |
| 12:55.38 | ScaredyCat | prolly... |
| 12:55.48 | ScaredyCat | depends if I actually get soem work done ;) |
| 12:55.50 | Bonzai090 | ScaredyCat just wanna chat to you about that.. |
| 12:56.37 | Bonzai090 | heh ScaredyCat so like the old cisco 1600 routers i can boot off the SD card and keep a backup of the sd card so if a clients pbx dies i can just put in the replacement card and charge them for it hehe |
| 12:56.51 | ScaredyCat | yes |
| 12:57.11 | ScaredyCat | and you can even put their configs on a http server and pull them off at boot :) |
| 12:57.25 | Bonzai090 | hmm ScaredyCat now we talking hehe :> |
| 12:57.40 | ScaredyCat | you haven't looked at them have you... |
| 12:57.45 | Bonzai090 | besides memory is faster than hdd |
| 12:57.47 | ScaredyCat | you bad bad person ;) |
| 12:57.58 | ScaredyCat | yes... CF is pretty slow... |
| 12:57.58 | *** join/#asterisk ymorin (n=ymorin@savoirfairelinux.net) |
| 12:58.15 | ScaredyCat | there's a pxe boot version too... but only really good for loacl booting |
| 12:58.21 | Bonzai090 | even maybe a usb2.0 memory stick hmmm |
| 12:58.41 | *** join/#asterisk coppice (n=chatzill@60.203.17.210.dyn.pacific.net.hk) |
| 12:58.42 | Bonzai090 | usb2.0 is about 400mb odd i think |
| 12:58.44 | ScaredyCat | yes, but I don;t have a machine that will boot off a usb stick, so I can't test it |
| 12:58.50 | littleball | Considering Google support Jabber/XMPP by launching its GoogleTalk, any indications from this? |
| 12:58.59 | Bonzai090 | ScaredyCat all the new pc's seemto be able to do it |
| 12:59.08 | ScaredyCat | i don;t have a new pc... |
| 12:59.11 | ScaredyCat | they're all old |
| 12:59.17 | ScaredyCat | :( |
| 12:59.21 | Bonzai090 | ScaredyCat time to invest in one man :) |
| 12:59.32 | ScaredyCat | should be here at the end of the week :) |
| 12:59.39 | Bonzai090 | mind you i am running a 7 use roffice off a celeron 1.7 with 256mb memory and a 40gb hdd lol |
| 12:59.58 | ScaredyCat | 1.7 celeron - pure luxury |
| 13:00.28 | littleball | Hi, anyone has some comments on what the future could be considering the emergency of GoogleTalk? I am interested in this? |
| 13:00.39 | coppice | We got evicted from our 1.7 celeron box |
| 13:00.41 | Bonzai090 | hehe ScaredyCat and my workstation is a lowely amd fx series with 1gb memory hehe |
| 13:00.49 | ScaredyCat | :/ |
| 13:01.13 | littleball | s/emergency/appear/ |
| 13:01.19 | coppice | Bonzai090: only one core? :-) |
| 13:01.37 | ScaredyCat | I don't know what my new one is going to be, someone ordered if for me though I did say 'lots of ram and disk space' |
| 13:02.03 | ScaredyCat | it's a dell, but it's better than what I have here atm... :) |
| 13:02.08 | Bonzai090 | coppice for now.. i am waiting to see what amd is oging to do |
| 13:02.10 | coppice | you'll get a big case with lots of space for RAM and disk |
| 13:02.28 | Bonzai090 | ScaredyCat dell?... *shivers* |
| 13:02.31 | ScaredyCat | finaly be ble to run vmwared boxen at real speed |
| 13:02.31 | coppice | X2s are nice :-) |
| 13:02.52 | Bonzai090 | well dell in south africa is bad cause there is only one service centre in the whole country |
| 13:03.02 | ScaredyCat | :O |
| 13:03.05 | Bonzai090 | coppice now if only the x2 can come out with fx :> |
| 13:03.17 | ScaredyCat | well dell service is generally bad anyway |
| 13:03.39 | coppice | Dell for small users is bad everywhere. for large corporates they are great |
| 13:03.44 | ScaredyCat | the number of times engineers turn up with the wrong part |
| 13:03.54 | yaboo | ScaredyCat, told the dell rep who rang me why my survey on dell service was bad |
| 13:03.55 | ScaredyCat | and then have to wait for a new part |
| 13:04.08 | Bonzai090 | heh |
| 13:04.18 | Bonzai090 | well i done a quote on a dell SQAN for one of my clients. |
| 13:04.19 | ScaredyCat | yaboo: an the reply was... |
| 13:04.47 | ScaredyCat | the only good thing is you can get consistancy in the hardware.. |
| 13:05.12 | coppice | I had a stunking hot PSU for a Dell notebook. our support guy called for a replacement "if it works we can't replace it". He hung up and called again saying it was dead. I got a new one the next day, and nobody complained the swapped out one still worked |
| 13:05.14 | yaboo | ScaredyCat, told them there india call center was crap and the local people were good |
| 13:05.22 | ScaredyCat | :) |
| 13:05.26 | Bonzai090 | 3 weeks later a truck pull sup with my quote.. a 5TB san with snap shot and racks the works |
| 13:05.31 | yaboo | ScaredyCat, they were upset with that |
| 13:05.38 | Bonzai090 | luckely i had the proff i only done a quote |
| 13:05.59 | Bonzai090 | pitty wish i could keep the kit though 5tb of space will be just enough for my downloads hehe |
| 13:06.03 | ScaredyCat | yaboo: well, it's the same for uk banks... it's just stupid... |
| 13:06.14 | coppice | Bangalore - the land of sex and drugs and call centres |
| 13:06.18 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 13:06.23 | ScaredyCat | particularly as they can't make decisions, they have to stick to the script |
| 13:06.53 | Akelavlk | Hello, I have problem with SIP phones. I can call from SIP phone, but when I try call from AIX I got " No route to destination" error. |
| 13:06.58 | Akelavlk | Where is a problem? |
| 13:06.59 | ScaredyCat | apparently one guy in one of the call centres was flogging CC numbers |
| 13:07.35 | Bonzai090 | heh ScaredyCat he's prolly some one sprison b!tch by now lol |
| 13:07.42 | *** join/#asterisk dsl (n=dsl@Ottawa-HSE-ppp260570.sympatico.ca) |
| 13:07.45 | ScaredyCat | hehe |
| 13:08.00 | ScaredyCat | sounds like a networking problem Akelavlk |
| 13:08.20 | ScaredyCat | right.. I must add this pricing to iax...bbl |
| 13:08.29 | Bonzai090 | ok lemme go get some work done.. not that i am in the mood to do some work today.. |
| 13:08.37 | Bonzai090 | chat later ScaredyCat |
| 13:08.49 | ScaredyCat | optimal temp is 15c for me |
| 13:09.06 | *** join/#asterisk yartelecom (n=no-email@82.211.129.231) |
| 13:09.29 | coppice | you must be a true penguin type |
| 13:09.29 | Akelavlk | ScaredyCat what kind of network problem? I have all phones and PCs on same network. |
| 13:10.26 | Bonzai090 | hehe well the IBm technicians love me already i know when they come in to work on my SAn so i turn the server room temp way way down and sit and watch them through the glass windows thats between my office and the server room :> |
| 13:11.05 | *** join/#asterisk Moc_ (n=mochouin@207-199-0-72-ppp.3menatwork.com) |
| 13:11.08 | Bonzai090 | nice to see them freeze then run for coffee then freeze again mwahahhaa |
| 13:16.16 | Katty | beep. |
| 13:16.33 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:16.50 | Uther_P | ploop |
| 13:18.39 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 13:19.23 | *** join/#asterisk Asylum (n=Asylum@dsl-58-6-126-60.qld.westnet.com.au) |
| 13:21.05 | *** join/#asterisk JessicaX^ (i=Jessie@86.112.145.168) |
| 13:23.06 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:24.44 | Hmmhesays | bye |
| 13:24.57 | bkw_ | blah |
| 13:25.04 | *** join/#asterisk secure75 (n=mic@ppp-82-135-4-122.mnet-online.de) |
| 13:25.26 | coppice | bkw: how's the leaking? |
| 13:25.32 | *** join/#asterisk lilalinux (i=e-trolle@deepthroat.deswahnsinns.de) |
| 13:25.35 | cypromis | wet |
| 13:26.11 | Katty | mrow. |
| 13:26.26 | Katty | bkw_: i'm tired. fix it. |
| 13:26.31 | Ariel_ | morning all |
| 13:26.57 | *** join/#asterisk Hmmhesays (n=Hmmm@66.173.103.107) |
| 13:27.03 | Hmmhesays | back |
| 13:27.38 | Hmmhesays | everyone miss me? |
| 13:27.41 | Mimmus | my telco said me that my PRI has 10 incoming and 10 outgoing channels. I suppose they are 1-10 and 11-15,17-21, no? |
| 13:27.58 | Mimmus | (E1 line) |
| 13:28.05 | JessicaX^ | An @! |
| 13:28.14 | Hmmhesays | Mimmus, what makes you think that? |
| 13:28.35 | Hmmhesays | define incoming and outgoing |
| 13:28.36 | *** join/#asterisk elriah (n=jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
| 13:29.02 | darkskiez | Mimmus: I'd imagine you have 10 channels that can be used for incoming or outgoing calls. |
| 13:29.09 | Mimmus | Hmmhesays: I tried 1-10 and 17-25 but noticed some strange beahaviour |
| 13:29.32 | Mimmus | darkskiez: 10 ONLY INCOMING AND 10 ONLY OUTGOING |
| 13:29.48 | darkskiez | OK, THATS INTERESTING |
| 13:29.52 | Mimmus | UFF! |
| 13:30.11 | elriah | Hi all. I have to setup a remote phone. It's a polycomo ip 300. I want to have full functionality - will the remote location need a static IP if I'm using SIP? (leaving out nat and port forwarding for now) |
| 13:30.25 | darkskiez | I dont define the channels on my E1, it seems to be auto-detected |
| 13:30.56 | Mimmus | it's a kind of magic! |
| 13:31.32 | bkw_ | coppice, it seems to be better but the box isn't busy for another hour or so |
| 13:31.42 | darkskiez | i just specify the whole range, and the network seems to take care of the rest |
| 13:32.09 | coppice | bkw: if should *definitely* be better :-) I fixed a bad one |
| 13:33.38 | Hmmhesays | elriah functionality doesn't have much to do with sip |
| 13:34.16 | *** join/#asterisk Dybdahl (n=Lars@cpe.atm2-0-7138.0x50a6f736.odnxx10.customer.tele.dk) |
| 13:35.06 | elriah | Hmmhesays: Well, what I mean by that is, I want to have a fully functional remote phone that sits behind someones home internet connection (understanding that the quality might suck, but this is an experiment). And, for example, when that extension gets a voicemail, I want the voicemail light to light-up like any local sip phones. Is this possible? |
| 13:35.15 | bkw_ | coppice, ;) |
| 13:35.40 | *** join/#asterisk Defraz_ (n=t0tal@24-119-12-238.cpe.cableone.net) |
| 13:36.09 | Ariel_ | Mimmus, most E1's line or channels are both inbound/outbound channels. I would very strange if they are just one way channels |
| 13:36.43 | Hmmhesays | elriah: yes |
| 13:37.08 | elriah | Hmmhesays: Even if the remote IP isn't static? i.e., it may change every so many days/weeks/etc? |
| 13:37.18 | Hmmhesays | elriah: yes |
| 13:37.19 | Katty | Hmmhesays: mew |
| 13:37.25 | Hmmhesays | Kattaaaaay |
| 13:37.59 | elriah | Hmmhesays: Awesome! Any recommendations on where to start? Maybe a how-to somewhere? doesn't have to be specific to the polycom phones ... (Thanks for the help, btw) |
| 13:38.21 | Hmmhesays | assuming you are going to use asterisk with it? |
| 13:38.29 | elriah | Yep. |
| 13:38.54 | Hmmhesays | asterisk should be on a public ip |
| 13:39.13 | elriah | It will be 1:1 natted ... |
| 13:39.34 | Hmmhesays | static ip? |
| 13:39.38 | elriah | Yep. |
| 13:39.41 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 13:40.42 | Hmmhesays | start on the wiki |
| 13:40.52 | elriah | That's where I am now ... |
| 13:40.57 | elriah | Reading through the articles ... |
| 13:41.08 | Katty | wikiwikiwikiwiki mushroommushroom |
| 13:41.40 | Hmmhesays | setting up asterisk is such a broad topic, you need to just dive in. unless you want to pay someone to teach you |
| 13:42.10 | elriah | No, I'm good on most accounts. I have a functional system that I use every day and has about 4 phones. |
| 13:42.18 | Hmmhesays | ahh ok |
| 13:42.21 | Hmmhesays | then what is the issue? |
| 13:42.49 | elriah | * is pretty easy to hack into running.. I wouldn't claim any expertise, but it works.. |
| 13:43.00 | Hmmhesays | so you know how to register phones |
| 13:43.14 | elriah | Yea, good there. |
| 13:43.17 | Hmmhesays | nat=yes |
| 13:43.27 | Hmmhesays | set your externip and localnet |
| 13:43.34 | Hmmhesays | ba da bing, done |
| 13:43.53 | elriah | Really? No port forwarding on the other end? |
| 13:44.15 | Akelavlk | Hello, I have problem with SIP phones. I can call from SIP phone, but when I try call from AIX I got " No route to destination" error. |
| 13:44.24 | Ariel_ | elriah, I would love to know why you think it's pretty easy to hack? you can always put firewall rules iptables with programs like shorewall. |
| 13:44.28 | Hmmhesays | if you have a stateful router set your sip registration expiry to 60 seconds |
| 13:44.38 | elriah | So the SIP registration is a persistent 'tunnel' of sorts? |
| 13:44.48 | elriah | Ariel: No, no.. hack as in get working, not break into. |
| 13:44.50 | Hmmhesays | no, it keeps the dynamic port map open |
| 13:45.07 | elriah | Hmmhesays: Ahh. |
| 13:45.08 | Hmmhesays | tunnel is not the correct term |
| 13:45.30 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
| 13:45.56 | Hmmhesays | pretty much any newer nat router is going to do that |
| 13:45.58 | *** join/#asterisk Stephnie (i=st@203.215.180.250) |
| 13:46.01 | Stephnie | hi |
| 13:46.02 | elriah | Hmmhesays: Makes sense. That sounds almost too easy. Any gotchas? |
| 13:46.07 | coppice | subterranean transport system be better? |
| 13:46.32 | Hmmhesays | if your router isn't stateful, it won't work |
| 13:46.50 | Stephnie | Math() returns value with DECIMAL ....how to get returns without decimal??? any help? |
| 13:46.53 | Hmmhesays | and if you don't send me a case of beer for helping you, the gods will be angered |
| 13:46.55 | *** join/#asterisk SwK (n=SwK@12-219-144-126.client.mchsi.com) |
| 13:47.08 | elriah | Hmmhesays: Well, it's probably something from BellSouth or a Cable modem. I guess only way to tell is to try it. |
| 13:47.20 | Faithful | is there a good tute for setting up call queues? |
| 13:47.31 | elriah | Hmmhesays: Beer? Done. PM me an address. |
| 13:47.31 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 13:47.44 | Hmmhesays | lol, I joke |
| 13:47.56 | Hmmhesays | but... yeah just try it. you'll know in a hurry if it works |
| 13:47.59 | elriah | Well, I appreciate the help. |
| 13:48.24 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 13:48.54 | elriah | I'll give it a shot.. If I see you in channel, I'll let you know how it went. |
| 13:50.06 | *** join/#asterisk sams2100 (n=sams@pcp424683pcs.naugus01.ga.comcast.net) |
| 13:51.03 | Hmmhesays | elriah: cool |
| 13:51.11 | *** part/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 13:51.23 | syle | is enabling MYSQL_FRIENDS in Makefile obsolete now? |
| 13:51.29 | Hmmhesays | I hang out in here to much, usually rambling about something or other |
| 13:52.08 | JessicaX^ | :( |
| 13:52.26 | Hmmhesays | whats wrong JessicaX^? |
| 13:52.27 | Asylum | Wondering if anyone can help, I have my ISDN card setup, I can dial out but number is busy when dialing in.. AUS PRI onramp10 |
| 13:52.36 | elriah | Does SIP only use port 5060 inbound? |
| 13:52.58 | Hmmhesays | it will use whatever you specify, 5060 is default |
| 13:53.17 | elriah | Ok, thanks again. I'm out! Later all!!! |
| 13:53.27 | *** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 13:53.59 | *** join/#asterisk adrianv (n=yoyo@193.239.135.5) |
| 13:54.12 | adrianv | hi all |
| 13:54.40 | netnameus | how can I initiate a restart command to a polycom phone through asterisk? |
| 13:54.44 | *** join/#asterisk [Jedi] (n=hhgds4@213.162.200.226) |
| 13:55.08 | cpatry | netnameus: theres a script called reboot-polycom.pl somewhere on the wikis. |
| 13:55.32 | adrianv | i have a question also :) |
| 13:55.50 | Hmmhesays | adrianv, ask it |
| 13:55.52 | Akelavlk | My SIP registration is failed, I got error "handle_request_register: Registration from 'sip:192.168.1.37' failed for '192.168.1.38'" What's wrong? |
| 13:56.02 | Hmmhesays | Akelavlk, bad credentials |
| 13:56.07 | adrianv | my problem is: i have a sip provider and an asterisk server @home |
| 13:56.09 | netnameus | thanks cpatry, i'll look |
| 13:56.17 | adrianv | some extensions on it |
| 13:56.20 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 13:56.23 | Akelavlk | Bad what? |
| 13:56.32 | Hmmhesays | username or password |
| 13:56.39 | *** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net) |
| 13:56.48 | adrianv | and i would like that all calls (except for my extensions) to be routed via my sip account |
| 13:57.07 | Akelavlk | My sip.conf is [jano_sip] |
| 13:57.07 | Akelavlk | type=friend |
| 13:57.07 | Akelavlk | username=jano_sip |
| 13:57.07 | Akelavlk | secret=supersec |
| 13:57.07 | Akelavlk | host=dynamic |
| 13:57.25 | Akelavlk | Client is set up correctly with same parameters.. |
| 13:57.26 | Ariel_ | adrianv, are you talking about outbound dialing or inbound calls |
| 13:57.35 | adrianv | outbound |
| 13:57.58 | syle | akelavlk |
| 13:58.03 | syle | do you have that in realtime at all? |
| 13:58.09 | Ariel_ | adrianv, make a outbound route for everyone that uses something like a 8 or 9 and one for yours that uses the other number |
| 13:58.25 | Akelavlk | syle, how do you mean that? realtime? |
| 13:58.37 | syle | http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+mysql+peers |
| 13:58.52 | adrianv | Ariel_, i've set up sip.conf ... register => user:pass@sip.provider.com/extension |
| 13:58.59 | *** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-48-89.dialin.kamp-dsl.de) |
| 13:59.05 | syle | i been trying to figure out how to set it up with friends, i can;t find mysql_friends definition in channels Makefile |
| 13:59.06 | Akelavlk | Aha realtime config. I have it in sip.conf file. |
| 13:59.06 | RoyK | with asterisk cvs head and sangoma/wanpipe, i get really bad noise on incoming calls from pstn. outgoing calls to pstn sound ok, though |
| 13:59.09 | Ariel_ | adrianv, that is not for outbound that is for inbound |
| 13:59.16 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 13:59.36 | syle | akelavlk: are you able to use friends? |
| 13:59.44 | adrianv | Ariel_, something like this: exten => _9., 1, Dial(SIP/${EXTEN:1}@sip.provider.com,,rT) ? |
| 13:59.47 | syle | akelavlk: or just peer and user |
| 14:00.01 | syle | i suppose could point them at same table |
| 14:00.17 | Akelavlk | syle, I just use friends. I didn't try anything else.. |
| 14:00.27 | Akelavlk | syle, should I try peer type? |
| 14:00.30 | syle | which howto did you use? |
| 14:00.32 | Ariel_ | adrianv, yes something like that. But you said you have asterisk@home why not use it's gui for it. |
| 14:00.57 | RoyK | adrianv: there really is no point of using ,r |
| 14:01.03 | adrianv | Ariel_, rackmount server... no graphics on it :D |
| 14:01.21 | Ariel_ | adrianv, so you still have asterisk@home on it don't you? |
| 14:01.23 | Stephnie | ${123456789.444444} <----- how to get {123456789} ??? lenth of digits before decimal are unknown but after decimal lenth of digits are only 6 .... any help?? |
| 14:01.41 | adrianv | Ariel_, i do have.. |
| 14:01.51 | Ariel_ | adrianv, then use the web setup on it |
| 14:01.58 | adrianv | Ariel_, it gives me the following error: |
| 14:02.15 | adrianv | Ariel_, Forbidden - wrong password on authentication for INVITE to '"xlite1" <sip:1@193.239.135.2>;tag=as0d49ad55' |
| 14:02.44 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 14:03.15 | syle | akelavlk: which howto did you use? |
| 14:03.28 | *** join/#asterisk ender (n=me@fedora/ender) |
| 14:03.36 | Ariel_ | adrianv, ether remove the conf files there and user your own made ones or use the gui. They will conflict with your own rules unless you set them up in the custom.conf files even then it will be a problem. |
| 14:03.43 | Ariel_ | ether/either |
| 14:04.15 | Ariel_ | the gui is on your via a web browser at http://IPADDRESS/admin |
| 14:04.38 | bkw_ | www.cnn.com |
| 14:04.44 | bkw_ | I sure hope someone let that dog out of that house |
| 14:04.52 | adrianv | Ariel_, hmmm.. |
| 14:04.54 | Akelavlk | syle, http://www.voip-info.org/wiki-SIP and next one vio-info.org page with configuration examlple. |
| 14:04.59 | Katty | bkw_: humans are horrible sometimes :< |
| 14:05.26 | bkw_ | I wouldn't leave my dog behind like that |
| 14:05.48 | Hmmhesays | the problem is... reproducing is so easy |
| 14:06.01 | Katty | human or otherwise :P |
| 14:06.35 | Akelavlk | syle, this one http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf |
| 14:07.26 | bkw_ | Ariel_, I think if they can help they should :P |
| 14:08.00 | Ariel_ | bkw_, I was in a similar problem back in 1993 belive me the last thing on your mind is dogs or cats but the kids and family. |
| 14:08.16 | bkw_ | my dog is part of the family |
| 14:08.18 | Ariel_ | sorry 1992 and I do belive in saving dogs /cats ect. |
| 14:08.19 | bkw_ | along with my cats |
| 14:08.41 | Akelavlk | syle, what SIP softphone are you using? May be it's problem with softphone.. |
| 14:08.47 | Hmmhesays | your animals have a higher chance of surviving on their own than children would |
| 14:08.59 | bkw_ | Hmmhesays, ya that is true |
| 14:09.01 | spackle | Ariel_: Where were you during the floods of '93? |
| 14:09.10 | Ariel_ | belive me If you let them free the do get by better then with you. |
| 14:09.22 | syle | akelavlk: ? that url isn;t realtime |
| 14:09.35 | Ariel_ | spackle, hurricane andrew 1992 took my house. and many others around me |
| 14:09.55 | spackle | Ariel_: ah, I see. |
| 14:10.22 | Ariel_ | bkw_, we found our dog actually she came back 3 days later with out a scratch. |
| 14:10.55 | Akelavlk | syle, I don't need load SIP accounts from DB. It's same like load data from file.. |
| 14:10.58 | adrianv | Ariel_, sorry.. but i couldn't find the web extensions.. my system is a gentoo (asterisk 1.0.9) should i use amportal? |
| 14:12.07 | *** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 14:13.24 | Faithful | This is wierd... I had a problem dialing out through a particular provider... it would complain about trunk timing being sent from the other end... but in the process of cleaning up my extensions.conf... the problem dissapeared ... does that make any sense??? |
| 14:15.21 | adrianv | Ariel_, sorry.. when i said '@home' i meant an installation of asterisk at home.. not that cd :) |
| 14:15.34 | adrianv | or distro |
| 14:15.41 | shido6 | process of the day, clean up your dialplan (extensions.conf) |
| 14:16.06 | *** part/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 14:16.31 | Akelavlk | syle, shit. It was problem with SIP softphone. So don't use sipXezPhone-0.35a |
| 14:17.18 | bjohnson | ewww .. I'm afraind of my dial plan |
| 14:17.40 | bkw_ | Now I wonder when the powers that be try to say "copy & paste" violates copyright law :P |
| 14:18.11 | coppice | bkw: have you seen that lexmark thing? |
| 14:18.13 | Ariel_ | adrianv, ok |
| 14:19.22 | Ariel_ | adrianv, no you should not use amportal if your able to learn the dial rules |
| 14:19.26 | Ariel_ | ~docs |
| 14:19.27 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 14:20.39 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 14:20.56 | memic | howto forward incoming calls to 2 sip phones? |
| 14:20.59 | many | *eek* "> AT+BLDN" "< ERROR" *bt-hs powered off and letnt power on again* |
| 14:21.10 | many | Dial(SIP/10&SIP/20) |
| 14:21.20 | memic | many thx |
| 14:21.22 | memic | ;) |
| 14:21.35 | many | yea many no problem. |
| 14:21.37 | many | ;) |
| 14:21.48 | *** join/#asterisk littleball (n=littleba@cm157.epsilon173.maxonline.com.sg) |
| 14:22.39 | adrianv | Ariel_, as i said... i've done everything by the book.. the only thing is that when i try to make an outbound call it gives me that error (Forbidden - wrong password on authentication for INVITE to '"xlite1" <sip:1@193.239.135.2>;tag=as0d49ad55'). |
| 14:22.56 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 14:23.23 | adrianv | Ariel_, of course my internal extensions don't exist on my providers server so i'm stuck |
| 14:23.59 | adrianv | Ariel_, i think i need to do something like NAT in order to place outbound calls |
| 14:24.27 | littleball | adrianv, what do you call? |
| 14:24.45 | *** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
| 14:24.48 | *** join/#asterisk mogorman (n=mogorman@digium.com) |
| 14:25.07 | adrianv | littleball, my setup is like this: some extensions for home and a sip account for everything else |
| 14:25.15 | *** join/#asterisk santiago (n=santiago@63.245.86.254) |
| 14:25.34 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 14:26.31 | adrianv | with NAT i mean, instead for the server at my provider he should see an inbound conn from myusername@sip.provider.com instead of my_extension@asterisk.server |
| 14:26.31 | littleball | enable your log and check SIP/XXX. What is XXX there? |
| 14:26.43 | littleball | i encount this before. |
| 14:27.14 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 14:27.18 | littleball | At the beginning,i try to Dial(SIP/1234), it failed. I need to call SIP/gwen etc. |
| 14:27.40 | adrianv | littleball, i've setup this: exten=_9., 1, Dial(SIP/${EXTEN:1}@sip.provider.com) |
| 14:27.52 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 14:28.01 | adrianv | littleball, when i call for example 98337 |
| 14:28.22 | adrianv | <PROTECTED> |
| 14:28.37 | adrianv | <PROTECTED> |
| 14:28.50 | adrianv | Sep 6 16:49:10 WARNING[509]: chan_sip.c:6864 handle_response: Forbidden - wrong password on authentication for INVITE to '"xlite1" <sip:1@193.239.135.2>;tag=as0d49ad55' |
| 14:29.05 | adrianv | and i end up with a busy tone |
| 14:29.44 | Bonzai090 | any one got a idea on how to enable callerid for in comming lines want to pass the caller id to my extentions |
| 14:30.21 | Stephnie | adrianv: do you have any peer [8337] in your sip.conf? |
| 14:30.25 | littleball | Can you change SIP/{EXTEN:1}#sip.provider.com to SIP/peer name? |
| 14:30.52 | littleball | yes, i also think so. you need to check sip.conf |
| 14:30.52 | adrianv | littleball, i've tried putting exten=_9., 1, dial(SIP/myuser:mypass@${EXTEN:1}@sip.provider.com) |
| 14:30.54 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 14:30.59 | *** part/#asterisk bkw_ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
| 14:30.59 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
| 14:31.07 | supaigtr | Hello. |
| 14:31.10 | littleball | check you sip conf for that peer |
| 14:31.12 | *** join/#asterisk Pj___ (n=pj@fernande.happycoders.org) |
| 14:31.29 | fulgas | trying to register a * on a sip proxy... can make calls but can't received :| |
| 14:32.08 | adrianv | Stephnie, [telip] |
| 14:32.08 | adrianv | type=peer |
| 14:32.08 | adrianv | secret=mypass |
| 14:32.08 | adrianv | username=8xx8 |
| 14:32.08 | adrianv | fromuser=8xx8 |
| 14:32.09 | adrianv | host=sip.provider.com |
| 14:32.11 | adrianv | fromdomain=provider.com |
| 14:32.13 | adrianv | insecure=very |
| 14:32.15 | adrianv | context=telip |
| 14:32.46 | *** join/#asterisk elriah (n=jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
| 14:33.05 | *** join/#asterisk wolfson (n=hehe@65.174.122.198) |
| 14:33.06 | adrianv | Stephnie, i'm trying to trunk for calls outside my asterisk pbx |
| 14:33.29 | littleball | i think you should call SIPtelip@sip.provider.com or SIP/telip |
| 14:33.45 | littleball | SIP/telsip@.... |
| 14:33.46 | Stephnie | so you want to dial 8337 through your SIP Provider... |
| 14:34.09 | *** join/#asterisk RoyK (n=roy@216-99-212.0506.adsl.tele2.no) |
| 14:34.29 | Stephnie | adrianv: it should be Dial(SIP/8337@telip) . . . |
| 14:34.32 | supaigtr | Anyone have a link on how to get IAX2 dial to work in both directions. I have it working in one but the reverse gets rejected. static - dynamic works dynamic -> static doesn't. I'm sure it has something to do with register. |
| 14:34.37 | RoyK | hi |
| 14:34.49 | adrianv | let me try that |
| 14:34.52 | RoyK | how can i dump the raw audio from a PRI? as from the bchan? i keep getting terrible noise on incoming calls from a PRI :( |
| 14:35.14 | Stephnie | adrianv: if you want to dial out 8337 through [telip] . . then that is the correct syntax |
| 14:35.28 | mogorman | hey stephnie, you gonna call in? |
| 14:35.35 | *** join/#asterisk bkw_ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
| 14:35.35 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:35.53 | Stephnie | me??? call where? |
| 14:36.28 | tzanger | quick 'top' question with asterisk |
| 14:36.37 | shido6 | top lies |
| 14:36.46 | tzanger | I'm showing VIRT of 18M, RES of 9M and SHR of 4.6M |
| 14:37.00 | tzanger | shido6: that's why I'm asking, I know it's not obvious |
| 14:37.23 | tzanger | does that mean that altogether, all the * processes are using 18M of memory, with 4.6 of that shared between all processes? |
| 14:38.38 | adrianv | Stephnie, thanks.. it works |
| 14:38.39 | adrianv | :) |
| 14:38.47 | Stephnie | :) .np |
| 14:38.49 | tzanger | I mean with unthreaded processes RES is the resident size with SHR being all the COW pages/shared libs |
| 14:41.30 | *** join/#asterisk elriah (n=jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
| 14:42.14 | coppice | cows can read. they don't need pages |
| 14:43.12 | elriah | Hmmheysays: Hey! Great success - can dial the remote extension and it works fine. One problem - When trying to make outbound calls, the remote SIP phone is using really high ports (i.e., 12xxx, 17xxx) .. Is it suppose to do that? I thought it was only 5060 .. |
| 14:43.18 | *** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com) |
| 14:43.30 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
| 14:43.45 | elriah | udp ports |
| 14:43.54 | lehel | hello |
| 14:44.16 | bjohnson | elriah: yes. set in /etc/asterisk config files |
| 14:44.17 | coppice | 5060 is only for the SIP traffic |
| 14:44.21 | Katty | mrow |
| 14:44.53 | lehel | in my meetme.conf can i define a multiple number like: conf => 81XX ? |
| 14:44.56 | elriah | Thanks - set where? I looked through sip.conf ... So SIP sets up the session, but calls go through other ports? |
| 14:45.15 | coppice | SIP is only a signalling protocol |
| 14:45.52 | *** join/#asterisk loick (n=loick@81.255.80.161) |
| 14:46.08 | elriah | I see. What config file holds these ports? |
| 14:46.14 | JamesDotCom | it's the rtp traffic |
| 14:46.19 | *** join/#asterisk RoyKa (n=roy@216-99-212.0506.adsl.tele2.no) |
| 14:46.21 | JamesDotCom | do a bit of basic googling/reading |
| 14:46.31 | ScaredyCat | rtp.conf |
| 14:46.37 | elriah | Thanks. Will do. |
| 14:48.14 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:48.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:48.39 | elriah | So is RTP only UDP or does it use TCP as well? |
| 14:48.56 | coppice | RTP is for streaming, so it only uses UDP |
| 14:49.19 | elriah | Great. Easy - thanks!!!! |
| 14:49.46 | Faithful | guys... I think I have my queues.conf pretty well configured ... what do I need to do to make it happen? |
| 14:50.53 | elriah | Works great now!!!! |
| 14:51.00 | shido6 | LOL |
| 14:51.21 | shido6 | I guess that works... my car is configured, how do I start her up |
| 14:51.39 | elriah | @shido6 eh? at me? |
| 14:52.09 | Katty | hi anthm (= |
| 14:52.14 | anthm | hi |
| 14:55.09 | bkw_ | I still don't get why people think that the SDP has anything to do with the size of the rtp packets |
| 14:55.23 | Hmmhesays | cause someone planted in their brain that it does? |
| 14:55.36 | Katty | people like that need hugs. |
| 14:55.50 | ScaredyCat | bkw_: ! |
| 14:55.56 | Hmmhesays | i prefer to deliver a swift kick in the nuts |
| 14:56.05 | Katty | Hmmhesays: you would. |
| 14:56.07 | Hmmhesays | <chuckle> j/k |
| 14:56.07 | bkw_ | ScaredyCat, yes? |
| 14:56.10 | Katty | Hmmhesays: you're the ANTI-HUG |
| 14:56.16 | ScaredyCat | I had an interesting idea... |
| 14:56.19 | ScaredyCat | 1 sec door |
| 14:56.29 | Bonzai090 | ScaredyCat :) got time to chat to me about SD instalation?.. |
| 14:56.51 | bkw_ | FYI I hate banks.. I have credit card processors.. I HATE THEM |
| 14:57.01 | Hmmhesays | is that something like the anti-drug? |
| 14:57.02 | bkw_ | they all need to be poked in the eye |
| 14:57.05 | tzanger | is it PCIX or PCIe that will take a standard PCI33 card? |
| 14:57.10 | mogorman | nah man |
| 14:57.19 | mogorman | you just need a swiss bankaccount bkw_ |
| 14:57.28 | bkw_ | no this is for credit card processing |
| 14:57.35 | bkw_ | they can't choose an API that is sane |
| 14:57.49 | bkw_ | they have some bastardized stuff to let us interface and its pissing m e off |
| 14:58.48 | bkw_ | the question is .. where in chan_sip are values from the pvt copied to the user/peer? |
| 14:59.02 | ScaredyCat | back |
| 14:59.03 | oej | Everywhere |
| 14:59.09 | oej | bkw_: Everywhere... |
| 14:59.10 | ScaredyCat | ok... bkw_ |
| 14:59.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:59.31 | ScaredyCat | you wont be particularly interested... since you're a sip kinda guy.. |
| 14:59.31 | bkw_ | oej, i'm going to try to make that one rtp patch per peer settings for packetization |
| 14:59.38 | bkw_ | ScaredyCat, shoot |
| 14:59.40 | ScaredyCat | but... |
| 14:59.59 | oej | bkw_: Just check for the other settings, like vmexten and you'll find every place it's copied |
| 15:00.02 | ScaredyCat | what about passing pricing info via iax during the call... |
| 15:00.15 | bjohnson | bkw_: grab their head, and swipe it |
| 15:00.32 | *** join/#asterisk muaddib (n=craig@c-67-177-183-28.hsd1.in.comcast.net) |
| 15:00.37 | muaddib | hello all |
| 15:00.52 | bkw_ | coppice, http://pastebin.com/356132 |
| 15:00.56 | *** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co) |
| 15:01.00 | muaddib | Question does Asterix support Radius and if so where can I find more information on it I can't find much mention on the main asterisk site |
| 15:01.07 | bkw_ | its Asterisk |
| 15:01.09 | bkw_ | not Astrix |
| 15:01.21 | mogorman | thanks bkw_ was gonna have to kick some one in the head |
| 15:01.22 | konrads | Asterisk and Obelix :) |
| 15:01.29 | mogorman | exactly |
| 15:01.32 | muaddib | bkw_: my mistake |
| 15:01.35 | mogorman | what is wrong in this picture |
| 15:01.51 | bkw_ | go search voip-info.org for info on using asterisk and radius |
| 15:01.57 | *** join/#asterisk litage (n=nick@203.201.96.60) |
| 15:01.58 | Wonka | konrads: no, Asterisk and Obelisk |
| 15:02.05 | ScaredyCat | not impressed then bkw_.. |
| 15:02.10 | ScaredyCat | lo Bonzai090 |
| 15:02.13 | Wonka | *mew* |
| 15:02.25 | bkw_ | ScaredyCat, passing variables via the IAX2 connection |
| 15:02.28 | bkw_ | that will never fly |
| 15:02.32 | bkw_ | we did a patch for that |
| 15:02.34 | mogorman | bkw_ your gonna extend your rtp thing to sip and iax2 peers? |
| 15:02.48 | bkw_ | mogorman, you work for digium you should know that iax2 doesn't use RTP |
| 15:02.49 | *** join/#asterisk CaNaBiS_ (i=canabis@ip68-111-51-215.br.no.cox.net) |
| 15:02.56 | mogorman | no not for that |
| 15:03.03 | mogorman | but variable frame size |
| 15:03.09 | muaddib | Also I seen this term mentioned alot on the website CDR? what is that |
| 15:03.10 | bkw_ | again IAX2 can't |
| 15:03.11 | mogorman | instead of 20ms frames |
| 15:03.14 | mogorman | huge frames |
| 15:03.19 | mogorman | really? |
| 15:03.25 | bkw_ | IAX2 doesn't suffer the same overhead as RTP |
| 15:03.28 | ScaredyCat | i mean specifically pricing |
| 15:03.29 | bkw_ | it wouldn't gain much from doing so |
| 15:03.29 | mogorman | i thought it was a #define |
| 15:03.32 | mogorman | yeah |
| 15:03.37 | tzanger | trunking kicks ass |
| 15:03.39 | mogorman | true |
| 15:03.42 | bkw_ | not even trunking |
| 15:03.45 | bkw_ | just raw IAX2 |
| 15:03.48 | Hmmhesays | file oh file where art thou |
| 15:03.53 | coppice | bkw_: what is this in the pastebin? |
| 15:04.05 | bkw_ | coppice, a crash in rxfax/spandsp |
| 15:04.09 | RoyKa | Hmmhesays: find / -type f -name file |
| 15:04.10 | RoyKa | :P |
| 15:04.25 | coppice | bkw: oh dear |
| 15:04.48 | bkw_ | memory usage is down because it crashes every few hours also :P |
| 15:04.49 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 15:04.55 | bkw_ | hehe |
| 15:04.56 | RoyKa | coppice: i can't load app_[tr]xfax anymore :( |
| 15:05.10 | bkw_ | RoyKa, did you recompile it? |
| 15:05.12 | mogorman | heh nice |
| 15:05.36 | Hmmhesays | if I dial sip:1235@foo and that returns a 404, will asterisk return CHANUNAVAIL as a dialstatus? |
| 15:06.01 | bkw_ | oej, i'm not even really putting much effort into the per peer rtp thing unless I can get feedback on an approved approach |
| 15:06.06 | oej | bkw_: "The ptime attribute in sdp gives the length of time in milliseconds represented by the media in the packet." |
| 15:06.06 | bkw_ | its not worth my time to even try otherwise |
| 15:06.09 | Ariel_ | argh why, why is there another storm forming off the coast of Floirda... Why.... Tropic storm warnings are not in effect from jupitor north. |
| 15:06.11 | brad_mssw | muaddib: http://www.voip-info.org/tiki-index.php?page=CDR |
| 15:06.23 | oej | bkw_: I should also wait until that was sorted out. It's easy to add peer/user stuff later |
| 15:06.37 | Hmmhesays | anyone know off hand? |
| 15:06.53 | bkw_ | oej but rtp stacks don't really care It seems |
| 15:06.55 | bkw_ | asterisk doesn't |
| 15:07.17 | oej | bkw_: Don't take Asterisk as a reference implementation :-) |
| 15:07.34 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 15:07.39 | oej | bkw_: I don't know if it's used, found it in rfc 2327. Let's look in the RTP/AVP profile |
| 15:07.55 | bkw_ | even my sipura doesn't care |
| 15:08.03 | bkw_ | it will chew what ever you throw at it |
| 15:08.19 | oej | So the packet length is included in the rtp packet? |
| 15:08.37 | bkw_ | it seems so |
| 15:09.05 | bkw_ | but if we need to support the ptime thing you spoke about we need this option in ast_rtp_write |
| 15:09.20 | bkw_ | I looked all over and found nothing about ptime when I was looking for it |
| 15:09.21 | malverian[work] | Ariel_, You from FL? |
| 15:09.27 | *** join/#asterisk dasuberdavid (n=dasuberd@digium.com) |
| 15:09.36 | spackle | Ariel_: it is supposed to be a busy storm year isn't it? |
| 15:09.37 | Ariel_ | malverian[work], yes |
| 15:09.42 | malverian[work] | Same here. |
| 15:09.49 | oej | ptime is sdp, let's check RTp |
| 15:09.53 | Ariel_ | spackle, yes it is suppose to be that way |
| 15:10.09 | oej | The RTP rfc might say "Don't use PTIME, we're doing it somewhere else" |
| 15:10.31 | RoyKa | Ariel_ Sharon? |
| 15:11.00 | bkw_ | oej well speex looks like the only one that needs to have a ptime param |
| 15:11.06 | Ariel_ | RoyK, no |
| 15:11.25 | Hmmhesays | anyone? anyone? ugh I'm going to have to test this |
| 15:11.47 | oej | bkw_: Did you try my patch I made for you earlier today? On the maxcalls stuff |
| 15:11.54 | bkw_ | not yet just seen it |
| 15:12.05 | RoyK | omg http://content.ytmnd.com//140000/140430/image.jpg |
| 15:12.09 | bkw_ | i'll have to try it later today |
| 15:12.12 | oej | bkw_: Please try the IAX2 stuff also... |
| 15:12.23 | bkw_ | will do |
| 15:12.32 | *** join/#asterisk huslage_ (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
| 15:12.46 | bkw_ | oej, did you see the nice thing in rtptimeout |
| 15:13.01 | lehel | in my meetme.conf can i define a multiple number like: conf => 81XX ? |
| 15:13.03 | bkw_ | if you're using app_record.. asterisk sends no rtp back |
| 15:13.15 | oej | bkw_: Yes, I made a comment I believe. Maybe not. But I fully understand. We might have to use the cng generator |
| 15:13.24 | bkw_ | hehe |
| 15:13.29 | oej | bkw_: What about voicemail? |
| 15:13.33 | Qwell | lehel: You can use dynamic confs |
| 15:13.38 | bkw_ | does the same thing |
| 15:13.40 | *** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net) |
| 15:13.47 | *** join/#asterisk Lars (n=Lars@cpe.atm2-0-7138.0x50a6f736.odnxx10.customer.tele.dk) |
| 15:14.18 | oej | bkw_: That worries me as well |
| 15:14.46 | spackle | should this work to record the channel as conf+datetime ? Monitor(wav,/var/spool/asterisk/monitor/conf${datetime},m) |
| 15:14.53 | SwK[Work] | RoyK: is that your video card? |
| 15:15.05 | RoyK | SwK[Work]: hehe. not me, no |
| 15:15.10 | RoyK | not my card either |
| 15:16.19 | spackle | all i get is conf.wav |
| 15:16.48 | Bonzai090 | heh wonder if there is a gsm file for sorry our city was hit by a huricane please tray again in 24 weeks time |
| 15:17.09 | tzanger | Bonzai090: go to thevoice.com and get Alison to make it up |
| 15:17.28 | lehel | Qwell: sorry, how do you mean dynamic confs? |
| 15:17.28 | malverian[work] | Hmmm.. |
| 15:17.42 | coppice | bkw_: this seg fault was after much running, right? |
| 15:17.45 | malverian[work] | Is spandsp still the primary method of receiving faxes in asterisk? |
| 15:18.04 | RoyK | malverian[work]: yes |
| 15:18.22 | Bonzai090 | hmm thevoice.com is some survey builder |
| 15:18.28 | bkw_ | coppice, maybe 3 hours |
| 15:18.28 | RoyK | malverian[work]: afaik the only way |
| 15:18.59 | coppice | malverian[work]: why would anyone want to create another soft-fax? |
| 15:19.01 | spackle | thevoice.digium.com is what he meant. |
| 15:19.06 | malverian[work] | coppice, True ;) |
| 15:19.13 | *** join/#asterisk LDybdahl (n=Lars@cpe.atm2-0-7138.0x50a6f736.odnxx10.customer.tele.dk) |
| 15:19.21 | Bonzai090 | ahh betta |
| 15:19.31 | malverian[work] | It appears that all of the "whole" packages that include the app_txfax/rxfax are spandsp-0.0.1 |
| 15:19.38 | malverian[work] | RoyK, I know, I saw in the readme :) |
| 15:20.13 | coppice | nobody should be using spandsp-0.0.1. That's dumb |
| 15:20.53 | tzanger | 0.7.0?? |
| 15:21.02 | RoyK | :) |
| 15:21.17 | malverian[work] | Where can the respective app_txfax and app_rxfax source be found? |
| 15:21.27 | malverian[work] | Not finding anything useful on the voip-info wiki |
| 15:21.44 | coppice | why would you look on the wiki? |
| 15:21.44 | tzanger | malverian[work]: www.opencall.org or www.soft-switch.org IIRC |
| 15:21.57 | RoyK | malverian[work]: http://www.voip-info.org/wiki-Asterisk+spandsp |
| 15:21.57 | *** join/#asterisk andreas_hecker (n=Andreas@p5497F4E5.dip.t-dialin.net) |
| 15:22.01 | lehel | Qwell you have any example of dynamic confs? |
| 15:22.09 | RoyK | lehel: see the wiki |
| 15:22.18 | RoyK | ~google realtime voip-info |
| 15:22.59 | malverian[work] | tzanger, I grabbed spandsp-0.0.3.tar.gz, doesn't appear to have the code for the apps, just the spandsp lib itself. |
| 15:23.21 | tzanger | malverian[work]: spandsp is a support library for app_rx/txfax, look around on the site a little, the faxing apps are separate |
| 15:23.21 | RoyK | malverian[work]: it's in a subdir iirc |
| 15:23.30 | lehel | RoyK: The requested URL /dynamic-meetme.tar.gz was not found on this server. ;) |
| 15:23.32 | coppice | use spandsp-0.0.2pre20. the apps are in the same directory |
| 15:23.34 | malverian[work] | tzanger, Alright, thanks. |
| 15:23.39 | malverian[work] | coppice, Okay. |
| 15:24.22 | coppice | bkw_: did it crash before the latest update? |
| 15:24.32 | bkw_ | no that was today |
| 15:24.34 | bkw_ | at 9am |
| 15:24.57 | tzanger | coppice: is 0.0.3 bad? |
| 15:25.00 | tzanger | or worse than 0.0.2pre20? |
| 15:25.38 | coppice | 0.0.3 has changed the FAX architecture to accomodate T.38. It isn't well enough tested for serious use |
| 15:25.46 | tzanger | coppice: aha. |
| 15:25.58 | *** join/#asterisk kg (n=kg@chello062179062077.chello.pl) |
| 15:25.58 | *** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
| 15:26.23 | lathos42 | Hello |
| 15:26.37 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 15:26.46 | spackle | hey lathos42, are you swimming in phones yet? |
| 15:27.11 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 15:27.27 | ScaredyCat | oh no |
| 15:27.47 | lathos42 | spackle: Not quite yet.. Still getting prices and doing the final convincing of management |
| 15:28.21 | coppice | bkw_: I think spandsp needs building with -g, so the traceback shows the parameters for the calls |
| 15:28.36 | bkw_ | coppice, i'll do that tonight when I rebuild it |
| 15:28.49 | coppice | OK |
| 15:28.51 | bkw_ | I fear touching it while its getting hammered |
| 15:28.55 | malverian[work] | coppice, My distro has a package for spandsp 0.0.2pre18, any major reason I shouldn't use that version? |
| 15:29.03 | bkw_ | MEMORY LEAK |
| 15:29.44 | RoyK | lehel: ah. i don't know about realtime meetme... |
| 15:30.02 | coppice | pre18 should be OKish. A couple of significant things were fixed in 19 and 20 |
| 15:30.14 | lathos42 | spackle: I did have a good laugh this morning when CDW sent me a quote for the IP-501 + PoE cable at $291 per phone |
| 15:30.37 | malverian[work] | Hm.. perhaps I'll commit a new version of the package to cvs.. |
| 15:30.53 | spackle | lathos42, I take it thats high? |
| 15:31.33 | lathos42 | spackle: I had seen web pricing for them at a little over $200 with the cable.. Once I replied telling the sales rep that, she sent me back a quote for $206 per phone with the PoE cable |
| 15:32.05 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
| 15:32.22 | tzanger | I fucking hate companies liek that |
| 15:32.26 | tzanger | I generally give them one kick at the cat |
| 15:32.35 | spackle | tzanger: ditto. |
| 15:32.39 | tzanger | I ask for their best price and if they dick me around I don't buy from them no matter what they say they'll come in at afterward |
| 15:32.52 | spackle | Lathos42: did you try www.pcconnection.com?? |
| 15:33.13 | lathos42 | spackle: Not yet, They're on my list though |
| 15:33.25 | *** join/#asterisk ManxPower (n=eric@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 15:33.28 | spackle | tzanger: I used to work for a guy whose small joy in life was pitting companies against each other like that. |
| 15:33.43 | Assid | heya bkw_ |
| 15:33.53 | spackle | tzanger: waste of time. |
| 15:33.59 | tzanger | ugh |
| 15:34.07 | Assid | i heard you did some iax vs sip connection tests |
| 15:34.16 | lathos42 | the sales rep is supposed to call me this afternoon.. I'm going to ask her why she tried to overcharge me by almost $100 :) |
| 15:34.22 | Assid | and found sip to be superior in clarity as compared to iax? |
| 15:34.54 | coppice | Assid: if you found that, something was buggy :-) |
| 15:35.14 | Assid | coppice: no not me.. there was a discussion earlier |
| 15:35.31 | Assid | if anything iax i thought would be better.. since it has less overhead usage |
| 15:35.32 | tzanger | Assid: I find it extraordinarily hard to believe the transport "sounds" any different |
| 15:35.45 | Assid | seriously.. someone was discussing that earlier |
| 15:36.03 | *** join/#asterisk grimse (n=grimse@p5481EB54.dip.t-dialin.net) |
| 15:36.04 | *** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com) |
| 15:36.24 | Assid | 6 hrs ago |
| 15:36.27 | Assid | aprox |
| 15:36.59 | *** join/#asterisk huslage_ (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
| 15:37.28 | file | FOOD |
| 15:37.34 | coppice | Assid: if there is *any* audio quality difference, something is broken. the protocols won't affect that |
| 15:37.58 | *** join/#asterisk Lars (n=Lars@cpe.atm2-0-7138.0x50a6f736.odnxx10.customer.tele.dk) |
| 15:38.17 | Assid | coppice: thats what i thought |
| 15:38.19 | steve___ | tzanger i didn't test it on the other pinab, but I was seeing a load of 2.00 on the pbx box - it's transcoding g.729 back to ulaw |
| 15:38.30 | RoyK | ka-ding |
| 15:38.48 | *** join/#asterisk RoyK (n=roy@216-99-212.0506.adsl.tele2.no) |
| 15:38.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:38.55 | Assid | but apparently they were arguing saying the sip uses a seperate thread or something about sip headers+rtp helping it out or something |
| 15:39.03 | Hogie | has anybody seen nufone ever return a busy for certain numbers all the time? |
| 15:39.04 | tzanger | has anyone here seen screwy load/general lack of many transcodes because of using ztdummy on sip or iax calls?? |
| 15:40.12 | file | the implementation of a protocol can effect audio quality |
| 15:40.39 | RoyK | file: with rtp-based protocols? |
| 15:40.42 | tzanger | file: yes, but none of my tcpdumps shows the packets being "warped"... they're being received at almost exactly 20ms intervals most of the time |
| 15:41.14 | tzanger | file: the only way a protocol implementation can fuck up audio is if it's sending packets out wrong or otherwise causing problems with the transmission or reception of said packets. |
| 15:41.30 | tzanger | file: or, if it's reaching up into the next protocol layer and mangling the packet data itself |
| 15:42.13 | file | tzanger: yup. |
| 15:42.28 | *** join/#asterisk Asylum (n=Asylum@dsl-58-6-126-60.qld.westnet.com.au) |
| 15:42.30 | Asylum | list |
| 15:42.33 | Asylum | woops |
| 15:42.41 | *** join/#asterisk ScaredyCat (n=ScaredyC@84.119.131.232) |
| 15:42.49 | Asylum | hey ScaredyCat :) |
| 15:42.56 | ScaredyCat | lo |
| 15:43.09 | Asylum | Hey have you got time for me to pick your brain? |
| 15:43.28 | ScaredyCat | yeah, sure... |
| 15:43.42 | Asylum | Ok, well, I have the ISDN card working now |
| 15:43.45 | Asylum | I can call out |
| 15:43.48 | Asylum | but i can't call in |
| 15:44.06 | ScaredyCat | do you get anything on the console at at=ll? |
| 15:44.08 | Asylum | i have setup the incomming parameters in amp portal.. but still no luck.. |
| 15:44.18 | ScaredyCat | oh god... not amp |
| 15:44.39 | Asylum | nope i looked in the asterisk cli and nothing comes up when i try and call in |
| 15:44.48 | ScaredyCat | ok... but can you see a call attempt coming in on the * console |
| 15:44.51 | ScaredyCat | ahh |
| 15:44.51 | *** join/#asterisk t3t (n=t3t@galley.pangalacticgargleblaster.com) |
| 15:44.54 | ScaredyCat | ok... |
| 15:44.55 | *** join/#asterisk [Lamer] (i=Lamer@221.128.89.196) |
| 15:45.12 | Asylum | I think it must have something todo with the zaptel config files |
| 15:45.24 | Asylum | I'm in australia remember.. |
| 15:45.42 | Asylum | haha |
| 15:45.57 | Asylum | Any Suggestions? |
| 15:46.00 | ScaredyCat | ok, but you should get someting on the console at least |
| 15:46.07 | Asylum | nothing.. |
| 15:46.07 | ScaredyCat | try turning on debug |
| 15:46.18 | Assid | sorry.. back |
| 15:46.24 | Asylum | and the command for that would be DEBUG= ?? |
| 15:46.37 | ScaredyCat | pri debug |
| 15:46.40 | Assid | tzanger: and they were saying something abut bkw_'s report or something |
| 15:46.47 | *** part/#asterisk secure75 (n=mic@ppp-82-135-4-122.mnet-online.de) |
| 15:47.02 | Asylum | pri no command... |
| 15:47.22 | Asylum | DEBUG=pri ? |
| 15:47.26 | ScaredyCat | no |
| 15:47.34 | ScaredyCat | PRI debug in the * console |
| 15:47.55 | Asylum | no such command |
| 15:48.05 | ScaredyCat | wtf |
| 15:48.59 | ScaredyCat | what version of * are you running... |
| 15:49.05 | Hmmhesays | thats cool, asterisk doesn't destroy a channel sent to an ip that is not reachable |
| 15:49.19 | Asylum | asterisk@home 1.5 |
| 15:49.28 | Hmmhesays | hey file |
| 15:49.30 | ScaredyCat | no, waht version of asterisk |
| 15:49.34 | ScaredyCat | show version |
| 15:49.41 | file | Hmmhesays: hail! |
| 15:49.43 | Asylum | 1.0.9 |
| 15:49.51 | ScaredyCat | should be there then... |
| 15:49.58 | Hmmhesays | 2005-09-01 15:19:47 UTC |
| 15:49.58 | Hmmhesays | <PROTECTED> |
| 15:50.01 | ScaredyCat | althoug |
| 15:50.11 | ScaredyCat | @home might be the clue .. |
| 15:50.16 | Asylum | lol |
| 15:50.17 | Hmmhesays | file I have a question, what do I need SER to send back to asterisk to make the dialstatus chanunavail if I can |
| 15:50.21 | ScaredyCat | maybe there's no pri stuff in that |
| 15:50.30 | ScaredyCat | which would be silly |
| 15:50.33 | file | Hmmhesays: hrm lemme look |
| 15:50.53 | *** join/#asterisk jonathh (n=asd@host81-154-159-222.range81-154.btcentralplus.com) |
| 15:51.15 | file | Hmmhesays: nothing. |
| 15:51.23 | file | you can't :) |
| 15:51.26 | Asylum | I dunno but it's annoying the heck out of me |
| 15:51.55 | *** join/#asterisk eKo1 (n=kino@metrored-gw.tropicohn.com) |
| 15:51.56 | ScaredyCat | is zaptel etc loaded? |
| 15:51.57 | Hmmhesays | hmm, ok so basically anything that fails is going to produce a noanswer |
| 15:52.09 | ScaredyCat | do an: lsmod |
| 15:53.02 | Asylum | yup |
| 15:53.30 | Asylum | zaptel 180128 26 [wcusb wcte11xp] |
| 15:53.50 | file | Hmmhesays: shouldn't |
| 15:53.54 | file | Hmmhesays: congestion maybe |
| 15:54.01 | Hmmhesays | hrm |
| 15:54.14 | file | yeah 404 should give you congestion |
| 15:54.24 | ScaredyCat | can I get on that box Asylum |
| 15:54.27 | file | and 403 forbidden... |
| 15:54.43 | file | and a 503 service unavailable |
| 15:54.47 | Asylum | yeah wait up |
| 15:55.09 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-141-149-248-22.buff.east.verizon.net) |
| 15:55.22 | SuPrSluG | hello |
| 15:56.40 | *** join/#asterisk hellagony (n=egutierr@200.121.129.180) |
| 16:00.23 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 16:00.45 | paryl | i'm trying to make a list of hardware that i'll need to replace my current pbx with asterisk |
| 16:01.14 | *** join/#asterisk jonathh (n=asd@host217-46-145-65.in-addr.btopenworld.com) |
| 16:01.14 | paryl | as far as the server goes... is anything special needed? like dual cpu's, lots of ram, etc? |
| 16:01.49 | paryl | (i ask because i've heard stories of people running asterisk on such little hardware |
| 16:02.19 | steve___ | paryl how many calls are you going to be doing? |
| 16:02.45 | paryl | steve: it's a full T1, 24 channels |
| 16:02.58 | paryl | plus VoIP traffic to other locations |
| 16:03.07 | *** part/#asterisk jonathh (n=asd@host217-46-145-65.in-addr.btopenworld.com) |
| 16:03.09 | paryl | ~40 stations |
| 16:03.29 | steve___ | paryl what codec? |
| 16:03.47 | paryl | no idea... yet another detail to address :) |
| 16:03.58 | *** part/#asterisk case_ (n=case@mailhost.seeft.com) |
| 16:04.04 | paryl | i need it to be at least as good as regular POTS |
| 16:04.18 | paryl | whatever codec will allow for that quality or better |
| 16:04.51 | SuPrSluG | use ulaw or alaw. which is what pots uses |
| 16:05.27 | paryl | ok.. are the others better/worse? |
| 16:05.38 | SuPrSluG | althoough that'll use 64 kbps. gsm uses 13 |
| 16:06.03 | brad_mssw | 64kbps over tcp/ip is more like 80kbps |
| 16:06.16 | SuPrSluG | gsm is cell phone quality. i use it and think it's fine |
| 16:06.59 | Hmmhesays | I suppose I can tell asterisk to retry the route somewhere else on congestion |
| 16:07.01 | SuPrSluG | for my needs. ymmv |
| 16:07.34 | coppice | gsm is more like ex-cellphone quality :-) |
| 16:07.43 | paryl | SuPrSluG: 'cell phone quality' is fairly subjective :) |
| 16:08.11 | coppice | The GSM networks gave up using that codec |
| 16:08.28 | paryl | the big thing is i want the finished product to be as similar to our current voice quality as possible |
| 16:08.29 | SuPrSluG | paryl:true enough. |
| 16:09.06 | *** join/#asterisk Stephnie (i=st@203.215.180.250) |
| 16:09.15 | paryl | anyway, as far as hardware goes... is it just 'the more the better', or is there some way to figure out how much i need? |
| 16:09.22 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 16:09.35 | Stephnie | Math(RV,42686305+9) = Result is 42686312.000000 <------- why asterisk returns the "wrong answer" ??? it should be 42686314 ...??????? |
| 16:09.43 | Stephnie | any help??? |
| 16:09.45 | *** join/#asterisk azrishahril (n=azrishah@60.50.197.222) |
| 16:10.22 | coppice | I guess it ran out of fingers :-) |
| 16:10.31 | paryl | haha |
| 16:10.32 | Stephnie | hehe.. |
| 16:11.02 | coppice | its beyond the precision they handle, I think. they seem to use single precision instead of double |
| 16:11.17 | SuPrSluG | really depends on concurrent calls. a call center will need more because the idea is to have everyone talking to customers all the time. |
| 16:12.15 | *** join/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com) |
| 16:12.18 | paryl | SuPrSluG: this is a call center, so i guess as much as i can pack into a box :) |
| 16:12.44 | *** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com) |
| 16:12.45 | Stephnie | coppice: any solution? |
| 16:12.45 | TheCops | Hi |
| 16:13.01 | SuPrSluG | they have a page on the wiki called asterisk at large where they try to anser some of these questions |
| 16:13.11 | TheCops | When I'm pressing # key during a call from a SIP client, I get asterisk say "Transfer", how the hell I can remove that ? |
| 16:13.45 | coppice | Stephnie: complain, and get them to use doubles |
| 16:14.03 | SuPrSluG | next thing to ask yourself is will i be using channel banks or doing it in software |
| 16:14.06 | paryl | another question: we have ~6 analog lines that are connected to fax machines. currently, our pbx has an 'analog' card that allows for regular modem connections over the T1. how do i do this with asterisk? |
| 16:14.19 | Stephnie | ahh...long procedure :) |
| 16:15.03 | anthm | TheCops, dont use the T or t flag or edit features.conf and make all the codes multikey combos not likely to be dialed |
| 16:15.10 | konrads | paryl: install analogue adapter :) |
| 16:15.15 | konrads | paryl: digium has FXO ports |
| 16:15.21 | konrads | paryl: so do some other. |
| 16:15.40 | konrads | paryl: odd that you do not use ISDN |
| 16:16.30 | *** join/#asterisk TheTeddy (n=bay_dogr@85.102.200.59) |
| 16:16.30 | TheCops | hrmmm anthm, this is the blindxfer ? |
| 16:16.31 | *** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg) |
| 16:17.23 | anthm | yes |
| 16:18.01 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 16:18.02 | paryl | konrads: awesome... so the FXO cards are the equivilent to their 'analog card'? |
| 16:19.09 | TheCops | anthm, I put blindxfer => ## and restarted asterisk, and the # still transfering |
| 16:19.17 | SuPrSluG | paryl. yes. look for tdm04b |
| 16:20.02 | anthm | try making it 77 or *1 or something there is likely a bug in it |
| 16:20.20 | SuPrSluG | the 04 designate the number of xfs/fxo modules. a 22 is 2 fxs/2fxo |
| 16:20.50 | TheCops | anthm same thing, hehe let me read on google, thanks for giving me that way ;) |
| 16:21.06 | anthm | np |
| 16:21.55 | paryl | so am i understanding correctly that the card here: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P can handle a total of 4 analog lines? |
| 16:22.04 | eKo1 | yes |
| 16:22.31 | konrads | paryl: yes. Make sure you get the FXS and FXO right |
| 16:22.59 | konrads | paryl: FXS is where the phone plugs in and FXO is where the phone line goes in |
| 16:23.10 | konrads | FXO for dial-out and FXS for phone plugging in |
| 16:23.41 | paryl | wait |
| 16:23.59 | paryl | but FXO can have incoming calls... right? |
| 16:24.01 | fugitivo | ~fxsfxo |
| 16:24.03 | jbot | [fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 16:24.05 | *** join/#asterisk Dolunay___ (n=kalamar@85.100.45.56) |
| 16:24.15 | *** join/#asterisk Patefield (n=denizgoz@85.96.14.216) |
| 16:24.19 | paryl | so i need FXS ports then? |
| 16:24.28 | JerJer[dead] | paryl: do you receive calls on your normal telephone line? from the legacy telco |
| 16:24.38 | fugitivo | if you need to provide dialtone, you need fxs, if you need to receive dialtone, you need fxo |
| 16:24.47 | denon | it's alive! |
| 16:24.55 | konrads | paryl: if you want to plug analogue line to the PSTN : you need FXO |
| 16:25.06 | konrads | if you want to plug a phone or faxmachine into asterisk, you need FXS |
| 16:25.07 | paryl | i need to provide dial tone to fax machines... receive/send phone calls across the T1 |
| 16:25.12 | *** join/#asterisk mavi`` (n=CrAzYbOy@81.215.165.85) |
| 16:25.18 | JerJer | then you need an FXS device |
| 16:25.23 | konrads | paryl: you need FXS |
| 16:25.25 | paryl | gotcha.. cool |
| 16:25.27 | denon | sup jer |
| 16:25.39 | paryl | are there any cards that handle more than 4? |
| 16:25.46 | denon | paryl: get a channelbank |
| 16:26.13 | spackle | paryl: it may help to think of FXS as S_upplying Dial tone. |
| 16:27.02 | paryl | denon: is that hardware or software? |
| 16:27.50 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj3u.dialup.mindspring.com) |
| 16:28.14 | paryl | o_O |
| 16:29.20 | *** join/#asterisk XTR (n=xtr@staff-nat.netnation.com) |
| 16:31.50 | SuPrSluG | paryl:sound like you'll need both. since you have 6 seperate analog lines outside the t1. is that your setup? |
| 16:32.19 | *** join/#asterisk ixx (i=foobar@cpe-24-27-44-163.austin.res.rr.com) |
| 16:32.49 | *** join/#asterisk seymen21 (n=A-K-I-N@85.99.90.49) |
| 16:33.32 | paryl | SuPrSluG: i don't know what you mean by 'outside the t1'. all incoming lines (a block of 100 numbers) are coming in across a T1. I need to provide dialtone to my fax machines for sending/receiving |
| 16:33.40 | spackle | should this work to record the channel as conf+datetime ? Monitor(wav,/var/spool/asterisk/monitor/conf${datetime},m) All I get is conf.wav is there another way to concatenate? |
| 16:33.41 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 16:33.52 | *** join/#asterisk jontow (i=jontow@ws.woflsys.net) |
| 16:33.58 | paryl | so it sounds like i just need two tdm40b cards? |
| 16:34.03 | Ayano | Where can I find a list of voip providers? |
| 16:34.40 | eKo1 | paryl: is the T1 connected to a channel bank? |
| 16:35.05 | fugitivo | any ip phone recommendation? |
| 16:35.06 | fugitivo | polycom? |
| 16:35.11 | Ayano | ip500 |
| 16:35.28 | fugitivo | money? |
| 16:35.37 | TheCops | anthm, god, it is very hard to make working features.conf, I put the right syntax and it seem to dont work |
| 16:35.37 | SuPrSluG | 200ish |
| 16:35.46 | *** join/#asterisk ^^__AvUkAt__^^ (n=VictoR@81.214.230.229) |
| 16:36.01 | paryl | eKo1: i don't know. :\ it comes in to an adtran box, and then goes straight into the pbx. |
| 16:36.02 | *** join/#asterisk oxcen (n=info@196.206.241.196) |
| 16:36.05 | Ayano | ip300 is about 150 I think |
| 16:36.10 | JerJer | blasphemy - thou shalt only have one god |
| 16:36.31 | fugitivo | thanks |
| 16:36.32 | SuPrSluG | paryl:that's a channel bank |
| 16:36.37 | fugitivo | it works ok with asterisk? |
| 16:36.43 | Ayano | yep |
| 16:36.55 | paryl | SuPrSluG: what's a channel bank? the adtran? |
| 16:36.59 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:37.01 | SuPrSluG | yes |
| 16:37.12 | JerJer | smells like someone needs to do some of his own research |
| 16:37.18 | SuPrSluG | go to the wiki |
| 16:37.21 | eKo1 | no kidding |
| 16:37.43 | *** join/#asterisk tq1 (n=pedro@200.117.234.254) |
| 16:37.45 | tq1 | nas |
| 16:37.48 | SuPrSluG | or google asterisk + channel banks |
| 16:37.49 | tq1 | alguien habla español? |
| 16:37.53 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 16:37.54 | eKo1 | si |
| 16:38.00 | JerJer | nien |
| 16:38.08 | tq1 | hola eKo1 |
| 16:38.12 | oxcen | hello there. does asterisk+digium TD400 support fax reception and emission ? |
| 16:38.23 | ScaredyCat | emmision! |
| 16:38.33 | eKo1 | you mean sending |
| 16:38.36 | fugitivo | oxcen: yes, if you don't have noise in the line |
| 16:38.40 | oxcen | yes sending |
| 16:39.01 | SuPrSluG | paryl:lots of info on the wiki as to what you want to do. |
| 16:39.02 | JerJer | only after you get the required California emission test |
| 16:39.03 | tq1 | eko1 has oido algo de asterisk en español (un proyecto de un mexicano, segun me cuentan) |
| 16:39.15 | eKo1 | JerJer: hehe |
| 16:40.13 | *** join/#asterisk ^^ezgisu^^ (n=_Agresif@85.97.168.171) |
| 16:40.19 | eKo1 | tq1: ¿huh? Asterisk es Asterisk en todo idioma. |
| 16:40.57 | doolph | uh uh |
| 16:41.27 | oxcen | how does asterisk save the received faxes ? |
| 16:41.53 | eKo1 | raw audio? |
| 16:42.14 | *** join/#asterisk ixx_ (i=foobar@cpe-24-27-44-163.austin.res.rr.com) |
| 16:42.22 | fugitivo | eKo1: asterisco :) |
| 16:43.24 | *** join/#asterisk Derkommissar (n=alberto@66.64.215.6.nw.nuvox.net) |
| 16:43.26 | Derkommissar | hello |
| 16:43.31 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 16:43.31 | Derkommissar | i have a sangoma card. |
| 16:43.39 | Derkommissar | and when i do an ztcfg i get this error |
| 16:43.40 | Derkommissar | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 16:43.47 | Derkommissar | i belive i installed everything rigth |
| 16:43.56 | Derkommissar | and configured it too |
| 16:44.13 | doolph | no you are not |
| 16:44.22 | doolph | check your config |
| 16:44.23 | Derkommissar | why not ? |
| 16:44.39 | Derkommissar | zaptel |
| 16:44.41 | Derkommissar | span=1,0,0,esf,b8zs |
| 16:44.42 | Derkommissar | bchan=1-23 |
| 16:44.42 | Derkommissar | dchan=24 |
| 16:44.46 | *** join/#asterisk vefas (n=G_O_K_H_@85.97.168.171) |
| 16:45.05 | Derkommissar | the module is there |
| 16:45.08 | Derkommissar | lsmod shows zaptel 208900 2 wct1xxp,wanpipe |
| 16:45.43 | Derkommissar | i mean what else can it be ? |
| 16:46.06 | spackle | Derkommissar> what distro are you running? have the devices been created or are they in udev? |
| 16:46.07 | eKo1 | try to reboot |
| 16:46.38 | Derkommissar | its fedora core 3 |
| 16:46.48 | stkn | Derkommissar: check your /dev/zap device nodes |
| 16:46.51 | spackle | Derkommissar> so udev |
| 16:46.58 | ManxPower | Derkommissar, ztcfg reads /etc/zapata.conf |
| 16:47.14 | ManxPower | sorry |
| 16:47.38 | ManxPower | ztcfg reads /etc/zaptel.conf and asterisk reads /etc/asterisk/zapata.conf |
| 16:48.03 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
| 16:48.23 | Derkommissar | udev ? /dev/zap is a directory what should i check in it ? |
| 16:48.48 | Ayano | What is a good voip provider that can provide dids all over? |
| 16:48.56 | *** join/#asterisk alexis101 (n=alexis@toronto-HSE-ppp4327833.sympatico.ca) |
| 16:49.30 | stkn | Derkommissar: any device nodes in there? |
| 16:49.38 | *** join/#asterisk IOscanner (n=IOscanne@c-67-166-160-64.hsd1.tx.comcast.net) |
| 16:50.34 | Derkommissar | how can i check the device nodes ? |
| 16:51.27 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:51.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:51.47 | ManxPower | Ayano, they all suck. Teliax seems to usually suck less. |
| 16:52.11 | Ayano | lol |
| 16:52.18 | Ayano | Thank you |
| 16:52.22 | ManxPower | Nufone also sucks less, but they don't have DIDs everywhere, just Mich DIDs and Toll Free DIDs |
| 16:52.25 | eKo1 | I don't get what the purpose of having multiple proxy setting in xten-xlite is. I mean, calls will always go through the default one. |
| 16:52.45 | Ayano | unless the primary one is down |
| 16:52.48 | denon | eKo1: redundancy, in case one sip gateway is down? |
| 16:53.10 | eKo1 | i thought I could switch which proxy i wanted to use. |
| 16:54.36 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 16:55.27 | *** join/#asterisk kalbin_zumrut_te (n=TEKiN@85.99.90.49) |
| 16:55.31 | stkn | Derkommissar: channel ctl timer pseudo should be in /dev/zap |
| 16:55.42 | Derkommissar | yes |
| 16:55.43 | Derkommissar | they are |
| 16:56.07 | Derkommissar | i did the thing in the udev folder, like the instructions say for kernel 2.6 |
| 16:56.19 | *** join/#asterisk astadmin (n=shafqat@pk-isb-trg-sc01-019.speedcast.com) |
| 16:56.35 | Derkommissar | /dev/zap/ |
| 16:56.35 | Derkommissar | channel ctl pseudo timer |
| 16:57.19 | *** join/#asterisk akrall_ (i=user@201.144.58.186) |
| 16:57.31 | stkn | wanpipe module is loaded? |
| 16:57.33 | *** join/#asterisk mike_jh (n=mike@81.187.90.205) |
| 16:57.41 | Derkommissar | yes |
| 16:57.57 | Derkommissar | wanpipe 721884 0 |
| 16:57.57 | akrall_ | Has anybody writeen a php script to connect via sockets to Asterisk Manager and parse the output of command to show on screen? |
| 16:58.07 | Derkommissar | wanpipe_syncppp 24608 1 wanpipe |
| 16:58.20 | Derkommissar | wanrouter 31688 4 wanpipe_lip,af_wanpipe,wanpipe,wanpipe_syncppp |
| 16:58.30 | Derkommissar | with all the zaptel modules and all the jazz |
| 16:59.12 | Derkommissar | zaptel 208900 2 wct1xxp,wanpipe |
| 16:59.20 | eKo1 | akrall_: why, so they can be displayed on a website or what not? |
| 17:00.00 | *** join/#asterisk Desombre (n=Sohbetim@85.99.90.49) |
| 17:00.09 | akrall_ | eKo1: exactly |
| 17:00.23 | eKo1 | that's disgusting |
| 17:00.59 | akrall_ | the manager returns a lot of lines before and after the actual command output so you need to flush those lines before you can do some explode or something and use them.. |
| 17:01.11 | akrall_ | eKo1: why? |
| 17:01.21 | Derkommissar | :-/ |
| 17:02.10 | eKo1 | because i'm sick and tired of porting everything to the web |
| 17:02.27 | stkn | Derkommissar: any error messages in the dmesg output? |
| 17:03.45 | Derkommissar | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 17:03.50 | Derkommissar | is the only message i get |
| 17:03.57 | Derkommissar | when i run ztcfg |
| 17:04.19 | Derkommissar | no errors in dmesg |
| 17:04.21 | Derkommissar | WANPIPE(tm) L.I.P Network Layer Stable 2.3.2-4 (c) 1995-2004 Sangoma Technologies Inc. |
| 17:04.21 | Derkommissar | WanpipeLIP: Protocols: FR PPP CHDLC |
| 17:05.04 | SuPrSluG | is it showing in lsmod? |
| 17:05.20 | SuPrSluG | zaptel that is |
| 17:05.54 | kusznir | Hello all: I'm having trouble placing toll-free calls through iaxtel. My console shows: |
| 17:05.54 | kusznir | <PROTECTED> |
| 17:05.54 | kusznir | Sep 6 10:00:41 NOTICE[25146]: chan_iax2.c:2742 auto_congest: Auto-congesting call due to slow response |
| 17:06.07 | file | iaxtel is down, it usually is |
| 17:06.17 | kusznir | the demo iax call to digium works. |
| 17:06.19 | kusznir | Ahh..ok. |
| 17:06.26 | file | it can't handle the load. |
| 17:06.33 | akrall_ | eKo1: the web is the future hahahhahahahaha |
| 17:06.52 | akrall_ | I personally love CLI's but the end user.. thats another story |
| 17:07.15 | Hmmhesays | it would be nice in asterisk to be able to set multiple variables with one cmd set |
| 17:07.34 | file | hot-n-sexy Hmmhesays, makes my world go 'round |
| 17:07.52 | Hmmhesays | woooo! |
| 17:08.06 | Hmmhesays | cause if I want to set like 9 variables it is a serious pain |
| 17:08.07 | file | what evil stuff are you up to? |
| 17:08.13 | kusznir | file: does asterlink charge for outbound calls to tollfree numbers? |
| 17:08.32 | Hmmhesays | file: just creating a dialplan for an ISP here |
| 17:08.40 | file | kusznir: I think so |
| 17:08.51 | kusznir | ok. |
| 17:08.57 | Hmmhesays | the ldap integration isn't ready yet so i'm using variables to indicate things like if the DID is enabled or not |
| 17:09.07 | *** join/#asterisk razu_ (n=razu@ip192.cab63.mus.starman.ee) |
| 17:09.11 | file | go for FWD |
| 17:09.17 | malverian[work] | astxs rocks.. |
| 17:09.23 | ManxPower | IAXtel goes down more often than a Venice Beach Babe |
| 17:09.36 | pifiu | im hating voip right now |
| 17:09.43 | pifiu | man i wish normal dsl connections had more upload |
| 17:09.43 | Hmmhesays | how do I get there? |
| 17:09.53 | Hmmhesays | er... |
| 17:10.14 | denon | pifiu: g.729 :) |
| 17:10.34 | file | everybody except the person I need to talk to is online |
| 17:11.02 | Guggemand | anyone know of a numberlist of the numbers that are free at voipbuster ? |
| 17:11.10 | *** join/#asterisk BuckRogers (n=steve@ool-44c29ac5.dyn.optonline.net) |
| 17:11.14 | BuckRogers | hello all |
| 17:11.24 | BuckRogers | anyone here use SER with asterisk |
| 17:11.43 | Hmmhesays | i'm in the process of it |
| 17:11.51 | *** join/#asterisk I0scanner (n=IOscanne@c-67-166-249-43.hsd1.tx.comcast.net) |
| 17:11.51 | Hmmhesays | should be up and running in 2 weeks |
| 17:11.56 | BuckRogers | yeah how is it going for you |
| 17:12.03 | Hmmhesays | tain't bad |
| 17:12.32 | BuckRogers | yeah we've been fine tuning * for a large deployment |
| 17:12.41 | Hmmhesays | how large? |
| 17:12.53 | BuckRogers | but the more ive been reading the more that i think ser is nessary |
| 17:13.04 | Hmmhesays | i'm limiting my calls to 120 per dual xeon box, so there isn't really much tuning to be done |
| 17:13.30 | BuckRogers | well i need to start with the ability to do 96 concurrent but we need to expand that over time |
| 17:13.41 | BuckRogers | 120 with or with out zap |
| 17:13.45 | Hmmhesays | without zap |
| 17:13.52 | BuckRogers | yeah same here |
| 17:14.05 | Hmmhesays | got an external gateway that can handle 29 t1's |
| 17:14.29 | BuckRogers | our provider is doing the transcoding when nessary |
| 17:14.39 | BuckRogers | from sip to pstn |
| 17:14.50 | Hmmhesays | so as soon as the call capacity has to be more than 120 then another asterisk box will be added |
| 17:14.57 | lehel | ppl: is this correct for a Dial command: ...{EXTEN:1},20,r,t) ??? |
| 17:15.04 | *** join/#asterisk paulc (n=paulc@S010600062586a0b4.vc.shawcable.net) |
| 17:15.05 | Hmmhesays | and 5 t1's will be allocated to that box |
| 17:15.14 | file | dear god it's paulc |
| 17:15.20 | paulc | It is it is! :-) |
| 17:15.20 | Hmmhesays | 20,rt |
| 17:15.24 | BuckRogers | what type of proccessor load are you looking at with the dual xeon at full load |
| 17:15.38 | JerJer | 1000 bogo mips |
| 17:15.52 | Hmmhesays | haven't turned it up yet in production, find out on friday though |
| 17:15.52 | lehel | Hmmhesays: or 20,rT ?? ok? |
| 17:16.06 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 17:16.15 | Hmmhesays | lehel, rT is different than rt |
| 17:16.17 | file | paulc: you pesky bugger! |
| 17:16.24 | BuckRogers | so you either celebrate over the weakend or work through it right :) |
| 17:16.28 | paulc | OI! LESS OF YOUR LIP YOUNG MAN! :-p |
| 17:16.34 | lehel | Hmmhesays: i know.. 't' -- allow the called user to transfer the calling user by hitting #. |
| 17:16.34 | lehel | <PROTECTED> |
| 17:16.35 | paulc | don't make me set your amish mother on you |
| 17:16.37 | file | paulc: Jan Polet :P |
| 17:16.39 | Hmmhesays | BuckRogers: from the research i've done there will not be a problem |
| 17:16.40 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 17:16.46 | paulc | file: LMAO |
| 17:16.53 | paulc | file: No, it's JAAAAN POOLLLEEEEE-EEETTTTTT! |
| 17:16.59 | JerJer | don't use 'r', 't' or 'T' unless you most absolutely have to |
| 17:17.00 | file | with Jan Polet! |
| 17:17.02 | BuckRogers | yeah every thing that ive read lead me to the same conclution |
| 17:17.05 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 17:17.10 | Hmmhesays | yeah r fscks a lot of things up |
| 17:17.29 | BuckRogers | 'r'? |
| 17:17.40 | Hmmhesays | the indicate ringing to the calling party flag |
| 17:17.56 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 17:18.08 | Hmmhesays | if you are having ringing problems it is better just to answer the call first and then send it to dial |
| 17:18.13 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 17:18.23 | lehel | what means this: {EXTEN:1} |
| 17:18.29 | lehel | ":1" ? |
| 17:18.32 | Sedorox | Say.... has anyone worked with Vina Integrators at all?? know a default password.. or a way for me to get into it? |
| 17:18.35 | Hmmhesays | remove the leading digit |
| 17:18.40 | BuckRogers | yeah we are pretty much going to be using astrisk to act as a bolt on for exisitng pbx's then connect them to our main * for voip terminatino |
| 17:18.41 | paulc | lehel: It means strip 1 digit off the front |
| 17:18.51 | lehel | ok! thanks |
| 17:18.52 | stkn | Derkommissar: did you build wanpipe with tdm voice support? |
| 17:18.57 | *** join/#asterisk pauldy (n=pauldy@c-67-173-198-40.hsd1.tx.comcast.net) |
| 17:19.17 | *** part/#asterisk intrepidhero (n=briand@67.189.59.49) |
| 17:19.18 | Hmmhesays | ok, I see a design fault in my dialplan, crap |
| 17:19.23 | ManxPower | We had three different people working on getting the phone lines back up today, all doing it in totally different and conflicting ways. |
| 17:19.34 | Hmmhesays | larry moe and curly? |
| 17:19.41 | BuckRogers | haha right on |
| 17:19.49 | ManxPower | Hmmhesays, something like that. |
| 17:20.02 | paulc | if you don't specify qualify=no, it's off by default right? |
| 17:20.11 | JerJer | yuk yuk yuk |
| 17:20.12 | BuckRogers | any of you guys work with isphone here |
| 17:20.47 | BuckRogers | Hey (jer jer) how is the 911 issue working out for you |
| 17:20.59 | *** join/#asterisk file[muon] (n=file@mctnnbsa30w-156034035106.nb.aliant.net) |
| 17:21.20 | Sedorox | anyway? vina default password? :p |
| 17:21.48 | JerJer | BuckRogers: there is no 911 issue for us |
| 17:21.50 | lehel | Hmmhesays: i still can't transfer the call:(.. what could be the problem?. i have now: ...20,rt) |
| 17:22.05 | JerJer | buy a phone with a transfer button |
| 17:22.15 | Hmmhesays | make sure you have transfers enabled in features.conf |
| 17:22.23 | konrads | But we just bought 200 wihtout the button! |
| 17:22.32 | JerJer | sucks to be you |
| 17:22.42 | file[muon] | paulc: okay puppy I'll call you when I'm finished :P |
| 17:22.42 | ManxPower | Yeah. Next time do more research. |
| 17:22.43 | file[muon] | lol |
| 17:22.44 | BuckRogers | o you got it all sorted out who did you go to for it, Introdo? |
| 17:23.06 | paulc | file: bzzzzt that joke's older than yer mum |
| 17:23.41 | konrads | j/k :) |
| 17:24.04 | lehel | Hmmhesays: pls have look at my features.conf .. |
| 17:24.09 | lehel | http://paste.debian.net/1815 |
| 17:24.39 | BuckRogers | (Hmmhesays) what type of deployment are you going for, commercial reseller or for one enterprise |
| 17:24.46 | lehel | i can't see anywhere about transferring calls.. |
| 17:24.52 | *** join/#asterisk kg (n=kg@chello062179062077.chello.pl) |
| 17:25.00 | BuckRogers | like are you going to be a service provider or more of a PBX installation |
| 17:25.03 | Hmmhesays | BuckRogers, commerical reseller |
| 17:25.18 | BuckRogers | we are in the same boat been developing for a year now |
| 17:25.20 | lehel | it is the same to parking calls.. maybe, but still nothing about enabled or disabled |
| 17:25.22 | Hmmhesays | they are going to offer phone service in conjuction with their current tv and internet service |
| 17:25.24 | file[muon] | why is authentication not occuring argh |
| 17:25.34 | DaPrivateer | can anyone recommand a SIP wifi phone that supports WPA? |
| 17:25.45 | Hmmhesays | BuckRogers, they gave me 2 weeks to light up the first site |
| 17:25.54 | BuckRogers | we are going for Small medium buisness market |
| 17:26.09 | lehel | Hmmhesays: http://paste.debian.net/1815 < pls have a look (features.conf) |
| 17:26.20 | BuckRogers | nice how is it looking for you? |
| 17:26.23 | file[muon] | this LCD is so very very clear |
| 17:26.26 | *** part/#asterisk akrall_ (i=user@201.144.58.186) |
| 17:26.28 | Hmmhesays | lehel that features.conf is missing some stuff |
| 17:26.45 | Hmmhesays | BuckRogers, the thing here is, these customers won't know they are on an ip system |
| 17:26.50 | lehel | hrmm.. |
| 17:26.53 | BuckRogers | im hopeing to get up and launched in 30-60 days, still negoatiating pricing with wholesalers |
| 17:26.58 | Hmmhesays | each sight is getting 1 or more 24 port gateways punched down in the phone room |
| 17:27.15 | ManxPower | There are like 50 billion providers of VoIP outgoing. |
| 17:27.31 | JerJer | no 51 billion |
| 17:27.36 | mutilator | 52* |
| 17:27.37 | ManxPower | And 5 providers of incoming non-toll free DIDs with a national footprint |
| 17:28.08 | BuckRogers | hmmhesays, how will they not know wont they have to use some type of a ata instead of Traditional phone line? |
| 17:28.56 | paulc | and few of those providers service the great white north :( |
| 17:29.23 | devel | huh. is there a way to factory reset a grandstream bt100? i have one that's gone stupid.... |
| 17:29.26 | Hmmhesays | they customers are all in apartment buildings |
| 17:29.30 | BuckRogers | yeah there are a bunch but most of them are reselling others stuff, the question is who;s got the quility and the right price and a 911 solution that doesnt cost a arm and a leg |
| 17:29.53 | paulc | devel: Go to menu, then reset, but type in the MAC address before you hit the select/ok/confirm button |
| 17:30.13 | BuckRogers | yeah we have been looking into getting in bed with a couple of the senor condo complexes that are going up like wild fire over here |
| 17:30.17 | devel | thanks, paulc, i'll give that a try |
| 17:30.28 | Hmmhesays | BuckRogers, 1. the gateways will be in the phone room, each apartment will have dialtone at the jacks. 2. these guys are terminating their own traffic |
| 17:30.29 | file[muon] | paulc knows too much |
| 17:30.36 | file[muon] | you will find me... time after time! |
| 17:30.38 | *** join/#asterisk simprix (n=simprix@24-231-248-225.static.aldl.mi.charter.com) |
| 17:30.39 | Hmmhesays | or not enough |
| 17:30.59 | Derkommissar | to dial trough a zap card.. i just have to use dial like ,Dial(Zap/${EXTEN} ? |
| 17:31.09 | simprix | What is the easist voip phone to setup with asterisk, I am looking at polycom |
| 17:31.15 | ManxPower | BuckRogers, Um, they are all reselling someone's service, even if it's their CLEC's service |
| 17:31.16 | paulc | file: but do you know any of the songs in MY hit test?! ;-) |
| 17:31.17 | BuckRogers | is the appartment managment have sometype of evergreen deal with you? |
| 17:31.54 | BuckRogers | in other words they get a piece of the monthly pie |
| 17:32.04 | ManxPower | The good outbound / inbound toll free providers all have PRIs. |
| 17:32.06 | file[muon] | paulc: I think I know the first one |
| 17:32.21 | Hmmhesays | BuckRogers, its an odd deal, this isp owns the apartments |
| 17:32.34 | *** join/#asterisk zoo (i=nobody@ip-11-16.travedsl.de) |
| 17:32.35 | BuckRogers | really that is an odd deal |
| 17:32.38 | Hmmhesays | 400 of them to be exact |
| 17:32.44 | file[muon] | why is this immediately throwing back a user is not authorized... |
| 17:33.32 | BuckRogers | ive never heard of an isp owing appartments, not a bad deal though they kind of have a mini monoply that way |
| 17:34.02 | mutilator | we do the same thing |
| 17:34.14 | *** part/#asterisk Morex (n=blah@host81-157-123-58.range81-157.btcentralplus.com) |
| 17:34.21 | Hmmhesays | BuckRogers, yeah they do, they also have a p2p networks between most of them |
| 17:34.26 | Hmmhesays | so they can garuantee qos |
| 17:34.30 | ManxPower | It really ruins your day when the president of the company calls you EVERY FUcKING 10 MINS wanting to know why the phone lines are not working. The answer is of course, THE DAMN CITY WAS LEVELED BY A HURRICANE. |
| 17:34.33 | ManxPower | </vent> |
| 17:35.00 | Hmmhesays | get direcway, port the numbers over to a voip provider |
| 17:35.01 | file[muon] | aha... |
| 17:35.02 | BuckRogers | ahh man what the #@#@# can you do in that situation |
| 17:35.04 | Hmmhesays | you have 10 seconds |
| 17:35.22 | ManxPower | BuckRogers, We had voice and data running by 9:30am. |
| 17:35.43 | ManxPower | paulc, I finally gave him the phone number of the NOC for our CLEC. |
| 17:36.05 | paulc | passing the buck! which gets him off your back.. at the risk of wrath from the NOC ;-) |
| 17:36.34 | ManxPower | paulc, No. I started answering the MIS manager's cell phone (we were on the road on the way to the office) and told him that reason I was answering the MIS manager's cell phone was "so we don't both die in a firey ball of death on the freeway" |
| 17:36.41 | *** join/#asterisk meppl (n=mephisto@p54AAE85A.dip.t-dialin.net) |
| 17:37.25 | simprix | How well does the polycom 301 work with asterisk |
| 17:37.30 | paulc | LOL.. funny how some people "just don't get it" |
| 17:37.32 | paulc | I can relate.. |
| 17:37.40 | paulc | simprix: Pretty well, although the 501/600 work better :) |
| 17:37.40 | luke-jr__ | Is a "50 pair line" == T1? |
| 17:37.41 | BuckRogers | Yah I would worry about your health too ManxPower |
| 17:37.56 | BuckRogers | screw the guy and get thier in one piece |
| 17:38.02 | paulc | luke-jr__: No.. a T1 is usually 2 copper pairs.. and a T1 = 24 pairs if your talking POTS |
| 17:38.10 | simprix | paulc: how is configuration |
| 17:38.17 | *** join/#asterisk FullyFaltoo (n=eDi_LaW@81.215.165.85) |
| 17:38.25 | luke-jr__ | paulc: hmm... any idea what a 50 pair would be? o.O |
| 17:38.40 | paulc | simprix: If you do it from the web interface it's "alright", if you do it from the XML config files via FTP it's a joy, mostly.. |
| 17:39.30 | paulc | luke-jr__: 50 pair cable doesn't really translate in to T1s or DS3s etc.. it's more like "here's a cable that you can connect 50 regular phones and/or lines to" (1 pair per line/device). Usually used at a punch down/demarc kinda deal.. or maybe a connection to a PBX that has line cards in odd sizes.. |
| 17:39.31 | simprix | if i am installing a pri what is a good phone to get or will asterisk just pick a line our of the trunk, or do i need a phone that will have that many line buttons |
| 17:39.32 | devel | thanks, paulc, that did the trick. my phone is still doing stupid though. i'll try firmware |
| 17:40.02 | paulc | simprix: Asterisk will do the clever stuff in the middle, choosing a line etc.. forget "line buttons" - asterisk is a PBX not a key system :) |
| 17:40.04 | file[muon] | paulc: is the first one Usher? |
| 17:40.14 | paulc | file: nope - the artists all begin with the letter A |
| 17:40.20 | file[muon] | bastard :P |
| 17:40.42 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 17:40.42 | *** mode/#asterisk [+o twisted] by ChanServ |
| 17:40.48 | Sedorox | anyone know the Vina Tech T1 integrator passowrd??? *puppy dog eyes* |
| 17:40.49 | visik7 | how can I write in the dialplan the 'R' key ? |
| 17:41.07 | ScaredyCat | it's flashhook... you don;t |
| 17:41.13 | PupenoL | Is it possible to make the Digium bug tracking system CC my mails to another account ? |
| 17:42.42 | paulc | file: ah yes, I can see why you'd think track 1 was Usher.. similar style.. but sadly not.. think UK act ;-) |
| 17:43.03 | file[muon] | I don't acknowledge the existence of the UK :P |
| 17:43.06 | ScaredyCat | ash |
| 17:43.12 | Ash | ash |
| 17:43.17 | ScaredyCat | ash |
| 17:43.20 | Ash | ash |
| 17:43.20 | ScaredyCat | ;) |
| 17:43.26 | paulc | Brim full of asher on the 45? |
| 17:43.29 | Ash | whoareyes? |
| 17:43.33 | Ash | who are you even |
| 17:43.33 | Ash | heh |
| 17:43.50 | file[muon] | brb |
| 17:44.35 | ScaredyCat | who are any of us... |
| 17:45.00 | mutilator | i am me |
| 17:45.06 | file[muon] | I'm file |
| 17:45.09 | file[muon] | nice to meet you. |
| 17:45.15 | mutilator | i r baboon |
| 17:45.27 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 17:45.42 | sigterm | im term, nine to meet you |
| 17:45.57 | file[muon] | paulc: shouldn't you be working? |
| 17:45.59 | file[muon] | :P |
| 17:46.27 | paulc | file: I AM working |
| 17:46.37 | file[muon] | uh huh |
| 17:46.43 | ScaredyCat | ./.s |
| 17:47.19 | paulc | file: I'll prove it with a work related question! |
| 17:47.24 | *** join/#asterisk subay_39 (n=EsS_SaBu@81.215.165.85) |
| 17:47.52 | paulc | I've got qualify=no in sip.conf and the phones are set to register every 60 seconds.. yet still sometimes a call destined for a phone goes straight to voicemail rather than ringing the phone.. can I blame the NAT on the router for expiring sessions too quickly? |
| 17:48.01 | file[muon] | yes. |
| 17:48.03 | file[muon] | :P |
| 17:48.26 | paulc | but.. seriously.. |
| 17:48.28 | ManxPower | paulc, I don't know of any NAT routers that expire sessions in under 60 seconds, but I guess they could exist. |
| 17:48.35 | file[muon] | they do exist |
| 17:48.37 | file[muon] | they're evil |
| 17:49.27 | file[muon] | parser parser do your parsing |
| 17:49.54 | TheCops | lol |
| 17:51.41 | lehel | ppl: ??> Sep 6 20:47:57 WARNING[31560]: cdr.c:421 ast_cdr_free: CDR on channel 'Zap/3-1' not posted |
| 17:52.19 | eKo1 | that sucks |
| 17:52.54 | lehel | eKo1: do you mean? i have some problem in mysql? |
| 17:53.05 | cpatry | lehel: cause it cant connect to db or u had a NoCDR() before. |
| 17:54.11 | lehel | cpatry: and what for is that NoCDR() ? |
| 17:54.40 | file[muon] | cpatry: Junky!!! |
| 17:55.24 | ManxPower | file[muon], Can you tell use some makes/models of NAT routers the time out NAT translations in less than 60 seconds? |
| 17:55.34 | cpatry | lehel: show application nocdr will tell ya. |
| 17:55.37 | cpatry | yo file!!!! |
| 17:56.07 | file[muon] | ManxPower: I haven't dealt with them in a long time, but I think my friend had a D-Link that did it |
| 17:56.38 | paulc | Manx: This customer says he's got some kind of Cisco/Linksys router.. but it seems more Linksys than Cisco cos he uses a web interface to change settings.. no command line IOS type thing.. |
| 17:57.18 | *** join/#asterisk oplog7640 (n=oplog764@206.222.29.50) |
| 17:57.19 | *** join/#asterisk orospakr (n=orospakr@ip-95.84.126.206.dsl-cust.ca.inter.net) |
| 17:57.29 | ManxPower | paulc, Linksys routers do not time out NAT sessions in under 60 seconds. HOWEVER, many of the older verisons of the router simply crash and reboot when they see SIP registrations. |
| 17:58.04 | orospakr | hi! I know that g729 offers decent performance for road-warror sip clients on connected to the internet via dialup, but can speex offer the same? |
| 17:58.07 | ManxPower | I had like three of the linksys rev 1 routers that did that. Had to replace them all |
| 17:58.52 | paulc | I've got Linksys routers (albeit it consumer versions, nothing "big") at home and at work and they work fine.. but this customer's unit is a pain in the ass.. we'll try with a firmware upgrade then tell 'em to sling it and get a proper router.. |
| 17:59.12 | paulc | interesting thing is.. Polycom phones also don't get the correct time via SNTP behind this router.. and I know the config is good cos "works alright for me!" applies.. |
| 17:59.27 | ManxPower | paulc, look on the bottom of the router for the hardware rev. I'll bet you have a rev 2 or 3 and he has a rev 1 (might not have a rev listed) |
| 17:59.30 | paulc | they'll sync to the correct minute, but the hours is always off.. might flip right for a few mins, then resync and go wonky again.. customer's not impressed. . |
| 17:59.45 | paulc | Manx: Alright - I'll get 'em to check.. |
| 17:59.54 | ManxPower | paulc, check your's too. |
| 18:00.14 | file[muon] | stupid auth api |
| 18:00.17 | Wonka | aaargh. someday i'll hit anyone for saying "wonky"! |
| 18:00.36 | paulc | mine's a BEFW11S4 rev 2 :-) |
| 18:00.51 | Wonka | *yummy* |
| 18:01.14 | *** join/#asterisk dant (n=dan@81-86-69-213.dsl.pipex.com) |
| 18:01.26 | *** join/#asterisk _sLave_ (n=EXECUTIV@85.99.90.49) |
| 18:01.46 | ManxPower | paulc, *nod* And I'll bet his router has no rev listed. |
| 18:03.00 | *** join/#asterisk [ViRii] (n=virii@24-159-155-42.dhcp.smrt.tn.charter.com) |
| 18:03.12 | *** join/#asterisk vmlinuz (n=nabudoco@ns1.ensenada.gob.mx) |
| 18:03.17 | ManxPower | I'm having a vision! Oh god! The horror! The horror! In my vision I see that your customer has a Linksys BEFSR41 Rev 1! |
| 18:03.18 | [ViRii] | hey guys, im new to asterisk find it really amazing so far i can only call into my polycom i600 phones. i cannot call out? |
| 18:03.44 | ManxPower | [ViRii], that's all controled by the SIP phone. |
| 18:04.00 | sivana | ManxPower: were you down in NO? |
| 18:04.20 | ManxPower | sivana, Waveland MS. Apparently in the %10 of Waveland that was not leveled. |
| 18:04.33 | sivana | I hope all is well |
| 18:04.47 | ManxPower | We evacuated to Jackson MS, then Texarkana TX, then I spent a night in Lafayette LA, and tonight I'll be in Baton Route LA. |
| 18:05.05 | ManxPower | sivana, There's a good chance all my stuff sat in water for 2 days. |
| 18:05.12 | ManxPower | But I can't get back to check on it yet. |
| 18:05.19 | *** join/#asterisk RoyK (n=roy@cm-80.111.22.187.chello.no) |
| 18:05.19 | sivana | ya |
| 18:05.29 | *** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com) |
| 18:06.03 | ManxPower | [ViRii], your extensive search of the asterisk mailing lists about polycom configs did not help you? |
| 18:06.20 | ManxPower | sivana, there is still no power in my area. |
| 18:06.37 | sivana | ya.. that whole thing is strictly amazing |
| 18:07.22 | [ViRii] | yes i even used configs from krisk.org |
| 18:07.24 | ManxPower | I'm at my client in Covington LA at the moment. Apparently one of the accounting people called up and said "I'm not coming back." |
| 18:07.27 | [ViRii] | i will try again |
| 18:07.31 | ManxPower | I think 8 people lost their houses. |
| 18:07.52 | sivana | ya, amazing how a storm can wipe out a city |
| 18:07.56 | ManxPower | [ViRii], Did you use a web browser to config the phone, or did you use the FTP/TFTP method? |
| 18:08.26 | ManxPower | and what is your dialplan on the phone? |
| 18:09.02 | mutilator | when hookin my te110p directly to a channel bank |
| 18:09.07 | mutilator | what should my timing be? |
| 18:09.11 | mutilator | 0 or 1 or does it matter? |
| 18:09.14 | ManxPower | http://www.dilbert.com/ |
| 18:09.16 | *** join/#asterisk dsfr (n=dsfr@digium.com) |
| 18:09.24 | ManxPower | mutilator, timeing should be 0 |
| 18:09.25 | [ViRii] | webbrowser currently |
| 18:09.36 | ManxPower | and you'll need a T-1 cross over cable, of course. |
| 18:09.40 | mutilator | yea |
| 18:09.43 | [ViRii] | entirely new to this first asterisk install |
| 18:09.49 | mutilator | k just trying to figure out why i'm getting a fast busy |
| 18:09.56 | mutilator | they pickup the phone and it's busy |
| 18:10.19 | spackle | mank, shouldn't it be a '1' if he is supplying timin to the channel bank? |
| 18:10.31 | spackle | er, manx, shouldn't it be a '1' if he is supplying timin to the channel bank? |
| 18:11.24 | Sedorox | anyone use lucent/vina tech/connect reach channel bank? I need the default password.... |
| 18:11.31 | spackle | mutilator, do youhave the channels assigned on the bank and set up for loop start - or whatever you are using? |
| 18:11.32 | *** join/#asterisk roche (n=roche@216.194.173.11) |
| 18:11.40 | mutilator | loop yea |
| 18:11.48 | mutilator | dialing into it seems to work fine, it rings and they answer, but when they pickup to make a call it's busy |
| 18:12.37 | spackle | mutilator: what channel bank? |
| 18:12.45 | mutilator | adtran ta624 |
| 18:12.59 | *** join/#asterisk roulduke_ (i=u3l0f0nt@p508D2112.dip0.t-ipconnect.de) |
| 18:13.18 | ManxPower | spackle, no, 1 means get timeing from line, 2 means get timeing from line is span with "1" fails, 0 means don't get timeing from line. |
| 18:14.00 | *** join/#asterisk mariogamboa (n=sudaikdd@201.138.187.101) |
| 18:14.09 | ManxPower | mutilator, you didn't do something *stupid* like set immediate=yes, did you? |
| 18:14.54 | mariogamboa | hi all |
| 18:14.59 | roche | Hi, I'm having a strange behavoir with my asterisk, I create phone peer in sip.conf with incominglimit=1,but the extension can't transfer a call , it can't put the incoming call on hold and call to other extension, in extension.conf I have tT in dial option |
| 18:15.07 | mutilator | no |
| 18:15.24 | *** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com) |
| 18:15.24 | mariogamboa | anyone here have a problem with disconnect in conversion the call in fxo modules |
| 18:15.52 | roche | when I change the incominglimit=2 Its works , but it doesn't have sense |
| 18:16.20 | mariogamboa | when that call from pstn incoming to asterisk sometimes cut or when i dial sometimes cut the call |
| 18:16.23 | mutilator | i'm not having much luck with these te110's |
| 18:16.43 | ManxPower | mutilator, they work fine for me. |
| 18:17.26 | mutilator | heh well first i had problem was cause it was on a riser card |
| 18:17.26 | spackle | manxpower, that's how you explained the timing to me, but Digium support explained it differently: 0, meant to accept timing from the line, 1 meant to provide timing to the line, which you would want to do for a channel bank, right? |
| 18:17.31 | mutilator | so i was gettin pci master abort errors |
| 18:17.41 | Hmmhesays | i see voipjet has taken a crap all over itself |
| 18:17.45 | mutilator | that went on for a while til i replaced it |
| 18:17.55 | mutilator | then saturday it died on me again so i go out there and flick the power back on |
| 18:18.06 | mutilator | and the harddrive starts smoking and i hear a couple snap crackle pops |
| 18:18.16 | mariogamboa | i have my signaling fxo_ks and callprogress=yes progozone=us but the call is cut is some cases |
| 18:18.24 | *** join/#asterisk dasuberdavid (n=dasuberd@digium.com) |
| 18:18.25 | hardwire | anybody having quite a few dropped call problems? |
| 18:18.26 | ManxPower | spackle, what did they say "2" did? |
| 18:18.27 | hardwire | sip to zap |
| 18:18.28 | mariogamboa | sorry is fxs_ks |
| 18:18.31 | hardwire | uh oh |
| 18:18.35 | mariogamboa | me |
| 18:18.36 | ManxPower | mutilator, pci master abort errors are motherboard issues |
| 18:18.40 | mutilator | now i replaced it with another pc and it just died again on me, so i have to have someone go out there and flick it on again and find out whats going wrong |
| 18:18.51 | mariogamboa | i have droped problem with FXO connect to the pstn |
| 18:19.05 | mutilator | manx: it was brand new, had a 2 port riser card |
| 18:19.10 | ManxPower | callprogress=yes should be renamed randomlydisconnectmycalls=yes |
| 18:19.15 | *** part/#asterisk BuckRogers (n=steve@ool-44c29ac5.dyn.optonline.net) |
| 18:19.32 | ManxPower | mutilator, Well, yeah. Many newer motherboards don't work well with Digium cards. |
| 18:19.49 | ManxPower | And riser card ones can be the worst. Just look at the mailing list archives |
| 18:19.49 | mutilator | wonderufl |
| 18:20.00 | mutilator | well |
| 18:20.16 | mutilator | i need to get a 1u server that'll work with these damned things |
| 18:20.51 | spackle | Manxpower - we didn't get into that. I have my channel bank set up as '1', maybe it works as a result of dumb luck. |
| 18:21.14 | ManxPower | spackle, Well, it's all documented in the sample config files |
| 18:21.37 | mariogamboa | max the rename callprogress works in asterisk 1.0.9 |
| 18:22.13 | mariogamboa | what that mean randomlydisconnectmycall what is the function of it |
| 18:22.37 | spackle | Manxpower: it is documented, but clear as mud as I recall. |
| 18:22.59 | Hmmhesays | ~seen Ariel_ |
| 18:23.01 | jbot | ariel_ <n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net> was last seen on IRC in channel #asterisk, 3h 11m 55s ago, saying: 'RoyK, no'. |
| 18:23.01 | mariogamboa | in the zapata.conf |
| 18:23.11 | mariogamboa | i don't see this option |
| 18:23.13 | ManxPower | mariogamboa, it means exactly what the sample config files says: This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, |
| 18:23.13 | ManxPower | ; so don't count on it being very accurate. |
| 18:23.47 | ManxPower | mariogamboa, try /path/to/asterisk/src/asterisk/configs/zapata.conf.sample |
| 18:24.27 | ManxPower | I have never, in my 2+ years of working with Asterisk, ever heard of callprogress=yes working correctly and not causing problems. don't use it. |
| 18:24.53 | ManxPower | mutilator, ask JerJer what he uses. |
| 18:26.13 | spackle | manxpower: I'm going to change my timing around and see if anything breaks. |
| 18:26.16 | Hmmhesays | back to perlinating |
| 18:26.55 | *** join/#asterisk Tili (i=Tili@202-133-65-215-dialup.sat.net.pk) |
| 18:26.56 | ManxPower | spackle, incorrect timeing can cause fax issues, as well as audio artifacts like mild clicks |
| 18:27.02 | hardwire | mutilator: there are lots of cheap 1u's that work |
| 18:27.12 | mariogamboa | well max the problem here in mexico i can maitain the comunication sudenly the calls drop when the another person quiet asterisk detect like a silence and cut the call |
| 18:27.59 | hardwire | damnit |
| 18:28.03 | hardwire | so calls are just being dropped |
| 18:28.21 | file[muon] | hardwire: I'm wearing an Alaska t-shirt :P |
| 18:28.26 | spackle | Manxpower, that describes what I was experiencing that led me to investigate timing. |
| 18:28.33 | hardwire | file[muon]: why?! |
| 18:28.36 | hardwire | :) |
| 18:28.51 | file[muon] | my parents brought it back... from Alaska! |
| 18:28.56 | hardwire | no kidding |
| 18:29.09 | hardwire | they have them on discount in Canada.. so maybe it was from there :) |
| 18:29.22 | hardwire | well uh.. welcome to the fan club? |
| 18:29.57 | brad_mssw | anyone here have the UTStarcom F1000 WIFI phone ? |
| 18:30.37 | mutilator | .. |
| 18:30.46 | hardwire | Got a FRAME_CONTROL (5) frame on channel Zap/1-1 |
| 18:31.07 | hardwire | is that after the d-channel is processed.. it just represents it as Zap/1-1 ? |
| 18:32.17 | hardwire | it looks like a timing drop |
| 18:32.23 | hardwire | as per http://lists.digium.com/pipermail/asterisk-users/2005-April/101324.html |
| 18:32.33 | mariogamboa | max the randomly |
| 18:32.48 | mariogamboa | doesn't exist in zapata.conf.example |
| 18:33.31 | mariogamboa | in the case show me the callprogress no the randomly |
| 18:34.52 | hardwire | anybody have zttest doing all 100%? |
| 18:35.01 | hardwire | --- Results after 41 passes --- |
| 18:35.01 | hardwire | Best: 100.000000 -- Worst: 99.987793 |
| 18:35.34 | cpatry | hardwire: it doesnt matter if isnt at 100%. |
| 18:36.33 | hardwire | well why am I getting FRAME_CONTROL frames on b-channels? |
| 18:36.52 | Hmmhesays | anyone know if asterisk dialstatus is congestion when a 404 is returned from a sip endpoint? |
| 18:37.21 | cpatry | dunno, cause ur d-chan has problem? |
| 18:37.48 | hardwire | I guess. |
| 18:37.53 | hardwire | it seems to be timing issues |
| 18:37.53 | *** join/#asterisk pilot51 (n=pi_Chulo@81.215.165.85) |
| 18:37.58 | hardwire | I will call and see if I am slipping |
| 18:38.00 | hardwire | grr. |
| 18:38.18 | Sedorox | anyone have a T1 card (for asterisk.. pci) that they are willing to sell for under $100? :p |
| 18:38.21 | *** join/#asterisk contegixmatthew (n=matthew@63.246.15.189) |
| 18:39.08 | contegixmatthew | greetings. i am looking for Brian Christie |
| 18:39.30 | hardwire | Sedorox: you just aren't that lucky. |
| 18:39.38 | Sedorox | I figured as much |
| 18:39.38 | Sedorox | :/ |
| 18:39.50 | Sedorox | need the password for this thing first anyway... which I can't seem to find :(:( |
| 18:40.19 | orospakr | hi! asterisk 1.0.9 isn't compiling on x86_64, with this message: http://pastebin.ca/22238 |
| 18:41.08 | brad_mssw | orospakr: you need to have at least gcc 3.4.4 to use -march=k8 |
| 18:41.15 | *** join/#asterisk scream_01 (n=xxexclus@85.97.20.105) |
| 18:41.20 | brad_mssw | orospakr: err, gcc 3.4.0 rather ... |
| 18:42.03 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 18:42.15 | brad_mssw | orospakr: I couldn't tell you if those cflags are built into asterisk or not though, i would assume not though ... what distro are you on ? |
| 18:42.25 | orospakr | brad_mssw, ubuntu 5.04. :) |
| 18:42.29 | Falle | i have a problem with my TDM card. It dont detect hangup's. the "channel" stays in the callqueue untill someone picks it up even if the caller hung up 10 minutes ago. Any ideas? |
| 18:42.36 | orospakr | yeah, gcc 3.3.5 |
| 18:42.51 | orospakr | is it work using gcc 3.4 or should I just change asterisk's makefiles? |
| 18:43.04 | brad_mssw | gcc 3.4 should definitely work |
| 18:43.21 | brad_mssw | but you could also edit the makefiles if the -march=k8 and -mcpu=k8 are hardcoded in them |
| 18:43.42 | *** join/#asterisk knight_ (i=[U2FsdGV@blackhole.phunc.com) |
| 18:43.43 | PupenoL | Anyone familiar with chan_agent.c sources ? |
| 18:43.43 | knight_ | hey |
| 18:44.02 | knight_ | anyone have asterisk@home hang on GRUB stage 2 after install? |
| 18:44.04 | SuPrSluG | when i have an inbound call and dial exten 211 asterisk dials 2 and comes back with invalid extension. why isn't it waiting for all the digits before dialing |
| 18:44.47 | bjohnson | contegixmatthew: I'm looking for Christie Brinkley |
| 18:45.58 | Hmmhesays | quick easiest way to regexp a value out from between quotes |
| 18:47.24 | file[muon] | Hmmhesays: YOU |
| 18:47.44 | orospakr | brad_mssw, thanks :) |
| 18:48.05 | *** join/#asterisk L|NUX (n=linux@202.5.145.14) |
| 18:48.29 | Hmmhesays | file what |
| 18:48.34 | Hmmhesays | 0_o |
| 18:49.15 | file[muon] | it's all your fault the hard drive in this box is failing |
| 18:49.20 | Hmmhesays | why is that? |
| 18:49.48 | Wonka | erm |
| 18:49.58 | spackle | ~seen ManxPower |
| 18:50.01 | jbot | manxpower is currently on #asterisk (3h 16m 36s). Has said a total of 55 messages. Is idling for 23m 5s |
| 18:50.06 | *** join/#asterisk erdal (n=JuSTiN__@85.99.48.34) |
| 18:50.26 | Katty | i have accomplished much today. |
| 18:50.32 | Katty | i moved my entire rest of office upstairs. |
| 18:50.34 | file[muon] | reallllly |
| 18:50.39 | file[muon] | that's nifty |
| 18:50.47 | Katty | yes |
| 18:50.50 | Katty | i must post gifs soon |
| 18:51.03 | file[muon] | gifs?!?!?!? |
| 18:51.07 | file[muon] | no no, jpgs or pngs |
| 18:51.18 | Wonka | ack |
| 18:51.43 | *** join/#asterisk tld (n=tld@253.80-203-96.nextgentel.com) |
| 18:51.52 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 18:52.05 | Katty | file[muon]: silly rabbit. |
| 18:52.29 | Katty | i'll put them in the gallery |
| 18:52.33 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 18:52.37 | Katty | with the other 38 pictures of me. |
| 18:52.37 | hardwire | has echocan ever caused dropped calls? |
| 18:52.43 | *** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
| 18:52.43 | *** mode/#asterisk [+o bkw__] by ChanServ |
| 18:52.56 | Katty | bkw__: afternoon (and you better say hi!) |
| 18:53.21 | file[muon] | high, high as a kite! |
| 18:53.25 | file[muon] | oh no it's Nix |
| 18:53.28 | file[muon] | *fear* |
| 18:53.35 | Nix | heh |
| 18:53.47 | Katty | Nix: now i have to disinfect. |
| 18:53.55 | Katty | Nix: but the lemony scented goodness will be mine! |
| 18:54.10 | afrosheen | bah, disinfect. |
| 18:54.20 | Katty | afrosheen: he has nerdy boy cooties. |
| 18:54.45 | afrosheen | hardwire: echocan? |
| 18:54.49 | *** part/#asterisk santiago (n=santiago@63.245.86.254) |
| 18:54.51 | hardwire | echo cancel |
| 18:55.00 | *** join/#asterisk Cresl1n (n=Cresl1n@digium.com) |
| 18:55.02 | afrosheen | the hardware card or the software option for tdm cards? |
| 18:55.08 | file[muon] | omfg it's Cresl1n |
| 18:55.23 | sivana | what does it take to make a copper pair into a T1? |
| 18:55.28 | afrosheen | you can't just say echo cancel :) |
| 18:55.33 | Cresl1n | file!!!! |
| 18:55.37 | Cresl1n | what's up my people? |
| 18:55.46 | visik7 | does zaphfc contains inline asm ? |
| 18:55.51 | Sedorox | sivana: some little box that just shocked the shit outta me about 30 mins ago |
| 18:55.51 | Sedorox | :p |
| 18:55.54 | Cresl1n | sivana: t1 cards :-) |
| 18:55.55 | gordonjcp | afrosheen: no, you need to say echochochocho cancelcelcelcelcelcel... |
| 18:56.11 | afrosheen | gordonjcp: gordonjcp |
| 18:56.13 | sivana | Cresl1n: what sort of cards? :) |
| 18:56.17 | afrosheen | that's far end echo |
| 18:56.22 | file[muon] | gah dang it |
| 18:56.26 | file[muon] | silly server, WORK! |
| 18:56.32 | Cresl1n | sivana: digium cards :-) |
| 18:56.32 | *** join/#asterisk t3t (n=t3t@galley.pangalacticgargleblaster.com) |
| 18:56.39 | Cresl1n | (I'm a bit biased) |
| 18:56.40 | Cresl1n | :-P |
| 18:57.10 | Cresl1n | TE110P, TE410P, TE405P |
| 18:57.19 | afrosheen | easy question: what's better, a bunch of TDM cards for analog or a channel bank |
| 18:57.21 | FuzzyCat | A104 |
| 18:57.22 | Sedorox | sivana: if you want to send it over just a pair (e.g. two wires) over a distance.. |
| 18:57.31 | Sedorox | your gonna need one of these (http://cgi.ebay.com/Adtran-HTOC-1242034L3-Enclosure-with-H2-TUR-1221026L1_W0QQitemZ5804315185QQcategoryZ80226QQrdZ1QQcmdZViewItem) at each end |
| 18:57.48 | sivana | and that's it? |
| 18:57.49 | Sedorox | then they would plug into your T1 device.. be it digium cards.. or a DSU/CSU... channel bank.. etc.. |
| 18:57.53 | sivana | ya |
| 18:57.54 | Sedorox | I'm not sure what the technical name is :p |
| 18:58.01 | Cresl1n | sivana: you want to use just a single pair of wire? |
| 18:58.04 | sivana | yes |
| 18:58.08 | Cresl1n | that's a different issue |
| 18:58.14 | Cresl1n | T1 is a 4 wire interface |
| 18:58.20 | Sedorox | T1 can be either.... |
| 18:58.22 | Cresl1n | one pair for tx and another pair for rx |
| 18:58.23 | Sedorox | well... |
| 18:58.30 | Sedorox | witht hat device.. |
| 18:58.54 | afrosheen | are you trying to do some kind of wacky closed-loop t1 between 2 sites with nothing in between |
| 18:58.55 | Cresl1n | you can transport T1 over hdsl which (IIRC) can be over a single pair |
| 18:58.56 | sivana | my PRI is delivered over 2 wire |
| 18:59.01 | *** join/#asterisk murat_15 (n=aLper26@85.96.97.74) |
| 18:59.04 | Cresl1n | but a true T1 is a 4 wire interface |
| 18:59.10 | Cresl1n | as is a PRI as well :-) |
| 18:59.15 | sivana | afrosheen: yes, over a dedicated copper pair |
| 18:59.39 | Cresl1n | sivana: if it's just a standard T1 you just need an interface card |
| 18:59.54 | sivana | Cresl1n: ie. that Adtran? |
| 19:00.10 | Sedorox | no |
| 19:00.16 | Sedorox | the itnerface card would be from digium |
| 19:00.16 | Cresl1n | sivana: what adtran? |
| 19:00.19 | Sedorox | or sangoma |
| 19:00.28 | Sedorox | Cresl1n: the link I posted.. from ebay |
| 19:00.40 | Cresl1n | ah, ok |
| 19:00.52 | sivana | ok.. I have a couple TE405... I want to use a 2 wire loop to produce T1 between two sites |
| 19:00.54 | mogorman | ebay the worlds largest fence... |
| 19:00.55 | Sedorox | I wish I new the technical name for that :p |
| 19:01.00 | sivana | with a Digium card on each end |
| 19:01.11 | Cresl1n | sivana: you need more than 2 wires for a T1 |
| 19:01.25 | Sedorox | sivana: I believe you need that then.. which would put it on a pair.. |
| 19:01.26 | *** join/#asterisk Tambiah (n=Sevgineh@81.214.224.107) |
| 19:01.34 | spackle | Sedorox: NIU or NUI - Network Interface Unit or something. |
| 19:01.37 | Cresl1n | sivana: you need 4 :-) |
| 19:01.43 | Sedorox | ahh ok |
| 19:01.51 | sivana | my PRI is over 2 wire with a blue box with HDSL |
| 19:01.53 | sivana | HDSL2 |
| 19:01.56 | Cresl1n | yep |
| 19:02.02 | Sedorox | Cresl1n: why can't you just run it over the NIU? |
| 19:02.05 | Cresl1n | you'd need a couple of those boxes at each end |
| 19:02.07 | Cresl1n | NIU? |
| 19:02.16 | Sedorox | the link I posted... |
| 19:02.22 | Cresl1n | maybe |
| 19:02.27 | Sedorox | takes a T1 rj45 and runs over a pair.. |
| 19:02.28 | Cresl1n | as long as it breaks it out to four wire at each end |
| 19:02.34 | Sedorox | yea... |
| 19:02.53 | Sedorox | actually does 6.. but the one I have only has 4 connected in the device |
| 19:02.54 | Sedorox | so... |
| 19:03.11 | Sedorox | "Network Controller Unit"? |
| 19:03.13 | Sedorox | *shrugs* |
| 19:03.13 | Sedorox | lol |
| 19:03.19 | Cresl1n | I see :-) |
| 19:03.24 | sivana | Sedorox: so with one at each end, i can create a T1 loop over a pair? |
| 19:03.30 | Sedorox | I would think so... |
| 19:03.36 | sivana | hehe |
| 19:03.38 | *** join/#asterisk ^^DuDu (n=IsLaK_Er@81.214.224.107) |
| 19:03.46 | jontow | can TDMoE do that sort of thing? |
| 19:03.53 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-38lc0k2.dialup.mindspring.com) |
| 19:04.05 | Cresl1n | jontow: you have to have an ethernet connection |
| 19:04.08 | Sedorox | that would provider the voltage over the line (yea.. be careful with that.. -200vdc.. nice little shock.. as I've recently experienced) |
| 19:04.18 | Cresl1n | jontow: how many wires are used in ethernet? |
| 19:04.19 | jontow | yes, that i understand |
| 19:04.20 | spackle | Jontow: on the same subnet |
| 19:04.24 | jontow | aha |
| 19:04.50 | *** join/#asterisk jayk- (i=jayk@vapid.reprehensible.net) |
| 19:05.13 | spackle | Ethernet uses two pair doesn't it? |
| 19:05.17 | Sedorox | if I had a camera.. I could take pics of how this one is setup... |
| 19:05.20 | konrads | spackle: yes. |
| 19:05.27 | file[muon] | aha... |
| 19:05.27 | konrads | spackle: rest shouldn't be used |
| 19:05.29 | Sedorox | spackle: yes.. except for gigabit |
| 19:05.38 | jayk- | i'm trying to set up asterisk so that i can transfer a call to voicemail. i put exten => _*1XX,1,Voicemail(u${EXTEN:1}) and exten => _*1XX,2,Hangup into extensions.conf, but when i dial *<EXT> from my cisco 7960 cisco phone, it doesn't go through. |
| 19:05.40 | Sedorox | gigabit uses all 4 pairs |
| 19:05.56 | jayk- | for some reason, the phone or the asterisk server isn't accepting "*" then the extension. anybody have any ideas? |
| 19:08.07 | [ViRii] | i can dial to my analog line. its attached to a x100p clone. from there the call is forwarded (configured via amp) to an extension. i can receive the calls there however when trying to dial from the phone i get a busy tone. |
| 19:08.44 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 19:10.01 | PupenoL | Is there any use in that agents prompts for username, checks it, and then prompts for the password, *instead* of prompting for both and then checking. |
| 19:10.19 | PupenoL | ? |
| 19:10.33 | *** join/#asterisk clive- (n=pirch@rrba-146-91-249.telkomadsl.co.za) |
| 19:12.28 | *** join/#asterisk huslage_ (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
| 19:12.31 | *** join/#asterisk tld (n=tld@253.80-203-96.nextgentel.com) |
| 19:12.58 | jayk- | anybody have any ideas? |
| 19:13.40 | FuzzyCat | [ViRii], that's because AMP is useless and does more harm than good |
| 19:13.43 | TheCops | I have callwaiting option on the line that asterisk using for my IVR system. You can hear "Beep beep" when you're on a phone and you get another line. for now, I'm transfering the line to an extension that have Flash cmd and Dial command for call back the guys who transfered the call to that extension. There's another way to manage call waiting with asterisk ? |
| 19:14.04 | Sedorox | PupenoL: yes |
| 19:14.30 | PupenoL | Sedorox: and what is it ? |
| 19:15.06 | Sedorox | exten => 221,1,AgentCallbackLogin(${CALLERIDNUM}|${CALLERIDNUM}@local) |
| 19:15.16 | Sedorox | I don't think you need that last part tho |
| 19:16.08 | PupenoL | Sedorox: I am sorry, but I don't get it. |
| 19:16.10 | [ViRii] | fuzzy: thanks |
| 19:16.45 | Sedorox | when I dial 221... I get prompted for my password, then #, so I type it, then it says agent logged in |
| 19:16.54 | [ViRii] | then how/what would i configure to get my phones to hit the pstn for outbound dials |
| 19:17.04 | Sedorox | it passes the number your calling from into the login, and uses that as the username |
| 19:17.55 | *** join/#asterisk pilot_tr (n=buzlar_p@81.214.224.107) |
| 19:18.31 | bkw_ | what? |
| 19:18.38 | PupenoL | Sedorox: ok, but that does force it to be impossible to ask for username and password and then checking when you are not using the caller id as username ? (I'll check it anyway). |
| 19:18.43 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 19:19.41 | Sedorox | well |
| 19:19.47 | Sedorox | if you just hit # |
| 19:19.57 | Sedorox | then it asks for agent ID and password |
| 19:20.06 | Sedorox | since it loops around.. but doesn't have the ID this time... |
| 19:20.58 | FuzzyCat | [ViRii], edit extensions.conf |
| 19:21.21 | [ViRii] | k |
| 19:21.37 | [ViRii] | ack ! |
| 19:22.14 | ender | is there anybody here that I was talking to about storing directory contacts in IP-301 and IP-501 phones w/ the latest firmware? |
| 19:22.40 | *** join/#asterisk Jackson-Grusby (n=SoN_GeMi@81.214.224.107) |
| 19:22.44 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:26.24 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:26.52 | Ariel_ | hello everyone |
| 19:28.23 | FuzzyCat | is that a real name? sounds like a place |
| 19:29.48 | MooingLemur | hmm.. asterisk segfaults in voicemailmain. trying to figure out if it's my config or due to the fact it's running in a vserver. |
| 19:30.07 | MooingLemur | everything else works :P |
| 19:31.12 | FuzzyCat | shouldn't make any difference |
| 19:31.15 | mutilator | Sep 6 15:29:41 NOTICE[2982]: chan_iax2.c:2230 auto_congest: Auto-congesting call due to slow response |
| 19:31.27 | mutilator | why might i be getting that when callin an iax line |
| 19:31.31 | FuzzyCat | I've run vmware *'s no probs |
| 19:31.42 | FuzzyCat | slow response... |
| 19:31.55 | mutilator | show peers shows it's there |
| 19:31.59 | mutilator | 1ms response time |
| 19:32.19 | mutilator | the other asterisk doesn't show anything |
| 19:32.31 | FuzzyCat | networking prob? |
| 19:33.14 | mutilator | .. |
| 19:33.17 | mutilator | of what kind |
| 19:33.21 | mutilator | they peer up fine |
| 19:33.26 | *** join/#asterisk clive-- (n=pirch@ndn-165-141-41.telkomadsl.co.za) |
| 19:33.26 | mutilator | but the call isn't doing anything |
| 19:33.43 | *** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 19:33.58 | Derkommissar | how come i dont get the calledid name in zap ? |
| 19:34.43 | Ariel_ | Derkommissar, hello long time no see. |
| 19:35.11 | mutilator | Dial(IAX2/m2k-trunk/1989${EXTEN}) should work right? |
| 19:35.31 | FuzzyCat | Derkommissar, from the PSTN? |
| 19:35.46 | FuzzyCat | you'll never get it from the pstn. |
| 19:35.50 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 19:36.13 | FuzzyCat | but if you mean from * to the phone, then the phone needs to support it |
| 19:36.50 | MRH2 | hi I have an error message on the console with current cvs i haven't seen before - ne1 know what it refers to |
| 19:36.54 | MRH2 | chan_local.c:134 local_queue_frame: blah wasn't locked while sending 1/35 |
| 19:37.50 | *** join/#asterisk Rankin (n=TEKiN@81.214.224.107) |
| 19:38.01 | *** join/#asterisk Tili (i=Tili@202-133-65-215-dialup.sat.net.pk) |
| 19:38.14 | *** join/#asterisk pauldy (n=pauldy@c-67-173-198-40.hsd1.tx.comcast.net) |
| 19:38.58 | Sedorox | anyone... connectreach/vina login password?!? ;-) |
| 19:39.18 | mutilator | i found the prob |
| 19:40.11 | *** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com) |
| 19:41.25 | mutilator | the fscking slash pimped me |
| 19:41.35 | mutilator | i had Dial(IAX2/m2k-trunk:1989${EXTEN}) |
| 19:43.18 | Hmmhesays | who's poling me |
| 19:43.20 | Hmmhesays | *poking even |
| 19:43.28 | Hmmhesays | 0_o |
| 19:43.40 | file[muon] | not I |
| 19:43.42 | *** join/#asterisk _hannes (n=hannes@port-212-202-55-34.dynamic.qsc.de) |
| 19:44.12 | Hmmhesays | it wasn't a nice poke either, it was one of those weird ones that makes you feel violated afterward |
| 19:44.59 | *** join/#asterisk Blackwell (n=B0RAN@81.214.224.107) |
| 19:46.51 | Katty | Hmmhesays: oh. |
| 19:47.38 | Hmmhesays | <chuckle> |
| 19:47.40 | Hmmhesays | 'ello |
| 19:48.06 | *** join/#asterisk zedkatuf (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk) |
| 19:48.10 | Hmmhesays | perl is making my head hurt today |
| 19:49.14 | *** join/#asterisk IshwariaRoye (n=Tutuklu@81.214.224.107) |
| 19:49.30 | fugitivo | what are you doing with perl? |
| 19:49.37 | bendy24 | holy mother of perl |
| 19:52.21 | Hmmhesays | i should probably have headphones on |
| 19:54.09 | spackle | should this work to record the channel as conf+datetime ? Monitor(wav,/var/spool/asterisk/monitor/conf${datetime},m) |
| 19:54.21 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 19:54.39 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 19:54.49 | spackle | I just get conf.wav instead of conf+datetime.wav |
| 19:57.51 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:00.52 | brad_mssw | what's the best way to access the asterisk database from a script (say like php via a webpage) ? Just need to issue some DBget and some DBput commands, but running asterisk -rx doesn't connect if you're the apache user (even if the apache user is in the asterisk group) |
| 20:01.33 | Hmmhesays | ok one of the salespeople just pawned me out for $60/hour |
| 20:01.36 | Hmmhesays | what a bunch of crap |
| 20:02.03 | file[muon] | poor Hmmhesays |
| 20:02.25 | oej | brad_mssw: Check the options for the pipe in README.asterisk.conf |
| 20:04.32 | brad_mssw | oej: err, what version of asterisk does that come with, I don't see it in 1.0.9 |
| 20:04.58 | *** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 20:05.16 | oej | brad_mssw: Sorry, that is 1.2beta |
| 20:05.46 | oej | brad_mssw: Check asterisk.conf to find the pipe that we use, then go to that directory and check the permissions for the pipe |
| 20:06.21 | oej | brad_mssw: Look for astrundir |
| 20:06.24 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 20:06.58 | PupenoL | Hello. |
| 20:07.56 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 20:08.08 | *** join/#asterisk izmirden_M (n=AntaLya@81.213.68.20) |
| 20:08.08 | brad_mssw | oej: cool, yeah doing a chmod g+w on the /var/run/asterisk/asterisk.ctl did the trick ... thanks |
| 20:08.27 | *** join/#asterisk kiwnix (n=egarcia@82.158.159.143) |
| 20:08.29 | oej | np |
| 20:08.34 | oej | Please test 1.2 |
| 20:08.36 | oej | :-) |
| 20:08.53 | clive-- | oej I am downloading it now |
| 20:09.04 | clive-- | what are the reports of it so far |
| 20:09.11 | brad_mssw | oej: err, wait, it always resets permissions upon asterisk restart ... crap |
| 20:09.33 | oej | Yes, we changed that in 1.2 |
| 20:09.33 | brad_mssw | oej: know of any way to change that other than to manually reset it every restart ? |
| 20:09.37 | brad_mssw | ah, ok ... |
| 20:09.45 | brad_mssw | hmm, maybe I'll just code to the manager IP interface ... |
| 20:10.00 | brad_mssw | I should be able to do dbput/dbget with that, no? |
| 20:11.25 | *** join/#asterisk zoo (i=nobody@ip-11-16.travedsl.de) |
| 20:11.49 | darkskiez | I'm messing around trying to add an UPDATE message to chan_sip to allow update to the outgoing callerid by adding a remote-party-id header on the cisco phones; Is this the right procedure? reqprep, add_header, add_header_contentLength, add_blank_header, send_request ? I'm not sure if i'm to increment p->ocseq or not, reqprep seems to do it, but it doesnt seem to get incremented in the sip trace. |
| 20:12.43 | eKo1 | darkskiez: i suggest you ask in #asterisk-dev |
| 20:12.58 | darkskiez | its as dead as elvis in there |
| 20:14.43 | FuzzyCat | elvis is alive and living in Chelmsford |
| 20:15.35 | Maveric | i thought elvis was alive in my pants |
| 20:15.36 | DrukenHME | pfft, elvis |
| 20:15.39 | darkskiez | out partying with the other dev's |
| 20:15.51 | file[muon] | ...oh joy |
| 20:16.00 | darkskiez | the point being, the aliveness of elvis is debateable. |
| 20:16.08 | darkskiez | and that of asterisk-dev |
| 20:16.18 | JessicaX^ | :D |
| 20:16.23 | JessicaX^ | alvis is alive |
| 20:17.54 | cpatry | sure, hes playing guitar with Kurt Cobain,. |
| 20:18.11 | *** join/#asterisk zoo^ (i=nobody@ip-11-16.travedsl.de) |
| 20:18.23 | bkw_ | <PROTECTED> |
| 20:18.51 | *** part/#asterisk zoo (i=nobody@ip-11-16.travedsl.de) |
| 20:19.24 | mutilator | anyone know where i can get some inexpensive 1u servers for making remote channel banks? |
| 20:19.40 | mutilator | other than ebay.. |
| 20:19.45 | afrosheen | mutilator: inexpensive and 1u are not compatible |
| 20:19.46 | paulc | define "inexpensive" ? |
| 20:19.47 | mutilator | lookin for like $300-$400 |
| 20:20.15 | Wonka | gas prices rose 0.2EUR/l _before_ oil prices rose... |
| 20:20.16 | mutilator | i missed a deal on some dual p3 servers |
| 20:20.20 | mutilator | lot of 5 for $750 |
| 20:20.26 | afrosheen | Wonka: welcome to speculative market pressure |
| 20:21.03 | DrukenHME | $1.90???? WTF do you live? |
| 20:21.07 | darkskiez | UK |
| 20:21.15 | afrosheen | he said 'litre' |
| 20:21.23 | afrosheen | that's nowhere near a gallon |
| 20:21.24 | DrukenHME | afrosheen: point? |
| 20:21.28 | Wonka | 1.149EUR/l super, this afternoon |
| 20:21.49 | DrukenHME | we are paying $1.30 a litre |
| 20:21.54 | Wonka | 1 gallon == 3.7854118 litre |
| 20:21.58 | eKo1 | It's about 4 USD per gallon over her. |
| 20:22.01 | DrukenHME | and that is expensive as shit |
| 20:22.06 | eKo1 | s/her/here |
| 20:22.09 | afrosheen | it'll be that high here in the US by the end of the year I bet |
| 20:22.12 | paulc | gas has been in the high $1.xx a litre in eastern Canada last week.. here in the west it's between $1.10 and $1.20 a litre right now |
| 20:22.19 | Hmmhesays | heh, i just got in trouble for "seeming" uninterested in helping a customer |
| 20:22.31 | paulc | I thought a gallon was 4.5 litres.. or maybe that's UK gallons not dodgy US ones.. |
| 20:22.35 | Wonka | Hmmhesays: use a cluebat on him |
| 20:22.54 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
| 20:22.55 | Wonka | hm, that was what units told me |
| 20:22.58 | Hmmhesays | I told him I was in the middle of something and I had to call him back |
| 20:23.02 | afrosheen | hahaha |
| 20:23.02 | paulc | of course.. the correct response is "Well if that's how I seem, it's probably because that's how I AM" |
| 20:23.04 | darkskiez | 0.99 (British pounds per litre) = 6.91499343 U.S. dollars per US gallon |
| 20:23.07 | Hmmhesays | and I told him I wasn't going to be around after 4 |
| 20:23.08 | mrfrenzy | hah you lucky bastards, here it's $1.59 / litre |
| 20:23.13 | *** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 20:23.16 | fugitivo | i think it's time to open #gas, it's been a hit lately |
| 20:23.18 | afrosheen | darkskiez: what are you guys paying for diesel |
| 20:23.32 | darkskiez | 1.04 (British pounds per litre) = 7.26423553 U.S. dollars per US gallon |
| 20:23.38 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 20:23.40 | FuriousGeorge | hey all |
| 20:23.42 | darkskiez | of diesel |
| 20:23.46 | afrosheen | weird how diesel used to be cheaper all the time |
| 20:23.57 | afrosheen | it's a less-refined distillate right |
| 20:24.01 | Wonka | $ units usgallon l |
| 20:24.01 | Wonka | <PROTECTED> |
| 20:24.27 | FuriousGeorge | i once bought gas in the late 90's for 87 cents/gallon |
| 20:24.37 | DrukenHME | afrosheen: actually, there's another step in making diesel |
| 20:24.40 | afrosheen | FuriousGeorge: 1999 |
| 20:24.43 | FuriousGeorge | in hindsight, shoulda got a few thousand gallons |
| 20:24.48 | Wonka | $ units brgallon l |
| 20:24.48 | Wonka | <PROTECTED> |
| 20:24.52 | afrosheen | FuriousGeorge: lol |
| 20:25.15 | Sedorox | lol |
| 20:25.20 | Wonka | darkskiez: biodiesel is foo. eats tubes... |
| 20:25.23 | Sedorox | I like the idea... |
| 20:25.25 | eKo1 | you mean methanol |
| 20:25.29 | *** join/#asterisk popvoxdave (i=user@dave2.toad.net) |
| 20:25.33 | Sedorox | but... I dunno about smelling like fries |
| 20:25.33 | Sedorox | :p |
| 20:25.33 | FuriousGeorge | how much does a gallon of butane cost here in the US? |
| 20:25.38 | DrukenHME | ethanol |
| 20:25.44 | FuriousGeorge | yeah thats it |
| 20:25.47 | afrosheen | gallon of butane..who knows |
| 20:25.51 | FuzzyCat | bio fuels are too inefficient... |
| 20:25.51 | *** join/#asterisk gooagle (n=goldenol@ns2.xoasisnetworks.com) |
| 20:25.52 | afrosheen | propane = better |
| 20:25.54 | FuriousGeorge | or is it propane |
| 20:26.02 | Wonka | afrosheen: brassica napus is better than corn |
| 20:26.09 | FuzzyCat | it takes more energy to grow and harvest it that it gives out |
| 20:26.12 | FuriousGeorge | what do they run their cars on in brazil? propane? |
| 20:26.16 | afrosheen | I say we all drive busses with 10 chinese guys pedaling on board |
| 20:26.19 | mutilator | so no leads on any servers? |
| 20:26.30 | afrosheen | labor is cheaper than gas |
| 20:26.30 | Wonka | biodiesel is "Rapsölmethylester" |
| 20:26.31 | paulc | we're still laughing at the $300 1U server idea.. |
| 20:27.18 | darkskiez | in that case I want people to push me to work |
| 20:27.18 | mutilator | why |
| 20:27.18 | afrosheen | mutilator: yeah that's too low |
| 20:27.18 | mutilator | i've found em |
| 20:27.18 | paulc | cos it's too dirt cheap? |
| 20:27.18 | mutilator | up to $500 i'm lookin] |
| 20:27.18 | Sedorox | granted.. needed to by ram... |
| 20:27.18 | Sedorox | and fix up the cooliing.. but still :p |
| 20:27.18 | afrosheen | Sedorox: what kind of parts are inside |
| 20:27.19 | Sedorox | dual PIII 1g |
| 20:27.21 | Sedorox | SCSI and SATA |
| 20:27.26 | afrosheen | sata on a p3? wow |
| 20:27.26 | gooagle | regarding the Sangoma a101u cards, anyone feel these have inferior echo canceling versus the new te110p? I am getting a ton of echo on the a101u |
| 20:27.29 | Sedorox | supermicro |
| 20:27.33 | Sedorox | yea.. addon |
| 20:27.33 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:27.33 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 20:27.34 | afrosheen | decent board |
| 20:27.45 | Sedorox | yea.. hopefully its shipping today :p |
| 20:27.49 | file | darumkilla! |
| 20:27.59 | afrosheen | but sata and linux is questionable at times |
| 20:28.09 | Sedorox | I know |
| 20:28.14 | eKo1 | stick with scsi |
| 20:28.14 | Sedorox | don't have a sata drive right now |
| 20:28.16 | drumkilla_laptop | file!!!!!!!!!!!!!!! |
| 20:28.16 | afrosheen | it's all about what distro, what kernel, what libs bla bla |
| 20:28.18 | FuzzyCat | never had any trouble with sata and linux |
| 20:28.19 | Sedorox | so only using the scsu... |
| 20:28.20 | drumkilla_laptop | I miss you, file. |
| 20:28.21 | *** join/#asterisk Blagg (n=ALBERT_M@81.213.68.20) |
| 20:28.21 | Sedorox | scsi* |
| 20:28.28 | Sedorox | but its actually gonna be our PBX |
| 20:28.29 | afrosheen | FuzzyCat: yes but others have |
| 20:28.29 | Sedorox | :p |
| 20:28.29 | paulc | aww.. drumkilla's no trouble :) |
| 20:28.32 | FuzzyCat | and there's 30 servers in the cab |
| 20:28.34 | Sedorox | so.. it'll be running asterisk |
| 20:28.40 | FuzzyCat | all linux+sata |
| 20:28.41 | Sedorox | fugitivo: hehe |
| 20:28.42 | Sedorox | er |
| 20:28.44 | Sedorox | FuzzyCat: |
| 20:28.52 | Sedorox | yea.. it depends on the sata chipset mainly.. I think |
| 20:29.06 | afrosheen | and there are like 30 of them out there |
| 20:29.10 | fugitivo | me? |
| 20:29.13 | fugitivo | hu? what? |
| 20:29.13 | Sedorox | no... |
| 20:29.15 | Sedorox | sorry |
| 20:29.15 | afrosheen | it'll improve with time just like anything else |
| 20:29.19 | Sedorox | stupid tab completetion |
| 20:29.19 | Sedorox | :p |
| 20:29.21 | FuzzyCat | well, we don't buy cheap shite servers that cost $300 ;) |
| 20:29.24 | Sedorox | well yea... |
| 20:29.34 | paulc | you want it to work, you pays your money.. |
| 20:29.40 | paulc | there's a thin line between frugal and sensible? |
| 20:29.42 | Sedorox | FuzzyCat: well yea.. but the opurtunity came up.. so.. I took it |
| 20:29.43 | Sedorox | :p |
| 20:29.43 | afrosheen | you knows it clarts |
| 20:29.50 | fugitivo | a complete server for $300? |
| 20:29.54 | FuriousGeorge | how big are the propane tanks home depot gives you for 12.95? |
| 20:30.02 | afrosheen | FuriousGeorge: call and ask |
| 20:30.07 | FuzzyCat | Sedorox, just buy another card... unless it's 1u ... then ur fecked |
| 20:30.10 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 20:30.19 | afrosheen | FuriousGeorge: you can still buy propane powered or supplemented trucks, at least here in Texas |
| 20:30.26 | Sedorox | its 1u |
| 20:30.27 | Sedorox | :p |
| 20:30.43 | Sedorox | but what do I need another card for? |
| 20:30.53 | FuzzyCat | to replace the crappy sata one |
| 20:30.56 | Sedorox | right now.. the pci slot is only used by the sata card.. so... |
| 20:31.02 | FuzzyCat | ahh ok |
| 20:31.02 | Sedorox | I duno what model it is yet |
| 20:31.06 | Sedorox | so all I know.. it could be good |
| 20:31.08 | afrosheen | since you're not using sata you're good to go |
| 20:31.13 | Sedorox | yea |
| 20:31.14 | Sedorox | right now... |
| 20:31.24 | *** join/#asterisk Heidi_ (n=Dostca@81.214.119.210) |
| 20:31.25 | afrosheen | personally I love sata |
| 20:31.29 | Sedorox | I was actually gonna get a 80-120gig sata to use just for asterisk (program.. voicemail.. etc...) |
| 20:31.31 | afrosheen | but I've been lucky with chipsets |
| 20:31.34 | mutilator | hmm |
| 20:31.35 | FuzzyCat | sata or satay |
| 20:31.37 | FuzzyCat | ? |
| 20:31.37 | FuriousGeorge | afrosheen: i bet i can find a good mechanic here in newark, nj who hook me up with a propane installation. assuming its only 10 gallon for 12.95, even at 4/5th efficency of 87 octane normal gas, thats still a helluva savings |
| 20:31.38 | Sedorox | and the scsi for the system. but we're saving money... |
| 20:31.45 | Sedorox | sata |
| 20:31.52 | Sedorox | :p |
| 20:32.08 | afrosheen | yeah those seagate 80g sata drives with NCQ are super fast |
| 20:32.11 | fugitivo | do you really need scsi for asterisk? |
| 20:32.20 | Juggie | ummm no |
| 20:32.21 | Juggie | hah |
| 20:32.23 | Sedorox | no.. but its in the server |
| 20:32.24 | FuzzyCat | you don't even need a hdd |
| 20:32.25 | Sedorox | so why not? :p |
| 20:32.30 | Juggie | no one needs a fast hdd for * |
| 20:32.38 | fugitivo | i'd use a CF for the system, ide or sata for data |
| 20:32.38 | Juggie | because your cdr should be on another server |
| 20:32.40 | eKo1 | you can run it of a cd |
| 20:32.48 | *** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com) |
| 20:32.51 | Maveric | i wouldn't waste money on a good controller until they get asterisk working async and multithreaded |
| 20:33.02 | Juggie | afrosheen, asterisk does almost NO hdd io |
| 20:33.02 | Maveric | any heavy i/o |
| 20:33.03 | opus_ | async, where |
| 20:33.12 | Maveric | can cause calls to be not very pleasent |
| 20:33.17 | afrosheen | that entry goes right next to gates' 'nobody will need more than 640k' comment |
| 20:33.18 | Maveric | cutting in and out |
| 20:33.27 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 20:33.29 | Sedorox | hehe |
| 20:33.41 | Juggie | afrosheen, give me a scenario when asterisk would do alot of hard drive IO |
| 20:33.45 | *** join/#asterisk Katty (n=katrina@68.112.15.110) |
| 20:33.49 | afrosheen | busy voicemail server |
| 20:33.53 | Sedorox | well... we're also debating on setting this as a hot spare main server.. so if needed.. we can fall over to it. |
| 20:34.01 | cpatry | Juggie: monitoring ? :) |
| 20:34.04 | FuzzyCat | no - you use a SAN for voicemail |
| 20:34.09 | FuriousGeorge | anyone know if any of these presence/im projects on the horizon are expected to work w/ exten eyebeam? its what we use at work |
| 20:34.13 | afrosheen | so you don't have a SAN |
| 20:34.13 | Maveric | Juggie there are times that you do have a need for heavy i/o in a production enviroment |
| 20:34.14 | Sedorox | lol |
| 20:34.16 | fugitivo | recording or some AGI that uses a lot of HD |
| 20:34.28 | Juggie | afrosheen, if your voicemail server is busy, and then that means its hopefully redundant... so your voice mail should be stored in a sql table :) |
| 20:34.29 | afrosheen | yeah there are plenty of scenarios, don't be such an elitist |
| 20:34.29 | Juggie | NEXT :P |
| 20:34.34 | Sedorox | yea... ivr and voicemail was thinking about putting on the sata... |
| 20:34.54 | afrosheen | not everyone using * has an IBM scale data center |
| 20:35.00 | Sedorox | lol |
| 20:35.06 | FuzzyCat | you don;t need to.,.. |
| 20:35.12 | FuzzyCat | it's just common sense |
| 20:35.16 | Juggie | if you are pushing enough voicemail to have to worry about HDD performance |
| 20:35.18 | mutilator | http://cgi.ebay.com/IBM-xSeries-330-1U-Server-Intel-1-26GHz-1GB-X330_W0QQitemZ5804633431QQcategoryZ11215QQrdZ1QQcmdZViewItem |
| 20:35.24 | Juggie | you should consider redundant voicemail servers with an sql backend |
| 20:35.29 | mutilator | like that is what i need |
| 20:35.39 | Sedorox | hmmm |
| 20:35.43 | fugitivo | i don't like the idea of sql for asterisk |
| 20:35.46 | FuzzyCat | if you spread the load over servers and they can register with any one (load balancing) then then need to be able to get their voicemail |
| 20:35.56 | FuzzyCat | what else ru gonna do rsync their voicemail |
| 20:36.01 | FuzzyCat | ? |
| 20:36.06 | fugitivo | that means another service, more problems |
| 20:36.30 | Juggie | SQL voicemail works fine |
| 20:36.42 | Sedorox | lol |
| 20:36.51 | fugitivo | sure, until it stops working fine :) |
| 20:36.52 | Juggie | its actually very sexy, leave voicemail on one server, pick it off another, etc. |
| 20:36.58 | afrosheen | FuzzyCat: it's not a race, you can spell out 'are you' ;) |
| 20:37.16 | FuzzyCat | ru takin da ps :P |
| 20:37.22 | afrosheen | r u 4 bbq |
| 20:37.30 | FuzzyCat | mmmmmm |
| 20:37.37 | FuzzyCat | fewd! |
| 20:38.06 | afrosheen | so what's everyone's favorite channel bank, the adtran? |
| 20:38.08 | FuzzyCat | anyone, it doesn't need to cost the earth... |
| 20:38.14 | FuzzyCat | anyway |
| 20:38.14 | Juggie | what linux distro should i put on a HP server, 2*3.2 Ghx proc, 4GB RAM, 3*75gb HD |
| 20:38.23 | FuzzyCat | windows-linux |
| 20:38.24 | FuzzyCat | :P |
| 20:38.25 | *** join/#asterisk jaybuffet (n=random@rrcs-24-227-53-138.se.biz.rr.com) |
| 20:38.27 | afrosheen | Juggie: centos..debian...whatever |
| 20:38.28 | fugitivo | what linux distro do you use? |
| 20:38.35 | Juggie | i normally use fedora |
| 20:38.38 | Maveric | gentoo |
| 20:38.40 | fugitivo | then use fedora |
| 20:38.47 | FuzzyCat | eeww gentoo... |
| 20:38.48 | afrosheen | I recommend centos4, highly |
| 20:38.49 | Juggie | nah, i dont want to wait for compiling |
| 20:38.53 | Juggie | i prefer apt-get love |
| 20:38.55 | FuzzyCat | yeah, stick with what u know |
| 20:39.00 | fugitivo | i recommend gentoo, but if you use fedora, why change? |
| 20:39.05 | Maveric | Juggie i like apt as well as the next person |
| 20:39.09 | Maveric | i'm a debian man |
| 20:39.17 | Juggie | apt-get rocks, yum/up2date blows |
| 20:39.19 | Maveric | but recently been using gentoo and it has a lot of nice things |
| 20:39.21 | afrosheen | you can use apt on rpm-based distros |
| 20:39.25 | jaybuffet | hello all... what cost am i looking at for hardware to support a small business (20-30 lines).. i believe we have a t1 coming is currently |
| 20:39.26 | FuzzyCat | wusses - whats wrong with tarballs |
| 20:39.26 | Juggie | i know |
| 20:39.34 | knight_ | apt-rpm rocks |
| 20:39.35 | Juggie | i use it on all my fedora systems |
| 20:39.37 | afrosheen | FuzzyCat: they stain the carpet |
| 20:39.38 | Maveric | fedora blows |
| 20:39.43 | knight_ | even though i despise rpm |
| 20:39.52 | mutilator | would a 500mhz 256mb ram run a 24 port channel bank -> iax trunk fine |
| 20:40.02 | afrosheen | jaybuffet: are you replacing all the phones with sip phones or what |
| 20:40.05 | FuzzyCat | I hate rpm, you never know what the buggers have done |
| 20:40.08 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 20:40.08 | fugitivo | i started using linux (slackware) 10 years ago, packages didn't exist |
| 20:40.16 | jaybuffet | afrosheen: yes probably |
| 20:40.16 | Juggie | mutilator, maybe, if you stick to ulaw only |
| 20:40.21 | gooagle | does the te110p versus the te100p have additional echo cancelling? |
| 20:40.23 | mutilator | ya, i would |
| 20:40.26 | afrosheen | back in my day, we had to type every line of code by hand..seriously...c64 :) |
| 20:40.29 | Juggie | then probally |
| 20:40.30 | mutilator | the backhaul pipe is huge |
| 20:40.34 | Juggie | as there is no codec translation |
| 20:40.46 | fugitivo | afrosheen: how do you type code today?? |
| 20:40.48 | afrosheen | jaybuffet: what does your current call volume look like there |
| 20:40.55 | afrosheen | fugitivo: bah you download it from india |
| 20:41.01 | Juggie | mutilator, if you are doing voicemail, convert all the gsm files to ulaw/wav files |
| 20:41.04 | Juggie | so theres no transcoding |
| 20:41.06 | fugitivo | afrosheen: lol |
| 20:41.12 | mutilator | if not then what.. go to gsm? |
| 20:41.16 | afrosheen | fugitivo: they do all the typing |
| 20:41.19 | jaybuffet | afrosheen: i would say 8-10 people on at one time usually |
| 20:41.24 | zedkatuf | Has anyone come across this problem: I can do inbound phone calls, but I don't hear the "ring ring" from the ohone that I'm calling from..the phones connected to my asterisk box ring OK..it's just a bit annoying... |
| 20:41.30 | afrosheen | jaybuffet: how about total call volume, how many hours per month |
| 20:41.38 | Juggie | mutilator, you cant go to gsm, because if you do gsm, you have to transcode when * talks to a t1 port |
| 20:41.42 | FuzzyCat | hours? minutes surely |
| 20:41.57 | afrosheen | either or |
| 20:42.09 | jaybuffet | afrosheen: oh.. you got me |
| 20:42.12 | mutilator | how about 2x700? |
| 20:42.13 | Juggie | does apt-rpn work on centos? |
| 20:42.22 | mutilator | handle 2 of those banks? |
| 20:42.22 | FuzzyCat | doens't everyone say 1 million possibly more then never do it |
| 20:42.23 | Juggie | er, apt-rpm |
| 20:42.24 | FuzzyCat | ;) |
| 20:42.25 | mutilator | using ulaw |
| 20:42.27 | afrosheen | Juggie: yeah it's binary compatible with rhel3 |
| 20:42.56 | Juggie | hrm, maybe i should try centos then |
| 20:42.57 | afrosheen | jaybuffet: assessment is half the deal when you're planning something like this |
| 20:43.07 | FuzzyCat | tao/centos etc... |
| 20:43.15 | FuzzyCat | but I'm going off centos... |
| 20:43.21 | afrosheen | Juggie: it's worth a shot, great distro as far as I'm concerned..good for a server |
| 20:43.35 | Juggie | which this is |
| 20:43.38 | jaybuffet | afrosheen: but i would need a look up table or something to compare my assessment to. does that exist ? |
| 20:43.41 | afrosheen | alot more stable than fedora as well |
| 20:43.45 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 20:43.53 | Juggie | i havnt had any problems with fedora |
| 20:43.53 | FuzzyCat | if you install php and mysql on a centos box from the install php doesn;t support mysql - how dumb is that |
| 20:43.55 | Juggie | stability wise |
| 20:43.55 | afrosheen | jaybuffet: the phone company will know how many minutes you used last month I think |
| 20:44.03 | darkskiez | zedkatuf: than can be down to configuration, also see the options you can specify to the dial command. |
| 20:44.13 | afrosheen | does the phone company track minutes on hard lines to businesses? |
| 20:44.32 | zedkatuf | darkskiez: Ta..am v much a newbie atm so tracking down problem could be tricky..but tnx for the pointer |
| 20:44.55 | FuzzyCat | fedora moves on too quickly |
| 20:45.10 | afrosheen | I heard fedora3 was a disaster from a few people |
| 20:45.29 | Juggie | works fine here |
| 20:45.30 | FuzzyCat | puzzled like fedora |
| 20:45.50 | FuzzyCat | but he's a rhe so has no choice ;) |
| 20:46.13 | eKo1 | rhe > fedora |
| 20:46.15 | afrosheen | jaybuffet: you're probably looking at spending about $200 per phone, if you want nice phones, around $2k on a server, and maybe adding a second t1 dedicated to voice |
| 20:46.22 | FuzzyCat | personally I think rh made a mistake... |
| 20:46.35 | opus_ | fuzzycat - i got php and mysql to work out of the box.. with centos 4? |
| 20:46.40 | FuzzyCat | 4.1 |
| 20:46.47 | jaybuffet | afrosheen: called accounting.. gonna go look at the phone bill |
| 20:46.51 | opus_ | yes 4.1 also |
| 20:46.58 | FuzzyCat | mmmm.... |
| 20:47.01 | opus_ | RHEL also supports it |
| 20:47.07 | afrosheen | jaybuffet: that'll just help you see what kind of bandwidth you'll probably be eating for the phones alone |
| 20:47.12 | FuzzyCat | it don't work here.... |
| 20:47.14 | opus_ | i think you are missing /etc/httpd/conf.d/ ? |
| 20:47.17 | Hmmhesays | ok, i'm going to quit this place now |
| 20:47.24 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:47.41 | opus_ | fuzzycat hmm. weird |
| 20:47.45 | *** join/#asterisk tq1 (n=pedro@200.117.234.254) |
| 20:47.54 | opus_ | fuzzycat doesn't matter. |
| 20:47.55 | FuzzyCat | yah... a pita if you ask me... |
| 20:47.55 | blitzrage | JunK-Y: j00 around? |
| 20:48.19 | FuzzyCat | opus_, it does to me ;) if I can get away with not rebuilding php I want to |
| 20:48.20 | opus_ | you can get a nice server for around $1100 |
| 20:48.25 | cpatry | yes |
| 20:48.39 | afrosheen | opus_: or two for 2200, nice to have a live backup |
| 20:48.39 | blitzrage | cpatry: sounds like I might be heading to Montreal in November |
| 20:48.47 | blitzrage | cpatry: just wanted to give you a heads up :) |
| 20:48.48 | cpatry | really? u have any date? |
| 20:48.50 | afrosheen | blitzrage: take your snow shoes |
| 20:48.50 | opus_ | afrosheen what servers do you use, curious |
| 20:48.59 | blitzrage | afrosheen: I live in Toronto - I know what its like :) |
| 20:49.05 | afrosheen | opus_: dell 2830 1u servers |
| 20:49.06 | blitzrage | afrosheen: November - won't be that much snow yet anyways |
| 20:49.08 | cpatry | let me know, i'll ask for some days off to chill in downtown. |
| 20:49.23 | blitzrage | cpatry: I've been told Nov. 11th - so I assume a day on either side of that too |
| 20:49.32 | *** join/#asterisk Corndawg (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 20:49.33 | cpatry | k |
| 20:50.04 | afrosheen | blitzrage: what is it with people in toronto and computer experts, I swear that town is a genius vortex |
| 20:50.19 | file | blitzrage is very silly with his expertise |
| 20:50.38 | Katty | darkskiez: beep. |
| 20:50.40 | jarrod | how would I create a new indiciation in asterisk to play the hangup tone that pulls battery for 900ms |
| 20:51.10 | afrosheen | file: I've known alot of canadian programmers in the past, all of them very unorthodox and very good |
| 20:51.22 | cpatry | jarrod: show application playtones |
| 20:52.04 | opus_ | it must be the w33d |
| 20:52.16 | file | afrosheen: scary |
| 20:52.17 | afrosheen | or the shrooms |
| 20:52.38 | R3DB0x | i need to order 8 corded VoIP phones and 2 cordless VoIP phones any suggestions on what to get for either or both? |
| 20:52.42 | *** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com) |
| 20:52.50 | FuriousGeorge | anyone looking to sell any fxo/fxs/tdmp? |
| 20:53.00 | FuzzyCat | R3DB0x, where are you located? |
| 20:53.06 | R3DB0x | Tx USA |
| 20:53.06 | afrosheen | for cordless get some iaxy's and plug regular cordless phones into them |
| 20:53.23 | jarrod | that doesnt list a disconnect tone |
| 20:53.26 | opus_ | redbox - Polycom Ip501's at $179 each, and two ATA's $69each, with cordless phone bases $59/each (at fry's) |
| 20:53.28 | afrosheen | for corded..polycom ip501s |
| 20:53.31 | FuzzyCat | ahhh shame... instead of wireless voip you could use dect, but iirc the US doesn't have it |
| 20:53.53 | *** join/#asterisk random_user324 (n=random@rrcs-24-227-53-138.se.biz.rr.com) |
| 20:54.12 | random_user324 | afrosheen: so about 500 calls monthly aveaging 1500 minutes |
| 20:54.15 | afrosheen | the polycoms are somewhat retarded to set up but they're a set and forget type fo deal |
| 20:54.25 | R3DB0x | ok |
| 20:54.26 | afrosheen | random_user324: ha nice nick |
| 20:54.35 | FuzzyCat | lol |
| 20:54.42 | R3DB0x | i dont have a fry's near me....the url is outpost.com right? |
| 20:54.50 | random_user324 | afrosheen: got disconnected... i hate that |
| 20:54.51 | afrosheen | R3DB0x: skip that, get the iaxys |
| 20:54.54 | opus_ | well, anywhere that sells a good cordless phone |
| 20:55.07 | afrosheen | yeah |
| 20:55.23 | afrosheen | random_user324: so how much bandwidth do you guys use for data currently and what is your t1 costing you |
| 20:55.26 | opus_ | you can also hook up CAT5 to a wireless DVD base station and use the transponder with the IAXY within 100 feet. (hehe) |
| 20:55.29 | R3DB0x | afrosheen, where can i buy the iaxy |
| 20:55.35 | brad_mssw | the UTstarcom F1000 works fine for wifi style cordless ... except for attended transfers :/ |
| 20:55.39 | afrosheen | R3DB0x: digium.com |
| 20:55.44 | R3DB0x | k |
| 20:56.56 | *** join/#asterisk royth (n=royth@200.121.129.178) |
| 20:57.20 | afrosheen | the iaxy's are cool to send home with people for remote extensions as well since iax penetrates firewalls |
| 20:57.30 | afrosheen | deals with nat gracefully actually |
| 20:57.35 | R3DB0x | afrosheen, may i msg you |
| 20:57.38 | afrosheen | sure |
| 20:57.55 | brad_mssw | well, except the iaxy's don't do g729 or g726 |
| 20:58.08 | FuzzyCat | PA168 |
| 20:58.13 | afrosheen | ulaw or alaw over cable/dsl is decent enough |
| 20:58.42 | brad_mssw | depends, on a 128/256 cheap-o dsl link, it's not that great |
| 20:59.12 | Hmmhesays | i just had to explain to a guy what a default route is |
| 20:59.14 | Hmmhesays | geebus |
| 20:59.20 | blitzrage | afrosheen: yah - it can be - I'm here. I think I make up 10% of the genious here |
| 20:59.22 | brad_mssw | Hmmhesays: where do you work |
| 20:59.23 | blitzrage | :D |
| 20:59.27 | FuzzyCat | save be geebus |
| 20:59.31 | FuzzyCat | mew |
| 20:59.33 | blitzrage | ok, I'm outta here - I got shit to do! |
| 20:59.34 | FuzzyCat | -w |
| 20:59.47 | Hmmhesays | brad_mssw, i shall not disclose that information, lets just say I won't be here much longer |
| 20:59.53 | Hmmhesays | they pissed me off a bit to much today |
| 21:00.00 | brad_mssw | heh, ok |
| 21:00.35 | twisted[asteria] | blitzrage, 10%? |
| 21:00.35 | brad_mssw | i deal with stupid people all day ... and these people call themselves developers too ... granted ... VB6, C#, RPG, crap |
| 21:00.38 | twisted[asteria] | blitzrage, thinking a little high are we? |
| 21:00.41 | *** join/#asterisk fungi (n=fungi@p54923383.dip0.t-ipconnect.de) |
| 21:01.08 | random_user324 | afrosheen: accounting is confusing me.. at one point they said they had a t1.. now im not so sure |
| 21:01.08 | Hmmhesays | Fuzzycat jobs yours |
| 21:01.29 | FuzzyCat | nota... |
| 21:01.30 | bjohnson | brad_mssw: he works at NASA |
| 21:01.32 | FuzzyCat | no ta |
| 21:01.38 | FuzzyCat | I only want your chair |
| 21:01.41 | afrosheen | random_user324: whoever is paying the bill should know :) |
| 21:02.07 | random_user324 | afrosheen: u would be surprised |
| 21:02.16 | afrosheen | random_user324: why are you considering a move to VOIP anyway |
| 21:02.29 | *** join/#asterisk Seba_soy (n=s@64.76.126.29) |
| 21:02.36 | Seba_soy | hi people |
| 21:02.48 | Seba_soy | somebody had problem with FAST-ENTERED dtmf? |
| 21:02.58 | Hmmhesays | i need a cigarette, now. |
| 21:03.03 | Seba_soy | :) |
| 21:03.04 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 21:03.12 | Seba_soy | I need a good Mate |
| 21:03.24 | Hmmhesays | do you have dangly bits? |
| 21:03.28 | FuriousGeorge | i need 2 CCs of cigarette, stat! |
| 21:03.35 | Seba_soy | dangly bits? |
| 21:03.39 | afrosheen | fruit basket? |
| 21:03.53 | random_user324 | afrosheen: because i need to become more indespensible than i already am.. :-) j/k |
| 21:03.53 | Hmmhesays | an outie |
| 21:04.02 | jarrod | does asterisk not support mgcp network disconnect? |
| 21:04.11 | jarrod | L/osi for 900ms |
| 21:04.22 | Seba_soy | someone had problems with FAST-ENTERED DTMF?, When I dial repeated numbers, Asterisk only take one. |
| 21:04.26 | afrosheen | omg people use mgcp? |
| 21:04.30 | FuzzyCat | errm... |
| 21:04.36 | jarrod | sometimes it is necessary |
| 21:04.37 | FuzzyCat | it's not even plugged in |
| 21:04.49 | afrosheen | random_user324: seriously though |
| 21:04.55 | Seba_soy | I am using 1.2.0beta1 |
| 21:05.07 | afrosheen | random_user324: what does your company seek to gain |
| 21:05.33 | random_user324 | afrosheen: because in my opinion it offers more flexibility.. lower tco.. |
| 21:05.39 | *** join/#asterisk Netgeeks_ (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 21:05.59 | afrosheen | random_user324: ok so you're clear on why you need it, have you already sold it to your boss? |
| 21:06.02 | Seba_soy | can somebody tell me why asterisk does not takes all dtmf when I pressed it too fast? |
| 21:06.22 | Seba_soy | specially repeated numbers |
| 21:06.24 | random_user324 | afrosheen: yeah.. they want it.. i just need to do a cost/time analysis |
| 21:06.25 | afrosheen | Seba_soy: enable mod_caffeine in line 41 of /etc/asterisk/sleepy.conf |
| 21:06.31 | Ariel_ | Seba_soy, is it asterisk or you phone not sending it to it. |
| 21:06.45 | brad_mssw | Seba_soy: how are you connected to asterisk ? Zap, IAX, SIP ? and is it an ATA? |
| 21:07.04 | afrosheen | random_user324: I'll tell you right now you'll probably spend a few months getting it perfect |
| 21:07.46 | devel | so how carried away can i get with my dialplan in extensions.conf? can i have 400 sip users, each with their own incoming context and sub-plans, so i might have as many as 1000 [context] entries in extensions.conf? |
| 21:07.49 | afrosheen | random_user324: probably the most important thing is your linux distro of choice and your hardware |
| 21:08.00 | Seba_soy | SIP, phone is a PSTN phone, call is received by a Cisco 5300, and then routed to an IVR in Asterisk USING SIP. IVR ask for a code, but when user dial fast dtmf and they are repetead, asterisk only recongnizes one of them |
| 21:08.04 | Seba_soy | EXAMPLE: |
| 21:08.11 | Seba_soy | if I diala 11112222 |
| 21:08.15 | Seba_soy | astersisk takes |
| 21:08.16 | Seba_soy | 12 |
| 21:08.27 | FuzzyCat | dtmfmode |
| 21:08.40 | afrosheen | yeah |
| 21:08.41 | distortio | are you using rfc2833? |
| 21:08.42 | Seba_soy | rfc2833 and CISCO-RTP |
| 21:08.47 | random_user324 | afrosheen: ive been trying to move to gentoo because ive 'heard' its fast, better, whatever.. i still like debian however |
| 21:08.57 | *** join/#asterisk mariogamboa (n=sudaikdd@201.138.152.159) |
| 21:09.02 | opus_ | <Seba_soy> can somebody tell me why asterisk does not takes all dtmf when I pressed it too fast? |
| 21:09.03 | Seba_soy | it only happens with repeated numbers |
| 21:09.04 | afrosheen | random_user324: stick with what you know, that's rule #1 when going into something new |
| 21:09.11 | opus_ | RFC2833 or inband? |
| 21:09.29 | random_user324 | afrosheen: well we currently spend about $900 a month on phone... so im hoping to cut that down a little |
| 21:09.29 | Seba_soy | RFC2833 on asterisk side, DTMF-RELAY CISCO-RTP in cisco 5300 |
| 21:09.29 | distortio | ive seen the repeated number problem a lot with inband |
| 21:09.39 | opus_ | seba_soy who is your provider? |
| 21:09.43 | Seba_soy | using g729 |
| 21:09.46 | afrosheen | random_user324: you'll definitely accomplish that, even if you buy another t1 dedicated to voice |
| 21:09.59 | distortio | set the cisco to rfc2833 |
| 21:10.08 | afrosheen | random_user324: another key is to find a good IAX trunk provider to handle your outbound and inbound calls |
| 21:10.09 | random_user324 | afrosheen: so looking at the contract.. it seems we have a fractional t1 (128k).. does that sound right ? (not to me) |
| 21:10.11 | Seba_soy | cisco does not have rfc2833 |
| 21:10.21 | afrosheen | fractional..big ouch |
| 21:10.25 | FuriousGeorge | i dont understand why everyone's setup wants to make the user dial 9 before getting an outside line. whats the benefeit of that? is that just a legacy thing from older pbx? people like to hit a number and hear a diatone again? |
| 21:10.32 | opus_ | Seba_soy does this happened only with a provider? |
| 21:10.36 | afrosheen | FuriousGeorge: yeah people expect that |
| 21:10.57 | afrosheen | FuriousGeorge: after we got our system up here we had people hitting 9 all the damn time |
| 21:11.05 | devel | FuriousGeorge, around here, we call those people "idiots" and tell them to get with it. |
| 21:11.07 | Seba_soy | opus_ if I use an aplicattion programmed in CISCO LANGUAGE it works ok |
| 21:11.11 | Ariel_ | FuriousGeorge, it's a PBX switch type system. IN asterisk you don't need that any more |
| 21:11.12 | FuriousGeorge | afrosheen: maybe cuz i dont spend my life in a cube but i kinda expect normal dialing |
| 21:11.17 | Seba_soy | TCL works OK |
| 21:11.27 | afrosheen | FuriousGeorge: creatures of habits users are. |
| 21:11.29 | opus_ | seba_soy what is cisco language? |
| 21:11.31 | FuriousGeorge | Ariel_: thats what i figured. devel: its the muscle memory |
| 21:11.41 | Seba_soy | TCL |
| 21:11.49 | opus_ | tcl/tk? |
| 21:11.50 | distortio | seba: rtp-nte RTP Named Telephone Event RFC 2833 |
| 21:11.59 | brad_mssw | Seba_soy: sometimes they call rfc2833 AVT |
| 21:12.02 | Seba_soy | maybe my CISCO IOS is too old |
| 21:12.06 | distortio | set "dtmf-relay rtp-nte" in your peer |
| 21:12.14 | Seba_soy | I dont have that option |
| 21:12.15 | FuriousGeorge | so we all agree there is no advantage to the "dial 9 for an outside line thing" |
| 21:12.24 | Ariel_ | falalalalal argh I hate meetings so we can set the agenta for the next meeting. argh |
| 21:12.24 | opus_ | furiousgeorge nope |
| 21:12.27 | afrosheen | FuriousGeorge: it's what you'd call a legacy feature |
| 21:12.34 | FuriousGeorge | good |
| 21:12.39 | Ariel_ | FuriousGeorge, none |
| 21:12.41 | devel | FuriousGeorge, we use prefixes sometimes for forcing to a particular provider (i.e. voicepulse over voipjet, etc) |
| 21:12.48 | distortio | then try to use dtmf in h245 signaling, that is second best |
| 21:13.00 | opus_ | Seba_soy well I've seen your problem but your before |
| 21:13.03 | Seba_soy | #dtmf-relay ? |
| 21:13.03 | Seba_soy | <PROTECTED> |
| 21:13.03 | Seba_soy | <PROTECTED> |
| 21:13.03 | Seba_soy | <PROTECTED> |
| 21:13.04 | FuriousGeorge | devel: i see, rather than breaking it down by every possible are code |
| 21:13.16 | FuriousGeorge | *area code |
| 21:13.22 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 21:13.28 | opus_ | seen yo problem before ,jezus christ |
| 21:13.29 | Seba_soy | so, I can try 245-signal? |
| 21:13.32 | devel | something like that, or for "testing" |
| 21:13.40 | FuriousGeorge | gotcha |
| 21:13.52 | Seba_soy | Actually I set CISCO-RTP |
| 21:15.18 | jaybuffet | afrosheen: seems we have 12 local numbers and a toll free.. |
| 21:15.20 | afrosheen | jaybuffet: you're definitely going to need alot more bandwidth, probably a full t1 just for voice |
| 21:15.42 | Seba_soy | distortio: so, you say I have to try 245-SIGNAL? |
| 21:15.46 | afrosheen | jaybuffet: I think we pay about $700 a month for bonded t1's = 300K/sec :) |
| 21:16.41 | *** join/#asterisk [hC] (n=hardcore@8.10.2.81) |
| 21:16.43 | jaybuffet | afrosheen: but as u can see we dont take that many calls (500 calls, 1500min/month avg) |
| 21:16.53 | Seba_soy | somebody has a A-Z list? |
| 21:17.00 | Seba_soy | I want to terminate calls on all world |
| 21:17.04 | afrosheen | jaybuffet: you said an average of 8-10 people on the phone at any given time, is that internally or externally |
| 21:17.05 | Seba_soy | prepay :) |
| 21:17.24 | jaybuffet | afrosheen: i would say half and half |
| 21:18.03 | distortio | seba, ive got what chu need |
| 21:18.29 | Seba_soy | yes= |
| 21:18.31 | Seba_soy | yes? |
| 21:18.39 | afrosheen | jaybuffet: so...8 calls simultaneously inbound and outbound, on an iax trunk..you'll use something like 50-60KB/sec for that, both directions |
| 21:19.01 | afrosheen | jaybuffet: maybe up to 110KB or so, depending on codecs, trunking efficiency, all that jazz |
| 21:19.01 | distortio | i havent registered so i cant msg you :D |
| 21:19.16 | Seba_soy | distortio: I tried h245-signal, with same results... :/ |
| 21:19.35 | afrosheen | someone has a big write-up of codec vs. codec somewhere |
| 21:19.44 | jaybuffet | afrosheen: so i guess that makes sense that we currently have an int-t 128kb (don't know what int-t stands for, but its on the phone bill) |
| 21:19.51 | *** join/#asterisk adam1234blah (n=adam@ip24-250-235-137.ga.at.cox.net) |
| 21:20.56 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 21:20.56 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:21.07 | afrosheen | jaybuffet: fractional t1, and yeah you'll need to upgrade that to a full t1 |
| 21:21.17 | afrosheen | jaybuffet: what are you paying for it now |
| 21:21.55 | jaybuffet | afrosheen: $200 |
| 21:22.18 | Seba_soy | I am looking for terminate to SPAIN, ITALY, EUROPE, CHILE, LATINAMERICA, USA |
| 21:22.45 | afrosheen | jaybuffet: pretty high for that much bandwidth, sdsl is cheaper at those levels I think..anyway plan on spending around $500-$600 per month for your full t1 |
| 21:22.59 | afrosheen | jaybuffet: where are you in the US |
| 21:23.31 | distortio | seba- msg me your email |
| 21:23.36 | jaybuffet | afrosheen: florida |
| 21:23.43 | afrosheen | btw is anyone offended by all this chit chat, should we take this private? |
| 21:23.57 | brad_mssw | jaybuffet: where in florida ? |
| 21:24.03 | Nugget | You guys are preventing me from starting an argument about mysql. |
| 21:24.12 | darkskiez | postgres! |
| 21:24.17 | jaybuffet | brad_mssw: tampa |
| 21:24.32 | afrosheen | emacs! |
| 21:24.41 | afrosheen | no wait, vim |
| 21:24.43 | twisted[asteria] | VIM |
| 21:24.47 | brad_mssw | jaybuffet: you should be able to get some cheap bandwidth around there, maybe even fiber to the door |
| 21:24.51 | darkskiez | yate! |
| 21:25.11 | brad_mssw | jaybuffet: sdsl should also be cheap around there |
| 21:25.15 | jaybuffet | brad_mssw: its close.. but not for business.. thats what they say |
| 21:25.44 | brad_mssw | jaybuffet: also, some cable companies (like Cox here in gainesville) have business offerings |
| 21:25.57 | brad_mssw | jaybuffet: for real cheap ... |
| 21:26.01 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 21:26.08 | afrosheen | comcast does too but I'm not really proud of their uptime if my home service is any indication |
| 21:26.30 | brad_mssw | heh, usually the business offerings are better |
| 21:26.32 | afrosheen | if you don't get 9 9's in your SLA forget it |
| 21:26.42 | Seba_soy | see you people in a while... |
| 21:26.44 | *** part/#asterisk Seba_soy (n=s@64.76.126.29) |
| 21:26.45 | brad_mssw | yeah, true ... cox does offer slas |
| 21:26.53 | devel | do the 9s have to all be together? :) |
| 21:27.00 | afrosheen | preferably with a decimal somewhere |
| 21:27.02 | jaybuffet | i mean i have roadrunner now.. no downtime issues... 6mb/768kb.. |
| 21:27.17 | afrosheen | jaybuffet: yeah but it's lopsided, no good for * |
| 21:27.20 | *** join/#asterisk santiago (n=santiago@63.245.87.180) |
| 21:27.34 | afrosheen | wonder if RR has business offerings in your area |
| 21:27.45 | jaybuffet | afrosheen: ok.. but basically t1 isnt my only option |
| 21:28.10 | adam1234blah | anyone from nufone here? |
| 21:28.36 | afrosheen | jaybuffet: yeah bandwidth is bandwidth, it's just that traditionally t1's have been super reliable |
| 21:28.55 | jaybuffet | afrosheen: thats what i figured.. but cost a premium too |
| 21:29.04 | afrosheen | well you get what you pay for once in awhile |
| 21:29.09 | *** join/#asterisk Nir_S (n=Nir@84.94.49.221.cable.012.net.il) |
| 21:29.10 | Nir_S | hey all |
| 21:29.15 | afrosheen | any company with a nice SLA will get my business though |
| 21:29.19 | brad_mssw | jaybuffet: http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption |
| 21:29.30 | brad_mssw | jaybuffet: just for reference |
| 21:29.36 | Nir_S | any AMA-BUF masters around here ? |
| 21:29.37 | jaybuffet | brad_mssw: thanks |
| 21:30.03 | *** part/#asterisk spackle (n=spackle@209.234.83.19) |
| 21:30.14 | jaybuffet | afrosheen: so what are my phone options if i want to go pure voip |
| 21:30.23 | jaybuffet | does that make sense |
| 21:30.29 | afrosheen | jaybuffet: http://rrbiz.com/RoadRunner/sec_formatted.asp?TRACKID=&CID=985&DID=1202 |
| 21:30.43 | afrosheen | phone options as in...handsets? |
| 21:30.58 | Ariel_ | polycom, polycoms, polycom... great phones |
| 21:31.09 | afrosheen | yeah we love polycoms, it's all we have here |
| 21:31.32 | knight_ | polycom's are not really worth the money imho |
| 21:31.36 | jaybuffet | afrosheen: sorry.. didnt finish.. phone providers |
| 21:31.39 | SwK[Work] | damnit |
| 21:31.40 | knight_ | overpriced junk |
| 21:31.43 | SwK[Work] | y0 knight_ |
| 21:31.46 | knight_ | hey SwK! |
| 21:31.51 | afrosheen | knight_: how many do you have |
| 21:31.55 | SwK[Work] | anyone have the default passwords for audiocodes MP10Xs handy? |
| 21:32.08 | knight_ | afrosheen, used them for years... they're nice, but not worth the cash. |
| 21:32.14 | jaybuffet | afrosheen: we have business RR right now.. thats our net connection |
| 21:32.17 | afrosheen | knight_: they're not that expensive |
| 21:32.27 | knight_ | afrosheen, you think they aren't overpriced? |
| 21:32.28 | afrosheen | jaybuffet: oh..then what's the fractional t1, is that through RR? |
| 21:32.41 | knight_ | SwK, nope |
| 21:32.49 | afrosheen | knight_: for a business, our 500's and 501's for less than $200 is a bargain |
| 21:32.54 | FuriousGeorge | does nufone allow you to call toll free's other voip numbers like sipphone? |
| 21:32.54 | afrosheen | each |
| 21:32.57 | jaybuffet | afrosheen: the frac t1 is dedicated to voice currently.. the RR is for our inernet |
| 21:32.57 | *** join/#asterisk mwgbc (n=junkmail@adsl-71-132-212-248.dsl.pltn13.pacbell.net) |
| 21:33.07 | Ariel_ | FuriousGeorge, yes |
| 21:33.12 | afrosheen | jaybuffet: oh so you guys have a channel bank? |
| 21:33.15 | Ariel_ | 48 state tollfree |
| 21:33.34 | FuriousGeorge | Ariel_: nice b/c sipphone has been ticking me off |
| 21:33.36 | jaybuffet | afrosheen: i believe so.. thats the 2 boxes with tons of phone lines in and out |
| 21:33.55 | mwgbc | I am having problems with NVBackgroundDetect. It is not transfering to the talk exten on a simple "hello" Any ideas? |
| 21:34.18 | afrosheen | jaybuffet: sounds about right |
| 21:35.06 | Ariel_ | FuriousGeorge, sipphone sometimes works sometimes does not for inbound on my setup. don't know why? |
| 21:35.35 | jaybuffet | afrosheen: is that good bad.. or just says a little more about our setup |
| 21:35.42 | *** join/#asterisk PyroSteve (n=pyrostev@24-159-79-219.dhcp.jcsn.tn.charter.com) |
| 21:35.49 | PyroSteve | hey |
| 21:35.58 | Ariel_ | ahh meeting over..... finally I was almost sleeping.... |
| 21:35.58 | PyroSteve | i figured out my damn dtmf problem |
| 21:36.11 | PyroSteve | and its not an easy dtmfmode issuse |
| 21:36.17 | mwgbc | Does anyone know anything about NVBackgroundDetect? Documentation is spotty |
| 21:36.21 | Ariel_ | PyroSteve, what dtmfmode problem |
| 21:36.31 | Ariel_ | mwgbc, not I |
| 21:36.38 | brad_mssw | mwgbc: http://www.voip-info.org/tiki-index.php?page=NVBackgroundDetect |
| 21:36.39 | *** join/#asterisk rg1_ (n=rg1@mail.airlinksystems.com) |
| 21:36.55 | FuriousGeorge | Ariel_: whats ticking me off about them currently is that ive tried to open accounts with 3 different emails and im not getting activations |
| 21:37.02 | PyroSteve | Ariel_, well i seem to get random problems with remote IVRs detecting my dtmf tones |
| 21:37.03 | mwgbc | brad_mssw: Yes, that is what I was referring to as being spotty |
| 21:37.07 | brad_mssw | mwgbc: should be pretty straight forward, no ? I haven't used it, but was thinking about doing so ... |
| 21:37.20 | PyroSteve | Ariel_, so after hours of troubleshooting |
| 21:37.23 | brad_mssw | mwgbc: what is it doing if it's not jumping to the 'talk' exten ? |
| 21:37.29 | rg1_ | can someone direct me to a URL/other for setting up a Polycom 501 IP phone on asterisk - I just got mine in today and no manual :( |
| 21:37.36 | mwgbc | brad_mssw: I am having problems with it transfering to the talk exten on a simple "hello" |
| 21:37.52 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 21:37.55 | PyroSteve | i found that IVRs that wont detect my key presses is because asterisk doesn't know the call is answered |
| 21:37.57 | brad_mssw | mwgbc: it says it will continue to the next priority if nothing is detected, or the 'fax' priority if 'fax' is detected ... |
| 21:38.01 | afrosheen | jaybuffet: it's good to know, you have some options...but you'll probably be converting that voice fractional t1 to a data full t1 |
| 21:38.08 | mwgbc | brad_mssw: It is dumping to the next dialed call |
| 21:38.13 | PyroSteve | and those calls are SIP calls going through broadvoice |
| 21:38.26 | PyroSteve | and ones that do work |
| 21:38.29 | brad_mssw | mwgbc: not sure what you mean |
| 21:38.41 | Ariel_ | rg1_, http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones#comments |
| 21:38.46 | rg1_ | thx ariel |
| 21:38.58 | mwgbc | brad_mssw: Yes, but I am saying hello, it is just not picking it up. I am using Asterisk in an autodialer sense |
| 21:39.06 | PyroSteve | the asterisk console says that the call is answerd and attempts a native bridge |
| 21:39.21 | jaybuffet | afrosheen: so you wouldnt recommend getting rid of the t1 and going with business class cable ? |
| 21:39.33 | brad_mssw | mwgbc: ok, what does your section of dialplan look like ? |
| 21:39.34 | Ariel_ | PyroSteve, so the problem is bv |
| 21:39.35 | *** join/#asterisk parky (n=kvirc@p5083E348.dip.t-dialin.net) |
| 21:39.54 | PyroSteve | so im not sure if its a remote PBX problem thats not communicating with bv or if its bv |
| 21:39.57 | afrosheen | jaybuffet: it's a toss-up, if you can call both of them and play them against each other for price, length of contract, etc. you'll probably have a winner |
| 21:40.23 | jaybuffet | afrosheen: thanks for your help |
| 21:40.24 | PyroSteve | because the problem never happends on call that Asterisk can detect weather the call is answered |
| 21:40.31 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 21:40.39 | PyroSteve | and that call status is passed to my xlite client |
| 21:40.44 | brad_mssw | mwgbc: I mean you should probably have right after NVBackgroundDetect() Goto(talk,1) or similar ... because it doesn't specifically detect talking doesn't mean anything ... it's doing it's detection off silence patters |
| 21:40.57 | PyroSteve | and when the buttons are pressed on xlite |
| 21:41.12 | PyroSteve | they dont get sent through unless the call is answered |
| 21:41.47 | jaybuffet | what do i need in order to go pure voip (what are my provider options, vonage, ?) |
| 21:42.13 | mwgbc | brad_mssw: Yeah, I was trying to avoid that due to answering machines. If it detected "Hello" -silence- then I thought it would go to talk, but if it continued for a longer period of time then it would mean an ans machine. |
| 21:43.14 | brad_mssw | mwgbc: wait, you're using NVBackgroundDetect() when dialing out ? |
| 21:43.30 | mwgbc | brad_mssw: yes, is there another way? |
| 21:43.54 | mwgbc | brad_mssw: I mean to get the detection I want for outbound? |
| 21:44.01 | PyroSteve | the same thing dtmf problems happens with voicepulse |
| 21:44.03 | *** join/#asterisk _mwoodj_ (n=mwoodj@24.96.145.218) |
| 21:44.08 | *** join/#asterisk wifi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 21:44.20 | brad_mssw | mwgbc: seems like there was another command ... hold on |
| 21:45.32 | brad_mssw | mwgbc: nvfaxdetect() maybe ?? |
| 21:45.49 | brad_mssw | or did you need to outgoing audio? |
| 21:46.19 | *** join/#asterisk huslage__ (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
| 21:46.48 | brad_mssw | mwgbc: there is a topic on basic answering machine detection here : http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20BackGroundDetect |
| 21:50.54 | *** join/#asterisk dflow (i=pch@yennefer.sisco.pl) |
| 21:51.12 | mwgbc | brad_mssw: This is for an autodialer. What it does is spawn a call using python and psql then answer and monitor for fax or voice. |
| 21:51.23 | mwgbc | brad_mssw: Thanks for the link |
| 21:51.28 | *** part/#asterisk dflow (i=pch@yennefer.sisco.pl) |
| 21:51.56 | nick125_lappy | when i try to make a call out to my asterlink trunk: |
| 21:52.10 | nick125_lappy | Sep 6 14:44:57 NOTICE[9262]: chan_sip.c:8985 handle_response: Failed to authenticate on INVITE to '"Host41 Communications" <sip:******@66.92.33.116>;tag=*********' |
| 21:52.50 | bkw_ | nick125_lappy, lets go to #asterlink and we'll get you fixed |
| 21:52.54 | nick125_lappy | the first ****** is the asterlink username |
| 21:52.54 | nick125_lappy | k |
| 21:53.08 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 21:54.10 | *** join/#asterisk shido6 (n=curtis@d57-81-54.home.cgocable.net) |
| 21:54.46 | afrosheen | doh |
| 21:59.25 | opus_ | fucking i hate dealing BULL SHIT fucking VOIP providers |
| 21:59.54 | shido6 | I know what the f$%C you mean |
| 22:00.47 | opus_ | am I smoking crack or shouldn't poviders set the ToS bit? |
| 22:00.48 | JessicaX^ | Woah |
| 22:00.49 | JessicaX^ | Easy now |
| 22:02.10 | PyroSteve | yeap |
| 22:02.16 | PyroSteve | With XLITE |
| 22:02.37 | fugitivo | wtf?? |
| 22:02.42 | PyroSteve | Ive found the some IVRs will not actually answer the call |
| 22:03.00 | PyroSteve | and xlite does not think the call is connected |
| 22:03.06 | PyroSteve | and dtmf tones will not be sent |
| 22:03.25 | *** join/#asterisk rob314[laptop] (n=rob314[l@cpe-65-185-169-238.neo.res.rr.com) |
| 22:03.40 | PyroSteve | but with other UAs like the SIpura SPA-841, the phone doesnt care and will still send dtmf tones |
| 22:03.56 | PyroSteve | and thats the results and broadvoice |
| 22:04.02 | PyroSteve | but with voicepulse |
| 22:04.36 | PyroSteve | niether will my xlite or my spa-841 can dial dtmf |
| 22:04.53 | PyroSteve | which in that case, its voicepulse that filters the digits |
| 22:05.05 | PyroSteve | because the call is 'aswered' |
| 22:05.15 | PyroSteve | must be something to screw tolls |
| 22:06.55 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 22:06.58 | PyroSteve | anybody ever run into this problem ? |
| 22:07.13 | *** join/#asterisk derek_1234 (n=derek@203.167.203.10) |
| 22:07.29 | *** part/#asterisk santiago (n=santiago@63.245.87.180) |
| 22:07.35 | derek_1234 | I have followed the asterisk list for a couple of years. |
| 22:07.49 | derek_1234 | in that time, I have been severely saddeded by the attitude. |
| 22:07.58 | derek_1234 | I have consulted with others. |
| 22:08.28 | derek_1234 | In the collective view, the asterisk devel & user lists has the worst attitude of all open source groups |
| 22:08.38 | FuzzyCat | is that directed at anyone in particular or are you ranting |
| 22:08.45 | derek_1234 | Asterisk people - be ashamed. |
| 22:08.54 | derek_1234 | there are several people in mind. |
| 22:09.06 | file[muon] | only several? |
| 22:09.08 | pfn | derek_1234 who made you such an authority? shush |
| 22:09.10 | derek_1234 | My question is, and I am not ranting, is how can we improve the general attitude. |
| 22:09.12 | FuzzyCat | 383 ? |
| 22:09.16 | pfn | :p |
| 22:09.19 | FuriousGeorge | pots <--> <--> pots = no echo pots <--> <--> sip = consistent echo IAX2 <--> * <--> POTS = echo and jitter. i use gsm and its one concurrent conversation so it cant be bandwidth |
| 22:09.22 | opus_ | <derek_1234> In the collective view, the asterisk devel & user lists has the worst attitude of all open source groups |
| 22:09.26 | fugitivo | derek_1234: if you don't like the attitude, don't read it |
| 22:09.27 | opus_ | EXACTLY |
| 22:09.29 | FuriousGeorge | ive tried pinging the servers so its not latency |
| 22:09.37 | opus_ | you should try to file a bug!!! |
| 22:09.38 | FuriousGeorge | so what is the source of my terrible echo |
| 22:09.47 | derek_1234 | file one bug ? |
| 22:09.49 | derek_1234 | bah. |
| 22:09.55 | derek_1234 | file 100000 bugs. |
| 22:09.59 | derek_1234 | 100000 lines of code. |
| 22:10.09 | FuzzyCat | file a bug... poor file |
| 22:10.17 | fugitivo | derek_1234: don't use asterisk |
| 22:11.01 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 22:11.19 | derek_1234 | sigh - i would love not to use * |
| 22:11.20 | file[muon] | derek_1234: fork your own asterisk if you don't like this community!!! |
| 22:11.34 | opus_ | file & derek i'll help with that |
| 22:11.34 | fugitivo | derek_1234: then don't use it |
| 22:11.35 | FuzzyCat | actaully he's right... |
| 22:11.37 | derek_1234 | However, there does not appear to be any other reasonably close project. |
| 22:11.50 | FuzzyCat | you're all horrible people... and I don;tlove u any more.. |
| 22:11.51 | derek_1234 | Anyhow, why fork asterisk ? |
| 22:11.55 | fugitivo | derek_1234: buy a call manager from cisco |
| 22:12.00 | derek_1234 | I would end up with my own copy of crap code. |
| 22:12.30 | file[muon] | derek_1234: but it would be your own, your very own! |
| 22:12.31 | derek_1234 | by all standards of quality code, asterisk is 100% shit. |
| 22:12.40 | derek_1234 | no, it would not be my own. |
| 22:12.47 | derek_1234 | it would be part digium's |
| 22:13.26 | bkw_ | HEY don't drag my ass into this.. I'm over here collecting info for an FBI Subpoena |
| 22:13.33 | fugitivo | derek_1234: your precious! |
| 22:13.35 | bkw_ | don't make my include your info too |
| 22:13.36 | malcolmd | feed the troll....yum, yum.... |
| 22:13.47 | FuzzyCat | lol, FBI... hahhaha... |
| 22:13.59 | FuzzyCat | I'll just tell em you hacked me bkw_ |
| 22:14.01 | derek_1234 | fearless Band Idiots == FBI |
| 22:14.05 | FuzzyCat | ..again |
| 22:14.55 | malcolmd | mmmm, muffinz |
| 22:14.56 | derek_1234 | enough shit stirring. |
| 22:15.02 | mtgh | Does anyone in here work for an ISP, I need to talk to someone about katrina things |
| 22:15.02 | file[muon] | ;( |
| 22:15.06 | mtgh | Please PM me |
| 22:15.23 | derek_1234 | It appears you all agree with me: the list attitude is pretty poor, when compared to other projects. |
| 22:15.33 | opus_ | yup |
| 22:15.47 | opus_ | bunch of bandits!!! |
| 22:15.50 | derek_1234 | So - how do we improve the list attitude... |
| 22:15.53 | FuzzyCat | the list is poor because of the holier than thou posters |
| 22:15.57 | file[muon] | derek_1234: drugs? |
| 22:16.03 | derek_1234 | right now it rates as abysmal. |
| 22:16.05 | file[muon] | Free drugs with mailing list subscription. |
| 22:16.10 | derek_1234 | what do we do ? |
| 22:16.30 | FuzzyCat | there's nothing you can do... |
| 22:16.37 | opus_ | fork bomb |
| 22:16.39 | FuzzyCat | how many ppl on it... 1000 |
| 22:16.40 | t3t | file, are you using a sub-atomic particle to communicate on the channel? |
| 22:16.41 | FuzzyCat | 2000 |
| 22:16.42 | derek_1234 | Rubbish - that is defeatist... |
| 22:16.53 | file[muon] | t3t: you better believe it! |
| 22:16.54 | FuzzyCat | no it's not it's being practical |
| 22:16.55 | FuriousGeorge | so the pc im using to gain some * skills is an old celeron 1ghz. what are the odds that these echo problems i get when voip is introduced into the equation are caused by a slow pc? |
| 22:17.14 | derek_1234 | There is something we can do - next time someone is called a troll, they can demand and get an apology. |
| 22:17.20 | t3t | file, sure beats that piece of junk notebook you usually use |
| 22:17.24 | t3t | ;) |
| 22:17.26 | file[muon] | PFFT |
| 22:17.29 | FuzzyCat | thousands of people from all over the world in one place, they're bound to piss each other off |
| 22:17.32 | file[muon] | my Powerbook is not a junk notebook |
| 22:17.43 | fugitivo | moderate the lists! |
| 22:17.45 | malcolmd | ...i'm not apologizing |
| 22:17.47 | t3t | file, it is if you turn on FileVault |
| 22:17.53 | t3t | ...but anyway... |
| 22:17.53 | file[muon] | pfft no |
| 22:17.55 | FuzzyCat | your a troll malcolmd |
| 22:17.57 | FuzzyCat | :P |
| 22:18.01 | FuzzyCat | you're |
| 22:18.04 | malcolmd | FuzzyCat: and? :) |
| 22:18.14 | FuzzyCat | can I cross your bridge |
| 22:18.23 | opus_ | spelling nazi!!! |
| 22:18.42 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 22:18.50 | FuzzyCat | that's, "spelling Nazi" I think you'll find opus_ |
| 22:18.51 | FuzzyCat | :P |
| 22:19.00 | malcolmd | FuzzyCat: did you bring me a pumpkin muffin? |
| 22:19.11 | t3t | I'm looking for some assistance from the 'seasoned veterans' around here... |
| 22:19.15 | FuzzyCat | yes, but bkw_ stole it |
| 22:19.33 | malcolmd | d'oh :( |
| 22:19.48 | FuriousGeorge | does anyone think that echo problems with one conversation over voip could be caused by a slow pc? (1ghz celeron) i have a dsl connection and i use gsm, my latency to the server is 50 ms, i'm starting to think its not a bandwidth issue with my dsl |
| 22:19.49 | FuzzyCat | I think he's using it as bait for the FBI |
| 22:20.10 | opus_ | furiousgeorge - iax or sip, echo usually means the problem is on the opposite side btw |
| 22:20.18 | FuzzyCat | FuriousGeorge, my * box is lower spec, no echo here |
| 22:20.23 | *** join/#asterisk Beccara (n=Beccara@222-152-27-117.jetstream.xtra.co.nz) |
| 22:20.33 | fugitivo | FuriousGeorge: imposible, mine is p2 400 and i don't have echo with ip calls |
| 22:20.46 | FuriousGeorge | opus_: what you mean on the other side? like with my VOIP dialtone provider? |
| 22:20.57 | FuzzyCat | ip calls should never get echo |
| 22:21.01 | opus_ | furious if you hear echo, then the problem is on the other side |
| 22:21.17 | opus_ | sometimes echo cancellation algorithms don't work well with jitter buffers |
| 22:21.29 | t3t | Take a look at http://distributedhelpdesk.com and join me in #distributedhelpdesk if you can lend a hand in a brainstorming session |
| 22:21.41 | opus_ | and if you are using jitter buffer, which is new in asterisk 1.2 beta for SIP atleats (what about IAX? has this been in IAX2 for quite a while) |
| 22:21.56 | opus_ | then turn it off |
| 22:22.29 | opus_ | www.beowolfdebuggers.com wtf |
| 22:22.49 | *** join/#asterisk Hmmhesays (n=Neg@24-117-213-113.cpe.cableone.net) |
| 22:23.07 | FuriousGeorge | opus_: caller <> IAX <> * <> POTS <fxo> POTS (remote office) |
| 22:23.30 | FuriousGeorge | if i hear echo on the pots on the right, the problem must be with * and the FXo according to what you say right? |
| 22:23.33 | opus_ | do you get echo : * <> POTS <fxo> POTS (remote office) |
| 22:23.35 | opus_ | ? |
| 22:23.38 | opus_ | or do you get echo |
| 22:23.42 | opus_ | <PROTECTED> |
| 22:23.47 | *** join/#asterisk r0d3nt|m (i=anonymou@tinfoilhat.net) |
| 22:23.55 | opus_ | OK on the right side |
| 22:24.18 | opus_ | FuriousGeorge try finding a zaptel expert; I don't know how any of the analog stuff works./\ |
| 22:24.43 | opus_ | there are hundreds of posts/websites about asterisk/zapta/echoing problems |
| 22:24.48 | *** part/#asterisk mogorman (n=mogorman@digium.com) |
| 22:24.54 | *** join/#asterisk mogorman (n=mogorman@digium.com) |
| 22:24.57 | opus_ | probabaly one of the best covered areas |
| 22:25.07 | FuriousGeorge | no echo with POTS <fxs> <fxo> POTS (remote office) but i do get echo with POTS <fxs> <sip> EYEBEAM |
| 22:25.08 | Hmm-home | use external gateways with good echo can |
| 22:25.18 | FuriousGeorge | damn bold |
| 22:25.23 | drray | there was just a mailing list thread that covered a lot of zap echo, and what to do |
| 22:25.42 | visik7 | how can I connect a RTCP phone to asterisk ? |
| 22:25.52 | FuriousGeorge | opus_: no echo with POTS <fxs> ASTERISK <fxo> POTS (remote office) but i do get echo with POTS <fxs> ASTERISK <sip> EYEBEAM |
| 22:26.04 | *** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr) |
| 22:26.43 | visik7 | sorry wrong question |
| 22:27.02 | opus_ | POTS <fxs> ASTERISK <sip> EYEBEAM: try a faster box first, then try FXS echo cancellation google search if persisting. echo cancellation can take CPU power i think |
| 22:27.23 | jskcr | hy all |
| 22:27.25 | FuriousGeorge | should mention there are two firewalls b/w eyebeam and * in that last scenario, but there is only one firewall on the serverside when i go IAX <dsl> ASTERISK <fxo> POTS |
| 22:27.37 | file[laptop] | Hmmhesays@home!!! |
| 22:27.46 | *** join/#asterisk burton27 (i=mimx@w201.ljudmila.org) |
| 22:27.47 | FuriousGeorge | and there is terrible echo AND jitter that way |
| 22:27.53 | Hmm-home | that is me |
| 22:27.55 | Hmm-home | at home |
| 22:28.00 | file[laptop] | scary |
| 22:28.06 | opus_ | furiousgeorge - make sure the ToS bit is set for each packet going to and being sent from each destination. Check with tcpdump -i eth0 -vvvv host xxx.xxx.xxx.xxx look for [ToS 0x18] (0x18 = good) |
| 22:30.43 | knight_ | is there an asterisk@home channel? |
| 22:30.47 | FuriousGeorge | opus_: installing tcpdump |
| 22:31.11 | jskcr | knight_: noper you better off looking at the sourceforge site. |
| 22:31.23 | jskcr | there are asterisk@home forums there |
| 22:31.30 | knight_ | indeed there are |
| 22:32.18 | jskcr | if its a asterisk related question you can ask it here |
| 22:32.18 | knight_ | i'm a long time asterisk user, but wanted to test asterisk@home, and all the zap extensions ring busy... odd. |
| 22:32.22 | knight_ | nothing on the mailing lists... |
| 22:33.04 | jskcr | does zttool show any alarms on the channels and is you fxo's setup as z/1 and not z/g1? |
| 22:33.15 | knight_ | yeah, no groups |
| 22:33.24 | FuriousGeorge | opus_: i see nothing about ToS when i do a tcpdump. am i missing a switch |
| 22:33.28 | knight_ | there is a red alarm on a x100p card that i'm NOT using |
| 22:33.37 | opus_ | yup |
| 22:33.38 | knight_ | but no alarms on the TDP400M |
| 22:33.40 | FuriousGeorge | im not really familiar with the command |
| 22:33.59 | FuzzyCat | knight_, ur the 5th person with the same issue today... |
| 22:34.02 | opus_ | FuriousGeorge what version of asterisk? check grep 'tos' /usr/src/asterisk/configs/* |
| 22:34.05 | knight_ | Fuzzy, really? |
| 22:34.07 | FuzzyCat | are you using AMP ? |
| 22:34.16 | FuzzyCat | to configure it |
| 22:34.20 | knight_ | well, asterisk@home 1.5 (which uses AMP, yep) |
| 22:34.26 | FuzzyCat | well don't |
| 22:34.30 | knight_ | heh |
| 22:34.31 | FuzzyCat | problem solved |
| 22:34.41 | knight_ | it's not really that kind of situation |
| 22:34.48 | knight_ | i'm just testing asterisk@home |
| 22:35.06 | knight_ | i use regular asterisk by hand for production uses |
| 22:35.15 | FuzzyCat | it's the same (or sounds like it) that all the others had ... whcih is a 2second fix by editing the extensoions.conf |
| 22:35.18 | FuriousGeorge | opus_: ok i found ToS in a bunch of the * configs |
| 22:35.35 | knight_ | FuzzyCat, what is wrong with the extensions.conf? |
| 22:35.53 | FuzzyCat | iirc [ext-local] is broke (AMP uses the manager i/f to add it... |
| 22:35.54 | opus_ | FuriousGeorge if you want to be evil, set that bit on your bittorrents haha |
| 22:36.16 | FuzzyCat | do a show dialplan ext-local and it's there, but useless |
| 22:36.36 | FuzzyCat | mod the [default] and add ur own... |
| 22:36.49 | FuzzyCat | besides, use realtime :) |
| 22:37.22 | FuriousGeorge | opus_: im afraid i dont get it :( what do you mean by setting that bit? |
| 22:37.39 | FuriousGeorge | i notice i have tos=lowdelay in iax.conf |
| 22:37.55 | FuriousGeorge | and whats ToS in * have to do with bt |
| 22:37.58 | knight_ | ext-local looks fine |
| 22:38.24 | opus_ | nevermind. |
| 22:38.41 | knight_ | the real problem exists inside dialparties.agi |
| 22:39.21 | FuzzyCat | agi, for dialing, sigh |
| 22:39.43 | FuriousGeorge | so i would set the type of service in BT to the same info in the header of iax packets and i send that to people i file share with vs. the data they look for? is that it |
| 22:39.47 | *** join/#asterisk RoyK (n=roy@cm-80.111.22.187.chello.no) |
| 22:40.09 | *** join/#asterisk Aboulafia (n=adlp@wurzel-0.adlp.org) |
| 22:40.28 | opus_ | it would prioritize bt traffic on the internet, causing massive chaos.. cats sleeping with dogs, etc |
| 22:40.39 | FuriousGeorge | i see |
| 22:40.52 | FuriousGeorge | but everyone would have to do it |
| 22:41.13 | opus_ | yeah |
| 22:42.01 | FuzzyCat | tbh knight_ I cba looking too deep when it's a 3 second fix... |
| 22:42.26 | knight_ | heh |
| 22:42.39 | FuriousGeorge | i guess before i figure out what the source of this echo is im gonna have to try IAX <dsl> ASTERISK <fxs> POTS, and POTS <fxo> ASTERISK <sip> LOCAL SOFTPHONE |
| 22:42.57 | knight_ | exten => s,3,Macro(dial,${RINGTIMER},${DIAL_OPTIONS},${ARG2}) |
| 22:43.00 | knight_ | that's the culprit |
| 22:43.40 | FuzzyCat | was there a new release today or something? |
| 22:44.05 | opus_ | {$ARG2} |
| 22:44.26 | opus_ | {$USE_FUCKEDUP_PROGRAMMING_CONVETNION}=true |
| 22:44.41 | FuzzyCat | ${ not {$ |
| 22:44.41 | opus_ | sorry i'm very sinical today. i'll leave |
| 22:45.24 | FuzzyCat | I'll let you off on the spelling ;) |
| 22:45.43 | JDLSpeedy | FuzzyCat: u use Asterisk? |
| 22:45.44 | JDLSpeedy | hehe |
| 22:45.49 | FuzzyCat | yes |
| 22:46.03 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
| 22:46.06 | JDLSpeedy | I didn't know that |
| 22:46.28 | RoyK | ding |
| 22:46.35 | FuzzyCat | why else would I be here |
| 22:46.45 | FuzzyCat | except for insanity... |
| 22:46.58 | FuzzyCat | which is fair enough, but not the reason |
| 22:47.38 | FuzzyCat | yum |
| 22:48.31 | FuzzyCat | stop hiding behind jbot_ you norweigan pussy |
| 22:48.40 | *** join/#asterisk iLuvCandy (n=ohad@18-231-13-72.cosmoweb.net) |
| 22:48.46 | RoyK | fugitivo: sorry |
| 22:48.56 | nick125_lappy | lol |
| 22:49.04 | nick125_lappy | poor kitty... |
| 22:49.05 | FuzzyCat | hehehe,tabcompletion... too complex for RoyK |
| 22:49.07 | nick125_lappy | :P |
| 22:49.21 | RoyK | :{ |
| 22:49.52 | iLuvCandy | hi all, how do i forward one of my did's to a number ? |
| 22:50.01 | iLuvCandy | what conf. file do i need to use? |
| 22:50.10 | DrukenHME | 150lbs of krispy kreme's would be what ? 12 donuts? |
| 22:51.05 | nick125_lappy | no fighting, children |
| 22:51.29 | DrukenHME | yeah, children are defenceless, and should not be fought |
| 22:52.30 | FuzzyCat | ~googlefight RoyK and Wet Lettuce |
| 22:52.39 | FuzzyCat | pffft useless bot |
| 22:52.45 | nick125_lappy | yeah |
| 22:52.46 | RoyK | ~lart FuzzyCat |
| 22:53.04 | FuzzyCat | ScaredyCat catbot, googlefight RoyK and Wet Lettuce |
| 22:53.04 | FuzzyCat | CatBot Hoooyah! |
| 22:53.04 | FuzzyCat | CatBot oooff!! |
| 22:53.04 | FuzzyCat | CatBot Woooooomph! |
| 22:53.04 | FuzzyCat | CatBot Wet Lettuce wins! (0 Vs. 525,000) |
| 22:53.16 | nick125_lappy | LOL |
| 22:53.31 | RoyK | mohaha |
| 22:54.20 | FuzzyCat | ~seen RoyK_do_anything_useful |
| 22:54.24 | jbot | FuzzyCat: i haven't seen 'royk_do_anything_useful' |
| 22:54.30 | FuzzyCat | me neither |
| 22:54.48 | RoyK | ~seen FuzzyCat_being_close_to_sane |
| 22:54.50 | jbot | RoyK: i haven't seen 'fuzzycat_being_close_to_sane' |
| 22:54.54 | RoyK | right |
| 22:55.22 | Royk_Shagin_cows | moo |
| 22:55.24 | nick125_lappy | idiots... |
| 22:55.27 | FuzzyCat | hi |
| 22:55.48 | FuzzyCat | you know, I'm a total idiot |
| 22:55.53 | RoyK | Me too |
| 22:55.55 | nick125_lappy | you both are |
| 22:56.11 | FuriousGeorge | i concur :) |
| 22:56.17 | DrukenHME | ditto |
| 22:56.19 | DrukenHME | :) |
| 22:56.19 | RoyK | yes, I, RoyK am a big retard |
| 22:56.41 | iLuvCandy | anyone? what conf file do i need to modify - i will like to forward calls that come to my DID 4444 to go to 1877 343 3333 or something like that.. where do i modify the conf file? |
| 22:56.52 | iLuvCandy | or which conf file do i need to modify? |
| 22:57.17 | DrukenHME | iLuvCandy: you need to modify your dialplan, extensions.conf |
| 22:57.29 | *** join/#asterisk fuzzycat (n=ScaredyC@84.119.131.232) |
| 22:57.35 | DrukenHME | that is as far as i'm willing to help if your not willing to RTFM |
| 22:57.43 | fuzzycat | :) |
| 22:57.48 | RoyK | :) |
| 22:57.58 | fuzzycat | ~seen Royk_Shagin_cows |
| 22:57.59 | jbot | royk_shagin_cows <n=ScaredyC@84.119.131.232> was last seen on IRC in channel #asterisk, 2m 37s ago, saying: 'moo'. |
| 22:58.17 | FuzzyCat | I hate lc |
| 22:58.31 | FuzzyCat | good |
| 22:58.33 | RoyK | FuzzyCat: fucking pot head |
| 22:58.44 | FuzzyCat | :) |
| 22:58.58 | FuzzyCat | Seal killer |
| 22:59.05 | FuriousGeorge | pot heads are good people |
| 22:59.13 | FuriousGeorge | until they run out and want to smoke mine :) |
| 22:59.15 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-18-136.cybersurf.com) |
| 22:59.20 | *** join/#asterisk Roldyx (n=Roldyx3@201.255.100.115) |
| 22:59.38 | Roldyx | fugitivo, |
| 23:00.36 | Roldyx | fugitivo, negocios en vista |
| 23:00.46 | Roldyx | fugitivo, enviame un privado |
| 23:00.55 | RoyK | with a spoon |
| 23:01.17 | FuzzyCat | llama llama duck |
| 23:01.43 | FuriousGeorge | opus_: i guess one thing i could do is set up this number on a different * box and see what happens |
| 23:02.08 | FuzzyCat | slurp |
| 23:02.11 | FuzzyCat | :) |
| 23:02.22 | FuriousGeorge | but will i miss an important part of the thrid consecutive law and order rerun, thats the problem |
| 23:02.42 | Katty | beep. |
| 23:03.54 | FuzzyCat | you mean you gave me a did on your system...:P |
| 23:04.16 | DrukenHME | i wonder how old my asterisk install is.... |
| 23:04.37 | visik7 | run version in console |
| 23:04.39 | nick125_lappy | i think my asterisk install is -20 minutes old... |
| 23:04.55 | DrukenHME | Asterisk CVS-HEAD-03/30/05 |
| 23:04.58 | DrukenHME | not tooooo old... |
| 23:06.00 | RoyK | Druken: either update regularly or use 'stable' |
| 23:06.30 | RoyK | 'stable' means 'the bugs won't get fixed, it's as stable as windows 95' |
| 23:06.31 | DrukenHME | what would i want to upgrade regularly? so everything can be broken ? |
| 23:06.39 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 23:06.39 | opus_ | FuriousGeorge : yup. try every possible combination. use the process of elimination. make your boss pay for the pizza |
| 23:07.04 | drray | if it's just one box (not hooked to other asterisk boxes) then I see no need to upgrade from head, unless I want new features |
| 23:07.44 | FuzzyCat | so long as it's not a 2004 install |
| 23:08.03 | DrukenHME | well, it's connected to other boxes.... but it works the way i want it to... without breaking :) |
| 23:08.22 | drray | I'd not upgrade |
| 23:08.31 | drray | I don't upgrade my kernel either |
| 23:08.41 | drray | but my box is not on the intraweb |
| 23:08.42 | DrukenHME | i haven't seen any new "features" that i'd be intrested in, since that install... |
| 23:09.09 | drray | as opposed to smax? |
| 23:09.12 | FuzzyCat | http://lists.digium.com/pipermail/asterisk-security/2005-June/000034.html |
| 23:09.17 | RoyK | wtf? http://bugs.digium.com/view.php?id=5079 is CLEARLY a security issue, but noone seems to care....... |
| 23:09.45 | FuzzyCat | http://tigger.uic.edu/~jlongs2/holes/mpg123.txt |
| 23:10.08 | RoyK | shit. asterisk bug tracking is BAD |
| 23:11.03 | FuriousGeorge | opus_: works much better on this other pc, but i havent tested the comparable scenario at that other location which is a call starting on IAX and terminating locally via fxs |
| 23:11.06 | FuzzyCat | RoyK, only allow connections from 127.0.0.1 and use ssh |
| 23:11.31 | RoyK | FuzzyCat: that's a workaround, not a solution |
| 23:11.36 | FuriousGeorge | opus_: all signs seem to point to a combination of voip with fxo being the problem |
| 23:11.55 | FuzzyCat | it's a safe way to use the manager interface.. |
| 23:12.01 | FuzzyCat | it's the only way I'd use it |
| 23:12.16 | RoyK | FuzzyCat: having a manager interface means you should allow other addresses in |
| 23:12.46 | FuzzyCat | telnet is insecure - end of story |
| 23:12.49 | RoyK | FuzzyCat: i know that's the 'safe solution', but still it should be possible to open up for others |
| 23:12.57 | *** join/#asterisk ianm (n=ianm@63.224.101.51) |
| 23:13.12 | knight_ | hmm |
| 23:13.36 | DrukenHME | i belive i only allow connections to manager from 127.0.0.1 and telnet isn't even installed... |
| 23:13.41 | knight_ | no wonder, there's no association between zap channel and callerid |
| 23:13.41 | DrukenHME | SSH only! |
| 23:13.44 | knight_ | weird |
| 23:13.48 | knight_ | AGI Tx >> agi_callerid: unknown |
| 23:13.48 | RoyK | FuzzyCat: asterisk mananger supports md5 auth, and that's enough for most of us (them), so it should be usable for people outside |
| 23:14.00 | *** join/#asterisk santiago (n=santiago@63.245.86.254) |
| 23:14.08 | FuzzyCat | the manager i/f is just dangerous... |
| 23:14.40 | DrukenHME | i guess asterisk was running as root? |
| 23:14.45 | opus_ | RoyK haha Closing request due to no response. |
| 23:14.47 | FuzzyCat | ya |
| 23:15.02 | RoyK | opus_: it wasn't my bug |
| 23:15.05 | opus_ | yeah |
| 23:15.06 | opus_ | i know |
| 23:15.09 | opus_ | thats terrible. |
| 23:15.22 | opus_ | how about somebody forward that to bugtraq and see if it gets reopened |
| 23:15.41 | RoyK | opus_: please do |
| 23:15.49 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-14-146.cybersurf.com) |
| 23:16.06 | RoyK | bjohnson: høh? flere nordmenn her? |
| 23:20.01 | FuzzyCat | The Asterisk development team was prompt and responsive to the |
| 23:20.01 | FuzzyCat | vulnerability alert. Portcullis was provided with an alternate means of |
| 23:20.01 | FuzzyCat | contact additional to email, if it was to be required. Wade Alcorn |
| 23:20.01 | FuzzyCat | (Portcullis), Mark Spencer (Asterisk), Kevin Fleming (Asterisk) |
| 23:20.01 | FuzzyCat | cooperatively provided and verified a solution to the problem. |
| 23:20.10 | FuzzyCat | looks like it was fixed but not closed |
| 23:20.54 | fugitivo | FuzzyCat: what alert? |
| 23:21.05 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-181.sw.biz.rr.com) |
| 23:21.08 | FuzzyCat | manager i/f overflow bug |
| 23:21.11 | *** join/#asterisk tld_ (n=tld@253.80-203-96.nextgentel.com) |
| 23:21.21 | FuzzyCat | http://seclists.org/lists/bugtraq/2005/Jun/0185.html |
| 23:21.45 | *** join/#asterisk Snake-Eyes (n=blog@203.201.96.60) |
| 23:24.26 | *** join/#asterisk DeanLand (n=chatzill@pool-68-161-110-19.ny325.east.verizon.net) |
| 23:25.13 | DrukenHME | who knows how to create dictionaries for voice activated systems? |
| 23:25.48 | DeanLand | hello, anyone here who can help a confused chatzilla user -- cannot get te registration syntax so it works |
| 23:26.29 | opus_ | <PROTECTED> |
| 23:26.32 | DeanLand | I thin k it is a syntax problem -- but not for sure |
| 23:26.34 | opus_ | PAB phat ass belly |
| 23:26.36 | FuzzyCat | DeanLand, /msg nickserv register <password> |
| 23:26.39 | FuzzyCat | then |
| 23:26.47 | RoyK | FuzzyCat: that's pretty bad..... |
| 23:26.50 | FuzzyCat | DeanLand, /msg nickserv idenfity <password> |
| 23:27.05 | FuzzyCat | nasty SIP expoit ... |
| 23:27.27 | Juggie | anyone know if the 3.2ghz xeon's are 64bit |
| 23:27.37 | FuzzyCat | An attacker could send "Messages-Waiting: yes" messages to all phones using the SIP-environment. Almost every phone processes this status message and shows the user an icon or a blinking display to indicate that new messages are available on the voice box. If the attacker sends this message to many recipients in a huge environment, it would lead to server peaks as many users will call the voice box at the same time |
| 23:27.50 | file[muon] | Juggie: well if they say they have EM64T, then yes :) |
| 23:28.02 | Juggie | file, not sure server came in from HP today |
| 23:28.05 | Juggie | i havnt looked @ it yet |
| 23:28.11 | Juggie | i dont know what model it is or anything |
| 23:28.14 | DeanLand | FuzzyCat, here's teh response I got: No default action for objects of type IRCNetwork |
| 23:28.25 | Juggie | all i know is 2x3.2ghz, 4gb ram, 3x72gb hdd |
| 23:29.25 | DeanLand | AH! It worked -- 2nd time around --MUCH THANKS |
| 23:30.03 | opus_ | who is the guy here that goes by Corydon76 on the bug list.. |
| 23:30.44 | Mulvane | What do I need to do to have multiple sip phones on one voip line act like pots? |
| 23:30.46 | FuzzyCat | Corydon76 oddly enough |
| 23:32.20 | DrukenHME | Mulvane: use something like SER |
| 23:32.51 | Mulvane | Cool..I already installed that package..Just can't seem to figure out how to get it to act like that |
| 23:32.53 | RoyK | Mulvane: SER can prolly do that |
| 23:32.59 | ianm | Mulvane: Do you mean DID's into SIP? |
| 23:33.00 | Mulvane | Least I know I have the right thing |
| 23:33.12 | RoyK | Mulvane: but then, SER needs to be configured correctly |
| 23:33.21 | DrukenHME | Mulvane: well, before ya get all excited... define "act like a pots line" |
| 23:33.23 | DrukenHME | hehe |
| 23:33.31 | Mulvane | I have voip. I have 1 harware sip, and 2 software sips..I want them to act like a plain old telephone system |
| 23:33.48 | FuzzyCat | lol |
| 23:33.51 | DrukenHME | well, in what aspect? |
| 23:33.57 | ianm | How is your POTS coming in ? |
| 23:33.58 | RoyK | rotfl |
| 23:34.06 | Mulvane | In that when 1 rings, they all ring and I can answer from any of them |
| 23:34.17 | FuzzyCat | easy, don'tneed SER |
| 23:34.22 | ianm | right |
| 23:34.24 | DrukenHME | ok, then that is easy |
| 23:34.25 | ianm | no ser needed |
| 23:34.37 | FuzzyCat | you just do Dial(SIP/phone1&Phone2&phone3) |
| 23:34.44 | FuzzyCat | oops |
| 23:34.54 | FuzzyCat | you just do Dial(SIP/phone1&SIP/Phone2&SIP/phone3) |
| 23:34.59 | DrukenHME | just didn't know if you wanted a handoff solution, so you can move from extension to extension... cause pots can do that.... but voip can't... |
| 23:34.59 | ianm | Excellent stuff :) |
| 23:35.02 | FuzzyCat | and it'll dial all 3... |
| 23:35.12 | ianm | SIP can too :) |
| 23:35.18 | FuzzyCat | in the real sip world it's called forking |
| 23:35.31 | RoyK | FuzzyCat: it's SIP/asdf&SIP/1234 etc |
| 23:35.32 | opus_ | i smoke POTS everyday |
| 23:35.39 | ianm | Dial(SIP/phone1,14&SIP/phone2,14,SIP/phone3,20) etc.... |
| 23:35.42 | Mulvane | So is that an asterisk solution? |
| 23:35.48 | ianm | will ring phone 1 and 2 for 14 seconds |
| 23:35.48 | DrukenHME | ianm: really? you can ring all three, pick one up, and then pick another one up and join the conversation ? |
| 23:35.55 | ianm | then call phone 3 if nobody has answered |
| 23:35.56 | FuzzyCat | I already corrected it |
| 23:36.12 | ianm | LOL |
| 23:36.19 | ianm | It's amazing ;) |
| 23:37.11 | FuzzyCat | DrukenHME, no you can't do that |
| 23:37.19 | ianm | Phew !!! |
| 23:37.24 | DrukenHME | FuzzyCat: i'm awear :) |
| 23:37.31 | FuzzyCat | you asked... |
| 23:37.33 | FuzzyCat | :) |
| 23:37.51 | Mulvane | FuzzyCat So what I need is asterisk? |
| 23:37.55 | FuzzyCat | yes |
| 23:37.59 | DrukenHME | not really.... was sorta proving my point to ianm |
| 23:38.00 | Mulvane | Ok..Thanks |
| 23:38.00 | ianm | Would be a cool feature though ;) |
| 23:38.02 | FuzzyCat | that's all u need |
| 23:38.02 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 23:38.10 | FuzzyCat | ianm |
| 23:38.14 | FuzzyCat | that's possible... |
| 23:38.26 | FuzzyCat | you just do Dial(SIP/phone1&SIP/Phone2) |
| 23:38.29 | ianm | <point proved !!> LOL |
| 23:38.33 | FuzzyCat | Dial(SIP/Phone2) |
| 23:38.38 | FuzzyCat | gahh! |
| 23:38.53 | FuzzyCat | you just do: exten => _X.,1,Dial(SIP/phone1&SIP/Phone2) |
| 23:39.04 | FuzzyCat | exten => _X.,2,Dial(SIP/Phone2) |
| 23:39.06 | visik7 | how can I write in the dialplan the 'R' key ? |
| 23:39.15 | FuzzyCat | it's flashhook, you don;t |
| 23:39.23 | FuzzyCat | for the 2nd time today |
| 23:39.29 | DrukenHME | bruised |
| 23:39.56 | ianm | Thank you ~Ian goes off to feel better about himself !!! ~ |
| 23:40.07 | opus_ | dude, who here works on bug tracking in asterisk |
| 23:40.43 | *** join/#asterisk kshumard_home (n=ksh@pcp08979908pcs.huntsv01.al.comcast.net) |
| 23:41.05 | Ariel_ | opus_, every time I see your name it brings me back to the days of the FidoNet BBS world...I used to run some Opus BBS systems. |
| 23:41.29 | opus_ | Ariel I remember Opus BBS systems |
| 23:41.38 | FuzzyCat | aahhh fidonet :) |
| 23:42.14 | Ariel_ | yes fidonet was the well in a way the internet. It would exchange messages and files via dial up. |
| 23:43.10 | DrukenHME | or even the internet email.... |
| 23:43.21 | DrukenHME | there was a email gateway |
| 23:43.21 | visik7 | Ariel_ are u of the colinux team ? |
| 23:43.32 | Ariel_ | visik7, no |
| 23:43.53 | visik7 | oh sorry |
| 23:44.21 | *** join/#asterisk ManxPower (n=eric@ip68-225-97-156.br.no.cox.net) |
| 23:44.40 | Ariel_ | ManxPower, did you get your card td410p |
| 23:44.44 | Ariel_ | te410p |
| 23:45.11 | fugitivo | fidone! |
| 23:45.16 | fugitivo | t |
| 23:45.40 | Ariel_ | yes I remember those days...... |
| 23:45.41 | DrukenHME | oh, and in gecho |
| 23:45.46 | fugitivo | i was ascii artist |
| 23:45.53 | Ariel_ | ansi colors |
| 23:45.57 | fugitivo | ansi too |
| 23:46.04 | DrukenHME | that was the day.... |
| 23:46.13 | ianm | Ansi color makes me feel good !! |
| 23:46.16 | ianm | LOL |
| 23:46.19 | Juggie | i had a fidonet node :p |
| 23:46.23 | DrukenHME | the only half descient bbs i can find that runs on linux is schronet... |
| 23:46.25 | FuzzyCat | 2:257/802 |
| 23:46.26 | *** join/#asterisk Byte (i=byte@2001:4bd0:1000:0:202:44ff:fe47:d3ee) |
| 23:46.29 | FuzzyCat | :D |
| 23:46.35 | Juggie | haha |
| 23:46.37 | DrukenHME | BBS over telnet baby!! |
| 23:46.39 | Juggie | well i dont remember it |
| 23:46.48 | Ariel_ | So did I but that was back in 1985 called Impulse... don't remember the number |
| 23:46.51 | Juggie | i used to kick ass in BRE though |
| 23:47.13 | Ariel_ | Mine ran on Opus |
| 23:47.17 | wunderkin | here we go with bbs talk again :) |
| 23:47.21 | Juggie | heh |
| 23:47.39 | Ariel_ | wunderkin, what has it become now. |
| 23:47.44 | fugitivo | mine ran on pcboard |
| 23:47.48 | DrukenHME | wunderkin: you know you miss them |
| 23:48.04 | wunderkin | hehe yeah i ran a small bbs, for me and one other occasional user haha |
| 23:48.06 | DrukenHME | icky, pcboard... almost as bad as wildcat |
| 23:48.14 | wunderkin | yaya i used wildcat |
| 23:48.18 | wunderkin | and a few others |
| 23:48.23 | fugitivo | pcboard was gret |
| 23:48.26 | fugitivo | great |
| 23:48.26 | Ariel_ | wildcat oh no |
| 23:48.34 | wunderkin | i got out of it when the new wildcat came out |
| 23:48.42 | Juggie | i ran iniquity, the coolest software around :P |
| 23:48.50 | Juggie | www.iniquitybbs.com :) |
| 23:48.50 | fugitivo | you could program your own modules... what was that language called |
| 23:48.52 | fugitivo | ppe ? |
| 23:48.52 | DrukenHME | i remember iniquity |
| 23:49.00 | FuzzyCat | Ariel_, what's ur real name |
| 23:49.05 | ianm | wildcat !! WOW !!! |
| 23:49.07 | Ariel_ | ariel |
| 23:49.14 | FuriousGeorge | i bet Ariel_ 's real name is FuzzyCat |
| 23:49.48 | ianm | That was an awesome piece of OS for it's day.... |
| 23:49.52 | fugitivo | http://www.geocities.com/SiliconValley/3492/ |
| 23:49.55 | FuzzyCat | Ariel_, one moment please, processing |
| 23:49.58 | DrukenHME | Ariel_: do you have red hair and no legs? |
| 23:50.11 | Ariel_ | rofl |
| 23:50.20 | fugitivo | pcboard bbs accessible via telnet! |
| 23:50.31 | FuzzyCat | 2:227/16 |
| 23:50.46 | Ariel_ | I am just a short fat,loosing his hair, rest going gray Cuban American.... |
| 23:51.05 | DrukenHME | nothing like being honest.... |
| 23:51.06 | fugitivo | This page last updated : March 6th, 1996 |
| 23:51.10 | fugitivo | lol |
| 23:51.22 | FuzzyCat | Ariel_, 'The Digital Impulse'? |
| 23:51.34 | Hmm-home | design wise, is there any problem in asterisk setting 4000 unique global variables in extensions.conf? |
| 23:51.46 | Ariel_ | No it was in Washington State nodes called The Impulse |
| 23:51.55 | fugitivo | Hmm-home: why you need 4000 global variables??? |
| 23:52.00 | Hmm-home | irrelevant |
| 23:52.13 | Hmm-home | I don't care to explain |
| 23:52.20 | fugitivo | then why ask? |
| 23:52.27 | DrukenHME | hmm,.... bit of an attitude... |
| 23:52.29 | Ariel_ | Hmm-home, ok try and see if you run out of memeory |
| 23:52.52 | Hmm-home | attitude? lol funny |
| 23:53.08 | Ariel_ | it's coffee time... |
| 23:53.26 | DrukenHME | it's coca-cola time! |
| 23:53.34 | fugitivo | whisky time! |
| 23:53.35 | Hmm-home | i need to store some peristent data, not going to use agi, not going to use sql |
| 23:53.37 | FuzzyCat | correction - if it was just Impulse.. or MPULSe_BBS,Vorselaar,Hannes_Van_De_Vel) then it's 2:292/8210 |
| 23:53.39 | fugitivo | mmmmmm whisky |
| 23:53.55 | Ariel_ | Hmm-home, use dbput/dbget |
| 23:54.05 | Hmm-home | I'd rather not use the database |
| 23:54.37 | fugitivo | Hmm-home: i think it's not a clean design using 4000 global variables in extensions.conf, but, it's my opinion |
| 23:54.37 | Hmm-home | I can't imagine 4k global vars taking up more than a few meg |
| 23:54.47 | Hmm-home | fugitivo: it is temporary |
| 23:54.59 | fugitivo | Hmm-home: one time use? |
| 23:55.01 | Hmm-home | fugitivo: tell me why it is not clean design? |
| 23:55.16 | Hmm-home | fugitivo: just in place till I can finish the ldap stuff |
| 23:56.39 | fugitivo | Hmm-home: it's like programming, i won't put 4000 constants in one code |
| 23:57.30 | FuzzyCat | naaa, put em in the .h file ;) |
| 23:57.43 | *** join/#asterisk asterisk99 (n=chatzill@d141-65-173.home.cgocable.net) |
| 23:57.49 | Ariel_ | Hmm-home, I think there is a size limite on the extensions.conf |
| 23:57.57 | Hmm-home | in theory each global var should not take more than 10 bytes |
| 23:58.35 | Hmm-home | 10 x 4000 = 40000 |
| 23:58.50 | fugitivo | Hmm-home: if it's temporary you could try with agi and a database or agi and a text file if it doesn't work from extensions.conf |
| 23:59.02 | Hmm-home | fugitivo: thats not going to happen |
| 23:59.02 | asterisk99 | anyone knoow why I would get a "No rule to make target 'clean' . Stop.' error when I try 'make clean' for zaptel?????? |
| 23:59.23 | Hmm-home | i am not firing up an interpreter each time a call comes in |
| 23:59.37 | fugitivo | asterisk99: because there's no make clean for zaptel? |
