| 00:05.12 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 00:06.09 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:08.49 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:09.10 | *** join/#asterisk iq|mobile (n=iq@71-215-58-212.omah.qwest.net) |
| 00:10.33 | *** join/#asterisk Lino` (n=Lino@i577BD69D.versanet.de) |
| 00:15.53 | *** join/#asterisk tier_1 (n=tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 00:16.03 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-145-155.gdrpmi.dsl-w.verizon.net) |
| 00:16.08 | tier_1 | ok anyone here having issues with meetme om 1.2.9.1 |
| 00:17.10 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:19.48 | *** join/#asterisk W9SH (n=Steve_He@adsl-068-209-117-205.sip.asm.bellsouth.net) |
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| 00:26.55 | *** part/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 00:27.26 | loonacy | Is it possible to disable call waiting in Asterisk? I have 4 lines set up for call hunting, but it comes in on line one as call waiting even if you're on the line. |
| 00:27.35 | loonacy | I want it to go to BUSY instead. |
| 00:27.56 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 00:28.10 | *** join/#asterisk jroysdon (n=jroysdon@c-67-181-65-139.hsd1.ca.comcast.net) |
| 00:28.28 | malverian | Is there a known regression with 'asterisk -r -x' usage between Asterisk 1.2.7.1 and 1.2.9.1 ? |
| 00:28.44 | malverian | Using 'asterisk -r - x "some command"' only outputs the first line of the result. |
| 00:29.27 | malverian | In 1.2.9.1 anyhow |
| 00:29.32 | malverian | In 1.2.7.1 this was not the case. |
| 00:31.15 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:32.16 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 00:32.19 | generalhan | whats up all |
| 00:32.53 | generalhan | im having issues with a "native bridge" and i dont know whats going on can some one take a look at my pastebin and let me know if you see something off ?? |
| 00:32.53 | generalhan | http://generalhan.pastebin.ca/65334 |
| 00:35.24 | *** join/#asterisk dongs (n=HPUX@h193012.ppp.asahi-net.or.jp) |
| 00:36.11 | dongs | how hard would it be to setup something that makes asterisk call to destination number + then call me and connect those two calls? |
| 00:36.25 | dongs | i guess its sorta like callback except the other end is already dialed elsewhere |
| 00:36.34 | dongs | instead of giving a dialtone or something. |
| 00:37.39 | *** join/#asterisk Lino` (n=Lino@i577BD65C.versanet.de) |
| 00:39.39 | *** join/#asterisk W9SH (n=Steve_He@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 00:40.32 | malverian | Anyone else noticed this problem with CLI I mentioned? |
| 00:46.38 | *** join/#asterisk techie (n=gus@voipops.net) |
| 00:46.39 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 00:48.30 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 00:49.28 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 00:50.24 | dongs | so um lol? |
| 00:50.34 | dongs | how to setup callback + bridge to another number with asterisk |
| 00:50.40 | dongs | looked at the wiki, nothing useful there |
| 00:51.04 | *** join/#asterisk endrin (i=daed@CPE-70-92-75-238.new.res.rr.com) |
| 00:51.10 | bon | tell me when you find out dongs .) |
| 00:54.19 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 00:59.38 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 01:01.28 | dongs | i love the support on mailing list. the guy posts like 20 lines asking how to do callback and some idiot replies with like 10 words 'ya just drop a file in a proper qcall dir' |
| 01:01.39 | dongs | holy shit, if you're not going to bother acutally giving USEFUL info, why even bother? |
| 01:02.08 | dongs | if the person asking the question knew what hte fuck 'qcall dir' was why would he ask in the first place? |
| 01:02.35 | clyrrad | some people are ass holes what can you do |
| 01:02.59 | dongs | well the problem is i'm seeing this in like 99% of opensource projects |
| 01:03.33 | dongs | people asking for help with detailed explanations of what they want are answered with one-liners like 'rtfm' or 'search mailing lists' or some perhaps helpful, but totally useless to a newbie answer |
| 01:03.45 | bon | dongs: mail me when you find it plz |
| 01:03.54 | bon | gotta go to get some sleep now |
| 01:03.55 | bon | 3am |
| 01:03.58 | dongs | heh |
| 01:04.00 | dongs | ive got all day |
| 01:04.05 | bon | lucky you :) |
| 01:04.50 | bon | check your notice |
| 01:04.52 | bon | thx bye |
| 01:04.56 | dongs | yeah. |
| 01:07.28 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 01:08.57 | *** join/#asterisk Lino` (n=Lino@i577BD08E.versanet.de) |
| 01:10.23 | dongs | so hwere are some examples of dialing a specific number and bridging to callback instead of calling and giving a dialtone |
| 01:10.28 | dongs | ?? |
| 01:10.30 | *** join/#asterisk endrin (i=daed@CPE-70-92-75-238.new.res.rr.com) |
| 01:12.54 | *** join/#asterisk Oshuma (n=poonanny@rrcs-24-73-218-218.se.biz.rr.com) |
| 01:13.06 | Oshuma | hello |
| 01:13.22 | Oshuma | has anyone ever used TouchStar? |
| 01:13.45 | Oshuma | i'm hoping to replace TouchStar with Asterisk if at all possible |
| 01:13.52 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 01:13.56 | dongs | let me ask you this |
| 01:14.09 | Oshuma | trying to migrate our networks over to entirely linux based |
| 01:14.09 | dongs | is 'touchstar' a product from a company that actually has good support? |
| 01:14.17 | Oshuma | no |
| 01:14.21 | dongs | really. |
| 01:14.24 | Oshuma | their support is shabby at best |
| 01:14.25 | dongs | well i gues you have nothing to lose hten |
| 01:14.30 | Oshuma | plus their application sucks |
| 01:14.43 | dongs | good luck getting any help on asterisk |
| 01:14.56 | Oshuma | why's that? |
| 01:15.53 | dongs | because there is no support. |
| 01:16.32 | glm2k | must be a slow day |
| 01:17.34 | Oshuma | well i figure it's the same with any open source app |
| 01:17.41 | Oshuma | it's up to you to find documentation |
| 01:19.23 | generalhan | dongs: is this your first time in here ? |
| 01:19.39 | dongs | does that in any way relate to my question? i dont htink so |
| 01:20.09 | Oshuma | dongs: ...ok? |
| 01:20.20 | dongs | i wasnt replying to you? |
| 01:20.21 | trelane | dongs, considering generalhan is a respected regular and I've never seen you before and you're rambling about a third party product I think his question is on-topic |
| 01:20.35 | dongs | what |
| 01:20.42 | generalhan | thats irrelovant ... you cant come in here and expect people to solve your problems ... if they can GREAT. but you cant complain about not getting tech support for a product that you didnt pay for |
| 01:20.42 | dongs | i asked how to fucking do callback with asterisk |
| 01:20.55 | dongs | and told that Oshuma dude that hes better off using a product with support |
| 01:20.57 | trelane | " |
| 01:20.58 | trelane | <dongs> is 'touchstar' a product from a company that actually has good support?" |
| 01:20.59 | dongs | now whats the problem |
| 01:21.04 | trelane | that was the last question I saw |
| 01:21.04 | dongs | yeah look at the line above |
| 01:21.07 | trelane | ooh |
| 01:21.09 | trelane | I know how to help |
| 01:21.34 | trelane | dongs: http//www.catb.org/~esr/faqs/smart-questions.html |
| 01:21.43 | trelane | it's an incredibly useful resource if you want help from the general community |
| 01:21.58 | trelane | realize please that everyone here is a volunteer with the exception of a handful of digium employees |
| 01:22.16 | trelane | purchasing a digium card entitles you to installation and configuration support for the card from digium (not from #asterisk) |
| 01:22.26 | trelane | digium and others offer paid dialplan and advanced configuration support |
| 01:22.32 | dongs | right. |
| 01:22.39 | trelane | we do the best we can but if noone knows/whoever's here is tied up |
| 01:22.42 | trelane | I mean we do what we can |
| 01:22.57 | Oshuma | i don't really care as much for support, i usually don't have a problem finding the info i need |
| 01:23.02 | Oshuma | google, usenet, friends, etc. |
| 01:23.06 | dongs | good luck |
| 01:23.19 | Oshuma | i'm just trying to talk my company into migrating everything over to linux |
| 01:23.26 | *** join/#asterisk juice (n=juice@doc-72-47-32-128.maryville.mo.cebridge.net) |
| 01:23.27 | dongs | i think you're making a big mistake |
| 01:23.32 | dongs | unlesss you're leaving right after this migration |
| 01:23.33 | generalhan | dongs: why do you think that ? |
| 01:23.42 | dongs | and i'm being serious too. |
| 01:23.51 | dongs | typical. |
| 01:24.05 | Oshuma | just curious if asterisk can replace all the features we're using |
| 01:24.15 | Oshuma | so i have a better argument when i talk to my boss. ;P |
| 01:24.30 | dongs | you still havent mentioned any of htem |
| 01:24.40 | generalhan | Oshuma: honestly .. ANYTHING that you can do with another PBX you can do with asterisk .. and typically more efficiently |
| 01:24.46 | dongs | haha. |
| 01:24.46 | file | what's going on in here? |
| 01:24.56 | generalhan | file: hey buddy ! |
| 01:24.56 | dongs | now thats bullshit. asterisk cant do any of key system features. |
| 01:25.12 | generalhan | file: we have an * hater in here trying to posion everyone |
| 01:25.13 | mog | err dongs |
| 01:25.17 | mog | with a little meetme magic |
| 01:25.18 | drray | :) |
| 01:25.20 | file | Asterisk is not a key system, it's a PBX |
| 01:25.22 | mog | you can impersinate a key system |
| 01:25.36 | Oshuma | generalhan: have you ever worked with TouchStar or Dialogic |
| 01:25.40 | Oshuma | which is what we're currently using |
| 01:25.55 | SplasPood | dongs: ragin against the open sores? |
| 01:26.01 | generalhan | Oshuma: i havent im sorry ... i only know they benefits vs cisco systems (the only others ive used) |
| 01:26.15 | dongs | file, how about telling me where to start if I wanted to setup something where i have asterisk dial a number, dial me, and bridge the two once they're connected? |
| 01:26.31 | Snake-Eyes | What would be some good things to test a voip network (server and client)? e.g. different kinds of NAT, QoS, stress test servers, codecs |
| 01:26.33 | file | call file. |
| 01:26.34 | glm2k | Oshuma: i'd rather make do with version 1.0 of * than Dialogic |
| 01:26.36 | file | or manager |
| 01:26.42 | dongs | file: i figured that much. |
| 01:26.51 | Oshuma | generalhan: well, either way, the only way to find out is to get my hands dirty and install it. heh |
| 01:26.52 | dongs | file: how would it connect the two. |
| 01:26.54 | Oshuma | glm2k: heh |
| 01:26.57 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 01:27.09 | Oshuma | trelane: hah |
| 01:27.10 | file | dongs: it would bridge them together in the Dial application... |
| 01:27.21 | dongs | really? how so |
| 01:27.27 | generalhan | Oshuma: before Asterisk i had never even touched linux ... and just in the past 9 months i have learned soo much just by trial and error |
| 01:27.35 | glm2k | Oshuma: as for features, have you looked at this list? : http://www.asterisk.org/features |
| 01:27.36 | SplasPood | dongs: in either the call file or via Originate in the manager API you'd specify the Channel as one number, and then use either Application or Exten/Priority/Context to Dial() the other number |
| 01:27.56 | generalhan | maybe i dont hang out in here as much as i should ... i have NEVER heard anyone talk to file like this. lol |
| 01:27.57 | dongs | ok now we're getting somewehre |
| 01:27.58 | Qwell | such a crappy troll... |
| 01:28.00 | file | it depends on a few factors exactly "how" it does it... |
| 01:28.15 | dongs | Qwell: if you can do better, I'm all ears. |
| 01:28.15 | glm2k | Qwell: lol |
| 01:28.24 | Qwell | dongs: oh, I'm trolling elsewhere |
| 01:28.28 | file | if it's doing a native bridge a native bridge function which is distinct to the channel driver is called with both channel pointers |
| 01:28.32 | file | so that it can be done more efficiently |
| 01:28.33 | dongs | ok, great. back to the task at hand |
| 01:28.37 | file | ie: exchanging RTP frames |
| 01:28.45 | dongs | file: both cahnnels would be sip |
| 01:28.52 | file | if it's doing a generic bridge, then it waits for asterisk specific frames on each channel and exchanges them |
| 01:29.58 | file | native bridges can also be overriden if you're using features, and the core has to listen for DTMF or stuff... or needs to inject audio (ie you're using the # transfer stuff) |
| 01:30.20 | dongs | all i want to do is specify 2 numbers, it calls both, and connects them |
| 01:30.21 | dongs | thats all |
| 01:30.29 | file | two people have already told you |
| 01:30.51 | SplasPood | yea |
| 01:30.52 | SplasPood | ok |
| 01:30.57 | SplasPood | now he's a verified troll |
| 01:31.01 | dongs | so i'm looking at http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 01:31.21 | dongs | and not getting it |
| 01:31.44 | Qwell | dongs: what's not to get? |
| 01:31.46 | dongs | file: "call files" is not exactly helpful if one doesnt understand the structure of them or how they work. |
| 01:31.58 | file | then that's why you go and look and learn |
| 01:32.13 | dongs | Qwell: i dont understand what i need to do to create tehse two calls. |
| 01:32.13 | denon | dongs: google will tell you more than we ever could in the allotted time |
| 01:32.13 | dongs | file: thats why I am here ASKING THE QUESTIONS so i can learn |
| 01:32.14 | glm2k | dongs: they're just files. if you just tried them it would take you less time to figure them out than to ask here |
| 01:32.16 | SplasPood | dongs: in either the call file or via Originate in the manager API you'd specify the Channel as one number, and then use either Application or Exten/Priority/Context to Dial() the other number |
| 01:32.20 | SplasPood | and one more time |
| 01:32.22 | SplasPood | for good measure |
| 01:32.24 | SplasPood | dongs: in either the call file or via Originate in the manager API you'd specify the Channel as one number, and then use either Application or Exten/Priority/Context to Dial() the other number |
| 01:32.30 | denon | dongs: http://www.google.com/search?sourceid=navclient&ie=UTF-8&rls=GGLD,GGLD:2005-15,GGLD:en&q=asterisk+call+files |
| 01:32.32 | file | you're asking the same question, how do you do it - and we have told you |
| 01:32.33 | denon | the very first result |
| 01:32.34 | dongs | SplasPood: "channel" being what? |
| 01:32.42 | file | you have not asked specifics in relation to call files |
| 01:32.46 | SplasPood | dongs: if you look at the url you referenced, it defines Channel: |
| 01:32.56 | dongs | yes, I can see that. |
| 01:32.59 | dongs | but I dont know what it is. |
| 01:32.59 | SplasPood | dongs: for example channel could be... SIP/1001 |
| 01:33.01 | SplasPood | if thats you |
| 01:33.07 | dongs | so that would be the first number to dial? |
| 01:33.08 | SplasPood | or SIP/12125551212@sipprovider |
| 01:33.11 | SplasPood | if thats you.. |
| 01:33.33 | generalhan | Im having an issue with my new fax lines on my TDM, rather its prolly a problem in my dialplan, when someone calls in lets say on Zap/13 and then i try and call Zap/50 (my fax line) i get this native bridge display in the CLI then the call just hangs up. http://generalhan.pastebin.ca/65334 |
| 01:33.35 | SplasPood | dongs: But I think you need to examine some more basics before you get into what you're doing.. |
| 01:33.45 | generalhan | can some one toss me a suggestion or two to look into to solve this ? |
| 01:33.45 | dongs | OK, so channel would be SIP/mynumber@provider. great. how do I specify the second number (the one I'm calling to)? |
| 01:34.02 | SplasPood | dongs: see the 2nd part of my original reply? |
| 01:34.05 | dongs | Context: + Extension: ? |
| 01:34.16 | SplasPood | dongs: you know what a dialplan is? |
| 01:34.21 | dongs | sure. |
| 01:34.27 | dongs | which call would be made first? the Channel: one in callfile? |
| 01:34.28 | SplasPood | dongs: ok.. you know what a context is? |
| 01:34.32 | dongs | yeah. |
| 01:34.41 | SplasPood | dongs: actually, I don't remember offhand.. I think the Channel one |
| 01:35.04 | file | Channel first, then either the application or extension |
| 01:35.11 | dongs | ok |
| 01:35.40 | dongs | so basically doing something like Channel: SIP/123456@provider would be equivalent to Dial(SIP/123456@provider), and it takes the audio from taht and will bridge it to the context/extension specified later |
| 01:35.45 | dongs | correct? |
| 01:35.52 | generalhan | dongs: DAMN and you were complaining about "no support" !! you have 5 people here working on JUST your problem ... i would call that better support than cisco would give you, and you have to pay A LOT for theirs |
| 01:35.58 | SplasPood | dongs: yep.. |
| 01:36.08 | dongs | ok |
| 01:36.09 | dongs | lets see |
| 01:36.14 | SplasPood | generalhan: ever think that maybe he knows what works? :P |
| 01:36.28 | dongs | :D |
| 01:36.37 | generalhan | lol trelane and i had been talking about that a long time ago ! |
| 01:36.44 | file | do it again and I won't be happy :) |
| 01:37.20 | Sponge_bob | how do i check how much CPU asterisk is using? |
| 01:37.35 | glm2k | Sponge_bob: top? |
| 01:37.37 | glm2k | :) |
| 01:37.38 | SplasPood | dongs: what's your application? trying to out click2dial google? :P |
| 01:37.45 | generalhan | bah... i learn more things that i end up implimenting from asking a question then sitting in here and waiting my turn ... i get to listen to soo many suggestions on how to fix other peopls problems that when i get to them i already know what to do |
| 01:38.02 | *** join/#asterisk juice (n=juice@doc-72-47-32-128.maryville.mo.cebridge.net) |
| 01:38.16 | dongs | SplasPood: pretty much |
| 01:38.26 | trelane | chattr +iu file |
| 01:38.27 | glm2k | generalhan: that's the lurker's fast tract way of learning about any open source project :) |
| 01:38.38 | glm2k | er, track |
| 01:38.41 | generalhan | damn straight ! |
| 01:39.31 | generalhan | Sponge_bob: i like to use 'top' and have a few people call in and out on serveral lines to test the cpu draw from * |
| 01:40.27 | generalhan | glm2k: sorry man i didnt mean to step on your toes ... i didnt see your response before i typed mine ! |
| 01:40.49 | glm2k | lol. no worries. fire away. |
| 01:41.28 | generalhan | ok let me see how far i can get with this before i have to leave ... |
| 01:41.52 | generalhan | http://generalhan.pastebin.ca/65334 can some one take a look at the native bridge line and then the hangup ... i need to know what i can do to make sure the call gets bridged properly |
| 01:42.07 | generalhan | this will be for our main business fax line and i need to make sure that we have 0 downtime |
| 01:43.12 | generalhan | I took the "tT" out after i pasted that ... all my Dial() cmds have that in it and i put it in there by accident. |
| 01:43.37 | timscott | So you're saying, you're calling one extension to another, and then when they connect, they both release? |
| 01:43.41 | dongs | nice |
| 01:43.43 | dongs | it worked |
| 01:43.45 | dongs | thanks dudes. |
| 01:43.51 | generalhan | timscott: is that to me ? |
| 01:43.56 | timscott | Yeah man. |
| 01:43.57 | timscott | That's to you |
| 01:44.08 | *** join/#asterisk Eric-xx (i=ericx@cm83.epsilon192.maxonline.com.sg) |
| 01:44.16 | timscott | Is that what is happening? |
| 01:44.31 | generalhan | timscott: no this is a fax coming in to my zap lines on channel 13 and trying to bridge to Zap/50 (which is my fax machine hooked to my TDM |
| 01:44.53 | timscott | ah. |
| 01:45.07 | timscott | could I perchance see the code you use to do that? |
| 01:45.12 | generalhan | sure |
| 01:45.18 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 01:45.25 | timscott | thanks mate |
| 01:45.55 | generalhan | timscott: its just a "exten => xxx,1,answer; exten => xxx,2,Dial(Zap/50,20)" |
| 01:46.03 | generalhan | and thats it |
| 01:46.13 | dongs | i dont think that would work |
| 01:46.25 | dongs | answer() wont return until the other end hangs up |
| 01:46.34 | dongs | though i'm probably talking out of my ass. |
| 01:46.40 | generalhan | well the retarded part about all of this is that it HAS worked and every 1 out of 20 tried it DOES work |
| 01:46.57 | generalhan | but the other 19 i get the same problem |
| 01:47.00 | dongs | define "not work"? does the fax fail? |
| 01:47.03 | pjchilds | generalhan, its probably a timeing issue with the fax to fax communciations... |
| 01:47.14 | generalhan | fax fails yes. |
| 01:47.28 | pjchilds | generalhan, just dial() and don't answer.. that what the originating fax machine wont start negotiating until its talking to the destination... |
| 01:47.34 | timscott | That should work... |
| 01:47.41 | dongs | yeah, that might be better |
| 01:47.41 | Sponge_bob | generalhan: i used top. it looks like an excelent tool. i noticed my total memeroy was 515000k total and 505000k used and 9000k free. is that normal? |
| 01:47.59 | generalhan | but everything looks like its gonna work right up until the hangup. the fax rings and answers says "connecting" for about 3 seconds then everything hangs up |
| 01:48.44 | dongs | WaitTime: <number> Seconds to wait for an answer < is this the time to wait while establishing Channel: // in a call file? |
| 01:48.53 | dongs | or is that time to wait for Context/Extension? |
| 01:48.54 | pjchilds | generalhan, but your originating fax machine has been trying to handshake with the asterisk box when it answers, and when its dialing... and then it connects... half way through the handshaking... |
| 01:49.20 | generalhan | ok gimme just one sec ... i took the dial out im gonna use our other fax machine and test it out a couple times to check |
| 01:49.21 | generalhan | brb |
| 01:49.25 | generalhan | thanks guys |
| 01:50.02 | Sponge_bob | what tool does everyone use to stress test asterisk? |
| 01:51.00 | pjchilds | i have used another asterisk machine with a shell script and lots of call files.. but it really only does load generation rather than feature coverage testing etc... |
| 01:51.06 | dongs | what exactly are you looking to stress test |
| 01:51.48 | Sponge_bob | dongs: the cpu, the quality, the load.... |
| 01:52.21 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-60-250.cybersurf.com) |
| 01:52.42 | Sponge_bob | dongs: well? |
| 01:53.07 | pjchilds | quality is pretty interesting thing to try and objectively test... |
| 01:53.16 | dongs | Sponge_bob: not sure :) |
| 01:53.39 | Sponge_bob | pjchilds: ok skip the quality test. clarity |
| 01:53.59 | Sponge_bob | pjchilds: any suggestions? |
| 01:54.09 | pjchilds | google 'hammer load tester' |
| 01:54.29 | pjchilds | its pretty expensive stuff, and i've only ever seen the hammer analyser, but its pretty sweet... |
| 01:54.41 | dongs | well |
| 01:54.41 | pjchilds | ethereal also can do analysis on RTP streams... |
| 01:54.46 | dongs | tehre is nothign to test as far as "clarity" |
| 01:54.52 | dongs | all codecs have known limitations. |
| 01:54.53 | generalhan | Sponge_bob: i cant remember any off hand, but i asked around until i found a couple of companies that have endless loops on their autoattendants and i called using half my lines and let them run for 24 hours and kept logs of the throuput on my server. ofcourse i had free minutes for 48 hours to test with so if you dont that my not be the best for you |
| 01:54.58 | dongs | its not going to get magically better or worse |
| 01:55.19 | pjchilds | in newer asterisk releases RTCP could be used to gather stats on RTP traffic in calls... |
| 01:55.44 | Sponge_bob | pjchilds: how do i use RTCP? |
| 01:56.42 | pjchilds | Sponge_bob, no idea.. i think its only in 'trunk' ... |
| 01:57.10 | Sponge_bob | hum... |
| 01:57.26 | generalhan | damn... took the Answer() out and test 4 faxes and none went through they all get to the "Attempting native bridge of ..." and then hangup |
| 01:57.38 | pjchilds | we would sometimes stuff in say 100 or 200 calls ... and then for the 101th call I would make the call and 'gauge' the call quality (whilst looking at CPU usage etc...) |
| 01:58.12 | Sponge_bob | i have a cisco 2821 hooked to a switch and from there my asterisk is connected also. I have a t1 pri connected to the cisco. I'm wondering what tests i can perform... |
| 01:58.59 | Sponge_bob | pjchilds: how can i generate 100-200 calls? |
| 01:59.00 | dongs | Sponge_bob: join #teenchat with a female nick and post a number there. then send all callers to a complicated menu system. that'll test all your lines quick |
| 01:59.13 | Sponge_bob | haha |
| 02:00.32 | pjchilds | Sponge_bob, well if you ask dongs he can explain asterisk call files ;) |
| 02:00.39 | generalhan | hahahaha |
| 02:01.01 | generalhan | ok all i cant sit here anymore ... 11 hours at work is just too much for me ... i gotta go home |
| 02:01.12 | generalhan | ill deal with this bridge issue tomorrow |
| 02:01.22 | generalhan | you all have good mash-pitting |
| 02:01.30 | Sponge_bob | generalhan: see you |
| 02:01.40 | pjchilds | our script is like ... http://pastebin.ca/65343 |
| 02:01.48 | *** join/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn) |
| 02:02.49 | pjchilds | and the context for testing looks like http://pastebin.ca/65344 |
| 02:03.26 | pjchilds | although we sometimes do other things... like play messages... wait for hold etc... (that one we were using to load test an IVR so we had to randomise menu selections...) |
| 02:04.23 | pjchilds | the 'bash' shell script can be 'saved' as a file and executed... of course the channels to be used, the number of calls, the delays etc would need to be set to whatever you were trying to achieve.... |
| 02:04.58 | Sponge_bob | pjchilds: ok, let me try it |
| 02:07.17 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
| 02:07.29 | Sponge_bob | pjchilds: what's the mk-call? |
| 02:08.14 | pjchilds | its commented out ... its a 'left over' from when we bashed the script together... |
| 02:09.24 | Sponge_bob | for the line that starts with channel: is that were the calls are originating? |
| 02:10.07 | *** join/#asterisk Lino` (n=Lino@i577BDE18.versanet.de) |
| 02:10.15 | pjchilds | thats the destination... so in this instance its a Zap channel in group 1 with an outbound number of 385144701.... |
| 02:10.48 | pjchilds | the call is then 'passed' to the [Context: outboundmsg] for futher processing ... |
| 02:11.03 | Sponge_bob | gottcha. i kind of understand it now |
| 02:11.51 | pjchilds | i kind of understand it, but only enought that it did what we wanted (ie called stuff lots of times, and did things -- made tones, or played wave files, or just put calls in waiting mode etc...) |
| 02:12.35 | pjchilds | we used it to test a IP IVR... so we knew after about 100 or so calls the IP IVR would CPU max out and the call quality sounded like shit... |
| 02:13.21 | pjchilds | also we could test what happened if we slammed 50 calls in one second (it didn't like that much...) |
| 02:13.55 | Sponge_bob | haha |
| 02:14.36 | Sponge_bob | how do you test the 50 calls/second? do i change the delay to 0? |
| 02:15.09 | pjchilds | or 0.1s or something like that... |
| 02:15.28 | pjchilds | it actually does one more call than the NUMBER_CALLS since I can't write a loop correctly :) |
| 02:15.55 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 02:16.02 | Sponge_bob | pjchilds: if i set the channel to sip/100 does that channel have to answer? |
| 02:16.21 | dongs | yes |
| 02:16.41 | nortex | Can sombody help me with faxing to a Sangoma A104d card with a rhino channel bank? |
| 02:16.46 | Sponge_bob | what i'm getting to is...in order to test it successfull how does the destination channel have to respond? |
| 02:16.51 | pjchilds | I guess thats what WaitTime, RetryTime and MaxRetries are about... |
| 02:17.01 | Sponge_bob | pjchilds: ok |
| 02:21.49 | nortex | Can sombody help me with faxing to a Sangoma A104d card with a rhino channel bank? |
| 02:29.00 | Sponge_bob | pjchilds: when i try to run the script it says : bad interpreter: no such file or directory |
| 02:30.57 | *** join/#asterisk Faithful (n=Faithful@202.6.145.116) |
| 02:31.28 | dongs | Sponge_bob: orly? what script |
| 02:31.48 | Sponge_bob | http://pastebin.ca/65343 |
| 02:32.22 | dongs | lol google ads. |
| 02:32.37 | dongs | anyway. make sure the file is in lunix line endings |
| 02:32.45 | dongs | did you edit it on windows? |
| 02:32.56 | Sponge_bob | opps |
| 02:33.01 | Sponge_bob | let me re-do it |
| 02:34.46 | Sponge_bob | dongs: its giving me the same thing. let me try something else |
| 02:36.34 | *** join/#asterisk NDT (n=noone@cpe-72-228-10-145.nycap.res.rr.com) |
| 02:36.36 | dongs | well |
| 02:36.38 | dongs | do you have /bin/bash? |
| 02:37.23 | Sponge_bob | its working |
| 02:37.33 | dongs | lunix line endings problem, right? |
| 02:38.35 | Sponge_bob | yup |
| 02:39.09 | dec | it's linux dude, not lunix |
| 02:40.32 | Sponge_bob | now i'm getting call failed to go through, reason 3 |
| 02:41.07 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 02:42.01 | dongs | thats better than before. |
| 02:42.26 | Sponge_bob | dongs: yup. :-) how should my destination answer the call? |
| 02:42.41 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 02:43.46 | variable_office | i am trying to compile zaptel, but i keep getting errors like this: http://pastebin.com/707825 |
| 02:43.47 | dongs | nice, got this shit working with a simple webpage. now i can take over google |
| 02:45.18 | variable_office | has anyone seen that before? because the little google turns up is useless |
| 02:46.18 | Sponge_bob | dongs: my destination is SIP/100 which gets answered by my softphone. it looks like it can only answer 3 calls at a time. how should i setup a dummy dialplan for the destination channel? |
| 02:48.50 | NDT | variable_office: need to get the kernel source for that the kernel you are using |
| 02:49.08 | variable_office | NDT i did get that |
| 02:50.19 | dongs | hm |
| 02:50.29 | Oshuma | is there a way to specify a PREFIX when installing? |
| 02:50.31 | dongs | Sponge_bob: make a dummy iVR |
| 02:50.33 | dongs | and send shit there |
| 02:50.38 | Oshuma | not sure how to do it without a ./configure |
| 02:50.38 | dongs | like a endless loop of "LOL YOU ARE ON HOLD" |
| 02:50.39 | dongs | or something. |
| 02:50.48 | Oshuma | editing the Makefile would work, i suppose |
| 02:50.53 | Sponge_bob | ok, let me try |
| 02:51.02 | dongs | exten=>101,1,Play(lol) exten=>101,2,Goto(1) |
| 02:51.03 | dongs | or somethign |
| 02:51.24 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 02:51.49 | variable_office | NDT i think i did at least, i got the kernel-devel |
| 02:52.18 | NDT | kernel-devel isn't the src |
| 02:53.20 | variable_office | humm, what would the package name be for the source because kernel-source doesnt exist |
| 02:53.46 | NDT | what distro is it? |
| 02:53.58 | variable_office | centos, a community rhel |
| 02:57.09 | NDT | http://altruistic.lbl.gov/mirrors/centos/4.3/updates/SRPMS/kernel-2.6.9-34.0.1.EL.src.rpm |
| 02:57.40 | variable_office | would i just wget that and then "rpm xxx" |
| 02:57.56 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:58.14 | NDT | yeah |
| 02:59.01 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 02:59.10 | Nugget | centos is redhat enterprise linux without the awful eula. |
| 02:59.54 | variable_office | Nugget do you know a good way to get the kernel source with just a package or something? |
| 02:59.54 | Nugget | it's functionally equivalent, truly free, but won't make dell or oracle happy if you have a support agreement |
| 03:00.11 | Nugget | dunno, I hate linux. |
| 03:00.19 | NDT | variable_office: That was the src rpm |
| 03:01.00 | variable_office | Nugget what do you use? |
| 03:01.01 | NDT | variable_office: wget http://altruistic.lbl.gov/mirrors/centos/4.3/updates/SRPMS/kernel-2.6.9-34.0.1.EL.src.rpm Then rpm -Uvh kernel-2.6.9-34.0.1.EL.src.rpm |
| 03:01.07 | Nugget | anything else. :) |
| 03:01.18 | variable_office | NDT ya, just wanted to see if there was a way to do it with packages |
| 03:01.27 | variable_office | NDT i am wget ing it now |
| 03:01.46 | Nugget | in all seriousness, though, my linux boxes are all redhat or slackware. on redhat I upgrade the kernel whenever redhat tells me to, and on slackware I do it with the source. |
| 03:01.55 | Nugget | the notion of a kernel package is kinda creepy to me. |
| 03:02.04 | Nugget | and the boxes I actually care about are all freebsd or os x. |
| 03:02.16 | *** join/#asterisk Lino` (n=Lino@i577BD5F8.versanet.de) |
| 03:03.36 | variable_office | freebsd seems like it would be cool, but i dont geel like trying to figure it out when i am still trying to grasp linux |
| 03:03.53 | Nugget | freebsd requires a lot less "figuring out" than linux, in my experience. |
| 03:04.12 | Nugget | it's a lot more consistent and unified, which deftly avoids exactly the sort of confusion you're facing right now. |
| 03:04.18 | variable_office | really? seems like it requires more, but thats probably because i dont find as many freebsd howtos |
| 03:04.24 | Nugget | it sucks for running asterisk, though. |
| 03:04.32 | variable_office | ah, that sucks |
| 03:04.51 | Nugget | there aren't many freebsd howtos because freebsd is well-documented and therefore doesn't generate the corpulent mess of user-created documentation and hints like linux does. |
| 03:05.20 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 03:05.20 | drray | and freebsd is not newbie friendly |
| 03:05.33 | variable_office | drray ooh thats bad for me :) |
| 03:05.34 | Nugget | A lot of people seem to have that opinion. |
| 03:05.47 | Nugget | I'm not sure where it comes from, other than from the fact that a lot of people have that opinion. |
| 03:06.05 | drray | :) |
| 03:07.12 | NDT | I am not a huge fan of freebsd because I am just used to linux...but I will tell you one thing...there is a freebsd box of ours that has been rebooted 4 times in over 2 years and thats just cause I wanted to reboot it heh |
| 03:07.37 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 03:08.04 | Qwell | NDT: meh, my linux box had it's uptime wrap around last week |
| 03:08.21 | Nugget | that's pretty lame. when does it wrap? |
| 03:08.30 | Qwell | it seemed to at 475 days |
| 03:08.31 | drray | My linux asterisk box has been up for 400 dyas |
| 03:08.54 | Nugget | uptime isn't important anyway, it's downtime that matters. :) |
| 03:09.04 | drray | I'm sorry 418 days |
| 03:09.38 | Nugget | a more interesting statistic is my freebsd box that I've source-level upgraded from version 2.2.5 (1997) to 6.1 (today). no re-installs, no wipes, no disruption. |
| 03:09.49 | Nugget | just steady "cvs up && make install" across all those versions |
| 03:10.09 | drray | is gentoo a bsd? or a linux? |
| 03:10.12 | anonymouz666 | I had a FreeBSD 4.6-Release load average around 9.0 / 10.0 running non-stop for 1003 days. |
| 03:10.21 | Sedorox | gentoo is linux |
| 03:10.25 | Nugget | I've never had a linux box go more than a few years before it was such a sloppy mess that a reinstall was the sanest way to upgrade. |
| 03:10.30 | Sedorox | but its package managemenbt is based off of bsd's ports |
| 03:10.34 | Sedorox | style anyway |
| 03:10.42 | *** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au) |
| 03:10.47 | drray | to be fair, I've not updated my linux box in a year |
| 03:10.52 | drray | or my asterisk |
| 03:10.55 | drray | but ssssshhh... |
| 03:10.56 | drray | :) |
| 03:10.57 | Sedorox | wow |
| 03:10.57 | P-NuT | Big up y'all. .... (and so forth..) |
| 03:11.03 | Sedorox | so whats your IP? :p |
| 03:11.06 | Nugget | heh |
| 03:11.10 | drray | it does not have an ip |
| 03:11.15 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:11.25 | drray | it's just for zap |
| 03:11.27 | Sedorox | if it has asterisk.. unless your doing all tdm stuff... |
| 03:11.31 | Sedorox | :p |
| 03:11.32 | Nugget | it has at least one. :) |
| 03:11.38 | drray | 127.0.0.1 |
| 03:11.40 | Sedorox | 127.0.0.1!! |
| 03:11.47 | Sedorox | oh noes! I know urs ip!! |
| 03:11.49 | Sedorox | elevenone |
| 03:11.54 | Nugget | I'll pingflood you! |
| 03:11.58 | [hC] | any idea why this context might act stupid? it answers the line, and wont play ringing to the calling party while the ext rings. If i do a playback() the calling party can hear it, if i put 'r' in the dial options, still no help: http://pastebin.ca/65352 |
| 03:12.02 | Sedorox | :p |
| 03:12.06 | [hC] | I have another context that does almost the same thing and it works fine. |
| 03:12.13 | [hC] | I dont get what the issue is here. |
| 03:13.29 | P-NuT | Does anybody have any idea how to configure an SPA3000 for asterisk? I just want to be able to accept and dial out from the PSTN. |
| 03:17.16 | P-NuT | no? |
| 03:17.18 | P-NuT | ok.... |
| 03:17.23 | [TK]D-Fender | [hC] : pastebin it in exectution.... |
| 03:17.54 | [TK]D-Fender | P-NuT : www.voxilla.com They have tons of guides on how to set it up for * in their forums, go check them out. |
| 03:18.12 | [hC] | http://pastebin.ca/65354 |
| 03:18.26 | P-NuT | awesome. thanks fender... |
| 03:18.30 | P-NuT | Again.... |
| 03:18.35 | [hC] | all very normal. everything looks okay, maybe im going crazy.. :) |
| 03:18.40 | P-NuT | Damn, how many times have you saved my ass now? |
| 03:19.19 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 03:19.41 | [TK]D-Fender | [hC] : Analog Zap originating the channel? Pastebin the dialplan of one that works, and one that fails, as well as the tech config for them. |
| 03:19.55 | [hC] | yeah analog zap originating. |
| 03:19.55 | [TK]D-Fender | P-NuT : Dunno... I'm out of fingers & toes ;) |
| 03:20.03 | [hC] | i'll do that when i get home, just gonna walk there now from work |
| 03:20.05 | [hC] | back in 10 |
| 03:20.05 | [hC] | :) |
| 03:20.14 | [TK]D-Fender | k |
| 03:20.23 | Sponge_bob | dongs: you still there? can i PM you? |
| 03:20.28 | dongs | im here :( |
| 03:21.48 | Sponge_bob | i'm still having trouble getting asterisk to answer more than a couple calls |
| 03:21.53 | dongs | hm |
| 03:21.58 | dongs | did you make that endless message playing loop? |
| 03:22.02 | Sponge_bob | i created a loop already but.. |
| 03:22.14 | dongs | does it work when yo ucall it? |
| 03:22.16 | Sponge_bob | what do i put in the channels for the script? |
| 03:22.46 | dongs | hm |
| 03:22.50 | dongs | good question |
| 03:22.59 | Sponge_bob | :-) |
| 03:23.11 | dongs | you put the loop extension in Extension: in clalfile |
| 03:23.43 | Sponge_bob | so do i leave the channel blank? |
| 03:23.48 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:24.32 | *** join/#asterisk bkw__ (n=brian@209.136.55.74) |
| 03:24.44 | dongs | no |
| 03:24.49 | dongs | the channel is |
| 03:24.49 | dongs | hm |
| 03:28.55 | dongs | duno :) |
| 03:29.09 | Sponge_bob | :-) |
| 03:29.19 | Sponge_bob | pjchilds: are you still there? |
| 03:29.48 | Sponge_bob | i wonder if i can make a dummy channel |
| 03:30.01 | dongs | how about |
| 03:30.02 | dongs | sending them |
| 03:30.04 | dongs | to MeetMe |
| 03:31.21 | Sponge_bob | let me try |
| 03:32.08 | *** join/#asterisk alephcom (n=Weibe@host75.net14.mcsnet.ca) |
| 03:32.16 | Sponge_bob | well, we still run into what to put in channel |
| 03:33.01 | *** join/#asterisk alephcom (n=Weibe@host75.net14.mcsnet.ca) |
| 03:33.04 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 03:33.28 | Sponge_bob | dongs: right? |
| 03:33.32 | *** join/#asterisk bkw__ (n=brian@209.136.55.74) |
| 03:33.34 | dongs | not really |
| 03:33.36 | dongs | make a meetme |
| 03:33.39 | dongs | and it creates a zap channel |
| 03:33.47 | dongs | i *think* you can use that as the channel name. |
| 03:34.15 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 03:34.28 | dongs | yea |
| 03:34.30 | dongs | Zap/pseudo-587132936 |
| 03:34.38 | dongs | once you get one user in a meetme it makes that channel |
| 03:34.43 | dongs | see if you can put that into callsource |
| 03:34.46 | dongs | er |
| 03:34.49 | dongs | Channel |
| 03:34.59 | Sponge_bob | hum... |
| 03:35.45 | dongs | just tried that. |
| 03:35.48 | dongs | that didnt work. |
| 03:36.18 | dongs | well, "that channel" the numbers being random. but anyway, it didnt work anyways. |
| 03:36.21 | Sponge_bob | your faster than me :-) |
| 03:40.13 | shmaltz | what processor type should I chose in menuconfig for a VIA Eden C7 CPU? |
| 03:40.41 | *** join/#asterisk Kerry_G (n=Kerry_G@216.70.131.136) |
| 03:40.46 | Kerry_G | ~ centosbug |
| 03:40.49 | jbot | centosbug is, like, a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 03:41.12 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:41.43 | dongs | you ever wonder why 'centos' is same length as 'redhat'? |
| 03:42.06 | Sponge_bob | i never knew that |
| 03:42.20 | dongs | here's a guide how to make centos iso out of redhat iso: cat redhat.iso | sed -i s/redhat/centos/ > centos.iso |
| 03:42.42 | Sponge_bob | haha |
| 03:43.16 | dongs | thats literally all they do, i mean not exactly this but they rip rhel and replace redhat -> centos and provide their own "free" updates or whatever. |
| 03:43.34 | dec | and package yum into it |
| 03:43.37 | Sponge_bob | true |
| 03:45.38 | *** join/#asterisk spine55 (n=erniee@c-69-180-53-201.hsd1.ga.comcast.net) |
| 03:50.09 | *** join/#asterisk bmg505 (n=leon@196.209.39.102) |
| 03:52.46 | variable_office | is asterisk real time something that needs to be done while compiling asterisk? something special i mean because cd /usr/src/asterisk make clean make install ? |
| 03:54.08 | *** join/#asterisk AeroIllini (n=aeroilli@c-71-197-210-101.hsd1.or.comcast.net) |
| 03:54.29 | AeroIllini | my music on hold is very choppy, but other audio is fine |
| 03:54.53 | AeroIllini | what could be causing this? |
| 03:55.07 | Sponge_bob | AeroIllini: i don't know but sometimes mine does that too |
| 03:55.24 | AeroIllini | mine's not sometimes, sponge_bob, it's all the time |
| 03:55.33 | Sponge_bob | what is the latency between the device and asterisk |
| 03:55.34 | Sponge_bob | ? |
| 03:55.34 | AeroIllini | every time |
| 03:56.01 | AeroIllini | none .... it's on a GigE internal network (for testing) |
| 03:56.18 | Sponge_bob | you got me there...i'm not sure |
| 03:56.36 | Sponge_bob | what's your call floow look like? |
| 03:56.37 | AeroIllini | what's so confusing is that all my other audio, calls, MP3Player(), Background(), etc, is fine |
| 03:57.09 | AeroIllini | Ekiga is registered as my softphone, and I dial extension 1000, which answers and plays MusicOnHold |
| 03:57.10 | Sponge_bob | i heard asterisk is picky on the files it plays for MOH. i could be wrong |
| 03:57.23 | AeroIllini | but it does this with the standard included ones, too |
| 03:57.31 | AeroIllini | which I hope are in a proper format :-) |
| 03:57.59 | Sponge_bob | try a different codec? |
| 03:58.04 | AeroIllini | did |
| 03:58.21 | AeroIllini | tried gsm, ulaw, native, and mp3 |
| 03:58.59 | Sponge_bob | how about a normal conversation? |
| 03:59.30 | AeroIllini | those are fine |
| 03:59.34 | nortex | Can sombody help me with faxing to a Sangoma A104d card with a rhino channel bank? |
| 03:59.44 | AeroIllini | I suspect it's a zaptel problem |
| 03:59.45 | *** join/#asterisk nobell (n=nobell@degraffenried.dsl.xmission.com) |
| 03:59.48 | *** join/#asterisk endrin (i=daed@CPE-70-92-75-238.new.res.rr.com) |
| 04:00.04 | Sponge_bob | yeah it could be a timing issue |
| 04:00.10 | Sponge_bob | how does meetme sound? |
| 04:00.25 | nobell | I have a question about an iax trunk I just set up. |
| 04:00.53 | AeroIllini | I just tried MeetMe, and it's choppy |
| 04:01.13 | AeroIllini | like it's playing a chunk, then waiting the length of that chunk, then playing another chunk |
| 04:01.15 | Sponge_bob | i think it could be a timing issue, which i am not familiar with |
| 04:01.20 | variable_office | wow this realtime stuff with pgsql/odbc seems awesome |
| 04:01.20 | AeroIllini | so the sound is twice as slow |
| 04:01.35 | variable_office | now i just need to figure out how to configure cards and stuff |
| 04:02.28 | Sponge_bob | recompile zap drivers? |
| 04:02.38 | Sponge_bob | get the latest and recompile |
| 04:07.22 | *** part/#asterisk alephcom (n=Weibe@host75.net14.mcsnet.ca) |
| 04:08.32 | variable_office | wheres a good place to start with configuration of asterisk? |
| 04:08.52 | tlowe_ | /etc/asterisk/ |
| 04:09.43 | variable_office | i gathered that much, i mean, i have it setup to have the confs in pgsql, what is the first thing i have to configure (i am using one of those cheapo $20 cards from ebay for pots connectivity) |
| 04:15.57 | variable_office | how can you tell if asterisk sees my card? |
| 04:18.17 | variable_office | its an x100p |
| 04:20.32 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 04:22.38 | *** join/#asterisk gcarrillog (n=gcarrill@dsl-201-133-121-242.prod-infinitum.com.mx) |
| 04:23.48 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 04:24.41 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 04:25.28 | *** join/#asterisk themikester60 (n=anthony@66-100-35-23-static.dsl.oplink.net) |
| 04:26.06 | themikester60 | Has anyone in here ever used call queues? I've setup a queue with music on hold, but for some reason all users in the queue have music on hold aside from the person who is next in line to be answered.. does anyone know what might be causing this? |
| 04:29.33 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:30.18 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:30.49 | harryvv | I killed the key in the Astdb regaring the call forward option on this ip500 that was enabled. Still shows as enabled on phone. Rebooted asterisk and ip500 still shows as call forward on the phone. Is there a known bug issue with asterisk as to why it refuses to release call forward on a phone? |
| 04:31.14 | file | harryvv: not up to Asterisk to do call forwarding... it's the phone |
| 04:31.23 | file | or you write your own dialplan logic if you really really want |
| 04:31.28 | variable_office | humm.. even though i compiled the zaptel, i have to zap command in the asterisk cli |
| 04:31.51 | harryvv | mmmm |
| 04:32.14 | harryvv | well it did show the CF key in Astdb and i killed it |
| 04:32.30 | file | AMP? |
| 04:32.34 | harryvv | nope |
| 04:32.35 | harryvv | cli |
| 04:33.08 | file | well, it's not part of the standard Asterisk to have logic to do CF in the dialplan... so it came from somewhere else... and if the forwarding is set on the phone, Asterisk can't override it... |
| 04:33.17 | file | phone just says "go here" when you call it |
| 04:33.59 | harryvv | well i tryed to kill it on the phone by typing in *73. I inititially set it on the phone and heard the asterisk voice enable it. |
| 04:34.09 | harryvv | Then used it again to kill it. It did not release. |
| 04:34.39 | file | that's some dialplan logic from somewhere else :) |
| 04:34.45 | harryvv | I believe it does show in the cli asterisk is diverting the call. |
| 04:34.49 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:35.07 | file | pastebin the CLI |
| 04:35.12 | harryvv | k |
| 04:35.26 | harryvv | when doing the *73 or calling that extention? |
| 04:35.35 | file | both |
| 04:35.37 | harryvv | k |
| 04:40.41 | *** join/#asterisk nick125 (n=nick@atarack/staff/nick125) |
| 04:43.56 | harryvv | file http://pastebin.ca/65371 |
| 04:44.28 | file | that's AMP, and the phone isn't forwarded using the AMP way |
| 04:44.32 | file | it's forwarded on the phone |
| 04:45.39 | harryvv | mmm |
| 04:46.51 | [TK]D-Fender | harryvv : Get off your but and hit the "forward" soft-key and disable it for crying out loud! |
| 04:47.05 | harryvv | I never enabled it |
| 04:47.15 | harryvv | I only use the asterisk CF |
| 04:47.39 | [TK]D-Fender | harryvv : Got the bouncing arrow? |
| 04:47.49 | harryvv | yes |
| 04:47.56 | [TK]D-Fender | harryvv : Then you're wrong. |
| 04:48.19 | harryvv | I use AAH *73 for CF only |
| 04:48.27 | [TK]D-Fender | Bouncing arrow = forward on the phone, no if's and's or but's |
| 04:48.36 | file | just turn it off... |
| 04:48.50 | [TK]D-Fender | harryvv : it doesn't matter WHAT you do with *, you've hard forwarded on the PHONE itself. |
| 04:51.07 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 04:53.36 | harryvv | up that was it. its a feature I never used |
| 05:08.24 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 05:11.26 | [TK]D-Fender | harryvv : Then disable it on your phones... I intend to |
| 05:13.33 | [TK]D-Fender | ok, bed time... later all |
| 05:13.33 | *** join/#asterisk alephcom (n=Weibe@host75.net14.mcsnet.ca) |
| 05:20.34 | *** join/#asterisk bmg505 (n=leon@196.209.39.102) [NETSPLIT VICTIM] |
| 05:20.34 | *** join/#asterisk Faithful (n=Faithful@202.6.145.116) [NETSPLIT VICTIM] |
| 05:20.34 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) [NETSPLIT VICTIM] |
| 05:20.34 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
| 05:31.11 | nick125 | is there a way to get ztdummy to work without RTC? in xen, I can't get a RTC device :/ |
| 05:31.13 | *** join/#asterisk carmen (i=ix@c-24-60-193-83.hsd1.ma.comcast.net) |
| 05:33.46 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 05:40.51 | *** part/#asterisk themikester60 (n=anthony@66-100-35-23-static.dsl.oplink.net) |
| 05:43.32 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 05:44.57 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 06:00.38 | *** join/#asterisk Mw3_ (i=mw3@national.t-error.hu) |
| 06:01.34 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 06:03.09 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 06:05.36 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 06:08.41 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:09.02 | *** join/#asterisk lorinc (n=ang@caracas-1511.adsl.interware.hu) |
| 06:18.10 | *** join/#asterisk mmmmmToop (n=mmmmToop@firewall.datapro.co.za) |
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| 06:25.00 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 06:28.56 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:39.22 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 06:57.44 | *** join/#asterisk zepmantra (i=wahhh@203.215.100.96) |
| 07:01.57 | *** join/#asterisk jeffik (n=Jeff@kns223.NetSurf.Net) |
| 07:06.40 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 07:07.48 | *** join/#asterisk NewSole (n=dave@d226-105-226.home.cgocable.net) |
| 07:07.56 | NewSole | anyone alive |
| 07:08.37 | kmilitzer | Yes, I am alive ... |
| 07:08.57 | NewSole | I need to make a test call |
| 07:09.12 | kmilitzer | NewSole: Define test call ... |
| 07:09.41 | NewSole | talk to someone voice... my dime |
| 07:11.17 | NewSole | u up for a call |
| 07:11.43 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 07:11.59 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 07:12.10 | FuriousGeorge | <PROTECTED> |
| 07:12.15 | *** join/#asterisk Sasch (n=Admin@host102-30.pool82107.interbusiness.it) |
| 07:12.20 | FuriousGeorge | noticed the damndest thing today |
| 07:12.21 | Sasch | hi all |
| 07:12.42 | *** join/#asterisk tparcina (n=tparcina@lns02-0080.dsl.iskon.hr) |
| 07:12.52 | Sasch | do you know a modem support list for asterisk ?? |
| 07:12.56 | FuriousGeorge | apparently if i call my pots lines and hang up real quick, asterisk doesnt notice that the call ended and keeps the fxo channel open |
| 07:12.57 | NewSole | whats that |
| 07:13.21 | FuriousGeorge | Sasch: afaik only one modem works with asterisk as an fxo |
| 07:13.28 | drray | and then it times out? |
| 07:13.42 | FuriousGeorge | drray: asking me? i dunno i restarted the server |
| 07:13.52 | NewSole | Can anyone do me a favor... and let me call them... |
| 07:14.18 | FuriousGeorge | although i noticed in the past that if i dont restart the server occasionally, eventually * will see all my lines as in ise |
| 07:14.19 | FuriousGeorge | use |
| 07:14.25 | FuriousGeorge | the analog ones that is |
| 07:14.40 | FuriousGeorge | i assumed it was a bug with zaptel |
| 07:14.46 | FuriousGeorge | it happens every few months |
| 07:14.51 | NewSole | it is Fur |
| 07:15.00 | NewSole | its signaling |
| 07:15.07 | NewSole | and volt |
| 07:15.35 | FuriousGeorge | is it a bug or is the card not detecting the volt drop off? i thought thats what kewl start was for |
| 07:15.49 | NewSole | card |
| 07:16.03 | NewSole | you have to boost the gains |
| 07:16.19 | FuriousGeorge | hmm |
| 07:16.26 | FuriousGeorge | both tx and rx? |
| 07:16.30 | NewSole | yup |
| 07:16.38 | FuriousGeorge | u think that will do it huh |
| 07:16.43 | NewSole | yup... |
| 07:16.48 | NewSole | it did with me |
| 07:16.56 | NewSole | FuriousGeorge... cau u do me a favor |
| 07:17.03 | FuriousGeorge | you wanna call me? |
| 07:17.15 | NewSole | can u let me call u.. I need to test out new soft phone |
| 07:17.37 | FuriousGeorge | not that i mind, but why dont you call some 1800 number like 1800 555 tell |
| 07:18.11 | NewSole | I need a two way convo so I can see debug info... |
| 07:18.30 | NewSole | I need to see packet data comming form both ends |
| 07:18.42 | FuriousGeorge | that number i gave you pormpts you for voice commands. lemme see if i have my cell phone though |
| 07:19.18 | NewSole | it does not have to be pots line |
| 07:19.24 | NewSole | a voip line is fine |
| 07:19.50 | syle | use /msg for private sex chats |
| 07:20.14 | NewSole | u only wish style.... in your dreams |
| 07:20.35 | *** join/#asterisk Sonderblade (n=mah@host-213.131.147.169.addr.tdcsong.se) |
| 07:22.12 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:25.28 | *** join/#asterisk alucard064 (n=alucard0@ABayonne-152-1-47-104.w83-193.abo.wanadoo.fr) |
| 07:25.45 | alucard064 | hello |
| 07:26.18 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 07:26.39 | alucard064 | someone can tell me what the difference between trixbox et asterisk@homev2.8 and the features please |
| 07:26.43 | alucard064 | ? |
| 07:29.22 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 07:29.26 | denon | alucard064: look at the topic |
| 07:29.27 | denon | -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 07:29.30 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:30.40 | alucard064 | ok |
| 07:30.42 | alucard064 | excuse me |
| 07:37.34 | *** part/#asterisk alephcom (n=Weibe@host75.net14.mcsnet.ca) |
| 07:37.36 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 07:40.40 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 07:41.41 | *** join/#asterisk littlejohn (n=little@host77-73.pool8717.interbusiness.it) |
| 07:42.33 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 07:44.12 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 07:44.16 | satlan32 | heelo |
| 07:44.21 | satlan32 | anyone here? |
| 07:44.27 | satlan32 | when i use the forwarding feature, (busy or no answer), i get busy tone and the call is not forwarded to the destination |
| 07:44.39 | satlan32 | any ideas why is that happening?? |
| 07:45.07 | satlan32 | the only time the call is forwarded is when i use forward all calls |
| 07:50.49 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:51.50 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 07:52.42 | RoyK | satlan32: it is quite impossible to help you out without you first pastbinning your dialplan and so on |
| 07:52.43 | RoyK | ~pb |
| 07:52.48 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 07:53.38 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 08:06.21 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 08:09.58 | *** join/#asterisk braniff (n=dfddfd@unaffiliated/braniff) |
| 08:11.11 | braniff | what kind of phone do i need to use asterisk to talk on one voip line ? |
| 08:18.14 | braniff | what do you guys recommend (out of http://www.asterisk.org/hardware) for the simplest possible asterisk-based voip ? |
| 08:18.50 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 08:19.04 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 08:19.18 | RoyK | braniff: just get anything. an ATA or a sip phone or a softphone or whatnot |
| 08:19.49 | luke-jr_ | braniff: why get hardware? |
| 08:20.29 | braniff | luke-jr_, good point...i guess i could just use a mic and earphones..? |
| 08:20.42 | RoyK | yes, and a softphone |
| 08:20.47 | RoyK | or two cans and some string :) |
| 08:20.54 | braniff | heh |
| 08:20.56 | luke-jr_ | braniff: yea |
| 08:21.06 | luke-jr_ | RoyK: Asterisk has an ALSA and OSS channel ;) |
| 08:21.27 | RoyK | iirc those are removed in recent versions |
| 08:21.43 | luke-jr_ | why? :/ |
| 08:21.44 | braniff | what about bluetooth headset ? |
| 08:22.11 | RoyK | because noone used them |
| 08:22.26 | RoyK | i think there are some works on bluetooth |
| 08:22.41 | luke-jr_ | RoyK: noone? IIRC, there was a nice intercom thing using it |
| 08:22.48 | RoyK | luke-jr_: you need asterisk 1.0.x to get them |
| 08:23.10 | RoyK | iirc app_intercom is gone as well |
| 08:23.36 | RoyK | yep |
| 08:24.37 | braniff | is there some way to transmit the sound from, say an openbsd asterisk server to a linux pc ? |
| 08:24.40 | luke-jr_ | my 1.2.7 has them |
| 08:25.25 | braniff | so i could then use a bluetooth headset |
| 08:25.36 | luke-jr_ | softphone |
| 08:26.58 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 08:27.58 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 08:30.01 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 08:30.06 | Zeeek | lo de lo |
| 08:32.00 | satlan32 | RoyK i'm using the default forwarding in asterisk@home 2.8 |
| 08:32.55 | braniff | what's the best platform to run asterisk on for a home voip user?? openbsd, openwrt router, linux pc? |
| 08:33.07 | Zeeek | linux is easiest |
| 08:33.11 | RoyK | satlan32: heh. don't use asterisk@home :P |
| 08:33.30 | RoyK | braniff: i'd say anything |
| 08:33.38 | satlan32 | and if i'm allready using it ? |
| 08:33.42 | RoyK | braniff: whatever makes you feel comfortable |
| 08:34.01 | braniff | well...openbsd gives me warm fuzzies about security... |
| 08:34.06 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 08:34.17 | RoyK | satlan32: well, asterisk@home is nice, but i don't know shit about debugging it. the config files are spread all over and so on. asterisk isn't that hard to learn from scratch |
| 08:34.30 | Greek-Boy | Where can I download cisco SIP firmware without having to register for a service contract? |
| 08:34.52 | RoyK | Greek-Boy: thepiratebay.org |
| 08:34.54 | RoyK | :) |
| 08:35.46 | luke-jr_ | RoyK: I see chan_oss and chan_alsa in HEAD... |
| 08:36.21 | RoyK | head? |
| 08:36.36 | RoyK | hm. trunk |
| 08:36.38 | RoyK | yes |
| 08:36.43 | RoyK | it seems it's there after all |
| 08:36.47 | RoyK | my fault |
| 08:36.50 | Greek-Boy | thanks RoyK |
| 08:36.51 | satlan32 | mmm.. i know, but for now i'm learning asterisk by this way |
| 08:37.19 | RoyK | Greek-Boy: dunno if you can find it there, but.... |
| 08:37.59 | Greek-Boy | no luck |
| 08:38.00 | Greek-Boy | :( |
| 08:38.01 | Greek-Boy | lol |
| 08:39.58 | drray | I jinxed myself this afternoon, I was shooting my mouth off about a 418 day uptime on my asterisk box, and I get a page from work 4 hours later about the pbx being down |
| 08:40.24 | braniff | doh! |
| 08:40.31 | drray | no kidding |
| 08:41.03 | braniff | what platform do you run asterisk on? |
| 08:41.09 | drray | fc3 |
| 08:41.11 | drray | :) |
| 08:41.22 | braniff | ah yes...SELinux enabled by default |
| 08:41.24 | *** join/#asterisk Formater (i=Formater@dhcp-87-116-136-201.cmtsns-ns.customer.sbb.co.yu) |
| 08:41.27 | Formater | hi |
| 08:42.26 | Formater | <PROTECTED> |
| 08:42.30 | Formater | [new_system] |
| 08:42.30 | Formater | type=peer |
| 08:42.30 | Formater | host=213.203.222.xxx |
| 08:42.30 | Formater | port=5061 |
| 08:42.31 | Formater | canreinvite=yes |
| 08:42.31 | Formater | allow=all |
| 08:42.39 | Formater | <PROTECTED> |
| 08:42.49 | Formater | <PROTECTED> |
| 08:42.53 | Formater | <PROTECTED> |
| 08:43.04 | Formater | <PROTECTED> |
| 08:43.04 | Formater | <PROTECTED> |
| 08:43.04 | Formater | <PROTECTED> |
| 08:43.04 | Formater | <PROTECTED> |
| 08:43.04 | Formater | <PROTECTED> |
| 08:43.05 | Formater | <PROTECTED> |
| 08:43.11 | Formater | <PROTECTED> |
| 08:43.12 | Formater | <PROTECTED> |
| 08:43.12 | Formater | <PROTECTED> |
| 08:43.13 | Formater | <PROTECTED> |
| 08:43.13 | RoyK | ~pb |
| 08:43.15 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 08:43.16 | RoyK | ~lart Formater |
| 08:43.27 | Formater | When I make call from old system using new system as termination provider, call goes to default context, and not to 'wholesale' as it is said in sip_buddies... so it does not authenticates properly :( Any idea? |
| 08:43.27 | Zeeek | ~RoyK |
| 08:43.28 | jbot | i guess royk is that viking asterisk guru, or your friend |
| 08:43.41 | RoyK | ~zeeek |
| 08:43.42 | jbot | from memory, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 08:43.53 | Zeeek | hahaha |
| 08:43.57 | braniff | that's funny |
| 08:44.00 | RoyK | :) |
| 08:44.03 | Zeeek | fucking paypal! |
| 08:44.32 | Zeeek | using paypal is like jerking off and not being able to come |
| 08:44.39 | RoyK | 'click here to verify your information' |
| 08:44.58 | Zeeek | Right now I'm verifying that it takes 5 days to make a payment |
| 08:45.17 | Zeeek | paypal sucks |
| 08:45.34 | Zeeek | e-bay is a more evil empire than m$oft |
| 08:45.39 | Zeeek | but I digress |
| 08:46.11 | Zeeek | <PROTECTED> |
| 08:46.52 | litage | are echo cancellers needed for asterisk boxes that don't have any telephony-specific hardware in them (eg: TDM4xx, A104, etc)? |
| 08:47.17 | Zeeek | depends on whether you cant to hear echo :) |
| 08:47.25 | Zeeek | s/cant/want/ |
| 08:47.52 | Zeeek | under the best conditions, there shouldn't be any |
| 08:50.33 | RoyK | litage: they are |
| 08:50.42 | RoyK | litage: but there's decent echocancel in zaptel |
| 08:51.46 | litage | so echo cancellers are sometimes very useful even in IP-only setups? (ie: doesn't use T1s, E1, FXOs, FXSs, etc |
| 08:56.00 | RoyK | oh |
| 08:56.01 | RoyK | no |
| 08:56.02 | RoyK | sorry |
| 08:56.26 | Zeeek | he said no hardware! |
| 08:56.58 | RoyK | Zeeek: running asterisk without hardware? |
| 08:57.17 | Zeeek | it's called asterisk@air |
| 08:57.24 | RoyK | hell. i need both cpu and memory and whatnot..... |
| 08:57.35 | Zeeek | naw, unnecessary |
| 08:57.37 | RoyK | asterisk@void |
| 08:57.43 | Zeeek | it's all virtual |
| 08:57.48 | Zeeek | no dialplan, either |
| 08:57.53 | RoyK | no need for it |
| 08:57.56 | Zeeek | it just guesses where the call should go :) |
| 08:58.16 | Zeeek | in fact, it doesn't answer, since there is no calls |
| 08:58.29 | Zeeek | and it's been up solid for 4 years! |
| 08:58.30 | RoyK | problem with actually running asteirsk on an os on real hardware etc, is it becomes so bloody insecure |
| 08:58.39 | RoyK | and unstable |
| 08:58.47 | Zeeek | just run everything as root and hope for the best |
| 08:59.04 | RoyK | yeah, the asterisk way |
| 08:59.49 | Zeeek | then you either have to 777 the outgoing directory or run everything as root to put stuff there :) |
| 09:05.16 | *** join/#asterisk Slabber (n=9105@jabber.keytrade.com) |
| 09:10.06 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-225-220-3.nsw.bigpond.net.au) |
| 09:10.13 | P-NuT | Hi all. |
| 09:10.35 | Zeeek | hi P |
| 09:10.42 | P-NuT | Does anyone know how to get the handset light to lightup on a cisco phone when you have a message? |
| 09:11.33 | Zeeek | the phone is subscribed to a mailbox? |
| 09:11.53 | P-NuT | yeeeeaahhhh..... um.... maybe? Sorry. I'm flying blind/ |
| 09:12.15 | P-NuT | I have set the mail number on the phone as 8500 |
| 09:12.20 | P-NuT | but it doesnt light up.. |
| 09:12.26 | InfraRed | is there any voodoo to get the mailbox working on * ? |
| 09:12.29 | P-NuT | there must be another number by default |
| 09:12.49 | P-NuT | * the number? or *=asterisk? |
| 09:12.50 | InfraRed | 1. voicemail.conf , 2. add mailbox=blah to sip.conf and fix extensions.conf |
| 09:12.54 | InfraRed | anything else ? |
| 09:13.08 | P-NuT | I thought that was it, |
| 09:13.25 | P-NuT | no the voicemail works fine, it's just the light doesnt show up. |
| 09:14.42 | P-NuT | hmm.. |
| 09:14.44 | P-NuT | strange. |
| 09:15.13 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 09:15.44 | [hC] | so, if i have a PRI (run by a sangoma a102) whats the most reliable way to receive fax on a channel and send it to me via email? |
| 09:15.51 | [hC] | do i need spandsp, or anything special here? |
| 09:17.45 | luke-jr_ | Zeeek: or just have your outgoing-generating app setuid |
| 09:19.29 | *** join/#asterisk ghenry (n=ghenry@81-174-216-113.pth-as9.dial.plus.net) |
| 09:20.05 | s-ndh-c | what could be the problem if incomming calls dont arive their destination? |
| 09:20.32 | s-ndh-c | i can call to outside from my sip phone via my misdn trunk |
| 09:20.40 | s-ndh-c | but not the otherway around |
| 09:22.32 | *** join/#asterisk ghenry (n=ghenry@81-174-216-113.pth-as9.dial.plus.net) |
| 09:23.53 | P-NuT | figured it out, sorry guys! |
| 09:24.12 | Zeeek | luke-jr_ hey tell us about that - I have no idea |
| 09:25.09 | s-ndh-c | asterisk opens a channel for the incomming call but it doesnt ring my sip phone |
| 09:26.53 | s-ndh-c | http://pastebin.com/708173 << this is what i see in my misdn.log |
| 09:27.01 | s-ndh-c | can someone maybe tell me whats wrong? |
| 09:31.21 | Slabber | hello everyone... I've got a problem with a TE110P missing IRQs.. it seems to be missing them all, i.e. all 1000 per second. Anyone know how to fix it? |
| 09:32.53 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:33.11 | puzzled | morning |
| 09:33.35 | [hC] | hmm. any of you using spandsp 0.0.3? |
| 09:33.54 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 09:34.24 | puzzled | [hC]: nope, asterisk 1.2.x which I use needs 0.0.2pre26 |
| 09:34.31 | [hC] | yeah thats what im just finding out |
| 09:34.39 | [hC] | they havent put up the app_ files for 0.0.3 yet |
| 09:34.46 | RoyK | [hC]: don't use 0.0.3 |
| 09:34.49 | RoyK | it's for development |
| 09:34.50 | RoyK | rtfm |
| 09:34.51 | RoyK | :) |
| 09:34.54 | *** join/#asterisk alucard064 (n=alucard0@ABayonne-152-1-47-104.w83-193.abo.wanadoo.fr) |
| 09:35.06 | puzzled | [hC]: check the main download dir from spandsp. there is a patch for libtiff which fixes garbled faxes iirc |
| 09:35.11 | drray | so my asterisk box lost its mind today |
| 09:35.28 | [hC] | oh dear.. ive installed libtiff from debian |
| 09:35.59 | puzzled | [hC]: I don't know if the patch has been accepted yet and in which version so seems some manual labor is involved :) |
| 09:36.07 | RoyK | [hC]: i use libtiff from debian in production |
| 09:36.12 | RoyK | [hC]: and have been for moths |
| 09:36.15 | Zeeek | has anyone has really good luck with spandsp? I've been using it for over a year and it receives one out of four faxes. The other just die (no distortion, just no fax) |
| 09:36.33 | [hC] | royk: well, i guess ill give it a go, and see if it causes me any problems |
| 09:36.44 | RoyK | Zeeek: as i said, been using rxfax for moths in production, lots of faxes daily |
| 09:37.00 | Zeeek | yes but does it work? :) |
| 09:37.05 | RoyK | indeed |
| 09:37.11 | RoyK | receive from a PRI |
| 09:37.14 | RoyK | send by email |
| 09:37.26 | [hC] | thats what im trying to get going here |
| 09:37.32 | [hC] | on my sangoma-driven pri |
| 09:37.39 | RoyK | that's what we use |
| 09:37.44 | [hC] | royk: which version of libtiff and spandsp have you got going? |
| 09:37.59 | Zeeek | I'm receiving on a X100P - that may not help |
| 09:38.07 | drray | so if I wanted to totally gut my asterisk install, and reinstall, what do I need to delete? and can I just copy zaptel. zapata. extensions. voicemail. sip. manager. and iax.conf? |
| 09:38.26 | Zeeek | these same faxes are received without problem on the same phone line with a PC freeware |
| 09:38.38 | RoyK | [hC]: /usr/lib/libtiff.so.4 -> libtiff.so.4.1.2 |
| 09:39.24 | Formater | my question, how to setup an entry in sip.conf to allow other asterisk to connect to it using IP auth only?:) |
| 09:39.25 | RoyK | [hC]: and spandsp 0.0.2pre26 |
| 09:39.26 | RoyK | iirc |
| 09:39.34 | RoyK | might be pre25, but i think it's pre26 |
| 09:44.01 | [hC] | apparently debian unstable (i know, my mistake) has a newer libtiff |
| 09:44.01 | [hC] | <PROTECTED> |
| 09:44.07 | [hC] | ill give it a shot and see how she goes. |
| 09:44.41 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-235-195.telkomadsl.co.za) |
| 09:46.12 | RoyK | [hC]: using unstable debian in production isn't really a good idea |
| 09:47.03 | RoyK | "The “unstable†distribution is where active development of Debian occurs. Generally, this distribution is run by developers and those who like to live on the edge." |
| 09:47.05 | RoyK | "The “unstable†distribution is called sid." |
| 09:47.13 | RoyK | "Sid is the kid that breaks toys" |
| 09:47.38 | RoyK | [hC]: http://www.debian.org/releases/ |
| 09:50.20 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 10:08.19 | *** join/#asterisk X-Rob_ (n=rob@CPE-58-166-164-176.qld.bigpond.net.au) |
| 10:08.51 | *** join/#asterisk Slabber (n=9105@jabber.keytrade.com) |
| 10:17.09 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 10:17.22 | Greek-Boy | Damn, I can't get the cisco firmware I need :( |
| 10:18.22 | InfraRed | too bad |
| 10:18.37 | InfraRed | try #hot-cisco-warez |
| 10:19.07 | Greek-Boy | lol |
| 10:19.18 | Greek-Boy | why does everyone think its a big joke? |
| 10:19.19 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 10:20.45 | InfraRed | buying cisco means you are bound to their terms and conditions of licensing firmware and paying for it |
| 10:20.58 | InfraRed | which may include being laughed at in irc channels |
| 10:21.24 | InfraRed | and the risk of gangbang by cisco executives |
| 10:25.10 | Greek-Boy | even for SIP firmware? |
| 10:27.26 | *** join/#asterisk DannyF (n=wizardon@dsl-cust-83-172-73-34.kringdata.net) |
| 10:28.44 | fenlander | especially for sip firmware |
| 10:28.59 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 10:31.08 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 10:31.16 | DannyF | lo folks |
| 10:32.21 | RoyK | .... .. |
| 10:34.30 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 10:42.44 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 10:48.13 | P-NuT | Hey all, |
| 10:48.22 | *** join/#asterisk Teeli (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 10:48.34 | P-NuT | when people call my asterisk server it's supposed to call my cisco phone, |
| 10:48.37 | P-NuT | which t does, |
| 10:49.03 | InfraRed | P-NuT: fix your extensions.conf |
| 10:49.10 | P-NuT | but when I want to redirect them to voicemail if unavailable, it goes to the voicemail main menu instead.. |
| 10:49.20 | P-NuT | I haven't asked yet! |
| 10:50.28 | InfraRed | touchy |
| 10:51.00 | P-NuT | sorry |
| 10:51.05 | P-NuT | here's the log |
| 10:51.05 | P-NuT | http://pastebin.ca/65440 |
| 10:52.30 | P-NuT | any ideas> |
| 10:52.47 | P-NuT | I'll paste my extensions... |
| 10:53.40 | P-NuT | http://pastebin.ca/65441 |
| 10:54.03 | P-NuT | that's what it diverts to, but it goes to the voicemail main menu |
| 10:54.07 | P-NuT | any ideas? |
| 10:54.41 | InfraRed | paste voicemail.conf |
| 10:54.44 | P-NuT | k |
| 10:54.46 | InfraRed | and extensions.conf |
| 10:55.11 | P-NuT | I cant paste the whole thing |
| 10:55.39 | P-NuT | actually let me try something... |
| 10:57.16 | DannyF | anyone been playing with penalties in realtime queues? |
| 11:08.02 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 11:08.31 | *** join/#asterisk Sonderblade (n=mah@host-213.131.147.169.addr.tdcsong.se) |
| 11:10.13 | Sonderblade | anyone know what is causing this error: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)? |
| 11:12.37 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 11:13.20 | RoyK | anyone here using sangoma cards with echocancel? |
| 11:15.36 | Formater | hm, when using real time auth, and want IP auth via SIP.. asterisk checks the 'name' field with the IP and not the 'host'.. very strange |
| 11:17.20 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 11:23.40 | *** join/#asterisk inspired (n=mikael@213.197.167.52) |
| 11:28.35 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 11:28.37 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-254-10.telkomadsl.co.za) |
| 11:30.07 | *** part/#asterisk littlejohn (n=little@host77-73.pool8717.interbusiness.it) |
| 11:32.25 | *** join/#asterisk nibbler_de (n=nibbler@some.host.name) |
| 11:32.35 | nibbler_de | hey ;) |
| 11:33.45 | nibbler_de | just wanted to know how you all solve the common problem that cisco 7960/7940 phones when asked to dial a sip url (a la home@voip.nibbler.de) try to dial it via asterisk which fails since it is not in the extensions.conf |
| 11:36.14 | P-NuT | hey guys |
| 11:36.15 | inspired | tell asterisk to look up srv records? |
| 11:36.19 | inspired | check in sip.conf for srv |
| 11:37.15 | P-NuT | if I want to create an voicemail extension which you dial, it works out your extension number and only prompts you for a password, how to I do that? |
| 11:42.16 | RoyK | anyone here using sangoma cards with echocancel? A104D? |
| 11:43.14 | nibbler_de | inspired: srvlookup=yes already :( |
| 11:45.43 | nibbler_de | P-NuT: try VoicemailMain but this won't ask for your pin |
| 11:48.40 | geoffl | nibbler_de: won't VoiceMailMain ask for the mailbox name if used without paramers, password only if you pass the name of the mailbox, and nothing if you prefix the mailbox name with "s". So, VoiceMailMain(101) would access the voicemail for 101 and ask for the password, but VoiceMailMain(s101) would access the voicemail for 101 and bypass the password? |
| 11:49.40 | nibbler_de | geoffl: uhm, yes. |
| 11:49.50 | *** join/#asterisk pjo (n=pjo@212.88.98.114) |
| 11:50.05 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 11:51.15 | nibbler_de | geoffl: so for P-NuT it would just be VoiceMailMain(${CALLERIDNUM}) |
| 11:51.19 | nibbler_de | which does auth |
| 11:55.11 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 11:55.53 | *** join/#asterisk Szolke (n=Szolke@22-36.adsl.etel.hu) |
| 11:56.49 | Szolke | Hi all. Can you help me to upgrade Our Sangoma card's driver? I downloaded the last version from Sangoma's web. |
| 11:58.37 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-11-243.w81-248.abo.wanadoo.fr) |
| 12:01.15 | s-ndh-c | Szolke: RTFM? |
| 12:03.58 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 12:03.59 | P-NuT | nibbler_de: THANKS MATEY!! woo hoo! |
| 12:04.04 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 12:06.31 | _4d4m_ | hi all. am looking at enabling ztdummy on 1.2.9.1 for meetme. an lsmod on the server shows usbcore, not usb_uhci or ohci. anyone know whether ztdummy will work? |
| 12:07.27 | Szolke | s-ndh-c: Thanx for the F help |
| 12:07.41 | *** join/#asterisk pigpen2 (n=mark@67.158.33.234) |
| 12:08.01 | s-ndh-c | hehe |
| 12:09.56 | _4d4m_ | nm, it aint gonna work.. will have to look at a 2.6 kernel |
| 12:10.16 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:11.45 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 12:12.34 | littleball | hello, who has experience of integrating ser with asterisk? asterisk works as media relay server . |
| 12:13.17 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 12:13.21 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:13.55 | *** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk) |
| 12:15.41 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 12:17.47 | RoyK | zoa: ping |
| 12:17.57 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 12:25.54 | *** join/#asterisk Godsey (i=jason@pdpc/supporter/sustaining/Godsey) |
| 12:26.47 | Godsey | might anyone know a reliable toll free inbound provider with either sip or iax2 that is responsive to customer support? |
| 12:28.38 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:29.37 | zoa | pong |
| 12:30.24 | SplasPood | Anyone here ever played /w Sphinx2 speech-to-text? If so, is it any good? |
| 12:31.18 | RoyK | zoa: I have this seriously strange problem with the jitterbuffer on a box here. sip/pstn calls are dejittered, but calls from pstn to sip are not :s |
| 12:32.35 | zoa | huh ? they are exactly the same ? |
| 12:32.43 | zoa | only the pstn side gets dejittered of course |
| 12:32.56 | zoa | but sip to pstn or pstn to sip should be 100% the same |
| 12:36.33 | RoyK | zoa: the way it looks, only calls initiated from SIP are dejittered on that box |
| 12:36.58 | RoyK | zoa: i'll do more testing, rolling back versions and so on before saying more, though |
| 12:37.11 | zoa | :) |
| 12:37.11 | zoa | oki |
| 12:40.19 | littleball | hello i try to integrate ser with asterisk. because i want to use asterisk to act as media relay server. how to make it work? |
| 12:41.39 | dongs | probably very carefully. |
| 12:41.42 | *** join/#asterisk \lart (i=nunya@neo.jasons.org) |
| 12:41.43 | dongs | and you better hope youre a programmer |
| 12:42.00 | RoyK | littleball: you do not want to use asterisk to do that |
| 12:42.03 | RoyK | littleball: beleive me |
| 12:42.20 | RoyK | littleball: you want something like mediaproxy or even whatever comes out of sitsotd.org |
| 12:42.27 | RoyK | littleball: you do NOT want asterisk |
| 12:43.03 | \lart | morning all.. about 2 weeks ago i got a linksys wip300 phone. Works great. Yesterday, my boss sent us all Nokia E61's, which also have wifi and sip capabilities. Obviously, at this point I can't return the phone, so before I throw it on eBay, I thought I'd see if anyone around here was in the market.. |
| 12:43.31 | \lart | if you're interested, /msg me. |
| 12:43.37 | dongs | i:ll take it for free if you pay shipping to japan |
| 12:43.56 | \lart | I had slightly more in mind than free. :) |
| 12:44.41 | iDunno | \lart: I'll give you 50p + postage for it ;) |
| 12:44.42 | mut | fie dolla! |
| 12:44.44 | \lart | a tempting offer, nonetheless. :) |
| 12:44.56 | iDunno | (assuming postage of < a 10GBP) |
| 12:45.00 | \lart | i paid $219 from voxilla. |
| 12:45.26 | iDunno | ahh - so it is only worth ~ 10GBP then ;) |
| 12:45.34 | dongs | the only time ive ever seen wifi sip phones was at trade shows |
| 12:45.39 | \lart | well, the dollar isn't as strong as it used to be... |
| 12:45.45 | dongs | japs dont sell any kind of voip shit here unless its some overpriced NTT crap |
| 12:46.09 | dongs | what a pity too, i heard BT101/s are like $40 in u.s. |
| 12:46.13 | *** join/#asterisk mvdk (i=mvdk@60-240-15-230-nsw-pppoe.tpgi.com.au) |
| 12:46.45 | mvdk | Quick Q: I just managed to get iaxmodem working with hylafax/asterisk |
| 12:47.18 | dongs | is iaxmodem what i think it is |
| 12:47.31 | mvdk | I would like to set up a system where asterisk indicates fast busy (congestion) if fewer than <n> lines are available |
| 12:47.43 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
| 12:47.45 | mvdk | (to particular clients only) |
| 12:47.49 | dongs | you can probably do that with some kinda script. |
| 12:47.51 | \lart | dongs, looks closer to $50 |
| 12:48.00 | key2 | Could someone tell me how to use an OutBound Proxy with asterisk ? |
| 12:48.19 | dongs | key2: dont, get a real IP. |
| 12:48.28 | mvdk | In particular, the ones that iaxmodem uses. I was wondering if there was some kind of way to do that with a GotoIf, or something |
| 12:48.35 | dongs | \lart: still better than i could get it locally. |
| 12:48.40 | mut | anyone know how to change the default paper size in internet explorer? i found registry settings to change margins and header/footer but not paper size or orientation |
| 12:48.54 | mvdk | Mut: Go ask the lusers channels |
| 12:49.03 | mvdk | This is for asterisk.... |
| 12:49.07 | mut | o hush |
| 12:50.07 | mvdk | dongs: BTW, iaxmodem is a softmodem that communicates over IAX |
| 12:50.21 | mvdk | It is available at iaxmodem.sourceforge.net |
| 12:50.24 | dongs | interesting |
| 12:50.50 | dongs | wow. |
| 12:50.56 | dongs | does it do better than 9600 for fax? |
| 12:51.01 | mvdk | Yeah, it does |
| 12:51.05 | dongs | no shit |
| 12:51.05 | mvdk | It can do 14.4 |
| 12:51.14 | dongs | so its not based on spandsp code? |
| 12:51.21 | mvdk | It is, but it's patched |
| 12:51.22 | dongs | something newer/better? |
| 12:51.25 | dongs | ahi see |
| 12:51.40 | mvdk | Only thing is, the code to do 14.4 is alpha |
| 12:51.50 | coppice | iaxmodem doesn't do 14.4K properly |
| 12:51.54 | mvdk | (So he recommends that it be disabled) |
| 12:52.04 | mvdk | And of course, I don't enable it |
| 12:52.28 | mvdk | But I don't care - I just use it to communicate the odd quote to someone that gets uptight about that kind of shit |
| 12:52.45 | \lart | oh well, train's pulling in.. later all.. |
| 12:52.51 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 12:53.17 | dongs | so what does it allow me to do anyway |
| 12:53.23 | dongs | fax over voip? omg |
| 12:53.27 | mvdk | Not quite |
| 12:53.36 | mvdk | I use the digium card to send faxes out.... |
| 12:53.47 | [TK]D-Fender | mvdk : For your <n> lines question "show application chanisavail" |
| 12:54.07 | dongs | i would use a digium card |
| 12:54.10 | mvdk | Oh, that sounds cool.... Does it say how many are available? |
| 12:54.11 | dongs | if all my modules werent burned |
| 12:54.39 | mvdk | Just googling it, don't worry :) |
| 12:54.47 | [TK]D-Fender | mvdk : Not all at once. You'd doa little diaplan script to count them. |
| 12:54.59 | mvdk | Hmmm.... |
| 12:55.19 | mvdk | It sounds like I might be better off using the math package, with a little skullduggery.... |
| 12:55.33 | *** join/#asterisk clive- (n=pirch@dsl-146-92-67.telkomadsl.co.za) |
| 12:55.49 | [TK]D-Fender | mvdk : What kind of lines are you checking? |
| 12:55.56 | mvdk | Zaptel lines |
| 12:56.02 | [TK]D-Fender | mvdk : PRI? |
| 12:56.08 | mvdk | No, POTS |
| 12:56.11 | mvdk | (ATM) |
| 12:56.11 | [TK]D-Fender | mvdk : And sequential? |
| 12:56.11 | *** join/#asterisk trelane_ (i=trelane@pdpc/supporter/sustaining/trelane) |
| 12:56.22 | mvdk | But possibly PRI in future |
| 12:56.30 | mvdk | Not necessarily |
| 12:56.36 | [TK]D-Fender | mvdk : a dozen lines of macro to do this..... |
| 12:56.37 | clive- | hi, will disabling uhci_hcd have a bad effect on asterisk? |
| 12:56.52 | [TK]D-Fender | clive- : Only if your relied on it for ztdummy... |
| 12:56.53 | mvdk | If you are using a 2.4 kernel, yeah |
| 12:57.10 | mvdk | (and then only if you don't have a zaptel card) |
| 12:57.27 | clive- | I am using ztdummy on kernel 2.6 |
| 12:57.35 | mvdk | Then you don't need to worry |
| 12:57.44 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 12:57.47 | clive- | so I just rmmod uhci_hcd and hope forthe best |
| 12:57.51 | clive- | :) |
| 12:57.55 | dongs | well |
| 12:57.58 | dongs | if you arent using it |
| 12:58.02 | dongs | why is it even loaded |
| 12:58.04 | dongs | or compiled |
| 12:58.07 | mvdk | Only if you're not using it |
| 12:58.17 | mvdk | Because you likely have such a device anyway? |
| 12:58.23 | mvdk | And you're using a distro kernel? |
| 12:58.50 | clive- | I dunno, my interrupts show this uhci_hcd loaded, and its like all over the place |
| 12:58.57 | mvdk | And so, the hotplug/udev scripts detect the device, and loads the module? |
| 12:59.07 | dongs | lol hotplug |
| 12:59.10 | dongs | people actually USE that? |
| 12:59.17 | mvdk | Some people do.... |
| 12:59.27 | mvdk | Depends on how ancient your installation is |
| 12:59.38 | clive- | I am running centos 4.2 |
| 12:59.52 | mvdk | Point is, you're using udev in all likelyhood |
| 13:00.03 | clive- | yes, I am |
| 13:00.11 | mvdk | Which means that short of renaming the module, then rebooting, you can't remove it |
| 13:00.32 | mvdk | But why does it bother you? |
| 13:00.54 | mvdk | God, it's perhaps 200k, I don't see it as a problem.... |
| 13:01.14 | mvdk | (200k would mean that it's *really* fat....) |
| 13:02.08 | *** join/#asterisk ManxPower (i=ewieling@52.sub-70-210-154.myvzw.com) |
| 13:02.18 | clive- | well, Its shares an interrupt with my eicon diva server card, and when the box gets busy, the voice quality gets worse, so I am thinking that this may be the cause of the quality issue |
| 13:03.03 | mvdk | Ah, when the box gets busy, voice quality will get worse due to echo cancellation, etc. not happening as quickly |
| 13:03.26 | mvdk | Asterisk cares very much about FPU performance.... |
| 13:03.54 | mvdk | The easiest solution is probably to renice asterisk |
| 13:04.06 | clive- | well the wierd thing is that eicon does hardware echo cancellation, so I can see any reason why its gets worse |
| 13:04.10 | clive- | renice? |
| 13:04.12 | *** join/#asterisk ast_freak (n=jesse@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 13:04.22 | mvdk | Give it a larger chunk of CPU time |
| 13:04.33 | clive- | how do you do that |
| 13:04.41 | dongs | i dont think thats goona help much |
| 13:05.02 | ManxPower | if you share interrupts, you will have problems |
| 13:05.08 | mvdk | "renice" is a command; Type "man renice" to find out more |
| 13:05.21 | ManxPower | Asterisk already has a command line option to run at pseudorealtime priority |
| 13:05.46 | mvdk | Yeah, so use it |
| 13:06.19 | clive- | manx, I think you missed teh beginnning of the discussion where my eicon E1 card shares an interrupt with uhci_hcd |
| 13:06.42 | clive- | manx, what command line option is this ? |
| 13:06.42 | *** part/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 13:06.43 | mvdk | Yeah, but if you're not using this USB device, how could it be causing problems? |
| 13:06.46 | ManxPower | clive-, Make it not share interrupts. If you share interrupts expect problems |
| 13:06.52 | kmilitzer | Anyone coming to Astricon Berlin next week? |
| 13:07.02 | ManxPower | if you are not using the USB, then make the driver not load. |
| 13:07.07 | clive- | this uhci_hcd doenst want to go away :( |
| 13:07.09 | ManxPower | This isn't rocket sicence, people. |
| 13:07.09 | mvdk | Maybe I'm just not thinking right..... |
| 13:07.19 | mvdk | In any case, you can put it in the blacklist |
| 13:07.23 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 13:07.23 | clive- | I disables USB on the motherboard and it still loads |
| 13:07.24 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 13:07.39 | ManxPower | how abour rmmod uhci_hcd |
| 13:07.49 | ManxPower | clive you need to disable it in /etc/modules.conf |
| 13:07.53 | clive- | I still classify myslef as a newbie ..:) |
| 13:08.00 | mvdk | It's going to want rmmod -f uhci_hcd |
| 13:08.02 | dongs | why dont you compile a damn kernel and get it done |
| 13:08.07 | mvdk | And that might cause a hard crash |
| 13:08.23 | ManxPower | if it causes a crash, then one might assume that SOMETHING is using that kernel module. |
| 13:08.35 | mvdk | Yeah, like this USB host |
| 13:08.49 | mvdk | But nothing's dangling off it, he seems to have said... |
| 13:08.57 | dongs | if you disable the device in the bios and it still loads, youve got other issues |
| 13:09.00 | mvdk | Perhaps you can hard-assign a different IRQ? |
| 13:09.10 | mvdk | (to the card)? |
| 13:09.39 | ManxPower | http://www.google.com/search?hl=en&sa=X&oi=spell&resnum=0&ct=result&cd=1&q=linux+kernel+module+prevent+loading&spell=1 |
| 13:10.22 | *** join/#asterisk jixi (n=damien@193.190.210.151) |
| 13:11.03 | mvdk | In any case, adding "alias uhci_hcd off" to modules.conf will get the job done |
| 13:11.21 | ManxPower | http://www.cyberciti.biz/nixcraft/vivek/blogger/2006/03/how-do-i-stop-linux-kernel-module-from.php |
| 13:11.28 | mvdk | You will, of course, have to reboot |
| 13:11.38 | ManxPower | mvdk, I usually expect dinner and drinks before holding someone's hand. |
| 13:11.56 | mvdk | Well, yeah, but I want to get back to my problem :) |
| 13:12.03 | jixi | RxFax("/var/spool/asterisk/file.tiff"||debug) doesn't give any debug |
| 13:12.05 | jixi | any hint? |
| 13:12.18 | mvdk | Yeah, use iaxmodem+hylafax instead :) |
| 13:12.33 | ManxPower | jixi, /etc/asterisk/logger.conf |
| 13:12.33 | jixi | seriously? |
| 13:12.34 | mvdk | RxFax, what a heap of junk |
| 13:12.49 | ManxPower | iaxmodem works with Zap? |
| 13:12.53 | mvdk | Yeah, it does |
| 13:12.57 | ManxPower | That's a suprize |
| 13:13.01 | mvdk | I've set it up, it's great |
| 13:13.23 | mvdk | I don't know why people waste their time on rxfax.... |
| 13:13.36 | ManxPower | mvdk, because it works and is trivial to set up. |
| 13:13.45 | ManxPower | Unlike the jedi magic required to do hylafax |
| 13:14.04 | mvdk | Hmm? On debian, anyway, there's not jedi magic involved.... |
| 13:14.39 | mvdk | Just take the config file from the iaxmodem distrib, and presto, everything works :) |
| 13:14.52 | *** part/#asterisk Godsey (i=jason@pdpc/supporter/sustaining/Godsey) |
| 13:14.54 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 13:15.01 | clive- | Thanks for your help , time to go try this all out |
| 13:15.10 | X-Gen | mvdk, would it be trivial to get hylafax to make printed pages from recordings of fax conversations ? |
| 13:15.19 | ManxPower | Regardless, iaxmodem is a bad name for it if it works on Zap |
| 13:15.20 | coppice | mvdk: i stopped counting the happy users of rxfax when I got to 10,000 |
| 13:15.23 | dongs | i played with rxfax/txfax a while ago and it was all fucked up |
| 13:15.38 | dongs | not to mention it never acutally received a fax successfully |
| 13:15.46 | coppice | ManxPower: its called iaxmodem, because it works over iax |
| 13:15.47 | mvdk | iaxmodem speaks over an IAX channel |
| 13:15.52 | X-Gen | ~rxfat |
| 13:15.58 | X-Gen | ~rxfax |
| 13:16.02 | mvdk | But of course, asterisk can bridge it to a Zap channel |
| 13:16.16 | ManxPower | Only an idiot would send faxes over VoIP |
| 13:16.25 | mvdk | It is recommended that you use it over a reliable netowkr |
| 13:16.28 | mvdk | *network |
| 13:16.35 | ManxPower | My rxfax gets 10 - 20 pages of faxes per day. |
| 13:16.36 | mvdk | Like the loopback interface.... |
| 13:17.07 | coppice | ManxPower: its only intended to work between well controlled boxes on a LAN. its just spandsp with a thin IAX wrapper around it |
| 13:17.26 | X-Gen | ManxPower, could rxfax handle an E1 (30 channels) of fax's comming in at the same time ? |
| 13:17.40 | ManxPower | coppice, *yawn* I'll stick to using Zap for faxing. |
| 13:17.49 | ManxPower | X-Gen, ask coppice, he wrote it |
| 13:17.50 | mvdk | Point is, you don't have to put up with the awful hacks that asterfax uses..... |
| 13:18.13 | jixi | Could that explain why app_rxfax doesn't work: "Dropping incompatible voice frame on mISDN/1-1 of format slin since our native format has changed to alaw" |
| 13:18.17 | ManxPower | mvdk, WHAT hacks? |
| 13:18.19 | dongs | why the hell od people still bother with faxes |
| 13:18.19 | coppice | X-Gen: some people have trouble at 3 or 4 concurrent FAXes. many saturate a couple of E1s |
| 13:18.28 | jixi | It was working nicely with Asterisk 1.0.9, but the upgrade to 1.2.9.1 broke everything |
| 13:18.37 | mvdk | Well, writing to the same spool directory.... |
| 13:19.00 | mvdk | Because it needs to be on the same machine, easy scalability is shot |
| 13:19.02 | ManxPower | mvdk, um, rxfax just writes the .tiff wherever you tell it to. |
| 13:19.12 | coppice | dongs: that is the real *key* question :-) |
| 13:19.17 | mvdk | Yeah, and txfax? |
| 13:19.27 | dongs | reads a tiff file from wherever you tel lit to. |
| 13:19.31 | mvdk | Yeah, it reads from the .tiff you tell it to |
| 13:19.32 | ManxPower | We are talking about rxfax, not txfax. |
| 13:19.32 | clive- | mvdk I dont have an /etc/modules.conf file |
| 13:19.37 | mvdk | Do you see the problem yet? |
| 13:19.59 | ManxPower | I have not figured out how to convert MS OFFICE documents (which are all my idiot users know) into a format a fax can send |
| 13:20.00 | jixi | nothing on the bugtracker, only unanswered emails on the mailing list about that NOTICE message... So I'm puzzled. |
| 13:20.06 | mvdk | clive-: What's the output of "which udevd" |
| 13:20.22 | ManxPower | so I don't use txfax |
| 13:20.25 | mvdk | Manxpower: That's a solved problem for me |
| 13:20.25 | jixi | btw, I'm not using Fax over IP, only app_rxfax with an mISDN channel |
| 13:20.34 | clive- | mvdk /sbin/udevd |
| 13:20.39 | ManxPower | mvdk, you killed all your users? |
| 13:21.02 | mvdk | clive-: Create a /etc/modules.conf with the contents that we spoke of, then |
| 13:21.20 | mvdk | Manxpower: No, I installed WHFC on their desktops, and used the redirecting print driver |
| 13:21.39 | puzzled | ManxPower: have you looked at the clients section of hylafax? iirc they have apps that you can use with Office docs |
| 13:21.42 | mvdk | That submits it to Hylafax, which submits it to iaxmodem |
| 13:21.57 | ManxPower | mvdk, Ah. We don't have the staff to do that on almost 400 machines, even if we could get the 200 or so laptops into the office. |
| 13:22.00 | mvdk | Which gets bridged over the PSTN :) |
| 13:22.19 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:22.26 | mvdk | Manxpower: Surely you can just hire a bunch of school kids for a few hours.... |
| 13:22.42 | ManxPower | mvdk, no we can't. |
| 13:22.53 | mvdk | Uh, are you a defence user, or something? |
| 13:23.06 | JackEStorm | mvdk: or runs a 900 service on * :) |
| 13:23.07 | ManxPower | mvdk, worse, real estate. |
| 13:23.12 | puzzled | lol |
| 13:23.39 | mvdk | JackEStorm: Could you clarify what you mean? |
| 13:23.46 | ManxPower | 3 years ago the web person moved into the marketing department. Management appoved hireing another MIS person LAST WEEK. |
| 13:24.01 | ManxPower | So it too three years to get back to the same staffing level, the company has doubled in size. |
| 13:24.01 | mvdk | Manx: I feel your pain.... |
| 13:24.39 | mvdk | Still, they won't listen to the idea of giving a few kids a job for a few hours? |
| 13:24.58 | ManxPower | Until Jun 1 there was 1 helpdesk person, 1 MIS manager (and pretty much anything else he had to be) and 1 consultant (me) for 390 users in 19 offices in 2 states. |
| 13:25.16 | mvdk | Or, it sounds like you guys are using SMS or something to distribute applications |
| 13:25.20 | mvdk | Is that the case? |
| 13:25.29 | ManxPower | distribute applications? |
| 13:25.40 | *** join/#asterisk tuxick (n=userMurf@tuxick.xs4all.nl) |
| 13:25.40 | mvdk | You know, install them on user's desktops |
| 13:25.45 | mvdk | *users' |
| 13:25.49 | ManxPower | HAHAHAHAH!!!!! |
| 13:25.50 | tuxick | whoa, big channel :) |
| 13:25.51 | ManxPower | no. |
| 13:25.55 | mvdk | Oh, OK |
| 13:26.03 | mvdk | No automation at all? |
| 13:26.05 | ManxPower | Users use MS Office, and web based applications. |
| 13:26.11 | mvdk | OMG |
| 13:26.24 | ManxPower | The web based applications use ActiveX. |
| 13:26.58 | dongs | awesome |
| 13:27.03 | dongs | windows is great. |
| 13:27.06 | mvdk | Not hardly.... |
| 13:27.11 | ManxPower | We would like to automate the corporate side of things, but we can't find anyone qualified to install Samba and LDAP |
| 13:27.17 | mvdk | I don't know in what parallel universe you live, dongs.... |
| 13:27.20 | ManxPower | so each office is stand alone. |
| 13:27.27 | mvdk | Manx: Where are you guys |
| 13:27.28 | dongs | mvdk: the one called "enterprise" |
| 13:27.39 | dongs | mvdk: where people are getting paid to get shit done |
| 13:27.41 | ManxPower | mvdk, Corporate HQ is 30 miles north of New Orleans |
| 13:28.00 | dongs | mvdk: as opposed to dicking around wiht half working shit |
| 13:28.00 | mvdk | Oh, OK |
| 13:28.01 | mvdk | dongs: I'm quite familiar with that environment |
| 13:28.08 | mvdk | It just so happens I work in it |
| 13:28.12 | dongs | well then using windows should be no surprise for you! |
| 13:28.20 | dongs | because corporations like things that work! and work good! |
| 13:28.25 | mvdk | Yeah, but the fiction about it working well..... |
| 13:28.33 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 13:28.44 | dongs | well,then perhaps you need to hire some real admins! not some unshaved lunix dorks who irc all day at work |
| 13:28.44 | ManxPower | the 1 windows box we have has to be rebooted twice a week |
| 13:28.49 | ManxPower | ..er... 1 windows SERVER |
| 13:28.54 | dongs | because wiht proper administration, windows networks are amazing |
| 13:29.05 | mvdk | dongs: Pull your head out of your butt |
| 13:29.15 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 13:29.29 | mvdk | MS fanboys should go join their own channel :) |
| 13:29.44 | ManxPower | With The New Guy finally here, I may be able to do more proactive projects. |
| 13:30.07 | mvdk | And Manx, sorry, I can't help you. I live in Sydney, AU.... |
| 13:30.33 | dongs | i will tell you onething |
| 13:30.39 | dongs | samba3 is a pile of feces |
| 13:30.39 | mvdk | Yes dongs? |
| 13:30.43 | ManxPower | I've basically given up on installing Samaba and LDAP. I'll do the LDAP stuff as a single place to authenticate against, then find a samba person later |
| 13:30.45 | dongs | performance is horrible |
| 13:30.54 | dongs | we upgraded a bunch of machines (dont ask) from 2.2 to 3 |
| 13:31.04 | ManxPower | dongs, Perhaps so, but none of the servers having more than 15 users..... |
| 13:31.08 | dongs | and disk reads/writes tanked, i mean we:re talking like 400k/sec on 100mbit network that used to be > 8000k/sec |
| 13:31.11 | Makenshi | heh i have to agree, we are migrating away from samba+openldap to server 2k3 |
| 13:31.12 | mvdk | dongs: You should perhaps have read the release notes, dongs..... |
| 13:31.17 | dongs | mvdk: really now |
| 13:31.20 | eBody | what hardware do i need to work with Asterisk and our existing 10 incoming analog lines? |
| 13:31.31 | mvdk | Yeah, no samba 4 for you.... |
| 13:31.34 | dongs | eBody: p3/p4 and 3 digium cards |
| 13:31.39 | mvdk | Oh, you went samba 3 |
| 13:31.40 | mvdk | Sorry |
| 13:31.51 | Makenshi | the biggest reason is the support for certificate autoenrollment which will make using our corporate wireless network dead simple, and yet very secure |
| 13:31.57 | ManxPower | We don't CARE about performance. We care about ease of admin |
| 13:32.05 | dongs | heh, that too |
| 13:32.12 | mvdk | But how the f*ck did you manage to get R/W performance to tank on Samba 3? |
| 13:32.15 | Makenshi | but there are plenty of other features that samba doesn't offer like group policy |
| 13:32.18 | mvdk | What version did you use? |
| 13:32.26 | dongs | mvdk: its consistent across every machine |
| 13:32.30 | sevard | In sip.conf I can set ToS for signaling, but what about RTP? |
| 13:32.31 | mvdk | And did you take this to the relevant mailing list? |
| 13:32.33 | dongs | its orders of magnitude worse than 2.2 |
| 13:32.41 | dongs | i took it to their irc channel few months ago |
| 13:32.44 | dongs | and promptly got banned |
| 13:32.48 | tRSS | how does ACD (automatic call distributor) work in asterisk? |
| 13:32.49 | mvdk | dongs: Which point release of 3.0? |
| 13:32.51 | ManxPower | So what WOULD it cost to buy 20 Windows server licenses and licenses for 400 users? |
| 13:32.59 | dongs | mvdk: whichever it was few months ago. |
| 13:33.11 | puzzled | ManxPower: your soul |
| 13:33.13 | mvdk | dongs: You should be aware that some difficult problems are better asked on the mailing list, and of course, it helps to be polite |
| 13:33.14 | dongs | mvdk: we dont care anymore, the few machiens that HAD to be samba went back to 2.2 and the rest were replaced with windows. |
| 13:33.14 | Makenshi | Manx, if you use virtualisation, you need only pay for 5 server licenses |
| 13:33.24 | ManxPower | Makenshi, Huh? |
| 13:33.32 | *** join/#asterisk bernardovieira (n=bvieira@c911935d.bhz.virtua.com.br) |
| 13:33.34 | dongs | eBody: p3/p4 is a processor. |
| 13:33.41 | Makenshi | Manx, each 2k3 r2 license permits you to run up to 4 virtual instances |
| 13:33.45 | dongs | digium card is a tdm400, but since it only has 4 ports, you:ll need 3 of them |
| 13:33.53 | dongs | i dunno if digum has a higher density card or not |
| 13:33.54 | bernardovieira | hi all! |
| 13:33.54 | mvdk | Which, of course, from the small amount of time I have spent conversing with you, appears to be an issue which may cause you some consternation.... |
| 13:33.55 | ManxPower | Most of the WAN runs 384K frame relay, I somehow don't think virtulaztition is the answer |
| 13:34.06 | coppice | there are people in this channel from strange other worlds. there are people from places where G.711 is a lossless codec. there are people from places where G.729 sounds exactly the same as G.711. now there are people from some place where windows works properly :-\ |
| 13:34.18 | ManxPower | puzzled, finally an honest answer. |
| 13:34.24 | mvdk | My point precisely, coppice! |
| 13:34.26 | bernardovieira | Is there a way to adjust the sample rate on the g729 codec supplied by digium? |
| 13:34.35 | Makenshi | Manx, not sure what that has to do with it, but i'm just pointing out that you don't have to buy so many licenses |
| 13:34.41 | dongs | bernardovieira: why would you want to! |
| 13:34.51 | tRSS | hellloooo!!!??? ;) |
| 13:35.00 | coppice | bernardovieira: sample rate or bit rate? |
| 13:35.01 | puzzled | bernardovieira: ask Digium support but I don't think so |
| 13:35.11 | ManxPower | Makenshi, As I understand it "virtualzation" would be running multiple instances on one piece of hardware. |
| 13:35.17 | tRSS | how does ACD (automatic call distributor) work in asterisk? |
| 13:35.34 | mvdk | tRSS: have you read the manual? |
| 13:35.39 | bernardovieira | dongs: the voip provider I need to use works on a lower sample rate, so we get a lot of dropped packets... |
| 13:35.48 | mvdk | It works really well, honest..... |
| 13:35.56 | tRSS | my search for ACD turned nothing |
| 13:36.00 | puzzled | tRSS: www.voip-info.org, buy the Asterisk book, google... |
| 13:36.06 | Makenshi | Manx, yes, several independant operating systems.. this also allows for high availability using vmware esx, since it can migrate a live machine across different physical hosts |
| 13:36.18 | dongs | bernardovieira: impossible, all voip is like 8khz/16bit/mono, how could the use a different sample rate. |
| 13:36.24 | mvdk | Anyway, to answer your question, you're talking about queues |
| 13:36.26 | clive- | mvdk that lias thing never worked but rmmod did the trick, lets hope tht those isseswill improve now |
| 13:36.31 | puzzled | but vmware esx does not come cheap |
| 13:36.33 | tRSS | lol @ puzzled. so I get a cold shoulder for helping others here |
| 13:36.36 | mvdk | Go look at queues.conf and agents.conf |
| 13:36.45 | tRSS | thanks mvdk |
| 13:36.56 | Makenshi | no it isn't, but i'd say it's worth it.. |
| 13:36.57 | tRSS | that should be sufficient to start me off! :) |
| 13:36.58 | puzzled | tRSS: nope, I don't know but point out where you might find the info |
| 13:37.11 | mvdk | clive-: Ah, that means that centos must do it differently again.... |
| 13:37.23 | dongs | does digium make a higher density than 4 port analog card these days? |
| 13:37.28 | puzzled | yes |
| 13:37.29 | dongs | last time i cared their biggest one was tdm400 |
| 13:37.29 | mvdk | I've not had any troubles convincing my debian box to do that kind of thing.... |
| 13:37.43 | mvdk | dongs: They have for quite a while now.... |
| 13:37.43 | Makenshi | in our case we don't have much alternative since we don't have a lot of rackspace.. using vmware and blades lets us pack a lot more into the same space |
| 13:37.46 | dongs | what do they have now? |
| 13:37.52 | mvdk | dongs: look at the TDM2400 |
| 13:38.13 | mvdk | It has capacity for up to 24 channels |
| 13:38.26 | dongs | oshit |
| 13:38.28 | coppice | bernardovierira: what do you mean by sample rate? there are three possible parameters. the actual sample rate is fixed at 8000/s. The bit rate could be 6.4k, 8k, or 11k per second, but * and most other things only support 8k. The packet size is usually 20ms in *, but that one can be changed |
| 13:38.31 | dongs | winner. |
| 13:38.39 | mvdk | (It uses a centronix connector to a patch panel) |
| 13:38.45 | iq | hi |
| 13:38.53 | dongs | well then |
| 13:38.59 | dongs | eBody: looks like thats your answer. |
| 13:39.08 | dongs | so i guess you only need one card then, for up to 24 lines!. |
| 13:39.09 | eBody | thanks guys, u really helped me out! |
| 13:39.15 | mvdk | No probs.... |
| 13:39.30 | mvdk | Of course, if you need a PRI, they have that too |
| 13:39.31 | bernardovieira | coppice: sorry... my question was dumb... hehehe I meant packet size... how do you go about doing that? |
| 13:40.13 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 13:40.16 | mvdk | (Those 24 lines are either FXO or FXS, not PRI) |
| 13:40.42 | satlan32 | hello |
| 13:40.50 | dongs | yea i figured that much |
| 13:40.55 | dongs | thats still quite a deal |
| 13:40.59 | satlan32 | i want to know if there is a way to install g729b codec in asterisk? |
| 13:41.14 | dongs | satlan32: yeah, ftp a copy from digium.com and then buy a license at $10/ea |
| 13:41.26 | ManxPower | satlan32, instuctions are available when you purchase a G279 license from Digium |
| 13:41.46 | satlan32 | is there a free version for testing? or only money option? |
| 13:41.53 | mvdk | God, do people still read these days? |
| 13:42.10 | mvdk | Yeah, if you don't mind being sued for patent violations.... |
| 13:42.31 | satlan32 | ??? |
| 13:42.37 | dongs | hm |
| 13:42.42 | mvdk | So yes, there is a free implementation, but it has patent issues |
| 13:42.51 | dongs | lol |
| 13:42.54 | mvdk | This is because g.729 is a *patented* codec! |
| 13:42.54 | Makenshi | depends which country you're in whether you get sued or not (devils advocate) |
| 13:43.31 | mvdk | Yeah, I assume we're not talking about some African country... |
| 13:43.52 | satlan32 | nop |
| 13:44.03 | mvdk | But I must point out that $10/line is hardly a huge expense.... |
| 13:44.13 | mvdk | And it only applies if it does transcoding |
| 13:44.29 | Makenshi | yeah it is hardly expensive |
| 13:44.33 | mvdk | If both endpoints of the call are speaking g.729, then you don't need a license |
| 13:46.41 | *** join/#asterisk pif (n=ldm@ATuileries-152-1-57-200.w82-123.abo.wanadoo.fr) |
| 13:46.59 | pif | hi, any ST-2030 user? |
| 13:48.29 | mvdk | So then, dongs, you've never had any vexing windows issues, places where windows is a brain-dead idea? Because I know of quite a few.... |
| 13:49.29 | clive- | coppice hi, how sensitive is iax2 trunking to low-ish zttest scores? |
| 13:49.43 | mercestes | WIll Microsoft ever live down Windows ME? |
| 13:50.03 | bernardovieira | satlan32: you could try http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1 for testing, but I don't think that code has been visited in a long time, your best bet would be to buy g729 from digium |
| 13:50.20 | mvdk | mercestes: We will confine ourselves to *real* issues, without talking about the past.... |
| 13:50.24 | satlan32 | ok thanks |
| 13:50.45 | mercestes | mvdk: Regurgatating something you read off of a hacker forum is hardly a "real" issue. |
| 13:51.00 | mercestes | Win2k has been, in my experience, nothing short of drastically acceptable. |
| 13:51.12 | mercestes | It just took Microsoft for freaking ever to get there. |
| 13:51.18 | mvdk | Yeah, acceptable being key.... |
| 13:51.29 | mercestes | as opposed to....? |
| 13:51.34 | mvdk | Barely adequate is a better descriptor for some things.... |
| 13:52.15 | mercestes | Should try using win2k /win2k3 every now and then, perhaps....instead of bashign it off topic in an IRC forum. |
| 13:52.25 | mvdk | I do, mercestes |
| 13:52.27 | Makenshi | windows is now posix compliant too |
| 13:52.29 | mercestes | Just because every grandma and retard on the planet has problems with it doesn't mean it's a bad OS.... |
| 13:52.43 | mercestes | It just means it's user friendly enough to attract grandmas and retards. |
| 13:52.48 | *** join/#asterisk Godsey (i=jason@pdpc/supporter/sustaining/Godsey) |
| 13:53.01 | mercestes | Linux has the advantage of being user abusive enough to require an IQ of 125 to boot the damn thing up...much less use it... |
| 13:53.04 | eBody | are you limited to the number of extentions with the TDM400?? |
| 13:53.09 | mercestes | That compensates for alot of what you call...inadequacies. |
| 13:53.09 | mvdk | No, the reason it's a bad OS is more because of the fact that it doesn't have things like CUPS, asterisk, hylafax, and the like |
| 13:53.13 | Godsey | I'm trying to build from svn, when I try building libpri it says I need a newer zaptel, when I try building zaptel it says I need a newer libpri |
| 13:53.18 | Godsey | how do I escape this catch22? |
| 13:53.25 | mercestes | ...............now that is a retarded statement... |
| 13:53.30 | mvdk | Ah, and Samba, of course |
| 13:53.34 | mercestes | Windows sucks because it doesn't have linux programs on it. |
| 13:53.43 | mvdk | Well, in a nutshell, yes |
| 13:53.49 | mercestes | Thank you for.....steering me away from you before I got too indepth with this. |
| 13:53.58 | coppice | i wonder if MS will ever both to fix up chinese windows so all the controls are actually visible in the dialogs. its been like that for 10 years |
| 13:54.43 | mvdk | For someone like me, used to working in an environment where stuff Just Plain Works (tm), working in an environment where one has to take servers offline regularly is just pure shit |
| 13:54.50 | file | Godsey: you don't need zaptel to build libpri, only it's test programs and things |
| 13:55.17 | mvdk | Once I get a hylafax config working, I don't ever need to touch it again |
| 13:55.39 | mvdk | OTOH, Someone came to me with this Windows fax serving program |
| 13:55.40 | mercestes | Why would Windows need a program to implement the illusion of a windows domain btw? |
| 13:55.45 | Godsey | file: ok |
| 13:55.46 | mvdk | It was pure shite..... |
| 13:56.07 | mercestes | When windows itself supports the very domains that Samba is trying to mask? |
| 13:56.08 | mvdk | Just like so many damn other Windows "Server" applications.... |
| 13:56.26 | mvdk | But samba does more than that, mercestes |
| 13:56.42 | mercestes | Yea, I read that earlier, obviously it causes R/W tanking too. |
| 13:56.43 | mvdk | It allows bridging between various configurations, and all other kinds of things |
| 13:57.05 | mercestes | Last time I checked Windows domains had drivers and protocols for various windows platforms, Linux, Apple, Macintosh and DOS. |
| 13:57.21 | mvdk | mercestes, you might wish to confine yourself to real issues, instead of people that can't be bothered talking about what their actual situation is |
| 13:57.27 | ManxPower | eBody, No. You are limited to 4 lines, but you can have an unlimited number of extensions, since extensions have NOTHING to do with lines. |
| 13:57.43 | mvdk | Yes, mercestes |
| 13:57.44 | mercestes | See, I don't even know what yoru ranting about anymore, MVdk..... |
| 13:57.54 | mvdk | Nor do I, fancy that |
| 13:58.06 | mercestes | Not surprised. |
| 13:58.07 | mvdk | Let's return to civil conversation, shall we? |
| 13:58.14 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:58.29 | mercestes | I guess if I were to spout of indirect and cryptic "retorts" I'd get lost in my own arguements too..:P |
| 13:58.49 | dongs | eBody: why dont you summarize what you:re "trying" to do, and perhaps someone will have some sort of a solution/suggestion for your setup! |
| 13:58.51 | mvdk | mercestes: I will not dignify that with a response |
| 13:58.54 | ManxPower | "can't we all just get a bong?" |
| 13:59.01 | dongs | or even better, a dong! |
| 13:59.16 | mvdk | eBody: What do you need? |
| 13:59.26 | mvdk | (to do)? |
| 13:59.37 | eBody | i'm doing this this: we need more extensions for our office, and want to move to VoIP |
| 13:59.47 | dongs | extensions = phones on the desk? |
| 13:59.53 | eBody | we have 10 analog lines coming in to our PBX |
| 13:59.55 | eBody | yes |
| 13:59.56 | mvdk | OK, by extensions you mean phones on peoples' desks |
| 14:00.00 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 14:00.06 | eBody | we need 48 |
| 14:00.06 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
| 14:00.07 | dongs | are you using a PBX or a key system? |
| 14:00.09 | mvdk | There are multiple ways you can do that |
| 14:00.20 | ManxPower | eBody, then you do NOT want a TDM400P! |
| 14:00.24 | vader-- | do you guys know what the pin layout for the rj45 connector is |
| 14:00.26 | mvdk | I would recommend using SIP phones on the desks, if that's practicle |
| 14:00.29 | mvdk | *practical |
| 14:00.34 | eBody | yeah i was looking for the TDM2400 |
| 14:00.35 | vader-- | to go from the CPE to the T1/PRI line card? |
| 14:00.41 | Godsey | ok how about this error? :) make[1]: Entering directory `/usr/src/asterisk-svn/libpri/channels' |
| 14:00.42 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:00.44 | Godsey | gcc -c -o chan_zap.o -I../include -I.. -fPIC -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -O6 -march=i686 chan_zap.c |
| 14:00.47 | Godsey | chan_zap.c:75:2: #error "You need newer libpri" |
| 14:00.53 | mercestes | White Orange/ Orange /White blue/ Green/ White Green/ blue /White brown /brown |
| 14:00.54 | __chris | when rebooting cisco 7940s (required for quite a lot of config changes) they can sit on the 'configuring vlan' screen for upto a minute - is there a way to speed this up? |
| 14:01.05 | mvdk | Godsey: fairly obvious, really.... |
| 14:01.05 | mercestes | oh...typed too soon. |
| 14:01.05 | dongs | godsey, which part of "you need a newer libpri" is unclear? |
| 14:01.10 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:01.11 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 14:01.13 | Godsey | I'm building libpri! |
| 14:01.20 | mvdk | Godsey: Look for libpri....... |
| 14:01.25 | mvdk | No, you're not |
| 14:01.25 | file | Godsey: uh, you're building chan_zap |
| 14:01.40 | dongs | did you like, check out zaptel into same dir with libpri? |
| 14:01.48 | ManxPower | vader--, orange/white, white/orange, white/breen, blue/white, white/blue, green/white. white/brown, brown/white |
| 14:01.57 | Godsey | I don't think so, I'll rm and try agian |
| 14:01.58 | mvdk | Godsey: Get libpri, your brain will thank you.... |
| 14:02.01 | ManxPower | or you could just look it up on Google |
| 14:02.32 | *** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 14:02.35 | dongs | libpri is like |
| 14:02.40 | dongs | 10 source files last i checked |
| 14:02.46 | dongs | and none of htem are named chan_zap |
| 14:02.48 | mvdk | So, ebody, are SIP phones on the people's desks a practical solution to your problem? |
| 14:02.59 | Godsey | that must have been it |
| 14:03.07 | ManxPower | __chris, tell the phone not to use CDP |
| 14:03.18 | dongs | eBody: your current phones on desks, do tehy have line buttons for 10 incoming lines, or what? |
| 14:03.27 | dongs | eBody: because if thats the case, asterisk is not for you! |
| 14:03.28 | *** join/#asterisk hener (n=mitka@62.76.244.194) |
| 14:03.50 | mvdk | Well, it can be for you, it's just that you may wish to engage a consultant |
| 14:04.02 | dongs | untraining old\stupid people from key system into something else is almost impossible |
| 14:04.43 | ManxPower | feel free to engage a consultant, just don't marry one! |
| 14:04.44 | mvdk | What's so hard about "press 9 for an outside line?" |
| 14:04.44 | SplasPood | dongs: how is it that 10 line buttons on a phone is a problem for asterisk? |
| 14:04.59 | dongs | SplasPood: because asterisk doesnt do that/ |
| 14:05.02 | ManxPower | SplasPood, because there are no really affordable 10 line IP phones that work with Asterisk |
| 14:05.07 | SplasPood | Well |
| 14:05.10 | SplasPood | there's the Polycom 601 |
| 14:05.13 | ManxPower | The polycom 601 with the BLF would be about as close as you can get. |
| 14:05.16 | SplasPood | with the side car |
| 14:05.17 | SplasPood | yea.. |
| 14:05.20 | SplasPood | whats wrong with that? |
| 14:05.22 | [TK]D-Fender | ManxPower : IP601 + 1 Attendant Module? |
| 14:05.28 | ManxPower | And that's like $400 for each phone. |
| 14:05.31 | coppice | ManxPower: eh? why not marry someone with a high income, then take things easy? |
| 14:05.32 | mvdk | You might be using a new definition of "affordable", splaspood.... |
| 14:05.49 | eBody | if we need new phones then we could get them but they are regular analog phones |
| 14:05.52 | SplasPood | I don't understand why you'd need everyone to have one tho.. |
| 14:06.04 | [TK]D-Fender | Only a receptionist could need a 10 line phone if even. the entire concept of "lines" as buttons is dated |
| 14:06.08 | mvdk | OK, then, ebody |
| 14:06.09 | ManxPower | SplasPood, because most users are too stupid to learn "dial 9" |
| 14:06.16 | SplasPood | why does one need to dial 9? |
| 14:06.24 | eBody | so yeah, we're going to have to get new phones either way i think |
| 14:06.26 | mvdk | splaspood: To get an outside line |
| 14:06.38 | SplasPood | why would you need to do that? |
| 14:06.42 | hener | anyone hre knows russian |
| 14:06.42 | ManxPower | SplasPood, Sorry, I had forgotten you are one of those "you don't need to dial 9" people. We have nothing to discuss. |
| 14:06.44 | mvdk | eBody: I'd suggest getting IP phones, and sticking them on desks |
| 14:06.44 | JackEStorm | mvdk: you don't HAVE to set itup like that. |
| 14:06.55 | dongs | [TK]D-Fender: tell that to old people using the phones |
| 14:06.56 | mvdk | Indeed, you don't |
| 14:06.57 | SplasPood | ManxPower: No I wanna learn.. why would one want to do it that way? |
| 14:07.08 | hener | Does anynone here from RUSSSIA |
| 14:07.15 | mvdk | Well, because you may wish to have an internal extension system |
| 14:07.19 | eBody | that does sound good. what does Asterisk support phone wise |
| 14:07.23 | clive- | does anyone know how sensitive iax2 trunking is to a bad zttest score ? |
| 14:07.32 | SplasPood | mvdk: we do, and the dialplan handles it fine |
| 14:07.32 | mvdk | Any SIP phone, or IAXy |
| 14:07.38 | ManxPower | SplasPood, Here is an example dialplan without dialing 9: NXXXXXX and 1NXXNXXXXXX and NXXX |
| 14:07.45 | ManxPower | Do you see the problem with this? |
| 14:07.55 | SplasPood | who allows 7 digit dialing? |
| 14:07.58 | mvdk | Well, I find it easiest to say "dial 9 for an outside line" |
| 14:08.12 | ManxPower | SplasPood, most people do. |
| 14:08.15 | mvdk | The dial plan is far more difficult in Australia, see.... |
| 14:08.18 | SplasPood | and 4 digits |
| 14:08.21 | [TK]D-Fender | mvdk : No need for a 9 prefix.... |
| 14:08.23 | SplasPood | all my extens start with a given number |
| 14:08.27 | SplasPood | so I wouldn't do NXXX |
| 14:08.30 | mvdk | Oh, I see |
| 14:08.43 | ManxPower | SplasPood, what number? |
| 14:08.45 | mvdk | So you say "dial N for an internal line" :) |
| 14:08.48 | SplasPood | 2 |
| 14:08.51 | Godsey | somehow I checked out asterisk zaptel and libpri into libpri |
| 14:09.02 | *** join/#asterisk SanketMedhi (n=sanketme@221.128.138.120) |
| 14:09.08 | SanketMedhi | hello |
| 14:09.14 | ManxPower | SplasPood, ok, the issue still stands. |
| 14:09.15 | mercestes | Dial 9 is like the retard interface for Asterisk....telling them to "dial 9" for an outside line just gives it that analogue flavor that old people crave....like plain vanilla. |
| 14:09.33 | ManxPower | when you dial 2XX, how does Asterisk know you are not dialing 2XXXXXX |
| 14:09.35 | mvdk | Well, it *is* trivial to set up.... |
| 14:09.37 | SanketMedhi | I am facing a problem with SIP ATAs |
| 14:09.46 | SplasPood | ManxPower: Ok explain it to me, cause I don't seem to be having any issues.. people dial 1+NXX NXXXX for a domestic call, 011 + for international, and 2XXX for internal |
| 14:09.53 | SanketMedhi | here is the output of "show peers" on the asterisk CLI |
| 14:09.56 | mvdk | Particularly as the Australian dial plan is nowhere near that simple |
| 14:10.08 | SanketMedhi | http://pastebin.com/708507 |
| 14:10.12 | dongs | these fucking jap adapters i:m using, they:re hardcoded to only allow numbers that look so EVERYTHING dialed must start with 0 if yo uwant a variable length number, otherwise it jsut rejects the call |
| 14:10.18 | ManxPower | SplasPood, *nod* If you don't premit 7 or 10 digit dialing then you don't have this specific problem in the USA. |
| 14:10.20 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 14:10.21 | SplasPood | ManxPower: well I also have no analog channels |
| 14:10.28 | SplasPood | ManxPower: ok I hear ya then |
| 14:10.32 | ManxPower | SplasPood, this has nothing to do with analog. |
| 14:10.53 | SplasPood | well since the dialplan on the phone decides when to send the digits, it sorta does.. |
| 14:11.00 | mercestes | ManxPower: There is also the timeout variable from 1-3 seconds you can set...NxxxT for example. |
| 14:11.08 | ManxPower | SplasPood, my example can apply to extensions.conf or the dialplan on the phone. |
| 14:11.22 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 14:11.24 | ManxPower | mercestes, Um, my users REQUIRED my to allow 20 second digit timeout |
| 14:11.42 | *** join/#asterisk {Sean} (n=sean@c-67-177-80-24.hsd1.mi.comcast.net) |
| 14:11.46 | mercestes | ManxPower: Yea, there is that issue in which people cannot seem to dial 1 digit every six seconds. |
| 14:12.10 | SplasPood | ADAPT OR PERISH! |
| 14:12.11 | Godsey | mercestes dial 9 an analog flavor? |
| 14:12.12 | mvdk | Point is, saying "dial 9 for an outside line" is a trivial thing to do.... |
| 14:12.13 | [TK]D-Fender | ManxPower : My permanent Polycom dial-plan ---> X.T|#.T|*.T send *whatever* to Asterisk as-is and STFU :) |
| 14:12.24 | mercestes | ManxPower: I mean, with a dozen buttons to choose from....that only allows a half a second decision time per button. |
| 14:12.26 | JackEStorm | ManxPower: see that, I set that time out on the phone, and bedone with it. |
| 14:12.32 | SplasPood | mvdk: so is saying "Just dial as you normally would" if yer in the US.. |
| 14:12.44 | Godsey | [TK]D-Fender: so you don't use polycom pickup groups? |
| 14:13.05 | [TK]D-Fender | Godsey : Not sure what you mean exactly... |
| 14:13.06 | mvdk | Do no local numbers start with 2 there? |
| 14:13.09 | ManxPower | What can I say. My users are both morons and assholes. |
| 14:13.10 | mercestes | I don't like the polycom handsets.....they're too light to abuse users with. |
| 14:13.23 | SanketMedhi | Somebody please help me with this : http://pastebin.com/708516 |
| 14:13.26 | SplasPood | mvdk: I don't allow 7 digit dialing |
| 14:13.38 | SplasPood | ManxPower: it happens |
| 14:13.48 | mvdk | Ah, I see |
| 14:14.06 | mvdk | OK, to most people here, that would be considered highly unusual |
| 14:14.12 | [TK]D-Fender | SanketMedhi : Sounds like they're behind the same router.... |
| 14:14.20 | SanketMedhi | yes they are |
| 14:14.21 | mercestes | SanketMedhi: That's line 1 and line 2 of the same ATA. |
| 14:14.28 | mercestes | *points* Or as D-Fender said, behind the same router. |
| 14:14.30 | ManxPower | SanketMedhi, sip device 503 is not registering to Asterisk. |
| 14:14.31 | [TK]D-Fender | SanketMedhi : Thats why... thats the ROUTER's IP. |
| 14:14.32 | mvdk | That's mainly because the area code covers an entire state |
| 14:14.35 | ManxPower | SanketMedhi, other than that there is no problem. |
| 14:14.36 | mercestes | Is this presenting a specific problem. |
| 14:14.37 | SanketMedhi | mercestes, no they arent |
| 14:14.47 | [TK]D-Fender | SanketMedhi : and NOTHING * can do about that. |
| 14:15.06 | SanketMedhi | the sip registry is kinda freezed .. within asterisk |
| 14:15.18 | SplasPood | mvdk: where is here? |
| 14:15.19 | SanketMedhi | 501 --> this ATA has died .. but registration is still present |
| 14:15.27 | [TK]D-Fender | SanketMedhi : Turn on SIP debug and restart the ATA and prove it... |
| 14:15.44 | SanketMedhi | is there a way to clear sip registry within asterisk ? |
| 14:15.59 | [TK]D-Fender | SanketMedhi : And make sure you have "qualify=yes" for them too |
| 14:16.17 | SanketMedhi | ok lemme check that [TK]D-Fender |
| 14:16.26 | RoyK | SanketMedhi: no, there isn't |
| 14:16.34 | SanketMedhi | qualify=yes is in global |
| 14:16.35 | hener | heeloo |
| 14:16.46 | RoyK | SanketMedhi: but looking at the code, it shouldn't be too hard to create a console command for it |
| 14:16.49 | SanketMedhi | RoyK, ok |
| 14:17.03 | hener | royk ios really smart |
| 14:17.05 | SanketMedhi | :) |
| 14:17.07 | mvdk | splaspood: Australia |
| 14:17.13 | RoyK | hener: ios? |
| 14:17.20 | hener | yeap |
| 14:17.26 | RoyK | what about ios? |
| 14:17.28 | RoyK | cisco ios? |
| 14:17.42 | hener | u are smart u should know |
| 14:17.44 | hener | dont worry |
| 14:17.47 | hener | cool down |
| 14:17.50 | RoyK | SanketMedhi: email me about it and i'll look around to see if i can find some old code. i started writing that a few months ago..... |
| 14:17.54 | RoyK | ~lart hener |
| 14:18.05 | SanketMedhi | RoyK, ok ur email? |
| 14:18.13 | SplasPood | mvdk: ahh here (nyc, us) I don't think verizon even allows 10 digit dialing, let alone 7 |
| 14:18.20 | RoyK | roy@karlsbakk.net |
| 14:18.25 | hener | do u need my email too |
| 14:18.26 | hener | ok |
| 14:18.27 | hener | thanks |
| 14:18.30 | hener | i will email u |
| 14:18.36 | hener | karlsbakk |
| 14:18.42 | SanketMedhi | for what? |
| 14:18.54 | SanketMedhi | :) |
| 14:18.55 | *** join/#asterisk acehunky (n=chat_jok@221.128.138.120) |
| 14:18.56 | hener | to dicuss |
| 14:18.58 | hener | with royk |
| 14:19.02 | ManxPower | SplasPood, regardless of what people say, NYC is not the entire world. |
| 14:19.10 | hener | her er roy karlsbakks lille hjemmeside |
| 14:19.25 | SplasPood | ManxPower: Damn man, you're all kinds of pissed off seeming |
| 14:19.37 | hener | who understood that |
| 14:19.48 | hener | sipnet.ru |
| 14:19.53 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:19.56 | acehunky | is there a howto to create the console command ? |
| 14:20.12 | SplasPood | acehunky: learn C in 21 days? |
| 14:20.28 | puzzled | I have that book |
| 14:20.34 | puzzled | didn't work for me |
| 14:20.35 | SanketMedhi | lol |
| 14:20.40 | mercestes | ManxPower is exhibiting a healthy level of cynicism with regards to our chosen industry. |
| 14:20.41 | [TK]D-Fender | acehunky : you may want to rephrase that question.... a LOT... |
| 14:20.42 | hener | i have another learn in 5 days |
| 14:20.47 | mvdk | The idea of putting "02" at the start of every local number would be met by a great deal of opposition here, I daresay.... |
| 14:20.47 | hener | but i burn it |
| 14:21.13 | esculapio__ | hola |
| 14:21.21 | hener | holala |
| 14:21.37 | SplasPood | well anyway, work calls |
| 14:21.40 | hener | i goin to fuck asterisk someday |
| 14:21.54 | mvdk | hener: I don't think you'll find it a good screw.... |
| 14:21.56 | esculapio__ | hener, estoy buscando un software de facturacion (billing) |
| 14:22.14 | ManxPower | I admit that even I, if I'm not careful, have a USA-centric viewpoint of telecom. |
| 14:22.20 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 14:22.39 | mvdk | Yep, so "dial 9 for an outside line" is fairly normal in Australia |
| 14:22.52 | ManxPower | mvdk, I thought it was "0" |
| 14:23.00 | mvdk | Depends what system |
| 14:23.05 | mercestes | hener: I guess if you directed a 60 volt ring current out of a zap channel...*ponders*.....it could be done. |
| 14:23.16 | shmaltz | I'm running this in my dial plan:GotoIf($[${CHANNEL:0:5}=SIP\/0]?550) |
| 14:23.17 | mvdk | I've always used 9 |
| 14:23.17 | shmaltz | if ${CHANNEL} contains Zap/1-1 then all is good and it evaluates to false, if ${CHANNEL} contains Zap/11-1 then it evaluates to ture, what's wrong? |
| 14:23.20 | mercestes | hener: we should write a howto on that. AsteriskXXX. |
| 14:23.33 | hener | Her finnes sma saker og ting som kan vare av interesse |
| 14:23.35 | mvdk | So that I can map 000 to the emergency number |
| 14:23.42 | ManxPower | shmaltz, use Cut() |
| 14:24.05 | shmaltz | ManxPower, what do you mean? how will cut help? |
| 14:24.10 | hener | or use Paste() |
| 14:24.32 | mvdk | You see, 0011 is the access code for international here, normally |
| 14:24.39 | ManxPower | For example, did you know that in most of the world incoming calls to cell phones are FREE? Did you also know that in most of the world it costs more to call a cell phone than a land line. Also, did you know that in most of the world ALL outgoing calls are billed by the min, even local calls? |
| 14:24.40 | hener | after u cut u will need to paste |
| 14:24.57 | hener | but in russia |
| 14:25.04 | hener | incoming calls to cellphone |
| 14:25.06 | hener | are charged |
| 14:25.12 | mvdk | So 00011... has 000 at the front |
| 14:25.26 | ManxPower | exten => s,4,Cut(TECHNOLOGY=CHANNEL,/,1) |
| 14:25.26 | ManxPower | exten => s,5,GotoIf($[${TECHNOLOGY} = "Zap"]?9:6) |
| 14:25.29 | *** join/#asterisk Borgon (n=l3orgon@host-69-59-103-160.nctv.com) |
| 14:25.30 | Borgon | <PROTECTED> |
| 14:25.31 | dongs | ManxPower: lol, what are you talkin about. local calls are free in any country that isnt a pile of steaming hot feces |
| 14:25.37 | mvdk | Which means that someone picking up the line might expect 000 to actually reach the emergency services |
| 14:25.47 | hener | treu |
| 14:25.47 | Borgon | any reason why am getting a not found error on the agi script? i checked the path and its correct |
| 14:25.48 | hener | heheheheh |
| 14:25.52 | mercestes | dongs: Unless your a service provider......then local cost cost more than long distance calls. |
| 14:25.54 | mvdk | dongs: Ah, that would include Australia |
| 14:26.11 | hener | how abt russia |
| 14:26.18 | mercestes | Although I must say, when I worked for a service provider, we were charged it 1/6 second increments...not by the minute. |
| 14:26.20 | [TK]D-Fender | Borgon : Does * have RIGHTS to that file? |
| 14:26.22 | mvdk | Perhaps, dongs, you might consider taking courses in etiquette.... |
| 14:26.32 | hener | u should |
| 14:26.33 | hener | really |
| 14:26.40 | hener | i do recommend it dongs |
| 14:26.41 | mercestes | mvdk: That's probably the most hypocritical thing you've said today. |
| 14:26.45 | Borgon | [TK]D-Fender: yup i chmod 755 |
| 14:26.55 | mvdk | Why thank you, mercestes..... |
| 14:26.57 | ManxPower | These are some of the reasons dialup internet did not do well in most of the world, since it cost per min to be dialup. |
| 14:27.04 | mercestes | mvdk np..:) *hugs* |
| 14:27.13 | ManxPower | also the reasons cell phones are more popular in much of the world than in the usa |
| 14:27.33 | mvdk | mercestes: *blows kiss* :) |
| 14:27.39 | mercestes | Woohoo! |
| 14:27.59 | pjo | from voip-info does an entry in extension.conf like exten => _7XXX,1,Dial(IAX2/serverB/${EXTEN:1},30,r mean dial whatever digits at serverB minus the 1st digit or minus the last digit? |
| 14:28.10 | hener | YAHOOOOOOOOOOOO |
| 14:28.27 | dongs | pjo: yes. |
| 14:28.28 | mvdk | pjo: Did you not understand what they said? |
| 14:28.35 | mercestes | minus the first digit, Pjo. |
| 14:28.42 | ManxPower | pjo, minus first digit AND "provide ring tone even when you should provide some other tone" |
| 14:28.59 | ManxPower | pjo, See README.variables in /path/to/src/asterisk/docs |
| 14:29.04 | hener | U are the men |
| 14:29.12 | hener | ManxPower u rock |
| 14:29.25 | ManxPower | uh, what did I do? |
| 14:29.28 | mvdk | Perhaps, though, you would be well advised to read the asterisk docs.... |
| 14:29.34 | pjo | ManxPower: mercestes . thx. |
| 14:29.41 | mvdk | Manx: you handed him an answer on a silver platter, that's what |
| 14:29.56 | mvdk | We need to teach monkeys like him to read documentation..... |
| 14:30.03 | ManxPower | mvdk, Ah. Yes, I'm giving it away for free again! |
| 14:30.18 | hener | Yeah |
| 14:30.27 | ManxPower | Build a man a fire and keep him warm for a night, set a man on fire and keep him warm for the rest of his life. |
| 14:30.29 | hener | MAxPower is the men over here |
| 14:30.39 | hener | max dont burn anoyne |
| 14:30.45 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 14:30.46 | [TK]D-Fender | ManxPower : LOL... I'll have to remember that one :D |
| 14:30.55 | pjo | mvdk: i did read. i just didn't understand. but thanks anyhow. |
| 14:31.17 | mvdk | Oh, OK, then |
| 14:31.26 | hener | u all take care |
| 14:31.31 | mvdk | Well, I'm sorry for implying that you didn't |
| 14:31.40 | hener | take a break |
| 14:31.40 | hener | http://www.waytorussia.net/WhatIsRussia/Women/YoungWomen.html |
| 14:31.50 | hener | ccheck out russian girls |
| 14:31.52 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 14:31.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:32.00 | hener | how is the suggestion |
| 14:32.25 | Borgon | [TK]D-Fender: problem was php was installed in another path |
| 14:32.27 | __chris | ManxPower - Am looking to disable CDP to try and get rid of the 'configuring vlan' error - the only commands / docs I can see are for when using Callmanager though - any ideas on how to do this using the sip firmware? |
| 14:32.52 | hener | ask ManxPower |
| 14:32.55 | hener | he knows alot |
| 14:33.00 | myiagy | some calls on my asterisk are dropping, debug tells me: "Didn't get a frame from channel: SIP/xxxx" |
| 14:33.16 | hener | then frame it |
| 14:33.17 | hener | heeh |
| 14:33.20 | mercestes | __chris: In polycom it's just....edit CDP and set to disable...what phone are you trying to set it in? |
| 14:33.28 | myiagy | hener what do you mean? |
| 14:33.29 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 14:33.29 | __chris | cisco 7940 |
| 14:33.35 | *** join/#asterisk jsolares (n=jsolares@125.209.191.2) |
| 14:34.18 | mercestes | __chris: I don't remember if there is a setting in the bootmenu for it or not..might wanna grep cdp in the config files. I seem to remember a vlan=no from back when I was using them tho |
| 14:34.22 | hener | well the first step is the using the firmware |
| 14:34.30 | hener | to frame sip protocol |
| 14:34.49 | hener | did u get it |
| 14:35.07 | mountainm2k | Grandstream / SIP -- anybody know how I can make it produce a second dialtone after "9" (for outside line)? |
| 14:35.08 | myiagy | not really, what firmware? |
| 14:35.15 | myiagy | of the phone dialing? |
| 14:35.15 | ManxPower | __chris, no. I don't use Cisco phones because they end up being much more expensive than polycoms |
| 14:35.21 | hener | the sip ...asterisk build in code |
| 14:35.24 | hener | zaptel |
| 14:35.41 | mercestes | myiagy: It sounds like a networking issue...just...ignore hener...he's a weiner.....hey...I'm a poet and I was even aware of it. |
| 14:35.55 | *** join/#asterisk ghenry (n=ghenry@81-174-212-187.pth-as5.dial.plus.net) |
| 14:36.00 | hener | then move on with |
| 14:36.00 | mountainm2k | Grandstream GXP2000 1.1.0.13 |
| 14:36.15 | hener | http://pastebin.com/708516 |
| 14:36.19 | hener | check it out |
| 14:36.23 | hener | ignore mercestes |
| 14:36.28 | hener | he is just jealous |
| 14:36.32 | myiagy | :P |
| 14:36.43 | hener | he likes to boost around |
| 14:36.46 | shmaltz | ManxPower I instead did: |
| 14:36.48 | shmaltz | GotoIf($[${CHANNEL:0:3}=Zap]?550) |
| 14:36.49 | shmaltz | GotoIf($[${CHANNEL:0:3}=SIP]?560) |
| 14:36.51 | shmaltz | Do you see a prolbem with that? it works now as inteded (or so I think). |
| 14:36.52 | mercestes | boast around...not boost. |
| 14:36.53 | drray | how do I blow out asterisk and reinstall everything from scratch? |
| 14:37.01 | mercestes | drray: rm -dvfr |
| 14:37.05 | hener | ask manxpower |
| 14:37.07 | hener | he knows |
| 14:37.08 | drray | /etc/asterisk? |
| 14:37.09 | mercestes | drray: from / preferably |
| 14:37.18 | ManxPower | shmaltz, that won't work for channel names that are not 3 chars, but it should work for you |
| 14:37.21 | hener | mercestes doesnt know .he just pretends |
| 14:37.30 | hener | fuck u mercestes |
| 14:37.36 | shmaltz | ManxPower, what channel names are not 3 char? |
| 14:37.38 | mercestes | drray: If you are in Gentoo an emerge -Ca would work. |
| 14:37.49 | ManxPower | MGCP, H323, SKINNY, SCCP |
| 14:37.51 | hener | it doesnt |
| 14:37.56 | ManxPower | pretty much ALL of them except SIP and ZAP |
| 14:37.59 | hener | dont lie mercestes |
| 14:38.01 | shmaltz | I'm using only SIP, and Zap so I know it will work for me |
| 14:38.02 | shmaltz | thank you |
| 14:38.03 | mercestes | drray: Umm....RPM has an uninstall syntax I don't remember.... |
| 14:38.09 | mercestes | drray: other than that I'd just make over the top of it. |
| 14:38.12 | dongs | rpm -e lol |
| 14:38.16 | hener | u are idiot u dont remember |
| 14:38.30 | hener | <mercestes> drray: Umm....RPM has an uninstall syntax I don't remember.... |
| 14:38.30 | hener | <mercestes> drray: other than that I'd just make over the top of it. |
| 14:38.37 | mercestes | hener is just mad because I made him rm -dvfr / once and it took him weeks to get back on IRC. |
| 14:38.40 | Godsey | rpm -e |
| 14:38.41 | drray | I made over an old version of asterisk, and a lot of modules were bogus |
| 14:38.51 | dongs | Godsey: welcome to 30 seconds ago |
| 14:39.06 | hener | <shmaltz> ManxPower I instead did: |
| 14:39.06 | hener | <shmaltz> GotoIf($[${CHANNEL:0:3}=Zap]?550) |
| 14:39.06 | hener | <shmaltz> GotoIf($[${CHANNEL:0:3}=SIP]?560) |
| 14:39.12 | mercestes | drray: ahh...could try rmdir the /user/source directories. Asterisk, zaptel, libpri, whatever, adn then redownloading. |
| 14:39.14 | hener | yeah mercestes |
| 14:39.17 | hener | shut up |
| 14:39.27 | hener | dont pretend to know everything |
| 14:39.31 | shmaltz | hener, yes? |
| 14:39.32 | hener | u are not GOD |
| 14:39.32 | mercestes | oh, Hener, don't cry...it was all in good fun. |
| 14:39.33 | drray | I did that as well, but my configs stayed |
| 14:39.36 | hener | hahah |
| 14:39.38 | hener | thank you |
| 14:39.42 | mercestes | drray: configs are in /etc/asterisk. |
| 14:39.59 | mercestes | drray: So I'd blast /etc/asterisk too just make samples when your done if you need a blueprint to work from later. |
| 14:40.22 | hener | yeah ..just keep on lying to him |
| 14:40.31 | hener | mercestes dont mislead ppl |
| 14:40.33 | drray | well, I was thinking of blasting /etc/asterisk and then copying zaptel, zapata, sip, iax, extension, and manager back in |
| 14:41.29 | mercestes | drray: Could try that or go for a fresh install.....I would attemp tthe fresh install...what get method did you use? CVS? |
| 14:41.32 | *** join/#asterisk adker (n=adker@74-33-195-209.br1.glv.ny.frontiernet.net) |
| 14:41.39 | drray | svn |
| 14:41.54 | hener | ehehe ..thats a great suggestion |
| 14:41.56 | hener | reinsstall |
| 14:41.58 | drray | my old version was a year ago |
| 14:42.08 | hener | mercestes...great suggestion |
| 14:42.10 | hener | any more |
| 14:42.13 | hener | bright idea |
| 14:42.15 | mercestes | drray: *sighs* I miss the days when the most technical expertise I needed was choosing between "install.exe" and "setup.exe." |
| 14:42.16 | hener | ideas |
| 14:42.29 | hener | how old are u now |
| 14:42.30 | hener | 100 |
| 14:42.44 | hener | those days....was it 100 years back |
| 14:43.01 | hener | mercestes....pls continue lyinh |
| 14:43.04 | hener | lyin |
| 14:43.15 | mercestes | drray: Yea, could try rming /etc/asterisk .....or , barring that, rm /etc/asterisk and rm /usr/src/asterisk ../zaptel ../libpri ../asterisk-addons ../asterisk-sounds you can't copy/paste that btw. |
| 14:43.17 | hener | if u dont know just shut up |
| 14:43.27 | mercestes | it's PSEUDOCODE....*makes cryptic hand wiggly motions* |
| 14:43.47 | mercestes | anyways..GTG socialize with customers... bbl. |
| 14:44.00 | mercestes | Hener: I hope they find a pill forwhatever you have...and when they do..ask yoru doctor if it is right for you. |
| 14:44.08 | hener | mercestes make ur balls ....wiggly motions |
| 14:44.19 | hener | shake it baby |
| 14:44.21 | mercestes | on second thought, Hener...don't ask..just take six a day and pray. |
| 14:45.33 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 14:47.27 | X-Gen | how can u see if your disk is a bottleneck ? |
| 14:47.44 | hener | open up with ur balls |
| 14:47.45 | hener | !! |
| 14:47.49 | *** join/#asterisk twilson (n=terry@69.17.122.227) |
| 14:47.56 | X-Gen | run hdparam while the system is very busy ? |
| 14:48.06 | RoyK | X-Gen: lots of processes in D state |
| 14:48.14 | RoyK | X-Gen: sar telling you lots of waiting for i/o |
| 14:48.18 | RoyK | X-Gen: stuff like that |
| 14:48.20 | mountainm2k | X-Gen: Take a look at "top", and look for processes in state "D" |
| 14:48.30 | X-Gen | RoyK, sweet, thanks, just what i wanted to know |
| 14:48.32 | hener | X-gen ....if not c then try f |
| 14:48.33 | X-Gen | ta mountainm2k |
| 14:48.47 | hener | X-gen try shankin |
| 14:48.51 | RoyK | X-Gen: setup sysstat/sar to run every 10 minutes or so, and just type 'sar' to get the info since 00:000 |
| 14:48.55 | RoyK | X-Gen: setup sysstat/sar to run every 10 minutes or so, and just type 'sar' to get the info since 00:00:00 |
| 14:48.56 | mountainm2k | X-Gen: or just look at the top of the display there, and depending on OS / version, it'll show you % IO Wait |
| 14:48.57 | RoyK | 00000000000 |
| 14:49.14 | Greek-Boy | any1 here got a service agreement with cisco? Please download cisco firmware for me. lol |
| 14:49.15 | RoyK | mountainm2k: top is clumsy compared to sysstat |
| 14:49.23 | RoyK | ~lart greed |
| 14:49.27 | hener | <PROTECTED> |
| 14:49.27 | RoyK | ~lart Greek-Boy |
| 14:49.40 | hener | l |
| 14:49.40 | hener | < |
| 14:49.47 | hener | jbot beats Greek-Boy over the head with a microkernel |
| 14:50.17 | hener | thanks men |
| 14:50.22 | hener | i needed that |
| 14:50.26 | Hmmhesays | anyone ever deal with an adtran atlas 550? |
| 14:50.43 | hener | * jbot fucks royK, courtesy of Helen |
| 14:51.10 | *** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 14:51.15 | hener | jbot sucks RoyK balls |
| 14:51.30 | RoyK | jbot: hener-- |
| 14:51.38 | Hmmhesays | can feel the lurve in here today |
| 14:51.48 | hener | jbot shakes RoyK dick |
| 14:52.01 | hener | shake it baby |
| 14:52.02 | Hmmhesays | ~8ball |
| 14:52.03 | jbot | ACTION rolls the eight ball and gets: Outlook not so good |
| 14:52.10 | puzzled | hehe |
| 14:52.14 | hener | hehehe |
| 14:52.27 | hener | jbot licks Royk |
| 14:52.32 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 14:52.32 | Hmmhesays | ~8ball did hener just accidently cut off his wang with a pair of safetly scissors? |
| 14:52.34 | jbot | Please ask again. |
| 14:52.41 | Seyr | is there any way to detect disconnect with SIP? |
| 14:52.43 | RoyK | ~hener? |
| 14:52.50 | hener | and RoyK licks again |
| 14:52.52 | Hmmhesays | ~8ball did hener just accidently cut off his wang with a pair of safetly scissorss? |
| 14:52.53 | jbot | Are you smoking crack? |
| 14:53.01 | Hmmhesays | ~8ball am I? |
| 14:53.02 | jbot | No. |
| 14:53.07 | [TK]D-Fender | :D |
| 14:53.15 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 14:53.26 | hener | who are u jbot.... |
| 14:53.32 | vader-- | hmmm |
| 14:53.36 | RoyK | ~hener? |
| 14:53.38 | jbot | from memory, hener is just a freak newbe that never learns |
| 14:53.38 | hener | the great one...i respect u |
| 14:53.56 | [TK]D-Fender | ~8ball is jbot useful? |
| 14:53.57 | jbot | Are you smoking crack? |
| 14:53.58 | hener | sorry...jbot |
| 14:54.00 | vader-- | im just took the loopback plug out of my t1 line and plugged in my cpe and now im getting a red flashing light |
| 14:54.02 | hener | i listen to always |
| 14:54.02 | Hmmhesays | ~hmmhesays |
| 14:54.13 | hener | what u want me to do jbot |
| 14:54.14 | vader-- | is there a command that will tell me whats going on with the pri card? |
| 14:54.16 | hener | just tell me |
| 14:54.26 | Hmmhesays | how do you add entries to jbot? |
| 14:54.36 | ambriento | vader--, what about zttool |
| 14:54.37 | RoyK | vader--: have you tried HELP!? |
| 14:54.50 | hener | jbot talk to me again |
| 14:54.52 | hener | pls |
| 14:55.08 | Hmmhesays | I find this mildly amusing |
| 14:55.18 | RoyK | lol |
| 14:55.40 | ambriento | ~me? |
| 14:55.52 | Hmmhesays | ~cockpunch |
| 14:56.07 | RoyK | jbot: tell hener about jbot |
| 14:56.12 | vader-- | ambriento what am i looking for in zttool |
| 14:56.20 | vader-- | i have a red alarm on my TE110P T1/E1 card |
| 14:56.22 | hener | where are u jbot... |
| 14:56.56 | RoyK | vader--: that means no cables plugged |
| 14:57.01 | hener | do not ignore jbot |
| 14:57.05 | vader-- | hmmm but it is |
| 14:57.07 | RoyK | vader--: try a loopback plug or even a crossed cable |
| 14:57.09 | hener | jbot is the master over here |
| 14:57.19 | satlan32 | i guys |
| 14:57.28 | vader-- | on both ends i put in a standard RJ45 pinned out end |
| 14:57.33 | hener | satlan32 respect jbot |
| 14:57.36 | satlan32 | i have problems transfering DTMF's through asterisk |
| 14:57.36 | vader-- | is that correct for this setup? |
| 14:57.38 | InfraRed | stop abusing the bot |
| 14:57.46 | hener | he is the master here |
| 14:57.59 | puzzled | bow for thy bot |
| 14:58.06 | hener | jbot tell them all |
| 14:58.16 | RoyK | vader--: you need a crossover to speak asterisk <-> asterisk. or what do you try to do? |
| 14:58.31 | hener | vader ask jbot |
| 14:58.33 | satlan32 | can anyone tell me if i can configure the DTMF parameters myself? |
| 14:58.34 | hener | he will help u |
| 14:58.47 | vader-- | im trying to go from my CPE to my T1 card in my asterisk box |
| 14:59.02 | InfraRed | then the cable wiring is wrong |
| 14:59.11 | vader-- | how should it be? |
| 14:59.30 | InfraRed | no idea! |
| 14:59.32 | InfraRed | google? |
| 14:59.38 | hener | ASK JBOT!!!!! |
| 14:59.45 | satlan32 | HOW??? |
| 14:59.46 | RoyK | hener: stfu |
| 14:59.53 | hener | AK JBOT..HE KNOWS |
| 15:00.04 | ids2500 | vader cpe to t1 card should be a straight thru |
| 15:00.07 | hener | he knows all |
| 15:00.14 | vader-- | ya thats what i have |
| 15:00.18 | ids2500 | uh, i take it back |
| 15:00.19 | ids2500 | sorry |
| 15:00.23 | ids2500 | use a t1 crossover instead |
| 15:00.28 | ids2500 | pins 1 and 2 go to pins 4 and 5 |
| 15:00.42 | hener | ids2500 tryin askin JBOT!!! |
| 15:00.52 | ids2500 | google "t1 crossover" to be sure |
| 15:00.55 | ids2500 | but i am 99% on that |
| 15:01.00 | RoyK | hener: have you forgotten to take your medication? |
| 15:01.05 | ids2500 | lol @ royk |
| 15:01.10 | RoyK | ids2500: it's 1,2-4,5 |
| 15:01.11 | RoyK | beleive me |
| 15:01.14 | RoyK | i just made one :) |
| 15:01.21 | *** join/#asterisk umay (n=chris@71-208-188-148.hlrn.qwest.net) |
| 15:01.27 | ids2500 | okay, i thought so |
| 15:01.29 | hener | royk give ur balls ..might be useful |
| 15:01.35 | hener | for him |
| 15:01.47 | hener | u are jsut jealous of jbot |
| 15:01.52 | hener | arent u |
| 15:02.03 | ambriento | wait |
| 15:02.05 | RoyK | hener: more valium for you, perhaps |
| 15:02.14 | hener | and more sperm for u |
| 15:02.18 | vader-- | so i flip the orange and blue |
| 15:02.25 | hener | suck it up baby |
| 15:02.35 | ambriento | vader--, what do you have? a straitgh cable with RJ45 ends? |
| 15:02.36 | hener | royK suck it up baby |
| 15:02.41 | vader-- | ya |
| 15:02.45 | RoyK | hener: will you please stfu? you know, we're trying to talk about telephony here |
| 15:02.52 | RoyK | some of us |
| 15:03.08 | ambriento | I think its correct vader-- |
| 15:03.13 | ambriento | BUT |
| 15:03.15 | hener | RoyK u are just pretending,....u just dont know abt shit...try askin jbot |
| 15:03.26 | RoyK | any OPs around? |
| 15:03.44 | ambriento | hener, would you please take it easy a little?kthx |
| 15:03.53 | hener | ok |
| 15:03.59 | hener | i will listen to u |
| 15:04.03 | ambriento | ty |
| 15:04.07 | *** join/#asterisk pengyong (n=lala@218.93.158.125) |
| 15:04.08 | hener | pls continue ur discussion |
| 15:04.10 | vader-- | so one side 1|white/orange, 2|orange, 4|blue, 5|white/blue |
| 15:04.21 | hener | u are kind |
| 15:04.29 | hener | not like royK |
| 15:04.32 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 15:04.38 | vader-- | the other side 1|blue, 2|white/blue, 4|white/orange, 5|orange |
| 15:05.02 | RoyK | vader--: it's critical not to use them green wires |
| 15:05.03 | RoyK | lol |
| 15:05.05 | RoyK | :) |
| 15:05.07 | vader-- | hehe |
| 15:05.17 | vader-- | does that look right though? |
| 15:05.21 | RoyK | indeed |
| 15:05.22 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-119-182.phlapa.east.verizon.net) |
| 15:05.26 | ids2500 | yes |
| 15:05.27 | vader-- | ok cool |
| 15:05.27 | ids2500 | looks good |
| 15:05.39 | coppice | RoyK: but plenty of greens is good for you |
| 15:05.41 | Strom_C | vader--: are you making a T1 cable? |
| 15:06.03 | ambriento | that doesn't look like a straight cable tome vader |
| 15:06.18 | McLazarus | hi. Anyone here do any testing/work with the t38passthrough stuff out of svn? |
| 15:06.34 | RoyK | McLazarus: setting it up this moment |
| 15:06.48 | RoyK | coppice: is there a way to determine an inbound call is a fax call? |
| 15:07.03 | coppice | t38 passthrough is sooo passe. we're perfecting T.38 termination right now :-) |
| 15:07.10 | zoa | roy, any updates for your one way jitter buffer ? |
| 15:07.12 | McLazarus | RoyK: cool. I have it set up, but I am getting a problem that I have seen reported a few times. |
| 15:07.17 | RoyK | well |
| 15:07.21 | RoyK | i'm not done setting it up :) |
| 15:07.23 | McLazarus | but I can't seem to track down the solution. |
| 15:07.33 | RoyK | McLazarus: ask coppice. he knows all about it |
| 15:07.40 | McLazarus | :) |
| 15:07.59 | *** part/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 15:08.28 | hener | i told u ROyK doesnt KNOW A SHIT |
| 15:08.45 | dongs | you can have your $0 back |
| 15:09.11 | ids2500 | school is obviously out for the summer |
| 15:09.15 | ids2500 | and the kiddies are playing on IRC (: |
| 15:09.17 | ids2500 | :( |
| 15:09.22 | McLazarus | well the problem is the: "Jun 14 10:48:03 WARNING[3621]: chan_sip.c:4368 process_sdp: Unknown SDP media type in offer: image 16398 udptl t38" |
| 15:09.29 | ids2500 | McLazarus |
| 15:09.33 | ids2500 | what equipment are you using? |
| 15:09.34 | McLazarus | I see other people reported it, but don't seem to see the response. |
| 15:09.36 | ids2500 | i get the same thing :( |
| 15:09.54 | McLazarus | spa2100 is the ATA and Vega 400 to do the gateway stuff |
| 15:10.04 | ids2500 | i am also using spa2100 |
| 15:10.07 | ids2500 | but lucent apx as term gateway |
| 15:10.21 | McLazarus | I was passing stuff through the 1.2.4 patched version, but I was having reliability problems |
| 15:10.31 | McLazarus | aka the faxes would come through garbled occasionally |
| 15:10.34 | oej | McLazarus: I would like to see a SIP debug of that session |
| 15:10.48 | ids2500 | 1.2.4-patch crashed my lucent apx :( |
| 15:10.51 | McLazarus | oej: sure, should I post it to the bug or email or something? |
| 15:11.05 | ids2500 | oej: you saw the capture I posted in 5090, right? |
| 15:11.09 | ids2500 | with the same issue? |
| 15:11.11 | ambriento | vader--, the cable you just describled looks like crossed to me |
| 15:11.48 | ambriento | guys, lunch time |
| 15:11.49 | ambriento | bbl |
| 15:11.54 | oej | McLazarus: yeah, something |
| 15:12.00 | oej | 5090? Will check |
| 15:12.14 | McLazarus | oej: :) ok |
| 15:12.17 | hener | 5080 |
| 15:12.39 | oej | ids2500: Which file is it? |
| 15:12.49 | ids2500 | one sec, loading 5090 |
| 15:13.04 | ids2500 | http://bugs.digium.com/file_download.php?file_id=10522&type=bug (bad capture 06 08 2006.txt) |
| 15:14.26 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 15:14.39 | oej | ids2500: Cool, thanks for telling me |
| 15:15.37 | hener | gtg... ROyK SHUT UP....SHAKE UR DICK....SPERM it |
| 15:15.41 | *** part/#asterisk hener (n=mitka@62.76.244.194) |
| 15:15.47 | dongs | lol. |
| 15:15.48 | coppice | ids2500: that lucent box sounds really robust :-) |
| 15:15.53 | dongs | another satisfied customer |
| 15:17.31 | RoyK | oej: do we really want scum like hener in here? |
| 15:17.56 | coppice | RoyK: they come free with every IRC channel |
| 15:17.57 | Ahrimanes | no? |
| 15:18.04 | ids2500 | coppice: yeah... :( |
| 15:19.48 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:21.09 | mutilator | yesssssss, i have direct access to verizons ordering and trouble ticket system |
| 15:21.51 | vader-- | hmm i crossed them over |
| 15:21.52 | vader-- | and still nothing |
| 15:22.08 | Strom_C | vader--: you're making a T1 cable? |
| 15:22.11 | vader-- | ya |
| 15:22.22 | Strom_C | one moment; I think I've got some diagrams handy |
| 15:22.23 | mutilator | man that new google building is huge |
| 15:23.00 | dongs | ~cables |
| 15:23.02 | jbot | somebody said cables was http://www.jaredsmith.net/misc/cables/ |
| 15:23.13 | *** join/#asterisk batphone (n=bugz@cpe-70-123-122-41.houston.res.rr.com) |
| 15:23.32 | batphone | how important is it to have an asterisk box firewalled? |
| 15:23.35 | dongs | i think that has a t! crossover |
| 15:23.43 | dongs | batphone: it isnt. infact, doing so is quite retarded. |
| 15:24.01 | *** join/#asterisk salviadud (n=ralfalfa@201.133.207.93) |
| 15:24.23 | batphone | dongs: so you should let taiwanese windows zombies brute force ssh 388 times per second 24x7? |
| 15:24.56 | batphone | i'll make a note of that for my CISSP... |
| 15:25.03 | ids2500 | change the ssh port |
| 15:25.15 | batphone | that aint happenning... as much as id like it to |
| 15:25.25 | ids2500 | guess that's your problem then :shrug: |
| 15:25.59 | *** join/#asterisk fnordian (i=fnord@spaceboyz.net) |
| 15:26.01 | fnordian | hi |
| 15:26.21 | dongs | batphone: well, then move ssh to a different port. |
| 15:26.22 | fnordian | is there a h.323 channel, that uses libpri for building q931-packets? |
| 15:26.55 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 15:27.19 | satlan32 | need help with dtmfs |
| 15:27.32 | dongs | satlan32: dtmf=rfcwhatevertahtnumberis |
| 15:27.43 | dongs | satlan32: also inband dtmf doesnt go across compressed channels. |
| 15:27.52 | *** join/#asterisk BertZ (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr) |
| 15:27.55 | satlan32 | ??? |
| 15:27.56 | BertZ | hello there |
| 15:28.08 | satlan32 | i want to use rfc2833 |
| 15:28.09 | BertZ | I would like to understand a thing about G723 codecs |
| 15:28.15 | batphone | another issue is that a given pbx might serve as a gateway/firewall for a lan or vpn endpoint |
| 15:28.32 | dongs | satlan32: ok, then use it |
| 15:28.35 | satlan32 | but for some reason my system get the dtmfs but can't recognize them |
| 15:28.36 | BertZ | what mean passthrough ?? I want to use Asterisk as a voice server to handle incoming calls |
| 15:28.46 | satlan32 | i see the packets in the ethereal capture |
| 15:29.17 | coppice | BertZ: passthrough means exactly what it says. the audio simply passes through *, without being processed. |
| 15:29.19 | BertZ | can I use this codec ? I mean I want to call someone through a Sip trunk. ca n I use G.723 ? |
| 15:29.21 | batphone | dtmfmode=rfc2833 |
| 15:29.24 | BertZ | okay |
| 15:29.26 | BertZ | perfect :) |
| 15:29.30 | BertZ | thx |
| 15:29.42 | BertZ | I just want to sue it no to do any transcoding |
| 15:29.44 | coppice | BertZ: if you want to call from the PSTN, then no |
| 15:29.44 | BertZ | use |
| 15:29.48 | BertZ | no |
| 15:30.00 | BertZ | I want to call from my sipphone, through my SoftSwitch |
| 15:30.15 | coppice | throught the soft-switch to where? |
| 15:30.19 | BertZ | well |
| 15:30.30 | BertZ | we have our own VoIP network |
| 15:30.41 | batphone | dongs: ids2500: i run snort on some of them to prevent access from IP's that do this |
| 15:30.45 | BertZ | handled with Nextone Softswitch |
| 15:30.50 | batphone | to detect portscans and issue iptables commands |
| 15:31.02 | batphone | but the problem is overhead on the cpu for that, especially on some of the busier machines |
| 15:31.10 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:31.16 | coppice | if is a G.723.1 capable phone talking to a G.723.1 capable box through * it should work OK |
| 15:31.41 | dongs | batphone: shhrug, unless taiwan zombies are causing you to have 100x loads, why even care |
| 15:31.42 | batphone | the portscans always precede the brute force attempts, usually from a different ip |
| 15:31.46 | InfraRed | you probably need a license |
| 15:32.02 | InfraRed | coppice: but that depends on where the call is going |
| 15:32.27 | mutilator | http://video.google.com/videoplay?docid=-728262218491668100 |
| 15:32.29 | mutilator | ^ teh awesome |
| 15:32.37 | coppice | I think I already said that |
| 15:32.37 | batphone | one of these days shit is gonna hit the fan with * security |
| 15:32.44 | batphone | and im gonna be there with a firewall |
| 15:33.00 | coppice | i thought it did yesterday |
| 15:33.30 | smackus | has anyone use DRUID Asterisk Management Interface? I am interested to hear what people think of the product. |
| 15:33.33 | Seyr | whats the best opensource billing solution for Asterisk? |
| 15:33.53 | coppice | Seyr: pencil and paper |
| 15:34.01 | Greek-Boy | can [globals] be used in sip.conf? |
| 15:34.03 | dongs | batphone: shit has already hit the fan plenty of time,s but good luck punching holes in ytour "firewall" for all the udp/rtp/otehr crap needed for voip |
| 15:34.05 | InfraRed | Seyr: look at voip-info and try them out |
| 15:34.16 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 15:34.20 | smackus | Seyr: if you have php/mysql experience, I have liked just using the CDR database and a simple php web page to query |
| 15:34.28 | InfraRed | best is very subjective |
| 15:35.31 | RoyK | except with editors, where vim is the best editor anyway |
| 15:35.39 | dongs | agreed |
| 15:36.06 | dongs | any "admiN" using pico should be fired on the spot |
| 15:36.10 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 15:36.36 | RoyK | s/fired/shot/ |
| 15:36.49 | batphone | dongs: i dont have any problem administering the firewalls. its other people that do. so they shut them down and the box gets hit all week while im out of the office |
| 15:36.59 | *** join/#asterisk smokes (i=SMOKEY@72.53.123.84) |
| 15:37.32 | batphone | its just that i found a box that someone was brute forcing known accounts with. |
| 15:37.41 | dongs | old news |
| 15:37.45 | dongs | shit happens daily |
| 15:37.47 | dongs | just ignore it |
| 15:37.58 | Juggie | changing the ssh port is just way easier |
| 15:37.59 | dongs | none of your enabled accounts hould have any kidn of bruteforceable passwords |
| 15:38.02 | dongs | that too |
| 15:38.02 | Juggie | removes 95% of that crap |
| 15:38.07 | RoyK | batphone: that happens every day |
| 15:38.12 | Juggie | move 22->2222 |
| 15:38.13 | Juggie | problem solved. |
| 15:38.23 | LoRez | or install sshwatch |
| 15:38.39 | Juggie | and if anyone cant figure out how to change the ssh port in their client they shoudnt be sshing anyway |
| 15:38.51 | *** join/#asterisk Jon335 (i=Jon335@unaffiliated/jon335) |
| 15:38.58 | batphone | Juggie: im with you on that |
| 15:39.09 | batphone | the problem with changing the port is one of policy |
| 15:39.23 | Juggie | why? |
| 15:39.27 | dongs | ?? |
| 15:39.29 | batphone | i cant get around it, ive tried for 8 months to get them to allow me to do that |
| 15:39.30 | dongs | policy of what |
| 15:39.41 | mountainm2k | anybody know about "Early Dial" ? |
| 15:39.41 | batphone | frankly, the policy of laziness |
| 15:39.52 | salviadud | batphone, just do it, bofh style |
| 15:40.03 | batphone | salviadud: its people like you who give me hope |
| 15:40.13 | mountainm2k | I thought I could solave my second dial tone question using it -- but it seems to lock up the outbound ZAP channel when I try to use it |
| 15:40.20 | salviadud | that way, you're the man |
| 15:40.22 | mountainm2k | (IE fast busy, and it stays that way for a while) |
| 15:40.27 | batphone | i can secure a voip network from layer 1 up, i just need the authority to tell someone not to fuck with my network... |
| 15:41.05 | salviadud | batphone, tell them to obey the fist, and just do what you need to do, they'll thank you for it later |
| 15:41.15 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 15:41.33 | salviadud | and you can say, "you can thank me later, mkay?" |
| 15:42.29 | batphone | too many people are telling me not to worry about the brute force ssh attempts.. my concern is some dumbass making a 'test' user one day |
| 15:42.34 | batphone | then bam there goes the whole network |
| 15:43.14 | Juggie | users shoudnt have root |
| 15:43.15 | dongs | why is a "dumbass" allowed to make accounts on your system |
| 15:43.49 | batphone | dongs: have you tried finding guys that are familiar with linux and * enough to do tier 2 support that dont require $50,000 a year? |
| 15:44.08 | dongs | batphone: no, we use Microsoft Windows XP Professional SP2, thank god |
| 15:44.22 | batphone | haha |
| 15:44.23 | SplasPood | batphone: one gets what they pay for |
| 15:44.25 | salviadud | lol@dongs |
| 15:44.28 | dongs | batphone: i:d probably shat myself if I had to deal with Linux and people around it |
| 15:44.52 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
| 15:45.39 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 15:45.43 | batphone | im shitting myself atm. some fucker tried to brute force my supervisors name about 40 times, then went on to some of my tech's names. |
| 15:45.56 | LoRez | batphone: use sshwatch |
| 15:48.44 | *** join/#asterisk IMG-SD (n=IMG-SD@as2.imperialgroup.ca) |
| 15:49.16 | vader-- | ok making a straight through cable brings up my CPE but the CPE has an alarm that comes on |
| 15:49.29 | vader-- | but it registers DSL1, DSL2, DS1, BZ1, etc |
| 15:49.34 | dongs | last year some idiot installed linux with root:admin as a pass, and within hours it was a part of some latvian botnet |
| 15:49.36 | vader-- | so im thinking straight through is the right way |
| 15:49.54 | *** join/#asterisk W9SH (n=Steve_He@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 15:50.53 | salviadud | latvian botnet? wow, the madness |
| 15:51.31 | dongs | or maybe romania. one of those shitty countries. |
| 15:51.38 | salviadud | i would disable root login on ssh if i had that kind of pass |
| 15:51.53 | dongs | salviadud: obviously, you missed the part about "some idiot" |
| 15:52.31 | vader-- | now im getting a yellow/red alarm in zztool |
| 15:52.33 | vader-- | what does the yellow mean? |
| 15:52.35 | LokeshIndian | romanians already entered once into my asterisk...they r just assholes |
| 15:52.49 | salviadud | dongs, i'm just kidding dude |
| 15:53.13 | salviadud | dongs, where the hell do you work at? |
| 15:53.24 | vader-- | what does it mean when you have a yellow/red alarm in zztool? |
| 15:53.49 | Juggie | link level error |
| 15:54.01 | dongs | salviadud: NTT |
| 15:54.13 | batphone | im just waiting on some remote root sip exploit to come out. |
| 15:54.57 | vader-- | juggie does that mean it's usually on the circuit end or asterisk's? |
| 15:55.00 | dongs | nobody is forcing you to run asterisk as root |
| 15:55.09 | batphone | dongs: on the contrary |
| 15:55.13 | dongs | orly? |
| 15:55.17 | dongs | i run mine as a user |
| 15:55.19 | dongs | with zero problems |
| 15:55.26 | batphone | so do the systems i design |
| 15:55.33 | *** join/#asterisk cyscapes (n=grayman@65.197.217.62) |
| 15:55.39 | *** part/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 15:55.40 | salviadud | what's the file permission number for that? |
| 15:55.48 | salviadud | i forgot |
| 15:55.54 | batphone | the problem comes from techs... people not familiar enough with linux to understand the implications of typing "asterisk" |
| 15:56.23 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 15:56.38 | batphone | im close to giving up on it. i mean, whats the point of asking someone how to secure some shit if you dont listen to them when their opinion compromises the level of effort the rest of the team is willing to undergo for security |
| 15:56.56 | g__ | My name is Geoff and I'm a Asterisk Administrator. |
| 15:57.09 | batphone | Hi Geoff. |
| 15:57.21 | salviadud | hey man |
| 15:57.23 | g__ | It's been 30 minutes since our last crash.. |
| 15:57.32 | g__ | .. and I'm allready in withdrawl. |
| 15:57.34 | dongs | (not surprising) |
| 15:57.43 | *** join/#asterisk jcollie[work] (n=jcollie@161.210.6.107) |
| 15:57.44 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 15:57.44 | salviadud | what version of asterisk? |
| 15:57.47 | batphone | I have a box up for 420 days ;) |
| 15:57.48 | g__ | 1.2.9.1 |
| 15:57.52 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-136-117.phlapa.east.verizon.net) |
| 15:57.58 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 15:58.01 | batphone | does nothing but faxes all day long in and out of 2 zap cards. |
| 15:58.02 | salviadud | 420 sounds stoner-like |
| 15:58.06 | McLazarus | oej: I emailed a sip trace / debug from trying the t38passthrough where the udptl stream was rejected as unknown. |
| 15:58.16 | *** part/#asterisk jcollie[work] (n=jcollie@161.210.6.107) |
| 15:58.22 | IMG-SD | Sorry to barge in, but I have a simple question. Is there a way to have Asterisk perform additional Dial commands AFTER the caller hangs up? For example, let's say I call someone, and I hang up, can I have Asterisk execute additional commands after the hangup occurs? |
| 15:58.24 | eKo1 | batphone: What zap cards? |
| 15:58.34 | McLazarus | I tried to add to bug 5090 but it was closed yesterday. |
| 15:59.02 | McLazarus | maybe that means I should be testing with straight trunk and not http://svn.digium.com/svn/asterisk/team/group/t38passthrough |
| 15:59.10 | g__ | It's one of those "pri_dchannel: Ring requested on channel 0/3 already in use on span 3. Hanging up owner" -like hangs. |
| 15:59.15 | eKo1 | IMG-SD: if the caller hangs up, who is going to be the caller of the call then? |
| 15:59.44 | oej | McLazarus: THank you |
| 15:59.45 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 15:59.47 | dongs | IMG-SD: yo ucan jsut add more sequences to the extension to proceed after hangup, right? |
| 15:59.59 | eKo1 | dongs: try it and find out |
| 16:00.01 | IMG-SD | AseKo1: Asterisk itself would be the caller... I need Asterisk to send DTMF digits to a FXO line after a call hangs up the line... |
| 16:00.01 | oej | McLazarus, yes you should now work with trunk for testing |
| 16:00.10 | dongs | IMG-SD: exten=>1,1,DIal(), exten=>1,2,DoshitAfteRHangup() |
| 16:00.17 | g__ | IMG-SD: you could look at using the 'h' extension, but there are a few warnings on the subject.. |
| 16:00.23 | dongs | or that too |
| 16:00.26 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 16:00.42 | *** join/#asterisk saftsack (n=saftsack@p54A7F024.dip.t-dialin.net) |
| 16:01.15 | dongs | thing is, i think once the user hangs up that fxo channel is already dead |
| 16:01.30 | g__ | Does anyone know how responsive Digium is to phone calls? |
| 16:01.38 | g__ | (support-like-phone calls?) |
| 16:01.47 | dongs | g__: im sure they would be if you paid them |
| 16:01.51 | IMG-SD | I can't use "g" because the callee is not the one hanging up... the caller is hanging up. As soon as the caller hangs up, adding another sequence to the dial plan doesn't work... |
| 16:01.52 | Juggie | call the support line and find out |
| 16:02.07 | Juggie | its free if you own a card isnt it |
| 16:02.07 | g__ | dongs: good point. |
| 16:02.32 | McLazarus | oej: thanks, I will recompile and retry |
| 16:02.50 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 16:03.01 | g__ | That's true. Although our PRI card is currently a Sangoma.. I guess I don't have to tell them that :) |
| 16:04.03 | g__ | Are there any Digium employees listening? |
| 16:04.38 | Strom_C | I'm just a contractor :) |
| 16:04.41 | file | I'm not in support though |
| 16:04.45 | g__ | You can keep a secret, right? |
| 16:05.03 | Juggie | i'm sure they do a proof of purchase routine |
| 16:05.13 | *** join/#asterisk tlowe_ (n=tlowe@bgp.terrorist.net) |
| 16:06.43 | IMG-SD | g__: Thanks for suggesting the "h" extensions.. I didn't know of its existence. I will give it a shot! :) |
| 16:07.28 | g__ | good luck.. |
| 16:07.37 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:08.16 | g__ | file & Strom_C, what do you guys do? Development? |
| 16:08.52 | file | Strom_C hangs around the office like a poser :D and I do development |
| 16:09.06 | g__ | ouch! |
| 16:09.16 | Strom_C | g__: Digium support is technically installation support for Digium hardware. I'm really not sure you're going to get very far with free support without having purchased a Digium interface card. However, if you want to enter into a paid support agreement, then things might be different |
| 16:09.35 | file | Strom_C: you rock Strommy boy! |
| 16:10.07 | g__ | There's a bug number and everything.. |
| 16:10.16 | Jason99 | Is there a way to know within a context if an extension is offline or online? |
| 16:10.18 | file | on mantis? |
| 16:10.29 | Strom_C | g__: I'd be interested to see what happens |
| 16:10.30 | batphone | Strom_C: how much different? The answer i get from support is always a cut and paste from the wiki that Ive read 43 times. |
| 16:10.47 | batphone | regardless of the 1000's of digium cards ive sold. |
| 16:10.52 | Strom_C | batphone: I don't work in support, so I'm not familiar with their procedures |
| 16:11.16 | batphone | dont get me started. |
| 16:11.21 | batphone | ;) |
| 16:11.32 | I-MOD | batphone, what kind of question? |
| 16:11.41 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 16:11.47 | batphone | basically i was having trouble a while back stacking T1 cards in a box |
| 16:11.47 | g__ | file: yup. |
| 16:11.55 | Strom_C | batphone: if you've got a gripe though, writing a letter would probably be a good idea |
| 16:11.55 | file | what's the number? |
| 16:12.04 | CrashHD | can anyone tell me why I can not get ringing to generate when dialing multiple sip phones (even using the r option)? |
| 16:12.21 | Strom_C | CrashHD: what do you mean |
| 16:12.48 | g__ | file: give me a sec.. |
| 16:12.51 | CrashHD | Dial(Sip/1&Sip/2,20,r) does not generate ringing for the calling party |
| 16:13.07 | CrashHD | just dead air until answered |
| 16:13.10 | file | CrashHD: pastebin the CLI output |
| 16:13.20 | file | plus info on the calling party - ie: technology |
| 16:13.27 | Strom_C | and the called party |
| 16:13.41 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 16:13.55 | file | and the cool party |
| 16:14.08 | nortex | Is bridgeing a fax from a T-1 to a Channel bank supported on Snagoma cards? |
| 16:14.12 | asterboy | Sangoma just lost me a days worth of billing. |
| 16:14.34 | file | asterboy: how so? |
| 16:14.39 | asterboy | and their IRC support...well, it does not exist. |
| 16:14.39 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
| 16:14.44 | *** join/#asterisk nazgool (n=oli@dip-108-135.bras.dsl.breisnet.com) |
| 16:14.45 | asterboy | Dead card. |
| 16:14.45 | *** join/#asterisk ManxPower (i=ewieling@204.sub-70-210-13.myvzw.com) |
| 16:14.53 | file | nobody is perfect |
| 16:14.54 | asterboy | confirmed from yesterday |
| 16:14.55 | nazgool | hi all |
| 16:15.08 | CrashHD | http://pastebin.com/708757 |
| 16:15.11 | *** join/#asterisk Qb3rt (n=jhgjkgui@216.252.87.8) |
| 16:15.16 | nortex | The irc serv for Sangoma is irc.irchighway.net there is a typo on thew wiki |
| 16:15.33 | asterboy | yes, but you would think if they are going to put up an IRC channel that they would at least man it with some guy from India for 2 cents an hour. |
| 16:15.42 | CrashHD | I'm using a sip phone on the same system to dial a 11 digit number which is terminating to another asterisk box (IAX2) and then sending back to the original box |
| 16:15.45 | file | CrashHD: do an iax2 debug and pastebin it |
| 16:15.59 | file | er wait |
| 16:16.01 | file | that's already answered |
| 16:16.24 | g__ | file: #6997, still open |
| 16:16.26 | coppice | indians don't work for 2cents an hour. someone in a call centre gets maybe $600-800 a month |
| 16:16.30 | sevard | Jun 14 11:16:05 WARNING[12368]: rtp.c:1017 ast_rtp_settos: Unable to set TOS to 184 |
| 16:16.31 | nortex | asterboy, No the feeling, I cannot even get a reply from support on my configuration. |
| 16:16.32 | sevard | :\ |
| 16:16.42 | LokeshIndian | i agree with coppice |
| 16:16.45 | asterboy | yikes! no reply |
| 16:16.50 | asterboy | that is so unprofessional |
| 16:17.01 | CrashHD | also the call is not answered, the user picks up the call says hello but the system shows the call as not being answered |
| 16:17.09 | nortex | asterboy, And no answer on the phone. |
| 16:17.31 | CrashHD | http://pastebin.com/708768 |
| 16:17.53 | CrashHD | the iax2 debug I just pasted, the user actually picked up (ext 100) and said hello but hung up because he could not hear me |
| 16:18.04 | asterboy | I'm talking to someone now |
| 16:18.05 | salviadud | how do i get a normal user to run asterisk, do i change the file permissions, or just add it to the path? |
| 16:18.19 | vader-- | what does it mean when you have a yellow/red alarm in zztool? |
| 16:18.22 | vader-- | does that mean it's usually on the circuit end or asterisk's? |
| 16:18.28 | vader-- | problem |
| 16:18.42 | InfraRed | red means line isnt connected |
| 16:18.45 | ManxPower | sevard, you are not running asterisk as root |
| 16:18.51 | vader-- | im getting both red/yellow |
| 16:18.54 | asterboy | Phone: 800·388·2475 x 119 |
| 16:18.57 | InfraRed | no idea |
| 16:18.58 | ManxPower | CrashHD, any NAT involved? |
| 16:18.59 | InfraRed | google ? |
| 16:19.00 | asterboy | for Sangoma TEch Support |
| 16:19.06 | file | CrashHD: try notransfer=yes in your iax.conf |
| 16:19.25 | ManxPower | CrashHD, you have purchased G729 licenses?? |
| 16:19.38 | CrashHD | ManxPower: ya some but extension to extension works fine |
| 16:19.42 | CrashHD | 711 set |
| 16:19.47 | asterboy | nortex, try the phone |
| 16:19.50 | nortex | asterboy, Is it David? Maybe your keeping them to busy |
| 16:19.56 | sevard | ManxPower: Right, what sort of permissions does my regular user need to set the ToS? |
| 16:19.59 | ManxPower | CrashHD, extension to extension would work fine without g729 licenses. |
| 16:20.18 | ManxPower | sevard, no idea. Try running asterisk as root and use the -U and -G command line options to Asterisk |
| 16:20.19 | asterboy | Didn't catch the name, I was forwarded to a lady in the RMA dept....Had to leave a messge...go figure |
| 16:20.24 | ManxPower | I think the Wiki also talkes about it. |
| 16:20.47 | nortex | asterboy, Thanks calling now. |
| 16:20.52 | mopri | hi.. i recorded the voices for the menu, i selected gsm (pcm) for output, but my asterisk is not recognizing the gsm, cause it causes an error. any suggestions recording stuff? I used sound forge and adobe audition. |
| 16:21.03 | CrashHD | file: still the same |
| 16:21.28 | Strom_C | mopri: gsm is not pcm |
| 16:21.30 | CrashHD | ManxPower: everythign is set to use 711 first |
| 16:21.43 | nazgool | i just installed my asterisk (1.2.9.1) with chan_capi-cm-0.6.5. my trouble is, an incoming (capi) call does show, but isn't connected to any extension. i get a capi debug message on the asterisk console such as: |
| 16:21.43 | mopri | ok.. |
| 16:21.44 | ManxPower | CrashHD, then disallow G729 in sip.conf and iax.conf |
| 16:21.45 | nazgool | <PROTECTED> |
| 16:21.45 | nazgool | <PROTECTED> |
| 16:21.48 | nazgool | <PROTECTED> |
| 16:21.55 | mopri | where can i get the gsm codec? |
| 16:21.59 | nazgool | any clue what i might be doing wrong? |
| 16:22.02 | Strom_C | mopri: downsample your adobe audition files to 8khz 16-bit audio and save them as wav |
| 16:22.03 | dongs | you dont, its built in |
| 16:22.11 | ManxPower | mopri, it's included in Asterisk |
| 16:22.13 | mopri | for adobe audition or any other, maybe a mp3 gsm converter o wav gsm .. |
| 16:22.19 | Strom_C | mopri: downsample your adobe audition files to 8khz 16-bit audio and save them as wav |
| 16:22.22 | CrashHD | ManxPower: already done (dis all, allow ulaw) |
| 16:22.32 | mopri | so i save them as wav |
| 16:22.35 | Strom_C | mopri: the resulting sound will be a lot better over the phone |
| 16:22.39 | CrashHD | hmm |
| 16:22.39 | mopri | and then what?.. asterisk has a converter |
| 16:22.41 | g__ | file, I think i found the bug in mantis.. #6997; the major symptom was the pri desyncronization. |
| 16:22.41 | CrashHD | on sip it is |
| 16:22.45 | CrashHD | let me set on iax |
| 16:22.46 | mopri | o do i put them in the sound folder? |
| 16:22.48 | Strom_C | mopri: no, asterisk will play the wav files |
| 16:22.49 | sevard | ManxPower: The wiki doesn't talk about what permissions you need to set ToS |
| 16:22.55 | CrashHD | file, any ideas? |
| 16:22.56 | sevard | Does it need direct dev write permissions or what? |
| 16:22.59 | ManxPower | sevard, Ak well. |
| 16:23.14 | mopri | excelent |
| 16:23.18 | ManxPower | sevard, Only root can set things like ToS, source, address, etc. |
| 16:23.21 | mopri | i'll try that :P |
| 16:23.22 | file | CrashHD: not off the top of my head |
| 16:23.38 | *** join/#asterisk ToyMan (n=stuq@74-32-26-30.dsl1.mdl.ny.frontiernet.net) |
| 16:23.54 | asterboy | Gentek requires I pay shipping for their defective product...that sucks |
| 16:24.07 | ManxPower | CrashHD, so you have disallow=all allow=ulaw in sip.conf and iax.conf and have notransfer=yes in iax.conf? |
| 16:24.56 | ManxPower | nazgool, do you have exten => 0123456789,1,Whatever in extensions.conf? |
| 16:25.11 | CrashHD | ManxPower: yes, and still no ringing and when someone picks up the Dial() call [made to multiple people] they can not hear the calling party and the asterisk shows as not picked up [even though I hear audio] and moves on |
| 16:25.49 | ManxPower | CrashHD, so you have TWO problems. Caller does not hear ringback and asterisk does not see the call is picked up. |
| 16:26.01 | ManxPower | for no ringback, make sure you have /etc/asterisk/indications.conf set up. |
| 16:26.10 | ManxPower | Did you say if you are using NAT or not? |
| 16:26.18 | CrashHD | indications is setup |
| 16:26.38 | CrashHD | all phones are passing through nat to the server |
| 16:27.08 | nazgool | ManxPower: ok you're right, i had only an extension s that did a goto to demo. i thought i would get an error, but it doesnt give me a "no such extension" error. just fails. with the extension it works |
| 16:27.10 | nazgool | thanx |
| 16:27.10 | ManxPower | CrashHD, Is the server behind NAT or just the phones? |
| 16:27.16 | dongs | lol nat |
| 16:27.17 | CrashHD | just the phones |
| 16:27.28 | ManxPower | so Asterisk is on a public IP address? |
| 16:27.32 | CrashHD | correct |
| 16:27.41 | ManxPower | and Asterisk is NOT running on a box that does NAT? |
| 16:27.47 | CrashHD | correct |
| 16:27.55 | ManxPower | Are the phones configured for NAT? |
| 16:28.22 | CrashHD | I haven't done anything special for them. |
| 16:28.27 | ManxPower | Good. |
| 16:28.37 | ManxPower | are the phone behind the same nat router? |
| 16:28.39 | *** join/#asterisk W9SH (n=chatzill@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 16:28.58 | CrashHD | most are, the one I'm calling from is behind a seperate nat router |
| 16:29.08 | CrashHD | my home has 1 phone, office has other phones |
| 16:29.29 | ManxPower | what nat router? |
| 16:29.45 | CrashHD | my home has a dlink something or another, the office uses a linux box |
| 16:29.46 | ManxPower | Cisco? Linksys? Dlink? |
| 16:30.07 | ManxPower | CrashHD, sounds like you need sip debug and learn how to read it. |
| 16:30.35 | ManxPower | I don't really do VoInternet since Asterisk isn't really ready for that. |
| 16:30.44 | CrashHD | but why would extension to extension work just fine |
| 16:30.51 | CrashHD | but extension to multi extension not |
| 16:31.36 | ManxPower | CrashHD, try extension to extension with Tt as the Dial options. Does extension to extension still work? |
| 16:33.00 | vader-- | hmm this sucks my pri is comming up with an alar |
| 16:33.01 | vader-- | m |
| 16:33.03 | CrashHD | ringback fine, waiting for someone to answer....... |
| 16:33.25 | ManxPower | CrashHD, you confirmed the Tt by watching the CLI? |
| 16:33.34 | sevard | Anyone else have any input on ToS as a non-root user? |
| 16:33.39 | ManxPower | vader--, what kind of alarm? |
| 16:33.47 | vader-- | well on asterisk it says yellow/red |
| 16:33.57 | vader-- | on my cpe it's just a red alarm light |
| 16:34.02 | ManxPower | vader--, call your telco, say "I'm getting a red alarm" |
| 16:34.12 | CrashHD | ManxPower: yes, confirmed |
| 16:34.18 | ManxPower | assuming you have confirmed it's not a cable problem |
| 16:34.35 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 16:34.53 | Strom_C | vader--: have you tried a hard loopback test? |
| 16:35.02 | vader-- | hard loopback? |
| 16:35.06 | Strom_C | yes |
| 16:35.11 | *** join/#asterisk alephco1 (n=Weibe@host75.net14.mcsnet.ca) |
| 16:35.12 | vader-- | how do you do that? |
| 16:35.16 | Strom_C | one sec |
| 16:35.18 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:35.21 | Strom_C | i'll show you a graphic :) |
| 16:35.25 | generalhan | whats up all !? |
| 16:36.14 | CrashHD | I get audio to and from my phone whenever I call the system. I figure I would be getting ringback at the very least |
| 16:36.35 | CrashHD | exten => 2,1,Dial(Sip/100&Sip/101&Sip/105&Sip/106&Sip/109&Sip/111&Sip/112,20,r) |
| 16:36.42 | CrashHD | you see anything wrong with the dial string? |
| 16:37.14 | ManxPower | CrashHD, other than the "r" no. |
| 16:37.18 | Strom_C | vader--: make a plug just like the diagram and see what happens when you plug it into the telco smartjack and then the T1 card |
| 16:37.27 | ManxPower | CrashHD, you know dialing 1 works, try 2, then three, then four, etc |
| 16:37.36 | CrashHD | r is to force ringback |
| 16:37.54 | CrashHD | why would I do that? the dtmf works fine |
| 16:38.14 | ManxPower | CrashHD, no, r is "override any message you should be getting and force a ringback even if the caller should not hear ringback." "r" HIDES issues. |
| 16:38.25 | ambriento | 4 |
| 16:38.33 | ManxPower | CrashHD, no, I means dialing 2 devices at a time, then three devices at a time. |
| 16:38.45 | ManxPower | Perhaps ONE of your devices is causing a problem. |
| 16:38.51 | CrashHD | but the man pages say that when dialing multiple parties ringback is flakey and should be forced |
| 16:39.00 | CrashHD | ok I'll give that a whirl |
| 16:39.07 | ManxPower | CrashHD, the man page says no such thing. |
| 16:39.13 | CrashHD | the wiki I mean |
| 16:39.25 | ManxPower | the Wiki is wrong. |
| 16:39.36 | vader-- | storm ya ive done that in the past |
| 16:39.44 | ManxPower | There was an issue in PRE 1.0.0 that r was needed if dialing more than 1 party. |
| 16:39.55 | ManxPower | Heck, I think that issue was fixed before 0.6.5 |
| 16:40.01 | CrashHD | lol |
| 16:40.19 | Strom_C | vader--: and what happens when you do it? |
| 16:40.26 | Strom_C | vader--: my name is Strom, not Storm |
| 16:40.59 | CrashHD | if I'm dialing an offline sip phone, would that hurt? |
| 16:41.10 | ManxPower | CrashHD, it should not. |
| 16:41.34 | ManxPower | Use "r" like you would use a format and reinstall of the OS. i.e. as a last resort after you have tried everything else. |
| 16:41.51 | CrashHD | even with just 2 extensions ringback is not played |
| 16:42.02 | CrashHD | (tried different extension combo's as well) |
| 16:42.12 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 16:42.16 | Strom_C | CrashHD: what kind of equipment is at the various extensions? |
| 16:42.33 | CrashHD | my phone is a polycom 601, other phones are astra 9133i |
| 16:42.38 | CrashHD | latest firmware on the astra's |
| 16:42.43 | CrashHD | dunno about the polycom |
| 16:42.54 | m4rkl4r | google is no help, how does one get the return value of an application in the dial plan |
| 16:42.55 | CrashHD | I do get CLI notice that the phones are ringing |
| 16:43.12 | [TK]D-Fender | CrashHD : IP 601 shipped with SIP 1.6.2 by default IIRC so you should have that or later |
| 16:43.19 | generalhan | CrashHD: thats odd ... im using 35 Aastra 9112i and i dail 15 at once with one queue, they all ring fine |
| 16:43.33 | [TK]D-Fender | CrashHD : Are none of the phones getting ringing? |
| 16:43.41 | CrashHD | they all ring fine |
| 16:43.49 | [TK]D-Fender | CrashHD : Whats the call source? |
| 16:44.15 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:44.22 | CrashHD | polycom phone that outbounds from switch 1 and bounces off switch 2 (via iax) back to switch 1 |
| 16:46.23 | [TK]D-Fender | CrashHD : So Polycom > *1 > IAX > *2 > Multiple phones ? |
| 16:46.51 | generalhan | Hey guys: http://generalhan.pastebin.ca/65526 : can some one take a look at this for me? this is a fax machine calling in on one zap channel and my dialplan dialing our fax machine on ZAP/50 but something goes wrong with the native bridge |
| 16:47.05 | generalhan | i need some suggestions |
| 16:47.05 | CrashHD | Poly > *1 > IAX *2 > IAX > *1 > Multi Phones |
| 16:47.11 | CrashHD | simulating true inbound |
| 16:47.37 | CrashHD | looks like *2 is getting out of the audio path |
| 16:48.00 | [TK]D-Fender | hmm |
| 16:48.57 | [TK]D-Fender | generalhan : Tried calling it by voice and seeing if it works? |
| 16:49.02 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 16:49.32 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 16:49.38 | Strom_C | droops! |
| 16:49.44 | droops | Strom! |
| 16:49.49 | generalhan | [TK]D-Fender: yes. i called even from my desk phone to the machine ... it rings .. the fax machine picks up and then i get the "Attempting native bridge" then the hangup |
| 16:49.50 | Strom_C | how are you, man? |
| 16:50.00 | droops | im loving asterisk |
| 16:50.03 | droops | im loving linux |
| 16:50.07 | droops | im loving php |
| 16:50.07 | m4rkl4r | perhaps this is a better question: in what context of asterisk usage are the return values documented in the applications accessable? |
| 16:50.20 | ManxPower | generalhan, I see nothing wrong with that |
| 16:50.21 | droops | im just hating the people who keep changing their minds about crap |
| 16:50.31 | syle | droops: have a woman in your life? |
| 16:50.39 | Strom_C | droops: crappy clients? |
| 16:50.39 | ManxPower | m4rkl4r, return values of the applications? |
| 16:50.42 | generalhan | ManxPower: i dont know what the deal is ... it just hangs up both lines |
| 16:50.43 | m4rkl4r | dial plans? perl/python/java agi? C code? |
| 16:50.44 | m4rkl4r | yes |
| 16:50.47 | CrashHD | ok new info |
| 16:50.47 | droops | crappy partners |
| 16:50.49 | [TK]D-Fender | generalhan : forget the fax, plug a PHONE on there and try and answer voice and chat... |
| 16:50.55 | ManxPower | m4rkl4r, C code. |
| 16:50.59 | droops | syle, yep all maried and everything |
| 16:51.03 | CrashHD | if I do not go out iax and back everything works |
| 16:51.10 | m4rkl4r | alright. i appreciate it |
| 16:51.31 | ManxPower | In fact the return code values of the applicaitons were removed from the "show application whatever" docs because they are not accessable from the dialplan |
| 16:51.32 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 16:51.43 | syle | droops: any kids? |
| 16:51.47 | ManxPower | of course many applications set dialplan variables, but thats not what we are talking about. |
| 16:51.48 | generalhan | [TK]D-Fender: ok ... i dont think i have any analog phones ... let me see if i can find a phone somewhere |
| 16:51.56 | m4rkl4r | no. |
| 16:52.11 | [TK]D-Fender | CrashHD : try just calling something on *2 and see if it rings before heading back... might tell you what leg of the trip it dies on... |
| 16:52.15 | droops | its epecially fun, the day we go to sell our product, my partners decided that something other than what was inportant yesterday is now the most important thing |
| 16:52.20 | droops | syle, not yet |
| 16:52.22 | ManxPower | CrashHD, still sounds like notransfer=yes is not being seen. perhaps it's not a global option, but onyl a peer/friend/user option |
| 16:52.52 | ManxPower | generalhan, you can get them for $9 at walmart. |
| 16:52.52 | droops | so it makes for fun times |
| 16:53.10 | syle | yeah make sure your in shape if you do, my god they are a workout and a half |
| 16:53.13 | droops | but asterisk is running great, and thats the key |
| 16:53.33 | CrashHD | notransfer=yes is in general on both machines (but ya you are right, I see the call being "transferred") |
| 16:54.04 | *** join/#asterisk Isamar (n=Isamar@200222128102.user.veloxzone.com.br) |
| 16:56.03 | CrashHD | argh |
| 16:56.18 | CrashHD | this is frustrating |
| 16:56.19 | CrashHD | lol |
| 16:56.49 | CrashHD | so the call is looping out of *1 and right back in. that is what is causing the problem....but why |
| 16:57.18 | dlynes_office | CrashHD: do you have a pastebin of your extensions.conf? |
| 16:57.28 | m4rkl4r | ManxPower: we are talking about, for example, page 261 of the oreilly book on MeetMeCount() that says: "returns 0 on success or -1 on a hangup" |
| 16:58.14 | CrashHD | while I'm listening to the AA the call state shows as this on *1 >http://pastebin.com/708842 |
| 16:58.18 | ManxPower | m4rkl4r, You'll notice that CURRENT docs no longer talk about the return value. |
| 16:58.25 | CrashHD | dlynes_office: I could get one for ya, hold on |
| 16:58.42 | m4rkl4r | Oh, I see |
| 16:58.49 | generalhan | Damn ... its not a problem with the bridge at all ... i plugged my reg. fax machine in there and it woked perfectly. so now i gotta figure out why the software fax is screwing me up |
| 16:58.49 | CrashHD | that call status is weird...why would it show as ringing |
| 16:58.55 | ManxPower | m4rkl4r, that book is pretty accurate, but a few things changed since it went to press, this is one of them |
| 16:59.00 | Delvar | ok a little help please: asterisk billing in regards to teansfered calls, i need to be able to bill all legs of the call, iv got a few ideas but i dont like them, iv looked on wiki but cant find anything to help, does anytone have a link or pointers for me? |
| 16:59.03 | dlynes_office | CrashHD: yeah...i just logged on, so if you posted it earlier, it was before i logged on |
| 16:59.37 | ManxPower | Delvar, If you use IAX then you must have notransfer=yes or you will not get CDRs for all legs of the call. |
| 16:59.56 | CrashHD | dlynes_office: what do you make of http://pastebin.com/708842? I'm actually listening to the auto attendant message but the system thinks it is still ringing? |
| 17:00.03 | file | unless you use media only transfers |
| 17:00.17 | Qwell[] | file: Nobody asked you :p |
| 17:00.22 | dlynes_office | CrashHD: i can't make anything of it, unless I can see your extensions.conf :) |
| 17:00.23 | m4rkl4r | ManxPower: Well, making simply a gratuitous, wild suggestion, I suppose it would be useful to export those return values into the dial plan as a variable. |
| 17:00.28 | CrashHD | heh one min |
| 17:00.29 | ManxPower | file, media only transfers are not part of any released version of asterisk. |
| 17:00.51 | ManxPower | m4rkl4r, not really, since all the apps should set the correct channel variables. |
| 17:01.04 | dlynes_office | CrashHD: i suspect it might be a collision of priorities...that's why i'm wanting to look at your extensions.conf file |
| 17:01.12 | Delvar | ManxPower: hehe thanks.. but thats not waht i mean :), i am not using asterisk CDRs for post pay, i have custom agi scripts for prepay one at teh start of the call and another at the end. |
| 17:01.29 | ManxPower | Delvar, so what is your SPECIFIC problem? |
| 17:02.17 | CrashHD | http://pastebin.com/708853 |
| 17:02.44 | CrashHD | a collision of priorities? but the call dials fine? |
| 17:02.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:03.06 | dlynes_office | CrashHD: didn't you say it kept looping back on itself? |
| 17:03.14 | CrashHD | no |
| 17:03.26 | dlynes_office | <CrashHD> so the call is looping out of *1 and right back in. that is what is causing the problem....but why |
| 17:03.32 | dlynes_office | sure looks like it to me |
| 17:03.39 | CrashHD | sorry, out of context |
| 17:03.54 | CrashHD | the call is sent out of *1 to *2 and back to *1 |
| 17:04.05 | CrashHD | to test *1 as a real inbound call |
| 17:04.17 | dlynes_office | Does it keep looping 1 to 2 to 1 to 2 to 1 to 2 to 1 to 2? |
| 17:04.32 | dlynes_office | Or just 1 to 2 to 1 and then stops? |
| 17:04.32 | CrashHD | when doing that if I Dial(Multiple extensions) I get no ringback or connection |
| 17:04.44 | CrashHD | 1 to 2 to 1 and 1 answers fine |
| 17:04.54 | Delvar | ManxPower: a is a voip user b and c are pstn destinations, a calls b, a transfers b to c, a is hung up and it calls my script in the h exten and bills ok. then c hangs up and again calls my script and bills ok. the problem is there is no h in the b leg of the call after teh transfer so i get billing for a to b for the first part and b to c second part but i dont get the billing for b on teh second part... |
| 17:04.57 | Delvar | feww... |
| 17:05.06 | CrashHD | 2's CLI tells me it gets out of the path |
| 17:05.16 | CrashHD | so it ends up 1 to 1 |
| 17:05.22 | dlynes_office | CrashHD: well, one good thing |
| 17:05.27 | ManxPower | Delvar, It sucks to be you. |
| 17:05.37 | dlynes_office | CrashHD: You don't have autofallthrough set to yes :) |
| 17:05.37 | Delvar | tell me about it |
| 17:05.48 | CrashHD | heh |
| 17:05.49 | m4rkl4r | ManxPower, does every official application set some variables, or do you just mean that applications for which a variable would be semantically useful have them? Meaning two seems to be the gist of the docs on voip-info |
| 17:05.53 | CrashHD | thanks to you, that is |
| 17:06.06 | dlynes_office | ah...you're the one i told to get rid of it :0 |
| 17:06.09 | dlynes_office | couldn't remember |
| 17:06.12 | CrashHD | lol ya |
| 17:06.14 | ManxPower | m4rkl4r, no, not all apps set variables. |
| 17:06.27 | CrashHD | I still say it shouldn't be the way it is...but it is so I deal |
| 17:06.38 | dlynes_office | CrashHD: I don't see a *2 or a *1 |
| 17:06.43 | m4rkl4r | ok. |
| 17:06.44 | m4rkl4r | thank you very much |
| 17:06.55 | dlynes_office | CrashHD: for that other call you've got that's indicating ringing, but playing back the autoattendant file |
| 17:06.57 | Delvar | so any ideas? links? guns? |
| 17:07.00 | CrashHD | *2 = asterisk box 2, *1 = asterisk box 1 |
| 17:07.15 | dlynes_office | CrashHD: which extension is that? |
| 17:07.26 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 17:07.44 | [TK]D-Fender | dlynes_office : Hey... what was the name of that rep at Williams you were mentioning? |
| 17:08.14 | CrashHD | I'm dialing an 11 digit number (1 + area + prefix + suffix) which hits the OUTBOUND (on system 1) which relays to system 2 (main trunk box) which in turn knows that the DID I'm trying is at system 1 and relays it back to system 1 (this is mainly for testing) |
| 17:08.25 | *** join/#asterisk key2 (n=ashdown@sd-420.dedibox.fr) |
| 17:08.29 | CrashHD | when it get's to system one it is hitting the INBOUND context |
| 17:08.34 | dlynes_office | [TK]D-Fender: We know two there....Don Williams (he's pretty good), and Roland Aucoin (he used to be at ummmm.....some nortel phone supplier....can't remember what it was offhand) |
| 17:08.50 | dlynes_office | Roland Aucoin just started there...Don Williams has made a career working for that company |
| 17:08.56 | dlynes_office | Erm Don Wright I mean |
| 17:08.58 | dlynes_office | not williams |
| 17:09.00 | dlynes_office | heh |
| 17:09.08 | dlynes_office | He's been there for about 30 years now |
| 17:09.29 | Delvar | ManxPower: so that means no then :( |
| 17:09.30 | [TK]D-Fender | dlynes_office : Roland.. thats the guy who abandoned me :( |
| 17:10.22 | dlynes_office | [TK]D-Fender: yeah, when I was going to place an order there the other day, I asked my boss whether he wanted me to use our regular guy (Don Wright), or to go with Roland (our sales rep from another company)...he didn't have any hesitation telling me to use Don Wright |
| 17:10.50 | [TK]D-Fender | dlynes_office : Well I've gone through one... just on to #2 now :) |
| 17:11.21 | dlynes_office | CrashHD: OUTBOUND == LONGDISTANCE? |
| 17:11.22 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 17:11.29 | dlynes_office | [TK]D-Fender: go for don wright |
| 17:11.30 | CrashHD | dlynes_office: correct |
| 17:11.45 | dlynes_office | [TK]D-Fender: he sends a lot of emails, but he's pretty solid |
| 17:11.56 | [TK]D-Fender | great, will call him up shortly. |
| 17:12.06 | dlynes_office | [TK]D-Fender: one sec, and i'll get his extension for you |
| 17:12.15 | [TK]D-Fender | Even better :) |
| 17:12.29 | dlynes_office | [TK]D-Fender: Just tell him Daniel at 24/7 Communications referred you, if he asks |
| 17:13.28 | [TK]D-Fender | gotcha |
| 17:13.30 | dlynes_office | oh yeah...Roland Aucoin was from Epik Networks |
| 17:13.49 | *** join/#asterisk liran_ (n=Coll@212.199.177.203.static.012.net.il) |
| 17:14.26 | dlynes_office | [TK]D-Fender: his direct line is 905-712-6371 |
| 17:15.02 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:15.27 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
| 17:16.58 | dlynes_office | [TK]D-Fender: he's out for lunch |
| 17:17.08 | dlynes_office | [TK]D-Fender: he won't be back until 1:40 or os |
| 17:17.12 | AlexCTI | Hi guys, quick question, If I set on sip.conf a user with "callwaiting = no" it only can receive one call at the time, right? |
| 17:17.14 | dlynes_office | s/os/so/ |
| 17:17.29 | g__ | Is there an easy was to set verbose=42 on asterisk' startup? |
| 17:17.43 | Qwell[] | g__: asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 17:17.48 | Qwell[] | That's about 42 |
| 17:17.51 | g__ | (besides `asterisk -rx 'set verbose 42'`?) |
| 17:17.51 | dlynes_office | g__: verbose=42? wtf for? none of the code looks for anything higher than 6 |
| 17:17.52 | CrashHD | g: asterisk.conf > [options] |
| 17:17.52 | CrashHD | <PROTECTED> |
| 17:17.54 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
| 17:17.54 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 17:18.01 | g__ | Qwell: that's our current method.. |
| 17:18.04 | Qwell[] | CrashHD: bah, going the "proper" way :p |
| 17:18.09 | CrashHD | lol |
| 17:18.15 | Qwell[] | dlynes_office: That's wrong, actually |
| 17:18.18 | g__ | Awesome guys.. |
| 17:18.28 | dlynes_office | Qwell[]: i've looked through the code extensively |
| 17:18.34 | dlynes_office | Qwell[]: so unless it's been changed |
| 17:18.35 | Qwell[] | dlynes_office: and I ran a grep the other day :P |
| 17:18.41 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 17:18.55 | dlynes_office | Qwell[]: so some idiot's gone and changed the max now? |
| 17:18.57 | Qwell[] | in fact... |
| 17:19.01 | Qwell[] | there never was a max |
| 17:19.03 | g__ | Qwell, yours certainly has style.. |
| 17:19.13 | dlynes_office | Qwell[]: sorry...lemme rephrase...the defacto max |
| 17:19.18 | CrashHD | so anyone have any thoughts on why an iax call that ends up back at the same system it orginiates from is having a problem? |
| 17:19.20 | Qwell[] | cdr_odbc uses 10 |
| 17:19.28 | g__ | dlynes_office: 42 was random. But that works for 6 as well. |
| 17:19.39 | dlynes_office | if( option_verbose>5 ) |
| 17:19.46 | dlynes_office | that's the biggest I've seen in the code |
| 17:19.46 | Qwell[] | 11, actually |
| 17:19.57 | Qwell[] | cdr/cdr_odbc.c: if (option_verbose > 10) |
| 17:20.05 | dlynes_office | Qwell[]: that's in a release? or trunk? |
| 17:20.07 | Qwell[] | res/res_jabber.c: if (option_verbose > 77) |
| 17:20.09 | Qwell[] | :D |
| 17:20.13 | g__ | Welcome the asterisk edition of "spinaltap" |
| 17:20.38 | dlynes_office | that's gotta be in trunk cause the release doesn't even have jabber |
| 17:20.42 | Qwell[] | dlynes_office: I doubt cdr_odbc verbosity levels were changed from release version |
| 17:20.46 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.238.136.Dial1.SanJose1.Level3.net) |
| 17:20.47 | g__ | Maybe if (option_verbose > 42 && option_verbose < 52) |
| 17:20.57 | Qwell[] | g__: I am so implementing that in chan_skinny |
| 17:21.07 | dlynes_office | Qwell[]: yeah...last time i looked at the verbose code was in 1.2.7.1 |
| 17:21.15 | *** part/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 17:21.40 | *** join/#asterisk assert_true (n=Sunil@59.176.26.247) |
| 17:21.56 | dlynes_office | CrashHD: actually |
| 17:22.04 | dlynes_office | CrashHD: I think I came across that problem before |
| 17:22.13 | dlynes_office | CrashHD: and solved it this way |
| 17:22.31 | CrashHD | dlynes_office: that's good because I only have so much hair left to pull |
| 17:22.59 | dlynes_office | CrashHD: Asterisk A -> call -> Asterisk B -> asterisk B extension & Local extension -> local extension -> Asterisk A extension |
| 17:23.26 | dlynes_office | CrashHD: i remember I couldn't go out to the originating box directly |
| 17:23.30 | CrashHD | so you had to hit a secondary extension |
| 17:23.35 | dlynes_office | CrashHD: i had to do it through a local extension |
| 17:23.50 | CrashHD | you have an example? |
| 17:23.52 | dlynes_office | not secondary...a local extension |
| 17:24.04 | CrashHD | I've never read about local extensions, never had the need |
| 17:24.04 | sevard | Does the Cisco CP-7960G support PoE, anyone know? |
| 17:24.05 | dlynes_office | as in Local/321 or something like that |
| 17:24.14 | Qwell[] | sevard: should |
| 17:24.14 | g__ | Qwell[]: oh dear.. |
| 17:24.41 | dlynes_office | CrashHD: anyways..I never thought to make a backup of that system |
| 17:24.49 | sevard | Qwell[]: I have a PoE switch laying around.. if it doesn't support PoE and I plug into it.. couldn't I brick it? |
| 17:25.03 | dlynes_office | CrashHD: And those losers are running Talkswitch now |
| 17:25.04 | CrashHD | dlynes_office: no worries I can figure it out, but doesn't this qual as a bug? |
| 17:26.10 | dlynes_office | CrashHD: i figure good ridddance...i was getting tired of "fixing" things whenever their network admin decided to screw with the network |
| 17:26.29 | *** join/#asterisk dandan (i=dandan@pacanka.com) |
| 17:26.33 | dandan | re all |
| 17:26.35 | dandan | :) |
| 17:26.49 | Qwell[] | sevard: only if you PoE switch is horribly flawed |
| 17:26.56 | dlynes_office | CrashHD: no idea...I never spent the time to hypothesize about it |
| 17:26.59 | dlynes_office | CrashHD: i just fixed it |
| 17:27.01 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 17:27.06 | Strom_C | sevard: the PoE switch should have a discovery protocol |
| 17:27.15 | CrashHD | dlynes_office: heh |
| 17:27.21 | Dandan | does anyone know /can help me with/ changing BEARER CAPABILITY field sent over PRI? |
| 17:27.26 | CrashHD | dlynes_office: bloody network admins ruin everything |
| 17:27.39 | sevard | So... It's plugged in now and the phone isn't on but the PoE light on the switch is blinking like mad |
| 17:27.50 | Strom_C | what kind of phone? |
| 17:27.52 | dlynes_office | CrashHD: now don't be saying that |
| 17:28.01 | sevard | Cisco CP-7960G |
| 17:28.03 | dlynes_office | CrashHD: i administer all our networks and systems :) |
| 17:28.13 | dlynes_office | CrashHD: and do all the bloody programming,t oo |
| 17:28.13 | Strom_C | sevard: Cisco uses its own wacly PoE protocol |
| 17:28.24 | Strom_C | er, wacky |
| 17:28.26 | sevard | oh friggen lame |
| 17:28.29 | Dandan | re dlynes_office, [TK]D-Fender |
| 17:28.29 | Dandan | :) |
| 17:28.46 | sevard | so I need to wire some sort of mutilated cross over or what |
| 17:28.53 | dlynes_office | Dandan: not this guy :) |
| 17:29.02 | dlynes_office | Dandan: I don't know anything about the inner workings of pri |
| 17:29.02 | Strom_C | sevard: you could theoretically do that, yes |
| 17:29.05 | CrashHD | dlynes_office: was a joke, I'm the network admin here |
| 17:29.14 | CrashHD | dlynes_office: phones are not my fortay |
| 17:29.26 | g__ | Does debug logging slow asterisk down, or does it just waste disk space? |
| 17:29.26 | dlynes_office | CrashHD: you missed my smiley above :) |
| 17:29.41 | dlynes_office | g__: impact is very minimal |
| 17:29.52 | Dandan | dlynes_office: BEARER CAPABILITIES field is also in h.323 too :/ I need to change it :) |
| 17:29.56 | g__ | k, thanks. |
| 17:30.01 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-2-21.red.bezeqint.net) |
| 17:30.10 | dlynes_office | Dandan: ah...but i have no idea what it even is |
| 17:30.10 | sevard | Strom_C: I'm plugging it into a netgear fs108p poe switch |
| 17:30.12 | CrashHD | heh |
| 17:30.15 | CrashHD | ok |
| 17:30.17 | CrashHD | so local chan |
| 17:30.22 | Dandan | dlynes_office: q.931 :) |
| 17:30.22 | CrashHD | brb |
| 17:30.33 | dlynes_office | Dandan: no...bearer capabilities |
| 17:30.46 | Dandan | that is explained in q.931 :) |
| 17:30.47 | dlynes_office | Dandan: i know what q.931 is |
| 17:31.02 | dlynes_office | Dandan: yeah...and like i said...i don't know the inner workings of pri |
| 17:31.12 | dlynes_office | Dandan: therefore I don't know the inner workings of q.931, either |
| 17:31.17 | Dandan | heh, thanks dlynes_office, i know you are trying to help :) |
| 17:31.26 | Dandan | should I post my problem to asterisk-users? |
| 17:31.28 | dlynes_office | Dandan: nope...not trying to help |
| 17:31.34 | dlynes_office | Dandan: i don't know enough to try |
| 17:31.45 | dlynes_office | Dandan: probably be a good place to start |
| 17:31.56 | dlynes_office | Dandan: more peeps read the mailing list than hop onto irc |
| 17:32.23 | Dandan | i gotta subscribe to it... |
| 17:32.41 | dlynes_office | Dandan: just subscribe to all the lists while you're at it |
| 17:32.44 | dlynes_office | they're all free |
| 17:32.56 | Dandan | yeah, and high-trafficky for sure |
| 17:32.58 | Strom_C | Dandan: then make sure you forward all the list mail to your cellphone |
| 17:33.07 | Strom_C | :) |
| 17:33.18 | dlynes_office | Dandan: yeah...asterisk-users and asterisk-biz and asterisk-dev are all quite high |
| 17:33.30 | dlynes_office | Dandan: asterisk-commits is extremely super duper high |
| 17:33.32 | Dandan | Strom_C: that's why I keep two separate email accounts :) |
| 17:33.45 | MikeJ__ | slashdot fun:7124327899 |
| 17:33.47 | *** join/#asterisk jtodd (n=jtodd@reserve-64-79-115-18.wiline.com) |
| 17:34.32 | liran_ | what is the function/purpose of defining a "context=something" argument in sip.conf for some sip extension? |
| 17:34.50 | dlynes_office | ~slashdot 7124327899 |
| 17:34.50 | Strom_C | liran_: that's where all outbound calls from that extension will start in the dialplan |
| 17:35.12 | dlynes_office | MikeJ__: ? |
| 17:35.16 | *** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net) |
| 17:35.16 | MikeJ__ | call the number |
| 17:35.24 | CrashHD | dlynes_office: that didn't work |
| 17:35.24 | dlynes_office | oh |
| 17:35.38 | dlynes_office | CrashHD: then maybe you didn't have the same problem i did |
| 17:35.57 | MikeJ__ | it hasn't been slashdotted yet.. we are working on load testing it first ;) |
| 17:36.02 | dlynes_office | CrashHD: does everything work if you don't make the call back to the originating asterisk server? |
| 17:36.02 | Qwell[] | "Welcome to movie^H^H^H^H^Hslashdot phone." |
| 17:36.07 | MikeJ__ | heh |
| 17:36.11 | CrashHD | dlynes_office: yes |
| 17:36.11 | MikeJ__ | indeed |
| 17:36.30 | CrashHD | what is odd |
| 17:36.30 | dlynes_office | CrashHD: but as soon as you add that call back in, it stops working properly? |
| 17:36.33 | Qwell[] | MikeJ__: lemme guess... $3.95/min chargeback? ;) |
| 17:36.33 | MikeJ__ | I think we have some other feeds on there too |
| 17:36.37 | MikeJ__ | no.. |
| 17:36.38 | philippel | hi all -looking for recommendations on good channel bank, need 20 FXS ports |
| 17:36.51 | MikeJ__ | normal call |
| 17:36.53 | Strom_C | philippel: Adtran Total Access 624 |
| 17:37.02 | Qwell[] | MikeJ__: kidding :) |
| 17:37.10 | philippel | Strom do you use some? |
| 17:37.17 | file | all your phones are belong to me |
| 17:37.18 | MikeJ__ | if I could get people to pay 4 buck a min for that.. I'd be rich |
| 17:37.27 | CrashHD | among the problem with the channels showing ringing when they are deffinitely answered is the 2nd asterisk is still transferring the call even though notransfer=yes is set in general and in the context (on both switches) |
| 17:37.27 | Strom_C | philippel: I use that at one of my client's locations. It is an awesome piece of hardware. |
| 17:37.28 | Qwell[] | MikeJ__: cepstral? |
| 17:37.32 | MikeJ__ | yeah |
| 17:37.38 | *** join/#asterisk pepepedo (i=ircap@200-42-84-111.cab.prima.net.ar) |
| 17:37.40 | generalhan | Is anyone in here using Fax2Email with * with good results ? |
| 17:37.45 | Qwell[] | Is the voice selectable? :p |
| 17:37.50 | MikeJ__ | yep |
| 17:37.51 | MikeJ__ | 5 |
| 17:37.51 | *** part/#asterisk assert_true (n=Sunil@59.176.26.247) |
| 17:37.52 | pepepedo | Hi |
| 17:37.58 | Qwell[] | nice |
| 17:38.00 | MikeJ__ | I only have 4 voices on there right now |
| 17:38.09 | MikeJ__ | and speed too |
| 17:38.13 | CrashHD | dlynes_office: if I call the aa directly in the system it works fine, if I dial and outbound number and it gets routed back it doesn't work |
| 17:38.19 | MikeJ__ | 2 and 8 make it speed up\slow down |
| 17:38.24 | philippel | Strom: how does it compare to Rhino products? |
| 17:38.25 | Qwell[] | heh |
| 17:38.47 | dlynes_office | what if you dial the extension in asterisk a directly from asterisk b? |
| 17:39.07 | Qwell[] | MikeJ__: nice cluecon ad :p |
| 17:39.07 | MikeJ__ | fun toys.... |
| 17:39.18 | MikeJ__ | hey.. what can ya do.. |
| 17:39.20 | Strom_C | philippel: I don't have any idea, since I've never used Rhino. The Adtran box is definitely a fantastic piece of hardware though; I wouldn't even consider using anything else |
| 17:39.30 | file | I have got to hear this ad |
| 17:39.32 | Qwell[] | sheesh...spammy |
| 17:39.33 | pepepedo | i nedd some help , i need some info where can i get meetme2 or app_cbmysql |
| 17:39.38 | [TK]D-Fender | generalhan : Works for me... |
| 17:39.41 | MikeJ__ | file, you already have |
| 17:39.47 | MikeJ__ | same one as last year |
| 17:39.50 | file | ah |
| 17:39.51 | [TK]D-Fender | Rhino = Set & forget. |
| 17:39.51 | pepepedo | because all the sites or down or 404 |
| 17:40.13 | pepepedo | somebody can help me pls? |
| 17:40.13 | MikeJ__ | fun toy at least.... |
| 17:40.20 | CrashHD | dlynes_office: if I use the /n with the local dial command it works perfectly |
| 17:40.32 | MikeJ__ | it's still not quite at the performance I want but... |
| 17:40.50 | MikeJ__ | cepstral guys are supposed to be profiling to help fix that :D |
| 17:40.52 | dlynes_office | CrashHD: /n? |
| 17:40.54 | philippel | D-Fender - so how many Rhino FXS banks do you have installed? |
| 17:40.54 | generalhan | [TK]D-Fender: i work for a law firm so faxes are vital and i cant afford to lose any. so i was thinking maybe using a regular fax machine for redundancy and still using Fax2Email for ease of use |
| 17:41.30 | MikeJ__ | generalhan, cisco gear can do good releiable faxing |
| 17:41.34 | MikeJ__ | but you pay for it. |
| 17:41.51 | pepepedo | someone knows what can i dot for connect meetme with mysql? |
| 17:41.59 | CrashHD | http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels |
| 17:42.01 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 17:42.02 | CrashHD | the /n thing |
| 17:42.05 | *** join/#asterisk slashlord (n=Jeff@digi29.ody.ca) |
| 17:42.05 | MikeJ__ | pepepedo huh? |
| 17:42.07 | tRSS | hey everyone |
| 17:42.08 | CrashHD | to not optimize |
| 17:42.14 | MikeJ__ | dot for connect? |
| 17:42.14 | CrashHD | and keep the local channel open during the call |
| 17:42.20 | pepepedo | i nedd admin meetme from a mysql |
| 17:42.25 | generalhan | see we have WinFax Pro (which worked great for us on our regular analog Qwest lines. but now on the asterisk server it freaks out a lot and hangs up on people |
| 17:42.28 | CrashHD | I guess it's the local equal to notransfer |
| 17:42.30 | MikeJ__ | like dynamic confs? |
| 17:42.41 | MikeJ__ | what do you want to change from mysql? |
| 17:42.44 | pepepedo | something like that |
| 17:42.45 | nortex | What are the possible reasons for fax problems when using a 4 port t-1 card to bridge between a channel bank with faxes and a T-1 with telco lines? |
| 17:42.59 | dlynes_office | CrashHD: not quite |
| 17:43.05 | pepepedo | but i would ike to pre assing to some users some rooms |
| 17:43.07 | dlynes_office | CrashHD: but yeah, it's probably what you want in there, anyways |
| 17:43.08 | MikeJ__ | pepepedo, just do dynamic confs and use dialplan mysql stuff to feed into the meetme options |
| 17:43.10 | tRSS | I am trying to configure realtime dynamic for my * box. I have the db, table and connection b/w * and mysql ready. but I can't see any peers when I do 'show sip peers' at the CLI. i have enabled rtcachefriends=yes in sip.conf. help would be appreicated |
| 17:43.37 | file | tRSS: sip show peer <name> load |
| 17:43.39 | nortex | generalhan, How do you lines from Quest come into asterisk? |
| 17:43.41 | tRSS | i meant sip show peers |
| 17:43.41 | file | tRSS: do that on the CLI |
| 17:43.42 | CrashHD | dlynes_office: so I guess with that left open it doesn't let it transfer the iax call....but again I'm confused as to why the notransfer=yes is not working |
| 17:43.44 | dlynes_office | CrashHD: another way that you might be able to solve this problem is by using iax2 trunking instead |
| 17:43.51 | generalhan | what if i write a script to record the incoming fax call ? then i could record the fax calls, send them to the fax machine then pipe the recorded audio thru the fax2email software ... that sound feasable ? |
| 17:43.53 | pepepedo | where can i get a good howto to do that? |
| 17:43.54 | Greek-Boy | if no one will download cisco firmware for me will someone atleast please tell me where I can purchase a service contract plan fast and easy? |
| 17:44.09 | MikeJ__ | generalhan, no |
| 17:44.09 | pepepedo | do u know? |
| 17:44.13 | MikeJ__ | fax is 2 way... |
| 17:44.27 | generalhan | hmm |
| 17:44.28 | philippel | Strom: how do you have your Adtran connected to asterisk? |
| 17:44.34 | CrashHD | dlynes_office: I hadn't thought about that...but deffinitely worth a shot |
| 17:44.38 | MikeJ__ | pepepedo, meetme's options, or the dialplan db stuff? |
| 17:44.43 | Strom_C | philippel: Asterisk box has a quad-span digium T1 card |
| 17:44.51 | dlynes_office | CrashHD: do you have static ips on both ends? |
| 17:44.59 | pepepedo | ok , thxs ... |
| 17:45.01 | generalhan | well i want to be sure that i have a hard copy incase the Fax2Email doesnt work for some reason. so i would want to send it to the fax machine first before the email software |
| 17:45.02 | Strom_C | philippel: two spans of the T1 card go to two channel banks |
| 17:45.04 | pepepedo | a lot |
| 17:45.05 | CrashHD | dlynes_office: ya |
| 17:45.08 | nortex | Greek-Boy, The SIP firmware was freely avaliable on Ciscos site last I checked. |
| 17:45.10 | Strom_C | third span goes to a PRI |
| 17:45.19 | dlynes_office | CrashHD: yeah...go with iax2 trunking, and hte switch statement then |
| 17:45.23 | MikeJ__ | generalhan, no way I can think of to do that |
| 17:45.26 | Strom_C | fourth span is open for ease of expanding station appearances in the future |
| 17:45.38 | tRSS | file: when you say <name>, what name are you referring to? the family name or the username created on the db server? |
| 17:45.39 | philippel | strom - ok, and you've been happy using the digium card for that or if you did it over would you choose a different T1 card? |
| 17:45.42 | dlynes_office | CrashHD: what that'll do is include the dialplans of the two asterisk boxes as one whole dialplan |
| 17:45.51 | Strom_C | philippel: perfectly happy with the digium card |
| 17:45.52 | CrashHD | dlynes_office: I was actually thinking about setting up dundi between all the switches...what do you think about that? |
| 17:45.54 | file | tRSS: the peer name |
| 17:45.59 | MikeJ__ | tRSS, he means the peer name |
| 17:46.00 | MikeJ__ | heh |
| 17:46.06 | dlynes_office | CrashHD: no idea...don't know anything about dundi |
| 17:46.07 | MikeJ__ | damn.. you beat me |
| 17:46.13 | file | MikeJ__: indeed, I r elite |
| 17:46.15 | tRSS | file: in that case, it didn't work |
| 17:46.24 | MikeJ__ | LIES |
| 17:46.28 | MikeJ__ | ;) |
| 17:46.29 | Greek-Boy | nortex; including the firmware for 7912? |
| 17:46.30 | AlexCTI | Hi, somoone knows how i disable the call waiting, so I have x-lite users and I set the "callwaiting = no" on sip.conf under the user context but the second call is still comming in, any ideas? |
| 17:46.35 | file | then your realtime is not setup correctly :) |
| 17:46.39 | *** join/#asterisk mmmmmToop (n=mmmmToop@firewall.datapro.co.za) |
| 17:46.39 | file | which is always a fun thing |
| 17:46.41 | generalhan | MikeJ__: if i have a recording of the fax and i called a different DID with that playback you dont think the Fax2Email software could talk to it ? |
| 17:46.42 | tRSS | file: found it, never mind |
| 17:46.43 | Greek-Boy | nortex; only 7960 seems to be free |
| 17:46.45 | MikeJ__ | I didn't think the 7912 had sip firmware |
| 17:46.50 | file | tRSS: excellent |
| 17:46.51 | [TK]D-Fender | generalhan : Good idea to leave critical faxes for a physically seperate line... |
| 17:46.52 | McLazarus | oej: FYI I get the same result trying t38 passthrough from trunk. That is: http://svn.digium.com/svn/asterisk-addons/trunk/ |
| 17:46.56 | Greek-Boy | MikeJ__ it does |
| 17:46.59 | McLazarus | oops |
| 17:47.00 | MikeJ__ | generalhan, fax is not just a 1 way thing.. it's like a modem |
| 17:47.03 | McLazarus | WARNING[16086]: chan_sip.c:4560 process_sdp: Unsupported SDP media type in offer: image 10072 udptl t38 |
| 17:47.07 | [TK]D-Fender | philippel : I run a single modular Rhino channel back FXS/FXO |
| 17:47.13 | file | McLazarus: I assume you have it enabled in the sip.conf? |
| 17:47.14 | MikeJ__ | if the other side isn't responding.. there is no fax |
| 17:47.19 | McLazarus | file: yep |
| 17:47.26 | nortex | Greek-Boy, My bad, I didn't see the 7912 model number. |
| 17:47.27 | CrashHD | is there a place where the channel options for iax2 are documented? |
| 17:47.40 | dlynes_office | ~book |
| 17:47.41 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:47.41 | Greek-Boy | nortex; np |
| 17:47.46 | file | T.38 is fun, because there's so many different ways to indicate it in SIP :D |
| 17:47.46 | dlynes_office | erm |
| 17:47.49 | dlynes_office | wrong one |
| 17:47.50 | Qwell[] | CrashHD: chan_iax2.c |
| 17:47.51 | dlynes_office | ~wiki |
| 17:47.53 | mmmmmToop | we have been struggeling with stability problems on the 1.2.9 branch in an inbound centre...anyone else having instability issues? |
| 17:47.58 | tRSS | file: alright, so I can see that the user is indeed loaded into asterisk. but this user is unable to register the softphone (xlite), although all other users from the flat file can register without a hitch. I have double and triple checked all the settings, but can't get it to work :( |
| 17:48.03 | dlynes_office | ~wikis |
| 17:48.04 | jbot | i heard wikis is http://www.voip-info.org |
| 17:48.08 | file | tRSS: I assume you mean peer? |
| 17:48.08 | dlynes_office | bleh |
| 17:48.09 | dlynes_office | ~docs |
| 17:48.11 | jbot | i guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:48.14 | MikeJ__ | T.38 is fun becuase the spec isn't speciffic enough for anything to be compatible |
| 17:48.16 | CrashHD | dlynes_office: I'd like to be able to just use a channel option so I do not need to do the local work around (with thousands of DIDS it will be a lot of work to duplicate and add the local stuff), any thoughts? |
| 17:48.19 | MikeJ__ | whee jbot... |
| 17:48.23 | MikeJ__ | ~cluecon |
| 17:48.24 | jbot | extra, extra, read all about it, cluecon is http://www.cluecon.com - The Open Source Telephony Expo and Developers Conference featuring presentations by Jim Van Meggelen, Greg Boehnlein, Ken Rice, Brian West, Craig Southeren, Derek Smithies, Kevin Lenzo, RJ Auburn, Nenad Corbic, David Sugar, Peter Nixon, and Anthony Minessale II. |
| 17:48.25 | dlynes_office | CrashHD: See the one on handbook-draft? |
| 17:48.27 | Qwell[] | ~spam |
| 17:48.28 | jbot | ACTION sings, Spam, Spam, Spam, Spam, Spam, Wonderfull spam! |
| 17:48.34 | MikeJ__ | you people are jbot happy |
| 17:48.43 | CrashHD | handbook draft |
| 17:48.44 | CrashHD | got it |
| 17:48.45 | generalhan | ~generalhan |
| 17:48.47 | jbot | somebody said generalhan was THE MAN |
| 17:48.47 | file | I'm just generally happy |
| 17:48.47 | MikeJ__ | ;) |
| 17:48.54 | CrashHD | ~crashhd |
| 17:48.56 | file | jbot: file? |
| 17:48.58 | jbot | well, file is a canadian that wants to be a russian. |
| 17:48.58 | mmmmmToop | wierd activity...like Asterisk droping zap channels just before going into a queue...? |
| 17:48.58 | MikeJ__ | ~MikeJ[Laptop] |
| 17:49.01 | dlynes_office | CrashHD: yeah...that one is the original book on iax2 |
| 17:49.05 | generalhan | HAhahaha |
| 17:49.10 | dlynes_office | CrashHD: but the problem with it, is that it's severely outdated |
| 17:49.23 | file | tRSS: I assume you mean peer? |
| 17:49.27 | dlynes_office | CrashHD: so you'll have to use a combination of that, and the sample iax2.conf file that comes with asterisk |
| 17:49.30 | MikeJ[Laptop] | .msg nickserv identify shhhhhh |
| 17:49.37 | *** join/#asterisk pengyong (n=lala@218.93.158.125) |
| 17:49.43 | MikeJ[Laptop] | hehe.. |
| 17:49.43 | tRSS | file: sorry, thats what I meant, my bad. I meant peer |
| 17:49.52 | MikeJ[Laptop] | that's not really my password :P |
| 17:49.53 | CrashHD | dlynes_office: ya there are no options in that book, at least not where they should be |
| 17:49.57 | file | tRSS: pastebin the CLI output when they try to register |
| 17:50.02 | CrashHD | dlynes_office: I'll take a look at the sample conf |
| 17:50.06 | file | plus a sip show peer of the peer |
| 17:50.16 | MikeJ[Laptop] | unlink file |
| 17:50.16 | CrashHD | dlynes_office: I just need an option to force to stay in media path |
| 17:50.24 | tRSS | file: give me a minute |
| 17:50.27 | file | Windows updates, always a part of a healthy breakfast! |
| 17:50.33 | MikeJ[Laptop] | heh |
| 17:50.49 | MikeJ[Laptop] | off to make a tarball! |
| 17:50.55 | MikeJ[Laptop] | time to make the donuts |
| 17:51.10 | file | MikeJ[Laptop]: may I have one? |
| 17:51.24 | dlynes_office | CrashHD: yeah, but if you set up iax2 trunking, you might not need to stay in the media path |
| 17:51.41 | Greek-Boy | so no one here has access to cisco smartnet? |
| 17:51.58 | dlynes_office | CrashHD: it just wasn't an option for me for one customer because that customer was on a dynamic ip |
| 17:52.06 | MikeJ[Laptop] | file, no.. you have been unlink'd |
| 17:52.07 | CrashHD | dlynes_office: I'm scared about the timing source, the main trunking box has an e1000 card in it with a digium pri card...I was told that was a big no no |
| 17:52.30 | dlynes_office | CrashHD: it's a big no-no because the two usually share the same interrupt |
| 17:52.56 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:53.03 | CrashHD | and I believe that would be my clock source with iax2 trunking, correct? |
| 17:53.12 | CrashHD | wouldn't that cause an issue |
| 17:53.18 | mmmmmToop | u need to check though it is board specefic cat /proc/interupts |
| 17:53.19 | dlynes_office | CrashHD: a motherboard that has e1000 card (two 1Gb ports), usually has the pci slot where the digium card needs to go sharing the same interrupt as the nic |
| 17:53.31 | mmmmmToop | or lspci -vv |
| 17:53.38 | dlynes_office | CrashHD: it's not always the case, but 99% of the time it is the case |
| 17:53.54 | CrashHD | so the pri card needs to go in a specific slot? |
| 17:54.01 | CrashHD | lol I know nothing |
| 17:54.05 | dlynes_office | CrashHD: you're going to notice issues with your pri communications long before you'll notice iax2 timing issues |
| 17:54.18 | dlynes_office | CrashHD: such as dropped calls, crappy call quality, ... |
| 17:54.21 | CrashHD | pri was already a problem |
| 17:54.22 | mmmmmToop | nice test for pri is to run zttest |
| 17:54.34 | dlynes_office | mmmmmToop: and patlooptest |
| 17:54.35 | CrashHD | zttest is 99.98 -> 100.00 solid |
| 17:54.35 | mmmmmToop | if you drop below 99.987 or something u need to be worried |
| 17:54.50 | mmmmmToop | mmmm.... |
| 17:54.56 | dlynes_office | CrashHD: yeah...99.98 is too low |
| 17:55.05 | dlynes_office | CrashHD: it's below 99.987 |
| 17:55.06 | MikeJ[Laptop] | :( |
| 17:55.37 | mmmmmToop | & your interupts...r they clean? |
| 17:55.42 | CrashHD | 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% |
| 17:55.46 | liran_ | the dial plans (or what i call "ini sections") in extensions.conf describe what happens to a call wen its inbound or outbound? (or both?) |
| 17:55.57 | dlynes_office | CrashHD: did you let it run for three minutes or so? |
| 17:55.58 | Qwell[] | liran_: depends |
| 17:56.28 | CrashHD | digium card is using irq 217 |
| 17:56.30 | CrashHD | lol |
| 17:56.32 | CrashHD | 217? |
| 17:56.38 | Qwell[] | apic |
| 17:56.43 | Dandan | dlynes_office: ok, i did pri debug span 1 |
| 17:56.48 | Dandan | and i have some debug... :) |
| 17:56.51 | CrashHD | 185 is the eth0 controller |
| 17:57.07 | liran_ | Qwell[]: exten => 1000,1,Dial(SIP/1000,40,t) , exten => 1000,2,Hangup -> can you explain me this setup? |
| 17:57.16 | mmmmmToop | can anyone recommend a stable branch/ release for queue functionallity? ...we are really struggeling with latest 1.2.9 branch |
| 17:57.18 | dlynes_office | Dandan: what part of do not know enough about it to help you, do you not understand? :P |
| 17:57.34 | Qwell[] | liran_: "inbound" and "outbound" are very fine, wavy lines |
| 17:57.41 | Qwell[] | easily crossed |
| 17:57.56 | dlynes_office | Dandan: i'm not really a telecom guy...i'm a computer guy |
| 17:57.57 | Dandan | lol! gimmie break, i have been struggling with that sh*t^H^H^Htuff since 6 am this morning :) |
| 17:58.03 | Dandan | dlynes_office: me too :) |
| 17:58.16 | CrashHD | so is apic ok to run? and does the pri card need to be in a specific slot? |
| 17:58.30 | Dandan | CrashHD: i have no probs with my sangoma |
| 17:58.31 | Dandan | and apic |
| 17:58.33 | tRSS | file: never mind, got it fixed. seems that xlite had an issue. tried another softphone (SjPhone) and it worked. surprising. but i got this error as soon as the peer registered on the CLI: 'MySQL RealTime: Failed to query database. Check debug for more info.' |
| 17:58.38 | Dandan | and i have a dell Server |
| 17:58.39 | dlynes_office | CrashHD: apic is probably the only way you're going to get that card to work in your machine properly |
| 17:58.51 | dlynes_office | CrashHD: otherwise you're probably going to have interrupt sharing issues |
| 17:58.59 | Dandan | i got a nice irq (21) and i am happy :) |
| 17:59.12 | liran_ | Qwell[]: what i mean is, does the "exten =>" configs define what happens to a call when it is on it's way to a sip client that registered to asterisk from another sip or when its outbound, when a sip client that registered to asterisk initiated? |
| 17:59.34 | CrashHD | zttest is > 99.987 |
| 17:59.38 | CrashHD | hmm |
| 17:59.45 | CrashHD | I'm not lucky enough to be the 1% |
| 17:59.49 | CrashHD | I'm probably missing something |
| 18:00.11 | dlynes_office | you mean unlucky enough? |
| 18:00.24 | dlynes_office | nvm |
| 18:00.26 | CrashHD | no, you said 1% have no issues |
| 18:00.27 | Dandan | CrashHD: what card is it? |
| 18:00.33 | CrashHD | the 4 port digium |
| 18:00.37 | dlynes_office | CrashHD: yeah...it might evne be less than 1% |
| 18:00.37 | Dandan | i had problems with voicetronix board! YUCK!!!! |
| 18:00.55 | CrashHD | dlynes_office: should the card be a specific pci slot? |
| 18:00.58 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:01.07 | dlynes_office | CrashHD: it should be in a slot where you're not sharing irqs |
| 18:01.17 | CrashHD | but other than that, no cares? |
| 18:01.17 | flujan | guys, I'm trying to configure asterisk to work with pri and got this error: http://pastebin.com/708987 |
| 18:01.41 | dlynes_office | CrashHD: ideally with the digium cards, you want acpi disabled, apic disabled, apm disabled, dma enabled on your hard drives, no sharing of interrupts |
| 18:02.09 | *** join/#asterisk mflorell (n=astmattf@www2.vicimarketing.com) |
| 18:02.49 | CrashHD | no more digium cards for me |
| 18:03.00 | CrashHD | everyone seems to have it easier with sangnoma |
| 18:03.21 | asterboy | except when they ship you dead cards and waste your day |
| 18:03.31 | CrashHD | lol |
| 18:03.41 | CrashHD | fair enough |
| 18:03.49 | flujan | I don't know why I'm having such message |
| 18:03.52 | dlynes_office | CrashHD: asterboy hasn't had good luck with sangoma :p |
| 18:04.01 | CrashHD | he is the minority |
| 18:04.04 | asterboy | we'll see how their RMA process stands up, still waiting for them to call back. |
| 18:04.07 | [TK]D-Fender | dlynes_office : You do know the e1000 = big trouble for Digium cards right? |
| 18:04.09 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 18:04.09 | flujan | when I plug it in a proprietaru pbx it works. |
| 18:04.21 | asterboy | Not sure why they have an IRC channel either |
| 18:04.26 | asterboy | no one is there. |
| 18:04.30 | flujan | [TK]D-Fender, any idea? http://pastebin.com/708987 |
| 18:04.32 | dlynes_office | [TK]D-Fender: yeah...i should know...i've got a system with a four port digium pri card with an e1000 card that's caused me no end of headaches |
| 18:04.36 | *** join/#asterisk wunderkin (n=wunderki@69.26.192.234) |
| 18:05.22 | flujan | here goes my zapata.conf file: http://pastebin.com/708996 |
| 18:05.25 | dlynes_office | [TK]D-Fender: but unfortunately, i had an idiot asterisk admin working for us before i took over that insisted on getting the digium cards that would only work in a small number of pci slots (it's the rarer voltage configuration) |
| 18:05.34 | gmfm | flujan, could it be that you are trying to send an 8 digit number beginning with 9 to the pstn? |
| 18:05.35 | [TK]D-Fender | flujan : Where are you located? |
| 18:05.53 | dlynes_office | [TK]D-Fender: brazil |
| 18:05.54 | flujan | [TK]D-Fender, Brazil. |
| 18:06.07 | [TK]D-Fender | flujan : And 8 digit #'s are legit there? |
| 18:06.44 | nortex | asterboy, I finally got done on the Phone, but the Sangoma tech could not get my faxes straight either. |
| 18:07.02 | [TK]D-Fender | nortex : A200 ot T1? |
| 18:07.09 | flujan | [TK]D-Fender, yes... our phone numbers have 8 digits... |
| 18:07.46 | [TK]D-Fender | nortex : A200 does not do faxing well ATM.... a new driver release is supposed to help that |
| 18:07.46 | nortex | T1 a104D |
| 18:07.46 | flujan | [TK]D-Fender, I already tried a lot of possibilities... I now that the protocol is euroisdn |
| 18:07.46 | CrashHD | dlynes_office: I'm just going to use the /t on the dial command at the 2nd asterisk box which will keep the media path open |
| 18:07.46 | flujan | [TK]D-Fender, and the signalling is pri. |
| 18:07.49 | nortex | It completely stumped Sangoma as to why it does not work |
| 18:07.51 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 18:08.07 | mmmmmToop | any thoughts on a stable Asterisk for inbound applications? |
| 18:08.25 | nortex | No signalling is FXSks, its a channelized T-1, the new PRI comes in tomorrow I hope. |
| 18:08.34 | mmmmmToop | anyone got a live inbound call centre running here? |
| 18:08.52 | dlynes_office | mmmmmToop: lots of peeps |
| 18:09.03 | generalhan | mmmmmToop: i dont have 100 users ... but i have about 40 on inbound calls |
| 18:09.14 | asterboy | nortex, someone else said that fax stuff and sangoma are not married yet. |
| 18:09.22 | asterboy | iirc it was that TK guy |
| 18:09.28 | dlynes_office | asterboy: that's for spandsp |
| 18:09.32 | mmmmmToop | generalhan: what version of Asterisk are you running? |
| 18:09.32 | asterboy | a |
| 18:09.37 | generalhan | 1.2.9.1 |
| 18:09.43 | flujan | gmfm, yes, I'm trying to send a 8 digits number beginning with a 9 but all number send the same message |
| 18:09.45 | dlynes_office | asterboy: if you're doing fxs->fxo bridging, i'm guessing it should "just work" |
| 18:09.57 | dlynes_office | asterboy: i'm getting a box set up for testing just that |
| 18:10.00 | mmmmmToop | are you using app_queues.c ...& AgentLogin...with staticly defined agents?\ |
| 18:10.11 | asterboy | would be good to know. |
| 18:10.17 | dlynes_office | asterboy: bought a sangoma a200 with an fxo module and an fxs module for testing just that |
| 18:10.34 | flujan | [TK]D-Fender, usually, why asterisk shows a congestion message? |
| 18:10.35 | generalhan | mmmmmToop: nope ... they are all stationed somewhere specific .. i have no need for agentlogin ... i just use call queues. |
| 18:10.59 | mmmmmToop | mmmm....maybe that is our problem... |
| 18:11.13 | generalhan | mmmmmToop: what IS your problem ? |
| 18:11.18 | mmmmmToop | we are running same version as you...but it dies at least once a day with AgentLogin... |
| 18:11.34 | mmmmmToop | I wish we knew! ; ) |
| 18:11.51 | dlynes_office | mmmmmToop: do you get an error in your log file? |
| 18:12.01 | generalhan | mmmmmToop: do you NEED agentlogin ?? i dont use it ... but not because it doesnt work ,,, ive heard tons of people talk about it in here with good results |
| 18:12.12 | generalhan | mmmmmToop: so if you need it .. dont scrap it cause i dont use it |
| 18:12.17 | *** part/#asterisk C (i=ix@c-24-60-193-83.hsd1.ma.comcast.net) |
| 18:12.22 | mmmmmToop | sure... |
| 18:12.46 | dlynes_office | mmmmmToop: can you pastebin a tail of your full log file from the time when your system dies? |
| 18:13.10 | nortex | asterboy, the tech said once I shutdown EC for the channels it should work fine. |
| 18:14.20 | [TK]D-Fender | nortex : pastebin yout zapata.conf and wanpipe config for that port. |
| 18:14.23 | mmmmmToop | we didn't have loggin on when it died today... : ( ...no usefull logs |
| 18:14.32 | *** join/#asterisk redondos_ (n=redondos@190.48.33.73) |
| 18:14.53 | dlynes_office | mmmmmToop: make sure you have full logging on |
| 18:14.55 | [TK]D-Fender | flujan : pastebin a "pri show span 1" |
| 18:15.02 | dlynes_office | mmmmmToop: and when you get it happening again |
| 18:15.12 | [TK]D-Fender | flujan : followed by "zap show channels" |
| 18:15.16 | mmmmmToop | agreed....will do so... |
| 18:15.18 | dlynes_office | mmmmmToop: make a backup of the log file |
| 18:15.18 | nortex | ~pastebin |
| 18:15.20 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 18:15.21 | [TK]D-Fender | ~pb |
| 18:15.22 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 18:15.34 | dlynes_office | mmmmmToop: that way when someone asks for it, you can produce it |
| 18:15.38 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:15.52 | dlynes_office | mmmmmToop: also, do you have a core dump? |
| 18:15.54 | mmmmmToop | we had 4Gigs of logs the other day...so we turned it off ; ) |
| 18:16.30 | stephane_ | soir |
| 18:16.33 | mmmmmToop | how do we get a core dump...? ...no wait let me google... |
| 18:16.49 | asterboy | nortex, is that hardware ec or software? |
| 18:17.00 | dlynes_office | mmmmmToop: it's a function of your shell allowing you to disable or enable core dumps |
| 18:17.20 | *** join/#asterisk jaike (i=jaike@210.5.116.108) |
| 18:17.29 | *** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
| 18:17.32 | [TK]D-Fender | asterboy : HWEC |
| 18:17.35 | dlynes_office | mmmmmToop: normally, I think core dumps are enabled in asterisk if you use safe_asterisk to start it |
| 18:17.38 | [TK]D-Fender | asterboy : Its the A104d |
| 18:17.46 | jaike | anyone experience seg faults with mixmonitor? |
| 18:17.55 | dlynes_office | jaike: yeah, on older versions |
| 18:18.13 | flujan | [TK]D-Fender, http://pastebin.com/709021 |
| 18:18.14 | jaike | dlynes: still do on 1.2.9.1 |
| 18:18.18 | mmmmmToop | ok...got it...i c ...safe_asterisk runs a: ulimit -c unlimited |
| 18:18.27 | flujan | [TK]D-Fender, as u can see I have no debug in the interface. |
| 18:18.52 | jaike | but its like one in every 3000 calls |
| 18:19.12 | mmmmmToop | where do you get the core dump then after safe_asterisk issues a ulimit -c unlimited ....? |
| 18:20.05 | dlynes_office | mmmmmToop: probably in whatever directory you ran safe_asterisk from |
| 18:20.55 | dlynes_office | jaike: ah...i don't use mix monitor that much |
| 18:21.09 | mmmmmToop | cool...thanks dlynes_office |
| 18:21.10 | dlynes_office | jaike: so it'll probably be a while before I get the system crashing on me |
| 18:21.20 | dlynes_office | mmmmmToop: did you find a core dump then? |
| 18:21.58 | jaike | dlynes: we make/receive like 10000 a day..so thats 3 crashes per day |
| 18:22.03 | tRSS | where are the debug logs saved from asterisk? |
| 18:22.05 | jaike | no crashes with monitor |
| 18:22.08 | dlynes_office | jaike: damn |
| 18:22.11 | [TK]D-Fender | flujan : Where is the "pri show span 1" ? |
| 18:22.14 | mmmmmToop | no...we weren't running safe_asterisk : ( |
| 18:22.14 | dlynes_office | jaike: i'd use monitor, then |
| 18:22.23 | dlynes_office | mmmmmToop: how were you running it then? |
| 18:22.58 | mmmmmToop | dylnes_office: asterisk ...that is all... last time we do that! ; ) |
| 18:23.06 | tRSS | where are the debug logs saved from asterisk? |
| 18:23.11 | jaike | dlynes: my servers starting to run out of resources..have to decide if going SMP will solve it |
| 18:23.25 | dlynes_office | mmmmmToop: damn...how are you going to use restart when convenient or anything like that, then? |
| 18:23.40 | flujan | [TK]D-Fender, http://pastebin.com/709036 |
| 18:23.55 | *** join/#asterisk littlejohn (n=little@host77-73.pool8717.interbusiness.it) |
| 18:24.00 | flujan | sorry, I did not notice it wasn't pasted. |
| 18:24.18 | jaike | am wondering which is preferred for asterisk, intel or amd dual core |
| 18:25.20 | ManxPower | most of our Asterisk servers have motherboards that can do dual processors, but only have 1 processor -- so we can add a CPU later if needed. |
| 18:25.54 | jaike | manxpower: intel? |
| 18:25.58 | *** join/#asterisk C4T3l (n=robert@216.54.143.2) |
| 18:26.12 | ManxPower | jaike, yes, but that's only because we only use intel for servers |
| 18:26.16 | mmmmmToop | dlynes_office: we have never had instability issues with Asterisk before...so we never really though of using safe_asterisk... |
| 18:26.20 | jaike | need advice on this area...dual cores/processors are expensive |
| 18:27.07 | mmmmmToop | we have some boxes with masive uptimes...only the latest pain is causing us to have masive rethink about safe_asterisk ; ) |
| 18:27.32 | C4T3l | hello, has anyone ever had thier asterisk server repeatedly state that Primary D-channel on span 1 is up? |
| 18:27.36 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 18:27.43 | dlynes_office | It's called 'safe'_asterisk for a reason :) |
| 18:27.53 | tamp4x | it wears a condom |
| 18:27.56 | mpruett | Anyone: Is it possible to install asterisk without Zaptel and use a different timing device outher than ztdummy? |
| 18:28.08 | Qwell[] | tamp4x: That would be safe_asstricks |
| 18:28.11 | tamp4x | anyone here have an e.164 internal formatted number? |
| 18:28.20 | tamp4x | international |
| 18:28.33 | dlynes_office | mpruett: there's a usb timing driver, also |
| 18:28.39 | *** join/#asterisk catlee (n=catlee@Z-pc1-959-S1.gw2.tor1.rogerstelecom.net) |
| 18:28.43 | catlee | Good afternoon |
| 18:28.54 | dlynes_office | mpruett: but again, it uses zaptel as the base |
| 18:29.01 | C4T3l | i just got a sangoma card up and running and that message keeps repeating |
| 18:29.13 | jaike | mpruett: whats wrong with ztdummy? |
| 18:29.34 | flujan | [TK]D-Fender, any idea? |
| 18:29.59 | [TK]D-Fender | flujan : there's your answer "Provisioned, Down, Active" <- no Dchannel |
| 18:30.10 | bernardovieira | does anyone know how to adjust the sample size for a sip channel? |
| 18:30.13 | [TK]D-Fender | flujan : pastebin your zaptel.conf as well now |
| 18:30.15 | mpruett | I am trying to install asterisk at a hosted site on a VM - They will not give me access to Kernal source so I can't compile Zaptel |
| 18:30.22 | flujan | [TK]D-Fender, this is saying I have no d-channel? |
| 18:30.42 | [TK]D-Fender | flujan : Pretty much.... |
| 18:30.51 | [TK]D-Fender | flujan : Zaptel.conf please.... |
| 18:31.09 | C4T3l | i just got a sangoma card up and running and that message keeps repeating over and over and over |
| 18:31.30 | jaike | mpruett: bummer..but what do you need ztdummy for? meetme? |
| 18:31.52 | flujan | [TK]D-Fender, http://pastebin.com/709057 |
| 18:32.02 | mpruett | jaike: Yes |
| 18:32.02 | flujan | [TK]D-Fender, thanks for the help!!!! |
| 18:32.25 | jaike | mpruett: your in a bind. get a dedicated server :) |
| 18:32.31 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 18:32.32 | C4T3l | hello, has anyone ever had thier asterisk server repeatedly state that Primary D-channel on span 1 is up? |
| 18:32.38 | C4T3l | i just got a sangoma card up and running and that message keeps repeating |
| 18:32.54 | nortex | [TK]D-Fender, Sorry got dropped. |
| 18:32.55 | catlee | I've been trying to figure out how to set up * so that users can have roaming extensions. What I've come up with is this...Each SIP phone has a unique ext., but it's not really used by users. When a person wants to transfer his ext. to a phone, he calls some number, which uses AgentCallbackLogin to log him in, and then AddQueueMember to add him to a call queue |
| 18:32.56 | mpruett | jaike: Yeah that is what they suggested - just thought I would see if there was an alternative |
| 18:33.02 | ManxPower | C4T3l, No, only when the PRI line was having problems. |
| 18:33.15 | [TK]D-Fender | flujan : Should be span=1,1,0,....... |
| 18:33.18 | catlee | In the dial plan, the user's extension transfers a call to his queue |
| 18:33.28 | [TK]D-Fender | flujan : SHould be used for timing... could be its due to lack of synch... |
| 18:33.36 | nortex | http://pastebin.com/709033 has the wanpip configs and my zaptel.conf |
| 18:34.02 | C4T3l | ManxPower: I need to get one more span up! |
| 18:34.04 | mpruett | jaike/anyone: SO just to be sure - If I want to use meetme Zaptel needs to be installed |
| 18:34.18 | ManxPower | C4T3l, how many PRIs do you have? |
| 18:34.19 | flujan | [TK]D-Fender, changed... I will reboot |
| 18:34.23 | flujan | be back soon. |
| 18:34.24 | flujan | :) |
| 18:34.33 | C4T3l | ManxPower: 4 |
| 18:34.49 | ManxPower | so only span 1 is giving that message over and over? |
| 18:34.54 | [TK]D-Fender | nortex : Hmmm, looks fine... |
| 18:35.14 | C4T3l | ManxPower: I only have one up at the moment, the other three are down,active |
| 18:35.16 | [TK]D-Fender | mpruett : Yes. You need a timing source. |
| 18:35.44 | [TK]D-Fender | mpruett : And you don't need to recompile the Kernel, just compile a kernel module.... |
| 18:35.54 | ManxPower | C4T3l, The only time I have seen that issue is when the telco had a bad card in their switch |
| 18:35.58 | *** part/#asterisk InfraRed (n=subhi@arpa-addr.in) |
| 18:36.41 | nortex | [TK]D-Fender, I did find one thing that you might have a quick answer to. Our old system had an Adtran TA 850 connected to this T-1 and the timing was set to loop, not local or remote. Is there a similar setting in Zaptel.conf? |
| 18:36.50 | C4T3l | ManxPower: cool, this is my first time using a PRI card. didn't think it would be such a pain |
| 18:37.07 | ManxPower | C4T3l, it's not. |
| 18:37.28 | Cresl1n | yeah, most of the time it's pretty easy |
| 18:37.30 | ManxPower | Since you are using Sangoma, you could have a Sangoma specific issue. Call them. Their cards come with support. |
| 18:37.53 | C4T3l | ManxPower: Cool, will do. Thanks for the info |
| 18:39.27 | [TK]D-Fender | nortex : they're sending you an analog T1, not PRI? |
| 18:40.01 | nortex | Currently yes. We have ordered a PRI, but it is not in yet. |
| 18:40.24 | [TK]D-Fender | nortex : Ok, well the rest of your setup looks kosher... dunno what dto say from here.... |
| 18:40.36 | [TK]D-Fender | nortex : pastebin an "ifconfig" |
| 18:40.42 | catlee | as far as I can tell this means that for each user I need an entry in agents.conf, and their own queue defined in queue.conf...and I don't know how voicemail would work :) |
| 18:41.07 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 18:41.24 | [TK]D-Fender | catlee : You don't want to make a system of 1 queue / person...... |
| 18:41.29 | catlee | plus an entry per phone in sip.conf |
| 18:41.38 | variable_office | when i do zap show status i get "Alarms Red" for my Gerneric Clone Board 1 |
| 18:41.42 | variable_office | what does that mean? |
| 18:41.44 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 18:42.01 | [TK]D-Fender | catlee : All calls would be queue calls and the setup for VM would be FUGLY at best.... |
| 18:42.06 | catlee | [TK]D-Fender: you're right, I don't :) |
| 18:42.07 | [TK]D-Fender | catlee : not the way to work this. |
| 18:42.10 | ManxPower | variable_office, that means you don't have a phone line plugged into the card |
| 18:42.11 | boch | could anyone helpme with php agi? |
| 18:42.23 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:42.25 | nortex | [TK]D-Fender, http://pastebin.com/709085 |
| 18:42.28 | variable_office | ManxPower oh, but it sees it then right |
| 18:42.29 | variable_office | ? |
| 18:42.39 | flujan | [TK]D-Fender, no success... Even with the changes in the zaptel.conf and a reboot... |
| 18:42.42 | ManxPower | variable_office, it sees the card. |
| 18:42.58 | flujan | [TK]D-Fender, pri debug span 1 is not returning any information. :( |
| 18:43.00 | ManxPower | You know that people that use clone cards go to hell, right? |
| 18:43.22 | variable_office | ManxPower its just a test setup before i get a pri |
| 18:43.46 | variable_office | ManxPower thats good, i am just trying to figure out why the system says 0 channels configured when i run ztcfg -vvvv ? |
| 18:44.07 | ManxPower | variable_office, put your /etc/zaptel.conf on pastebin.ca |
| 18:44.11 | dlynes_office | variable_office: becuase you have 0 channels configured |
| 18:44.40 | variable_office | ok, just a second |
| 18:44.48 | boch | the method get_data() from PHP AGI class is not waiting for the ms i specified, do you know why ? |
| 18:44.53 | mpruett | Fender/Jaike: appreciate the help - thanks |
| 18:44.58 | flujan | [TK]D-Fender, the pri show span 1 are still the same |
| 18:45.34 | variable_office | ManxPower http://pastebin.ca/65568 |
| 18:46.40 | ManxPower | variable_office, you don't have any channels configured. you have trunkgroups configured. |
| 18:46.49 | mpruett | Fender: Any chance of compiling Zaptel on different server with same OS and just moving the files I need over to server? |
| 18:47.00 | catlee | would Dial(Agent/agentId) work? |
| 18:47.21 | ManxPower | variable_office, remove the first line, try again |
| 18:47.38 | jaike | mpruett: am also on the lookout for a good hosting service for asterisk |
| 18:47.53 | variable_office | ManxPower remove the [trunkgroups] ? |
| 18:48.03 | [TK]D-Fender | nortex : Looks clean... I'm stumped |
| 18:48.20 | [TK]D-Fender | flujan : I said "pri show span 1" |
| 18:48.21 | ManxPower | variable_office, correct. that is not a valid option i zaptel.conf, only in /etc/asterisk/zapata.conf |
| 18:49.06 | flujan | [TK]D-Fender, sorry... pri show span 1 is the same... no changes... :( |
| 18:49.14 | flujan | [TK]D-Fender, could it be a hardware problem? |
| 18:49.19 | flujan | [TK]D-Fender, my digum card... |
| 18:49.36 | nortex | [TK]D-Fender, Any chance the PRI magicly fixes the problems? |
| 18:49.44 | [TK]D-Fender | flujan : "cat /proc/interrupts" |
| 18:50.11 | [TK]D-Fender | nortex : Only if you promise your first-born to the PRI-Fairie |
| 18:50.23 | ManxPower | That would be me. |
| 18:51.06 | flujan | [TK]D-Fender, http://pastebin.com/709104 |
| 18:51.06 | [TK]D-Fender | note to self : "ManxPower=Faerie" |
| 18:51.17 | variable_office | ManxPower still 0 channels configured, is there another file i need to edit besides zaptel.conf ? |
| 18:51.41 | [TK]D-Fender | flujan : Could be your card... I had 2 TE405P's that simply would not clock and wobbled on synch.... |
| 18:51.56 | *** join/#asterisk \lart (i=nunya@neo.jasons.org) |
| 18:52.09 | nortex | [TK]D-Fender, Well, no luck there. Thanks for looking it over, I'm off to get a bite to eat and tackle it again later. |
| 18:52.10 | variable_office | ManxPower to note, i did not have any zaptel.conf when i started, i created it from blank, does that matter? |
| 18:52.46 | [TK]D-Fender | \lart : Just missed from last night, but on confirmation of your list of needs, the IP 501 is for you.... $170 well spent. |
| 18:52.48 | flujan | [TK]D-Fender, I will call the card vendor... :) |
| 18:53.05 | flujan | [TK]D-Fender, just to clean my conscience... please, take a look http://pastebin.com/709109 |
| 18:53.10 | flujan | [TK]D-Fender, extensions.conf :) |
| 18:53.33 | \lart | Thanks.. Just got a call last night from a buddy that works for a company that got scooped up by cisco a few months back.. He's sending me a "spare" phone. :) |
| 18:53.45 | \lart | So, $0 is decidedly better. :) |
| 18:53.57 | [TK]D-Fender | flujan : No need... I saw the CLI output... if you say that number is cool, hey, thats all there is... but your problem is the link is DOWN. |
| 18:54.16 | [TK]D-Fender | \lart : argv[-1] |
| 18:54.20 | flujan | [TK]D-Fender, maybe the cable? |
| 18:54.31 | [TK]D-Fender | flujan : Possible.... try another |
| 18:54.42 | flujan | [TK]D-Fender, ok... thanks... :) |
| 18:54.57 | jsolares | flujan: have you tried changing pri_net for pri_cpe and viceversa? |
| 18:55.33 | vader-- | ok so friday they are going to bring my T1 line up for testing |
| 18:55.44 | \lart | so I just had a conversation from the train. My boss got us those verizon ev-do cards.. I vpn into my house, and used x-lite with a headset to make a call. the stuff actually worked. Color me shocked. |
| 18:55.59 | *** part/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 18:56.17 | ManxPower | variable_office, not for ztcfg -vvv |
| 18:56.18 | [TK]D-Fender | \lart : Careful on bandwidth charges.... |
| 18:56.26 | Qwell[] | \lart: no "shocked", sorry |
| 18:56.42 | jaike | can anyone suggest a good asterisk hosting site? |
| 18:57.03 | *** join/#asterisk Wowzers10 (n=pbaker@nnat-gw.adeptra.com) |
| 18:57.22 | Wowzers10 | hello all, does anyone know the time format for the API return of AgentLoggedInTime |
| 18:57.48 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:58.18 | flujan | jsolares, yes... no success at all... I will call the card vendor... Since first, he sold the card as a T1, and without the jumpers... :) now this. |
| 18:58.23 | Wowzers10 | IE if I were to log an agent in now, the time would reflect : 1150311613 |
| 18:58.25 | flujan | [TK]D-Fender, thanks for the help. |
| 18:58.28 | *** join/#asterisk pjo (n=pjo@212.88.98.114) |
| 18:58.37 | flujan | [TK]D-Fender, I'm using a straight T1 cable for this... |
| 18:58.45 | Wowzers10 | I assume thats epoch |
| 19:05.13 | *** part/#asterisk jaike (i=jaike@210.5.116.108) |
| 19:05.17 | pjo | i've successfully setup my local asterisk server to call out using the coprate asterisk box. however, if I try to call extension 100 on my local box (from the coporate phones) my machine says "Rejected connect attempt from a.b.c.d who was trying to reach 100@ ... CAUSE : No authority found". any pointers on what could be missing in my local iax.conf? |
| 19:06.02 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 19:06.29 | asterboy | So far not too impressed with Sangoma tech support...seems like poor David is the only guy for the whole world. |
| 19:06.47 | asterboy | Still no call back from Sangoma on my RMA. |
| 19:06.49 | asterboy | :( |
| 19:06.49 | tamp4x | anyone here can call a US # from an international #? |
| 19:06.56 | asterboy | yep |
| 19:07.02 | asterboy | pm me |
| 19:07.18 | [TK]D-Fender | asterboy : Ask for Nenad Korvic specifically. |
| 19:07.29 | asterboy | ah, ok...that's good to know. |
| 19:08.39 | file | it's Corbic |
| 19:10.09 | boch | the method get_data() from PHP AGI class is not waiting for the ms i specified, do you know why ? |
| 19:10.48 | *** join/#asterisk willy123 (n=icechat5@62.231.36.194) |
| 19:11.53 | [TK]D-Fender | file : Hmmm, no idea why I had it memorized wrong... |
| 19:12.23 | *** join/#asterisk ReD-MaN (i=daemon@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 19:13.31 | tamp4x | anyone here have a internation # they can call from |
| 19:13.44 | *** join/#asterisk clive- (n=pirch@dsl-145-44-59.telkomadsl.co.za) |
| 19:14.11 | alephco1 | if Canada is international. :-) |
| 19:14.30 | clive- | hi, does any expereince ztdummy losing score as the cpu load goes up ? |
| 19:14.36 | tamp4x | i dont think do |
| 19:14.38 | Ariel_ | tamp4x, I guess some place other then the US |
| 19:14.50 | clive- | does anyone I mean |
| 19:15.21 | dlynes_office | tamp4x: what constitutes international? we have no idea what country you're in |
| 19:15.31 | tamp4x | usa |
| 19:15.42 | dlynes_office | tamp4x: and so what constitutes international? |
| 19:15.42 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 19:15.44 | Ariel_ | dlynes_office, he is in the US I just saw he put that up earlier. |
| 19:15.58 | dlynes_office | Ariel_: yeah...i just scrolled back and saw it, myself |
| 19:15.59 | tamp4x | im not sure if canada coutns as internaional |
| 19:16.14 | Ariel_ | there in the same 1NXX segment |
| 19:16.17 | Qwell[] | I'm not sure canada counts as a country |
| 19:16.18 | dlynes_office | tamp4x: does alaska or hawaii count as international? |
| 19:16.29 | Ariel_ | yes and no |
| 19:16.40 | Ariel_ | yes for billing no for same 1NXX area |
| 19:17.04 | Katty | hi lads. |
| 19:17.08 | dlynes_office | tamp4x: so i'm guessing you mean a country that's not in the same country code as the US? |
| 19:17.28 | alephco1 | Qwell[]........ |
| 19:17.39 | dlynes_office | Katty: heya sweetie |
| 19:17.59 | Katty | what's goin on? |
| 19:18.19 | dlynes_office | asterisk? |
| 19:18.23 | Katty | pfft |
| 19:18.28 | Katty | actually. |
| 19:18.36 | Katty | you reminded me of something. |
| 19:18.40 | Katty | twisted[asteria]: you around? |
| 19:18.53 | dlynes_office | i reminded you of a twisted individual? |
| 19:18.57 | Katty | nope. |
| 19:19.01 | dlynes_office | i'm not sure whether to take that as a compliment or ot |
| 19:19.04 | dlynes_office | s/ot/not/ |
| 19:19.29 | dlynes_office | let's try that again not ot |
| 19:19.36 | *** join/#asterisk cardiffit (n=sb@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
| 19:19.39 | dlynes_office | s/\<ot\>/not/ |
| 19:19.54 | dlynes_office | s/<ot>/not/ |
| 19:20.00 | Qwell[] | s/ot$/not/ |
| 19:20.01 | dlynes_office | hrm...his regex is broken |
| 19:20.01 | Qwell[] | :D |
| 19:20.17 | dlynes_office | \< means the beginning of a word, \> means the end of a word |
| 19:20.47 | twisted[asteria] | Katty, yeah |
| 19:20.56 | Katty | twisted[asteria]: are you going to st. louis anytime soon? |
| 19:21.16 | twisted[asteria] | hmm... don't have any plans to do so |
| 19:21.21 | Katty | kk |
| 19:21.29 | twisted[asteria] | why, is it going to implode? |
| 19:21.46 | Katty | no. i'm just going to be up there for a few days at a conference |
| 19:21.51 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 19:21.51 | twisted[asteria] | oooooh |
| 19:21.52 | Katty | are you going to chicago in august? |
| 19:22.16 | *** part/#asterisk alephco1 (n=Weibe@host75.net14.mcsnet.ca) |
| 19:22.18 | twisted[asteria] | i dunno. I know i'll be in dallas again soon though |
| 19:22.29 | Katty | dallas is practically to the moon |
| 19:22.35 | shmaltz | what could be the reason for having echo to the same number when they call me, but no echo when I call that number, this is very consistent. WHy? and what can I do? |
| 19:22.37 | tamp4x | anyone here have an international # thats is outside of US & canada |
| 19:22.39 | twisted[asteria] | no, more like the sun |
| 19:22.44 | twisted[asteria] | *hot* |
| 19:22.48 | Katty | minor detail. |
| 19:22.54 | dlynes_office | tamp4x: you mean country code 1 |
| 19:23.02 | Qwell[] | at least it's in Oct, and not July |
| 19:23.07 | twisted[asteria] | Qwell, yeah. |
| 19:23.09 | dlynes_office | tamp4x: most of the carribbean is country code 1, also, as is Mexico |
| 19:23.12 | Katty | october is a grand month |
| 19:23.16 | twisted[asteria] | Katty, i'll show you a minor detail. |
| 19:23.17 | twisted[asteria] | er |
| 19:23.22 | Qwell[] | pwned |
| 19:23.24 | twisted[asteria] | yeah, october is a most excellent month |
| 19:23.32 | Qwell[] | most excellent? |
| 19:23.39 | Qwell[] | going to San Dimas? |
| 19:23.40 | twisted[asteria] | two holidays |
| 19:23.44 | Katty | and birthdays |
| 19:23.45 | Katty | ;) |
| 19:23.52 | twisted[asteria] | *nods* |
| 19:23.55 | dlynes_office | shmaltz: turn on your echo canceller |
| 19:24.09 | asterboy | Finally, got a call from Sangoma RMA. |
| 19:24.09 | Katty | twisted[asteria]: i need your contact inflimation again, by the way |
| 19:24.15 | twisted[asteria] | z0mg |
| 19:24.15 | dlynes_office | shmaltz: you probably have it disabled |
| 19:24.18 | Katty | twisted[asteria]: sim card is not transferable to blackberry |
| 19:24.23 | twisted[asteria] | oooh |
| 19:24.26 | Katty | twisted[asteria]: which makes me /real/ sad. |
| 19:24.30 | twisted[asteria] | but but |
| 19:24.33 | asterboy | have to pay shipping. |
| 19:24.34 | asterboy | doh |
| 19:24.35 | shmaltz | dlynes_office its on |
| 19:24.38 | twisted[asteria] | use your old phone and upload to NOLA |
| 19:24.46 | Katty | nah |
| 19:24.47 | twisted[asteria] | then use your new phone and download it ;) |
| 19:24.48 | Katty | don't tell....... |
| 19:24.49 | NDT | Heh try and make 2 of these in a Dell 2850 work with 2 TE410P cards...---> Ethernet controller: Intel Corporation 82541GI/PI Gigabit Ethernet Controller (rev 05) |
| 19:24.51 | dlynes_office | shmaltz: ok, try changing the echo canceller you're using then |
| 19:24.53 | Katty | i just synced it to outlook |
| 19:24.54 | Katty | and then back again ;) |
| 19:24.55 | Qwell[] | NOLA? |
| 19:24.59 | twisted[asteria] | Katty, OUCH |
| 19:25.02 | dlynes_office | shmaltz: two of the better ones are MARK2, and MG2 |
| 19:25.04 | Katty | it really wasn't that bad. |
| 19:25.09 | Katty | i just lost my DC field |
| 19:25.15 | twisted[asteria] | i don't have dc anymore |
| 19:25.18 | Katty | so i heard |
| 19:25.25 | shmaltz | dkynes_office I'm using MG2 |
| 19:25.28 | Katty | since i kept tryin to beep you |
| 19:25.35 | Katty | every friggin friday night |
| 19:25.37 | shmaltz | dlynes, b4 I upgraded to 1.2.x it worked fine |
| 19:25.38 | Katty | and you don't answer! |
| 19:25.39 | dlynes_office | sounds kinky |
| 19:25.41 | file | those T-Mobile people, taking over the world |
| 19:25.45 | twisted[asteria] | brb |
| 19:25.55 | Katty | file: you, sir, need a hug. |
| 19:25.57 | dlynes_office | shmaltz: 1.2.x has a different default echo canceller |
| 19:26.02 | file | a chocolate hug? |
| 19:26.12 | Katty | sure! |
| 19:26.22 | iq | Anyone here uses YATE ? |
| 19:26.26 | dlynes_office | shmaltz: i think the default was mark2 in 1.0 |
| 19:26.33 | dlynes_office | iq: try asking on #yate? |
| 19:26.37 | shmaltz | dlynes true |
| 19:26.41 | shmaltz | so I'll try mark2 again |
| 19:26.42 | iq | dlynes_office: they don't use it |
| 19:26.56 | dlynes_office | iq: the developers of yate don't use yate? |
| 19:27.01 | dlynes_office | iq: that makes so much sense |
| 19:27.25 | iq | dlynes_office: apparently they are too busy to help. I hope they use it |
| 19:27.38 | dlynes_office | iq: on that channel, you might have to give them 24 hours to respond |
| 19:27.44 | dlynes_office | iq: it's not a terribly busy channel |
| 19:27.47 | iq | dlynes_office: lol |
| 19:27.51 | dlynes_office | iq: it's busier at 1am, PDT |
| 19:27.55 | dlynes_office | iq: most of them are in europe |
| 19:28.11 | iq | dlynes_office: yes. I'm aware of that. Its been 24 hours btw :) |
| 19:28.16 | Katty | iDunno: why so blue panda bear? |
| 19:28.28 | iq | dlynes_office: I shall try #yate. Thanks :) |
| 19:28.58 | dlynes_office | iq: yeah, but like i said...it's not usually lively in there until about 1am or so, Pacific Daylight Time |
| 19:29.14 | dlynes_office | iq: after about 3am or so, PDT, it quietens down again |
| 19:29.23 | iDunno | Katty: just got given a weeks worth of work to do tomorrow... which is always a bit of a pain ;) |
| 19:29.33 | iq | dlynes_office: i see. I will try in 6 hours or so |
| 19:29.44 | dlynes_office | iq: most of the participants in there are developers, not users |
| 19:29.50 | iDunno | (well, technically, it has to be done by tuesday, but I'm not in the office friday or monday, I booked them as holiday) |
| 19:30.08 | dlynes_office | iq: so they might be busy working on code, or preparing for cluecon, too |
| 19:30.09 | iq | dlynes_office: yes. all of them are very nice people. Just busy or I guess sleeping right now |
| 19:30.26 | twisted[asteria] | Katty, did you get my sms? |
| 19:30.31 | Katty | oh wait |
| 19:30.32 | Katty | no, not that phone |
| 19:30.43 | twisted[asteria] | you didn't port? |
| 19:30.46 | Katty | not yet |
| 19:30.50 | twisted[asteria] | ahh |
| 19:30.56 | Katty | it's a line 2 |
| 19:31.02 | Katty | blackberry's only one line |
| 19:31.14 | Katty | sec, i'll get you the new number. |
| 19:31.19 | twisted[asteria] | ;) |
| 19:31.33 | clive- | hi, does anyone also have the expereince of ztdummy losing score as the cpu load goes up ? |
| 19:35.20 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 19:35.59 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
| 19:39.17 | AlexCTI | Hi, someone has a DID provider around here? |
| 19:39.28 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 19:39.41 | MikeJ[Laptop] | AlexCTI, any number anywhere? |
| 19:39.53 | Qwell[] | I don't think #asterisk has it's own NPA |
| 19:40.01 | MikeJ[Laptop] | it should |
| 19:40.11 | cardiffit | wassup |
| 19:40.28 | MikeJ[Laptop] | cardiffit, nothin |
| 19:40.38 | AlexCTI | well, it most be outside florida USA, and inside USA |
| 19:40.51 | Qwell[] | outside of FL? |
| 19:40.53 | MikeJ[Laptop] | yeah.. I have some did's. |
| 19:41.06 | MikeJ[Laptop] | but no way to give em out yet..... |
| 19:41.06 | Qwell[] | MikeJ[Laptop]: 626 573? :p |
| 19:41.08 | AlexCTI | qwell yes |
| 19:41.16 | MikeJ[Laptop] | 712-432 |
| 19:41.17 | Qwell[] | erm, not 573 |
| 19:41.21 | Qwell[] | just 626 :p |
| 19:41.35 | catlee | Is there a way for asterisk to assign an extension to a SIP phone? Or does each phone have to be configured individually? |
| 19:41.37 | MikeJ[Laptop] | I have some 213 numbers too |
| 19:41.47 | Qwell[] | hmm |
| 19:42.08 | MikeJ[Laptop] | and 858 I think.. other los angeles area stuff |
| 19:42.12 | AlexCTI | where are those are codes? |
| 19:42.22 | MikeJ[Laptop] | 712 is northern iowa |
| 19:42.27 | Qwell[] | yeah |
| 19:42.28 | MikeJ[Laptop] | the rest are around LA |
| 19:42.38 | MikeJ[Laptop] | los angeles that is |
| 19:42.38 | Qwell[] | wanna check 626? |
| 19:42.41 | [TK]D-Fender | catlee : You are describing the very function of extensions.conf |
| 19:43.40 | Qwell[] | and check PM :p |
| 19:44.19 | catlee | [TK]D-Fender: hmmm, how so? |
| 19:44.40 | MikeJ[Laptop] | where is 626 |
| 19:44.52 | Qwell[] | Covina area..LA |
| 19:45.00 | Greek-Boy | does anyone have 7912 cisco firmware? |
| 19:45.03 | Qwell[] | san gabriel valley, I believe it is |
| 19:45.07 | Qwell[] | Greek-Boy: Cisco does |
| 19:45.14 | Qwell[] | and they're the only ones you can get it from |
| 19:45.26 | MikeJ[Laptop] | Qwell, probably.. you need for voice or fax? |
| 19:45.34 | Qwell[] | voice |
| 19:46.01 | catlee | I thought I had to have an entry in sip.conf for each phone? |
| 19:46.02 | Greek-Boy | Qwell, they require a useless contract |
| 19:46.11 | Qwell[] | Greek-Boy: better sign it then.. |
| 19:46.19 | ManxPower | Us to Sales Rep: "We want to bump up the port speed of the frame to a full T-1." Today everyone is on a conference call to do the upgrade and I find out the order is for 20 channels not 24 channels. |
| 19:46.22 | [TK]D-Fender | catlee : The concept of "extension" is a number assigned to a device. sip.conf defines DEVICES, extensions.conf defines the #'s people can dail to go "somewhere", but it an script or to dial DEVICES, etc.... |
| 19:46.24 | generalhan | Greek-Boy: it not useless if you need it to get something you need |
| 19:46.36 | MikeJ[Laptop] | Qwell, sec |
| 19:46.43 | Greek-Boy | lol |
| 19:46.44 | generalhan | there is DEFINATELY a use for it |
| 19:46.45 | Qwell[] | MikeJ[Laptop]: see PM too |
| 19:46.59 | Greek-Boy | when I get my hands on that firmware i'm going to distribute it all over the net |
| 19:47.02 | Greek-Boy | i swear by it |
| 19:47.08 | Qwell[] | Greek-Boy: have fun |
| 19:47.10 | catlee | ok, that makes sense |
| 19:47.27 | Qwell[] | Cisco has very angry lawyers. I suggest not crossing them. |
| 19:47.33 | generalhan | Greek-Boy: what about the next firmware version .. and the next .. and the next ? |
| 19:47.41 | catlee | So, new question then :) Can I have DEVICES automatically defined for me? |
| 19:47.47 | ManxPower | catlee, no. |
| 19:48.00 | *** join/#asterisk kevinfcn_ (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 19:48.00 | ManxPower | Just like you can't have email addresses automatically assigned for you |
| 19:48.14 | catlee | I was thinking something like DHCP for SIP |
| 19:48.22 | [TK]D-Fender | catlee : not really... SIP phones have to register to a server, and that isn't dynamic.... |
| 19:48.34 | catlee | ah, ok |
| 19:48.42 | ManxPower | catlee, Good SIP devices have the ability to be provisioned without having to have the device in hand |
| 19:48.44 | catlee | so each phone has to be programmed manually? |
| 19:48.55 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 19:49.10 | [TK]D-Fender | catlee : How many people are you planning on having as "mobile"? And what kind of link between your 2 servers? VPN? |
| 19:49.21 | ManxPower | for example Polycom phones can be sent an option in the DHCP response to point the phone to the provisioning server, that server contains text files that the phone downloads as it's configuration |
| 19:49.32 | clive- | ok, ztdummy officially sucks |
| 19:49.51 | [TK]D-Fender | ManxPower : He's specifically looking to have something like a "roaming"regsitration so his users can be at site A or B at any time and they always get their calls... |
| 19:50.00 | catlee | ManxPower: cool...and it tells the phone where the SIP server is presumably |
| 19:50.11 | generalhan | what about agentlogin from connected boxes ? |
| 19:50.19 | [TK]D-Fender | catlee : Not what you're looking for..... |
| 19:50.20 | ManxPower | [TK]D-Fender, that would work as long as the phone was provisioned BEFORE it started roaming. |
| 19:50.46 | ManxPower | catlee, remember the current RTP support in the release verison Asterisk does NOT support a jitterbuffer. |
| 19:51.02 | catlee | ManxPower: Sorry, I don't know what that means |
| 19:51.07 | [TK]D-Fender | ManxPower : But then it dials HOME.... but its the USER that roams, not the PHONE. There's the problem... he wants a user to be able to ID as themself from any phone at another site... |
| 19:51.23 | *** join/#asterisk lorinc (n=ang@caracas-4853.adsl.interware.hu) |
| 19:51.31 | ManxPower | [TK]D-Fender, GOOD GOD! That's a design for disaster! |
| 19:51.32 | [TK]D-Fender | ManxPower : Mobile users, stationary phones.... |
| 19:51.38 | [TK]D-Fender | ManxPower ;: Indeed |
| 19:51.39 | catlee | [TK]D-Fender: About 12 regularly mobile users |
| 19:52.00 | catlee | w/ a VPN connection between servers |
| 19:52.04 | Qwell[] | sounds like a job for astdb |
| 19:52.26 | generalhan | [TK]D-Fender: if the phone is connected to the server via VPN cant each user just use an Agentlogin type thing to identify themselves ? then they get thei own calls to that phone |
| 19:52.56 | ManxPower | generalhan, that's far more complicated than it needs to be. |
| 19:52.59 | ManxPower | use astdb |
| 19:53.00 | generalhan | not to just put my 2 cents in on this .. but i had planned on doing something similar and thats how i was plann to do it |
| 19:53.11 | catlee | The phone question was unrelated to the roaming users question...I'm just thinking of how to connect a few dozen phones without having to put too much into sip.conf |
| 19:53.41 | ManxPower | catlee, regardless, asterisk has to know about the phones and |
| 19:54.11 | [TK]D-Fender | generalhan : VM id problems, having to USE a queue, lack of VM integration, large # of queues just to support DIALING... its th wrong tool.... |
| 19:54.25 | boch | do you know why GET DATA is not waiting for the ms i specified ? |
| 19:54.32 | [TK]D-Fender | ManxPower : SHHH!!! Don't just GIVE them the answer! Make them do a trick first! |
| 19:54.33 | asterboy | lol, this is a line in zconfig.h |
| 19:54.34 | asterboy | #define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT |
| 19:54.34 | *** join/#asterisk liran_ (n=Coll@212.199.177.203.static.012.net.il) |
| 19:54.36 | generalhan | [TK]D-Fender: well im glad you told me now ... BEFORE i starting messing with it' |
| 19:55.07 | asterboy | scooby snacks?! |
| 19:55.12 | [TK]D-Fender | generalhan : Can be done, but some ways are less painful/stupid than others. |
| 19:55.15 | catlee | heheh |
| 19:55.32 | generalhan | [TK]D-Fender: ohh ive learned that many times over since i started playing with asterisk ! lol |
| 19:55.51 | generalhan | and i usually do pick the hardest way before i learn about the easier way |
| 19:55.52 | catlee | I was just thinking that it would be nice to have any SIP phone on a network assigned a unique device ID by * |
| 19:56.10 | [TK]D-Fender | catlee : I've already figured the way to do this. The only downside would be that you wouldn't have a VM indicator on the remote phone saying if there is a VM waiting. |
| 19:56.30 | catlee | Yeah, with AgentCallbackLogin() |
| 19:56.31 | catlee | ? |
| 19:56.36 | catlee | I have that sort-of working |
| 19:56.49 | generalhan | pfft my stupid Cisco phones' MWI never goes off anyway ! lol |
| 19:57.02 | [TK]D-Fender | catlee : try getting that to work CROSS-SERVER :) Thats what will kill you.... |
| 19:57.09 | catlee | Yeah, I bet :) |
| 19:57.12 | [TK]D-Fender | generalhan : Polycom > Cisco :D |
| 19:57.25 | catlee | I take it not too many people use roaming "extensions"? |
| 19:57.25 | [TK]D-Fender | catlee : Drop queues, its not the way... |
| 19:57.29 | catlee | I did |
| 19:57.32 | generalhan | [TK]D-Fender: lol AGAIN ... doing the hard way before i learn the easy way |
| 19:57.35 | [TK]D-Fender | catlee : Few.... |
| 19:58.04 | catlee | I have long extensions that call the SIP devices |
| 19:58.08 | [TK]D-Fender | generalhan : I've never done it "officially", but the tool describes the means... I know what will suck, and what will suck LESS :) |
| 19:58.29 | catlee | And short extensions that use Dial(Agent/${EXTEN}) |
| 19:58.39 | generalhan | [TK]D-Fender: thats what i need to learn how to do then ! hah |
| 19:59.03 | [TK]D-Fender | catlee : Just remember - cross-server = DOA |
| 19:59.39 | catlee | ManxPower: What did you mean by "the current RTP support in the release verison Asterisk does NOT support a jitterbuffer"? |
| 19:59.57 | ManxPower | catlee, read up on voip jitterbuffers |
| 20:01.03 | [TK]D-Fender | catlee : And what he means is extremely explicit in that phrase :) |
| 20:02.07 | mpruett | Anybody know a hosted services company that is asterisk friendly? |
| 20:02.11 | catlee | for me, the meaning is how that phrase impacts my work, which isn't evident to me :) |
| 20:02.41 | ManxPower | catlee, it manifests it'self as poor audio quality in 1 direction |
| 20:03.42 | catlee | Hmmm... |
| 20:04.11 | catlee | So SIP uses RTP as the media transport mechanism, and * doesn't have a jitter buffer for RTP, does that mean that SIP isn't a good option with *? |
| 20:04.32 | file | the current releases don't, but 1.4 will have a jitterbuffer yay |
| 20:04.40 | clive- | catlee depends on the network you have |
| 20:05.02 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:05.02 | ManxPower | catlee, many people would disagree with me, but I feel that the current release of Asterisk is not suitable for use with SIP devices on the internet or on any network where you cannot control the packet jitter. |
| 20:05.23 | catlee | But on an intranet it would be fine? |
| 20:05.27 | *** join/#asterisk copantl (n=galel@190.4.22.82) |
| 20:05.35 | copantl | hello guys |
| 20:05.40 | catlee | max 2 hops |
| 20:05.46 | ManxPower | catlee, as long as you have control of all legs of the network between Asterisk and the phone. |
| 20:05.46 | [TK]D-Fender | catlee : Depends... your intranet running on internet based VPN? :) |
| 20:06.00 | catlee | I'm thinking just the local part |
| 20:06.01 | ManxPower | As long as you can do QoS on the WAN links. |
| 20:06.05 | catlee | The VPN is another issue |
| 20:06.13 | ManxPower | if there are no wan links, then it's much less of an issue. |
| 20:06.38 | catlee | I don't consider the remote part of the VPN "local" |
| 20:06.45 | copantl | its safira's SS7 library in a production state? |
| 20:07.19 | ManxPower | Now, "WAN" can consist of links that CANNOT do QoS, like a Frame Relay link with no CIR. |
| 20:07.56 | catlee | And even over a fibre connection you're at the mercy of the routers between you and your destination |
| 20:08.12 | ManxPower | catlee, that would depend on the type of connection. |
| 20:08.45 | ManxPower | Any sort of "T-1" or "E-1" should not be an issue, since those have VERY strict timing requirements |
| 20:08.59 | catlee | E-10? |
| 20:09.04 | copantl | some one use safiras SS7 librarys? |
| 20:09.25 | *** join/#asterisk Jon335 (i=Jon335@unaffiliated/jon335) |
| 20:09.28 | ManxPower | or pretty much DSx or OCx as well, of course. |
| 20:09.59 | nortex | Why does this show up in the CLI? WARNING[21456] : chan_zap.c:3925 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| 20:10.00 | *** join/#asterisk noky (n=noky@200.69.211.18) |
| 20:10.10 | noky | hi |
| 20:10.15 | noky | boch: :D |
| 20:10.35 | noky | buddies i want to know if anybody test the application MeetMe to look the performance... |
| 20:10.38 | SplasPood | How much should an LNP port cost... ? |
| 20:10.49 | noky | or a page with a benchmark? |
| 20:10.57 | ManxPower | nortex, it just does. don't worry abut it. |
| 20:11.49 | nortex | ManxPower, I wouldn't execept it shows up on every attempted fax, which is completely useless at this point. |
| 20:12.39 | catlee | [TK]D-Fender: Our VPN right now has an average latency of 95ms, with peaks up to 228ms...So I'm not sure how well SIP will work over it :) |
| 20:12.56 | catlee | Is there a way to measure jitter? |
| 20:13.24 | nick125 | catlee: SIP usually freaks out at >200ms |
| 20:13.29 | copantl | any body know something about SS7? |
| 20:14.47 | [TK]D-Fender | catlee : It'd be fine, its a question of the impact of failure, and its reliability |
| 20:16.21 | *** join/#asterisk radhios (n=radhios@bue215-194.is.net.ar) |
| 20:16.31 | radhios | Hi!! All |
| 20:16.47 | clive- | copantl very few of us use ss7 in here |
| 20:16.47 | radhios | I have a little problem!! |
| 20:17.23 | pjo | doing asterisk -rvvvvv increases the verbosity. is there a way I can decrease the verbosity without restarting asterisk? |
| 20:17.25 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
| 20:17.27 | eKo1 | copantl: I do. |
| 20:17.31 | radhios | Currently i using a Asterrisk@Home PBX with Soyo ATA |
| 20:17.45 | copantl | clive-: are you heare about the new ver of sefira SS7 Library |
| 20:17.55 | eKo1 | I'm actually looking for someone who can give me the low down on chan_ss7 |
| 20:18.04 | clive- | I heard it exists, I never tried it |
| 20:18.11 | radhios | and when I try to make a call, I have a long delay before the ring sond!! |
| 20:18.26 | [TK]D-Fender | radhios : you need to change that on your ATA then. |
| 20:18.27 | radhios | anybody have any idea?? |
| 20:18.58 | [TK]D-Fender | radhios : The ATA determines whether or not it needs to wait before passing the # you dial to *. |
| 20:19.03 | copantl | i was read about that, and i gonna tested in my network |
| 20:19.18 | eKo1 | copantl: What are you going to test? |
| 20:20.11 | copantl | a connection between my linux/asterisk box with a lucent PSTN switch using SS7 link |
| 20:20.20 | ManxPower | nortex, Um, I get those messages all the time with no issues at all |
| 20:21.38 | copantl | eKo1: do you know how many links ( channels) i can open with ss7 |
| 20:22.15 | radhios | Can you be more specify??? |
| 20:22.29 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 20:24.19 | *** join/#asterisk roche (n=roche@crsj-dc1-fw001.accuhosting.com) |
| 20:24.46 | *** part/#asterisk pjo (n=pjo@212.88.98.114) |
| 20:25.37 | [TK]D-Fender | radhios : No. Read your ATA's manual and learn how to configure its dial-plan. |
| 20:26.08 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 20:26.11 | *** part/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 20:26.21 | radhios | tnx!!! |
| 20:26.47 | noky | [TK]D-Fender: |
| 20:26.52 | noky | i want to know if anybody test the application MeetMe to look the performance... |
| 20:26.56 | noky | some bencharmk |
| 20:27.00 | noky | benchmark |
| 20:27.03 | noky | something like that |
| 20:29.00 | *** join/#asterisk RoyK (n=roy@120.80-203-21.nextgentel.com) |
| 20:29.09 | noky | nobody test the performance of meetme's application????! |
| 20:29.42 | eKo1 | copantl: What do you mean by links? |
| 20:30.08 | copantl | links what? |
| 20:30.27 | eKo1 | Each link in an SS7 networks is either a T1 or E1. |
| 20:30.39 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 20:30.42 | noky | performance??? |
| 20:31.33 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:31.44 | copantl | eKo1: but how many simultaneos call i can have with a SS7 library on a Te110p e1 card? |
| 20:32.32 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 20:33.38 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 20:34.11 | blaylock | can anyone tell me how to generate a call to a zap channel from the asterisk command line? |
| 20:34.18 | boch | eKo1: are you running * with ss7 ? |
| 20:35.00 | RoyK | noky: ? |
| 20:35.37 | noky | RoyK: ? |
| 20:35.58 | Katty | hi RoyK! |
| 20:36.16 | RoyK | hi, Katty |
| 20:36.57 | eKo1 | boch: No, I'm using a signaling gateway that does PRI<->SS7 |
| 20:37.30 | eKo1 | copantl: With 4 E1s, you can have 120 concurrent calls. |
| 20:47.07 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 20:47.57 | techman97_andy | hey all - I have a odd question for ya'll. My system is working fine - no complaints there...no SIP proxies or anything *wink*...but I was just looking at my SIP phone as a call came in, and I saw the callerID number...I thought to myself, how can I get the callerID name to come across too? |
| 20:48.56 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 20:49.00 | CunningPike | techman97_andy: If you do NoOp(${CALLERID(name)}, what do you get? |
| 20:49.11 | Qwell[] | CunningPike: a parse error |
| 20:49.13 | techman97_andy | good point - I get a number. |
| 20:49.27 | techman97_andy | context aside - yes, I do get a number |
| 20:49.37 | techman97_andy | so my SIP provider doesn't send the name |
| 20:50.00 | CunningPike | techman97_andy: So do we for most calls - our telcos mumble something about CID across exchanges, but no-one's ever explained it properly |
| 20:50.03 | *** join/#asterisk SparFux (n=player@e182017229.adsl.alicedsl.de) |
| 20:50.07 | CunningPike | Qwell: :P |
| 20:50.30 | CunningPike | Syntax, schmyntax |
| 20:50.50 | techman97_andy | s,2,NoOp(${CALLERID(all)}) |
| 20:50.51 | [TK]D-Fender | techman97_andy : You SHOULD have CID names... if you don't that should only be for calls coming from carriers that don't transmit it (often cell phones) |
| 20:50.53 | techman97_andy | I do that on my inbound calls |
| 20:50.54 | SparFux | Hi all. I have a realtime issue. I am using libpam-modules from ubuntu studio version. I configured /etc/security/limits.conf and still I don't get programs to run in realtime mode. |
| 20:50.59 | techman97_andy | it puts the number in all fields |
| 20:51.08 | CunningPike | techman97_andy: Where are you located? |
| 20:51.13 | techman97_andy | MN |
| 20:51.16 | techman97_andy | (USA) |
| 20:51.41 | SparFux | ps ax -O ni,rtprio gives me ni=0 and rtprio="-" so I guess I don't really have realtime caps activated. |
| 20:51.43 | CunningPike | techman97_andy: We're in BC (Can) ;) and we rarely get name |
| 20:52.14 | CunningPike | techman97_andy: Our name goes out OK, and we get name on some calls, but most we get number/number |
| 20:52.28 | techman97_andy | that's the same thing I'm seeing, but 99% number/number |
| 20:52.38 | techman97_andy | inbound |
| 20:52.40 | CunningPike | techman97_andy: Yep - us, too. it's frustrating |
| 20:52.46 | techman97_andy | well crap. |
| 20:52.48 | techman97_andy | =) |
| 20:53.27 | CunningPike | techman97_andy: Here's the great thing - I phone home from the office, I get name at home. The other way around, no name. I phone most anywhere else from home, name is displayed |
| 20:53.37 | CunningPike | techman97_andy: Nuts |
| 20:54.13 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:54.32 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 20:54.40 | techman97_andy | =/ |
| 20:54.50 | flujan | guys, can I run zapbarge in a extension? For instance, IAX2/100 :? |
| 20:56.18 | [TK]D-Fender | flujan : Zapbarge = ZAP, not IAX2. You need ChanSpy |
| 20:57.24 | flujan | chanspy is a app? I have the o'relly book... doesn't find it in the appendix. :P |
| 20:58.58 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 20:59.54 | flujan | [TK]D-Fender, thanks again [TK]D-Fender |
| 20:59.55 | flujan | :) |
| 21:02.02 | [TK]D-Fender | ywc |
| 21:04.28 | dlynes_office | [TK]D-Fender: did you call don wright yet? |
| 21:04.48 | [TK]D-Fender | dlynes_office : not yet, and no chance now : taking down my server for upgrade :) |
| 21:04.56 | [TK]D-Fender | dlynes_office : Will tomorrow... |
| 21:05.02 | dlynes_office | [TK]D-Fender: it appears the only reason roland hasn't gotten back to you yet, is because their sales team is running 5 guys short |
| 21:05.14 | dlynes_office | [TK]D-Fender: there's five of them all down at a trade show somewhere |
| 21:05.24 | [TK]D-Fender | dlynes_office : This is a MONTH AGO.... |
| 21:05.25 | dlynes_office | [TK]D-Fender: so even don wright is behind on 30 calls |
| 21:05.32 | dlynes_office | [TK]D-Fender: ah...different story then :) |
| 21:05.39 | [TK]D-Fender | yup... me = forgotten |
| 21:07.11 | [TK]D-Fender | dlynes_office : I want to see how agressive their new pricing is.... |
| 21:07.20 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 21:07.21 | dlynes_office | [TK]D-Fender: on polycom or aastra? |
| 21:07.26 | dlynes_office | [TK]D-Fender: they don't have the 430's yet |
| 21:07.27 | [TK]D-Fender | dlynes_office : CCP gets the job done bet they are god-aweful slow |
| 21:07.38 | dlynes_office | [TK]D-Fender: you mean cccp? |
| 21:07.44 | [TK]D-Fender | dlynes_office : Yeah, Polycom primarily.... |
| 21:07.54 | [TK]D-Fender | dlynes_office : Canadian Communications Products |
| 21:08.00 | dlynes_office | Canadian Communications Products...I can't remember what the third 3rd C was for |
| 21:08.10 | [TK]D-Fender | dlynes_office : No 3rd C :) |
| 21:08.13 | dlynes_office | ah |
| 21:08.14 | dlynes_office | anyways |
| 21:08.19 | [TK]D-Fender | dlynes_office : And no-one has 430's yet... |
| 21:08.19 | dlynes_office | CCP is damned expensive, too |
| 21:08.51 | dlynes_office | [TK]D-Fender: pm me, and i'll give you the pricing I just got from don |
| 21:09.02 | [TK]D-Fender | :) |
| 21:09.18 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 21:11.02 | Cresl1n | heh |
| 21:14.45 | *** part/#asterisk C4T3l (n=robert@216.54.143.2) |
| 21:17.57 | RoyK | <PROTECTED> |
| 21:19.53 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe199.t-net.net.ve) |
| 21:20.07 | epablo | Hi people.. how's it going? |
| 21:20.15 | RoyK | bad |
| 21:20.21 | RoyK | asterisk sucks |
| 21:20.35 | macTijn | how's that ? |
| 21:21.20 | *** join/#asterisk evilrabbi (i=evilrabb@hi.onlineok.com) |
| 21:22.25 | *** join/#asterisk pjo (n=pjo@212.88.98.114) |
| 21:22.57 | pjo | hi all, i get lots of static on an openswitch12 when i say callerid = on in my vpb.conf. any ideas? |
| 21:24.22 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 21:27.15 | dlynes_office | pjo: you're running OpenPBX, or something, right? |
| 21:28.00 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 21:28.26 | pjo | dlynes_office: asterisk |
| 21:28.49 | pjo | openswitch12 is a voicetronics fx0/fxs card |
| 21:29.04 | dlynes_office | pjo: ah...thought that particular product was one of the products from voicetronix that wasn't compatible with stock asterisk |
| 21:29.31 | CunningPike | dlynes_office, [TK]D-Fender: We get our Polycoms from Microserve |
| 21:29.32 | pjo | it works fine *until* i try to turn on callerid |
| 21:29.48 | dlynes_office | CunningPike: they've got good pricing? |
| 21:29.57 | CunningPike | dlynes_office: Better than CCP :) |
| 21:30.06 | dlynes_office | CunningPike: ah...they're local? |
| 21:30.17 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:30.20 | CunningPike | dlynes_office: I think we're paying around $225 for a 501 with PoE cable |
| 21:30.28 | CunningPike | dlynes_office: yes - right here in Vancouver |
| 21:30.46 | CunningPike | dlynes_office: www.microserve.ca, I think |
| 21:31.10 | dlynes_office | CunningPike: cool |
| 21:31.18 | *** part/#asterisk pjo (n=pjo@212.88.98.114) |
| 21:31.27 | dlynes_office | thought they were a pc company |
| 21:31.46 | dlynes_office | oh damnit |
| 21:31.52 | dlynes_office | they're not based out of toronto |
| 21:32.01 | CunningPike | dlynes_office: They are - I was just about to say don't go expecting any support - they're a "stuff in brown boxes" company |
| 21:32.19 | CunningPike | dlynes_office: I think they'll drop ship anywhere in Canada |
| 21:32.22 | dlynes_office | CunningPike: so no firmware updates from them, or anything? |
| 21:32.23 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 21:32.31 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:32.42 | CunningPike | dlynes_office: They probably wouldn't know what you were talking about ;) |
| 21:32.53 | dlynes_office | CunningPike: so how do you get the firmware updates then? |
| 21:32.57 | *** part/#asterisk radhios (n=radhios@bue215-194.is.net.ar) |
| 21:33.37 | CunningPike | dlynes_office: We did two things - we bought a couple of Polycoms from, I think it was voipsupply. Also we got onto the Polycom NDA list, so we get them direct from Polycom now |
| 21:33.53 | dlynes_office | nda? |
| 21:34.05 | CunningPike | dlynes_office: Non-Disclosure Agreement |
| 21:34.11 | CunningPike | ~NDA |
| 21:34.12 | jbot | I'm not allowed to tell you |
| 21:34.18 | dlynes_office | CunningPike: i know what an nda is :p |
| 21:34.26 | dlynes_office | I thought you meant something else |
| 21:34.50 | dlynes_office | how do you get onto that list though? |
| 21:35.06 | hads | He's not allowed to tell you :) |
| 21:35.13 | dlynes_office | heh |
| 21:35.45 | CunningPike | dlynes_office: We discovered that their R&D dept is in North Vancouver - they put us in touch with the sales guy for the territory, we told him we would be buying 400 sets and he came and visited us |
| 21:36.08 | dlynes_office | iow, we're not going to be able to do that :p |
| 21:36.24 | dlynes_office | we're not going to be buying 400 sets any time soon, unless we get a huge client |
| 21:36.44 | dlynes_office | like maybe the district of north van |
| 21:36.45 | CunningPike | dlynes_office: Ours are over a couple of years....... |
| 21:37.20 | CunningPike | dlynes_office: I heard they're the pits to work with. Some hot shot guy there things he knows everything |
| 21:37.43 | dlynes_office | heh |
| 21:37.48 | CunningPike | :D |
| 21:39.05 | [TK]D-Fender | CunningPike : I am a client of MicroServ's already, but their pricing is no better than CCP is for me already... |
| 21:39.24 | [TK]D-Fender | CunningPike : And while CCP is slow, MicroServ actively piss me off :) |
| 21:39.57 | CunningPike | [TK]D-Fender: We've found them OK - we do all our server business through them already |
| 21:39.57 | dlynes_office | and you can still get firmware updates from ccp right? |
| 21:40.25 | dlynes_office | CunningPike: how long can you wait for a shipment? |
| 21:40.56 | CunningPike | dlynes_office: We usually get them inside a week |
| 21:41.03 | CunningPike | dlynes_office: Maybe we get yours :P |
| 21:41.09 | dlynes_office | CunningPike: that's probably why they're too slow for tk |
| 21:41.27 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 21:41.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:41.30 | dlynes_office | CunningPike: usually for the telecom industry you need them in a couple days, cause your customers are bitching where's the phone system |
| 21:41.44 | dlynes_office | CunningPike: cause they order the phone system 2 days before they move into their office |
| 21:41.57 | dlynes_office | CunningPike: but everything else is taken care of two months beforehand |
| 21:41.57 | CunningPike | dlynes_office: I see - we keep a stock at our place, so we always have a couple dozen |
| 21:42.17 | dlynes_office | we've got one customer moving in on Friday |
| 21:42.30 | Strom_C | why customers do that I will never know |
| 21:42.32 | dlynes_office | They still haven't given us a check to tell us to move on it |
| 21:42.47 | dlynes_office | so the cabling's not finished yet, and no phone system ordered yet |
| 21:42.50 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 21:42.50 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:43.30 | dlynes_office | the phone system's usually the last thing they think about when moving in, and the first thing they think about when there's problems |
| 21:44.42 | CunningPike | lol - Microsoft's IE security blocks downloads from............ downloads.microsoft.com! |
| 21:44.51 | dlynes_office | cool |
| 21:44.57 | drray | probably smart |
| 21:44.59 | dlynes_office | Did you see the new google spreadsheet? |
| 21:45.18 | dlynes_office | I've already signed up to be a tester :) |
| 21:45.42 | dlynes_office | heh |
| 21:45.52 | dlynes_office | royk just logs on to say asterisk sucks and then logs off again? |
| 21:46.20 | CunningPike | Must be having a Bad Day |
| 21:46.28 | dlynes_office | obviously |
| 21:46.31 | dlynes_office | he said he was :) |
| 21:46.31 | nettie | damn I lost him again |
| 21:46.46 | nettie | I needed to know how to enable jb in sip.conf.. |
| 21:46.48 | nettie | doh! |
| 21:46.59 | nettie | I patched asterisk but the syntax is unknown to me :) |
| 21:47.19 | dlynes_office | nettie: look at the code for the patch |
| 21:47.44 | nettie | yeah |
| 21:47.51 | nettie | that was my last chance :) |
| 21:47.52 | nettie | eheh |
| 21:47.59 | dlynes_office | it is opensource, you know? |
| 21:48.01 | dlynes_office | heh |
| 21:48.09 | nettie | what the patch? |
| 21:48.22 | dlynes_office | yeah :) |
| 21:48.22 | nettie | it's supposed to be imho considering how GPL works |
| 21:48.33 | nettie | lemme read |
| 21:49.12 | nettie | This program is free software, distributed under the terms of |
| 21:49.12 | nettie | + * the GNU General Public License Version 2. |
| 21:49.12 | nettie | seems |
| 21:49.52 | *** join/#asterisk pdunkel (n=pdunkel@213.235.192.21) |
| 21:50.31 | nettie | found it |
| 21:50.39 | nettie | :) |
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| 22:23.15 | smackus | where do I specify where incoming connections to asterisk are allowed from. for example some of the things I am playing with making connections from a php file to the manager are only allowed by localhost, even though I have added permit= lines |
| 22:23.38 | smackus | but not from other ip addresses |
| 22:34.08 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 22:34.16 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 22:34.35 | liran_ | is there a free service which provides me with a "real" number which i can route to asterisk? |
| 22:35.08 | bon | hm |
| 22:35.12 | bon | if you hear of one |
| 22:35.14 | bon | tell me :) |
| 22:36.12 | *** join/#asterisk gcdtech (n=agough@gcdtechnologies2.plus.com) |
| 22:36.32 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 22:36.59 | dlynes_office | liran_: "real" numbers are called DIDs |
| 22:37.16 | dlynes_office | liran_: you need to find a service that provides free DID origination |
| 22:37.16 | gcdtech | hey, need a little help with Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p |
| 22:37.37 | dlynes_office | liran_: good luck though :) |
| 22:38.28 | gcdtech | anyone any good with Junghanns Quad BRI cards? |
| 22:38.56 | gcdtech | got a wierd outboun cal problem |
| 22:39.01 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:39.11 | dlynes_office | smackus: in your sip.conf, you can have permit=ip.address/net.mask, or deny=ip.address/net.mask |
| 22:39.33 | dlynes_office | smackus: oh...nvm...you asked for manager |
| 22:39.58 | dlynes_office | smackus: yeah...actually...same ones for manager.conf, too |
| 22:41.49 | gcdtech | I'm in the UK, I have the card working for inbound clas and I can dail outbound to local numbers |
| 22:42.04 | gcdtech | eg 92123456 |
| 22:42.11 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:42.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:43.29 | gcdtech | but if I dail a national or mobile number such as 02892123456 I get BT operator message |
| 22:43.35 | liran_ | thanks dlynes_office |
| 22:43.51 | gcdtech | the number you have dailled has not been recognised |
| 22:44.31 | dlynes_office | gcdtech: i don't knwo about the uk |
| 22:44.35 | gcdtech | <PROTECTED> |
| 22:44.35 | gcdtech | <PROTECTED> |
| 22:44.35 | gcdtech | <PROTECTED> |
| 22:44.36 | gcdtech | <PROTECTED> |
| 22:44.53 | dlynes_office | gcdtech: but in north america, whne you dial long distance, you need to dial 011 and then the number |
| 22:45.22 | ManxPower | gcdtech, what is the value of DIALSTATUS |
| 22:45.53 | ManxPower | gcdtech, changes are you have issues in the *dialplan enries in zapata.conf |
| 22:46.12 | ManxPower | even though it's not a PRI, consider it one for pridialplan, prilocaldialplan, etc |
| 22:46.45 | gcdtech | ManxPower: sorry for the Noob question but where do I get DAILSTATUS? |
| 22:47.03 | gcdtech | I have pridialplan = local |
| 22:47.12 | gcdtech | and |
| 22:47.13 | gcdtech | prilocaldialplan = dynamic |
| 22:47.14 | ManxPower | gcdtech, in the priority after the dial Noop(DIALSTATUS=${DIALSTATUS}) |
| 22:47.45 | ManxPower | gcdtech, I'll bet you need to use something different for those. check the mailing list archives. |
| 22:47.50 | ManxPower | ~mailinglist |
| 22:47.51 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 22:48.28 | ManxPower | I think you want your nationalprefix=0 and internationalprefix=00 or something like that. |
| 22:48.53 | ManxPower | I don't know, I just remember seeing it discussed on the mailing lists or on this channel. I'm in the USa where pridialplan=unknown is what you want |
| 22:48.53 | gcdtech | I have both those values in zapata.conf |
| 22:48.58 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 22:49.07 | dlynes_office | RoyK: asterisk still sucks? |
| 22:49.30 | RoyK | well |
| 22:49.40 | RoyK | sometimes it sucks so hard you don't want to know |
| 22:49.43 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:49.46 | RoyK | but most of the time it works |
| 22:49.46 | dlynes_office | heh |
| 22:49.56 | RoyK | ~nickometer [TK]D-Fender |
| 22:49.57 | generalhan | RoyK: take it back ... you know ytou dont mean that ! lol |
| 22:49.57 | ManxPower | Asterisk always sucks. But as long as you make sure it is careful about the teeth.... |
| 22:50.03 | [TK]D-Fender | Work upgrade to 1.2.9.1 successful |
| 22:50.12 | gcdtech | ManxPower: Thanks, I'll take another look through the maillist archive |
| 22:50.26 | RoyK | generalhan: i mean that so perfectly exact you don't know it |
| 22:50.32 | generalhan | hshshs |
| 22:50.33 | dlynes_office | [TK]D-Fender: btw...realized part of why I might have been having difficulties with sangoma on 2.6 |
| 22:50.34 | generalhan | hahaha erven |
| 22:50.41 | [TK]D-Fender | RoyK : You're just jealous because JBOT even notices me ;) |
| 22:50.47 | generalhan | lol |
| 22:50.56 | generalhan | ~RoyK |
| 22:50.58 | jbot | somebody said royk was that viking asterisk guru, or your friend |
| 22:50.58 | dlynes_office | [TK]D-Fender: I was using a lower version of udev than 2.6.15.5 recommended |
| 22:51.05 | [TK]D-Fender | dlynes_office : Do tell... I jsut upgraded mine to 2.3.4-4 on mine an hour ago :) |
| 22:51.13 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 22:51.24 | dlynes_office | [TK]D-Fender: but, the kernel compile doesn't balk about it |
| 22:51.42 | dlynes_office | [TK]D-Fender: i just found out by reading the 'Changes' document in the Documentation directory of the kernel |
| 22:51.59 | [TK]D-Fender | dlynes_office : Hmmm... is that to say you're now running happily on your typically customized kernel again? |
| 22:52.07 | dlynes_office | Not yet |
| 22:52.11 | dlynes_office | Getting there |
| 22:52.18 | dlynes_office | I'm installing on four fresh machines atm |
| 22:52.28 | [TK]D-Fender | dlynes_office : Fun for all... |
| 22:52.33 | dlynes_office | Three VIA Nehemiah C3's |
| 22:52.38 | dlynes_office | One Dual P3 |
| 22:52.47 | dlynes_office | and a Pentium 75 - 200 |
| 22:53.20 | [TK]D-Fender | dlynes_office : I did my old Wanpipe setup and * compile from "straight source" since my work PBX's inception, but have now switched to their SRPMS and all is working well... almost turn-key now. |
| 22:53.24 | dlynes_office | I'm just itching to see how well the Pentium 75 runs asterisk :) |
| 22:53.37 | dlynes_office | I'm going to be running an x100p in that machine :) |
| 22:53.43 | [TK]D-Fender | dlynes_office : Run implies a certain momentum... good luck ;) |
| 22:54.10 | [TK]D-Fender | dlynes_office : DON'T..... Zaptel SWEC would KILL the CPU ;) Get an A200d ;) |
| 22:54.27 | *** join/#asterisk CAPS-LOCK (i=deadbeef@c-71-197-166-39.hsd1.or.comcast.net) |
| 22:54.35 | dlynes_office | [TK]D-Fender: it's just going to be for a home pbx, with one extension, one incoming line |
| 22:54.49 | dlynes_office | [TK]D-Fender: the extension will be ulaw to a sipura 2000 |
| 22:55.01 | [TK]D-Fender | P75? You cheap friggen bastard! And I though it was only your BOSS who needed to invest! |
| 22:55.03 | dlynes_office | [TK]D-Fender: it's mostly for blacklisting incoming calls |
| 22:55.08 | dlynes_office | lol |
| 22:55.22 | *** part/#asterisk CAPS-LOCK (i=deadbeef@c-71-197-166-39.hsd1.or.comcast.net) |
| 22:55.40 | dlynes_office | It's for his home pbx |
| 22:55.47 | dlynes_office | I dont' want to waste a perfectly good machine on it |
| 22:56.13 | [TK]D-Fender | dlynes_office : Waste? No you can do a LOT more with it..... |
| 22:56.35 | [TK]D-Fender | I use mine as my router / HTPC / file / FTP / Web server, etc.... |
| 22:56.39 | gcdtech | ManxPower: unknown did the trick! Thanks |
| 22:56.47 | [TK]D-Fender | dlynes_office : Used to make coffee too! |
| 22:56.52 | dlynes_office | heh |
| 22:57.04 | dlynes_office | I've got a 586dx/133 i use for the firewall |
| 22:57.07 | generalhan | dlynes_office: the best is to use the home box for VM boxes ... with your blacklist if you know its a creditor or something you just have a buddy record the message as some forigen name, and they tend not to call back again ! |
| 22:57.55 | dlynes_office | Hello, you have reached the residence Ho Chi Minh. Please leave a message at the beep. |
| 22:58.03 | generalhan | hahaha |
| 22:58.10 | generalhan | they will think you moved numbers ! |
| 22:58.49 | dlynes_office | Wei? Wo shi Zhang Ze Min. Ni hao ma? |
| 22:58.58 | generalhan | i work for a BK attiorney and i hear clients call in all the time crying cause creditors wont leave them alone ... i always think about bargin in on the call and offering my services ! lol |
| 22:59.09 | dlynes_office | BK? |
| 22:59.16 | generalhan | Bankruptcy |
| 22:59.19 | dlynes_office | ah |
| 22:59.49 | dlynes_office | [TK]D-Fender: damn, you've got a short commute |
| 23:01.59 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 23:02.09 | RoyK | methinks hawkins has been smoking som rather funny stuff |
| 23:02.26 | [TK]D-Fender | dlynes_office : 10 mins... you like? :) |
| 23:02.39 | dlynes_office | [TK]D-Fender: i take it you live downtown? |
| 23:02.46 | generalhan | [TK]D-Fender: it really only takes you 10 minutes to get to work ? |
| 23:02.48 | [TK]D-Fender | Hell no.. West-Island... |
| 23:03.00 | dlynes_office | [TK]D-Fender: eh? I thought you were in T.O.? |
| 23:03.01 | [TK]D-Fender | generalhan : Yup... it used to take *5* ;) |
| 23:03.10 | [TK]D-Fender | dlynes_office : Nope, Montreal. |
| 23:03.30 | dlynes_office | [TK]D-Fender: ah...that's what I thought originally |
| 23:03.30 | [TK]D-Fender | You can't get ANYWHERE in TO in 5 minutes :) |
| 23:03.30 | rene- | D-Fender: there is a way around the waiting times for agent pickup, remember, from yesterday, queues used for outbound? well im using agentlogin instead of agentcallbacklogin, works like a champ |
| 23:03.35 | generalhan | im soo jealous ... it takes me an hour .. IF i leave AFTER rush hour is over (which i usually do just to miss it) |
| 23:03.47 | dlynes_office | [TK]D-Fender: but then I clicked on waht i thought was you earlier today, and seen a Toronto domain |
| 23:04.05 | *** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc) |
| 23:04.06 | dlynes_office | generalhan: i've got about a ten minute commute, too |
| 23:04.16 | generalhan | i hate all you .... im gonna cry |
| 23:04.20 | dlynes_office | generalhan: i live about 20 blocks from the office |
| 23:04.56 | [TK]D-Fender | I live 3 streets from work :) |
| 23:04.57 | dlynes_office | it all depends on how long the wait is for the left turn signal |
| 23:05.00 | generalhan | god that would be sooo nice ... just last week i got all the way home and sat down with a Captain and Coke and the owner called me up because his interent connection and all phone lines went down .. so i got to come all the way back |
| 23:05.04 | dlynes_office | [TK]D-Fender: ah...ten minute walk? |
| 23:05.11 | dlynes_office | [TK]D-Fender: yeah..iv'e got a ten minute drive |
| 23:05.22 | [TK]D-Fender | dlynes_office : nope, 10 minute drive. I used to be a 20 minute walk though :) |
| 23:05.29 | rene- | i sleep under my desk does that count |
| 23:05.30 | generalhan | i would care about that if it only took me 10 minutes to get to work |
| 23:05.37 | *** join/#asterisk backblue (n=moo@87-196-4-132.net.novis.pt) |
| 23:05.37 | dlynes_office | how does three streets take ten minutes to drive? |
| 23:05.40 | dlynes_office | probably faster to walk |
| 23:05.44 | generalhan | rene-: sure that counts ... if you have a fridge and a TV too ! |
| 23:06.03 | [TK]D-Fender | dlynes_office : No. I USED to live closer to work... I'm now farther, but still only a 10 min drive. |
| 23:06.33 | rene- | it is an improvement over my last job where we lived in a ship off shore, i had to go downstairs to the computer room, im closer now |
| 23:07.08 | dlynes_office | rene-: you work for the navy, or something? |
| 23:07.25 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:07.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:15.04 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:15.27 | rene- | dlynes_office: heh, no i was just trying to be funny, you know like those chinese boats where they build TVs in the open sea |
| 23:15.36 | RoyK | <PROTECTED> |
| 23:16.18 | rene- | but i did sleep some days next to a REALLY dell 2650 and i mastered the heavy sleep skill at it |
| 23:18.01 | *** join/#asterisk AeroIllini (n=kevin@c-71-197-210-101.hsd1.or.comcast.net) |
| 23:18.10 | dlynes_office | i think royk needs an enema |
| 23:18.16 | dlynes_office | He's full of blanks |
| 23:18.22 | RoyK | <PROTECTED> |
| 23:18.31 | [TK]D-Fender | :D |
| 23:18.46 | AeroIllini | can I have a dialplan that rings a telephone and then joins that phone to a meetme() meeting when it answers? |
| 23:19.44 | *** join/#asterisk NewSole (n=dave@d226-105-226.home.cgocable.net) |
| 23:20.20 | NewSole | hello... anyone alive |
| 23:20.26 | RoyK | <PROTECTED> |
| 23:21.06 | NewSole | :P |
| 23:21.17 | RoyK | [01:21] *c888 19:18 r(300) the terms and conditions are as follows, by agreeing to contribute to asterisk you are disclaiming any rights you may or may ever have to own any of your own code. you also must relinguish your first born male child to digium and at least 100 liters of blood per year. please be advised that these terms are non-reversable and are binding forever |
| 23:22.03 | dlynes_office | RoyK: old news |
| 23:22.42 | NewSole | I need some one to do a free test for me |
| 23:23.02 | NewSole | here is a stupid question who is running an IAX server |
| 23:23.48 | dlynes_office | i think what you really meant was who isn't? |
| 23:23.54 | RoyK | free test? don't want to pay me for it? |
| 23:24.05 | NewSole | need some one to test out iax softphone on differnet server |
| 23:24.48 | [TK]D-Fender | back later... |
| 23:24.49 | AeroIllini | I'm looking for a dialplan guru to help me with a question |
| 23:25.01 | RoyK | i love this laptop. when it's dark, my keyboard lightens up |
| 23:25.13 | ManxPower | I usually point people to this: http://www.digium.com/disclaimer.txt |
| 23:25.18 | NewSole | who wants to be a beta tester |
| 23:25.23 | dlynes_office | ~suggestions |
| 23:25.25 | jbot | hmm... suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite and ... |
| 23:25.30 | ManxPower | This (with a couple of very small modifications) is the disclaimer I sent to digium |
| 23:25.45 | dlynes_office | ManxPower: they've got two disclaimers you can use |
| 23:26.07 | NewSole | dlynes_office... u still want a copy of soft phone |
| 23:26.09 | RoyK | ManxPower: well, the digium discaimer has been hard to see for some. |
| 23:26.20 | dlynes_office | NewSole: i didn't |
| 23:26.24 | dlynes_office | NewSole: my friend did |
| 23:26.31 | ManxPower | dlynes_office, I know, but when people complain about disclaimers, they are usually saying "I have to assign copyright to Digium" |
| 23:26.39 | dlynes_office | NewSole: i don't really have a need for it atm, unless it runs under linux |
| 23:26.52 | dlynes_office | ManxPower: yeah, and you don't |
| 23:26.53 | NewSole | ok... |
| 23:27.04 | ManxPower | dlynes_office, exactly. |
| 23:27.10 | dlynes_office | ManxPower: one of those disclaimers allows you to retain title to the code |
| 23:27.18 | Jason99 | If you have nat=yes will the media always go through the server? |
| 23:27.28 | dlynes_office | ManxPower: which is the one i sent in, myself |
| 23:27.29 | rene- | yes yes |
| 23:27.35 | ManxPower | The modifications I made to the one I sent in where basically limiting my disclaiming to patches and code posted to the bug tracker under my specific userid. |
| 23:27.59 | RoyK | well, someone i talked to asked digium to send him his disclaimer, since he never sent it |
| 23:28.04 | ManxPower | so if I posted code to the mailing list or a web site, it would not accidently be disclaimed |
| 23:28.06 | RoyK | and they refused to do so |
| 23:28.33 | RoyK | also, they meant it was absolute, for all time |
| 23:28.34 | Jason99 | !nat |
| 23:28.50 | RoyK | Jason99: ? |
| 23:28.55 | dlynes_office | RoyK: i have no idea what you're talking about |
| 23:28.57 | dlynes_office | ~nat |
| 23:29.00 | jbot | hmm... nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 23:29.00 | Jason99 | trying to make jbot help me |
| 23:29.00 | Jason99 | hehe |
| 23:29.25 | RoyK | ~lart |
| 23:29.31 | dlynes_office | AeroIllini: btw...that ~suggestions was meant for you |
| 23:29.34 | dlynes_office | ~lart RoyK |
| 23:29.37 | RoyK | ~lart Jason99 for fun |
| 23:29.51 | RoyK | ~kill dlynes_office |
| 23:29.53 | jbot | ACTION shoots a super-inverse quark gun at dlynes_office |
| 23:29.56 | Jason99 | I'm wondering if nat=yes will make the media go through the server even if the phone isnt being nat |
| 23:30.07 | RoyK | ~disclaimer? |
| 23:30.09 | jbot | I disclaim all of you!, or "fortune -m 'Void where'" |
| 23:31.01 | AeroIllini | dlynes_office, I did just state my problem, and was ignored |
| 23:31.38 | dlynes_office | AeroIllini: i just saw you say you had a dialplan problem...you never stated what your problem was |
| 23:32.04 | dlynes_office | a dialplan problem could be any of a million differnet problems |
| 23:32.25 | pjchilds | AeroIllini, are you looking for asterisk to call someone and then put them in a conference ? |
| 23:32.49 | pjchilds | AeroIllini, ... you could probably use a 'call' file for that |
| 23:32.51 | AeroIllini | dlynes_office, when I first came in the room, I said "can I have a dialplan that rings a telephone and then joins that phone to a meetme() meeting when it answers?" |
| 23:33.07 | dlynes_office | AeroIllini: ah...never seen that |
| 23:33.07 | AeroIllini | pjchilds, yes, that's what I'm trying to do |
| 23:33.18 | dlynes_office | AeroIllini: was probably when i was working on another machine |
| 23:33.59 | pjchilds | AeroIllini, try a call file... it should work... Channel: would be the number you are calling... set context to a context and extension... put meetme() in that... walla ! |
| 23:33.59 | AeroIllini | dlynes_office, ok |
| 23:34.57 | AeroIllini | pjchilds, ahh, I didn't know about call files ... it looks like exactly what I'm looking for |
| 23:35.06 | AeroIllini | pjchilds, thank you |
| 23:35.45 | AeroIllini | pjchilds: one more question |
| 23:36.18 | AeroIllini | pjchilds, can I have an incoming call create a dynamic meeting room, then call out with the .call file and join another caller to it, or many callers? |
| 23:37.03 | AeroIllini | pjchilds, i.e., person 1 calls an extension, which creates a meeting room (or uses a static one) and then asterisk calls person 2 and person 3 and joins them to the room when they answer |
| 23:37.22 | pjchilds | can't see why not... |
| 23:38.03 | pjchilds | there may be a 'nicer' way of doing it.. but your original inbound could call (system ???) to echo out multiple call files, or just run a script that does it.... |
| 23:39.32 | Jason99 | Jun 14 19:38:38 WARNING[15111]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 |
| 23:39.44 | Jason99 | i get that over and over |
| 23:40.05 | dlynes_office | becuase you're trying to use dtmfmode=inband on a sip connecction that's using g729 |
| 23:40.22 | dlynes_office | Jason99: g729 can only use rfc2833 or info |
| 23:40.23 | pjchilds | if you compress DTMF tones with g729 it makes detection really difficult (they get distorted...) |
| 23:40.44 | Jason99 | ah ok so I should switch all phones to rfc2833 |
| 23:40.58 | dlynes_office | Jason99: inband is only officially supported on ulaw and alaw |
| 23:41.07 | dlynes_office | Jason99: correct...rfc2833 is the sanest choice |
| 23:41.18 | dlynes_office | Jason99: occassionally, you have a fubar service provider that requires info |
| 23:41.20 | pjchilds | plus it makes it easier to use a sniffer and grab everones banking-by-phone information etc.... |
| 23:41.43 | *** join/#asterisk |dennis| (n=dennis@200.32.215.84) |
| 23:42.47 | NewSole | http://updates.virttel.com/SoftBeta.zip |
| 23:43.39 | NewSole | anyone want to test it and send me feed back mworkman@virttel.com |
| 23:45.23 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 23:45.27 | AeroIllini | pjchilds, thanks for your help ... I will try it out |
| 23:46.20 | dlynes_office | NewSole: emailed it to my friend for ya |
| 23:46.57 | NewSole | ok |
| 23:47.18 | NewSole | its not all working but connection and callin is |
| 23:48.02 | Strom_C | lklkjl |
| 23:48.54 | dlynes_office | NewSole: so you can place calls from it, and calls can come in? |
| 23:49.23 | NewSole | yup |
| 23:49.32 | dlynes_office | NewSole: and it registers? |
| 23:49.35 | NewSole | it has a voice mail server and call routing and conferance calling... |
| 23:49.43 | NewSole | those are not running right now... just the connection and calling |
| 23:49.58 | dlynes_office | but receive call, make call and register all work right? |
| 23:50.18 | NewSole | yup |
| 23:50.26 | NewSole | and uses g729 |
| 23:50.30 | dlynes_office | NewSole: yeah...those are the only features he uses, anyways :) |
| 23:51.15 | NewSole | o ya and call recording is not enabled yet |
| 23:51.49 | NewSole | the call recording and voice mail server will be in next build... also g723.1 will be added |
| 23:51.53 | dlynes_office | NewSole: you can do that in microsoft sound recorder, anyways :) |
| 23:52.35 | NewSole | ya but it can record single call or conf calls |
| 23:52.50 | NewSole | just like meet me |
| 23:52.59 | dlynes_office | ah |
| 23:53.54 | NewSole | o ya and next build on weekend will also include SMS and Web/Txt MSG sharing |
| 23:56.26 | dlynes_office | incidentally, how many developers are working on it? |
| 23:56.36 | dlynes_office | just yourself? |
| 23:56.38 | NewSole | 2 |
| 23:56.46 | dlynes_office | ah |
| 23:56.49 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |