| 00:00.03 | *** join/#asterisk tomcontr3 (n=gcontrer@198-77-246-201.adsl.terra.cl) |
| 00:00.23 | tomcontr3 | hello |
| 00:00.25 | dlynes | MikeJ[Laptop], Yeah, I noticed that big mess in there when I was going through there, looking for something to do with dtmf tones the other day |
| 00:00.33 | tomcontr3 | Im havinfg a problem with outbound calls |
| 00:00.44 | dlynes | tomcontr3, You need to be more specific |
| 00:00.52 | tomcontr3 | sure, let me tell you the problem |
| 00:01.07 | tomcontr3 | I have 4 lines or 4 trunks from the same provider |
| 00:01.20 | De_Mon | i haven't spent enough time with gdb to use breakpoints. /me adds that to his list of things to do |
| 00:01.21 | tomcontr3 | so |
| 00:01.22 | dlynes | tomcontr3, including letting us knwo what zaptel/sip hardware you're using |
| 00:01.33 | tomcontr3 | no hardware |
| 00:01.38 | tomcontr3 | just SIP connection |
| 00:01.51 | dlynes | Ok, so four SIP upstream channels |
| 00:02.01 | tomcontr3 | the problem is that I can only use the last trunk that I added to make outbound calls |
| 00:02.08 | *** part/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
| 00:02.08 | tomcontr3 | right |
| 00:02.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-241-16.red.bezeqint.net) |
| 00:02.25 | dlynes | tomcontr3, pastebin your sip.conf? (http://pastebin.ca/) |
| 00:02.29 | MikeJ[Laptop] | russellb, I don't actually see where it's doing the name comparison... is it in sip, or rtp? |
| 00:02.38 | tomcontr3 | hold on |
| 00:02.56 | MikeJ[Laptop] | tomcontr3, scrub passwords |
| 00:03.23 | dlynes | MikeJ[Laptop], do people actually upload config files iwth all their passwords in there? |
| 00:03.30 | MikeJ[Laptop] | yes |
| 00:03.32 | dlynes | damn |
| 00:03.41 | tomcontr3 | I also have the file sip_additional.conf |
| 00:03.42 | MikeJ[Laptop] | I've seen it many times before |
| 00:03.54 | MikeJ[Laptop] | tomcontr3, amp/freepbx? |
| 00:03.55 | dlynes | MikeJ[Laptop], I guess most asterisk users aren't sysadmins then |
| 00:04.04 | tomcontr3 | http://pastebin.com/686042 |
| 00:04.09 | MikeJ[Laptop] | and my cc number is..... |
| 00:04.22 | file | 1234567890123456? |
| 00:04.33 | MikeJ[Laptop] | yes file, that's correct |
| 00:04.36 | file | sweet |
| 00:04.47 | MikeJ[Laptop] | and the sig panel code of course is... |
| 00:04.52 | tomcontr3 | freepbx |
| 00:05.00 | file | 123? |
| 00:05.03 | MikeJ[Laptop] | no, 321 |
| 00:05.08 | dlynes | tomcontr3, ummm...you don't have any sip channels defined in that file |
| 00:05.09 | file | sweet |
| 00:05.13 | *** part/#asterisk file (i=jcolp@216.237.114.82) |
| 00:05.14 | MikeJ[Laptop] | tomcontr3, how did I guess... |
| 00:05.17 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 00:05.19 | *** join/#asterisk file (i=jcolp@216.237.114.82) |
| 00:05.22 | file | bleh |
| 00:05.32 | MikeJ[Laptop] | dlynes, it;s freepbx.. they are in some included file.. |
| 00:05.33 | tomcontr3 | lolol, wait, the channels are in the sip_additional.conf |
| 00:05.36 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
| 00:05.53 | dlynes | MikeJ[Laptop], Yeah, mine are all in included files, too, but I don't use freepbx |
| 00:06.13 | dlynes | MikeJ[Laptop], I separate all mine out by line/upstream, and further separated by provider/customer |
| 00:06.41 | MikeJ[Laptop] | I don't use config files... |
| 00:06.49 | MikeJ[Laptop] | they are so last year becky |
| 00:06.52 | dlynes | MikeJ[Laptop], ah...you use the odbc realtime config? |
| 00:07.01 | MikeJ[Laptop] | see the second line |
| 00:07.23 | dlynes | It doesn't say what you use |
| 00:07.30 | tomcontr3 | http://pastebin.com/686048 |
| 00:07.30 | tomcontr3 | there |
| 00:07.32 | MikeJ[Laptop] | OMG becky.. can't you read... |
| 00:07.35 | MikeJ[Laptop] | sigh.. |
| 00:07.46 | dlynes | I thought becky was some expression from your neck of the woods, or something |
| 00:07.59 | MikeJ[Laptop] | dlynes, where you from? |
| 00:08.12 | dlynes | Vancouver |
| 00:08.16 | MikeJ[Laptop] | how old? |
| 00:08.20 | dlynes | 36 |
| 00:08.37 | MikeJ[Laptop] | oh cmon... that hits your age range... |
| 00:09.04 | file | OMG Becky, your butt is soooooo big |
| 00:09.04 | dlynes | Maybe it's an Americanism? |
| 00:09.06 | *** join/#asterisk The_Isle_of_Mark (n=mark@c-68-85-63-96.hsd1.ga.comcast.net) |
| 00:09.06 | MikeJ[Laptop] | http://en.wikipedia.org/wiki/Valley_girl |
| 00:09.06 | JonR800 | not really.. |
| 00:09.10 | JonR800 | haha |
| 00:09.24 | JonR800 | Maybe Sir Mix A Lot didn't make it that far north |
| 00:09.41 | tomcontr3 | some one told me, that my problem is because all the trunks are from the same provider, and that * does get confused with the same IP |
| 00:09.44 | dlynes | Nah...I've heard Sir Mix-a-lot |
| 00:09.45 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:09.55 | MikeJ[Laptop] | http://www.azlyrics.com/lyrics/sirmixalot/babygotback.html |
| 00:09.56 | dlynes | And I've heard some Valley girl speak, but most of that crap I tuned out |
| 00:10.01 | dlynes | Too damned irritating |
| 00:10.05 | Ariel_ | evening everyone |
| 00:10.15 | MikeJ[Laptop] | it's all about the sir mix a lot.... |
| 00:10.17 | JonR800 | irritation is the spice of life. |
| 00:10.27 | dlynes | nothing about becky on that valley girl page, either |
| 00:10.53 | MikeJ[Laptop] | it goes along with the sir mix a lot... |
| 00:10.55 | tomcontr3 | <PROTECTED> |
| 00:11.09 | MikeJ[Laptop] | but it all about the 70's-80's valley girl thing |
| 00:11.40 | dlynes | tomcontr3, You shouldn't need to specify ports for every sip user |
| 00:11.53 | DoktorGreg | <PROTECTED> |
| 00:11.54 | tomcontr3 | freepbx did that |
| 00:12.00 | MikeJ[Laptop] | tomcontr3, not to be rude.. but that whole bit about scrubbing passwords.. I wasn't kidding |
| 00:12.05 | DoktorGreg | Wii = Why? |
| 00:12.18 | MikeJ[Laptop] | you just posted all your passwords for the whole world to see |
| 00:12.23 | dlynes | tomcontr3, and mikej told you not to leave your passwords in there |
| 00:12.31 | tomcontr3 | no, those are not the password |
| 00:12.36 | JonR800 | DoktorGreg: Awful name, I agree. |
| 00:12.51 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 00:12.51 | MikeJ[Laptop] | ok |
| 00:12.56 | DoktorGreg | is it pronounced "Why"? |
| 00:13.02 | justinu|laptop | "we" |
| 00:13.02 | JonR800 | "We" |
| 00:13.18 | *** part/#asterisk seb- (n=seberino@netblock-72-25-103-59.dslextreme.com) |
| 00:13.21 | dlynes | tomcontr3, so your provider doesn't just take calls from you, they transfer calls back to you, also? |
| 00:13.35 | tomcontr3 | no, |
| 00:13.39 | DoktorGreg | how do you get ee from ii? |
| 00:13.47 | JonR800 | Ask the Japanese |
| 00:13.53 | tomcontr3 | if I call and use the trunk 5804760 |
| 00:14.00 | dlynes | tomcontr3, then why do you have incomings defined for eveyr sip channel? |
| 00:14.05 | tomcontr3 | I can make calls, because that is the last trunk that I added, |
| 00:14.23 | tomcontr3 | if I try to make a call using the trunk 5804759 I cant |
| 00:14.39 | tomcontr3 | but If I delete the 5804760 the I can use the 5804759 |
| 00:14.42 | dlynes | tomcontr3, You've probably got other issues too |
| 00:14.50 | dlynes | tomcontr3, Have you tried asking on #freepbx? |
| 00:16.07 | tomcontr3 | yes, some one told me, that is becuase all the trunks uses the same IP |
| 00:16.18 | tomcontr3 | so that * gets confused |
| 00:16.26 | dlynes | nope |
| 00:16.28 | dlynes | Not true |
| 00:16.35 | tomcontr3 | mmm |
| 00:16.44 | dlynes | Asterisk doesn't have a problem with that |
| 00:16.46 | tomcontr3 | <PROTECTED> |
| 00:16.46 | tomcontr3 | <PROTECTED> |
| 00:16.54 | tomcontr3 | that is what I get if I dont use the last trunk |
| 00:17.10 | dlynes | That's a problem with SIP/25804757, not asterisk |
| 00:17.30 | tomcontr3 | hmmm |
| 00:17.39 | tomcontr3 | what kind of problem could it be? |
| 00:17.50 | dlynes | probably incominglimit=1 |
| 00:17.50 | MikeJ[Laptop] | crank up the sip debug and see what it is doing |
| 00:18.00 | dlynes | Or maybe a codec mismatch issue |
| 00:18.08 | dlynes | Or maybe it's something else altogether |
| 00:18.10 | MikeJ[Laptop] | or fire and brimstone? |
| 00:18.17 | MikeJ[Laptop] | provider busted |
| 00:18.29 | dlynes | That's what I said...not an asterisk issue |
| 00:18.35 | tomcontr3 | hmmm, |
| 00:18.38 | dlynes | it's a sip end point issue |
| 00:18.41 | tomcontr3 | so it could be configuration |
| 00:18.43 | MikeJ[Laptop] | lookie at the debug and you can see what the provider is sending you |
| 00:19.01 | MikeJ[Laptop] | off to dinner.. buh bye! |
| 00:19.31 | tomcontr3 | http://pastebin.com/686057 |
| 00:19.37 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:19.37 | tomcontr3 | there is the debug |
| 00:20.17 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 00:20.55 | dlynes | tomcontr3, You see that part about 403 forbidden? |
| 00:21.06 | tomcontr3 | nop |
| 00:21.12 | dlynes | tomcontr3, look closer |
| 00:21.57 | tomcontr3 | SIP/2.0 403 Forbidden |
| 00:21.58 | tomcontr3 | v: SIP/2.0/UDP 201.246.77.198:5060;branch=z9hG4bK63a45af2;received=201.246.77.198;rport=5060 |
| 00:22.02 | tomcontr3 | you are right |
| 00:22.02 | dlynes | there ya go |
| 00:22.09 | dlynes | It means you're forbidden to access it |
| 00:22.13 | tomcontr3 | but why could that happedn? |
| 00:22.24 | dlynes | That's something you're going to have to ask your provider |
| 00:23.09 | *** part/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 00:23.12 | dlynes | One of the problems with SIP error codes is that they can often mean more than one thing |
| 00:23.31 | tomcontr3 | hmmm |
| 00:23.32 | dlynes | So it all depends on what your provider's chosen to map to that sip error code, what it could mean |
| 00:23.42 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 00:23.56 | dlynes | It could mean you don't have enough ports to dial out on, it could mean your password and username didn't match, ... |
| 00:24.00 | dlynes | It could mean any number of things |
| 00:24.24 | tomcontr3 | the wird thing is that if I delete the other trunks, I will be able to us it |
| 00:24.28 | TheCops | I'm using a IP601 and trying to transfer a call. I receiv a call, put on hold, call another number, and want to transfer de first call to the second...someone know how ? |
| 00:24.31 | dlynes | When a SIP provider sets you up, they often set up a certain number of ports |
| 00:24.45 | dlynes | If you exceed that number of ports, they start refusing connections |
| 00:25.11 | tomcontr3 | ohhh |
| 00:25.12 | tomcontr3 | I see |
| 00:25.21 | tomcontr3 | that might be the problem |
| 00:25.32 | [hC] | if you delete other sip trunks, and that one works |
| 00:25.37 | [hC] | youve got a syntax error in your sip.conf file |
| 00:25.38 | tomcontr3 | but, I have all the trunks registered |
| 00:25.38 | dlynes | TheCops, if it's like most other sip phones, you receive the call, hit the transfer button, dial the other number, and then hit transfer again, after the second line starts working |
| 00:25.42 | [hC] | and its sending the wrong auth info |
| 00:25.57 | dlynes | s/working/ringing |
| 00:26.05 | tomcontr3 | Host Username Refresh State |
| 00:26.05 | tomcontr3 | 200.27.79.52:5060 25804760 105 Registered |
| 00:26.05 | tomcontr3 | 200.27.79.52:5060 25804759 105 Registered |
| 00:26.05 | tomcontr3 | 200.27.79.52:5060 25804757 105 Registered |
| 00:26.05 | tomcontr3 | 200.27.79.52:5060 25804758 105 Registered |
| 00:26.09 | TheCops | dlynes, yeah, but I can't talk to the person I'm calling (the second person) before transfering the call...:/ |
| 00:26.26 | dlynes | TheCops, that's cause you're hitting hold, not transfer |
| 00:26.33 | TheCops | hrmm |
| 00:26.43 | tzanger | evening all |
| 00:26.45 | dlynes | TheCops, the 501 does have a transfer button, right? |
| 00:26.49 | tzanger | dlynes: yes |
| 00:26.55 | TheCops | yup |
| 00:27.04 | dlynes | TheCops, so use it, not the hold button |
| 00:27.09 | tomcontr3 | see |
| 00:27.24 | TheCops | Wow |
| 00:27.26 | TheCops | very nice |
| 00:27.27 | TheCops | VERY NICE |
| 00:27.41 | TheCops | IP 601 own really.. |
| 00:27.51 | TheCops | XML browser is totaly crazy |
| 00:28.02 | dlynes | yeah, so i've heard |
| 00:28.14 | dlynes | I haven't had the luxury of using a polycom voip phone yet |
| 00:28.23 | dlynes | I've only used their analog conferencing phones |
| 00:28.40 | dlynes | their analog conferencing phones rock |
| 00:28.44 | TheCops | it's for a project of a call center |
| 00:28.51 | TheCops | I have a 501 at home |
| 00:29.03 | TheCops | not many difference, XML, more line... |
| 00:29.18 | dlynes | 501 doesn't have the xml mini browser then? |
| 00:29.26 | TheCops | nop |
| 00:29.42 | TheCops | you can't hook a expansion too...with the 601 you can |
| 00:30.24 | TheCops | you don't want snom 320 phone ?! I have 6 to sell hehe |
| 00:30.36 | TheCops | I want to change all stuff for IP601 that's too nice |
| 00:30.55 | Ariel_ | tomcontr3, asterisk has a bug with sip. That if it's sending to the same IP address it takes information from the last one in the conf file so if it's one of the other ones that is not the last it's going to fail due to password or other things |
| 00:30.56 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
| 00:31.38 | dlynes | TheCops, hell, no .... we've got a snom220...they freaking suck |
| 00:31.43 | TheCops | lol |
| 00:31.51 | dlynes | TheCops, it's the worst user interface i've ever experienced |
| 00:31.56 | TheCops | 320 are nice too, but have some firmware bug causing audio glitch |
| 00:32.10 | dlynes | TheCops, the software is solid on the snom220 |
| 00:32.16 | dlynes | TheCops, it's just the user interface i hate |
| 00:32.20 | TheCops | hehe |
| 00:32.29 | Ariel_ | the 220 and the 200/190 are far too lite for my customers. They complain all the time about that. |
| 00:32.46 | dlynes | TheCops, i don't want to have to train someone for two or three hours on how to use something as simple as a phone |
| 00:32.55 | TheCops | Ariel_, my customers are in love with 601 phone's |
| 00:33.05 | TheCops | lol |
| 00:33.09 | Ariel_ | I have most of my customers now on polycom's |
| 00:33.12 | dlynes | Ariel_, yeah...the handsets are way too light |
| 00:33.18 | Ariel_ | I can get the 501 for about 169 dollars |
| 00:33.29 | TheCops | Ariel_, what's your network dimension ? |
| 00:33.30 | dlynes | Ariel_, I don't know about the polycoms, but the aastra 9133i's have nice heavy handsets |
| 00:33.40 | TheCops | Ariel_, netxusa ? |
| 00:33.41 | [hC] | the aastra's are garbage phones. |
| 00:33.55 | dlynes | Yeah...I keep hearing they're garbage phones |
| 00:33.58 | [hC] | one of my partners keeps using them for himself, but i dont know why. |
| 00:33.58 | Ariel_ | TheCops, I have many customers from 2 phone sets to over a 1000 units |
| 00:34.00 | dlynes | But I haven't had a single issue with them |
| 00:34.06 | [hC] | the audio quality sucks, the firmware is flaky |
| 00:34.07 | dlynes | I actually like them |
| 00:34.16 | TheCops | Ariel_, wow, 1000 phone, how many simultanous line ? |
| 00:34.19 | [hC] | his crackles alot |
| 00:34.34 | dlynes | [hC], Are you using the 1.3.1 firmware? |
| 00:34.36 | [hC] | only the recent firmwares are usable |
| 00:34.44 | [hC] | I dont use them at all |
| 00:34.47 | [hC] | I use my 7970, or my ip601 |
| 00:34.48 | Ariel_ | TheCops, no I get them from here:http://www.tritechcoa.com/product/791437.html |
| 00:34.57 | dlynes | ah....I've only used the 1.3.1 firmware (the newest) |
| 00:35.05 | dlynes | So I don't know what the older stuff is like |
| 00:35.14 | Ariel_ | TheCops, that customer has 12 asterisk boxes |
| 00:35.16 | TheCops | nice price |
| 00:35.24 | TheCops | Ariel_, fuck |
| 00:35.43 | TheCops | Ariel_, how many calls per box ? |
| 00:35.44 | TheCops | avg |
| 00:36.42 | Ariel_ | TheCops, depends on this one. There is one box that has about 240 sip calls going through it and others that only have 10 but it's due to where there located. |
| 00:36.59 | TheCops | nice |
| 00:37.53 | Ariel_ | Funny thing is that only 2 of there boxes has more then 100 users using it at one time. Rest is smaller |
| 00:38.02 | dlynes | Ariel_, That problem you describe with multiple accounts on the same ip address |
| 00:38.18 | Ariel_ | dlynes, yes it's one that happens all the time |
| 00:38.20 | TheCops | Ariel_: I bought a x226 ibm xseries. xeon 3ghz 1gb de ram, 73gb scsi 10k 320mbps raid mirror...for maximum 8 simultanous line |
| 00:38.22 | dlynes | Ariel_, I have that set up for six phones, all of which don't exhibit that problem |
| 00:38.27 | TheCops | do you think it will be ok enought ? |
| 00:38.37 | Ariel_ | phones are different |
| 00:38.47 | Ariel_ | accounts like providers is the problem. |
| 00:38.54 | dlynes | Ariel_, well, if it works for phones, it should work for providers, too |
| 00:39.08 | dlynes | Ariel_, if it doesn't work for providers, it sounds like it's a provider issue |
| 00:39.27 | Ariel_ | dlynes, don't know why but it's a problem when it sends out to a provider. I think it's more on how the provider answers back |
| 00:39.54 | Ariel_ | dlynes, could be alittle of both. But we found it doing asterisk to asterisk sip trunks |
| 00:40.31 | dlynes | Ariel_, ah...yeah...if it was a problem, you wouldn't be able to use different accounts wiht the same sip provider for different routing |
| 00:40.54 | *** join/#asterisk wolfson (n=wolfson@65.167.193.253) |
| 00:40.55 | Ariel_ | inbound is not an issue due to you route on did's |
| 00:41.20 | dlynes | Ariel_, but then again, I don't need to use that method, myself...I just send all to provider, he routes it based on source ip, and i use different prefixes, depending on how i want to route |
| 00:41.22 | Ariel_ | but sending out and them getting the right one is. So we ended making one main accounts |
| 00:41.46 | dlynes | Ariel_, so no need to even use an account name or password for me |
| 00:41.49 | Ariel_ | dlynes, well that might be one way to do it. |
| 00:42.39 | dlynes | anyways...gotta run |
| 00:42.59 | Ariel_ | I have posted this bug a few times and they kept being signed off by people saying it's a design issue and not a bug. |
| 00:43.02 | The_Isle_of_Mark | quick question: I have 2 pstn lines and one sip account coming into my * box. I have a sip ATA setup as extension 200. Incoming calls work just fine, but outgoing calls give me a busy signal. I have setup my outbound routes correctly *I think*. Where should I start to troubleshoot? Is there a trouble shooting proceedure documented somewhere? |
| 00:43.47 | *** part/#asterisk tomcontr3 (n=gcontrer@198-77-246-201.adsl.terra.cl) |
| 00:43.50 | Ariel_ | The_Isle_of_Mark, if it's sip do sip debug and read what the log says. |
| 00:44.24 | The_Isle_of_Mark | Ariel_, is that a command I can run from a console? |
| 00:44.58 | Ariel_ | on the cli yes sip debug |
| 00:45.05 | Ariel_ | the to turn it off sip no debug |
| 00:45.16 | Ariel_ | you can also do verbose 9 |
| 00:45.20 | Ariel_ | it will help as well. |
| 00:45.22 | The_Isle_of_Mark | Ariel_, I see, much like a cisco |
| 00:45.32 | Ariel_ | argh Cisco |
| 00:46.00 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
| 00:46.02 | The_Isle_of_Mark | Ariel_, I haven't worked with their voice stuff....only data...their data stuff is rock solid |
| 00:46.58 | Ariel_ | The_Isle_of_Mark, I don't like that they charge for everything. Like if you get the phone you have to buy the smartnet. Then the Sip Lic is extra so on and so on. |
| 00:47.47 | The_Isle_of_Mark | Ariel_, again, no experience with that. Data stuff is the same way really. Hey, my box doesnt have the sip command in the paths |
| 00:48.08 | Ariel_ | ??? |
| 00:48.10 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
| 00:48.26 | The_Isle_of_Mark | Ariel_, yeah "which sip" gets me nothing |
| 00:48.27 | Ariel_ | you have to be in the asterisk console |
| 00:48.30 | Ariel_ | asterisk -rvvvvv |
| 00:48.34 | The_Isle_of_Mark | ah |
| 00:48.38 | The_Isle_of_Mark | <---newbie |
| 00:48.41 | The_Isle_of_Mark | heheh |
| 00:48.44 | Ariel_ | yes I see. |
| 00:49.00 | The_Isle_of_Mark | unix guy, but not asterisk YET |
| 00:49.17 | russellb | The_Isle_of_Mark: i think you have the longest possible nick |
| 00:49.32 | Ariel_ | hello russellb |
| 00:50.00 | Ariel_ | I got a new line... I know nothing...nothing I said... |
| 00:50.04 | The_Isle_of_Mark | russellb, thanks! tab completion in xchat completes it for ya |
| 00:51.06 | russellb | The_Isle_of_Mark: as most clients do :-p |
| 00:51.20 | The_Isle_of_Mark | ok neat...sip debug gives me a bunch of stuff I don't have a clue about |
| 00:51.59 | russellb | don't forget iax2 debug, rtp debug, pri intense debug, and set debug 10 |
| 00:52.03 | Ariel_ | wow too much info |
| 00:52.12 | Ariel_ | not flood him will you |
| 00:52.15 | russellb | and enabled debug logging in logger.conf |
| 00:52.23 | The_Isle_of_Mark | not using iax2 |
| 00:52.27 | The_Isle_of_Mark | or pri |
| 00:52.34 | russellb | well, it was a joke anyway |
| 00:52.43 | The_Isle_of_Mark | come on take it easy on the newbie |
| 00:53.10 | russellb | hehe, sorry |
| 00:53.15 | The_Isle_of_Mark | content-length: 0 could be the problem I assume |
| 00:53.24 | The_Isle_of_Mark | either way it destroyed the call |
| 00:54.03 | The_Isle_of_Mark | Ariel_, there is a special place in hell for people that give that line as advice |
| 00:54.25 | The_Isle_of_Mark | but it is funny none the less |
| 00:54.57 | Ariel_ | The_Isle_of_Mark, yes I know...I had it happen to me actually my ex employee did it.... I was ROFL for hours. |
| 00:55.08 | ManxPower | ~ecfo |
| 00:55.11 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 00:55.40 | Ariel_ | ManxPower, how are you tonight? |
| 00:55.45 | The_Isle_of_Mark | Ariel_, ok so what should I look for on this sip debug |
| 00:55.47 | The_Isle_of_Mark | ? |
| 00:55.59 | The_Isle_of_Mark | Ariel_, I can paste it |
| 00:56.15 | Ariel_ | pastebin.ca |
| 00:57.04 | ManxPower | Ariel_, pretty good. |
| 00:57.20 | ManxPower | They reserved a Ditch Witch for this weekend |
| 00:57.30 | The_Isle_of_Mark | http://pastebin.ca/51882 |
| 00:57.59 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 00:58.12 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:58.33 | dlynes_ | russellb: ah cool...didn't know about rtp debug or pri intense debug |
| 00:58.39 | The_Isle_of_Mark | funny thing is, it worked last night...incoming and outgoing |
| 00:59.05 | Ariel_ | The_Isle_of_Mark, that is not enough info what did you change between last night and today |
| 00:59.35 | The_Isle_of_Mark | Ariel_, I have thought about that, and the only thing I did was plug everything back in |
| 00:59.45 | dlynes_ | btw...anyone on atm that would know a reason why a zaptel driver would fail to load after rebooting? |
| 00:59.59 | ManxPower | dlynes, many reasons. |
| 01:00.19 | dlynes_ | i.e. ztdummy and wcfxo...neither one will load on this one machine anymore |
| 01:00.25 | dlynes_ | doesn't matter how many times i reboot |
| 01:00.25 | ManxPower | the most common of which is that some older cards and some bioses seem to change the PCI ID to something the drivers don't know about. |
| 01:00.29 | dlynes_ | and nothing had changed |
| 01:00.43 | Ariel_ | kernel updates is another |
| 01:00.49 | Ariel_ | and other updates |
| 01:00.55 | dlynes_ | absolutely zero has changed, though |
| 01:01.04 | The_Isle_of_Mark | Ariel_, I am setting this up in my house to use a few different lines to do a very simple outgoing routing. I only have 3 cables going into the box |
| 01:01.20 | dlynes_ | no hardware has changed, no software has changed, no configuration files have changed, absolutely nothing |
| 01:01.34 | dlynes_ | All i did was shut the machine down for twelve hours or so |
| 01:01.38 | dlynes_ | and then turned it back on |
| 01:01.53 | dlynes_ | and hten all hell broke loose |
| 01:01.53 | The_Isle_of_Mark | 2 pstn (well technically one ata from a sip provider and one pstn) and one sip provider configured on the asterisk box |
| 01:01.56 | *** join/#asterisk TUplink (n=Tommy@68-232-82-147.chvlva.adelphia.net) |
| 01:02.17 | ManxPower | dlynes, do you have a machine with the exact same card that does work? |
| 01:02.22 | TUplink | i have a prob.... "Unknown RTP codec 101 received" only when i press a number... any ideas |
| 01:02.28 | dlynes_ | It's on a Soltek VIA C3 Nehemiah chipset machine, if that means anything |
| 01:02.33 | dlynes_ | ManxPower: Yeah...several |
| 01:02.43 | ManxPower | TUplink, you cannot get the message on your Zap ports. |
| 01:02.47 | Ariel_ | damm for some reason when I have gmail open in the back ground in Firefox it pots up to the forground on its own... really not nice. |
| 01:02.58 | TUplink | not a ZAP |
| 01:03.02 | TUplink | TelaSIP |
| 01:03.21 | dlynes_ | Ariel_: In X11, or Windows? |
| 01:03.22 | ManxPower | dlynes, look as "lspvi -v" look at the PCI ID on the nonworking box and on the working box, see if they are different. |
| 01:03.34 | dlynes_ | ManxPower: ah..ok |
| 01:03.46 | dlynes_ | ManxPower: but how would that cause ztdummy to cease working? |
| 01:03.47 | Ariel_ | dlynes, windows |
| 01:03.59 | ManxPower | TUplink, none of the 106 mailing list messages with "TelaSIP" were helpful? |
| 01:04.06 | dlynes_ | Ariel_: no idea how to fix it then...sometimes changing window focus behaviour will fix that on x11, though |
| 01:04.06 | ManxPower | dlynes, ZTDUMMY????? |
| 01:04.12 | ManxPower | dlynes, I have never heard of that. |
| 01:04.17 | dlynes_ | ManxPower: Yeah...ztdummy and wcfxo |
| 01:04.22 | dlynes_ | ManxPower: both stopped working |
| 01:04.31 | dlynes_ | ManxPower: I tried ztdummy when wcfxo stopped working |
| 01:04.33 | ManxPower | dlynes, um, ztdummy is useless if you have a digium card. |
| 01:04.36 | The_Isle_of_Mark | anyone have any input on my paste? |
| 01:04.39 | Ariel_ | The_Isle_of_Mark, like I said more info is needed then what you posted in the debug |
| 01:04.51 | TUplink | MAX 106? |
| 01:04.57 | dlynes_ | ManxPower: ztdummy doesn't work properly if you have an x100p card? |
| 01:04.57 | TUplink | MANX* |
| 01:05.08 | CpuID | ManxPower, fyi, it was the lan cards, they werent on the same irq, but there was a marvell 88E8001 onboard and an nforce3 onboard, disabled both stuck 2 realtek pci's in and the issue disappeared |
| 01:05.23 | ManxPower | dlynes, no, it's not used by Asterisk if you have a digium card in the system |
| 01:05.29 | Ariel_ | TUplink, what codec and what dtmfmode are you using? |
| 01:05.31 | CpuID | came to the conclusion it wasnt hdds, since copying locally on the hdds (not over network) doesnt affect things |
| 01:05.39 | dlynes_ | ManxPower: I'm not even getting to asterisk at that point |
| 01:05.40 | CpuID | so yea, no weird static interference now :) |
| 01:05.43 | The_Isle_of_Mark | Ariel_, ok, but unfortunately I don't know what info you would need. Can you ask me pertinent questions and I answer them? |
| 01:05.45 | dlynes_ | ManxPower: just trying to load the drivers |
| 01:05.47 | TUplink | all codecs and inband |
| 01:05.52 | TUplink | i think |
| 01:05.56 | TUplink | let me look |
| 01:06.01 | ManxPower | dlynes, there was some talk about making ztdummy throw an error message and refuse to load if it sees a digium card. |
| 01:06.09 | TUplink | dtmfmode=rfc2833 |
| 01:06.28 | Ariel_ | ulaw/alaw are the only ones that use inband rest rfc2833 unless your using some Grandstream devices they use info |
| 01:06.30 | dlynes_ | ManxPower: Ok, so maybe if i downgrade to zaptel-1.2.5 it might work fine? |
| 01:07.03 | Ariel_ | why do you need ztdummy and a digium board? |
| 01:07.06 | ManxPower | dlynes, are the PCI IDs of the X100P the same on the working .vs. the non-working system? |
| 01:07.20 | dlynes_ | Ariel_: if wcfxo fails to load, i still need a timing device |
| 01:07.30 | dlynes_ | Ariel_: but wcfxo and ztdummy are both failing to load |
| 01:07.51 | dlynes_ | ManxPower: don't know offhand...I'm going to have to take a look at them later tonight |
| 01:08.02 | dlynes_ | ManxPower: they're both at the office, and I've headed home for supper |
| 01:08.09 | *** part/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4159016.sympatico.ca) |
| 01:08.53 | dlynes_ | ManxPower: they're both running zaptel-trunk-apr15 though |
| 01:09.43 | dlynes_ | ManxPower: only reason I'm using trunk is because it was more stable for me than the release on my main asterisk machine |
| 01:10.06 | dlynes_ | ManxPower: but if it's not all that stable on x100p cards, I could downgrade it back to 1.2.5 |
| 01:10.19 | dlynes_ | ManxPower: and just keep trunk on my main machine |
| 01:10.28 | dlynes_ | ManxPower: 1.2.5 was unstable as hell on my pri |
| 01:10.39 | dlynes_ | I was lucky if i could get 1 day uptime |
| 01:11.29 | *** join/#asterisk tjcomp911 (n=NNSCRIPT@CPE-72-128-104-163.wi.res.rr.com) |
| 01:13.05 | *** join/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
| 01:13.16 | TUplink | hum..... |
| 01:13.34 | TUplink | what dose this meen.... Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 01:15.01 | The_Isle_of_Mark | ok question: when I add a trunk using a zap card, and the card is in group 0, channel 1, what should be the syntax in the trunk config file? |
| 01:19.07 | dlynes_ | TUplink: those are voice and/or video capabilities that are not encoded streams |
| 01:22.03 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 01:24.46 | tzanger | tainted-: around? |
| 01:25.25 | dlynes_ | ~seen tainted- |
| 01:25.37 | jbot | tainted- is currently on #asterisk (17h 34m 55s). Has said a total of 130 messages. Is idling for 1h 55m 30s, last said: 'Jaxxan which one'. |
| 01:25.51 | dlynes_ | He's been gone for 2 hrs, tzanger |
| 01:26.52 | dlynes_ | ~seen dlynes |
| 01:26.54 | jbot | dlynes is currently on #asterisk (2h 13m 4s). Has said a total of 137 messages. Is idling for 44m 15s, last said: 'anyways...gotta run'. |
| 01:30.04 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 01:31.48 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 01:35.33 | shmaltz | I'm trying to set PRI_CASUE on my PRI to 22 or 27 then issuing Hangup and I see that the client (caller) is not happy and keeps trying again. Is there something new in 1.2.x as I have this working nicely in 1.0.x |
| 01:38.10 | shmaltz | looks like 22 or 27 has changed in 1.2 as 1 still works as advertiesed |
| 01:40.57 | *** join/#asterisk Mike (n=mike@dsl-201-129-119-118.prod-infinitum.com.mx) |
| 01:41.14 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 01:43.39 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 01:44.39 | *** part/#asterisk NineIron (n=root@gateway.digium.com) |
| 01:45.27 | *** join/#asterisk BadPacket (n=BadPacke@unaffiliated/badpacket) |
| 01:49.00 | *** join/#asterisk MGSsancho (n=weee@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
| 01:51.43 | *** join/#asterisk nite (i=synaptic@68.62.176.196) |
| 01:53.46 | *** join/#asterisk cuban (n=djimenez@johnny5.cuban.cc) |
| 01:53.49 | cuban | sup to thou |
| 01:54.22 | cuban | So this nufone jank is ghetto. Anyone have any alternatives? I have a pretty good feeling they will be closin down soon... |
| 01:54.35 | cuban | or is there someway I can get my damn numbers back?? hah... |
| 01:55.52 | Ariel_ | cuban, there outbound is working there 800 and mi did are not. |
| 01:57.31 | xachen | I myself would have moved the first time they had problems |
| 01:58.03 | xachen | it makes me edgy just thinking that CLEC won't cooperate with a VoIP provider |
| 01:59.14 | cuban | xachen: yeah but I was hopeful :( |
| 01:59.21 | cuban | xachen: Now I have a whole 800 number down |
| 01:59.28 | cuban | xachen: oh wells |
| 01:59.56 | *** join/#asterisk Qwell[] (n=chatzill@unaffiliated/qwell) |
| 02:00.18 | *** join/#asterisk angom_h (n=angom@red-corp-201.143.97.166.telnor.net) |
| 02:01.10 | cuban | Can anyone recommend an alternate? |
| 02:02.51 | Ariel_ | cuban, I am looking for someone that can do it in a few days so far everyone says 10 days to port the number some say 30 days |
| 02:03.04 | cuban | Ariel_ Ouchhhhh |
| 02:03.05 | xachen | It takes a few days to port |
| 02:03.15 | BadPacket | voicepulse did it for me in 30 minutes |
| 02:03.16 | xachen | and AFAIK Nufone sometime takes a couple requetss to hand over the number |
| 02:03.16 | Ariel_ | I have found one race.com that said they might be able to it in less but they can't say yes. |
| 02:03.17 | xachen | very slow |
| 02:03.34 | Ariel_ | xachen, yes I know |
| 02:04.41 | *** join/#asterisk zx225 (n=joel@node-65-183-34-2.dsl.ipglobal.net) |
| 02:06.15 | cuban | I like how voicepulse's website has no pricing info. |
| 02:06.32 | Ariel_ | cuban, stay away from broadvoice |
| 02:06.43 | cuban | Ariel_ I can't even believe they are still around, haha. |
| 02:06.56 | Ariel_ | bkw_, how long do you guys take to port a 800 number over from nufone???? |
| 02:06.57 | BadPacket | http://connect.voicepulse.com/ |
| 02:07.08 | BadPacket | it pretty much *only* has pricing info |
| 02:07.16 | cuban | hahaha |
| 02:07.20 | cuban | man I need to eat. I'm freaking blind |
| 02:07.22 | xachen | Arial_, we can have it expedited |
| 02:07.40 | Ariel_ | xachen, your from where? |
| 02:07.47 | xachen | Then it takes up to 10 days. The problem is mainly Nufone handing them over |
| 02:07.51 | xachen | I work with bkw_ |
| 02:07.56 | Ariel_ | I see |
| 02:09.12 | cuban | Man nufone was much cheaper though, phew |
| 02:09.25 | justinu | and look what happened |
| 02:09.26 | xachen | Price, Support, Reliability - Choose only 2 :) |
| 02:09.30 | cuban | yep :) |
| 02:09.57 | xachen | I personally think Nufone is finished |
| 02:10.01 | cuban | I agree. |
| 02:10.14 | cuban | No way they can survive this, they probably have very few customers left. |
| 02:10.25 | cuban | Their MI DID's are out too huh? |
| 02:10.29 | xachen | Yup |
| 02:11.19 | dlynes_ | Well, once vonage's funding has gone dry, they'll probably be finished, too |
| 02:11.26 | xachen | Hehe |
| 02:11.29 | cuban | haha |
| 02:11.31 | cuban | hmm |
| 02:11.31 | dlynes_ | I can't see them actually turning a profit at the rates they're charging |
| 02:11.41 | cuban | in via voicepulse, out via nufone |
| 02:11.43 | cuban | sounds uber cheap |
| 02:11.58 | justinu | monthly service fees is how the telcos did it for years |
| 02:12.25 | dlynes_ | justinu: yeah, but telcos didn't give unlimited north american minutes, either |
| 02:12.39 | justinu | vonage isn't truly unlimited |
| 02:12.44 | justinu | they'll shut you down and backbill you |
| 02:12.53 | dlynes_ | they will? |
| 02:12.55 | justinu | yes |
| 02:13.10 | dlynes_ | I asked the stupid operator on the phone if their unlimited was truly unlimited |
| 02:13.12 | justinu | read the ToS closely |
| 02:13.14 | dlynes_ | and she insisted it was |
| 02:13.21 | justinu | try and use them as a trunk :) |
| 02:13.35 | dlynes_ | I haven't even tried them, much less tried using them as a trunk |
| 02:13.44 | justinu | their stuff actually seems to work |
| 02:13.58 | dlynes_ | The ToS says unlimited means certain number of minutes per month? |
| 02:14.06 | justinu | no, it says it's at their discretion |
| 02:14.10 | dlynes_ | ah |
| 02:14.13 | Qwell[] | ~unlimited |
| 02:14.16 | jbot | unlimited is, like, <Nugget> unlimited voip == punch the monkey to win a free ipod |
| 02:14.23 | justinu | usage must follow normal residential patterns |
| 02:14.29 | justinu | or business patterns |
| 02:14.35 | justinu | if you buy the business plan |
| 02:16.53 | *** join/#asterisk BadPacket_ (n=BadPacke@unaffiliated/badpacket) |
| 02:17.34 | dlynes_ | So I guess 1-900 numbers called regularly after 11pm is out of the question? |
| 02:20.37 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 02:24.02 | *** join/#asterisk nortex (n=breeves@adsl-69-149-172-106.dsl.amrltx.swbell.net) |
| 02:27.38 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 02:27.51 | cuban | dammit bitchx |
| 02:27.57 | cuban | how many times do I disable that crap |
| 02:30.17 | Alystair | ~commands |
| 02:30.21 | Alystair | :[ |
| 02:31.18 | [TK]D-Fender | ~suggestions |
| 02:31.19 | jbot | [suggestions] 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 patient. 6) ... |
| 02:31.22 | [TK]D-Fender | \:D |
| 02:31.28 | [TK]D-Fender | Stop being so pushy :) |
| 02:32.10 | nortex | Anybody here use Flash Operator with astmanproxy? |
| 02:33.41 | cuban | man |
| 02:33.50 | cuban | DSBL blacklisted a customer of mine's IP for NO reason |
| 02:33.52 | cuban | twice. |
| 02:34.09 | Hmmhesays | obviously homicide is your only course of action |
| 02:34.10 | cuban | Claims they run a socks5 proxy on some port that isn't even open. I h8 DSBL. |
| 02:34.42 | sleepy_one | who is DSBL ? |
| 02:36.24 | BadPacket_ | . |
| 02:41.54 | cuban | stupid mail blacklisting |
| 02:43.08 | xachen | I refuse to use DNS BL's for that reason |
| 02:43.12 | xachen | because they are unfairly run |
| 02:46.41 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:51.15 | wolfson | that all depends on what dnsbl you use |
| 02:51.24 | wolfson | some are quite loose in who they add, and stringent in who they remove |
| 02:51.42 | wolfson | others have nearly 0 false posatives |
| 02:56.01 | *** join/#asterisk nortex (n=breeves@adsl-69-149-172-106.dsl.amrltx.swbell.net) |
| 03:05.59 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 03:06.19 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 03:10.31 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 03:10.46 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 03:12.55 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 03:15.18 | cuban | wolfson Well this one is scanning one ip and adding another |
| 03:15.21 | cuban | it's rediculous |
| 03:16.23 | wolfson | who is it? |
| 03:21.23 | sleepy_one | -A RH-Firewall-1-INPUT -p tcp -m tcp -j DROP |
| 03:24.00 | sleepy_one | cya all l8r |
| 03:28.57 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:29.38 | harryvv | looking for a DID/Porting number supplier for the 604 area code area or 206 area code. Thanks |
| 03:35.56 | SplasPood | Can musicclass=default be set for a specific sip user, or is it only a global? |
| 03:41.17 | *** join/#asterisk inv_Arp (i=junya@adsl-10-153-159.mia.bellsouth.net) |
| 03:48.00 | *** join/#asterisk I-MOD (n=opticron@gateway.digium.com) |
| 03:50.17 | *** join/#asterisk erick_az (n=erickj_a@ip67-91-82-163.z82-91-67.customer.algx.net) |
| 03:51.56 | erick_az | I have an asterisk server that has been up and running for several months. A few hours ago all my phones stopped being able to dial out. when a call comes in there is a 1 to 2 second delay in the voice. nothing in the config has changed and the network seems to be fine. |
| 03:52.02 | *** join/#asterisk bmg505 (n=leon@dsl-146-6-211.telkomadsl.co.za) |
| 03:52.06 | erick_az | Does anyone have any ideas? |
| 03:57.06 | znoG | anyone know a cheap way to build a network tone generator? |
| 03:57.10 | dlynes | erick_az, any windows machines on the same network? |
| 03:57.23 | znoG | USD $44 + international shipping + waiting time is a bit much |
| 03:57.53 | dlynes | znoG, do you not have any electrical wholesalers in argentina? |
| 03:58.06 | znoG | yes, and they like to charge about 3-4 times that amount |
| 03:58.11 | dlynes | ah |
| 03:58.13 | dlynes | suckage |
| 03:59.33 | orlok | i know theres tone generation software in python |
| 03:59.38 | orlok | but its for series 60 mobiles |
| 04:00.05 | jake1932 | i'm trying to figure out how dns can half work - i can ping sites and they resolve, yet asterisk wont resolve the dns - says dns error |
| 04:01.23 | justinu|laptop | i've seen asterisk act wonky like that too, jake |
| 04:02.18 | jake1932 | the only thing i can guess is that ping uses a different api call or something like that... |
| 04:02.39 | dlynes | jake1932, restarting asterisk doesn't help? |
| 04:02.41 | jake1932 | gethostbyname? |
| 04:02.53 | jake1932 | right - restarting doesnt help |
| 04:03.07 | dlynes | weird |
| 04:04.06 | jake1932 | Probably a DNS error for registration to me@host.com, trying REGISTER again |
| 04:04.19 | dlynes | erick_az, the reason I'm wondering if there's windows machines on the network or not, is because there might be increased network activity if any of them got hit with a virus or a worm, or something...if it's severe enough, it might affect voip traffic |
| 04:04.50 | jake1932 | justin - did you have this error? |
| 04:05.48 | *** join/#asterisk MacDome (n=eseidel@adsl-69-226-232-199.dsl.pltn13.pacbell.net) |
| 04:12.38 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 04:16.48 | websae | how's everyone doing? |
| 04:18.25 | Strom_C | i'm having phun |
| 04:18.32 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:18.33 | websae | what are you doing? |
| 04:18.41 | Strom_C | tinkering |
| 04:19.27 | *** join/#asterisk groogs_ (n=greg@d221-73-237.commercial.cgocable.net) |
| 04:19.37 | websae | that's fun then :) |
| 04:21.10 | *** join/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
| 04:23.46 | *** part/#asterisk angom_h (n=angom@red-corp-201.143.97.166.telnor.net) |
| 04:28.55 | *** join/#asterisk chiang (i=KuChax@202.6.239.3) |
| 04:31.52 | chiang | hi, i'm trying to use FWD to call US toll free, any idea how to do the "Dial Patterns" |
| 04:41.02 | litage | i'm not currently on a redhat-based box. would someone using fedora/centos/redhat/RHEL/etc mind doing an ``ls -al /etc/sysconfig/'' and putting it in a pastebin for me please? (eg: http://rafb.net/paste) |
| 04:41.17 | websae | i wish i was still on redhat |
| 04:41.23 | websae | is that fedora you're running? |
| 04:42.06 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:44.06 | litage | websae: i'm running debian, but i need to find out the contents of /etc/sysconfig/ in redhat-based distros (ie: fedora, centos, rhel, etc) |
| 04:44.40 | websae | ohh |
| 04:44.48 | websae | why do you need to know that |
| 04:45.30 | *** join/#asterisk rushowr (n=rushowr@cpe-65-189-186-30.columbus.res.rr.com) |
| 04:46.25 | orlok | litage: hold on |
| 04:46.42 | litage | websae: i'm converting some RPMs to DEBs |
| 04:47.12 | websae | ohh ok |
| 04:47.27 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:48.27 | *** part/#asterisk rushowr (n=rushowr@cpe-65-189-186-30.columbus.res.rr.com) |
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| 04:53.31 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:55.59 | websae | anyone alive out there? |
| 05:01.46 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 05:11.13 | dlynes_ | no |
| 05:11.30 | websae | dlynes_how's it going |
| 05:11.35 | dlynes_ | not too bad |
| 05:11.41 | dlynes_ | Just got home |
| 05:11.43 | websae | what time is it by you? |
| 05:11.46 | websae | same here |
| 05:11.49 | dlynes_ | 10pm |
| 05:11.58 | websae | 12:11AM here |
| 05:12.03 | websae | California? |
| 05:12.07 | dlynes_ | Vancouver |
| 05:12.11 | websae | ahh |
| 05:12.11 | websae | ok |
| 05:12.57 | dlynes_ | one headache after another with network problems with sip phones today |
| 05:13.20 | dlynes_ | not to mention zaptel drivers |
| 05:13.26 | websae | what's going on with sip phones? |
| 05:13.32 | websae | what do you use? |
| 05:13.50 | dlynes_ | they seem to be kind of erratic as to whether they'll register with asterisk or not |
| 05:13.53 | dlynes_ | Aastra 9133i |
| 05:14.01 | websae | hrm never have played with those |
| 05:14.08 | dlynes_ | I've got a few of them deployed out in the field without any problems |
| 05:14.12 | websae | only grandstream,cisco,sipura here |
| 05:14.20 | dlynes_ | but these two in particular, especially the one i've been having problems with |
| 05:14.26 | websae | hrm....having registration issues |
| 05:14.34 | websae | pointing them to a STUN server? |
| 05:14.36 | dlynes_ | sipura ata's I've had all kinds of problems with |
| 05:14.44 | dlynes_ | No point...no nat involved |
| 05:14.47 | websae | my 2100 seems to work fine |
| 05:14.59 | websae | are you a VoIP provider? |
| 05:15.02 | dlynes_ | 2100 is a sip ata/router all in one, right? |
| 05:15.08 | dlynes_ | voip provider and interconnect |
| 05:15.09 | websae | yep with QoS |
| 05:15.12 | websae | ohh sure |
| 05:15.28 | websae | if you ever need some good wholesale rates, let me know :) |
| 05:15.30 | websae | hehehe |
| 05:15.41 | dlynes_ | Well, we're getting pretty good rates already |
| 05:17.57 | *** join/#asterisk bzbw (n=wlwzhang@68-190-223-129.dhcp.mtpk.ca.charter.com) |
| 05:20.02 | bzbw | anyone online? |
| 05:20.22 | websae | hey |
| 05:20.47 | bzbw | I just have a question to see anyone can help to answer |
| 05:21.07 | bzbw | <PROTECTED> |
| 05:21.08 | bzbw | <PROTECTED> |
| 05:22.02 | bzbw | the problem is that it does not send SIP invite to the broadvoice sip server, it went to somewhere else, strange:( |
| 05:23.11 | websae | odd |
| 05:23.44 | bzbw | damn, it's so quiet here. |
| 05:23.57 | _diLLec | =) |
| 05:25.08 | bzbw | ok, another question, does anyone know 1.2.7.1 support H.264? |
| 05:26.28 | DoktorGreg | heya |
| 05:26.49 | DoktorGreg | how do i make mpg321 play my music on hold files? |
| 05:32.06 | brookshire | doktor: don't use mpg321 use native moh! |
| 05:33.33 | _diLLec | bzbw: it should support it since asterisk is one part of the consortium deploying it. |
| 05:36.29 | *** join/#asterisk Pageus (n=FreePBX9@ip70-190-19-6.ph.ph.cox.net) |
| 05:38.46 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
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| 05:39.05 | DoktorGreg | whats wrong with mpg321? |
| 05:40.28 | Pageus | I have a question about the differences between pri and regular voice T1's and how the T100P card handles them |
| 05:42.37 | Pageus | I have my local sip systems talking, and my voicepulse connect IAX2 trunk working, and while i am waiting for my local line provider to run the cable to my rack, i setup the t100p card... |
| 05:42.57 | brookshire | dok: native is better ;) |
| 05:43.32 | Pageus | however it seems that they changed horses on me and went from PRI to just DID based voice T1. |
| 05:44.16 | brookshire | pageus: so they are sending you CAS? |
| 05:44.25 | Pageus | Working with PRI was much easier.. there is loads of info on how to set it up and the hunting system i don't have to worry about since it is handled by the provider |
| 05:45.16 | Pageus | i believe so.. i am awaiting an email with the information.. however were on a deadline and i am hoping they will drop the line to the rack by tommarow.. since i have to turnthis system up by monday at 8am |
| 05:45.47 | brookshire | phone companies are known to screw up orders ;) |
| 05:45.49 | Pageus | he gave me a basic rundown on what they were sending me. and i had already setup the card to handle pri based. |
| 05:45.56 | brookshire | was a pri ordered? |
| 05:45.56 | Pageus | yeah.. something about saving us money |
| 05:46.05 | brookshire | of course.. |
| 05:46.09 | brookshire | so are these outgoing lines? |
| 05:46.17 | Pageus | apparently not.. to give you a brief rundown |
| 05:46.55 | brookshire | so, is the T1 for incoming or outgoing? |
| 05:47.03 | brookshire | or both? |
| 05:47.05 | Pageus | we went with the company and they installed a local t1 with a channel bank.. everything worked fine in testing till we went live |
| 05:47.06 | Pageus | yes |
| 05:47.07 | Pageus | both |
| 05:47.12 | brookshire | k |
| 05:47.19 | Pageus | 6 trunks and 26 DID |
| 05:47.28 | brookshire | does the company have 1 phone number? |
| 05:47.51 | Pageus | my company.. no we have 7 but that last one is moving to an efax service |
| 05:48.11 | Pageus | big question here is does asterisk allow for hunting. |
| 05:48.15 | brookshire | k.. i don't know how you're going to handle the different numbers, but oh well ;) |
| 05:49.42 | Pageus | yeah when i found out it wasn't pri.. i got a bit frazzelled.. i used to be in telco(analog) and have always dabbled in VOIP since back in the days of Voxtalk. i worked with PRI systems and they were fairly easy to configure.. |
| 05:50.28 | brookshire | well.. by hunting, you mean.. which number is it coming from, correct? |
| 05:50.33 | Pageus | so i am now at the mercy of those in irc land for help being that i feel i have exhausted my resources in google and yahoo and any other way you can get info on the net |
| 05:50.34 | Pageus | lol |
| 05:50.37 | Pageus | no |
| 05:50.52 | Pageus | hunting as in when one number is busy it rolls over to the next one |
| 05:50.58 | Pageus | that i tell it 2 |
| 05:51.25 | Pageus | so say i have xxx-4060 it's busy so it rings xxx-4061 |
| 05:51.28 | Pageus | and so forth |
| 05:51.37 | dlynes_ | hunt groups |
| 05:51.45 | Pageus | yeah |
| 05:52.07 | dlynes_ | similar to overlines, but not quite |
| 05:52.20 | Pageus | i know i can do hunting on the extension side.. i had setup aah here at the house to start focusing more on it |
| 05:52.25 | dlynes_ | hunt groups are like overlines, but every overline has a real phone number associated with it |
| 05:52.37 | Pageus | nod |
| 05:52.47 | dlynes_ | overlines don't necessarily have a real phone number |
| 05:53.07 | dlynes_ | and the hunting is all done on the telco side, not on the CPE |
| 05:53.14 | Pageus | we have 6 DID's that we have had for close to 15 years. that were real lines. |
| 05:53.33 | DoktorGreg | ok musiconhold is really obviously playing the mpg files wrong |
| 05:53.38 | Pageus | these are now going to be apparently just pushed to us as DID's through the T1 |
| 05:53.45 | dlynes_ | Pageus: correct |
| 05:53.46 | brookshire | doktor: use native |
| 05:53.48 | brookshire | :) |
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| 05:54.14 | dlynes_ | DoktorGreg: Did you try what I told you last time with mpg123 0.53r or whatever it is, or madplay? |
| 05:54.15 | *** part/#asterisk vinkoval (n=vinkoval@c-24-10-180-226.hsd1.ut.comcast.net) |
| 05:54.17 | Pageus | so i would need to tell them to hunt my numbers for me no matter what? |
| 05:54.28 | dlynes_ | Pageus: you mean for incoming calls? |
| 05:54.38 | Pageus | correct, Dlynes |
| 05:54.51 | dlynes_ | Pageus: I don't know how it works when you're using a channel bank |
| 05:55.00 | Pageus | I'm no longer using a channel bank |
| 05:55.01 | dlynes_ | Pageus: Normally I would just plug it straight into the t1 card |
| 05:55.21 | dlynes_ | And if a call comes in on 604-266-3352, another call can come in on the same phone number |
| 05:55.27 | dlynes_ | it'll only ring busy if you tell it to ring busy |
| 05:55.30 | Pageus | that system failed misserably due to line problems. so they just hunted and forwarded the numbers to live lines |
| 05:55.39 | Pageus | OK!!! |
| 05:55.53 | Pageus | that took a load off my mind there |
| 05:55.57 | dlynes_ | So you can have five simultaneous calls all coming in on the same phone number |
| 05:56.02 | dlynes_ | No need to do hunt groups |
| 05:56.08 | Pageus | whew |
| 05:56.24 | dlynes_ | At least with what we're doing |
| 05:56.29 | Pageus | glad i finally broke down and decided to get back into IRC |
| 05:56.30 | Pageus | lol |
| 05:56.32 | dlynes_ | I'm not sure if what we have is a Voice T1, or a pri |
| 05:56.41 | Pageus | hmm |
| 05:56.42 | dlynes_ | I think it's a voice t1 |
| 05:56.45 | Pageus | nod |
| 05:56.56 | Pageus | nod as in understood |
| 05:57.06 | dlynes_ | It's got 23 b(?) channels and 1 d(?) channel |
| 05:57.23 | brookshire | that's pri |
| 05:57.26 | Pageus | sorry.. comming from an old background of dial-up bbsing and teleconf and had avoided irc for a LONG time |
| 05:57.31 | Pageus | hmm |
| 05:57.42 | dlynes_ | brookshire: so what's the diff then? |
| 05:57.50 | dlynes_ | brookshire: i thought voice t1 and pri were the same thing? |
| 05:58.00 | Pageus | the provisioner told me that they would send dids with wink??? |
| 05:58.20 | Pageus | which is understand the wink part.. but not how the card would interact with it. |
| 05:58.23 | brookshire | well.. pri has b and d channels |
| 05:58.26 | brookshire | it's like isdn |
| 05:58.30 | brookshire | i don't know what voice t1 is |
| 05:58.40 | brookshire | but if it is CAS |
| 05:58.42 | dlynes_ | wink is a proprietary method, as an alternative to kewl start i think pageus |
| 05:58.47 | brookshire | that's 24 lines |
| 05:59.09 | Pageus | yeah we have a possibility of 24 lines(trunks?) |
| 05:59.16 | Pageus | still trying to figure that one out |
| 05:59.27 | brookshire | pageus: it's not digital, basically |
| 05:59.31 | brookshire | well.. it is |
| 05:59.37 | brookshire | ugh |
| 05:59.38 | brookshire | confusing |
| 05:59.41 | dlynes_ | i've got 23 channels and a pseudo channel |
| 05:59.44 | dlynes_ | whatever tha tis |
| 05:59.51 | Pageus | with my voicepulse trunk i have 2 numbers that work seperatly and send a did throuh |
| 05:59.58 | brookshire | dyl: sounds like pri |
| 06:00.00 | Pageus | right and i have 24 channels |
| 06:00.11 | dlynes_ | yeah...pri |
| 06:00.29 | Pageus | 23 on a t1 used to represent PRI when i was working on the old PM3's |
| 06:01.00 | Pageus | i remember getting a line that was NOT pri and it wouldn't work |
| 06:01.16 | brookshire | it should work with asterisk |
| 06:01.24 | brookshire | you just can't do any of the fun pri stuff :( |
| 06:01.31 | Pageus | but the phone company told me i had 24 lines instead of 23 |
| 06:01.34 | Pageus | lol |
| 06:01.47 | Pageus | i never viewed anything with pri as fun. it was a job |
| 06:01.47 | Pageus | lol |
| 06:01.52 | brookshire | yeah.. so you can't change caller id, or map 1231231231 billion phone numbers |
| 06:01.56 | Pageus | i consider voip fun |
| 06:02.02 | Pageus | lol |
| 06:02.08 | brookshire | you're stuck with whatever the line is |
| 06:02.15 | dlynes_ | Pageus: sounds like you're going to have to set up hunt groups with the phone company |
| 06:02.17 | Pageus | our numbers are already listed in the callerid registries and such |
| 06:02.28 | Pageus | hmm ok |
| 06:02.34 | brookshire | pageus: with pri, you can set that :) |
| 06:02.41 | dlynes_ | Pageus: I think it's a cash grab on the phone company's part |
| 06:02.54 | dlynes_ | Pageus: i.e. they make more money off you when you use voice t1 instead of pri |
| 06:03.01 | Pageus | not sure.. hunting never cost us anything |
| 06:03.03 | brookshire | i don't think so.. |
| 06:03.05 | dlynes_ | ah |
| 06:03.08 | Pageus | actually the pri was more expensive |
| 06:03.15 | dlynes_ | yeah...pri's more expensive |
| 06:03.16 | brookshire | i think they just want to upsell pri |
| 06:03.22 | Pageus | lol |
| 06:03.36 | Pageus | they have been losing money on us since they took the account |
| 06:03.47 | dlynes_ | but to be able to do all the stuff you can do on a pri and emulate it on a voice t1, it sounds like you'd need at least two voice t1's, instead of one pri |
| 06:03.57 | Pageus | they have to pay the local phone companies lines and sell them to us at lower rate |
| 06:04.07 | Pageus | hmm |
| 06:04.14 | dlynes_ | why's that? |
| 06:04.23 | Pageus | i do know i can break out the t1 into 24 trunks |
| 06:04.50 | Pageus | just from reading.. but it never specified if it was pri or CAS(voice?) |
| 06:04.58 | *** join/#asterisk vinkoval (n=vinkoval@c-24-10-180-226.hsd1.ut.comcast.net) |
| 06:05.03 | dlynes_ | No, I meant why do they have to pay the normal local phone company line rates? |
| 06:05.14 | dlynes_ | Pageus: don't they get them at a competitive wholesale rate? |
| 06:05.45 | dlynes_ | or sell their own product? |
| 06:05.46 | Pageus | oh.. they quoted us a flat rate of 350 a month for service plus 2.5cent/min LD. when they couldn't get the channelbank to work they had to come up with a solution |
| 06:06.37 | dlynes_ | ah...so maybe they're giving you a fractional voice t1 then |
| 06:06.42 | Pageus | the company is on Indian owned land.. so the local company has no issue with charging them normal rates and not cutting them a deal |
| 06:07.06 | Pageus | ok what is the difference between that and a frac data t1? |
| 06:07.38 | dlynes_ | a fractional voice t1 is usually 8-12 analog lines |
| 06:07.45 | Pageus | hmm |
| 06:08.14 | dlynes_ | also with t1, you get echo from what i understand |
| 06:08.17 | dlynes_ | with pri, you don't |
| 06:08.36 | Pageus | ack.. i hope not |
| 06:08.37 | Pageus | lol |
| 06:08.50 | dlynes_ | well, that's why you use echo cancellers |
| 06:08.55 | Pageus | ahh ok |
| 06:09.02 | Pageus | that i understand |
| 06:09.03 | Pageus | :) |
| 06:09.12 | vinkoval | Does anyone work with any Mitel stuff? I'm looking for information about Mitie API.. |
| 06:09.21 | dlynes_ | vinkoval: heh |
| 06:09.29 | vinkoval | hey |
| 06:09.33 | dlynes_ | vinkoval: We've got sipura units going into mitel |
| 06:09.38 | dlynes_ | I hate mitel |
| 06:10.14 | dlynes_ | vinkoval: Did you try www.techtips.com? |
| 06:10.49 | DoktorGreg | has anyone gotten a cease and decist letter from using the signate free music on hold mp3's? |
| 06:10.50 | Pageus | another question i have.. should i just rebuild the system from a "howto" and scrap the aah system or stay with the aah system until i get a better understanding of it all.. it was nice that it Identified the t100p right off the bat. |
| 06:11.04 | vinkoval | Yea.. Me too.. My boss wants me to develop an application using Mitel.. He mentioned somthing about Mitie API... |
| 06:11.07 | dlynes_ | DoktorGreg: huh? |
| 06:11.16 | DoktorGreg | http://www.signate.com/moh.php |
| 06:11.32 | DoktorGreg | is it safe to use those files as music on hold? RIAA et al |
| 06:11.36 | snitt | morgen |
| 06:12.27 | dlynes_ | DoktorGreg: is their website running off their asterisk server? |
| 06:12.31 | Pageus | i guess it would depend on how large your company was and if they have pissed off teh RIAA lately |
| 06:12.46 | dlynes_ | DoktorGreg: afaik for their moh files |
| 06:13.04 | dlynes_ | DoktorGreg: it's legal to use them, as long as you own a copy of their book, or something like that |
| 06:13.14 | dlynes_ | DoktorGreg: as I recall, they had some kinda special licensing with it |
| 06:13.27 | dlynes_ | DoktorGreg: oh...nvm |
| 06:13.33 | dlynes_ | DoktorGreg: not those ones...those are free |
| 06:13.39 | dlynes_ | DoktorGreg: completely free |
| 06:13.40 | *** join/#asterisk bzbw (n=wlwzhang@68-190-223-129.dhcp.mtpk.ca.charter.com) |
| 06:14.08 | dlynes_ | DoktorGreg: why would you get a cease and desist letter for using those? |
| 06:14.19 | dlynes_ | DoktorGreg: the songs are hundreds of years old |
| 06:14.35 | dlynes_ | DoktorGreg: their copyright (if it existed) has long since expired |
| 06:14.40 | DoktorGreg | We get cease and decist letters when the employees play radio in store... |
| 06:14.48 | dlynes_ | lol |
| 06:14.54 | dlynes_ | from who? |
| 06:14.57 | DoktorGreg | I kid you not |
| 06:14.59 | DoktorGreg | Ascap |
| 06:15.03 | dlynes_ | hahahhaa |
| 06:15.08 | dlynes_ | that's freaking hilarious |
| 06:15.19 | Pageus | In store is different than music on hold |
| 06:15.36 | Pageus | at least the RIAA and ASCAP feel so |
| 06:15.41 | DoktorGreg | kk just checking |
| 06:15.59 | dlynes_ | our government had a bill tabled similar to the RIAA bill in the US |
| 06:16.08 | dlynes_ | It ended up getting thrown out |
| 06:16.19 | dlynes_ | The music industry is still pushing for it |
| 06:16.23 | Pageus | music on hold is a free service where a directed person is listening for a short time and has no control on how long they listen |
| 06:16.37 | dlynes_ | But a number of recording labels left the canadian music industry association in protest over it |
| 06:16.46 | dlynes_ | and now they're fighting the bill against the riaa |
| 06:16.49 | DoktorGreg | here is the freedom loving USA they are debating making it a 1 year in jail and civil asset forfeiture for copying windows |
| 06:16.51 | Pageus | whereas in store, a person can stay for an indefinite(sp) period of time and listen.. or even take a tape recorder in and record it |
| 06:17.06 | Pageus | lol |
| 06:17.13 | Pageus | more power to linux users |
| 06:17.31 | dlynes_ | DoktorGreg: I disagree |
| 06:17.48 | dlynes_ | DoktorGreg: if you pirate windows, you should be locked in the insane asylum |
| 06:17.54 | DoktorGreg | lol |
| 06:18.06 | Pageus | lol. |
| 06:18.18 | Pageus | but but.. why pay for a product that runs like a virus |
| 06:18.20 | DoktorGreg | well linux is gonna get a huge boost from it i think |
| 06:18.31 | dlynes_ | Linux already has |
| 06:18.47 | dlynes_ | Especially after Microsoft has started viciously defending their trademarks and copyrights |
| 06:19.08 | Pageus | i mean they throw virus releaser and creators in jail.. why not microsoft heads? |
| 06:19.18 | dlynes_ | China and Russia are in the ASP.... 's target scope |
| 06:19.32 | dlynes_ | Canada's 3rd or 4th on the list of the most wanted countries for pirating |
| 06:19.48 | DoktorGreg | because microsoft gives large bags of cash to people in DC |
| 06:20.10 | dlynes_ | nah...if it wasn't for windows |
| 06:20.13 | Pageus | just goes to show.. we have no government. we have a company that runs the government |
| 06:20.28 | dlynes_ | how could blonde secretaries gather up enough brain power to use a computer? |
| 06:20.34 | Pageus | ROFL |
| 06:21.12 | snitt | lol |
| 06:21.12 | dlynes_ | there's no chance in hell they'd be able to figure out how to use Linux |
| 06:21.19 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 06:21.30 | kristalino | i wonder how much piracy helps windows on the marketing topic |
| 06:21.32 | Pageus | secretaries like that arn't made for running computers damnit.. there there to look cute, answer phones in a sexy voice.. and flirt with the boss |
| 06:21.49 | Pageus | yeah.. see microsoft never looks at that aspect of it.. |
| 06:21.53 | DoktorGreg | Pageus, you took the words out of my mouth |
| 06:22.02 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 06:22.08 | dlynes_ | the piracy of windows has really helped the us government marketing campaign |
| 06:22.22 | dlynes_ | the marketing campaign of america, the free, america the proud, america the ... |
| 06:22.25 | Pageus | there are alot of music artists that look at copying of there music as a great marketing tactic. |
| 06:22.38 | dlynes_ | i.e. the whole american branding thing |
| 06:22.54 | dlynes_ | and then all the immigrants land in the us, and find out life isn't a bowl of cherries after all |
| 06:22.55 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 06:23.25 | DoktorGreg | latino immigrants are planning a national strike this summer |
| 06:23.36 | Pageus | you figure if they put enough ppl in jail.. wait.. where did all the tax dollars go.. oh yeah.. they can't pay as much as the normal person so all of a sudden the tax dollars go down. |
| 06:23.48 | dlynes_ | they're getting tired of all the pot getting confiscated at the border? |
| 06:24.18 | Pageus | wait.. i saw a movie about that once.... where all the latino's or was it hispanics .. anyway.. they all disapeared from california |
| 06:24.38 | dlynes_ | Yeah...I think...I think....I think that was a Cheech and Chong movie, man |
| 06:24.42 | Pageus | ROFL |
| 06:24.46 | *** part/#asterisk vinkoval (n=vinkoval@c-24-10-180-226.hsd1.ut.comcast.net) |
| 06:24.57 | dlynes_ | I was born in East L.A. |
| 06:25.12 | dlynes_ | Heh...Chong's actually from here :) |
| 06:25.23 | DoktorGreg | well i think they will be successful with their strike |
| 06:25.24 | Pageus | naw.. was actually a good movie.. and somehow i doubt it ever got the viewing that it should have |
| 06:25.29 | dlynes_ | He's an original BC pothead |
| 06:25.30 | Pageus | well yeah |
| 06:25.35 | Pageus | lol |
| 06:26.04 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 06:26.10 | dlynes_ | Cheech was probably one of the draft dodgers that came up here, to perfect hydro weed :) |
| 06:26.19 | Pageus | who else is going to pick the fruit.. and do all the work that none of us thinskinned ppl want to do |
| 06:26.20 | tparcina | gooooooood morning! |
| 06:26.29 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:26.32 | dlynes_ | .hr = honduras? |
| 06:26.32 | tparcina | how is everybody? |
| 06:26.41 | Pageus | morning.. what continent are you on? |
| 06:26.44 | tparcina | not even cloose! |
| 06:26.50 | dlynes_ | Hungary? |
| 06:26.57 | tparcina | hnugary is hu |
| 06:26.58 | dlynes_ | no...wait |
| 06:27.00 | dlynes_ | yeah |
| 06:27.17 | dlynes_ | Herzogovina? |
| 06:27.19 | tparcina | on english my country starts with cr..... |
| 06:27.28 | dlynes_ | oh |
| 06:27.32 | dlynes_ | Croatia? |
| 06:27.32 | Pageus | croatia(sp)? |
| 06:27.34 | Pageus | lol |
| 06:27.41 | mut | sweet |
| 06:27.44 | tparcina | herzegovina isn't country, its region in bisnia and hercegovina (bih) |
| 06:27.57 | tparcina | yes, croatia! |
| 06:27.57 | mut | this sangoma card is working spiffingly |
| 06:28.01 | mut | got a dialup connection thru my ata @ 31.2kbps |
| 06:28.11 | dlynes_ | ah...but the tld came from 'Herzegovina'? |
| 06:28.14 | tparcina | pegeus, what (sp) means |
| 06:28.27 | dlynes_ | tparcina it means he's not sure about the spelling |
| 06:28.49 | tparcina | thanks dlynes_ |
| 06:29.04 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 06:29.08 | dlynes_ | but herz is why croatia is '.hr'? |
| 06:29.24 | tparcina | because it's Hrvatska |
| 06:29.27 | dlynes_ | ah |
| 06:29.33 | Pageus | i tend to use it alot.. lol.. spend to much time in IM's.. and such.. and being a programmer.. the words that i spell right.. arn't normally used in the english language |
| 06:29.56 | tainted- | it's amazing the diversity of people who come in here |
| 06:30.10 | dlynes_ | Yeah...the one guy i work on asterisk the most with, is originally from Croatia |
| 06:30.33 | Pageus | it's been close to 15 years since i logged into ANY IRC channel.. glad i managed to find one that wasn't just junk |
| 06:30.34 | dlynes_ | tparcina: do you know Alex Pavlovic? |
| 06:30.46 | brookshire | hey! i think i know him |
| 06:30.53 | tparcina | dyles_, no, I don't know him |
| 06:30.57 | tparcina | where is he from? |
| 06:31.05 | dlynes_ | Originally from Croatia |
| 06:31.09 | brookshire | somewhere near russia right? |
| 06:31.11 | dlynes_ | He lives in Vancouver, now |
| 06:31.29 | dlynes_ | brookshire: yeah, if you consider several thousand miles to be close :) |
| 06:31.34 | tainted- | Pageus a/s/l? |
| 06:31.34 | Pageus | ROFL |
| 06:31.49 | brookshire | lol |
| 06:31.51 | Pageus | 31/m/english |
| 06:31.54 | Pageus | eer |
| 06:31.57 | Pageus | sorry |
| 06:31.57 | tparcina | brookshire, no where near russia. on your map, look right from italy |
| 06:32.00 | Pageus | AZ |
| 06:32.02 | Pageus | ROFL |
| 06:32.10 | dlynes_ | Yeah...it's across the inlet from the italian boot |
| 06:32.12 | brookshire | oh well.. don't know him |
| 06:32.14 | brookshire | :D |
| 06:32.18 | tainted- | wanna cyber?!!!!?~ |
| 06:32.23 | Pageus | always get a kick out of the a/s/l calls |
| 06:32.24 | Pageus | ROFL |
| 06:32.48 | dlynes_ | I still don't even know what the hell a/s/l means |
| 06:32.55 | tainted- | age/sex/location |
| 06:33.00 | dlynes_ | ah |
| 06:33.04 | Pageus | your just to sexy for me.. i could never.. |
| 06:33.05 | Pageus | lol |
| 06:33.06 | Pageus | ROFL |
| 06:33.09 | dlynes_ | so why'd pageus say 'english' for his location? |
| 06:33.15 | Pageus | god i hsould be drunk for this |
| 06:33.42 | tainted- | that's his preferred position |
| 06:33.52 | tainted- | there's enlgish/greek etc etc |
| 06:33.58 | Pageus | depending on what system i am on.. l stands for language or location |
| 06:34.00 | Pageus | lol |
| 06:34.16 | Pageus | hmm not a bad one there tainted.. will have to use that one next time |
| 06:34.24 | tainted- | this stuff is going way over dlynes's head |
| 06:34.27 | tainted- | haha |
| 06:34.30 | dlynes_ | nah |
| 06:34.36 | Pageus | old geeks have it bad.. lol. |
| 06:34.37 | dlynes_ | I'm just not as young as you kids |
| 06:34.56 | tainted- | how dare u expose the truth! |
| 06:34.59 | tparcina | question from yesterday, when I transfer call from sip phone, does it goes to context what is default context for that phone? I belive it does, just checking to be sure |
| 06:35.06 | brookshire | O RLY? |
| 06:35.09 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 06:35.15 | Pageus | i wish i was younger.. i would have kept computers as a hobby instead of as a carreer |
| 06:35.23 | dlynes_ | tparcina: whatever you've stated is your context= in your sip.conf |
| 06:35.26 | brookshire | pageus: haha.. amen to that! |
| 06:35.40 | mut | Pageus: and done what instead? |
| 06:35.59 | tparcina | dlynes_, thanks |
| 06:36.01 | dlynes_ | damn...i thought tainted was one of those kids :) |
| 06:36.07 | Pageus | at 31.. i really can't change my career.. if needed.. without going back to school |
| 06:36.20 | mut | what? |
| 06:36.22 | mut | yes you can |
| 06:36.25 | mut | pfft |
| 06:36.27 | tainted- | what! i navigated horrible DOS manuals with the best of them |
| 06:36.28 | mut | 31 is nothin |
| 06:36.28 | dlynes_ | no kidding |
| 06:36.31 | tparcina | Pegeus, you talk like you are 61 |
| 06:36.33 | brookshire | LOAD "*", 8,1 |
| 06:36.43 | dlynes_ | i know lots of peeps that go back to school when they're in their late 30's/early 40's |
| 06:36.48 | brookshire | hah.. i hate the kids who grew up with DOS |
| 06:36.50 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 06:36.50 | tainted- | Pageus what do u do |
| 06:36.54 | tparcina | and why you would like to change your carier? |
| 06:37.02 | tparcina | don't you like to chat with us? :)) |
| 06:37.04 | dlynes_ | brookshire: ummm...what you just typed was dos |
| 06:37.07 | Pageus | i would have prolly either stay'd on the track to become a minister.. or a social worker.. |
| 06:37.08 | dlynes_ | brookshire: cbm dos |
| 06:37.13 | brookshire | dlynes: no.. it was commodore! |
| 06:37.14 | Pageus | hell i even loved doing framing and roofing. |
| 06:37.17 | Pageus | lol |
| 06:37.21 | Pageus | Vic-20 here! |
| 06:37.29 | dlynes_ | brookshire: like i said...cbm dos |
| 06:37.32 | Pageus | nod |
| 06:37.47 | brookshire | *pffft* commodores had no disks ;) |
| 06:37.56 | dlynes_ | I started out on the Sinclairs and the Commodore Pets |
| 06:37.58 | brookshire | at least my first one |
| 06:38.03 | dlynes_ | ummmm |
| 06:38.04 | Pageus | I'm a v.p. and programmer for a web development company that just recently went out of one market and into another |
| 06:38.08 | brookshire | tape drive baby |
| 06:38.09 | tainted- | that's old |
| 06:38.13 | Pageus | yeah |
| 06:38.26 | dlynes_ | brookshire: load "*",8,1 means load the directory from the '8' (1541 floppy disk drive) |
| 06:38.27 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 06:38.28 | tparcina | Pegeus, what is v.p.? |
| 06:38.38 | tainted- | Greek-Boy <---- LOL |
| 06:38.44 | Pageus | Vice President |
| 06:38.44 | tainted- | that couldn't be better placed |
| 06:38.50 | brookshire | dly: o rly? haha.. i had to look it up |
| 06:38.55 | dlynes_ | brookshire: load "*",1,1 means load the directory from the tape drive |
| 06:39.06 | brookshire | ,8,1 could be a great name for a geek rock band |
| 06:39.07 | dlynes_ | brookshire: erm...nvm |
| 06:39.14 | tainted- | dlynes psssh.. 001101001110100100!!! |
| 06:39.14 | tparcina | geee Pegeus, will you hire me? |
| 06:39.15 | dlynes_ | brookshire: tape drives didn't have directories |
| 06:39.25 | Pageus | ROFL |
| 06:39.43 | tainted- | think ur old school |
| 06:39.47 | tainted- | my keyboard has two keys |
| 06:39.58 | brookshire | omg.. you have a keyboard? |
| 06:40.12 | tainted- | and i type by stomping my left foot |
| 06:40.12 | dlynes_ | heh...remember the old altairs? |
| 06:40.13 | Pageus | part of the reason i am in a crisis with this asterisk system is that the president of the company had a heart attack and were all moving to home offices to cut back on his stress and increase our profits right now since were starting out clean in a new field |
| 06:40.20 | dlynes_ | 8 toggle switches at the front :) |
| 06:40.25 | tparcina | ok, I'll buy your company and fire you :=) |
| 06:40.32 | Pageus | hehe please do |
| 06:40.32 | brookshire | asterisk should be ported to the commodore |
| 06:40.33 | Pageus | lol |
| 06:40.37 | Pageus | wait.. the company already sold |
| 06:40.38 | Pageus | lol |
| 06:40.45 | Pageus | and then was repurchased |
| 06:40.53 | Pageus | can't fire an owner |
| 06:40.54 | Pageus | ROFL |
| 06:40.57 | tainted- | that sounds familiar |
| 06:40.59 | Greek-Boy | lol |
| 06:41.15 | tainted- | sell during dot com boom, re-buy when shares plummet |
| 06:41.21 | Pageus | been with them for 7 years.. LONG years.. |
| 06:41.26 | Pageus | never went public.. |
| 06:41.29 | Pageus | and survived.. |
| 06:41.44 | Pageus | and we were single style built |
| 06:41.48 | Pageus | all in the RV industry |
| 06:41.59 | Pageus | sold that part off.. |
| 06:42.03 | tainted- | whenever someone says 'can u design a simple ecommerce site with just like registration and shopping cart' i cringe |
| 06:42.09 | Pageus | ROFL |
| 06:42.12 | brookshire | hehe |
| 06:42.19 | tainted- | 'i heard from a friend that it's like 6.00 and takes two days' |
| 06:42.19 | brookshire | do you want flash with that? |
| 06:42.43 | brookshire | 6.00 per sale maybe! ;) |
| 06:42.49 | Pageus | i have built to many of those. have one that i built 5 years ago that is still running solid though so i'm happy.. only one customer though |
| 06:43.17 | *** join/#asterisk sfollo81 (n=stefano@81.88.224.6) |
| 06:43.40 | *** join/#asterisk DarthClue (n=DarthClu@adsl-69-152-249-37.dsl.snantx.swbell.net) |
| 06:43.40 | Pageus | see the old timer programmers are starting to enjoy the relaxation of using a gui to build stuff.. but at least we know how to fix what the gui borked in the backend |
| 06:43.58 | tainted- | i used to build websites in thedraw |
| 06:44.07 | Pageus | LOL |
| 06:44.09 | Pageus | LOL |
| 06:44.11 | brookshire | gui? what.. i still use vi |
| 06:44.28 | Pageus | hehe i prefer pico/nano but use VI when i have to |
| 06:44.39 | Pageus | ascii based websited eh tainted |
| 06:44.48 | Pageus | bet they looked great on lynx |
| 06:44.49 | Pageus | lol |
| 06:44.57 | tainted- | good times |
| 06:45.04 | brookshire | heh |
| 06:45.12 | brookshire | asterisk in trunk has ascii art |
| 06:45.25 | Pageus | used to run an MBBS board back in the days.. |
| 06:45.31 | Pageus | fun as hell |
| 06:45.41 | brookshire | omg.. i remember those |
| 06:45.51 | Pageus | lol.. wait.. how many of u member ascii porn.. ROFL |
| 06:45.52 | brookshire | most of the boards around here were vbbs though |
| 06:45.53 | dlynes_ | All your bases are belong to VIM!!! |
| 06:45.58 | Pageus | lol |
| 06:46.05 | tainted- | what ascii porn was the best!!~ |
| 06:46.07 | dlynes_ | ewww |
| 06:46.09 | brookshire | pageus: ANSI was the big upgrade to ascii porn |
| 06:46.10 | dlynes_ | vbbs sucked |
| 06:46.19 | tainted- | remember the first animated ascii porn u saw |
| 06:46.22 | Pageus | very true.. the right spots had good color |
| 06:46.24 | tainted- | DAMNN |
| 06:46.28 | DarthClue | i used to use pico/nano, then i was forced to use vi, then someone tried to convince me to use emacs. whatever gets the job done. |
| 06:47.02 | dlynes_ | DarthClue: Left hand usually works better than the right hand |
| 06:47.07 | Pageus | the first animated one i saw i think i was 14 or so.. killed the connection on telix.. |
| 06:47.17 | Pageus | lol |
| 06:47.21 | tainted- | lol telix |
| 06:47.38 | dlynes_ | yeah...good old salt scripts |
| 06:47.42 | DarthClue | dlynes, unless you have properly trained yourself to be ambidextrious... |
| 06:48.34 | tainted- | wait telix was windows based |
| 06:48.39 | Pageus | i had some old plato terminals from a company that was upgrading there systems, i wrote a pascal program that dialed out to the bbs's and retransmitted the data to the terminal at my bedside.. then i incorporated telix salt scripts to handle different things |
| 06:48.42 | dlynes_ | tainted-: it was dos based |
| 06:48.48 | Pageus | telix was dos based originally |
| 06:48.51 | dlynes_ | tainted-: it was later guified and ported to windows |
| 06:48.55 | tainted- | really? i remember running it in a window |
| 06:49.13 | dlynes_ | tainted-: it didn't run on windows until v4.xx |
| 06:49.19 | tainted- | oh that's right |
| 06:49.28 | tainted- | there was another popular package |
| 06:49.33 | tainted- | by qualcomm wasn't it? |
| 06:49.34 | dlynes_ | procomm plus |
| 06:49.38 | dlynes_ | and i still use it |
| 06:49.42 | Pageus | lol |
| 06:49.43 | dlynes_ | for logging into phone systems |
| 06:49.45 | tainted- | really? |
| 06:49.46 | Pageus | wow that is old |
| 06:49.50 | tainted- | yea that's old |
| 06:49.53 | Pageus | i use my tandy model 100 for that |
| 06:50.00 | Pageus | laptop.. and very portable |
| 06:50.10 | Pageus | 6 AA's and a cable.. |
| 06:50.10 | dlynes_ | or hyperterminal...whichever's on the laptop i'm using |
| 06:50.20 | tainted- | did u guys ever use xtreepro |
| 06:50.28 | dlynes_ | tainted-: xtree gold? |
| 06:50.29 | tainted- | or pctools |
| 06:50.29 | Pageus | god how old are you tainted |
| 06:50.30 | Pageus | lol |
| 06:50.38 | tainted- | hex editors and stuff |
| 06:50.39 | Pageus | here here |
| 06:50.48 | dlynes_ | pctools is pretty new, actually |
| 06:51.03 | Pageus | was referring to the other stuff he was taling about |
| 06:51.05 | dlynes_ | It didn't come out until ms-dos v3.xx or something |
| 06:51.22 | tainted- | i think dlynes is the oldest among us |
| 06:51.27 | tparcina | i need help with dialplan: my outgoing call from sip phone goes to "sip" context. when I go avay I transfer all incoming calls to my mobile. the problem is that I can't transfer incoming call to my mobile because I need to change his caller id to make it goe thrue. how to solve this? |
| 06:51.41 | dlynes_ | Well, i've used cp/m-80, if that gives you a reference point |
| 06:51.41 | Pageus | i used pctools to hack the text in my ms-dos boot to make it look like my own os.. lol.. |
| 06:52.09 | dlynes_ | had cp/m-80 on my apple ][+ |
| 06:52.13 | tainted- | i remember having nightmares about TSR programs taking up my memory |
| 06:52.24 | dlynes_ | and used it on the Osborne II as well |
| 06:52.36 | tainted- | WHERE THE FUCK IS MY EXTENDED MEMORY~!!~ |
| 06:52.41 | Pageus | 8" floppy's and 8MB hds that sounded like a car that was revving up it's engine |
| 06:52.49 | dlynes_ | snicker |
| 06:53.09 | tainted- | yea the ferrari drives? |
| 06:53.16 | dlynes_ | tparcina: yeah...I ran into that problem today, myself |
| 06:53.20 | DoktorGreg | ahhh getting more memory through expanded memory controllers, and 64k windows.... |
| 06:53.33 | Pageus | i wish i could help u there tparcina.. why would you need to change the callerid.. when i have my voicepulse transfer the calls to my cell phone i never had a problem with it.. |
| 06:53.39 | tparcina | dlynes-, and, have you solve it? |
| 06:53.43 | Pageus | then again i don't care what the callerid said. |
| 06:53.57 | dlynes_ | tparcina: well...not quite the same problem |
| 06:54.00 | Pageus | tainted.. yes |
| 06:54.15 | dlynes_ | tparcina: i ended up having to use a certain method to change his caller id |
| 06:54.25 | tparcina | pegeus, i must change it because it's E1 interface and my provider demands specific caller id |
| 06:54.37 | Pageus | hmm.. this is one of the few channels where i can say 8" floppy and not get a bunch of sick jokes.. |
| 06:54.37 | dlynes_ | tparcina: only one way worked...the others would act like they were dialing sip, but then autocongest |
| 06:54.38 | Pageus | nice |
| 06:54.55 | DoktorGreg | I speced a system... nothing special really, that is the same as 1,000,000 3.5 inch floppy disjs |
| 06:55.12 | tainted- | remember when u discovered that u could double your floppy disk capacity by punching a hole in it? and then rummaging through your whole collection trying optimize your storage space? |
| 06:55.17 | Pageus | 720k or 1.44? |
| 06:55.21 | Pageus | or 2.8? |
| 06:55.21 | DoktorGreg | take that back, 1,500,000 floyyy disks |
| 06:55.25 | DoktorGreg | 1.44 |
| 06:55.41 | DoktorGreg | oh wait |
| 06:55.47 | DoktorGreg | original 1000000 was closer |
| 06:55.51 | dlynes_ | Pageus: yeah...i never did see any of those 2.88MB floppy dirves or disks for that matter :) |
| 06:56.02 | Pageus | 2.16mb? |
| 06:56.07 | Pageus | or is that gig |
| 06:56.08 | Pageus | hmm |
| 06:56.16 | Pageus | gb |
| 06:56.20 | DoktorGreg | its a server will have 1,5 terrabytes |
| 06:56.24 | Pageus | for the 1.44 |
| 06:56.34 | Pageus | missed a zero |
| 06:56.35 | Pageus | lol |
| 06:56.36 | DoktorGreg | for video storage |
| 06:56.38 | Pageus | nice |
| 06:56.43 | dlynes_ | there's a company called 1999 that makes a 64TB drive |
| 06:56.51 | DoktorGreg | and its all off the shelf |
| 06:57.01 | dlynes_ | and that was the biggest drive you could get in 1998 |
| 06:57.02 | DoktorGreg | nothing special |
| 06:57.07 | Pageus | we managed to change our 2 tb storage system to a 950+ system.. lol.. |
| 06:57.35 | dlynes_ | the price tag on it was $64,000USD |
| 06:57.37 | DoktorGreg | it holds the video cache for all the video cameras in all the stores |
| 06:58.07 | DoktorGreg | which holly cow are all streamed over the internet |
| 06:58.17 | Pageus | nice.. |
| 06:58.29 | Pageus | wouldn't it have been easier to use a flashcomm server |
| 06:58.35 | DoktorGreg | well they will be everywhere in next two years |
| 06:58.38 | Pageus | less space and faster transfer with more compression |
| 06:59.01 | DoktorGreg | as laws are passing in all the states that require a video camera pointed at any cc transaction |
| 06:59.08 | dlynes_ | or use a $300 dvr system |
| 06:59.32 | tainted- | DoktorGreg u scare me |
| 06:59.33 | brookshire | hopefully.. not at the cards |
| 06:59.57 | DoktorGreg | tainted-, why do I scare you? Im not the passign the regulations |
| 07:00.11 | DoktorGreg | iirc california already passed it |
| 07:00.17 | dlynes_ | let them pass it |
| 07:00.24 | tainted- | hmm |
| 07:00.28 | dlynes_ | I just won't pay by credit card when i go to the us then |
| 07:00.38 | tainted- | it does seem like there are cameras everywhere in california |
| 07:00.49 | Pageus | that there are |
| 07:00.50 | dlynes_ | well yeah |
| 07:00.54 | DoktorGreg | there are cameras everywhere already... |
| 07:00.56 | tainted- | dlynes_ u can't afford anything in the US |
| 07:01.08 | dlynes_ | all the actors and actresses ahve to get all their faces on the camera wherever they can |
| 07:01.10 | Pageus | i take it he's canadian? |
| 07:01.11 | Pageus | lol |
| 07:01.11 | dlynes_ | tainted-: eh? |
| 07:01.26 | DoktorGreg | thats ok, i cant affor the 10 a pack smokes in canada |
| 07:01.28 | dlynes_ | tainted-: dood...our currency is the strongest it's been against the greenback for quite some time |
| 07:01.32 | tainted- | what's that all aboot |
| 07:01.36 | brookshire | hey now, the canadian dollar is catching up to the US |
| 07:01.54 | DoktorGreg | no, its just that the dollar is weak |
| 07:01.56 | Pageus | that it is.. but i would still like to live in canada and get paid in US |
| 07:02.25 | tainted- | don't get me wrong, i love our baby seal clubbers to the north |
| 07:02.26 | dlynes_ | the us dollar is only worth about $1.15 Cdn now |
| 07:02.40 | dlynes_ | the euro is much stronger |
| 07:02.47 | DoktorGreg | we have had massive inflation down here last few years |
| 07:02.51 | Pageus | yes but the euro is going away |
| 07:02.59 | brookshire | where is it going? |
| 07:03.03 | mut | ugh just say it |
| 07:03.05 | brookshire | canada? |
| 07:03.06 | DoktorGreg | I get sticker shock whenever i go to grocery |
| 07:03.17 | mut | americans are superior and there is nothing anyone can say against it |
| 07:03.27 | tainted- | it used to be 1USD ~ 1.3CDN |
| 07:03.28 | Pageus | the different countries are wanting there own currency back.. was reading about it about a month ago |
| 07:03.30 | DoktorGreg | 4 fir a gallon of milk |
| 07:03.32 | tainted- | that was the best |
| 07:03.32 | dlynes_ | DoktorGreg: yeah...exactly....and we've been getting no inflation up here, with the exception of gasoline |
| 07:03.39 | Pageus | ugh |
| 07:03.44 | Pageus | GAS.. |
| 07:03.49 | brookshire | pageus: that is the worse thing they could ever do |
| 07:03.49 | Alystair | yeah gase |
| 07:03.52 | Pageus | 3 -4 bux a gallow |
| 07:03.53 | DoktorGreg | $18 for a lbs of steak |
| 07:03.53 | Pageus | gallon |
| 07:03.55 | Alystair | lol 109.9 |
| 07:03.56 | dlynes_ | It's almost $1.20 a litre here, now |
| 07:03.59 | dlynes_ | freaking sucks |
| 07:04.08 | brookshire | the euro is the single biggest reason to why the dollar sucks |
| 07:04.18 | Alystair | dlynes: you still pay more for soda ;D |
| 07:04.20 | tainted- | brookshire why? |
| 07:04.20 | DoktorGreg | 5 pack of bratworst 9 bucks |
| 07:04.35 | Pageus | as long as the beer is cheap though.... |
| 07:04.37 | Pageus | ;) |
| 07:04.38 | dlynes_ | Alystair: no i don't...i don't buy soda |
| 07:04.42 | brookshire | because everyone traded dollar for euro.. now there is too many dollars in the economy |
| 07:04.59 | tainted- | damn Mac buttons... so pretty i click on stuff even though i don't need to |
| 07:05.11 | Pageus | lol |
| 07:05.21 | brookshire | and the dollar is also no circulating as fast as it use to be |
| 07:05.25 | brookshire | s/no/not |
| 07:05.32 | DoktorGreg | I wonder what Bush's lawyers knows, that he insists all his clients pay him in gold bars |
| 07:05.33 | dlynes_ | Alystair: you're in Ontario? |
| 07:05.46 | Pageus | lol |
| 07:06.28 | brookshire | the reason the dollar was strong in the past, was because almost every country used it in trade with other countries... |
| 07:06.32 | Alystair | yep |
| 07:06.35 | Alystair | toronto! :D |
| 07:06.50 | DoktorGreg | well that and the US wasnt filled with the biggest group of loosers in all history |
| 07:06.55 | Alystair | You wanna fight about it? |
| 07:06.58 | dlynes_ | lol |
| 07:06.59 | dlynes_ | dood |
| 07:07.07 | dlynes_ | Canucks kick the Leafs' ass :) |
| 07:07.10 | DoktorGreg | but the boomers are the biggest group of loosers history has ever seen |
| 07:07.34 | Alystair | I don't follow hockey damn it |
| 07:07.36 | DoktorGreg | I say that as a boomer |
| 07:07.39 | Alystair | soccer is my thing ;) |
| 07:07.46 | dlynes_ | ah |
| 07:07.47 | DoktorGreg | Go Red Wings! |
| 07:07.50 | Alystair | and even that I have a hard time following |
| 07:07.56 | tainted- | Alystair can't u have your canuck citizenship revoked for that? |
| 07:08.17 | DoktorGreg | All I know is my town wins the big Hockey game just about every year! |
| 07:08.26 | dlynes_ | tainted-: I'm not a hockey fan either...was just trying to be an antagonist :) |
| 07:08.30 | tainted- | brookshire so folks trade in euros now? |
| 07:08.48 | *** join/#asterisk mightnare (n=maikeru@p6034-ipad01motosinmat.mie.ocn.ne.jp) |
| 07:08.51 | brookshire | all of europe does, and russia |
| 07:08.52 | tainted- | dlynes british columbians.. always incitin shit |
| 07:08.57 | dlynes_ | lol |
| 07:09.02 | DoktorGreg | And Iran does |
| 07:09.04 | DoktorGreg | too |
| 07:09.13 | DoktorGreg | which is the real reason Bush wants to nuke em |
| 07:09.21 | mightnare | has anybody tried configuring asterisk with sangoma a200 on japan's ntt lines? |
| 07:09.23 | tainted- | oh wait, it's the quebecois who wanted to split |
| 07:09.41 | tainted- | who care.. it's like 5 people arguing |
| 07:09.51 | dlynes_ | tainted-: bc, alberta, oregon, washington, and montana want to split apart and start their own country, too |
| 07:10.05 | tainted- | that's nonsene |
| 07:10.20 | DoktorGreg | Michigan wants to split and join canada... |
| 07:10.24 | tainted- | no one in washington wants to split |
| 07:10.40 | dlynes_ | well, it does make some sense...there's a lot of intercommerce between those states and provinces |
| 07:10.53 | dlynes_ | but it's mostly just a pipe dream from some idealists |
| 07:11.02 | DoktorGreg | We have a canadian legislature |
| 07:11.15 | dlynes_ | huh? |
| 07:11.27 | DoktorGreg | Governor of Michigan is a canadian |
| 07:11.36 | dlynes_ | eh? |
| 07:11.46 | dlynes_ | How can be governor, and be a Canadian citizen? |
| 07:11.50 | tainted- | i think california is about to join mexico though |
| 07:12.06 | DoktorGreg | dlynes, this is michigan!!! |
| 07:12.16 | dlynes_ | You mean he's originally from Canada, don't you? |
| 07:12.23 | DoktorGreg | she |
| 07:12.27 | dlynes_ | i.e. she's not Canadian anymore? |
| 07:12.31 | DoktorGreg | oh whats her name |
| 07:13.05 | dlynes_ | but michigan will never become part of Canada |
| 07:13.05 | DoktorGreg | JEnnifer Granholm |
| 07:13.06 | Pageus | well with arnie in office.. be glad it's not austria |
| 07:13.07 | Pageus | lol |
| 07:13.25 | dlynes_ | people in detroit would never give up their guns |
| 07:13.27 | brookshire | Arnold isn't from the US and he's governor of cali |
| 07:13.28 | tainted- | arnold is amazing |
| 07:13.37 | DoktorGreg | naw, the US is gonna break up the same way soviet union did |
| 07:13.39 | tainted- | he calls rival politicians 'bitches' and 'faggots' |
| 07:13.40 | dlynes_ | brookshire: yeah, but arnie is still a us citizen |
| 07:13.46 | DoktorGreg | KAtrina was our Chernobyl |
| 07:13.55 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 07:13.55 | Pageus | yes.. he became one. |
| 07:13.59 | dlynes_ | exactly |
| 07:14.19 | Pageus | but he still can't be president.. so no following in good ol' ronnies footsteps |
| 07:14.20 | DoktorGreg | its obvious now that DC isnt gonna sort things out |
| 07:14.22 | *** join/#asterisk lorinc (n=ang@caracas-3754.adsl.interware.hu) |
| 07:14.23 | DoktorGreg | never again |
| 07:14.40 | tainted- | i thought he was going to get the constitution amended |
| 07:14.44 | Pageus | ROFL |
| 07:14.55 | dlynes_ | never gonna happen |
| 07:15.03 | Pageus | he will terminate that part of the constitution.. |
| 07:15.10 | DoktorGreg | Constitution will never be amended again |
| 07:15.11 | dlynes_ | the good ol' boys don't want an immigrant getting in there :) |
| 07:15.16 | DoktorGreg | not in our lives anyhow |
| 07:15.17 | Pageus | OMG.. it's late.. i can't believe i thought that one was funny |
| 07:15.48 | Pageus | brb need some stiff meds and a good strong soda. |
| 07:15.55 | tainted- | arnie should use iax2 |
| 07:16.15 | tainted- | it does congress traversal way better than sip |
| 07:16.33 | brookshire | oh well... SIP happens |
| 07:16.36 | Pageus | back |
| 07:16.37 | Pageus | lol |
| 07:16.46 | tainted- | haha |
| 07:16.48 | tainted- | i'm gonna register that |
| 07:16.52 | tainted- | siphappens.com |
| 07:16.54 | tainted- | lol |
| 07:17.06 | Pageus | would be a good company name |
| 07:17.07 | tainted- | DAMN IT |
| 07:17.15 | Pageus | someone has it? |
| 07:17.16 | brookshire | sip-happens.com |
| 07:17.17 | brookshire | ?? |
| 07:17.24 | Pageus | lol |
| 07:17.31 | tainted- | NOOOOO |
| 07:17.34 | dlynes_ | iaxu.to |
| 07:17.41 | tainted- | voipwank registered to dlynes |
| 07:17.55 | Pageus | ROFL |
| 07:17.58 | brookshire | siphappens.org |
| 07:17.59 | brookshire | ;) |
| 07:18.10 | dlynes_ | faqueue.to |
| 07:18.11 | tainted- | iaxu.to is nice |
| 07:18.13 | tainted- | lol |
| 07:20.09 | dlynes_ | iaxu.to is available |
| 07:20.22 | dlynes_ | so is faqueue.to |
| 07:21.14 | Pageus | hmm just had something odd happen.. i dialed into one of my iax2 provider numbers with 2 phones.. one went to one extention.. and the other to voicemail.. |
| 07:21.20 | tparcina | in the dialplan, how to call variable that holds calleridname? and how to get out only first 4 letters from that varialbe? |
| 07:22.07 | mightnare | has anybody configured asterisk in japan? |
| 07:22.12 | tainted- | u weirdos |
| 07:22.12 | Pageus | and that number ONLY goes to one core extention |
| 07:22.13 | dlynes_ | ${CALLERID(name)}, ${CALLERID(name):0,4} |
| 07:22.17 | tainted- | ask some regular questions |
| 07:22.24 | Pageus | i just did |
| 07:22.35 | Pageus | hehe figured we had enough fun.. back to work |
| 07:22.37 | tparcina | thanks dlynes_ |
| 07:23.00 | *** join/#asterisk stoffell (n=stoffell@93-204.240.81.adsl.skynet.be) |
| 07:23.06 | dlynes_ | tparcina: actually...that might be :0:4 |
| 07:23.19 | dlynes_ | tparcina: Check the variables section on the asterisk wiki on voip-info.org to make sure |
| 07:23.20 | tainted- | how are u dialing the provider |
| 07:23.26 | tainted- | through pstn or dial() |
| 07:23.35 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.218.49) |
| 07:23.44 | mightnare | i always get a busy signal when calling up the number attached to the fxo port of sangoma's a200 |
| 07:24.09 | tainted- | mightnare how is that card? |
| 07:24.26 | Pageus | with my 2 cell phones |
| 07:24.34 | mightnare | it's an a200 with 2fxo and 2fxs ports |
| 07:24.35 | Pageus | one after the other one started ringing |
| 07:24.58 | mightnare | one of the fxo ports is connected to ntt pots |
| 07:25.07 | dlynes_ | btw...which modem cards were they that sucked again? |
| 07:25.14 | dlynes_ | the motorolas or the intels? |
| 07:25.16 | tainted- | Pageus the DID might be limited to one channel |
| 07:25.32 | brookshire | mightnare: sounds like you are using the wrong signaling :) |
| 07:25.38 | *** join/#asterisk TallAndy (n=sdad@82.163.6.213) |
| 07:25.41 | Pageus | was supposed to be 4 channels avail. |
| 07:25.42 | Pageus | hm |
| 07:25.50 | tainted- | Pageus voicepulse? |
| 07:25.53 | Pageus | yeah |
| 07:25.57 | Pageus | i have 2 numbers right now |
| 07:25.59 | mightnare | tainted-, i can't make any outgoing call using sip through the fxo port.. |
| 07:26.01 | tainted- | i've always had issues with their DIDs |
| 07:26.08 | Pageus | ahh |
| 07:26.18 | dlynes_ | Pageus: What's your root password and your ip address? |
| 07:26.19 | Pageus | i went with them cuz incomming is free.. |
| 07:26.24 | Pageus | ROFl |
| 07:26.25 | dlynes_ | Pageus: I'll take a look |
| 07:26.26 | mightnare | tainted-, must be the signalling i guess.. any suggestions? i'm using kewlstart right now... tnx |
| 07:26.31 | Pageus | umm |
| 07:26.41 | tainted- | does voicepulse support 729 yet |
| 07:26.43 | Pageus | 192.168.1.69 and gofuckyourself |
| 07:26.45 | Pageus | ROFL |
| 07:26.49 | tainted- | last i used them they only did gsm |
| 07:27.09 | Pageus | j/k tainted |
| 07:27.21 | Pageus | since you and i both know that i'm not THAT out of it |
| 07:27.22 | Pageus | lol |
| 07:27.23 | tainted- | mightnare try brookshire, i don't touch analog |
| 07:27.28 | dlynes_ | mightnare: did your provider tell you what kind of lines they are? |
| 07:27.32 | tainted- | i'm voiperati like that |
| 07:27.41 | Pageus | lol |
| 07:27.43 | dlynes_ | mightnare: i..e ground start, loop start, kewl start, or wink start? |
| 07:27.44 | brookshire | i'm not going help sangoma card though :) |
| 07:28.22 | KeX_WorX | hi |
| 07:28.34 | KeX_WorX | I've a question about the extension.conf / dialplan |
| 07:28.38 | dlynes_ | mightnare: kewl start is usually called loop start with disconnect supervision |
| 07:29.00 | mightnare | dlynes, i have no idea what NTT uses.. #-o |
| 07:29.00 | KeX_WorX | is it possible to get the value of callgroup/pickupgroup from a phone, in the dialplan ? |
| 07:29.04 | KeX_WorX | is this possible ? |
| 07:29.11 | KeX_WorX | or only with an agi script `? |
| 07:29.12 | tainted- | i asked voicepulse for an API a year ago |
| 07:30.02 | dlynes_ | mightnare: you can usually find out by monitoring the voltage when someone hangs up |
| 07:30.08 | tainted- | recently they released a wsdl with GetBalance() as the only function |
| 07:30.15 | Pageus | yeah.. i saw that |
| 07:30.24 | dlynes_ | mightnare: Hook up an analog phone to your trunk line |
| 07:30.31 | Pageus | and of course now you have to sign a new agreement with them or get restricted access come june |
| 07:30.40 | tainted- | i should write some code to provision DIDs via soap |
| 07:31.03 | mightnare | dlynes, then...? |
| 07:31.23 | Pageus | i have been poking around working on a way to integrate my flashcomm server and my coldfusion mx system to tie into the asterisk server |
| 07:32.35 | Pageus | give my customers a way to quickcontact our office through an interface that 98% of the computers that use windows already have |
| 07:33.11 | brookshire | pageus: head has a xml webservice thing built in |
| 07:33.16 | dlynes_ | mightnare: Just trying to find a good article for you on google, that will help explain it a bit better |
| 07:33.19 | brookshire | pageus: other option is manager |
| 07:33.47 | Pageus | hmm i'm running aah so havn't looked into those |
| 07:34.01 | brookshire | and by head.. i mean trunk |
| 07:34.17 | brookshire | trunk = development branch of asterisk.. the unstable stuff |
| 07:34.41 | tainted- | brettnem what!? |
| 07:34.46 | tainted- | brookshire i mean |
| 07:34.47 | dlynes_ | mightnare: http://computing-dictionary.thefreedictionary.com/ground+start |
| 07:34.51 | tainted- | xml webservice!? |
| 07:34.58 | brookshire | yes |
| 07:35.02 | tainted- | no way |
| 07:35.04 | brookshire | yes way |
| 07:35.06 | tainted- | SOAP? |
| 07:35.06 | dlynes_ | mightnare: you can also try typing in wink start, loop start, and kewl start into that search mechanism, too |
| 07:35.09 | brookshire | not soap |
| 07:35.14 | tainted- | NOOOOOOOOOOOOOOOOOOOOOOOOOOOOooooooooooooooooooooooooo |
| 07:35.15 | mightnare | oh ok... thanks very much |
| 07:35.20 | tainted- | why not soap? |
| 07:35.24 | mightnare | i'm trying other signallings now.. |
| 07:35.31 | brookshire | i guess it's soap like |
| 07:35.37 | brookshire | you don't have to use soap |
| 07:35.43 | brookshire | xml is xml is xml :) |
| 07:35.51 | tainted- | where where |
| 07:35.57 | brookshire | it's whatever xml works with ajax |
| 07:36.01 | tainted- | does it extend manager api? |
| 07:36.07 | dlynes_ | mightnare: you might want to make sure there weren't any errors loading your driver, and also make sure there weren't any errors initializing the interfaces under asterisk |
| 07:36.12 | tainted- | that just made absolutely no sense |
| 07:36.45 | dlynes_ | mightnare: I had a similar problem with my pri card, but it was usually because the libpri and/or zaptel code had some stability issues |
| 07:36.49 | brookshire | yes.. basically.. it's a xml interface to manager |
| 07:36.59 | tainted- | SWEET! |
| 07:37.06 | dlynes_ | xmlrpc |
| 07:37.24 | tainted- | are there docs yet? |
| 07:37.33 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:37.59 | brookshire | http://svn.digium.com/view/asterisk/team/oej/test-this-branch/doc/ajam.txt?rev=17044&view=markup |
| 07:37.59 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 07:38.00 | *** join/#asterisk bartpbx (n=bartpbx@u14-1.dsl.vianetworks.de) |
| 07:38.06 | bartpbx | hello from germany |
| 07:38.15 | Pageus | howdy |
| 07:38.17 | dlynes_ | goodbye from Canada |
| 07:38.22 | bartpbx | I have a litle question concerning iax realtime |
| 07:39.19 | bartpbx | I've found a patch providing: "asynchronous processing of IAX2 registrations and call setups (no more audio drops when using realtime)" is this a known problem using realtime? |
| 07:39.47 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:39.49 | bartpbx | or is only res_config_mysql affected by this Problem? |
| 07:40.08 | brookshire | bart: it maybe when dealing with 1000s of registrations |
| 07:40.21 | brookshire | i think trunk handles iax registrations better |
| 07:40.53 | dlynes_ | the res_config_mysql isn't under development regularly, either |
| 07:41.05 | dlynes_ | so it might be out of sync with the main asterisk tree |
| 07:41.13 | brookshire | no.. i suggest using odbc |
| 07:41.16 | brookshire | it's maintained |
| 07:41.29 | brookshire | and performs better |
| 07:41.33 | bartpbx | we're using odbc but by the patch mysql is mentioned |
| 07:41.36 | brookshire | (from what i've been told) |
| 07:41.43 | bartpbx | what do you mean with trunk? the generall trunking in iax or somthing special? |
| 07:41.58 | bartpbx | how does the iax trunking affect the registration process? |
| 07:41.59 | mightnare | dlynes, thanks very much for your help.. i've already tried kewlstart, groundstart, and loopstart.. |
| 07:42.05 | mightnare | ...still no go :~ |
| 07:42.06 | brookshire | trunk = svn development branch.. the unstable stuff |
| 07:42.19 | macTijn | re |
| 07:42.22 | dlynes_ | mightnare: yeah, and winkstart i think might be particular to pri's |
| 07:42.26 | brookshire | iax is not multithreaded in 1.2 |
| 07:43.12 | bartpbx | so this patch could be usefull for 1.2 |
| 07:43.20 | bartpbx | we have currentyl 200 active iax2 users |
| 07:43.30 | brookshire | also, if you're using iax trunking... as long as host is not 'dynamic' you shouldn't register |
| 07:43.59 | brookshire | only time you need to register is if the user is dynamic |
| 07:44.13 | nextime | mightnare : stupid question: are you using fxs_XX signalins or fxo_XX? |
| 07:44.56 | dlynes_ | nextime: he's got two fxs ports, and two fxo ports, so he should be using both |
| 07:45.25 | dlynes_ | mightnare: Here's a link that might be useful for you, assuming you can read nihongo: http://www.infoscience.co.jp/technical/voip/nokia/n_text02.html |
| 07:45.40 | stoffell | hm, queue announcements work perfect, but can I allow the caller to press 0 at any time to leave a voicemail for example.. ? |
| 07:45.45 | nextime | dlynes_ : right, are you combine fxo and fxs signaling with the right ports? |
| 07:46.04 | nextime | so, fxs ports with fxo_XX signaling and fxo ports with fxs_XX signaling? |
| 07:46.05 | dlynes_ | nextime: Me? I'm not using the sangoma.... mightnare is |
| 07:46.23 | nextime | dlynes_ sorry you right |
| 07:46.33 | nextime | mightnare... |
| 07:46.39 | mightnare | dlynes, the handset that i've attached to the fxs ports work fine... |
| 07:47.08 | dlynes_ | mightnare: did you see the link i gave you for InfoScience, Japan? |
| 07:47.26 | mightnare | dlynes, yes i'm checking it now.. thanks very much |
| 07:47.31 | dlynes_ | no problem |
| 07:47.34 | dlynes_ | It's all in Kanji |
| 07:48.01 | dlynes_ | mightnare: also, nextime has some suggestions for you |
| 07:48.36 | dlynes_ | nextime: I guess he's off again :) |
| 07:49.25 | tparcina | dlaynes_, i have solved my problem. maybe this can help you |
| 07:49.40 | dlynes_ | ? |
| 07:49.42 | tparcina | now i can transfer all incoming calls to call phone |
| 07:49.53 | tparcina | do you ned the solution? |
| 07:49.57 | dlynes_ | sure |
| 07:50.02 | dlynes_ | if someone else runs into the problem |
| 07:50.07 | dlynes_ | I can tell them how to solve it, too |
| 07:50.20 | tparcina | every sip phone has name "Lama - name last name" |
| 07:50.45 | mightnare | nextime, yes i think i got the right signalling combinations for the fxo and fxs ports.. |
| 07:50.57 | tparcina | and when somebody leaves the office, on phone he transfers all calls to his mobile |
| 07:51.16 | mightnare | nextime, the handset that i've connected to the fxs ports can make calls to sip phones |
| 07:51.22 | bartpbx | brookshire: so you don't thik iax registration can cause audio dropouts when using realtime + odbc? |
| 07:52.21 | tparcina | when incoming calls comes, dialplan checks does it have "Lama" as first four names on his callerid(name), if it doesn't have than callerid(name) is changed to "External call" and callerid(number) is changed to number that is allowed by my provider to call out |
| 07:52.47 | tparcina | do oyu get it? |
| 07:52.51 | dlynes_ | Yeah |
| 07:52.53 | tparcina | do you get it? |
| 07:53.01 | dlynes_ | Your provider insists on your caller id being a certain value |
| 07:53.04 | dlynes_ | Otherwise you can't dial out |
| 07:53.13 | tparcina | yes |
| 07:53.27 | TallAndy | I have a SIP 400 error question with a circuit busy error. |
| 07:53.29 | dlynes_ | Is that for pri calls, or voip calls? |
| 07:53.38 | tparcina | for pri - E1 |
| 07:54.04 | dlynes_ | tparcina: yeah...some providers don't allow you to fake caller id values, unless they have those caller id values registered |
| 07:54.05 | tparcina | once more, thank you, you have been weary helpfull |
| 07:54.30 | brookshire | bart: it could if your db is on the same box |
| 07:54.37 | tparcina | now I need to solve some other problem, talk to you (I hope soon :)) |
| 07:54.42 | dlynes_ | TallAndy: sip 400 can mean numerous different things |
| 07:54.44 | brookshire | i mean.. anything is possible |
| 07:55.02 | dlynes_ | TallAndy: You usually need to look at your sip debug to see what the real problem is |
| 07:55.05 | TallAndy | yeh its the cli thats causing a problem |
| 07:55.11 | TallAndy | i'm bridging 2 calls |
| 07:55.41 | TallAndy | e.g call 1 -> cli = 111, target = 222 |
| 07:55.53 | TallAndy | call2 (Bridge) cli = 222, target = 111 |
| 07:56.01 | dlynes_ | eh? |
| 07:56.05 | TallAndy | if i change the cli to something different in the 2nd call it works fine |
| 07:56.05 | dlynes_ | what's this 111 and 222? |
| 07:56.05 | bartpbx | ok. I'll setup a milliwatt test and test it. currently the db is on the same server |
| 07:56.14 | TallAndy | just as an example |
| 07:56.15 | bartpbx | but the load ist normaly around 0,1 |
| 07:56.17 | TallAndy | it would be a real number |
| 07:56.28 | bartpbx | with 20 concurrent calls |
| 07:56.31 | dlynes_ | TallAndy: but how are the 111 and 222 related? |
| 07:56.43 | TallAndy | they need to be paired up for billing purposes |
| 07:56.46 | dlynes_ | TallAndy: i.e. what is a "target"? |
| 07:56.53 | dlynes_ | and what is a "cli"? |
| 07:56.57 | TallAndy | target is the person we are calling |
| 07:57.08 | dlynes_ | ok |
| 07:57.09 | dlynes_ | and cli? |
| 07:57.13 | tainted- | dlynes i think that's telemarketing speak |
| 07:57.17 | tainted- | command line interface |
| 07:57.18 | TallAndy | cli is the inital requestor of the phone call |
| 07:57.26 | dlynes_ | ah |
| 07:57.31 | TallAndy | sorry to confuse u :) |
| 07:57.32 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 07:57.33 | dlynes_ | why not just say sip A and sip B? |
| 07:57.59 | dlynes_ | you start saying cli, and i'm thinking command line interface |
| 07:58.04 | TallAndy | sure :) |
| 07:58.06 | gr0mit | hi - is anyone familiar with configuring chan_ss7 ? |
| 07:58.29 | dlynes_ | TallAndy: anyways...that's what a bridge normally is |
| 07:58.32 | TallAndy | do u see why i get the problem now, it thinks the channel is in use coz i've already used sip a |
| 07:58.43 | bartpbx | can anyone point me to a simple testscript to make iax registrations |
| 07:58.43 | TallAndy | the number for it |
| 07:59.16 | dlynes_ | TallAndy: no , i don't see the problem, unless you've got incominglimit=1 or outgoinglimit=1 on the sip.conf file for that phone |
| 07:59.27 | TallAndy | right |
| 07:59.36 | dlynes_ | you've got those set, then? |
| 08:00.02 | TallAndy | just looking |
| 08:00.11 | dlynes_ | TallAndy: also, even if those are set |
| 08:00.19 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 08:00.28 | dlynes_ | TallAndy: you shouldn't be getting sip 400, circuit is busy here |
| 08:00.38 | dlynes_ | You should be getting ummmm |
| 08:00.47 | dlynes_ | Trying to recall the sip error now |
| 08:00.59 | dlynes_ | But the text is 'Busy here', not 'circuit is busy' |
| 08:01.09 | dlynes_ | It's something like 403 or something |
| 08:01.32 | TallAndy | one sec |
| 08:01.35 | dlynes_ | If you're getting 400, that's a legitimate error |
| 08:01.45 | dlynes_ | it means something's gone horribly wrong |
| 08:02.16 | TallAndy | right |
| 08:02.20 | TallAndy | hmmm |
| 08:02.27 | TallAndy | stranger |
| 08:03.45 | dlynes_ | TallAndy: I suggest typing sip debug peer 111, or whatever your peer is |
| 08:03.55 | dlynes_ | TallAndy: and looking at the debug output, to determine what's gone wrong |
| 08:04.21 | TallAndy | thanks dlynes_ |
| 08:04.49 | *** join/#asterisk heka (n=Mango@80.80.175.130) |
| 08:05.50 | dlynes_ | afk for a while |
| 08:07.17 | *** join/#asterisk pif (n=ldm@sargon-nerim.lncsa.com) |
| 08:07.26 | dlynes_ | btw, Pageus ... you still having problems with the voice t1? |
| 08:07.30 | pif | hi, what is the best asterisk manager? |
| 08:07.51 | dlynes_ | pif: define "asterisk manager" |
| 08:08.18 | pif | a client that allows me to control channels |
| 08:08.29 | dlynes_ | pif: asterisk -r |
| 08:08.55 | pif | I'd lin |
| 08:09.03 | dlynes_ | pif: or do you mean from a receptionist's perspective? |
| 08:09.19 | pif | yes |
| 08:09.27 | dlynes_ | pif: Flash operator panel |
| 08:09.48 | pif | it leaks memory like there's no tomorrow |
| 08:09.49 | bartpbx | does anyone konws the status of astertest? |
| 08:09.58 | dlynes_ | pif: so fix it, then |
| 08:10.04 | pif | no |
| 08:10.08 | dlynes_ | pif: and submit your patches back to the development team |
| 08:10.08 | heka | Hello, what can I use instead of hangup to disconect the callers line and continue to execute extensions? |
| 08:10.17 | pif | I'm not a programmer |
| 08:10.28 | dlynes_ | pif: well, you asked what the best one was |
| 08:10.30 | dlynes_ | pif: I told you |
| 08:11.23 | dlynes_ | heka: soft hangup is the only one i know of |
| 08:12.32 | heka | dlynes_: Im trying to make a callback, and I have to drop the line and callback the caller! |
| 08:12.36 | heka | any way to do that? |
| 08:15.38 | bartpbx | I could still not find a tool to test iax2 registration load |
| 08:15.55 | bartpbx | does anyone has souch a tool |
| 08:16.02 | bartpbx | anything like sipp for iax? |
| 08:16.02 | zoa | yes |
| 08:16.06 | zoa | i made something like that before |
| 08:16.40 | bartpbx | are you the zoa from astertest? |
| 08:17.04 | zoa | http://www.astertest.com/downloads/iax_mass_authorization.tgz |
| 08:17.09 | zoa | yes |
| 08:17.25 | bartpbx | is the work on astertest still going on? |
| 08:17.31 | zoa | not for the moment no |
| 08:17.38 | zoa | too much other things to focus on |
| 08:18.02 | bartpbx | ok. |
| 08:19.34 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
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| 08:26.32 | mightnare | gtg... thanks guys, thanks dlynes and nextime =) |
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| 08:31.02 | pif | what is the iax trunk monitoring command again? I'm trying "iax2 trunk debug" |
| 08:31.17 | pif | to check whether trunking is active |
| 08:32.37 | zoa | its iax2 trunk debug yes |
| 08:32.47 | pif | thanks |
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| 08:52.22 | Sebb | florz: there are some changes in bristuff-0.3.0-PRE-1o that zaphfc_0.3.0-PRE-1f_florz-11.diff.gz won't apply anymore (rejects in zaphfc.c, not just in the makefile.. ;) |
| 08:57.56 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 09:02.45 | Snake-Eyes | does asterisk convert DTMF, eg IP phone uses inband, asterisk is set to RFC2833 for trunk and VSP uses RFC2833. I have this setup and DTMF works nicly.. |
| 09:02.45 | kmilitzer | Is there a known issue with the detection of rtp-timeout? |
| 09:03.50 | kmilitzer | I have set rtptimeout in sip.conf to 30, started a call that way UAC->SER->Asterisk1->Asterisk2->PSTN(SS7) and then did a killall -9 asterisk on Asterisk1 |
| 09:04.17 | kmilitzer | IMHO the call should then be terminated after 30 seconds on Asterisk2, but it isn't :( |
| 09:11.09 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
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| 09:12.21 | *** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au) |
| 09:14.47 | Thazza | Hey all. |
| 09:16.36 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-38-dsl.end.estpak.ee) |
| 09:18.56 | viperdude | kmilitzer: are you still there? |
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| 09:23.27 | *** part/#asterisk syneus (n=syneus@81.88.224.6) |
| 09:25.10 | kmilitzer | viperdude: Yes |
| 09:25.38 | viperdude | hi, i was just wondering about your SER - Asterisk setup does it work well? |
| 09:26.48 | kmilitzer | Yes, sure ... I use SER as a registrar server and asterisk for call distribution and VM, etc |
| 09:27.07 | viperdude | so Asterisk is like the media proxy? |
| 09:29.19 | kmilitzer | Yes, kind of ... techically its a b2bua, so all calls are accounted and i am able to monitor calls for lawfull interception and so on |
| 09:30.32 | viperdude | are you accounting on SER or Asterisk? I have played around with SER but the CDR's produced are not as good as Asterisk |
| 09:32.35 | Thazza | Has anyone been able to make a call via a IAX connection, and this redirects via asterisk, and goes out via a SIP connection? |
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| 09:36.40 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:37.37 | puzzled | morning |
| 09:39.13 | tparcina | good day puzzled |
| 09:40.00 | tparcina | viperdude, you are counting from cdr that are stored in mysql, right? |
| 09:40.18 | viperdude | yes |
| 09:40.30 | tparcina | how acurate that data is? |
| 09:40.48 | viperdude | yes |
| 09:41.05 | tparcina | yes, like 100% acurate? |
| 09:41.49 | viperdude | yes and also able to tell total time and billable time etc |
| 09:43.14 | tparcina | have you ever tried to use advice of charge (aoc), billing informations that provider sends o you? |
| 09:43.43 | viperdude | we dont have a provider we are the provider |
| 09:43.58 | tparcina | ooo.. |
| 09:44.39 | tparcina | do you need to reset timing, or you just use what asterisk gives you? |
| 09:45.04 | tainted- | viperdude how are you the 'provider' |
| 09:45.06 | tparcina | because i have noticed, that cdr shows that not established call has last for more than 10 sec |
| 09:45.36 | viperdude | yes thats the issue we need to know established time |
| 09:45.41 | tainted- | i guess it depends on your definition of provider |
| 09:45.50 | kmilitzer | viperdude: I do accounting on asterisk |
| 09:46.02 | tparcina | well, how doid you solve that ishue? |
| 09:46.12 | viperdude | so all calls are bridged through Asterisk? |
| 09:47.39 | kmilitzer | viperdude: Yes, right ... this way I can be sure, that all information travel over my asterisk ... |
| 09:47.41 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:48.20 | tainted- | anyone work with manager api? |
| 09:48.22 | viperdude | hmm i wanted only PSTN calls to go throgh Asterisk, looks like I might need on-net calls to go through Asterisk as well if i want the CDR |
| 09:49.32 | tainted- | what would be ur alternative |
| 09:49.47 | tainted- | SER? other gateway/proxy? |
| 09:50.26 | viperdude | i have done some maanger stuff |
| 09:51.04 | tainted- | do u 'connect as needed' or maintain connection |
| 09:51.26 | tainted- | wondering about performance hit |
| 09:51.28 | viperdude | we have astmanproxy running and everything connects to that |
| 09:53.04 | tainted- | meaning u send manager actions via http querystring? |
| 09:53.31 | tainted- | or meaning u cluster asterisk boxes with it |
| 09:53.44 | viperdude | no there is a program called astmanproxy that connects to the manager and proxies incoming and outgoing manager requests |
| 09:53.58 | viperdude | http://www.voip-info.org/tiki-index.php?page=AstManProxy |
| 09:58.48 | poisoner | Has anyone linked asterisk with an Cisco Call Manager 3.1? via H.323 ? |
| 10:01.49 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
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| 10:08.55 | puzzled | anyone heard of a Samsung WiFi SIP phone? |
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| 10:21.17 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 10:23.28 | Zeeek | hey ladies and germs... |
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| 10:31.34 | *** join/#asterisk Kazukichi (n=ryan@82-33-205-227.cable.ubr11.newt.blueyonder.co.uk) |
| 10:31.42 | Kazukichi | Blah, still having outbound issues. :/ |
| 10:34.24 | puzzled | hi Zeeek |
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| 10:35.02 | *** join/#asterisk cornet (n=nathan@cpc1-heck3-0-0-cust893.hudd.cable.ntl.com) |
| 10:35.31 | cornet | using the asterisk queue viewer (http://asterisk.toad.net/qview.pl) and its only returning one queue with asterisk 1.2.5 |
| 10:35.34 | cornet | um 1.2.6 |
| 10:38.02 | Zeeek | hi puzzled |
| 10:38.36 | Zeeek | Anyone around here heard anything about Nufone? (prolly too early, but I'm leaving soon) |
| 10:38.50 | Zeeek | ~seen JerJer |
| 10:38.54 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #asterisk, 5d 15h 43m 2s ago, saying: 'but that's cvs code from like pre-1.2'. |
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| 10:45.03 | Zeeek | anyone here in the USA/Canada at the moment? |
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| 11:07.20 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:17.22 | SparFux | How to hook flash a ZapHfc channel? |
| 11:18.38 | wasim | Flash() ? |
| 11:22.50 | SparFux | That does only work with FXO it says. Like analog lines. ZapHfc is a bri channel. |
| 11:23.00 | SparFux | I need an equivalent for capicommand(hold) |
| 11:23.10 | ManxPower | SparFux, Why do you need to FLASH? |
| 11:23.35 | SparFux | I have asterisk connected to a hardware pbx and want to use its functionality. Therefore I need it. |
| 11:24.15 | SparFux | I try this with zaphfc now. Before I tried capi, but it lacks the ability to send digits over the line. Not DTMF but data. |
| 11:24.18 | ManxPower | SparFux, I thought ISDN did it differently, like maybe using the Transfer() or a BRIStuff app. |
| 11:24.26 | SparFux | Like ZapSendKeypadFacility |
| 11:24.29 | macTijn | did anyone here read shelleytherepublican.com/2006/04/linux-european-threat-to-our-computers.html ? |
| 11:25.31 | SparFux | Manx: I don't know. With capi there is capicommand(hold) |
| 11:26.58 | Frogzoo | macTijn: this is pretty great propaganda "Linux is a computer program that was initially developed in Finland as a means of circumventing valuable copyrights and patents owned by an American company called SCO Group." |
| 11:27.33 | ManxPower | SparFux, have you tried the transfer() app? |
| 11:27.49 | ManxPower | Results 1 - 10 of about 221 from lists.digium.com for transfer bristuff. (0.38 seconds) |
| 11:28.27 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 11:29.01 | luke-jr_ | Any ideas why Asterisk would have silence on one or both ends of a SIP reinvite *randomly*? |
| 11:29.24 | luke-jr_ | eg, sometimes it works, other times it doesn't |
| 11:29.46 | luke-jr_ | (I guess a better word would be intremittantly) |
| 11:31.05 | wasim | luke-jr_: network? |
| 11:31.36 | ManxPower | luke-jr_, NAT |
| 11:31.44 | luke-jr_ | ManxPower: intermittantly... |
| 11:31.49 | ManxPower | luke-jr_, nat |
| 11:32.12 | luke-jr_ | that doesn't explain intermittant... >.> |
| 11:32.52 | ManxPower | If your NAT router has an unusually long translation cache, then as long as the phone uses the same RTP ports it will work. If the phone uses different RTP ports (which it is allowed to do) you would not get audio. |
| 11:33.14 | ManxPower | Also if you do port forwarding on the client NAT router this could happen if the phone uses different ports than you think it does. |
| 11:33.19 | luke-jr_ | there's no phone involved |
| 11:33.24 | viperdude | i am getting 2-3 sec audio drops anyone know where to start looking for cuase, the inbound calls from PRI and the outbound calls on VoIP |
| 11:33.33 | luke-jr_ | I'm reinviting an incoming call to an outgoing one on the same ITSP |
| 11:34.29 | ManxPower | viperdude, English is not your native language? |
| 11:34.56 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 11:35.01 | viperdude | I am British lived in England all my life |
| 11:35.14 | luke-jr_ | ManxPower: so you'd say it's definately a problem on the ITSP's end? |
| 11:35.19 | ManxPower | viperdude, then rewrite your question so we can understand it. |
| 11:35.50 | ManxPower | luke-jr_, I can't say anything. |
| 11:36.14 | luke-jr_ | well, you said "nat"... and the RTP stream post-reinvite is nowhere near my networks |
| 11:36.36 | luke-jr_ | theoretically, it should be looping back to the ITSP's Asterisk system |
| 11:36.37 | viperdude | lol |
| 11:36.52 | ManxPower | is the phone behind NAT? |
| 11:36.53 | viperdude | I am getting audip drops on my calls, what difficult about that? |
| 11:37.32 | luke-jr_ | ManxPower: the phones are both on POTS |
| 11:38.06 | ManxPower | viperdude, you said 2-3 sec audio drops. Well is that 2-3 audio drops per seconds, or audio drops every 2 - 3 seconds, or are you getting audio drops that last for 2-3 seconds |
| 11:38.31 | viperdude | audio drops that last for 2-3 seconds and are minutes apart |
| 11:38.40 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 11:38.46 | luke-jr_ | latency maybe? |
| 11:39.00 | tainted- | 2-3 drops seconds calls, pri iax but channel sip compile yes? |
| 11:39.01 | ManxPower | there! That wasn't so bad, was it? Sounds to me that you don't have enough bandwidth? |
| 11:39.15 | luke-jr_ | ManxPower: I'm getting an incoming SIP call from SellVoIP and Dial()ing another number via SellVoIP and reinviting those |
| 11:39.16 | ManxPower | tainted-, exactly |
| 11:39.30 | ManxPower | luke-jr_, then I have no idea what to suggest. |
| 11:39.38 | viperdude | err its a Gigabit network with 50 phones on it, network utilisation under 1% |
| 11:39.42 | tainted- | viperdude he's saying your english pipe is a tad teensy |
| 11:39.54 | luke-jr_ | ManxPower: no idea of anything on either end that might cause it? |
| 11:40.20 | ManxPower | tainted-, I don't mind if people have trouble with English if it's not their native language. |
| 11:40.20 | luke-jr_ | viperdude: tried Ethereal? |
| 11:40.34 | ManxPower | luke-jr_, since it has nothing to do with asterisk, no. |
| 11:40.45 | viperdude | not sure what I need to look for with Ethereal |
| 11:40.51 | tainted- | neither i nor myself esl problems |
| 11:40.57 | luke-jr_ | ManxPower: so it's ITSP's end? |
| 11:41.00 | ManxPower | viperdude, what codec are you using? |
| 11:41.07 | viperdude | ulaw |
| 11:41.09 | ManxPower | luke-jr_, as you said if audio isn't going thru asterisk..... |
| 11:41.14 | viperdude | cisco 7940 phones |
| 11:41.16 | ManxPower | viperdude, what bandsidth do you have? |
| 11:41.40 | luke-jr_ | ManxPower: and if the ITSP uses Asterisk, you'd suspect their network config? |
| 11:41.40 | viperdude | gigabit LAN and 200Mbs internet connect |
| 11:41.57 | viperdude | i am sure its not a bandwidth issue |
| 11:42.00 | SparFux | Manx: Transfer() doesn't seem to do the job with bri. |
| 11:42.03 | ManxPower | viperdude, I doubt it's a bandwidth issue then. 8-) What router? |
| 11:42.15 | viperdude | cisco routers we are a ISP |
| 11:42.23 | ManxPower | SparFux, and none of the 200+ google results were helpful? |
| 11:42.47 | viperdude | I think its either a interrence issue on the LAN or maybe interrups on digium card? |
| 11:42.48 | luke-jr_ | I wish I was an ISP ;) |
| 11:43.14 | luke-jr_ | viperdude: is the Asterisk running on a dedicated box? |
| 11:43.15 | ManxPower | viperdude, WHAT ROUTER are you using? |
| 11:43.32 | SparFux | Manx: I'll search. |
| 11:43.34 | tainted- | check the frequency demodulator |
| 11:44.36 | ManxPower | viperdude, "cat /proc/interrupts" will tell you what IRQ the Digium card is on. Digium cards CANNOT share interrupts with other cards. |
| 11:44.57 | viperdude | cisco 4000 switch and cisco 7513 router |
| 11:45.04 | poisoner | viperdude ever tried to link an ccm with asterisk? |
| 11:45.32 | viperdude | ecte11xp is on IRQ 25 and nothing else on 25 |
| 11:45.34 | ManxPower | viperdude, If you insist on using good equipment and have high bandwidth connections, I won't be able to blame the hardware. |
| 11:46.15 | viperdude | ManxPower: that is what I am thinking but I am getting complaints from the users so where do i look next? |
| 11:46.19 | ManxPower | viperdude, your problem is not common, except when really bad routers are used or when you are using ulaw and a 64k link. |
| 11:46.44 | ManxPower | viperdude, could there be some traffic shaping / QoS on the router or switch? |
| 11:46.44 | viperdude | i use ulaw on my DSL line for 3-4 calls without problem |
| 11:47.04 | ManxPower | viperdude, not on a 64Kbps DSL link you don't. |
| 11:47.06 | viperdude | the switches are running QoS on the vlans |
| 11:47.25 | ManxPower | viperdude, can you turn off QoS and see if that fixes things? |
| 11:47.28 | viperdude | I have a 8 meg DSL link |
| 11:47.38 | ManxPower | viperdude, exactly. |
| 11:47.39 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
| 11:48.02 | viperdude | thing is another part of the building has the same config on the switches but no audio drops |
| 11:48.06 | *** join/#asterisk startled (i=startled@d58-105-31-172.dsl.vic.optusnet.com.au) |
| 11:48.09 | ManxPower | Just remember that ulaw and alaw take about 80kbps |
| 11:48.28 | viperdude | could there be electrical interference on the CAT5/6 somewhere like a relay kicking in |
| 11:48.38 | viperdude | yeah I know |
| 11:48.44 | ManxPower | viperdude, that would be beyond the scope of this channel |
| 11:49.03 | luke-jr_ | I'd try Ethereal |
| 11:49.20 | ghost99 | hi all , does anyone have any hints about an x100P card not being recognised ....I have surfed and fiddled all day but no luck .. I really need a few hints if you know anything :) ? |
| 11:49.33 | ManxPower | viperdude, Either I'm terrible at doing QoS or Cisco's QoS is not very good because EVERYTIME I try to put QoS on our network, stuff breaks. |
| 11:49.53 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:49.55 | ManxPower | ghost99, talk to the company you bought the clone card from. |
| 11:50.12 | viperdude | ManxPower: interesting maybe disable QoS and retry |
| 11:50.35 | ManxPower | viperdude, I doubt it would make things any worse. |
| 11:51.11 | ghost99 | oh .. ebay and i have two cards .. but none recognised ... box had www.x100.com which is a useless site - The point being i knows these are just play cards but if they work them i'm all in to buy a real card :) |
| 11:51.22 | viperdude | we had a issue on monday where by a network cable had to be moved when a new electric plant in the building was switched on and the problem seems to be related |
| 11:51.29 | wasim | the real cards work, the play cards are only good to pee on |
| 11:51.32 | ManxPower | ghost99, talk to the company you bought the clone card from. |
| 11:53.06 | ManxPower | ghost99, Digium no longer makes X100P cards, and has not done so in over a year. |
| 11:56.03 | ghost99 | manzpower: fair statement but still apparantly the ones i have are originals and not clones ...?? or so they say on ebay |
| 11:56.21 | ghost99 | manxpower : typo on your name opps :) |
| 11:56.28 | gr0mit | hi - is anyone here using chan_ss7 with asterisk? |
| 11:57.16 | ManxPower | ghost99, best of luck. |
| 11:59.44 | ghost99 | manxpower: ... is there any particular reason that rather than say .. go here you saying bugger off and work it out ? ... surely there value in helping me if I'm prepared to play and then once proven buy the real gear ? or is there ppls that can telnet in take a look being that I'm no unix guru |
| 11:59.50 | ManxPower | That would be something for you to talk to John [the MIS manager] about, as I am not qualified to deal with users of your caliber. |
| 11:59.57 | ManxPower | Must resist sending |
| 12:00.55 | ManxPower | ghost99, because many companies are selling cards, claiming to be compatable and expecting support for them and digium doesn't get a dime from it. |
| 12:01.12 | ManxPower | If they are REAL digium cards then call digium, as there is obviously a hardware problem. |
| 12:01.15 | gr0mit | ghost99, what exactly is the problem you have with the X100P card? |
| 12:01.27 | ManxPower | If they are not real digium cards, then bugger off. We don't suppor clones. |
| 12:01.53 | viperdude | do you need to pay digium for telephone support on their hardware? |
| 12:02.02 | ghost99 | the ISO as downloaded does not detect the card .. re if i have a clne then fair call but am under the assumption that i don't ?? |
| 12:02.07 | ManxPower | viperdude, no, you do not. |
| 12:02.46 | gr0mit | ghost99, what do you see when you type lspci ? |
| 12:02.50 | viperdude | hmm ok I will give them a call about this audio drop just to make sure not the PRI card |
| 12:02.52 | ManxPower | Digium provides free tech support for their HARDWARE. If it's a software issue, like an issue with Asterisk, then you have to pay for support. |
| 12:02.59 | ghost99 | manxpower: i have tried 2 differnet cards and 2 p.cs ... just does not detect ?? which is why I'm here after some 10-12 hours of trying :( |
| 12:03.01 | viperdude | ok cheers |
| 12:03.32 | ManxPower | ghost99, as I said, contact Digium if you have real cards. Digium supports their cards. |
| 12:03.52 | ghost99 | manxpower is there any special way of telling if i do have a clone ? or real one ? |
| 12:04.07 | ManxPower | ghost99, but Digium will prolly make you build the zaptel drivers from scratch, so do that first. |
| 12:04.11 | ManxPower | ghost99, I don't know. |
| 12:04.21 | I-MOD | what's the serial number on it? |
| 12:04.28 | ManxPower | for one thing a real digium card should have a serial number sticker. |
| 12:04.46 | ManxPower | a real digium card also has a small heatsink (but so do some clones) |
| 12:05.28 | gr0mit | ghost99, I have had both originals and clones working |
| 12:05.49 | gr0mit | they are identical insofar as they were both rather disappointing |
| 12:06.29 | ghost99 | k: serial 2006020845 ... but does have authentic holographic sticker (lol) ..x100P.com on the sticker |
| 12:06.39 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:06.43 | gr0mit | neither works well on non-US lines |
| 12:06.47 | *** part/#asterisk bartpbx (n=bartpbx@u14-1.dsl.vianetworks.de) |
| 12:06.48 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:06.52 | gr0mit | due to the impedance mismatch. |
| 12:07.30 | gr0mit | if you are in a place where you can get a BRI I would recommend you use a Cologne chipsetp-based card and a BRI line from your telco |
| 12:08.58 | I-MOD | ghost99: its a fake |
| 12:09.17 | I-MOD | unless there is a different serial number on teh board somewhere |
| 12:09.32 | I-MOD | has to be a sticker tho |
| 12:10.31 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 12:11.05 | ghost99 | I-Mod - no no other stickers .... The person is still selling them on ebay.com.au even says they are authentic :( |
| 12:11.11 | ManxPower | ghost99, sounds like it's an authentic clone card. |
| 12:11.27 | ManxPower | i.e. an authentic x100p.com card |
| 12:12.09 | ghost99 | manxpower :) so really do i check it away send them a nasty email and buy a real one or keeptrying or email those jerks .. oh how the world misleads us |
| 12:12.22 | I-MOD | report them to ebay |
| 12:12.33 | ManxPower | ghost99, As I said, I don't think you CAN buy a real X100P anymore. |
| 12:12.35 | I-MOD | it'll get fixed pretty quick |
| 12:12.51 | I-MOD | not new anyway |
| 12:12.56 | luke-jr_ | ManxPower: does the prior destination of a call matter when reinviting? |
| 12:14.16 | coppice | No summer of code stuff from Asterisk this year. never really saw too many results from last year's efforts |
| 12:14.20 | ghost99 | I-mod and Manxpower ..... thanks for your help .. I'm about to kick up a dust storm then! |
| 12:14.38 | luke-jr_ | ghost99: maybe he peeled off the serial #? |
| 12:15.31 | I-MOD | lol |
| 12:15.48 | [TK]D-Fender | ManxPower : I own an authentic Folex watch... what are you tlking about!? |
| 12:16.47 | *** join/#asterisk evilbuny (n=evilbunn@203-214-157-4.perm.iinet.net.au) |
| 12:16.53 | russellb | leave negative feedback, that'll teach 'em |
| 12:19.13 | [TK]D-Fender | russellb : Yeah, nothing like a good spam-filterable scolding to show them whot-for! |
| 12:20.33 | [TK]D-Fender | if intext(fake shit "not real" hate "money back") >> Trash :D |
| 12:20.34 | *** join/#asterisk tdonahue (n=tdonahue@www.vonworldwide.com) |
| 12:21.35 | mut | [TK]D-Fender: i'm getting a double ring when calling out |
| 12:21.38 | mut | first is immediate |
| 12:21.45 | mut | next is delayted quite a bit |
| 12:21.50 | mut | got this sangoma card in now |
| 12:22.03 | mut | call from voip going out a pri |
| 12:25.52 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:26.50 | [TK]D-Fender | mut : What kind of phones? |
| 12:27.07 | mut | spa-2002 ata |
| 12:27.11 | [TK]D-Fender | mut : Watch out for redundant progress indications... |
| 12:27.24 | mut | dial plan has no progress indications but some busy outs |
| 12:27.27 | [TK]D-Fender | mut : That can cause double-ring from what I've heard |
| 12:27.41 | [TK]D-Fender | mut : it'd be in the sip.conf config |
| 12:28.06 | [TK]D-Fender | mut : pastebin your zapata if you'd like me to take a look at this. |
| 12:34.26 | luke-jr_ | Is it possible to make Asterisk only use an 'externip' when the source IP would otherwise be a LAN IP? |
| 12:42.33 | [TK]D-Fender | luke-jr_ : You want to force the extern even for INTERNAL clients? |
| 12:42.41 | stoffell | tzafrir, is it possible to use # for transferring calls on an astribank? (i read something on in audio dtmf not working, does this mean....?) |
| 12:43.12 | [TK]D-Fender | luke-jr_ : I suppose you could set the LOCALNET to a subnet that is NOT local, but still reserved.... |
| 12:43.47 | [TK]D-Fender | Astribank... *shudder* |
| 12:43.57 | stoffell | shudder? |
| 12:44.47 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 12:44.52 | [TK]D-Fender | stoffell : indeed. USB, tied to *, and questionable standard features... |
| 12:44.52 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 12:45.12 | [TK]D-Fender | stoffell : Stardized + Agnostic = good. |
| 12:45.30 | [TK]D-Fender | stoffell : Standardized + Agnostic = good. |
| 12:46.04 | [TK]D-Fender | WOW, feel the dyslexia! I have *NO* idea how that came out so wrong :) |
| 12:46.10 | stoffell | hm, ok i understand what you mean... |
| 12:46.16 | luke-jr_ | [TK]D-Fender: no |
| 12:46.34 | luke-jr_ | [TK]D-Fender: I want s/192\.168\.77\.76/24\.30\.x\.x/ |
| 12:46.35 | [TK]D-Fender | luke-jr_ : Ummm... "no" to what? |
| 12:46.37 | stoffell | don't worry [TK]D-Fender , I'm also having keyboard-dyslexia since a few weeks.. guess it's a human virus |
| 12:46.45 | luke-jr_ | [TK]D-Fender: only for non-local clients |
| 12:46.50 | [TK]D-Fender | luke-jr_ : I can't read or write REGEX.... |
| 12:47.02 | luke-jr_ | [TK]D-Fender: replace 192.168.77.77 with 24.30.x.x for non-local |
| 12:47.10 | [TK]D-Fender | luke-jr_ : thats what externIP is FOR... |
| 12:47.23 | luke-jr_ | [TK]D-Fender: externip replaces *anything* with 24.30.x.x for non-local |
| 12:47.26 | [TK]D-Fender | You just need to define what IS local and everything else gets the externip. |
| 12:47.31 | luke-jr_ | the Asterisk box has two routes |
| 12:47.39 | coppice | [TK]D-Fender: a true dyslexic couldn't have spelled dyslexia :-) |
| 12:47.57 | [TK]D-Fender | coppice : Flukes happen... look at the platypus! |
| 12:47.58 | luke-jr_ | the primary route is a public IP; I want Asterisk to leave packets from that alone |
| 12:48.11 | luke-jr_ | only when the packets are being sent via the backup IP should they be treated as NATted |
| 12:49.00 | coppice | [TK]D-Fender: does a platypus have a lot of trouble with flukes? i guess most water living animals do :-) |
| 12:49.40 | [TK]D-Fender | coppice : ;p |
| 12:50.44 | coppice | I had fluke trouble today. the NiCds in one of their scopes have been ousing and corroding the scope |
| 12:51.39 | *** join/#asterisk Tepes (n=a@omegasoft-fo.b.astral.ro) |
| 12:52.52 | Tepes | hi, anyone knows what it takes to compile zaptel on a i386:x86_64 arch? |
| 12:53.27 | Tepes | i386 userland, x86_64 kernel |
| 12:53.44 | coppice | just compile it |
| 12:54.03 | luke-jr_ | coppice: doubt it |
| 12:54.06 | Tepes | i tried, it failed on link |
| 12:54.10 | Tepes | :) |
| 12:54.11 | Pj_ | try zelda |
| 12:54.14 | Pj_ | :( |
| 12:54.19 | coppice | it works for everyone else |
| 12:54.25 | Tepes | it seems that zaptel is made to be compiled on pure arch |
| 12:54.51 | Tepes | zaptel 1.2.5? |
| 12:55.05 | coppice | last one I tried was 1.2.4 |
| 12:56.07 | Greek-Boy | how does one ensure that when phones move around extensions don't move with it? Ie, extension stays with wall socket |
| 12:56.08 | Greek-Boy | :P |
| 12:56.35 | ManxPower | Greek-Boy, you can't. |
| 12:56.40 | chapeaurouge | need to downgrade asterisk from 1.2.5 to 1.2.4... if i run a make install, will it overwrite my config files? |
| 12:56.41 | ManxPower | at least with IP Phones. |
| 12:56.47 | *** join/#asterisk MrTelephone (n=MrTeleph@c199-243-115-15.hb.picriverisp.net) |
| 12:57.06 | ManxPower | chapeaurouge, not supposed to, but back them up just in case. |
| 12:57.25 | chapeaurouge | yea, that's done. Thanks ;) |
| 12:57.36 | MrTelephone | Can someone tell me more about the echo VPM100 Module on the TDM2400P card? I have revision A.. When I turn echo cancellation on in zapata.conf the phone calls sound like a walkie talkie half-duplex call |
| 12:58.05 | MrTelephone | I thought zapata.conf was for software cancellation but does it control the hardware cancellation if it is detected? |
| 12:58.10 | ManxPower | MrTelephone, contact Digium, that should be something covered under the free tech support. |
| 12:58.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:58.51 | [TK]D-Fender | chapeaurouge : No, only "make samples" would kill them. |
| 12:59.35 | chapeaurouge | hey TK... ok, cool |
| 12:59.37 | [TK]D-Fender | MrTelephone : Zapata controls the hardware EC as well for both Digium and Sangoma cards. |
| 12:59.57 | [TK]D-Fender | chapeaurouge : np. Still using * for pure P2P SIP only? |
| 13:00.03 | chapeaurouge | no. |
| 13:00.07 | chapeaurouge | got ISDN Cards and all now |
| 13:00.10 | MrTelephone | are there any new tricks adjusting rxgain and txgain values? |
| 13:00.36 | chapeaurouge | that's why i need to downgrade... BRIstuff drivers from junghanns require 1.2.4 |
| 13:00.44 | chapeaurouge | the * patching isn't working on 1.2.5 |
| 13:00.49 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 13:00.49 | [TK]D-Fender | chapeaurouge : I'm talking about your home use... |
| 13:00.53 | chapeaurouge | aaah |
| 13:01.03 | [TK]D-Fender | chapeaurouge : But yeah, you finally got work on * now? |
| 13:01.05 | chapeaurouge | well, im not often at home anymore, so... |
| 13:01.06 | chapeaurouge | yes. |
| 13:01.06 | MrTelephone | Im confused why echo is a big problem when you can plug in a 15 dollar phone and have no echo ? |
| 13:01.16 | chapeaurouge | got like 10 IP300 polycom, a few ip500.. |
| 13:01.17 | stoffell | chapeaurouge, bristuff is at 1.2.7 now |
| 13:01.39 | [TK]D-Fender | chapeaurouge : Good choices. I just got my IP 301 & 501 for home as well. |
| 13:01.51 | chapeaurouge | stoffell, i see bristuff experimental 0.3.0-PRE1l |
| 13:02.00 | AsteriskAlbania | any one has implemented AGI script for radius authentication ? |
| 13:02.01 | stoffell | chapeaurouge, yes, but go to junghanns.net/downloads/ |
| 13:02.14 | stoffell | chapeaurouge, there's already an 'n' (the homepage updates don't seem to work well :p) |
| 13:02.24 | mut | MrTelephone: because sometimes it isn't that easy |
| 13:02.29 | chapeaurouge | i see.. they could update their homepage.... |
| 13:02.35 | chapeaurouge | thx for the tip stoffell |
| 13:02.46 | stoffell | or even 'o' :) yes, i also always check the downloads dir :) |
| 13:02.48 | chapeaurouge | there's an 'o' even |
| 13:02.49 | chapeaurouge | lol |
| 13:02.55 | chapeaurouge | nice, will do that now too |
| 13:02.59 | [TK]D-Fender | MrTelephone : There are greater things involved once you put a pbx between the analog line and phones of any kind... |
| 13:03.04 | coppice | Mr Telephone: the answer lies in latency |
| 13:03.27 | [TK]D-Fender | MrTelephone : And a few other things like impedence, etc... |
| 13:03.46 | luke-jr_ | I thought the answer was the analog<->digital conversion |
| 13:03.55 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 13:04.41 | chapeaurouge | i hope the config from 1.2.5 to 1.2.7 will go well then :) |
| 13:05.14 | stoffell | chapeaurouge, yeah, but 1.0.9 to 1.2.x did have some glitches :p |
| 13:05.24 | chapeaurouge | so i heard ;) |
| 13:05.29 | [TK]D-Fender | luke-jr_ : Thats victim to all the other parts, latency particularly. |
| 13:06.20 | *** join/#asterisk drkfiber (n=drkfiber@207.241.43.114) |
| 13:06.24 | *** join/#asterisk zaf (n=zaf@wsip-68-228-9-79.br.br.cox.net) |
| 13:06.36 | luke-jr_ | [TK]D-Fender: I meant how the analog lines were full duplex, whereas digital splits the two channels |
| 13:08.39 | [TK]D-Fender | luke-jr_ : And remixing causes latency, can have impedence issues, and the whole gamut of complications. |
| 13:09.08 | coppice | luke_jr_ the echo is purely about latency. without latency we call is reverb and like the effect |
| 13:09.12 | [TK]D-Fender | I thry to think of it as "Yeah you need EC, PERIOD" for anything but a raw phone plugged on a raw line... |
| 13:10.00 | [TK]D-Fender | coppice : reverb, delay, and chorus are great on my guitar effects processor, NOT on my phone calls :) |
| 13:10.47 | coppice | reverb is fine on phone calls. its normal, and acceptable. packetising causes heavy latency, and that's where the trouble comes from |
| 13:11.12 | coppice | without any reverb at all, calls sound rather dead |
| 13:11.13 | chapeaurouge | Link /usr/src/linux-2.6 to your kernel sources first! |
| 13:11.14 | chapeaurouge | make: *** [linux26] Error 1 |
| 13:11.17 | chapeaurouge | wtf... the link is there |
| 13:11.26 | Pj_ | [TK]D-Fender: and don't use your fender as a phone too, that may help. |
| 13:11.59 | *** part/#asterisk drkfiber (n=drkfiber@207.241.43.114) |
| 13:12.35 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:12.40 | coppice | the pickups on a fender don't make the best voice mics |
| 13:12.52 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 13:14.20 | mut | wha |
| 13:14.30 | mut | peter frampton was pretty good at it |
| 13:16.41 | wasim | depends, the single coil ones are passable |
| 13:16.51 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 13:17.03 | coppice | no. he took the output of a fender and played in into his mouth. he then picked up the output from his mouth |
| 13:17.30 | [TK]D-Fender | Pj_ : My nick has nothing to do with music, but rather my FPS history. I'm 100% voip @ hom running Polycom phones, and terminating my calls at my office on a PRI :) |
| 13:17.44 | rkr245 | guys need help in implementing proxy for asterisk |
| 13:17.46 | coppice | wasim: don't you find the strings a mite resonant for getting a flat voice response :-) |
| 13:17.53 | [TK]D-Fender | coppice : D>A>D>A :) |
| 13:18.09 | [TK]D-Fender | rkr245 : Go take a look at SER |
| 13:18.10 | trelane_ | [TK]D-Fender, I'm running Asterisk... At home (but not Asterisk@home) |
| 13:18.12 | wasim | coppice: only the flat-wound ones, tee hee ... |
| 13:18.42 | [TK]D-Fender | trelane : I still want nothing to do with AMP unless I'm making money off of it... |
| 13:18.56 | blitzrage | [TK]D-Fender: $500? |
| 13:18.58 | wasim | [TK]D-Fender: thats like the first two chord song i learnt too ... |
| 13:19.08 | [TK]D-Fender | blitzrage : $500 for what? |
| 13:19.13 | blitzrage | [TK]D-Fender: documentation |
| 13:19.28 | [TK]D-Fender | blitzrage : To doc it? Is that an offer? |
| 13:19.47 | trelane_ | [TK]D-Fender, strongly concur, and I actually turned down a chance to make money on AMP and told them I would sooner delete it and write the configs myself |
| 13:19.47 | [TK]D-Fender | wasim : What was? |
| 13:19.56 | wasim | [TK]D-Fender: D A D A |
| 13:20.17 | blitzrage | [TK]D-Fender: well, I got offered to write docs for A@H for $500... :) |
| 13:20.20 | [TK]D-Fender | trelane : I've done the same, but it may cripple business opportunities, so I may get another box for testing.... |
| 13:20.30 | blitzrage | [TK]D-Fender: I'm just curious if you'd have taken the offer :) |
| 13:21.01 | rkr245 | no body wakeup there iam trying there since three weeks |
| 13:21.02 | [TK]D-Fender | wasim : I was referring to Digital > Analog > Digital > Analog for coppice's amusement :) |
| 13:21.31 | coppice | why might this amuse me? |
| 13:21.34 | [TK]D-Fender | blitzrage : Depends on how much work it'd take naturally.... but I might suck it up "on a curve" as it opens new possibilities as well... |
| 13:21.50 | [TK]D-Fender | coppice : just my assessment of the kind of audio path you were just joking about :) |
| 13:22.10 | wasim | [TK]D-Fender: i know, it was a bad attempt at a joke by me, for coppice's amusement as well |
| 13:22.24 | blitzrage | lol -- why is everyone trying to entertain coppice ? :) |
| 13:22.31 | [TK]D-Fender | rkr245 : I already answered you....YOU wake up :) |
| 13:22.33 | coppice | gee, and people here complain about my humour |
| 13:22.41 | blitzrage | :) |
| 13:22.55 | ManxPower | Docs? We don't need docs! Asterisk was hard to write, it should be hard to setup! |
| 13:23.02 | [TK]D-Fender | blitzrage : If we stare at him long enough he might do a trick! (or succeed at T.38 for *) |
| 13:23.23 | [TK]D-Fender | $50 ON THE FORMER!!! |
| 13:24.16 | ghost99 | Manpower: are you one of the developers ? |
| 13:24.16 | blitzrage | ooooooo, a trick! |
| 13:24.23 | ghost99 | Monxpower :) opps : there i go again dislexia didn't someone say that ? |
| 13:24.30 | ManxPower | ghost99, I am not a developer. I do not work for Digium. |
| 13:24.40 | blitzrage | I do not like green eggs and ham. |
| 13:24.52 | tzanger | try them, try them, you will see! |
| 13:25.11 | Hmmhesays | i like it when they are being served by nekkid wimmins |
| 13:25.13 | Hmmhesays | sam I am |
| 13:25.20 | ManxPower | If I worked for Digium, then I would have to be nice to people that are idiots. |
| 13:25.23 | ManxPower | And that's no fun at all. |
| 13:25.25 | blitzrage | lol |
| 13:25.28 | tzanger | Hmmhesays: hahaha that's the R-rated version of the book |
| 13:25.29 | blitzrage | hail! |
| 13:27.59 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.37.Dial1.SanJose1.Level3.net) |
| 13:28.28 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.37.Dial1.SanJose1.Level3.net) |
| 13:28.36 | ghost99 | well yes re idiots , trouble is they are everywhere .... so rather than be a killer so to speak .. being diplomatic should we say is still a good approach and outcome |
| 13:28.56 | AsteriskAlbania | what is the best distro for implementing ASTERISK? |
| 13:29.11 | AsteriskAlbania | RHEL4, FEDORA or ???? |
| 13:29.17 | luke-jr_ | Foobarist |
| 13:29.21 | [TK]D-Fender | to coin a phrase from Fight Club "I am ManxPower's raging bile duct" |
| 13:29.33 | ghost99 | so hope you weren't referring to me ? ;) |
| 13:29.35 | [TK]D-Fender | AsteriskAlbania : Whichever you are most comfortable with. |
| 13:29.38 | ghost99 | never saw that movie :) |
| 13:29.40 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:29.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:30.00 | [TK]D-Fender | ghost99 : You seriously need to. EVERYONE of sufficien maturity does.... |
| 13:30.03 | ManxPower | ghost99, nope. You were just foolish. 8-) |
| 13:30.13 | coppice | ManxPower: I've seen T.38 working with * :-) |
| 13:30.31 | ManxPower | coppice, but only after eating wild mushrooms? |
| 13:30.34 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:30.44 | ghost99 | yeah but hey ... times are diffeent .. well sort of .. 15 years ago in the irc bar ... say what you want :) !~ |
| 13:30.46 | Ariel_ | morning folks |
| 13:30.56 | coppice | wild? when you eat them they get bloody furious |
| 13:31.08 | wasim | and nekkid |
| 13:31.19 | AsteriskAlbania | [TK]D-Fender: I am trying to update mysql4 -> mysql5 for astbill and I am having a lot of troubles |
| 13:31.35 | AsteriskAlbania | [TK]D-Fender: on RHEL4 |
| 13:32.05 | [TK]D-Fender | AsteriskAlbania : find something more comfortable. And while you're at it try #mysql , this is #asterisk :) |
| 13:33.11 | AsteriskAlbania | [TK]D-Fender: everyone is sleeping at #mysql :) |
| 13:33.32 | [TK]D-Fender | AsteriskAlbania : NMP <- |
| 13:33.44 | AsteriskAlbania | ? |
| 13:33.51 | [TK]D-Fender | Not My Problem :) |
| 13:34.32 | AsteriskAlbania | [TK]D-Fender: do you know any AGI script for talking to a radius server ? |
| 13:35.02 | [TK]D-Fender | AsteriskAlbania : TOYWY : The One You Write Yourself |
| 13:35.05 | MrTelephone | Has anyone tried to adjust the gain values using the 1004 tone method? |
| 13:35.09 | kardecallan | <PROTECTED> |
| 13:35.21 | *** join/#asterisk keith80403 (n=keith@24-56-189-80.co.warpdriveonline.com) |
| 13:35.26 | ghost99 | so back to the subject .. and i must say that seems 2 days ago the digium cards were telepermitted here in New Zealand , yippee i spose .. so now i won't now be doing anything illegal : .. but anyone here know about running these cards over ADSL/DSL lines and that might cause problems? |
| 13:35.28 | ManxPower | kardecallan, you might want to also try #asterisk-dev |
| 13:35.34 | AsteriskAlbania | [TK]D-Fender: YAGGTSLT |
| 13:35.36 | [TK]D-Fender | MrTelephone : Adjusting gains is the last deperate ploy of those with inferior echo cancellation... |
| 13:35.52 | AsteriskAlbania | [TK]D-Fender: You are a great guy to speka like that |
| 13:36.00 | MrTelephone | ghost99.. should work fine over dsl? |
| 13:36.07 | ManxPower | [TK]D-Fender, so basically all Asterisk users then. |
| 13:36.11 | Katty | heylo. |
| 13:36.15 | MrTelephone | the biggest problems with these cards is the echo problems on the pstn -> sip conversion |
| 13:36.31 | MrTelephone | otherwise they are great |
| 13:36.35 | [TK]D-Fender | ManxPower : Nope, mine are flat-line everywhere and ) echo :) |
| 13:36.39 | [TK]D-Fender | 0* |
| 13:36.44 | ManxPower | MrTelephone, no. The echo is there when you are using a plain analog phone, you just don't HEAR it as an echo because the latency is so low. |
| 13:37.10 | iDunno | lunch good. |
| 13:37.21 | ManxPower | the zaptel-trunk echocan (MG2?) seems to work well, but we are still going to install more tellabs boxes. |
| 13:37.26 | MrTelephone | where is the latency from sip->pstn pstn->sip is this conversion causing 10ms?? |
| 13:37.28 | iDunno | long lunch very good :) |
| 13:37.37 | Katty | iDunno: it's lunchtime there already? |
| 13:37.42 | [TK]D-Fender | ManxPower : I switched to the Dark Side, and am not coming back :) |
| 13:37.49 | tzanger | MrTelephone: the latency is in the zaptel driver and the PCI bus and hte network |
| 13:37.54 | ManxPower | MrTelephone, all VoIP has latency |
| 13:37.58 | tzanger | MrTelephone: and the conversion from ulaw->whatever and back |
| 13:38.05 | *** join/#asterisk unixgeek (n=unixgeek@12.45.238.189) |
| 13:38.08 | ManxPower | [TK]D-Fender, which Dark Side? Sangoma? |
| 13:38.15 | iDunno | Katty: it's 14:38 now, I've been at lunch for an hour and a half :) |
| 13:38.19 | MrTelephone | so can adjusting the rxgain txgain eliminate the echo? |
| 13:38.28 | [TK]D-Fender | MrTelephone : And you CPU doing things like managing pr0n over BitTorrent... |
| 13:38.35 | [TK]D-Fender | ManxPower : INDEED |
| 13:38.36 | ManxPower | MrTelephone, No. But it can make the echo soft enough you don't hear it. |
| 13:38.50 | ManxPower | It could also make the outgoing audio volume so low the far end can't hear you. |
| 13:38.57 | ManxPower | ECHO is one of the hardest things to fix. |
| 13:38.59 | MrTelephone | so is it better to have a distinct echo so the echo cancellation will remove it properly? |
| 13:39.17 | [TK]D-Fender | ManxPower : tremendous satisfaction with their entire line. Have consulted with A200 installs recently as well (even without HWEC). |
| 13:39.19 | ManxPower | MrTelephone, no. |
| 13:39.36 | unixgeek | Does anyone have experience connecting asterisk to another switch over a t1 tie line? |
| 13:39.46 | ManxPower | [TK]D-Fender, I may go with Sangoma when I get a 2-port T-1 card later in the summer. |
| 13:39.50 | ghost99 | MrTelephone: you saying they work great becasue you are in this region ? |
| 13:40.24 | MrTelephone | I have a problem.. when I place outgoing calls the echo is nominal.. incoming calls the echo is bad but gets trained in about 10 seconds.. does this sound filmiliar? |
| 13:40.27 | unixgeek | I can receive calls from the switch, but trying to make calls from the asterisk box to the switch fail immediately. |
| 13:40.33 | [TK]D-Fender | ManxPower : Thats all I'm using mine for right now... its SO worth it... |
| 13:40.40 | puzzled | anyone know what a rapidly flashing green led on a sipura spa-3000 means when I hook up a (known good) lan cable? |
| 13:40.55 | [TK]D-Fender | ManxPower : And by then they may have released a 2-port version of their HWEC card... |
| 13:40.56 | MrTelephone | unixgeek: asterisk -rvvvv and check the output |
| 13:40.56 | ManxPower | puzzled, I think it means "I see traffic" |
| 13:41.11 | ManxPower | [TK]D-Fender, I won't use onboard EC anyway. |
| 13:41.12 | puzzled | ManxPower: nope since the Cisco switch shows no connection |
| 13:41.28 | unixgeek | MrTelephone: Let me make a call and post the output on pastebin |
| 13:41.29 | [TK]D-Fender | ManxPower : Still sticking with ZAptel? |
| 13:41.30 | ManxPower | puzzled, bad cable then |
| 13:41.32 | MrTelephone | puzzled, switch out ur cable try a different port.. plug it into a different switch |
| 13:41.45 | puzzled | MrTelephone: nope, cable works fine on another phone |
| 13:41.57 | MrTelephone | what is the echo like then from pstn -> asterisk -> digium fxs? |
| 13:42.47 | MrTelephone | Nortel Meridian boxes seem to do a good job with the echo.. is the technology just more advanced (they are very expensive) |
| 13:42.47 | unixgeek | OK. console messages are at http://pastebin.com/686962 |
| 13:42.51 | ManxPower | [TK]D-Fender, Huh? |
| 13:43.10 | ManxPower | MrTelephone, you mean nortel meridian VoIP boxes? |
| 13:43.28 | [TK]D-Fender | ManxPower : Saying you won't use onboard EC.... you plan on sticking with Zaptel software EC, or porting over your tellabs? |
| 13:43.56 | ManxPower | Um, tellabs takes a t-1 in and puts an echo canceled t-1 out to the PBX |
| 13:44.07 | [TK]D-Fender | MrTelephone : No, Nortel jsut includes decent EC right away in their hardware and you pay through the nose for it |
| 13:44.10 | MrTelephone | manxpower, i guess, even the older systems |
| 13:44.38 | MrTelephone | is there a hardware unit that will go in between POTS lines and asterisk fxo module? |
| 13:44.45 | tzanger | actually the nortel ec isn't all that great |
| 13:44.49 | coppice | there's no EC in a meridian |
| 13:45.03 | MrTelephone | meridian is just a switch then? |
| 13:45.13 | tzanger | I have a TE405 going into a Norstar MICS... it maybe has 32 or 16ms EC, but not enough for me |
| 13:45.22 | coppice | there isn't even a meridian in a meridian. there's a nice one in Greenwich :-) |
| 13:45.34 | unixgeek | The switch technician is telling me that everytime an outgoing call is placed, he gets a message stating that "the loop number is out of range (BCH2TN)" Anyone have any ideas? |
| 13:45.48 | kardecallan | ManxPower, this question I already made, but nobody still answered. |
| 13:45.51 | MrTelephone | sorry unixgeek i never worked with pri or bri |
| 13:45.57 | [TK]D-Fender | coppice : We're not caffeinated enough for geographical humour! Cool it! |
| 13:46.12 | tzanger | coppice: heh |
| 13:46.12 | MrTelephone | unix is that just an iax link over IP? |
| 13:46.15 | ManxPower | tzanger, Um you have more than 32ms PSTN latency? |
| 13:46.23 | unixgeek | MrTelephone: Thanks for taking a look. |
| 13:46.25 | tzanger | unixgeek: are you trying to do 2BCT? |
| 13:46.38 | tzanger | ManxPower: PRI -> TE405 - 1-hop IAX2 - TE405 -> MICS |
| 13:46.43 | MrTelephone | is 15ms enough to hear urself? |
| 13:46.46 | I-MOD | unixgeek: pastebin a pri debug of the outgoing call attempt |
| 13:46.59 | unixgeek | The call orginates from one * box to the one that has the pri to the nortel switch. |
| 13:47.08 | ManxPower | tzanger, ah, no EC will be able to handle that |
| 13:47.33 | unixgeek | tzanger: I need 2bct defined. Is that part of a signaling protocol? |
| 13:47.41 | MrTelephone | who here knows how to use fxotune properly? |
| 13:47.57 | tzanger | ManxPower: well I've tried MG2 on the PRI-side TE405 and it mostly works but still some that's too much. I've got an A104D but it echo cancells too well... D channel works great but no audio whatsoever |
| 13:48.09 | MrTelephone | unixgeek, did u try making the call from the box thats connected to the pri? |
| 13:48.38 | unixgeek | MrTelephone: That is actually something we are setting up right now. |
| 13:49.09 | MrTelephone | looks like ur iax link is working though eh? |
| 13:49.29 | MrTelephone | I havn't done that but would like to in the future |
| 13:49.44 | MrTelephone | how are you plugged into the nortel switch.. is there a pri port on it? expansion card? |
| 13:49.56 | tzanger | unixgeek: 2BCT = 2 B-channel Transfer. Relatively new feature in libpri that allows certain switches to negotiate a bridge without keeping your channels in the loop. i.e. call comes in your PRI, and bridges to another B chan going out. 2BCT allows the Asteirsk box and switch to say "hey, both of these B channels are handled by you, why don't you just bridge them for me and let me take two more calls?" |
| 13:50.15 | unixgeek | MrTelephone: yes, the iax link is working just fine. It seems to be when the signalling is occuring over the t1. |
| 13:50.15 | tzanger | unixgeek: it only works with 5ESS switches at the moment I think. I know it doesn't work for me. :-( |
| 13:50.35 | ManxPower | tzanger, you would need EC at PRI -> TE405 AND TE405 -> MICS |
| 13:51.15 | tzanger | ManxPower: I have MG2 at PRI->TE405 and MG2 at TE405->MICS. I even bumped MG2 out to 1024 taps (128ms) on both with no real change on the nasty calls, which is why the A104D got ordered. |
| 13:51.15 | MrTelephone | so noone hear has a problem with the echo canceller on the tdm2400p making a call sound half-duplex |
| 13:51.38 | MrTelephone | ? |
| 13:51.46 | unixgeek | tzanger: Interesting. I have a nortel switch at the other end. |
| 13:51.48 | tzanger | MrTelephone: you have the hardware echo can option on the TDM2400P? "half-duplex" is what the software echo canceller's agressive mode does. |
| 13:52.15 | tzanger | * jsharp never had a problem with echo. |
| 13:52.24 | wasim | lol |
| 13:53.11 | MrTelephone | tzanger, i think i enabled aggressive as a last resort.. There is no documentation on the distinction between software and hardware echo cancellation settings. |
| 13:53.25 | MrTelephone | I have the hardware card |
| 13:54.01 | jsharp | Its strange. I've been diddling with * for 5+ years now and have never had to tweak or fiddle with echo cancellers. Its always Just Worked. |
| 13:54.03 | tzanger | MrTelephone: you must have disabled it in order to get the software echo can settings to have any effect. probabloy something like "vpmsupport=0" or commenting out the VPM_SUPPORT define in the wct2400 driver. |
| 13:54.09 | tzanger | jsharp: are you close to the CO? |
| 13:54.15 | mut | jsharp: you've been lucky |
| 13:54.32 | tzanger | I mostly have no echo but the odd call |
| 13:54.42 | tzanger | likely from farmers 15 miles from the nearest CO |
| 13:54.54 | mut | software ec wouldn't get rid of mine |
| 13:54.55 | MrTelephone | tzanger, the echotraining= function fails.. would that mean the hardware card is enabled? |
| 13:54.56 | mut | cisco box did |
| 13:55.02 | mut | and this sangoma card is too |
| 13:55.03 | jsharp | Been one rack over from the CO...been 18 miles from the CO. |
| 13:55.37 | MrTelephone | my locations are 15 miles away from CO and there is a mini CO switch 4 miles away to switch local calls |
| 13:55.41 | MrTelephone | i have nasty echo problems |
| 13:55.43 | ManxPower | jsharp, mostly analog? |
| 13:55.45 | mut | shoot i can dialup at 32.6k over my voip with the sangoma card |
| 13:55.46 | ManxPower | or PRI? |
| 13:55.50 | ManxPower | i.e. no VoIP? |
| 13:55.56 | jsharp | A mix. |
| 13:56.15 | jsharp | Either been POTS lines or PRI circuits. |
| 13:56.31 | ManxPower | yeah. You're not going to get echo if you're not doing VoIP |
| 13:56.53 | jsharp | Oh, there's been plenty of voip. |
| 13:57.20 | jsharp | Heck, right now I'm running 10 PRIs from the MCI POP 9 miles away over 600ms satellite hops with no echo. |
| 13:58.00 | MrTelephone | the echo is a problem with DIGIUM FXO modules |
| 13:58.15 | MrTelephone | from what i read most pri connections don't have much echo problems |
| 13:58.24 | ManxPower | jsharp, I've noticed that telcos put EC on LD trunks |
| 13:58.34 | *** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 13:58.45 | ManxPower | MrTelephone, the far end is still frequently analog. |
| 13:58.52 | MrTelephone | my echo has nothing to do with telco because all the callers say the call is perfect |
| 13:59.04 | SplasPood | does MusicOnHold() have some sort of absolute timeout? After a while of listening it'll just drop to congestion |
| 13:59.06 | ManxPower | MrTelephone, you do not understand echo. |
| 13:59.28 | MrTelephone | why not? |
| 13:59.38 | ManxPower | echo can happen ANYWHERE there is a 2-wire circuit. |
| 13:59.50 | coppice | or a handset |
| 14:00.02 | mut | or a pissed off gnome |
| 14:00.10 | MrTelephone | so why does a caller not hear echo when I can hear echo inside my asterisk network |
| 14:00.21 | wasim | earwax |
| 14:00.22 | MrTelephone | does the telephone company echo cancel the call? |
| 14:00.45 | blitzrage | echo is generated by the far end.. if you hear echo, its the other side (or some circuit in your network...) |
| 14:00.50 | jsharp | sunspots. Or your dishwashing detergent. |
| 14:00.53 | ManxPower | i.e. Analog POTS <-> Telco <-> PRI <-> Asterisk <-> SIP Phone. In this example the person on the SIP phone hears his own voice echoed back because of the POTS<->Telco connection |
| 14:01.06 | mut | solar flares |
| 14:01.22 | MrTelephone | so what is the solution to that? |
| 14:01.24 | ManxPower | The person on the Analog POTS won't hear echo because there is no analog connection on the far end. |
| 14:01.30 | wasim | qtips |
| 14:01.41 | mut | sunscreen |
| 14:01.41 | ManxPower | MrTelephone, There is no solution except for a hardware or software echo canceler. |
| 14:02.03 | coppice | don't make phone calls. that works |
| 14:02.23 | MrTelephone | is there a way to test the impedence of the lines with a tool to get an idea what the rxgains and txgains should be? |
| 14:02.32 | file | don't even TOUCH a phone |
| 14:02.38 | ManxPower | MrTelephone, fxotune? |
| 14:02.56 | MrTelephone | i tried that but i get results like.... 1:5,0,0,0,0,0,0 |
| 14:03.18 | MrTelephone | same value for all the fxo lines |
| 14:03.47 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 14:04.00 | ManxPower | now in a Analog POTS <-> Telco <-> PRI <-> Asterisk <-> Analog POTS There will STILL be echo, but because the latency of the telco is so low, the user does not perceive the echo, but it's still there. |
| 14:04.39 | ManxPower | Because cell phones have such high latency all Cell <-> PSTN interfaces must do Echo Cancel |
| 14:06.10 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:06.28 | MrTelephone | ok i have to go out on call but im gonna fiddle with my gains a little bit when I come back |
| 14:06.47 | coppice | Some naming their product "AlternaTIFF" deserves bankruptcy |
| 14:06.53 | MrTelephone | I have a feeling my hardware echo canceller isn't working because it should be a matter of turning it on and no echo. |
| 14:07.14 | tzanger | MrTelephone: in theory, yes. |
| 14:07.42 | MrTelephone | when i modprobe wctdm24xxp it shoulds the that module is there.. something about U-LAW codec dtmf 1024 |
| 14:08.04 | MrTelephone | tzanger, are you filmiliar with this card? |
| 14:08.30 | tzanger | not hte TDM2400 specifically, but I have sevearl TE405/406 cards and have used the (1st gen) VPM |
| 14:10.50 | MrTelephone | I have a feeling maybe the hardware echo cancel is turned on with echocancel=yes but the software cancel is still trying to cancel as well.. do you think that could happen? |
| 14:11.23 | SplasPood | hrm yes.. MusicOnHold seems to hang up without fail |
| 14:12.04 | MrTelephone | check the console message |
| 14:12.07 | MrTelephone | asterisk -rvvvv |
| 14:13.14 | tzanger | MrTelephone: nope, echocancel=yes turns on echo cancel support. If the hardware is detected (check dmesg when you load the driver) then it uses hardware EC. If the hardware echo cancel is NOT detected, it uses the software EC |
| 14:14.00 | heka | Is there any way to disconnect the callers line but to continue to execute other extensions after hangup? |
| 14:14.07 | SplasPood | MrTelephone: Nothing specific, just says it stopped MusicOnHold and I hung up |
| 14:14.51 | *** join/#asterisk noname32 (n=noname@38.113.5.165) |
| 14:16.03 | MrTelephone | i have to go for a few mins i'll be back |
| 14:17.20 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 14:18.48 | brookshire | splas: how are you calling musiconhold? |
| 14:18.54 | *** join/#asterisk barros (n=carlos@zaphod.inexo.com.br) |
| 14:19.01 | SplasPood | brookshire: yep, just testing my music |
| 14:19.10 | SplasPood | oh |
| 14:19.11 | SplasPood | HOW |
| 14:19.14 | SplasPood | MusicOnHold(class) |
| 14:19.23 | SplasPood | no Answer() no nothin |
| 14:19.26 | brookshire | ahh.. so like exten => 1234,1,MusicOnHold |
| 14:19.28 | SplasPood | exten, 1 just calls Music.. |
| 14:19.28 | SplasPood | yes |
| 14:19.30 | SplasPood | exactly |
| 14:19.34 | SplasPood | cept I'm passing a class |
| 14:19.36 | brookshire | yeah.. without answer, it will hang up |
| 14:19.40 | SplasPood | ahh |
| 14:20.21 | brookshire | splas: and are you using native? |
| 14:20.23 | barros | anyone know about the final result of the bug #5374, about the silence suppress? |
| 14:20.29 | brookshire | native musiconhold |
| 14:20.39 | SplasPood | newp |
| 14:20.52 | brookshire | native musiconhold > mpg123 |
| 14:21.10 | SplasPood | try doing native and sticking a file in there that asterisk doesn't like |
| 14:21.13 | SplasPood | I've had crashes |
| 14:21.35 | brookshire | heh |
| 14:21.38 | brookshire | true i guess |
| 14:21.39 | redondos | I am wondering how I could implement this: someone needs to edit the configuration files via FTP. I know it's insecure, impractical, in*; but it's something that needs to be done. How would I configure it? Shall I add the user to the "asterisk" group (/etc/asterisk/* chowned asterisk:asterisk) and set his umask to be 0007? Shall I give the "asterisk" user a valid shell instead of /bin/false and give it a password so that the user can log-in using those |
| 14:21.59 | barros | i've just found the bug entry but nothing was implemented in the source tree and no more comments about this issue! |
| 14:22.03 | SplasPood | brookshire: and of course I was fucking with this on my production box... luckily it was late at night |
| 14:22.32 | redondos | Or is there a web application that would let him edit those files? I'm just thinking here... |
| 14:23.25 | *** join/#asterisk ghost100 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
| 14:25.08 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 14:25.26 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:27.27 | barros | any hint??? |
| 14:28.57 | [TK]D-Fender | redondos : Symlink the whole folder... |
| 14:29.15 | [TK]D-Fender | redondos : And force the userid to root. |
| 14:29.28 | *** join/#asterisk }cytrak{ (n=noneon1@208.63.19.179) |
| 14:30.30 | redondos | [TK]D-Fender: Symlink /etc/asterisk into /home/$USERNAME? Force the userid to root-> what do you mean by that? |
| 14:30.54 | }cytrak{ | hi... i finaly fixed my echo problem by adjusting my rxgain and txgains .. but now users tell me that when I call them on a regular phone .. from iax-softphone to regular phone my senteces breaks up |
| 14:31.04 | [TK]D-Fender | redondos : on FTP, jsut have it assume the user is root so you don't have to worry about permissions. |
| 14:31.09 | *** part/#asterisk heka (n=Mango@80.80.175.130) |
| 14:31.30 | redondos | [TK]D-Fender: Oh, ok. Thank you for the suggestion. |
| 14:31.34 | }cytrak{ | the sound quality is fine though .. does this also have something to do with rx/tx gains ? |
| 14:31.51 | redondos | [TK]D-Fender: What daemon do you use? Is that doable with vsftpd? |
| 14:32.38 | [TK]D-Fender | redondos : Everything I suggested is hypothecal and I'm not qualified to answer those questions, sorry :) |
| 14:33.20 | [TK]D-Fender | redondos : I COULD cheat by using sama locally and connecting to that to virtualize the userid, but thats a 4-bit hack, even by my standards :) |
| 14:33.38 | luke-jr_ | redondos: FTP doesn't do editing |
| 14:34.26 | redondos | luke-jr_: I know, but this Windows user can use an FTP client that would download the files, open them on an editor, and upload them everytime the guy modifies them. So that's convenient for him. |
| 14:34.52 | redondos | [TK]D-Fender: hehe, yeah. I'll see what I do... perhaps a simple web application that shows a list of config files and has a textbox for editing them. That would come in handy. |
| 14:35.38 | luke-jr_ | redondos: honestly, someone who can't use SSH and nano shouldn't be editing Asterisk configs... >.> |
| 14:36.02 | [TK]D-Fender | luke-jr_ ....... I use MC... watch it buddy :) |
| 14:37.05 | luke-jr_ | [TK]D-Fender: ? |
| 14:37.06 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
| 14:37.29 | luke-jr_ | [TK]D-Fender: use != can use |
| 14:37.36 | luke-jr_ | [TK]D-Fender: I trust you *can* use SSH... |
| 14:37.56 | luke-jr_ | and nano |
| 14:40.17 | *** join/#asterisk koenvi (n=kova@tech.ascom.be) |
| 14:42.00 | *** join/#asterisk guigouz (n=guigouz@201-42-161-117.dsl.telesp.net.br) |
| 14:42.07 | *** join/#asterisk Samoied (n=Samoied@201-14-148-91.fnsce7002.dsl.brasiltelecom.net.br) |
| 14:42.12 | [TK]D-Fender | luke-jr_ : I have no practical experience with any other CLI text-editor..... |
| 14:42.30 | luke-jr_ | [TK]D-Fender: It's like pico |
| 14:42.30 | guigouz | if I have a sip phone, can I call it using only its ip address (without registering to a proxy) ? |
| 14:42.36 | [TK]D-Fender | luke-jr_ : I think I tried nano twice, and got into vi not knowing how to get out :) |
| 14:42.56 | luke-jr_ | [TK]D-Fender: :q |
| 14:43.09 | [TK]D-Fender | luke-jr_ : Thats like someone describing the taste of a Canada Goose to you as similar to that of a bald eagle :D |
| 14:43.11 | luke-jr_ | [TK]D-Fender: I just used 'nano' because anyone can use that |
| 14:43.23 | *** join/#asterisk sl16 (n=root@212.73.131.4) |
| 14:43.24 | RoyK | vim rules |
| 14:43.25 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 14:43.35 | a1fa | sip.broadvoice.com 147.135.0.128 5060 OK (54 ms) |
| 14:43.37 | a1fa | hehe this is so messed up |
| 14:43.44 | koenvi | how should channels in zapata be set for a qsig link? |
| 14:43.46 | a1fa | i am unable to recieve calls |
| 14:43.48 | a1fa | again |
| 14:43.52 | a1fa | sip show peers |
| 14:43.53 | a1fa | show this |
| 14:43.55 | a1fa | sip.broadvoice.com 147.135.0.128 5060 OK (54 ms) |
| 14:44.05 | guigouz | a1fa: broadvoice problems ? I gave up on them in the past. |
| 14:44.12 | a1fa | yeah |
| 14:44.16 | a1fa | sip debug ip 147.135.0.128 |
| 14:44.22 | a1fa | doesnt show any output |
| 14:44.26 | luke-jr_ | a1fa: checked voip-info.org, I assume? |
| 14:44.31 | [TK]D-Fender | a1fa : change peers |
| 14:44.41 | a1fa | [TK]D-Fender : i am going to.. looks like it |
| 14:44.47 | a1fa | i called them 10 times regarding the same issue |
| 14:45.04 | a1fa | and they make it sound like its my fault |
| 14:45.08 | a1fa | "no everything is working fine" |
| 14:45.09 | a1fa | yes |
| 14:45.14 | a1fa | it works fine for the first 30 minutes |
| 14:45.17 | a1fa | when you register |
| 14:45.20 | a1fa | then shit hits the fan |
| 14:45.23 | a1fa | and nothing works |
| 14:45.29 | guigouz | do any of you know if I can receive ip calls on my sip phone (for people who dial my ip address) without registering to a proxy ? or should I use h323 ? |
| 14:45.29 | a1fa | i can still make outbound phone calls |
| 14:45.33 | a1fa | which is weerd |
| 14:45.44 | jsharp | Yup. They're a telco. "Must be a problem in your equipment, sir". |
| 14:46.08 | [TK]D-Fender | a1fa : I mean add a hosts entry for sip.broadvoice.com to point to a different node |
| 14:46.21 | [TK]D-Fender | a1fa : THEN you can give up on them entirely :) |
| 14:46.25 | luke-jr_ | jsharp: I got that from iConnectHere. Jerks. |
| 14:47.11 | a1fa | [TK]D-Fender : i've done that.. if i use different peer, it re-registers every 30s |
| 14:48.19 | [TK]D-Fender | :/ |
| 14:48.45 | [TK]D-Fender | All ITSP's suck, just to varying degrees at different intervals... |
| 14:48.59 | brettnem | [TK]D-Fender: heh |
| 14:49.06 | a1fa | they suck big time |
| 14:49.08 | *** join/#asterisk lzhang (n=rjrae@67.95.13.46) |
| 14:49.09 | a1fa | m0x0r f0x0rs |
| 14:49.15 | *** part/#asterisk Samoied (n=Samoied@201-14-148-91.fnsce7002.dsl.brasiltelecom.net.br) |
| 14:51.40 | a1fa | anybody wrote any asterisk policies? |
| 14:52.03 | a1fa | such as, to establish standards across organizations |
| 14:52.25 | a1fa | configuration stards, deployement |
| 14:52.40 | a1fa | how about HAA Pair of *? |
| 14:52.54 | a1fa | that would be sweet |
| 14:53.29 | a1fa | have two A records |
| 14:53.34 | a1fa | for one URL |
| 14:53.42 | a1fa | 1 IN A sip |
| 14:53.44 | a1fa | 2 IN A sip |
| 14:54.09 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 14:54.09 | a1fa | so depending where the phone registers, have a script in asterisk that will determine, to which one it will send |
| 14:54.36 | a1fa | if (200 - unavailable on this system) { send to alternative system } |
| 14:54.43 | a1fa | so if one box goes down |
| 14:55.03 | a1fa | you will still have a reliable service |
| 14:55.15 | a1fa | [TK]D-Fender : hot or not? |
| 14:56.13 | boddy | hii all when local phones thats on meridiand 1c calling sip client via E1 link thats between asterisk and merdian which codec will they use |
| 14:56.34 | *** join/#asterisk cian (n=cian@g5.cian.ws) |
| 14:56.55 | *** join/#asterisk stoffell_h (n=stoffell@d5153F9E0.access.telenet.be) |
| 14:57.05 | [TK]D-Fender | a1fa : What do I know? I'm a 2-bit hack! |
| 14:59.00 | boddy | ? |
| 14:59.01 | a1fa | [TK]D-Fender : does it sound like a good idea or bad? or * has a native HA pair support? |
| 15:00.17 | blitzrage | no HA wiht asterisk |
| 15:01.07 | a1fa | blitzrage : but, you can make it "artificially" |
| 15:02.09 | blitzrage | chan_sip doesn't look at more than the first returned address of an SRV record though |
| 15:02.17 | a1fa | blitzrage : two ips with same A record |
| 15:02.31 | a1fa | 1 IN A sip.tld |
| 15:02.33 | a1fa | 2 IN A sip.tld |
| 15:02.43 | [TK]D-Fender | a1fa : broadvoice is still toast right now, right? |
| 15:02.48 | a1fa | right |
| 15:02.51 | a1fa | i didnt do a sip reload |
| 15:02.52 | a1fa | hold on |
| 15:03.01 | [TK]D-Fender | Shit, one of my clients is whinining to me about that right now! |
| 15:03.07 | blitzrage | I'm having problems too now -- res_odbc cores on me when I try to load it |
| 15:03.17 | a1fa | let me check |
| 15:03.33 | a1fa | [TK]D-Fender : its working, after sip reload |
| 15:03.55 | a1fa | blitzrage : so in incomming extensions |
| 15:03.58 | viperdude | is there anyway to save the SIP DEBUG output to a file? |
| 15:04.33 | a1fa | 200 -> if (200@this.system == unavailable) { call 200@other.system) |
| 15:05.12 | a1fa | so that way, if 200 is registered to 1, it can still call other 20x extensions on 2 |
| 15:05.26 | a1fa | so the users can register to either one via dns |
| 15:05.37 | a1fa | it is very fugly, but in theory, this can work |
| 15:05.53 | a1fa | you can also make roaming clients with this theory |
| 15:06.22 | a1fa | have a SIP in US and a SIP in EU |
| 15:06.36 | a1fa | and have two extensions on both systems |
| 15:06.44 | a1fa | so when you fly from .EU to .US |
| 15:06.58 | a1fa | your wireless will login to the .US, and still be local to the system |
| 15:07.05 | a1fa | cuts down on the latency |
| 15:07.28 | a1fa | but HA pair is what i am thinking would work |
| 15:08.15 | syzygybsd | could you combine that in more then 10 lines? |
| 15:09.27 | a1fa | syzygybsd |
| 15:09.28 | a1fa | n |
| 15:09.28 | a1fa | o |
| 15:09.39 | a1fa | why |
| 15:09.39 | a1fa | do |
| 15:09.39 | a1fa | you |
| 15:09.40 | a1fa | ask |
| 15:10.05 | ghost99 | whats the command to start the asterisk server ? |
| 15:10.22 | syzygybsd | ghost99: depends on how you want to start it |
| 15:10.26 | boddy | [TK]D-Fender: |
| 15:10.37 | boddy | have any idea about my question |
| 15:10.37 | nDuff | ghost99: depends on your packaging. May be just "asterisk", may be /etc/init.d/asterisk, may be something else. |
| 15:10.39 | syzygybsd | asterisk -c will start it from the console |
| 15:10.42 | ghost99 | i dunno it stopped .. and dunno whay that is either .. but logging yeah |
| 15:11.13 | *** join/#asterisk doolph (n=doolph@cpe-00c0a8f7b32d.cpe.cableonda.net) |
| 15:13.40 | [TK]D-Fender | boddy: |
| 15:13.47 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 15:13.59 | [TK]D-Fender | I love these meaningful conversations in IRC :) |
| 15:14.38 | a1fa | baaaaaah |
| 15:14.39 | a1fa | :; |
| 15:15.50 | *** part/#asterisk cian (n=cian@g5.cian.ws) |
| 15:16.03 | a1fa | [TK]D-Fender : how many customers do you support, if it aint a secret |
| 15:16.17 | *** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com) |
| 15:16.56 | b00mer_ | anyone know if I use a Cisco 7914 expansion module... I can use it with SIP if I only use it for speed dials? |
| 15:20.23 | viperdude | b00mer_: AFAIK Cisco 7914 only works with skinny/SCCP not SIP |
| 15:20.35 | znoG | damnit, 80% of faxes are failing |
| 15:20.47 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:21.22 | Zeeek | wazzzaaT? wasim? |
| 15:21.26 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 15:21.28 | MikeJ[Laptop] | 80.123526896% |
| 15:21.28 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 15:22.24 | twisted[asteria] | nice made up statistic |
| 15:22.39 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:22.54 | Zeeek | it's a spam excerpt as in "Dear 80.1235%" |
| 15:23.18 | twisted[asteria] | oh, i usually get "Dear [%name%] |
| 15:23.29 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:23.38 | Zeeek | I got one from an inept spammer with variables for name and subject |
| 15:23.49 | Zeeek | I call it "trailer spam" |
| 15:23.53 | twisted[asteria] | lol |
| 15:23.57 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 15:24.05 | blitzrage | OT: anyone happen to know how to make postgresql log where a transaction is coming from? (i.e. hostname) |
| 15:24.21 | tzanger | blitzrage: it's in the logger configuration |
| 15:24.21 | Zeeek | RTFM :) |
| 15:24.22 | pif | i noticed I get disconnected after a few minutes when routing my calls through an outside * server (using iax2), is there some maximum call lenght setting ? |
| 15:24.26 | twisted[asteria] | blitzrage, make the logging host provide it |
| 15:24.41 | blitzrage | tzanger: I turned on log_hostname=true and I don't see anything... |
| 15:25.25 | twisted[asteria] | ruh roh |
| 15:25.31 | tzanger | did you get the logger putting it somewhere expected? |
| 15:25.32 | blitzrage | getting pushed into a bush is probably the highlight of it... |
| 15:25.34 | MrChimpy | hmm. no DIAL command in AGI? |
| 15:25.35 | tzanger | twisted[asteria]: he likes bushes |
| 15:25.41 | Zeeek | maybe it's log_hostname=1 |
| 15:25.43 | tzanger | twisted[asteria]: I prefer trimmed though |
| 15:25.53 | twisted[asteria] | tzanger, i like dunes |
| 15:25.58 | tzanger | hehe |
| 15:26.02 | blitzrage | tzanger: yah -- its going to logfile and messages it looks like |
| 15:26.35 | Zeeek | since we're in OT mode, does anyone here know anything about IIS6? |
| 15:26.45 | twisted[asteria] | ew.. you mentioned M$ |
| 15:26.59 | Zeeek | I know, I just wasted another day at a cust site with that crap |
| 15:27.05 | tzanger | blitzrage: hmm, that should get it working |
| 15:27.12 | twisted[asteria] | Zeeek, we've had to deal with it before |
| 15:27.23 | twisted[asteria] | Zeeek, anything in particular? |
| 15:27.58 | Zeeek | after some installation we were getting Server Error which after much googline turned out to be pool related and we fixed that |
| 15:28.07 | twisted[asteria] | yeah |
| 15:28.13 | twisted[asteria] | servers in the pool generally don't work right. |
| 15:28.13 | Zeeek | now IIS6 seems to start fine but won't serve pages |
| 15:28.15 | nettie | Hi guys, I would like to know if asterisk 1.2.6 supports VAD/CNG/silet suppression ? I'm having issues with MOH when calling the pstn. |
| 15:28.29 | twisted[asteria] | Zeeek, at all? |
| 15:28.37 | Zeeek | however, telnet locahost 80 <ctrlc> returns an html error |
| 15:28.52 | twisted[asteria] | hmm. |
| 15:29.01 | Zeeek | proving that the server does serve if you piss it off :) |
| 15:29.07 | MikeJ[Laptop] | nettie, you should disable vad on your sip phones and such |
| 15:29.22 | nettie | MikeJ[Laptop] but asterisk supports it? |
| 15:29.25 | twisted[asteria] | Zeeek, if you hit it with a browser does it give an error? |
| 15:29.36 | Zeeek | I don't hate MS because it's fashionable, I hate it because it SUCKS!!!! |
| 15:29.47 | Zeeek | twisted no it hangs |
| 15:29.53 | twisted[asteria] | nettie, no, asterisk does not support VAD/CNG as of yet |
| 15:29.57 | Zeeek | ie can't find..; times out |
| 15:30.00 | twisted[asteria] | nettie, but it's being worked on |
| 15:30.03 | twisted[asteria] | Zeeek, hrm.. |
| 15:30.04 | MikeJ[Laptop] | nettie, it does not deal with it.... |
| 15:30.29 | Zeeek | but looking at the original pool problem ZIllions of people have had it ! |
| 15:30.30 | twisted[asteria] | Zeeek, sorry man, no idea then.. |
| 15:30.30 | nettie | MikeJ[Laptop] and why I should do it on the client moh played by the server |
| 15:30.35 | Zeeek | ok, thx anyway |
| 15:30.58 | Zeeek | I never claimed to be able to fix that crap, they'll have to find someone who can and pay him/her |
| 15:30.59 | MikeJ[Laptop] | nettie, because it wont send your client audio if the client is not sending it audio |
| 15:31.10 | nettie | MikeJ[Laptop] ahhhhhh |
| 15:31.23 | MikeJ[Laptop] | hense, "it does not deal with it" |
| 15:31.37 | nettie | MikeJ[Laptop] ok, I have to figure out how to disable it on my polycom soundpoiunt 5001 |
| 15:31.38 | nettie | 501 |
| 15:31.59 | znoG | Apr 28 12:24:05 NOTICE[14224]: chan_sip.c:3608 process_sdp: No compatible codecs! |
| 15:32.05 | Zeeek | nettie I don't remember touching anything on the 500 |
| 15:32.14 | znoG | the call is from alaw to g729, which is should be able to transcode between... |
| 15:32.31 | Zeeek | znoG is 729 working for sure? |
| 15:32.33 | znoG | and the call is established and they can hear each other.. so why the error message? |
| 15:32.34 | MikeJ[Laptop] | you have g729 lic installed? |
| 15:33.06 | MikeJ[Laptop] | the call is fine, but you are getting no compatible codecs? |
| 15:33.12 | znoG | yup |
| 15:33.27 | ghost99 | manxpower: gr0mit ended up helping me .. I got that dodgy card working .. hes a cool guy .. hes just help me for last hour or so ... thankyou gr0mit and thanks. time for me to go to bed bed ... tomorrows and new day and thaks to you guys in some months we get the kick arse gear becasue I will have shown my business partner how great this stuff is ... Cheers all I'm on me way !! |
| 15:34.03 | vader-- | do you guys know how many lines the cisco 7940 can handle? |
| 15:34.11 | vader-- | i see buttons for two extensions |
| 15:34.11 | nettie | guys, if the called make noises or the environment has a lot of background noise MOH is fine.. if he's totally silent he cant hear it anymore.. |
| 15:34.20 | nettie | called == called party |
| 15:34.22 | vader-- | but the config file shows settings for 6 extensions |
| 15:34.51 | mercestes | The cisco 7960 has six buttons....and they use the same .cfg file. |
| 15:34.58 | vader-- | gotcha |
| 15:34.58 | mercestes | you may ignore the extra four placeholders. |
| 15:35.16 | twisted[asteria] | nettie, that's because you have silence suppression on |
| 15:35.20 | twisted[asteria] | nettie, again, turn that off |
| 15:35.22 | vader-- | so the 7940 only handles 2 lines? |
| 15:35.29 | Zeeek | can anyone in the US make a 800 call on a regular phone line for me? |
| 15:35.35 | nettie | twisted[asteria] on asterisk or on the clients? |
| 15:35.38 | mercestes | It can only handle two presences..... |
| 15:35.42 | MikeJ[Laptop] | nettie, yeah, that silence supression on the phone... |
| 15:35.46 | nettie | phone.. |
| 15:35.46 | nettie | damn |
| 15:35.50 | mercestes | which may be two instances of a single sip presence or two seperate sip presences... |
| 15:36.00 | vader-- | gotcha |
| 15:36.12 | mercestes | In PBX-speak you could consider it "two lines." |
| 15:36.21 | MikeJ[Laptop] | asterisk is using blocking reads from the inbound rtp.. it won't send anything to the phone till it gets somthing from the phone |
| 15:36.25 | nettie | I have to figure out how to do it.. my voip carrier confirmed me that they dont have silence suppression enable, so it must be pghone. |
| 15:36.28 | mercestes | if your used to turnkey systems. Each of those "lines" however can handle multiple callls. |
| 15:36.48 | pif | how to check the current AbsoluteTimeout value? |
| 15:37.02 | twisted[asteria] | nettie, if your call is coming in from the provider and you're hitting hold on your phone, then it *IS* turned on on the provider side |
| 15:37.19 | brettnem | asterisk + HA.. har har har |
| 15:37.22 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 15:37.33 | twisted[asteria] | the phone just returns the RTP path back to asterisk to provide MOH. once you hit the hold button on your phone, your phone is no longer involved in the media path |
| 15:38.09 | nettie | twisted[asteria] ok I try to replicate what you said. |
| 15:38.16 | twisted[asteria] | k |
| 15:38.17 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 15:39.48 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 15:40.01 | nettie | twisted[asteria], that's exaclty my thought.. anyway I tried calling in from the provider and put the call on hold. I confirm you that the MOH is jumpy. |
| 15:40.17 | bkw__ | VAD |
| 15:40.27 | twisted[asteria] | no kidding bkw |
| 15:40.32 | bkw__ | :P |
| 15:40.34 | bkw__ | hey ho.. ltns |
| 15:40.37 | vader-- | mercestes thanks for the info on the phones |
| 15:40.44 | twisted[asteria] | heh |
| 15:40.46 | twisted[asteria] | bkw__, how's life |
| 15:40.50 | nettie | twisted[asteria] also.. if I put on hold one of my collegue here at the office MOH is perfect |
| 15:40.58 | twisted[asteria] | yea |
| 15:41.05 | bkw__ | twisted[asteria], GREAT!!! |
| 15:41.09 | twisted[asteria] | it's probally off on the phone |
| 15:41.43 | twisted[asteria] | bkw__, :) |
| 15:41.52 | MrChimpy | so, any AGI implementation type people here? |
| 15:42.27 | mog_work | twisted[asteria], where did you get a moonpie? |
| 15:42.35 | twisted[asteria] | mog_work, at teh store |
| 15:42.42 | MrChimpy | since I can't dial using AGI, i'm attempting to set a variable then dial that from the dialplan |
| 15:42.53 | mog_work | real moonpie or chattanooga variety? |
| 15:43.26 | pif | does * have dns issues? |
| 15:43.29 | twisted[asteria] | mog_work, last time i checked they're all from chattanooga |
| 15:43.41 | pif | troubling resolving iax hosts for instance? |
| 15:43.54 | MrChimpy | so I do my AGI call then a Dial($TODIAL}@demo) |
| 15:43.56 | pif | s/troubling/trouble/ |
| 15:43.57 | unixgeek | Question concering a pri. I am looking at a pri with zttool. No matter what I set the clocking source to, it always shows "internally clocked". That does not seem right. |
| 15:43.59 | MrChimpy | in the dialplan |
| 15:44.03 | MrChimpy | doesn't appear to work |
| 15:44.26 | mog_work | nope twisted[asteria] they are from everywhere, at least ones at mardi gras |
| 15:44.31 | rpm | does anyone know how to transfer a call with astmanproxy? |
| 15:44.43 | MikeJ[Laptop] | twisted[asteria], was it 7-11? |
| 15:45.26 | twisted[asteria] | mog_work, The south?s favorite snack food, the MoonPie originated in 1917 in Chattanooga. |
| 15:45.32 | *** join/#asterisk steve12345 (n=steve@store-fw.porchlight.ca) |
| 15:45.32 | twisted[asteria] | www.moonpie.com |
| 15:45.34 | twisted[asteria] | pwnt. |
| 15:45.42 | mog_work | it started there |
| 15:45.47 | mog_work | i wont deny that |
| 15:45.57 | mog_work | but i dont like the cahttanooga moonpie factory moonpies |
| 15:46.02 | mog_work | they are too thick |
| 15:46.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 15:46.21 | twisted[asteria] | mog_work, then you don't like real moonpies |
| 15:47.03 | mog_work | meh i guess i dont |
| 15:47.33 | [TK]D-Fender | a1fa : I've got about a half-dozen places that I've consulted for that I still do occasionaly maintenance for. |
| 15:47.42 | [TK]D-Fender | a1fa : its a side business |
| 15:49.42 | bsdfreak | :] |
| 15:50.28 | *** join/#asterisk manalive (i=yashy@mail.yashy.com) |
| 15:50.30 | *** join/#asterisk avogadro (n=rj@67.95.13.46) |
| 15:50.53 | jarrod | is there a better pkg than high availability for a virtual ip configuration between two servers |
| 15:51.02 | manalive | exten => s,1,Wait(4) |
| 15:51.02 | manalive | exten => s,2,Answer |
| 15:51.27 | manalive | I have this.. is there a way to not go to the next step (answering machine/record) if one of the telephones in my house is picked up? |
| 15:52.34 | ManxPower | manalive, how about exten => s,1,Wait(15) exten => s,2,WaitForRing |
| 15:52.46 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 15:52.49 | ManxPower | That way if you pick up the house phone in 15 seconds, Asterisk will never pick up. |
| 15:53.06 | nettie | twisted[asteria] if local calls with my collegues MOH is fine and when calling trough my voip carrier I have issues I suppose it's definitely related to the carrier.. |
| 15:53.08 | ManxPower | But the REAL solution is to not be such a cheap bastard and run all calls thru Asterisk |
| 15:53.13 | Zeeek | I need an emergency replacement for nufone DID. Not for business, my wife is leaving for the US for a week. I need it so she can call me. Any suggestions folks? |
| 15:53.18 | manalive | ManxPower: okay thanks.. will try. |
| 15:53.32 | ManxPower | manalive, "show application waitforring" |
| 15:53.40 | Zeeek | as it happens, my asterlink DID doesn't seem to be working |
| 15:53.43 | MrTelephone | has anyone had trouble with tdm2400p card where one person hears another conversation on there line? |
| 15:53.56 | ManxPower | Yay! My agile RF modulators are "out for delivery" |
| 15:54.06 | tzanger | ManxPower: nice... what are you doing? |
| 15:54.29 | ManxPower | tzanger, going to play with setting up a "mini cable service" here at the campground. |
| 15:54.46 | tzanger | ahhh |
| 15:54.50 | tzanger | mini-circuits amps? |
| 15:55.13 | ManxPower | the ditch witch should be here this afternoon, so we should have all the cat5, coax, and fiber in the ground by next week. |
| 15:55.17 | manalive | ManxPower: exactly what I was looking for.. thanks. |
| 15:55.37 | ManxPower | the modulators to allow you to turn up the power quite a bit. |
| 15:55.51 | rpm | Is it possible for me to transfer calls from an Web-based attendant console to another destination? I'm trying to figure out how to communicate with astmanproxy and tell it to do a transfer. |
| 15:56.11 | MrChimpy | hmm! |
| 15:56.15 | MrChimpy | Apr 28 17:01:03 WARNING[4781]: app_dial.c:1563 dial_exec_full: Had to drop callbecause I couldn't make Zap/1-1 compatible with IAX2/tonyh-4 |
| 15:56.31 | MrChimpy | what's up with that? missing codecs I assume |
| 15:56.41 | rpm | MrChimpy: show translation |
| 15:56.52 | rpm | MrChimpy: does it say it can transcode between your codecs? |
| 15:56.56 | ManxPower | MrChimpy, any messages about can't translate or can't make compatable |
| 15:59.16 | ManxPower | tzanger, i.e. put the camera at the gate on one channel, pipe a couple of DVD players into the modulators for...educational programming, there is some Free-to-Air satallite programming that is also available that I can put on channels, also have a channel with local events and a channel for local weather. |
| 15:59.25 | *** part/#asterisk manalive (i=yashy@mail.yashy.com) |
| 15:59.59 | tzanger | ManxPower: yep |
| 16:01.02 | tzanger | why DVD players though, I'd imagine a simple PC streaming content would be more robust, DVB-S will handle your FTA stuff... |
| 16:01.58 | *** part/#asterisk SparFux (n=player@e182018201.adsl.alicedsl.de) |
| 16:02.05 | LostFrog | lol.. 'educational programming' :) |
| 16:04.06 | MrChimpy | yep, cool |
| 16:04.18 | MrChimpy | made my IAX phone use gsm and it'll do the call fine |
| 16:05.38 | LostFrog | I think I just realized that my users can dial 9911 to get 911. :( |
| 16:06.29 | MikeJ[Laptop] | you don't want anyone to be able to call 911? |
| 16:06.35 | mut | you think you just realized? |
| 16:06.37 | MikeJ[Laptop] | what if they have an emergency? |
| 16:06.42 | LostFrog | Not from other offices.. |
| 16:06.48 | LostFrog | Especially India. |
| 16:07.15 | LostFrog | Local users can call 911 and it goes out on pots. |
| 16:07.26 | LostFrog | But, we all have cell phones too. |
| 16:07.46 | MikeJ[Laptop] | hmmm |
| 16:07.58 | MikeJ[Laptop] | I don't think that will work jsharp |
| 16:08.16 | *** join/#asterisk Skarmeth (n=Skarmeth@201009037175.user.veloxzone.com.br) |
| 16:08.19 | MrChimpy | my agi app works. i'm so clever. |
| 16:08.20 | Skarmeth | hi all |
| 16:08.22 | LostFrog | Depends on what result he wants, MikeJ[Laptop]. If he wants a vacation, it will. |
| 16:09.14 | jsharp | A vacation at the Gray Bar Inn. |
| 16:09.29 | LostFrog | Exactly. |
| 16:09.38 | LostFrog | A cot and three hots a day. |
| 16:10.14 | *** join/#asterisk saftsack (n=saftsack@p54A7F649.dip.t-dialin.net) |
| 16:10.54 | jsharp | Actually, that doesn't sound too bad. |
| 16:11.10 | Skarmeth | I have a doubt with a Asterisk setup here. I have a old PBX and I connect Asterisk to it throught a PBX Extension. In that old PBX, when a call it's transferred to another extension, the first extension becomes idle. But if a call comes to asterisk via that extension and Asterisk tryes to transfer it to another extension on the old PBX, it use another extension to do so. How can I simulate the old PBX behavior beetwen Asterisk an |
| 16:11.11 | Skarmeth | d the old PBX? |
| 16:14.13 | *** join/#asterisk Igorz (i=Igorz@82.193.84.145) |
| 16:16.16 | mut | Apr 28 12:15:26 WARNING[9991]: chan_sip.c:4683 build_rpid: Unsupported callingpres (64) |
| 16:16.19 | mut | ??? |
| 16:16.29 | Igorz | hi |
| 16:16.43 | Igorz | where i can bou asterisk hardware |
| 16:16.51 | Igorz | boy |
| 16:16.51 | Igorz | buy |
| 16:17.07 | [TK]D-Fender | Igorz : All sorts of places. Depends best on where you are. |
| 16:17.08 | praet | voipsupply has a lot |
| 16:17.18 | [TK]D-Fender | And what you want... |
| 16:18.00 | C4T3l | how do i start my own telecom??? |
| 16:18.36 | zoa | haha lol |
| 16:18.39 | zoa | with a lot of money |
| 16:18.40 | zoa | :) |
| 16:18.48 | mut | doesn't that THAT much |
| 16:19.01 | mut | more paperwork and legal shit than money |
| 16:19.04 | mut | it's a pain |
| 16:19.15 | mut | and long |
| 16:19.37 | twisted[asteria] | and hard |
| 16:19.56 | file | and filled with |
| 16:20.41 | mut | we're on our way tho! |
| 16:20.53 | mut | another year and it'll be official i hope |
| 16:21.35 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:22.04 | LostFrog | Yay.. you can split gsm files with dd. |
| 16:22.09 | rpm | is it possible to dynamically create a queue with asterisk without using realtime? |
| 16:22.54 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 16:23.26 | C4T3l | can you guys point me to a site or something that can help with starting my own telecom? :) |
| 16:23.40 | eric_hill | http://www.att.com |
| 16:23.40 | mercestes | http://www.bankruptcy.com |
| 16:23.44 | mut | call a lawyer |
| 16:25.39 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:26.42 | SplasPood | That question was too funny. |
| 16:27.29 | *** join/#asterisk IceManRISK (n=kart@201-14-4-45.mganm7005.dsl.brasiltelecom.net.br) |
| 16:28.42 | eric_hill | Anyone have any ideas how to remove a dynamic queue member that doesn't answer the phone? |
| 16:29.15 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 16:29.18 | C4T3l | rm -rf / |
| 16:29.38 | russellb | C4T3l: so funny! |
| 16:29.38 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:29.42 | eric_hill | C4T3l: Let me try that - just a second... |
| 16:29.49 | eric_hill | [NO CARRIER] |
| 16:29.51 | C4T3l | good luck! |
| 16:30.00 | *** join/#asterisk MrBelvedr (n=tt@ip68-100-113-84.dc.dc.cox.net) |
| 16:30.00 | C4T3l | ha |
| 16:31.36 | twisted[asteria] | ever seen the /disco lights in irc? |
| 16:31.55 | zoa | show me |
| 16:31.56 | zoa | :p |
| 16:32.08 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 16:32.13 | eric_hill | I tried it and it didn't do anything. What happens when you try it? |
| 16:32.15 | twisted[asteria] | i can't show you, you have to show yourself |
| 16:32.26 | zoa | omg |
| 16:32.27 | zoa | this is cool |
| 16:32.31 | zoa | what an easter egg |
| 16:32.34 | zoa | haha |
| 16:32.40 | zoa | i feel like partying now |
| 16:32.40 | LostFrog | lol |
| 16:32.48 | LostFrog | Noone did it. :( |
| 16:32.53 | LostFrog | /disco lights |
| 16:32.57 | LostFrog | Didn't work for me. |
| 16:33.02 | twisted[asteria] | aw |
| 16:33.04 | zoa | you need to type it fast enough |
| 16:33.13 | LostFrog | /disco lights |
| 16:33.39 | russellb | twisted[asteria]: come join my new channel, we're having a party in #2,000 |
| 16:33.45 | twisted[asteria] | russellb, word! |
| 16:34.16 | russellb | LostFrog: i guess you're invited too |
| 16:34.22 | noname32 | lol |
| 16:34.49 | noname32 | does alt f4 change the speed of the strobes? |
| 16:35.19 | twisted[asteria] | only in windows |
| 16:35.19 | russellb | yep |
| 16:35.20 | eric_hill | So does Ctrl-D |
| 16:35.25 | noname32 | haha alt f4 works in alot of x11 wm to ahah |
| 16:35.34 | *** part/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 16:35.40 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 16:35.48 | mut | haha |
| 16:35.49 | mut | fool |
| 16:35.50 | noname32 | lol |
| 16:35.54 | [TK]D-Fender | :D |
| 16:36.16 | [TK]D-Fender | was quicker than "/clear" :D |
| 16:36.17 | noname32 | ctrl z for shell users :) |
| 16:36.24 | mut | riiight |
| 16:36.32 | rpm | is it possible to create queues on the fly? |
| 16:36.36 | *** join/#asterisk daeamarth (n=daeamart@166.70.238.106) |
| 16:36.39 | twisted[asteria] | dunno, is the fly big enough? |
| 16:36.47 | russellb | rpm: you can put queues in realtime |
| 16:36.49 | LostFrog | I can never stand on a fly long enough to answer calls. |
| 16:36.53 | mut | they do make some pretty cool microdevices no a days |
| 16:36.54 | noname32 | lol lot of smart asses today haha |
| 16:36.58 | [TK]D-Fender | rpm : The real trick is pinning the fly long enough to insall :) |
| 16:36.59 | mut | so you may be able to do it |
| 16:37.03 | LostFrog | lol @ twisted[asteria]. Twisted minds think alike. |
| 16:37.09 | twisted[asteria] | LostFrog, :P |
| 16:37.20 | russellb | at least i had a serious answer :-p |
| 16:37.31 | twisted[asteria] | russellb, heh. |
| 16:37.39 | mut | it's friday |
| 16:37.42 | mut | who wants to be serious |
| 16:37.46 | LostFrog | THANK GOD! |
| 16:37.51 | twisted[asteria] | oooh |
| 16:37.55 | mut | shite i feel like goin home already |
| 16:37.56 | twisted[asteria] | and i don't work at the bar tonight! |
| 16:37.59 | mut | day is dragging |
| 16:38.02 | twisted[asteria] | woohoo! |
| 16:39.30 | rpm | it seems the only way i can do this is: 1. calling party calls remote party, 2. remote party picks up phone, 3. mysql dynamically creates a unique queue id, 4. dial plan jumps to new context/extension/priority and adds both parties to the queue, 5. from my web based interface i transfer a call... some how it needs to communicate and remove the called party and transfer to a part of the dialplan. |
| 16:40.09 | twisted[asteria] | rpm -e brain |
| 16:40.24 | LostFrog | lol |
| 16:40.51 | twisted[asteria] | that made my head spin |
| 16:41.08 | C4T3l | lol |
| 16:42.11 | noname32 | yum update brain |
| 16:42.12 | rpm | its making my head spin trying to find a way to do this.. i need to integrate some kind of attendant console to asterisk. |
| 16:43.48 | LostFrog | I wonder if FOP can handle more than one * box at a time.. |
| 16:43.51 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 16:44.25 | LostFrog | I'm 99% sure it doesn |
| 16:44.27 | rkr245 | rpm:try your asterisk box with SER |
| 16:44.29 | LostFrog | doesn't, even. |
| 16:44.47 | viperdude | LostFrog: we have monitered 2 * boxes with 1 FOP |
| 16:45.30 | viperdude | however we have now split them to two instances of FOP as the number of extensions grew to make it easier on the eye for those that needed to use it |
| 16:45.55 | LostFrog | viperdude: Cool. I may have to talk to you about that later. |
| 16:47.17 | viperdude | i am off out in a few minutes but the docs explain how to do it.. in op_server.cfg you set up multiple manager logins and then in op_buttons.cfg you specify against each button which server it is on |
| 16:47.23 | daeamarth | asterisk work with mysql 5.0? |
| 16:47.34 | C4T3l | has anybody made * and ser work together?? |
| 16:47.43 | Igorz | where i can find asterisk resseler in europe? |
| 16:47.49 | Igorz | east europe |
| 16:47.50 | viperdude | C4T31: yes and also running on the same box :-) |
| 16:48.29 | LostFrog | viperdude: thanks. |
| 16:48.40 | viperdude | LostFrom: yw |
| 16:48.54 | b00mer_ | is there a good link on why I would want to run SER or a SIP proxy and asterisk together... I want to try an understand the benefits |
| 16:49.29 | viperdude | SER off loads the registry from asterisk and it also has powerful routing capabilitites |
| 16:49.56 | C4T3l | viperdude: load-balance asterisk?? |
| 16:50.24 | b00mer_ | if I have a office of 100 people... all using hard phones on their desks... does a proxy provide an advantage? |
| 16:50.45 | b00mer_ | what about if I have people connecting from their homes over the internet? |
| 16:50.58 | viperdude | yes you could along with DNS round-robin for instance.. i.e if asterisk was the PSTN gateway you would not need to specify each SIP UA in the * boxes |
| 16:51.00 | [TK]D-Fender | b00mer_ : for 100 I wouldn't bother. |
| 16:51.19 | viperdude | b00mer_: yes thats fine there is a nat helper module in SER |
| 16:51.33 | viperdude | yes 100 is ok on * |
| 16:51.59 | C4T3l | viperdude: any sample ser.cfg's out there that you know of? |
| 16:52.04 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 16:52.07 | b00mer_ | i've been getting asked for people to get hardphones in their houses that will connect to the company PBX |
| 16:52.11 | viperdude | ok well there is a beer waiting with my name on it... so i am outta here |
| 16:52.18 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 16:52.19 | b00mer_ | I initially wanted to use a VPN |
| 16:52.22 | [TK]D-Fender | SER comes in when you want better billing control or need STUN, scalability, etc |
| 16:52.22 | C4T3l | viperdude: noooooooooooooo!! |
| 16:52.34 | b00mer_ | but it seems like that is getting resistance |
| 16:52.47 | florz | |
| 16:52.57 | b00mer_ | they want a non-vpn solution... does a SIP proxy help with that? |
| 16:53.21 | b00mer_ | so I don't have to have direct (or maybe I would) access from the internet to my pbx |
| 16:53.57 | cornet | why non-vpn ? |
| 16:54.10 | b00mer_ | can't do the QoS they want to do |
| 16:54.36 | [TK]D-Fender | b00mer_ : how many remote phone connections? |
| 16:54.41 | b00mer_ | and since its a hard phone... they would need to setup soho routers |
| 16:54.47 | b00mer_ | with VPN capability |
| 16:54.54 | stakk | daeamarth: yes, i'm running asterisk on ubuntu dapper MySQL 5. works fine |
| 16:54.56 | b00mer_ | and then route IP space etc. |
| 16:55.06 | b00mer_ | [TK]D-Fender : about 10 |
| 16:55.16 | [TK]D-Fender | b00mer_ : No big deal. Just use * normally..... |
| 16:55.20 | daeamarth | stakk: thanks! |
| 16:55.47 | stakk | welcome |
| 16:56.11 | b00mer_ | normal use... is there a way to have an asterisk server in the DMZ that would then communicate to the PBX on the internal networks? |
| 16:56.14 | [TK]D-Fender | b00mer_ : Good tip though, for that many extensions, you might want to make a small scripts to generate a merge file for sip.conf and extensions.conf. |
| 16:56.38 | [TK]D-Fender | b00mer_ : Mine here is on the main LAN, and not DMZ'd, just a few ports forwarded. |
| 16:56.58 | [TK]D-Fender | in fact its even on a seperate subnet than my internet router |
| 16:56.59 | b00mer_ | yea... thats possible... just violates corp policy |
| 16:57.19 | [TK]D-Fender | b00mer_ : Policy says NO forwarding of any kind allowed? |
| 16:57.45 | b00mer_ | [TK]D-Fender : no forwarding from the internet to internal without proxy or DMZ |
| 16:58.15 | cornet | sounds like a fair policy |
| 16:58.48 | b00mer_ | so I could have 2 pbx linked right? IAX2 trunk between? |
| 16:59.02 | b00mer_ | one on a dmz for internet phones |
| 16:59.12 | [TK]D-Fender | b00mer_ : Well DMZ means no internal access.... thats BS. |
| 16:59.20 | b00mer_ | trunked to the one internal for inside |
| 16:59.43 | [TK]D-Fender | b00mer_ : but then what you're now suggesting says that traffic CAN be allowed from DMZ to main... how is THAT safe? |
| 16:59.45 | b00mer_ | [TK]D-Fender : I disagree... DMZ means that it needs to pass though before coming in or out |
| 17:00.14 | [TK]D-Fender | b00mer_ : All too messy... |
| 17:00.15 | b00mer_ | [TK]D-Fender : safer than directly from some k12 school in south korea |
| 17:00.33 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 17:00.48 | [TK]D-Fender | b00mer_ : Well double firewall it an only all SIP UDP. |
| 17:00.53 | [TK]D-Fender | allow* |
| 17:01.19 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 17:01.31 | cornet | [TK]D-Fender: there are no hard set rules for DMZ |
| 17:01.32 | Druken | [TK]D-Fender: don't you work man?? :) your always here |
| 17:01.34 | MikeJ[Laptop] | you need to allow rtp too |
| 17:01.58 | b00mer_ | but a SER / sip proxy wouldn't help in this case? instead of another asterisk |
| 17:02.06 | [TK]D-Fender | Druken : I'm loosely monitored, and I *am* the IT dept :) |
| 17:02.12 | cornet | hehe |
| 17:02.13 | russellb | does anyone here have an FC5 system that you could test something on for me? |
| 17:02.15 | bsdfreak | >:D |
| 17:02.24 | Druken | [TK]D-Fender: since when is irc IT ?? :) hehehe |
| 17:02.41 | [TK]D-Fender | MikeJ[Laptop] : I considered that somewhat included in my "bulk statement". |
| 17:02.57 | [TK]D-Fender | Druken : it is when I'm in #asterisk and I'm supporting an # server here :) |
| 17:03.13 | [TK]D-Fender | I'm looking for help here pbviously! |
| 17:03.30 | SplasPood | I'm getting tons of these on console, anyone know? Apr 28 13:06:58 WARNING[19598]: frame.c:1252 ast_codec_get_len: Unable to calculate sample length for format unknown |
| 17:03.37 | Druken | oh, right... sorry... i must have missed that :) |
| 17:03.41 | LostFrog | There aren't any psychiatrists here, [TK]D-Fender, AFAIK. |
| 17:04.24 | b00mer_ | hey! [TK]D-Fender has help my lousy ass more times than I can count... |
| 17:04.38 | Druken | LostFrog: who needs one of those? just tell a stranger all your troubles :) |
| 17:04.38 | b00mer_ | I value your presence |
| 17:04.51 | LostFrog | Druken: that's why there are bartenders, I think. |
| 17:05.10 | b00mer_ | helped with my lousy ass asterisk problems... needed clarification |
| 17:05.20 | LostFrog | lol |
| 17:05.26 | [TK]D-Fender | ... |
| 17:05.31 | [TK]D-Fender | *cough* |
| 17:05.33 | Druken | bartenders for the first session, prostitues for the second, medical examiners for the third |
| 17:06.11 | LostFrog | Shouldn't the VD clinic enter into that somewhere? |
| 17:06.25 | [TK]D-Fender | LostFrog : Thats the ME part :) |
| 17:06.33 | Druken | well, wouldn't you find a M.E, at the VD clinic? |
| 17:06.41 | [TK]D-Fender | LostFrog : Just extended to a natural conclusion... |
| 17:06.57 | LostFrog | I wouldn't think there would be ME at the VD clinic.. |
| 17:07.03 | b00mer_ | unless he went to the third session as well |
| 17:07.05 | LostFrog | A ME is at the morgue. |
| 17:07.11 | [TK]D-Fender | LostFrog : Who needs a VD Clinic when its already killed you? :) |
| 17:07.13 | LostFrog | An, even. |
| 17:07.19 | [TK]D-Fender | LostFrog : See above :D |
| 17:07.34 | LostFrog | [TK]D-Fender: I was responding to Druken. |
| 17:07.41 | Druken | :) |
| 17:07.42 | b00mer_ | I am confused |
| 17:07.55 | Druken | wow, that wasen't very hard :) |
| 17:07.55 | [TK]D-Fender | Hello confused, nice to meet you! |
| 17:08.06 | b00mer_ | usually isn't |
| 17:08.16 | b00mer_ | don't worry... I'll get it in 3-4 minutes |
| 17:08.21 | Druken | atleast your honest :) |
| 17:09.00 | Druken | not hard to confuse me either... try to explain to me how ss7 works... lost after it.... |
| 17:09.12 | praet | "call me a cab" "ok.. you're a cab" |
| 17:09.22 | [TK]D-Fender | Druken : Now if my work were to ask about my trolling in LavaLife, IM & SSH windows open to my outside clients, and so on, that'd be another matter :) |
| 17:09.42 | MikeJ[Laptop] | Druken, honestly there are some great ss7 descriptions on ss7box.com |
| 17:09.44 | Druken | lavalife? |
| 17:09.46 | Druken | hehe |
| 17:09.48 | LostFrog | Not to mention the pr0n' downloads. |
| 17:10.18 | Druken | is lavalife just in t.o. or like everywhere? |
| 17:10.24 | wasim | ss7box rocks |
| 17:10.52 | MikeJ[Laptop] | that's just because mr. muehler is an insane protocol man |
| 17:10.53 | Druken | MikeJ[Laptop]: the only reason i don't know ss7 is because i've never looked at it, i have no use for it at the moment |
| 17:11.03 | [TK]D-Fender | LostFrog : No that be a blatant violation of corp policy and give them grounds to fire me. As it is, this would merely be a reprimand. |
| 17:11.06 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 17:11.12 | MikeJ[Laptop] | Druken, it's really quite cool stuff |
| 17:12.03 | Druken | i can imagine, i'd love to be able to signal bells switches to switch the calls instead of mine :) |
| 17:12.37 | Druken | would be great for call forwards |
| 17:13.09 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:14.14 | [TK]D-Fender | Druken : Sounds like 2BT. |
| 17:14.25 | Druken | 2bt? |
| 17:14.29 | starlein | can someone tell me why asterisk no more doesn't fork its processes? only got 1 main process without threads. i think it's related to my glibc update?! |
| 17:15.25 | file | Asterisk doesn't fork, it creates threads... make up your mind :) |
| 17:15.42 | starlein | ah okay i said it in wrong term |
| 17:15.54 | starlein | but i think the smp is useless now |
| 17:15.54 | file | how do you know the threads are not up and running? |
| 17:16.07 | starlein | they don't appear at process list? |
| 17:16.19 | file | they don't have to, that doesn't mean they are not going |
| 17:16.33 | starlein | ahhhh |
| 17:16.42 | file | is Asterisk working? |
| 17:16.45 | starlein | yeah sure |
| 17:16.47 | starlein | just was wondering |
| 17:16.52 | file | then you have multiple threads |
| 17:16.53 | starlein | if smp gets used |
| 17:16.59 | starlein | ah okay thanks man |
| 17:17.28 | Druken | hehe generally if it works, i dun ask :) |
| 17:17.49 | Druken | it's when it doesn't work that i start to wonder why... |
| 17:18.13 | starlein | hehe yes same here, but was only wondering about smaller process count at system |
| 17:18.30 | starlein | before glibc update asterisk had up to 30threads |
| 17:18.32 | file | people have freaked out about the same thing on the mailing list |
| 17:18.38 | file | "why are they so many asterisk processes?!?" |
| 17:18.54 | file | "why do I only have one and this guy has so many?" |
| 17:19.12 | starlein | i liked it, because i know SMP is working with many threads better |
| 17:19.20 | [TK]D-Fender | file : since I run 2.4 w/o SMP I'm used to seeing tons of processes. I got freaked out when they disappeared on a new install :) |
| 17:19.22 | Druken | "does it work?", "yes", "then who cares!" :) |
| 17:19.23 | starlein | i doubt that this 1 process will get 2x100% cpu |
| 17:19.43 | file | threads get distributed over the CPUs, Asterisk is multithreaded, it will be fine |
| 17:19.56 | file | [TK]D-Fender: I bet |
| 17:20.19 | starlein | hehe okay |
| 17:20.20 | file | if threading was broke on your system then lots of the Asterisk modules and some of the core would choke |
| 17:21.33 | russellb | file: that's an understatement |
| 17:21.33 | file | russellb: I'm trying not to scare them |
| 17:21.33 | starlein | :D |
| 17:21.33 | starlein | you can't |
| 17:21.33 | starlein | i dream of asterisk |
| 17:21.33 | file | oh yeah? |
| 17:21.33 | starlein | sure |
| 17:21.33 | Druken | sick bastard |
| 17:21.33 | russellb | starlein: that's hot |
| 17:21.33 | starlein | about complex dialplans |
| 17:21.34 | file | I woke myself up out of sleep because I was writing chan_iax2 code in my sleep |
| 17:21.34 | starlein | fat agi scripts |
| 17:21.34 | starlein | and so on |
| 17:21.34 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 17:21.42 | russellb | file: i do that sometimes :/ |
| 17:21.42 | starlein | heh |
| 17:21.43 | file | just this morning.; |
| 17:21.44 | [TK]D-Fender | I do too... they just end with me up screaming in a cold sweat :) |
| 17:21.49 | Druken | file: that's scary man... |
| 17:22.26 | russellb | file: we're pretty hardcore ... |
| 17:22.28 | Druken | i had a telemarketing job years ago, i caught myself doing my job in my sleep, i quit a week later... that's just wrong |
| 17:22.32 | [TK]D-Fender | "You know you're in trouble when you wake up from a nightmare, only to realize you weren't ASLEEP" |
| 17:22.36 | file | russellb: 'tis scary |
| 17:24.00 | russellb | MrChimpy: Asterisk is the hotness |
| 17:24.05 | starlein | yeps |
| 17:24.10 | starlein | it is definitly |
| 17:24.18 | MrChimpy | AGI seems to lack a lot of things. i'm assuming there are reasons |
| 17:24.25 | file | now, you will have to buy lots of Digium hardware |
| 17:24.26 | MrChimpy | like there's no DIAL |
| 17:24.35 | MrChimpy | and there's no SAY FESTIVAL :) |
| 17:24.38 | file | MrChimpy: you can execute outside applications, like Dial |
| 17:24.47 | MrChimpy | howso? |
| 17:24.59 | file | there's an AGI call to execute applications. |
| 17:25.16 | MrChimpy | at the moment i'm setting a channel variable from the AGI and having the dialplan then dial the number if the variable is set |
| 17:25.29 | MrChimpy | ah, i'll have a further look. this is my first AGI thing |
| 17:25.39 | file | you can execute most anything you can execute in the dialplan in an AGI |
| 17:25.43 | russellb | MrChimpy: "exec" |
| 17:25.43 | MrChimpy | file: we'll buy LOTS of digium hardwarer |
| 17:25.58 | MrChimpy | like 10 TE411s |
| 17:26.09 | russellb | MrChimpy: yay, that lets file and I eat! |
| 17:26.14 | xachen | I sure hope your not putting them all in the same box :P |
| 17:26.20 | MrChimpy | :) |
| 17:26.24 | *** join/#asterisk }cytrak{ (n=noneon1@208.63.19.179) |
| 17:26.24 | MrChimpy | two per box ;) |
| 17:26.31 | file | we like eating |
| 17:26.34 | file | means we can code more! |
| 17:26.45 | xachen | blah |
| 17:26.46 | xachen | whats food? |
| 17:26.46 | Druken | food? share! |
| 17:27.16 | }cytrak{ | is there anyway I can tell a PBX box to forward any signals coming in from PRI-2 to check the dialplan and route the call out to thw world ? |
| 17:27.37 | Druken | i should invite the toronto aug up for a bbq some weekend... |
| 17:27.37 | MrChimpy | i'll make you guys some money getting this place buying lots of digium stuff, then I can get payback doing easy voip installs as a contractor :) |
| 17:28.11 | }cytrak{ | I got my * box talking to a siemens PBX over a second PRI/T1 and I can call from * to any internal phone but I can't dial out |
| 17:28.14 | }cytrak{ | to the worls |
| 17:28.15 | file | russellb: pfft |
| 17:28.16 | }cytrak{ | to the world |
| 17:28.18 | MrChimpy | digium could/should make tons of money employing people to do that crap |
| 17:28.22 | file | russellb: I ate... plus probably doing a family dinner thing tonight |
| 17:28.24 | MrChimpy | lots of cash in it |
| 17:28.55 | russellb | MrChimpy: well, that's what our partners do |
| 17:29.25 | file | russellb knows the story, I don't... I'm new! |
| 17:29.44 | russellb | MrChimpy: so did you see the "exec" agi command? |
| 17:29.54 | MrChimpy | never noticed it rb. |
| 17:30.24 | myiagy | MrChimpy you're doing agi in php? |
| 17:30.34 | MrChimpy | pel |
| 17:30.37 | MrChimpy | perl |
| 17:30.39 | Druken | asterisk is easy, so many people make it complicated.... |
| 17:30.47 | myiagy | oh, can't help then :) |
| 17:30.57 | *** join/#asterisk ToTo (n=ToTo@host113-87.pool8256.interbusiness.it) |
| 17:31.01 | MrChimpy | agi in php. ick :) |
| 17:31.17 | MrChimpy | perl's bad enough :) |
| 17:31.37 | myiagy | i find it quite easy to do it in php, even easier to integrate with mysql for IVR or something |
| 17:32.17 | MrChimpy | sorry, it's my language snobbery |
| 17:32.38 | MrChimpy | i've gone from 100% C to reluctantly poking perl |
| 17:33.27 | Hmmhesays | can you still sign up for msn messenger with a non MS email address? |
| 17:36.08 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 17:36.31 | Druken | yes |
| 17:40.17 | Hmmhesays | yeah I just found the link |
| 17:42.57 | Druken | :) |
| 17:43.28 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:45.10 | *** join/#asterisk Splas (n=jwb@206.252.198.101) |
| 17:45.20 | msw | has anyone had any success/failure stories with spandsp and app_rxfax? I tried it out, and it seems to work pretty well. I'm looking for any reports of problems/instability/etc. |
| 17:45.44 | msw | (using a T1 PRI, fwiw) |
| 17:46.08 | *** join/#asterisk Gnat42 (n=gnat42@S010600119576abd3.gv.shawcable.net) |
| 17:46.43 | MrChimpy | cool. |
| 17:47.03 | eric_hill | msw: I set up spandsp and app_rxfax on a test server and sent several dozen pages from another (brooktrout-based) system with only 1 issue -- the faxes came through without header details. |
| 17:47.06 | MrChimpy | that got rid of the hacky passing numbers through channel vars... exec dial works nicely |
| 17:47.57 | msw | eric_hill: looks like I'm getting the header details from the sending fax without any trouble in my tests |
| 17:48.14 | }cytrak{ | is there any other free dial out network besides the FWD ? |
| 17:51.32 | vader-- | Do any of you guys know how to create a null byte character? |
| 17:53.19 | dlynes_ | vader--: Ctrl-VCtrl-@ |
| 17:53.46 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 17:53.46 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:54.26 | Hmmhesays | little pig, little pig let me in |
| 17:54.59 | vader-- | how about in windows dlynes? |
| 17:55.36 | [TK]D-Fender | Hmmhesays : Right... time to crank up the Green Jelly :D |
| 17:55.56 | Hmmhesays | yeah the jam band does a medley of that and "don't phunk with my heart" |
| 17:56.00 | Hmmhesays | it's crazy |
| 17:57.11 | [TK]D-Fender | *cringes* |
| 17:57.15 | *** part/#asterisk Gnat42 (n=gnat42@S010600119576abd3.gv.shawcable.net) |
| 17:57.24 | *** join/#asterisk mocker (n=ks@in.kansas.but.not.a.republi.cn) |
| 17:57.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:58.16 | mocker | I'm having a problem with the agents.conf createlink option. Even if I have it set to no, it still adds the comment to the userfield. |
| 17:59.07 | Hmmhesays | [TK]D-Fender its pretty funny |
| 17:59.24 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 18:00.52 | *** join/#asterisk stoffell_h (n=stoffell@81.83.249.224) |
| 18:03.20 | *** join/#asterisk websae (n=icechat5@h69-129-251-26.69-129.unk.tds.net) |
| 18:03.30 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
| 18:04.05 | japerry | heya, does anyone here have experience with a digium T1 card? |
| 18:04.20 | japerry | I keep getting a yellow alarm, we're using e&m wink |
| 18:04.39 | japerry | loopback works fine, but that is the only time I get a green light |
| 18:07.20 | coppice | its probably a problem with the wires from you to the telco |
| 18:07.40 | japerry | hmm, well I have a person from Verizon here using her test equip.. tests are showing okay |
| 18:07.49 | MikeJ[Laptop] | coppice, the wires from my telco are string.. is that good or bad? |
| 18:08.11 | japerry | but in the zttool all I see is us receieving the alarn, there is no transmit |
| 18:08.19 | mocker | japerry: Have you tried using the patlooptest to verify your card? |
| 18:08.26 | MikeJ[Laptop] | japerry, is it set up right? |
| 18:08.33 | japerry | yup. and we made a loopback plug |
| 18:08.35 | coppice | string is good, unless it gets overly tight when it rains |
| 18:08.37 | [TK]D-Fender | MikeJ[Laptop] : Depends.... did you get a pair matching resonance tin-cans rated for the kind of string you're using? :) |
| 18:08.39 | *** join/#asterisk inv_arp[work] (i=junya@adsl-10-153-159.mia.bellsouth.net) |
| 18:08.41 | mocker | japerry: I had the same type of symptoms and it turned out that it was a bad card. |
| 18:08.47 | japerry | hmm |
| 18:08.49 | japerry | span=1,1,0,esf,b8zs |
| 18:08.49 | japerry | e&m=1-4 |
| 18:08.49 | japerry | loadzone = us |
| 18:08.49 | japerry | defaultzone = us |
| 18:09.08 | japerry | mocker: did it past the looptest? |
| 18:09.15 | MikeJ[Laptop] | japerry, let me introduce you to pastebin... japerry, pastebin.. pastebin, japerry |
| 18:09.20 | MikeJ[Laptop] | ~pastebin |
| 18:09.23 | jbot | pastebin is probably 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 |
| 18:09.29 | japerry | oh thanks |
| 18:09.31 | mocker | japerry: It would go green w/ a loopback adapter. |
| 18:09.32 | japerry | sorry :-/ |
| 18:10.06 | coppice | a yellow alarm is the other end telling you it cannot see your signal |
| 18:10.06 | japerry | okay, damnit... |
| 18:10.34 | MikeJ[Laptop] | and the color of the traffic light that tells you that you need to speed up |
| 18:10.46 | japerry | coppice: yup. thats what we see.. so a misconfiguration would probably still result a in a green light if the card was good? |
| 18:10.50 | japerry | lol |
| 18:11.04 | MikeJ[Laptop] | japerry, no... |
| 18:11.28 | MikeJ[Laptop] | is this pri or strait T1? |
| 18:12.14 | japerry | its called a 'flexgrow' -- basically its a e&m_wink T1... verizon says its not a pri |
| 18:12.33 | japerry | but they control the dialtone.. which would indicate that it is... |
| 18:12.49 | MikeJ[Laptop] | it's not a pri |
| 18:13.17 | MikeJ[Laptop] | is it mixed voice and data? |
| 18:13.23 | websae | I can't believe it, there's a company here that will allow you 5 PRIs at like $270/each |
| 18:13.27 | japerry | nope, all voice -- 4 channels voice and the rest is emtpy |
| 18:13.31 | ManxPower | japerry, You do realize that a non-PRI T-1 is about as useful as a POTS line, right? |
| 18:13.50 | MikeJ[Laptop] | japerry, so the flex is that they can add channels at will |
| 18:13.56 | eric_hill | ManxPower, albeit with faster call setup/teardown that is... |
| 18:13.56 | coppice | what's wrong with RBS T1? |
| 18:14.09 | japerry | ManxPower: yup. Verizon said they can direct the 20did's to the 4 channels |
| 18:14.17 | japerry | and yes, MikeJ you can add channels at will |
| 18:14.31 | MikeJ[Laptop] | coppice, I dunno.. I hear some weird british guy in hong kong wrote drivers for that.. but I don't trust it |
| 18:15.00 | ManxPower | eric_hill, Huh? You still have to outpulse DTMF |
| 18:15.03 | *** join/#asterisk exonic (n=exonic@sig.triton.net) |
| 18:15.09 | exonic | What's that program used for jitter testing? |
| 18:15.20 | nahirean | linux enviornment or windows? |
| 18:15.28 | japerry | hmm and now Verizon is saying that we need to make it configured for 4digit DTMF .. is that different than e&m_wink? |
| 18:15.42 | eric_hill | Yes, but you can outpulse the dtmf much faster from my experience. The T1 is a clearer signal. |
| 18:15.53 | coppice | ManxPower: the call setup is still much faster, and you have full supervision, and DID |
| 18:16.07 | coppice | the audio is clean, too |
| 18:16.18 | eric_hill | 4 digit DTMF is *after* the call has been set up. e&m wink start is the signalling method. |
| 18:16.26 | ManxPower | coppice, how is call setup faster? |
| 18:16.49 | distortion | anyone know of a way to add ip ranges to sip.conf, or is there a way to add multiple ip's to the same host= line? |
| 18:16.51 | ManxPower | japerry, that basically means that your exten => lines only have the 4-digits, not the whole number. |
| 18:17.02 | coppice | outgoing isn't much different, but incoming is much faster |
| 18:17.09 | ManxPower | distortion, host= is only for OUTGOING |
| 18:17.29 | ManxPower | for incoming you want permit/deny |
| 18:17.50 | distortion | well, i have a wholesale customer with 10 cisco 5300's and i have it setup as "insecure=port,invite" so it auth's off of the host=ip entry |
| 18:18.00 | japerry | ManxPower, so basically the 4digit DTMF has nothing to do with the initial provision of the device (the e&m=1-4 part) |
| 18:18.18 | brettnem | wait, T1 is a clearer signal than what?! |
| 18:18.42 | coppice | analogue |
| 18:18.45 | eric_hill | Yes, a T1 is clearer than an analog line. |
| 18:19.57 | japerry | HAHHAHHA |
| 18:20.07 | japerry | verizon has the channels possibly provisioned for Data |
| 18:20.08 | japerry | HAHAHAHAAH |
| 18:20.13 | Hmmhesays | lovely |
| 18:20.19 | japerry | that'd break it, eh? |
| 18:20.33 | eric_hill | Minor detail. Tell them you don't speak binary... :) |
| 18:21.02 | [TK]D-Fender | There are only 10 kinds of people, those that know binary, and those that don't :) |
| 18:21.04 | *** join/#asterisk SparFux (n=player@e182018201.adsl.alicedsl.de) |
| 18:21.16 | japerry | heh, they're doing some more tests, but I bet the old setup here was half data and half voice.. w00t :-P |
| 18:21.39 | SparFux | There is MP3Player. But I'd like a more universal approach. Is there an app already, which will support an external application giving sound to stdout in a certain format? |
| 18:21.51 | *** join/#asterisk AK01 (n=none@nv-65-173-88-46.sta.sprint-hsd.net) |
| 18:21.59 | doolph | hi there |
| 18:22.10 | dlynes_ | SparFux: why not use the internal player? |
| 18:22.26 | SparFux | dlynes: You mean MP3Player? |
| 18:22.33 | dlynes_ | SparFux: internal, not external |
| 18:22.34 | AK01 | Is aseterisk a alternative for to a PRI? |
| 18:22.38 | dlynes_ | SparFux: mode=files |
| 18:22.58 | SparFux | dlynes: I don;'t get what you mean? |
| 18:22.59 | dlynes_ | AK01: asterisk will use a PRI |
| 18:23.18 | ManxPower | japerry, correct |
| 18:23.20 | dlynes_ | SparFux: mpg123 is an external player that asterisk used to use as the default player |
| 18:23.38 | dlynes_ | SparFux: with asterisk 1.2, asterisk now has an internal player, so no need for an external player |
| 18:23.50 | SparFux | dlynes: Ah, didn't know that. |
| 18:23.53 | dlynes_ | SparFux: produces less load on the system because there's no latency in spawning processes now |
| 18:23.55 | *** join/#asterisk g0tw00d (n=greenbea@24.246.200.205.diodecom.net) |
| 18:24.14 | dlynes_ | SparFux: and you can play gsm, pcm, mp3, ... |
| 18:24.30 | dlynes_ | SparFux: so you can create a file that best matches the codec coming in, so no need for transcoding |
| 18:25.07 | AK01 | dlynes_ car if I PM you? |
| 18:25.10 | SparFux | dlynes: it probably can't play *.m3u files and *.pls files I suppose. |
| 18:25.12 | dlynes_ | SparFux: see www.astrecipes.net for some examples |
| 18:25.27 | dlynes_ | AK01: no...please keep it in the channel |
| 18:25.33 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:25.43 | dlynes_ | AK01: I'm guessing you're going to ask a bunch of questiosn that have already been asked here |
| 18:25.51 | AK01 | I'm getting a new connection http://www.mpowercom.com/products/integrated/mpower_enterprise.shtml |
| 18:26.00 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 18:26.13 | g0tw00d | Hi, I'm an asterisk's novice.....I just bought a Digium TDM11b. If you would like to make an attemped at helping me plz message me. Thanks |
| 18:26.38 | ManxPower | ~thebook |
| 18:26.39 | jbot | it has been said that thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 18:26.39 | dlynes_ | AK01: go on |
| 18:26.42 | g0tw00d | I can setup asterisk and make it work with cheap FXO cards |
| 18:26.43 | AK01 | and a new cisco system with some switch, phones, and other misc. here is there router http://www.cisco.com/en/US/products/ps5882/index.html |
| 18:26.56 | SparFux | dlynes: what's the internal player called? |
| 18:26.59 | AK01 | Where does Asterisk fit into my setup? |
| 18:27.04 | dlynes_ | SparFux: mode=files |
| 18:27.09 | sevard | Can somebody explain something to me... I'm not really grasping something here |
| 18:27.12 | SparFux | dlynes: aha. |
| 18:27.15 | SparFux | thx! |
| 18:27.28 | dlynes_ | SparFux: yeah...no need to specify the application= line |
| 18:27.30 | sevard | There's a phone somewhere on my box that's streaming music with cmd ControlPlayback, which is fine |
| 18:27.43 | sevard | but I don't think that phone is online, infact sip show peer 1000 (that phone) says it's unregistered |
| 18:28.00 | dlynes_ | AK01: I don't know what the cisco system is |
| 18:28.03 | sevard | I'm not sure where that soft client is in this building but i'm almost positive it's not online |
| 18:28.03 | eric_hill | AK01: Asterisk = call management & voicemail. Cisco equipment does the on-ramp to the telco and handsets. Pretty trivial. |
| 18:28.09 | [TK]D-Fender | AK01 : You should simply forget the Cisco stuff altogether and just Asterisk as your PBX. |
| 18:28.14 | sevard | how could asterisk be streaming audio to a phone that's not registered? |
| 18:28.33 | eric_hill | AK01: Also, DO NOT USE CISCO'S CALL MANAGER EXPRESS. Just don't. Please. Seriously. |
| 18:28.57 | doolph | heh |
| 18:28.59 | doolph | y not |
| 18:29.30 | dlynes_ | AK01: Looks like you've got a whole bunch of people able to help you...I know nothing about cisco |
| 18:29.52 | eric_hill | I'd have to use Pastebin and ROT13 my answers so I didn't offend anyone. I have a TAC case open right now that has been open for two weeks and they still don't know why VM is ... um... odd. |
| 18:29.53 | sevard | can anyone explain that? |
| 18:30.25 | eric_hill | sevard: Why does it need to be registered? You can SIP/call and IP address, right? |
| 18:30.28 | dlynes_ | sevard: Don't need to register it to talk to it |
| 18:30.29 | SparFux | dlynes: you mean music on hold? |
| 18:30.43 | dlynes_ | dlynes: correct...usually /etc/asterisk/musiconhold.conf |
| 18:30.52 | dlynes_ | erm |
| 18:31.08 | dlynes_ | lol |
| 18:31.08 | SparFux | *LOL* |
| 18:31.10 | dlynes_ | SparFux: there's plenty of examples in there on how to configure it, including the files= mode |
| 18:31.24 | g0tw00d | With a TDM11b and a fresh load of asterisk, is the card suppose to be detected and installed automatically like cheap FXO's? Everytime I restart asterisk tells me I have removed the card and need to reconfigure. Can anyone help me? |
| 18:31.28 | sevard | dlynes_: I just found the computer with the client on it. It was off. I turned it on, the phone registered yet asterisk is still streaming audio to a non-existant client |
| 18:31.30 | SparFux | dlynes: Ill check, but I am unsure, if this is what I wan.t |
| 18:31.52 | sevard | dlynes_: is this a fairly present problem with large scale asterisk systems? |
| 18:32.05 | dlynes_ | sevard: I wouldn't know...I don't have any large scale asterisk systems |
| 18:32.23 | dlynes_ | sevard: closest I've got is about 40 or 45 remote clients |
| 18:32.43 | sevard | dlynes_: I have about 10 but haven't seen this problem for a while until now |
| 18:33.02 | dlynes_ | sevard: if some of them aren't getting connectivity, it's usually because of a defective sipura unit |
| 18:33.31 | sevard | dlynes_:I've seen it with good and fully working Aastra phones, sipura 2002s and eyebeam clients |
| 18:33.38 | sevard | so I'm not sure if it's client specific. |
| 18:34.08 | sevard | See, in a smaller server setup like this it's fairly easy to see if something has gone haywire, but in a large setup it'd be near impossible |
| 18:34.33 | sevard | and I don't even know how to destroy that open channel ;;/ |
| 18:34.39 | SparFux | dlynes: it's not what I want. Because I cannot use my Microphone as a sound source, or my tv-card and I cannot play *.m3u files. I cannot let asterisk download URLs from a homepage and play the radio stations from the net. :-( |
| 18:34.40 | *** join/#asterisk jake1932 (n=Administ@68.236.22.143) |
| 18:34.53 | g0tw00d | can someone help me? |
| 18:34.56 | *** join/#asterisk apr (n=apr@r200-40-234-53-dialup.adsl.anteldata.net.uy) |
| 18:35.00 | dlynes_ | sevard: Just look for Status: UNKNOWN, or Status: UNREACHABLE |
| 18:35.20 | dlynes_ | g0tw00d: you probably don't have the driver loading automatically at bootup |
| 18:35.30 | sevard | dlynes_: it is status UNKNOWN, how do I go about destroying that channel without rebooting *? |
| 18:35.49 | dlynes_ | SparFux: try shoutcast, or some other streaming media server |
| 18:36.03 | g0tw00d | should there be a problem with it if i just reloaded asterisk with the card in? |
| 18:36.08 | dlynes_ | sevard: if it's unknown, it's not being access |
| 18:36.09 | dlynes_ | sevard: if it's unknown, it's not being accessed |
| 18:36.31 | dlynes_ | g0tw00d: you probably don't have the driver loading automatically at bootup |
| 18:36.32 | sevard | dlynes_: sip show channels shows an active channel and I can see via cli output that there is activity |
| 18:36.49 | [TK]D-Fender | g0tw00d : Pastebin your zaptel & zapata.conf |
| 18:36.50 | [TK]D-Fender | ~pb |
| 18:36.52 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 18:37.00 | sevard | 192.168.1.2 1000 f63b525fe01 00101/00002 ulaw No Rx: ACK |
| 18:37.01 | g0tw00d | ok |
| 18:37.03 | g0tw00d | one sec |
| 18:37.19 | dlynes_ | goddamn, asterisk is annoying sometimes |
| 18:37.56 | dlynes_ | Why is it that set verbose 0 doesn't get rid of the ever present, ever annoying: Apr 28 11:29:31 NOTICE[30897]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end message? |
| 18:38.27 | sevard | soft hangup SIP/1000-c603 |
| 18:38.52 | Hmmhesays | hmm does addqueuemember not have priority jumping anymroe? |
| 18:38.55 | dlynes_ | sevard: and that machine's not online? |
| 18:39.12 | jake1932 | can someone from digium help me with g729 licenses - i bought some a while back and I lost them. |
| 18:39.21 | sevard | dlynes_: it wasn't, i booted it, the phone registered, and * was still streaming although the client wasn't getting anything. I destroyed the channel. |
| 18:39.43 | dlynes_ | jake1932: try their 1-888 number? |
| 18:39.59 | jake1932 | that would be my next step :) |
| 18:40.09 | dlynes_ | jake1932: that's the fastest step |
| 18:40.23 | g0tw00d | ok pasted |
| 18:40.41 | [TK]D-Fender | g0tw00d : links please... |
| 18:40.58 | dlynes_ | sevard: Yeah, no idea what it would be...never ran into that problem myself |
| 18:41.07 | dlynes_ | sevard: btw...are you running the latest firmware for the Aastras? |
| 18:41.15 | Splas | Hrm, asterisk 1.2.4 just up and died on me after spewing a ton of Apr 28 14:42:37 WARNING[19595] frame.c: Unable to calculate sample length for format unknown |
| 18:41.19 | g0tw00d | http://pastebin.com/687437 |
| 18:41.32 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 18:41.53 | dlynes_ | Splas: probably passing it a file that has a higher sample rate than 8K |
| 18:42.17 | dlynes_ | Splas: it's for music on hold, right? |
| 18:42.22 | *** join/#asterisk blkcamarozr28 (n=wchan@cpe-72-134-204-179.hawaii.res.rr.com) |
| 18:42.27 | Splas | dlynes: could be, the error wasn't specific... |
| 18:42.38 | Splas | dlynes: i'm using mpg123 tho, not native |
| 18:42.51 | dlynes_ | Splas: yeah...mpg123 is very picky about the formats |
| 18:43.01 | [TK]D-Fender | g0tw00d : Running AMP... please read the channel topic... |
| 18:43.08 | Splas | dlynes: odd, there's only 1 file... |
| 18:43.13 | Hmmhesays | anyone using addqueuemember? |
| 18:43.14 | Splas | dlynes: and its working now... |
| 18:43.21 | dlynes_ | Splas: try to use a different mp3 file, and see if you still encounter that problem |
| 18:43.47 | g0tw00d | sorry to bother you thx for your time |
| 18:43.50 | [TK]D-Fender | g0tw00d : An you also didn't provide all the required parts for zapata due to linking. |
| 18:44.09 | g0tw00d | huh |
| 18:44.12 | ManxPower | g0tw00d, so try the channel listed in the /topic |
| 18:44.14 | g0tw00d | my zaptel is there |
| 18:44.19 | Splas | dlynes: ok, I killed the additional context |
| 18:44.20 | g0tw00d | i will thx |
| 18:44.46 | ManxPower | g0tw00d, AMP is SO customized basically anything we tell you will be wrong for your install |
| 18:45.37 | sevard | dlynes_: yes |
| 18:46.40 | *** join/#asterisk natmlt (n=natmlt@ip24-251-50-106.ph.ph.cox.net) |
| 18:47.09 | dlynes_ | sevard: just asking because i've heard of people running into all kinds of problems using firmware before 1.3.1 |
| 18:47.13 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 18:48.04 | *** part/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 18:48.12 | dlynes_ | Splas: i didn't mention anything about a context |
| 18:48.23 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 18:48.40 | stack_ | afternoon, everyone |
| 18:49.45 | Splas | dlynes: The context with my additional music on hold. the music on hold context... erm... 'class' |
| 18:50.01 | Splas | dlynes: it happened again so now its JUST the default as-shipped moh |
| 18:50.07 | stack_ | We are getting a PRI in within the next couple of days. Once it is in, we need to extend it from the DMARK to our telco room and then to our server and into a TE110P... what kind of wiring do I need to do for this? |
| 18:50.10 | dlynes_ | Splas: again...I didn't mention anything about a context |
| 18:50.25 | dlynes_ | Splas: I said try changing your mp3 files |
| 18:50.47 | dlynes_ | Splas: as in the actual .mp3 files, not the musiconhold.conf file |
| 18:50.51 | Splas | dlynes: In musiconhold.conf you can setup multiple different music on hold classes (what I referred to as a context). I removed the ones I created. |
| 18:51.04 | dlynes_ | dlynes: yeah, i'm fully aware of that |
| 18:51.06 | Splas | dlynes: hence the mp3 files I added are no longer being included. |
| 18:51.21 | dlynes_ | Splas: ah...so just using the ones that come with asterisk now? |
| 18:51.24 | Splas | yes |
| 18:51.31 | Splas | since my other "context/class" had only that one file |
| 18:51.31 | dlynes_ | and you're still getting the problem? |
| 18:51.36 | Splas | which played fine when testing... |
| 18:51.39 | Splas | we'll see if I still get it now |
| 18:51.45 | Splas | yes |
| 18:51.47 | Splas | its happening right now |
| 18:51.53 | dlynes_ | and which version of mpg123 are you using? |
| 18:53.26 | doolph | for what |
| 18:53.41 | dlynes_ | Splas: ? |
| 18:54.17 | dlynes_ | Splas: try mpg123 --version, and if that doesn't work, try mpg123 --longhelp |
| 18:54.31 | Splas | dlynes: this was FINE for weeks |
| 18:54.37 | Splas | then I started tryinng to add new MOH |
| 18:54.39 | Splas | and blamo |
| 18:54.43 | Splas | it seems moh reload |
| 18:54.47 | Splas | wasn't killing the old mpg123s |
| 18:54.57 | dlynes_ | Splas: oh...didn't know you were using that command |
| 18:55.06 | dlynes_ | Splas: moh reload wasn't terribly stable in asterisk 1.2.4 |
| 18:55.16 | [TK]D-Fender | stack_ : Typically a stright cat5 patch. nothing special. |
| 18:55.19 | Splas | go figure.. for a stable release :) |
| 18:55.29 | dlynes_ | Splas: try upgrading to 1.2.7.1 |
| 18:55.33 | tzanger | heh |
| 18:55.34 | tzanger | Anyways, back on topic. The day I see a mobile HDD survive an 3-story fall onto solid concrete, then I'll be impressed. And yes, I've seen it. Except for the surviving part. |
| 18:55.35 | dlynes_ | Splas: You'll find it a lot more stable |
| 18:55.55 | dlynes_ | Splas: and 1.2.7.1 fixes some sip subscription problems, too |
| 18:56.25 | dlynes_ | Splas: heh...a lot of times, trunk is more stable than the releases :) |
| 18:56.38 | Splas | yes, I'll do that tonight |
| 18:56.45 | Splas | been trying to hold off till the end of the week/off hours |
| 19:00.08 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
| 19:00.16 | Splas | dlynes: and this whole 'trunk being more stable than release' shit shouldn't fly anymore... |
| 19:00.33 | skydog | i remember somone on here had experience with installing orca for stats on * boxes. Which/where did the dependency packages are needed? |
| 19:00.39 | stack_ | [TK]D-Fender: so if I just punch down the 4 used wires for ethernet, it will work? |
| 19:00.47 | Splas | Of course nothing better than asterisk currently exists in the OSS realm |
| 19:00.52 | Splas | stable or not.. |
| 19:00.54 | dlynes_ | Splas: yeah, no kidding |
| 19:01.15 | Splas | its just so upsetting sometimes. |
| 19:01.20 | dlynes_ | Splas: I'm actually having to run libpri-trunk and zaptel-trunk on one of my production machines just to get it to have more than 24 hour uptime |
| 19:01.51 | Splas | thankfully I'm not using any actual zaptel hardware |
| 19:01.55 | Splas | its all SIP to an AS5330 |
| 19:02.09 | [TK]D-Fender | stack_ : pretty much. |
| 19:02.36 | stack_ | [TK]D-Fender:k, we are gettings our PRI any day now and that was pretty much the last thing I was worrying about |
| 19:02.38 | [TK]D-Fender | stack_ : You only really need 2 pairs, but buy yourself some sanity... |
| 19:02.57 | blitzrage | anyone here do time tracking of their hours? |
| 19:03.13 | blitzrage | looking for a nice simple program to track time on projects and daily work |
| 19:03.16 | dlynes_ | blitzrage: you're wanting a program to manage it for consulting? |
| 19:03.17 | [TK]D-Fender | blitzrage : I come up with ad-hoc numbers for billing :) |
| 19:03.21 | Hmmhesays | excel |
| 19:03.24 | Hmmhesays | notepad |
| 19:03.26 | [TK]D-Fender | blitzrage : MS Project? |
| 19:03.27 | Hmmhesays | vi |
| 19:03.38 | dlynes_ | blitzrage: Try timeslips (runs under msdos, macintosh, and maybe windows now) |
| 19:03.50 | dlynes_ | blitzrage: it's quite cheap, too |
| 19:04.17 | dlynes_ | blitzrage: it'll also integrate with simply accounting and a couple of other accounting packages |
| 19:04.24 | blitzrage | dlynes_: yah -- I just want somethign REAL simple though to track how many hours I actually work a week -- I think I'm working more than I charge for |
| 19:04.43 | blitzrage | I don't need anything that complicated |
| 19:05.06 | blitzrage | something that sits in the toolbar that I can click on to start, then when I click to stop, it asks for a log entry, and thats it |
| 19:05.17 | blitzrage | if its complicated, I won't use it -- or won't have time to use it :( |
| 19:05.19 | dlynes_ | blitzrage: do you already use quickbooks? |
| 19:05.48 | blitzrage | yah -- but thats a huge app to have open just to track time... |
| 19:06.00 | denon | he's talking about the timer app |
| 19:06.01 | dlynes_ | blitzrage: it's got a button you can click on it, to start a time |
| 19:06.01 | denon | its different than qbooks |
| 19:06.02 | [TK]D-Fender | blitzrage : Maybe just some dumb spreadsheet? |
| 19:06.03 | dlynes_ | blitzrage: it's got a button you can click on it, to start a timer |
| 19:06.25 | dlynes_ | blitzrage: it's for time tracking for projects in quickbooks |
| 19:06.42 | blitzrage | dlynes_: right -- but then I have a pretty heavy program open... |
| 19:06.57 | dlynes_ | blitzrage: i think it's only available in the accountant/contractor's edition though |
| 19:07.00 | blitzrage | problem is I can picture exactly how something would work -- just can't program good enough :) |
| 19:07.03 | denon | that's like .. cross-posting |
| 19:07.07 | blitzrage | denon: *gasp* |
| 19:07.22 | blitzrage | but different people in different channels :) |
| 19:07.22 | denon | blitzrage: the timer app isnt qbooks, its a tiny app, fits on a floppy or somethin |
| 19:07.33 | dlynes_ | blitzrage: so you don't leave quickbooks open all the time? |
| 19:07.37 | denon | blitzrage: ya, but I had the same answer for you in the other chan, and you didnt even see it |
| 19:07.45 | Hmmhesays | god i hate it when customers have everything on a local network "i can't let you in, just tell me what to configure" |
| 19:07.50 | blitzrage | dlynes_: nope -- I just work from home, so there is very little use in having it open all the time |
| 19:08.11 | dlynes_ | ah...yeah...i always have it open to look up customer contact info and vendor contact info |
| 19:08.25 | natmlt | There is an new little app for Mac called iRatchet that might track time like you want |
| 19:08.30 | denon | the point is .. the QB Timer App has nothing to do with quickbooks |
| 19:08.41 | denon | you just export the timer files to Quickbooks later |
| 19:08.44 | blitzrage | natmlt: mac? ewww ;) |
| 19:08.45 | denon | and import the customer lists |
| 19:08.49 | dlynes_ | denon: could be...i've only used it inside of quickbooks, myself |
| 19:08.53 | blitzrage | denon: ok -- I didn't know it was a separate app |
| 19:08.58 | denon | blitzrage: you would if you read :) |
| 19:09.07 | dlynes_ | yeah...no kidding |
| 19:09.09 | natmlt | Mac Rocks :P |
| 19:09.09 | blitzrage | denon: I did read it |
| 19:09.15 | dlynes_ | denon's only mentioned it like 3 or 4 times now |
| 19:09.27 | blitzrage | ok, I get it, I'm a really bad person |
| 19:09.30 | denon | heheh |
| 19:09.40 | denon | I'm good for overall morale :) |
| 19:09.45 | blitzrage | at least I know denon and respect him :) |
| 19:09.53 | denon | but yes, it like fits on a floppy or somethin |
| 19:09.57 | denon | comes with QB Pro and up |
| 19:10.10 | denon | its really cheezy, but doesnt really need to be fancy for most uses |
| 19:10.12 | blitzrage | denon: coolio -- I got that app, didn't know it was able to run as a separate app -- that might work |
| 19:10.18 | blitzrage | yah exactly |
| 19:10.20 | syzygybsd | How can i set the extension in the dialplan? Set(EXTEN=s) doesnt' work |
| 19:10.28 | denon | we dont use it here, so your results may vary |
| 19:11.21 | denon | blitzrage: the thing I really like about it, though .. is its actually a separate app .. you dont even need quickbooks open .. pretty cool |
| 19:11.59 | blitzrage | denon: yah thats perfect, I didn't know I coudl run it separate from QB since I don't leave it open all the time |
| 19:12.04 | denon | haha |
| 19:12.09 | blitzrage | :D |
| 19:12.17 | denon | that just flew right over |
| 19:12.21 | blitzrage | I'll keep re-answering your comments all day long if you want :) |
| 19:12.31 | denon | but yeah, its separate because obviously your workers dont have qb installed |
| 19:12.32 | denon | or shouldnt |
| 19:12.37 | natmlt | Ok, actual Asterisk question now... |
| 19:12.41 | blitzrage | well I'm the only worker :) |
| 19:12.41 | natmlt | I found a line in my context exten => a,1,Goto(Main,s,1), I have not seen "a" used anywhere. Have any ideas? |
| 19:12.42 | denon | a who? |
| 19:12.56 | blitzrage | natmlt: you must be new here -- we don't talk about asterisk in here |
| 19:13.12 | blitzrage | natmlt: its used by app_voicemail |
| 19:13.14 | denon | a is just a name |
| 19:13.17 | denon | oh, is it? |
| 19:13.18 | natmlt | But my trainer from digium told me to come here...lol |
| 19:13.20 | denon | whoops |
| 19:13.28 | blitzrage | denon: yah -- when you hit 0 or * or something in the login I think |
| 19:13.34 | denon | huh |
| 19:13.41 | denon | makes sense |
| 19:13.54 | msw | is there any built-in function for doing shell escaping on a string? |
| 19:14.15 | natmlt | I think I see why it is there now...kind of what I thought. |
| 19:14.19 | russellb | natmlt: they probably meant #asterisk, not #asterisk-dev |
| 19:14.28 | denon | he's not in -dev |
| 19:14.45 | russellb | well look at that. |
| 19:14.46 | dlynes_ | russellb: you've forgotten where you are |
| 19:14.46 | natmlt | Thought I was in #asterisk |
| 19:14.49 | russellb | lol! |
| 19:14.52 | russellb | natmlt: you are |
| 19:14.54 | russellb | i'm just being stupid |
| 19:14.57 | denon | good grief, you guys need Windows worse than anyone I know |
| 19:14.58 | Hmmhesays | anyone know of a decent program so I can access a serial port via tcp? |
| 19:14.59 | natmlt | Oh and actually if you can amaging...I am not a he |
| 19:15.11 | natmlt | ;) |
| 19:15.22 | Splas | dlynes: so far so good... must've been my use of moh reload |
| 19:15.22 | natmlt | oops...imagine |
| 19:15.38 | denon | natmlt: ruh roh, now you've done it .. you're going to get like 10 privmsgs, and attention from every guy in this channel, whether they know what they're doing or not |
| 19:15.41 | denon | welcome to irc |
| 19:15.42 | dlynes_ | Hmmhesays: slip? ppp? |
| 19:15.49 | natmlt | LOL |
| 19:16.19 | *** join/#asterisk lionelp (n=lionel@ip-128.net-82-216-65.rev.numericable.fr) |
| 19:16.56 | nDuff | natmlt: "a" indicates that the user pressed *. |
| 19:16.59 | *** join/#asterisk mikefoo (n=mikefoo@64.124.169.2) |
| 19:17.02 | nDuff | natmlt: see http://www.voip-info.org/wiki/index.php?page=Asterisk+standard+extensions |
| 19:17.15 | mikefoo | Anyone do a test call into me real quick from outside of the u.s.? |
| 19:17.15 | nDuff | (err, wait, that was already answered). |
| 19:17.37 | natmlt | Doh, thanks, I search there but obviously not very well. |
| 19:17.48 | *** join/#asterisk tomcontr3 (n=gcontrer@68-78-246-201.adsl.terra.cl) |
| 19:18.03 | tomcontr3 | is there any way of killing a call from the command line? |
| 19:18.07 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 19:18.23 | blitzrage | soft hangup |
| 19:18.33 | dlynes_ | tomcontr3: using a cli command: asterisk -rx "soft hangup Zap/1-1" |
| 19:18.36 | syzygybsd | could someone enlighten me as to why GotoIf("SIP/home-b8de", "[4 = 4]?1001:12") is going to 12 instead of 1001? |
| 19:18.56 | blitzrage | syzygybsd: you're missing a $ in front of the [ ] |
| 19:18.59 | tomcontr3 | and how could I do this? End the call, and play a recording? |
| 19:19.07 | syzygybsd | thanks a ton blitzrage |
| 19:19.29 | jsharp | Once you hang up a call, the call is gone. |
| 19:19.31 | dlynes_ | tomcontr3: you mean play a recording and end a call? you can't play a recording to a terminated caller |
| 19:19.32 | blitzrage | I just answered like 3 asterisk questions in a row -- I think that exempts me from ridicule for my question earlier :) |
| 19:20.02 | blitzrage | although this is #asterisk, and am a realist, so I'm sure it doesn't |
| 19:20.04 | tomcontr3 | what I want to do, is a little program, that ends the call, because he run out of mony, |
| 19:20.11 | tomcontr3 | and then play the recording |
| 19:20.15 | doolph | mikefoo? |
| 19:20.20 | dlynes_ | tomcontr3: Check out the documentation on the Dial() command |
| 19:20.31 | tomcontr3 | ok |
| 19:20.43 | mikefoo | ? |
| 19:20.59 | mikefoo | need someone to give me a test call from outside of the US |
| 19:21.02 | dlynes_ | tomcontr3: There's a number of modifiers for it that allow you to let the caller know how much time they've got left, ... |
| 19:21.07 | doolph | whats the number |
| 19:21.10 | brettnem | hey, so anyone have any recommendation on who to go to now that nufone is hosed? :) |
| 19:21.20 | dlynes_ | brettnem: everyone else |
| 19:21.36 | doolph | what happen with nufone |
| 19:21.38 | brettnem | everyone? that's a lot of providers to sign up with |
| 19:21.39 | mikefoo | doolph: thanks.. |
| 19:21.48 | mikefoo | doolph: I sent it, in pm |
| 19:21.51 | brettnem | nufone's getting the Voip axe |
| 19:22.03 | brettnem | thank your politicans |
| 19:22.07 | *** part/#asterisk g0tw00d (n=greenbea@24.246.200.205.diodecom.net) |
| 19:22.10 | brettnem | just remember, your next |
| 19:22.13 | mikefoo | brettnem: yeah? whats the scoop? |
| 19:22.28 | brettnem | oh I don't know their scoop.. but I know what's going on all over.. so I can guess |
| 19:22.51 | brettnem | anyway, anyone have an actual recommendataion.. I don't like playing darts with ITSPs without a recommendation |
| 19:23.12 | mikefoo | doolph: thanks, thats was my co-worker |
| 19:23.21 | doolph | cool |
| 19:23.30 | doolph | what are you testing |
| 19:23.38 | mikefoo | level3 |
| 19:23.43 | doolph | whats that |
| 19:23.47 | C4T3l | could anyone recommend a voip company that i could connect my * to - to call out on the PTSN? |
| 19:23.50 | ManxPower | brettnem, ALL ITSPS SUCK. Teliax seems to suck less than most. |
| 19:24.06 | mikefoo | doolph: where was call originated from? |
| 19:24.24 | doolph | umm its voip |
| 19:24.25 | ManxPower | as for a secondary provider, I don't know. |
| 19:24.35 | mikefoo | yeah, where are you located? |
| 19:24.47 | brettnem | ManxPower: thanks, that's what I was looking for.. and yes.. they suck.. agreed |
| 19:24.58 | doolph | panama |
| 19:25.01 | brettnem | ManxPower: btw, all ITSPs will get shut down like nufone.. just a matter of time. |
| 19:25.01 | mikefoo | what number are you sending out rather? |
| 19:25.04 | mikefoo | a US number? |
| 19:25.29 | doolph | nop |
| 19:25.41 | doolph | its connected via unknown provider |
| 19:25.53 | natmlt | If you want a small but very reliable ITSP we use Starnetworks |
| 19:25.57 | mikefoo | yeah, it was originated in US. |
| 19:26.01 | mikefoo | which is no good to me, heh |
| 19:26.26 | rajiv | looking at the sangoma remora a200. how do i know if i need it with or without echo cancellation? |
| 19:26.29 | doolph | well, every call will be originated in US |
| 19:27.14 | MikeJ[Laptop] | there are several issues with ITSP's 1) impossible to make good money on minutes w/o huge volume, and 2) the model of delivering voice over public internet will never be as reliable as pstn |
| 19:28.00 | brettnem | MikeJ[Laptop]: actually, the problem is more fundamental and political than that.. |
| 19:28.06 | MikeJ[Laptop] | that being said, I don't beleive all ITSP's will get shutdown |
| 19:28.10 | brettnem | and the politics will catch up to it before the technical issues will |
| 19:28.32 | MikeJ[Laptop] | market fources I think will still drive this in the end |
| 19:28.45 | MikeJ[Laptop] | which means a race to 0 for the minutes business |
| 19:28.47 | mocker | What are some recommended applications for doing screen pops when calls come in? |
| 19:28.53 | ManxPower | I don't actually do any voice over internet anymore |
| 19:28.57 | brettnem | nah, the politicans are in bed with the incumbants and they don't want it to happen.. really |
| 19:29.05 | brettnem | voice over internet has nothing to do with it.. |
| 19:29.12 | brettnem | in fact.. voip almost has nothing to do with it.. |
| 19:29.23 | brettnem | in a few years, ITSPs won't be able to find a CLEC willing to sell them service |
| 19:29.44 | brettnem | or at least, not for less than a couple pennies a minute (in or outbound) |
| 19:29.56 | brettnem | it'll be a real shock to the industry.. but mark my words.. |
| 19:30.02 | MikeJ[Laptop] | well.. an ITSP will need to be a CLEC to pull it off soon enough |
| 19:30.02 | ManxPower | brettnem, kind of like what happened with the DSL market 8-) |
| 19:30.22 | brettnem | MikeJ[Laptop]: even then, you can't do it because of politics.. I'm a CLEC and an ITSP |
| 19:30.40 | coppice | seems the US government is currently voting to kill these services |
| 19:30.49 | brettnem | coppice: yes, in general, they are |
| 19:30.51 | MikeJ[Laptop] | as a CLEC you do have the ability to force interconnect to a better extent |
| 19:31.00 | brettnem | MikeJ[Laptop]: not really |
| 19:31.02 | MikeJ[Laptop] | and I know who you are... :P |
| 19:31.13 | brettnem | MikeJ[Laptop]: we've spoken on the phone.. ;) |
| 19:31.17 | dlynes_ | Does anyone know of any two port ATAs that you can do two simultaneous g729 channels? |
| 19:31.18 | MikeJ[Laptop] | yes... |
| 19:31.28 | dlynes_ | s/you// |
| 19:31.29 | coppice | i wasn't thinking in general. i was thinking of the vote not to require the internet to be unsegmented |
| 19:31.46 | brettnem | anyway.. CLECs typically terminate voip traffic as "local" which the incumbants do NOT allow |
| 19:31.48 | MikeJ[Laptop] | I thought CLEC interconnect was somewhat regulated... it was the last mile charges that the clec's just got screwed on |
| 19:31.56 | [TK]D-Fender | brettnem : Mediatrix 2102 |
| 19:32.13 | dlynes_ | [TK]D-Fender: was that intended for me? |
| 19:32.28 | [TK]D-Fender | dlynes : Yeah... bad aim :) |
| 19:32.31 | brettnem | MikeJ[Laptop]: it's all regulated.. yes.. and the incumbants will treat a VoIP provider like a LD company and make them pay intercarrier access fees (like an LD carrier) to terminate VoIP calls |
| 19:32.32 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 19:32.54 | ManxPower | *shrug* Our CLEC doesn't bill us for any calls within the state of MS or LA. |
| 19:32.58 | dlynes_ | [TK]D-Fender: ah...any others? I think i seem to recall the Mediatrix is quite expensive? |
| 19:33.11 | brettnem | ManxPower: because they are dealing with it now.. all CLECs go through this.. |
| 19:33.13 | MikeJ[Laptop] | brettnem, that makes sense |
| 19:33.29 | coppice | i wonder how much the telco charge each other between countries these days. I can call anywhere in the US for about 1 cent per minute, just by picking up my phone and dialing the normal IDD code |
| 19:33.43 | MikeJ[Laptop] | at the same time.. there are lots of ways to route calls... |
| 19:33.55 | brettnem | MikeJ[Laptop]: right now, many CLECs terminat ethe calls as local and dispute the charges, that can last for years, but it pops eventually and the CLEC says "I don't want to deal with the VoIP companies anymore." |
| 19:34.01 | asterboy | Polycom IP500 gets a call transfer, but you can't pick it up...what gives? |
| 19:34.30 | asterboy | Anyone else have this bug with a Polycom phone? |
| 19:34.34 | brettnem | MikeJ[Laptop]: All the same, when the incumbant see a call with a CID that is not local terminate on a local trunk, it spawns an investigation |
| 19:34.45 | Splas | It'd really be nice if this had been implemented already http://bugs.digium.com/view.php?id=1082 |
| 19:34.47 | asterboy | You can REJECT the call but not Answer |
| 19:35.11 | Splas | since its been open for more than a year... I should brush up my C so I could actually just fix it up myself rather than complaining :) |
| 19:35.23 | ManxPower | asterboy, no, we have not seen that. |
| 19:35.32 | asterboy | ok, thanks for the response |
| 19:35.34 | brettnem | ok, enough soapboxing.. sorry.. back to work |
| 19:35.41 | asterboy | I'll try another phone. |
| 19:39.42 | *** join/#asterisk tomcontr3 (n=gcontrer@68-78-246-201.adsl.terra.cl) |
| 19:39.52 | tomcontr3 | does any one know this company |
| 19:39.56 | tomcontr3 | www.x100p.com |
| 19:39.59 | tomcontr3 | ? |
| 19:40.09 | *** join/#asterisk critch (i=critch@steven.basesys.com) |
| 19:40.13 | dlynes_ | yeah |
| 19:40.20 | dlynes_ | I've got two of their cards |
| 19:40.22 | dlynes_ | Quite flaky |
| 19:40.24 | MikeJ[Laptop] | tomcontr3, there are tons of people making x100p clones |
| 19:40.30 | critch | anyone know what kind of switch TDS uses in the Chicago area? |
| 19:40.32 | MikeJ[Laptop] | the cheap ones will be flakey |
| 19:40.53 | dlynes_ | MikeJ[Laptop]: the x100p.com ones are the more expensive ones |
| 19:41.04 | tomcontr3 | sorry to ask but what does flaky means? |
| 19:41.06 | dlynes_ | MikeJ[Laptop]: that's why i bought them, figuring they'd be better |
| 19:41.08 | MikeJ[Laptop] | how much are they? |
| 19:41.20 | coppice | a sensible price for an x100p is only a few dollars. how much cheaper can they get? |
| 19:41.22 | dlynes_ | MikeJ[Laptop]: $25USD, if I remember correctly |
| 19:41.37 | dlynes_ | and then they get shipped from hk, to boot |
| 19:41.42 | coppice | $6-7 is more reasonable |
| 19:41.51 | MikeJ[Laptop] | coppice, in practice every one I see that cheap has had caller id issues |
| 19:42.00 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 19:42.09 | dlynes_ | MikeJ[Laptop]: That's the biggest issue i had with the x100p.com cards |
| 19:42.10 | MikeJ[Laptop] | I do not know the technical reasons behind it |
| 19:42.33 | dlynes_ | MikeJ[Laptop]: But I've had other issues lately, too...mostly drivers not loading |
| 19:42.39 | coppice | that's pretty bad if a modem card can't even do an elementary FSK modem :-) |
| 19:42.46 | *** join/#asterisk wolfson (n=wolfson@65.167.193.253) |
| 19:42.48 | tomcontr3 | the x100p.com says that the had fix the caller id issue |
| 19:42.53 | coppice | however, the caller ID modem in * is pretty crappy |
| 19:43.10 | dlynes_ | tomcontr3: that's bs, tomcontr3...I just bought the card less than two weeks ago |
| 19:43.31 | tomcontr3 | hmmm |
| 19:43.52 | tomcontr3 | and what is the caller id issue? it does not appear? |
| 19:43.55 | dlynes_ | tomcontr3: i've heard from others on here though, that this one out of norway is pretty good |
| 19:44.03 | dlynes_ | tomcontr3: sometimes it gets caller id, sometimes it doesn't |
| 19:44.07 | jake1932 | i'm calling dial and getting and error (Had to drop call because I couldn't make X compatible with Y). I expect this error for now, but asterisk won't continue down to the next priority (a different provider). trying to do failover. |
| 19:44.10 | dlynes_ | tomcontr3: about one in seven tries it gets it |
| 19:44.13 | guigouz | if I have an account at ekiga.net, and want to call someone at sip.freeipcall.com, do I need to get a sip.freeipcall.com account ? |
| 19:44.28 | MikeJ[Laptop] | coppice, I thought the fsk in asterisk was yours? |
| 19:44.33 | jake1932 | s and/an |
| 19:44.36 | coppice | nope |
| 19:44.40 | MikeJ[Laptop] | oh |
| 19:44.43 | tomcontr3 | so, are x100p.com good ones? |
| 19:44.44 | MikeJ[Laptop] | where was it from? |
| 19:44.50 | dlynes_ | tomcontr3: i would say no |
| 19:44.50 | tomcontr3 | or I can get a better card? |
| 19:44.57 | coppice | jesus wrote it |
| 19:45.02 | dlynes_ | tomcontr3: one sec, i'll give you a link someone on here recommended to me |
| 19:45.07 | MikeJ[Laptop] | oh yeah.. .I remember now.... |
| 19:45.37 | asterboy | ~polycom |
| 19:45.39 | jbot | from memory, polycom is the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 19:45.39 | MikeJ[Laptop] | the spanish commnets.. |
| 19:45.46 | dlynes_ | tomcontr3: http://cgi.ebay.com/OEM-Digium-FXO-X100P-X101P-for-ASTERISK-Extras-NEW_W0QQitemZ9713231563QQcategoryZ34165QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 19:45.57 | dlynes_ | tomcontr3: that item has ended, but they've got other items |
| 19:46.04 | coppice | its part of the original zapata stuff from mexico |
| 19:46.35 | nDuff | Careful buying polycom hardware except from a licensed reseller; they pretty much refuse to support it otherwise. |
| 19:47.00 | dlynes_ | nDuff: paradyne does the same thing :) |
| 19:47.17 | ManxPower | nDuff, Um, Polycom doesn't do end user support anyway. |
| 19:47.31 | dlynes_ | actually that would be cool, getting voip to work over paradyne units |
| 19:47.37 | ManxPower | <-- is Polycom certified |
| 19:47.56 | nDuff | ...and careful of their videophones. One of our developers used to work there, and indicated that the development team doing the videophones was pretty much made up of idiots. |
| 19:47.56 | dlynes_ | then you could have phone lines easily in remote locations |
| 19:48.10 | nDuff | That said, the developer in question is on the high end of the talent pool, so his standards could be skewed. |
| 19:48.20 | ManxPower | Polycom Certified is like Microsoft Certified, you really don't have to know much to get the certification |
| 19:48.30 | jake1932 | um |
| 19:48.36 | jake1932 | take that back |
| 19:48.54 | ManxPower | I'm referring to the basic certification of course. |
| 19:49.03 | jake1932 | ok |
| 19:49.04 | jake1932 | :) |
| 19:49.21 | ManxPower | I can't comment on the advanced certifications of either company |
| 19:49.27 | coppice | polycom make phones. I assume certification requires you can dial a number :-) |
| 19:49.42 | ManxPower | "I'm a MCSE! Where's the Start button?" |
| 19:49.50 | ManxPower | coppice, actually it didn't. |
| 19:50.08 | jake1932 | the MCSD is quite different |
| 19:50.33 | MikeJ[Laptop] | the newer mcse is actually a little better.. but certs in general I have found are crap |
| 19:50.39 | coppice | and there I was assuming they would give you a comprehensive evaluation |
| 19:50.56 | ManxPower | I'm considering getting the more advance certs from Polycom. |
| 19:50.59 | coppice | the cisco tests seem fairly meaningful |
| 19:51.07 | MikeJ[Laptop] | I was about to say... |
| 19:51.18 | ManxPower | I'm expecting my largest customer to go bankrupt in a few years. |
| 19:51.37 | MikeJ[Laptop] | but again, they are very cisco specific... you tend tot leave understanding the cisco way well.. but maybe not the protocol |
| 19:52.08 | MikeJ[Laptop] | from what I saw of the bgp tests, it taught you how to set it up well.. but maybe not necessarily to get it |
| 19:52.37 | coppice | these things are supposed to be specific. at least the cisco certs mean you have shown the ability to fix a few real problems |
| 19:53.08 | LostFrog | coppice: take said equipment and throw it at the wall?? |
| 19:54.31 | coppice | www.openvox.com.cn want 19.95 for an x100p. the inflation rate must be high :-) |
| 19:54.57 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 19:55.33 | paryl | has anyone here compiled zaptel drivers on ubuntu with one of the stock apt-getable kernels? |
| 19:55.48 | LostFrog | paryl: yes. |
| 19:55.58 | apr | what Asterisk-friendly service provider would you recommend to get an incoming US phone number? (if it is okay to ask here, otherwise please ignore the question) |
| 19:56.33 | paryl | LostFrog: i've done everything i can honestly think of, and i keep getting "Invalid module format" during modprobe |
| 19:57.52 | ManxPower | brettnem, ALL ITSPS SUCK. Teliax seems to suck less than most. |
| 19:58.01 | ManxPower | ..er... apr ALL ITSPS SUCK. Teliax seems to suck less than most. |
| 19:58.31 | ManxPower | paryl, what does uname -a say? |
| 19:58.35 | apr | Like the Mutt MOTO :) |
| 19:58.56 | ManxPower | apr, Yup! I stole it from Mutt |
| 19:59.10 | paryl | Manx: Linux pbx-okc 2.6.12-10-k7-smp #1 SMP Sat Mar 11 17:18:23 UTC 2006 i686 GNU/Linux |
| 19:59.11 | apr | ;) got it |
| 19:59.16 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 19:59.31 | *** join/#asterisk darby_t (i=darby_t@aapd85.neoplus.adsl.tpnet.pl) |
| 19:59.34 | asterboy | ok I switched the Polycom phone but STILL have a very annoying problem. |
| 19:59.44 | CunningPike | Has anyone had any luck with the g(#) option for Voicemail? |
| 19:59.50 | asterboy | I can TRANSFER calls to the phone, BUT... |
| 20:00.10 | CunningPike | It doesn't seem to make any difference - VoiceMail(2348@default|ug(10)) |
| 20:00.11 | paryl | Manx: and i have installed the headers for that kernel. when i compile zaptel, it installs the modules into /lib/modules/2.6.12/misc though, instead of the correct directory |
| 20:00.20 | *** part/#asterisk critch (i=critch@steven.basesys.com) |
| 20:00.47 | asterboy | ...if the call is coming in on the 1-877-xxx-xxxx number, I can transfer it to the other GXP-2000 phones, but the Polycom IP500 will ring but won't allow you to pickup. Anyone have some ideas? |
| 20:01.15 | paryl | i tried making a link from /lib/modules/2.6.12-10-k7-smp to /lib/modules/2.6.12, and they finally get put in the right spot, but i still get the same error |
| 20:01.35 | *** join/#asterisk noname32 (n=noname@38.113.5.165) |
| 20:01.51 | paryl | Manx: but everything else compiles perfectly... so i don't know if it's just a strange setting somewhere, or if it's a problem in the zaptel source |
| 20:07.36 | brettnem | hey anyone know how to troubleshoot a "Unable to find key 'keyname' to decode shared key from" where "keyname" |
| 20:07.45 | brettnem | obviously has a public key listed in "show keys" |
| 20:07.46 | dlynes_ | Is it the motorola or the intel modems that suck the most? |
| 20:08.10 | brettnem | is there something special I need to do to make it start using those keys? |
| 20:08.54 | [TK]D-Fender | asterboy : ...huh? |
| 20:09.24 | dlynes_ | [TK]D-Fender: he can only refuse transfers on the polycom ip500; he can't accept them |
| 20:09.56 | tainted- | anyone using vpf? |
| 20:10.06 | brettnem | any ideas on the keys, anyone? |
| 20:12.42 | brettnem | wooOOoOOooo silence.. |
| 20:14.13 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:15.17 | [TK]D-Fender | dlynes : Can't accept a transfered call? |
| 20:15.23 | [TK]D-Fender | dlynes : makes no sense... |
| 20:15.46 | asterboy | ][TK[ |
| 20:15.52 | [TK]D-Fender | dlynes : I'd far sooner blame the GXP. I'm running SIP 1.5.2 & 1.6.x here and at home and have never heard of anything like that before... |
| 20:16.42 | asterboy | [TK]D-Fender, the Polycom just won't allow me to pickup a transfered call from specifically a call that comes in on the 1-877-xxx-xxxx number, but all other transfers work. |
| 20:16.55 | asterboy | SIP/2.0 401 Unauthorized |
| 20:17.02 | CunningPike | asterboy: When you say "won't accept", what do you mean? Phone doesn't ring? You pick up and no-one's there? |
| 20:17.06 | asterboy | from, sip debug peer |
| 20:17.12 | asterboy | phone rings. |
| 20:17.17 | dlynes_ | asterboy: sounds like an authentication problem |
| 20:17.21 | asterboy | but you try to pickup and it does nothing. |
| 20:17.22 | dlynes_ | asterboy: not a polycom problem |
| 20:17.33 | asterboy | ya, after getting the SIP/2.0 401 Unauthorized |
| 20:17.42 | asterboy | I'm thinking something is not right. |
| 20:17.53 | ManxPower | What would you say is the best mp2 encoder? (mp2 being audio/video) |
| 20:17.56 | asterboy | strange that it works for any other number |
| 20:18.07 | asterboy | you would think NO transfers would work. |
| 20:18.09 | ManxPower | asterboy, what version of the polycom firmware? |
| 20:18.14 | dlynes_ | but yeah, i'd think the grandstream would be having a problem before the polycom would have a problem |
| 20:18.18 | asterboy | 1.5.3 SIP |
| 20:18.20 | CunningPike | Different context? |
| 20:18.24 | ManxPower | asterboy, BTW, we had that problem with calls coming from QUEUES |
| 20:18.38 | nDuff | ManxPower: I recall seeing a review lately... the best one was commercial, about $500. |
| 20:18.53 | ManxPower | What would you say is the best free mp2 encoder? (mp2 being audio/video) |
| 20:18.58 | ManxPower | 8-) |
| 20:19.00 | asterboy | hmmm...not using queues |
| 20:19.30 | asterboy | sip show peers looks right |
| 20:19.58 | asterboy | sip.conf has the right username and pass |
| 20:20.06 | asterboy | the phone has the right username and pass |
| 20:20.15 | *** part/#asterisk blkcamarozr28 (n=wchan@cpe-72-134-204-179.hawaii.res.rr.com) |
| 20:20.39 | [TK]D-Fender | Queues works fine here with 1.5.x |
| 20:20.56 | ManxPower | [TK]D-Fender, this was only with agents and was in 1.0 |
| 20:21.35 | CunningPike | asterboy: Try canreinvite=no in sip.conf |
| 20:22.17 | CunningPike | Anyone? |
| 20:22.51 | asterboy | ok thanks |
| 20:23.00 | asterboy | here is my config line: |
| 20:23.01 | asterboy | <reg reg.1.displayName="Ed1" reg.1.address="Ed1" reg.1.label="Ed1" reg.1.type="private" reg.1.thirdPartyName="Ed1" reg.1.auth.use |
| 20:23.08 | asterboy | rId="Ed1" reg.1.auth.password="" reg.1.server.1.address="192.168.1.208" reg.1.server.1.port="5060" reg.1.server.1.transport="DNSnapt |
| 20:23.08 | asterboy | r" reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="" reg.1.server.1.register="" reg.1.server.1.retryTimeOut="" reg.1.ser |
| 20:23.10 | asterboy | ver.1.retryMaxCount="" reg.1.server.1.expires.lineSeize="" reg.1.acd-login-logout="0" reg.1.acd-agent-available="0" reg.1.ringType=" |
| 20:23.13 | asterboy | 1" reg.1.lineKeys="1" reg.1.callsPerLineKey="3" |
| 20:23.16 | asterboy | shit |
| 20:23.16 | asterboy | thought that would one liner |
| 20:23.18 | generalhan | WOAH |
| 20:23.19 | generalhan | lol |
| 20:23.32 | nDuff | ManxPower: See http://www.transcoding.org/cgi-bin/transcode?FFmpeg_Vs._Mpeg2enc |
| 20:23.38 | [TK]D-Fender | asterboy : no password? |
| 20:23.45 | *** join/#asterisk drkfiber (n=drkfiber@207.241.43.114) |
| 20:23.46 | asterboy | lol, yes.. |
| 20:23.53 | [TK]D-Fender | reg.1.auth.password="" <------------- |
| 20:23.59 | asterboy | ya I have one |
| 20:24.05 | [TK]D-Fender | that looks like a problem... |
| 20:24.05 | asterboy | just took it out for testing |
| 20:24.08 | nDuff | ManxPower: that said, if you don't have a hard requirement for MPEG2, you might seriously consider Theora. |
| 20:24.16 | [TK]D-Fender | asterboy : Stop polluting our test!! |
| 20:25.10 | ManxPower | nDuff, and mencoder? |
| 20:25.28 | ManxPower | nDuff, not unless DirecTivo supports it 8-) |
| 20:25.39 | ManxPower | nDuff, someone said mencoder is the best, but I'm more of a text console guy |
| 20:25.44 | nDuff | ManxPower: mencoder uses ffmpeg. |
| 20:25.52 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:25.59 | nDuff | ManxPower: that said, mencoder *is* a console app. |
| 20:26.01 | tzanger | mencoder is text console |
| 20:26.05 | ManxPower | nDuff, ah |
| 20:26.09 | tzanger | think mplayer, but backwards |
| 20:26.50 | tzanger | What kind of music ya get here, ma'am? We got both kinds, Country AND Western! |
| 20:27.09 | ManxPower | I' |
| 20:27.27 | vader-- | any of you guys know much about doing vendor-specific options in dhcp? |
| 20:27.31 | ManxPower | I think the current system I'm using uses ffmpeg and gawd the output sucks |
| 20:27.36 | ManxPower | even at 8Mbps |
| 20:27.49 | vader-- | im trying to send a specific tftp server address to a phone based off of what phone it is |
| 20:28.44 | tzanger | vader--: yep |
| 20:28.54 | tzanger | it's not difficult in isc |
| 20:29.24 | tzanger | <PROTECTED> |
| 20:29.28 | tzanger | for example |
| 20:29.29 | vader-- | im using microsoft dhcp |
| 20:29.39 | tzanger | man dhcp-options |
| 20:29.39 | tzanger | oh |
| 20:29.44 | tzanger | well I can't help you there :-) |
| 20:30.00 | tzanger | anyway I'm outta here |
| 20:30.01 | vader-- | and for some reason dhcp is sending option 66 encapsulated into option 43 |
| 20:30.06 | vader-- | so the phone doesn't know what it is |
| 20:31.21 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 20:31.33 | mocker | Woo. |
| 20:31.41 | mocker | yaacid is awesome. ;) |
| 20:31.50 | ManxPower | nDuff, thanks for the link btw |
| 20:32.21 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 20:32.24 | PakiPenguin | evening |
| 20:33.21 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 20:33.24 | *** join/#asterisk Eggplant (i=No@dsl-664.cascadeaccess.com) |
| 20:34.01 | nDuff | ManxPower: np. |
| 20:37.18 | paryl | ManxPower: did you get my previous msgs? |
| 20:40.47 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 20:41.30 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 20:43.36 | jsharp | glorp |
| 20:46.03 | asterboy | Here is a pastebin of the Polycom IP500 *not* able to answer a trasnfered call from 1-877-xxx-xxxx numbers. It *is* able to answer calls transfered from other numbers. By *not* answered, I mean it rings but you can't pickup it just keeps ringing the phone. |
| 20:46.09 | asterboy | http://pastebin.ca/index.php |
| 20:46.28 | brettnem | fun |
| 20:46.47 | asterboy | The output is from 'sip debug peer' |
| 20:46.52 | asterboy | ya real fun |
| 20:47.09 | asterboy | canreinvite=no. |
| 20:47.19 | brettnem | what rev? |
| 20:47.31 | brettnem | hey, nice pastebin link |
| 20:47.38 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 20:47.39 | brettnem | mr.funnypants |
| 20:47.47 | asterboy | lol |
| 20:47.54 | asterboy | 1.5.3 SIP |
| 20:48.13 | asterboy | The big clue here, I think, is the # |
| 20:48.15 | asterboy | SIP/2.0 401 Unauthorized |
| 20:48.31 | asterboy | checked and rechecked the name and pass |
| 20:48.41 | brettnem | hmm.. I wonder what the trace looks like.. hmm |
| 20:49.05 | asterboy | ya that is what I was just going to ask, is there a way to get more information? |
| 20:49.14 | asterboy | more debug info |
| 20:49.23 | asterboy | I'm maxed on the vvvvvvvvvvvvvvvvvvvvv's |
| 20:49.33 | brettnem | well i haven't seen any info.. soo I don't really know what "more" means |
| 20:50.01 | asterboy | oh shit...that's what you meant |
| 20:50.03 | asterboy | http://pastebin.ca/52090 |
| 20:50.10 | brettnem | but I am having a blast at guessing |
| 20:50.11 | brettnem | ;) |
| 20:50.25 | asterboy | don't you guys have crystal balls as standard issue? |
| 20:53.47 | LostFrog | I have lead balls. |
| 20:54.25 | asterboy | kindakinadsfasdf;lkjasdf |
| 20:54.57 | asterboy | balls to walls |
| 20:55.16 | *** join/#asterisk Tangent (n=Tangent@connerdata.plus.com) |
| 20:55.20 | LostFrog | nDuff: We have the biggest balls of them all?? |
| 20:55.21 | Hmmhesays | my ball room's always full, and everybody comes and comes again |
| 20:56.26 | file | Hmmhesays: you're coming? :P |
| 20:56.28 | *** join/#asterisk swytch (n=ezcall@d213-103-233-200.cust.tele2.fr) |
| 20:57.02 | Hmmhesays | always |
| 20:57.25 | Hmmhesays | btw file is there any easy way in SER to replace the contact field in the sdp? |
| 20:59.04 | file | rephrase that so it makes sense |
| 20:59.51 | *** join/#asterisk gursikh (n=guriskh1@adsl-69-151-246-132.dsl.hstntx.swbell.net) |
| 21:00.12 | Hmmhesays | haha |
| 21:00.18 | Hmmhesays | sorry doing too much |
| 21:00.49 | rajiv | any sangoma a200 users? |
| 21:00.56 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
| 21:01.10 | Hmmhesays | can SER replace the contact field on a sip invite with something that I specify |
| 21:01.15 | C4T3l | anyone use mythphone plugin with * ? |
| 21:02.36 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com) |
| 21:02.38 | jake1932 | if i have a provider that is g729 only and a phone that will do g729 - shouldn't asterisk select g729 first (I have the digium licenses)? |
| 21:02.57 | *** part/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:04.05 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net) |
| 21:04.08 | tainted- | jake1932 just disallow=all |
| 21:04.13 | tainted- | and allow=g729 |
| 21:04.27 | jake1932 | right - that'll work |
| 21:04.55 | jake1932 | but if the phone can do ulaw and g729, shouldn't asterisk select g729 when using that provider? |
| 21:06.00 | jake1932 | to avoid transcoding? |
| 21:06.05 | tainted- | depends how u order it |
| 21:06.09 | *** join/#asterisk hads|home (n=hads@203.109.245.87) |
| 21:06.21 | tainted- | if allow=711 is before allow=g729, then it'll try 729 first |
| 21:06.26 | jake1932 | got it |
| 21:06.30 | jake1932 | i'll try that - tnx |
| 21:09.36 | asterboy | can I upload sip 1.6.5 with bootrom 2.6? |
| 21:10.06 | *** part/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 21:10.14 | asterboy | I'm out of things to try as to why the polycom's won't answer a 1-877-xxx-xxxx number |
| 21:11.36 | *** join/#asterisk lzhang (n=rjrae@67.95.13.46) |
| 21:11.44 | fourcheeze | asterboy: is it only polycoms? |
| 21:11.50 | fourcheeze | have you asked polycom? |
| 21:11.51 | asterboy | YES |
| 21:11.59 | asterboy | not asked them yet. |
| 21:12.04 | fourcheeze | well then, ask |
| 21:12.16 | ManxPower | asterboy, Yes, 2.6 bootrom will work with 1.6.x SIP |
| 21:12.24 | twisted[asteria] | what do you mean it won't answer it? |
| 21:12.29 | asterboy | can't get 1.6.x |
| 21:12.39 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:12.40 | asterboy | dumb policy |
| 21:12.41 | ManxPower | asterboy, a link to it is on the wiki |
| 21:12.47 | asterboy | ah ok. |
| 21:12.51 | ManxPower | freedomfones or freedomphones or some such site |
| 21:13.10 | [TK]D-Fender | Upgrading firmware, huh? |
| 21:13.12 | [TK]D-Fender | heh |
| 21:14.00 | [TK]D-Fender | Polycom want to ensure that they don't have to think about end-users... |
| 21:14.09 | asterboy | lol |
| 21:14.43 | [TK]D-Fender | Fortunately I have a "decent" reseller and keep up to date on things. |
| 21:15.27 | [TK]D-Fender | Looking to upgrade your 1.5.x to 1.6.x? |
| 21:15.34 | asterboy | zip |
| 21:16.00 | asterboy | what linux unzip exists? |
| 21:16.07 | [TK]D-Fender | Keep in mind you REALLY should redo your provisioning files from scratch for at least SIP.CFG as there are some important differences. So not jsut replace the sip.ld file |
| 21:16.16 | asterboy | probably have to download something from sourceforge |
| 21:16.29 | [TK]D-Fender | "unzip" |
| 21:16.32 | asterboy | don't think bzip and uncompress will recognize it |
| 21:16.32 | nDuff | asterboy: well, there's one called just "unzip"; most distributions have it. |
| 21:16.52 | asterboy | ok, that's one disadavantage of using lfs |
| 21:17.03 | [TK]D-Fender | MC also support it an every other major archive |
| 21:20.02 | CunningPike | asterboy: Those unatuthorized messages are normal - part of the SIP challenge response process |
| 21:20.14 | CunningPike | It goes Register --> |
| 21:20.24 | CunningPike | <--- Unauthorized |
| 21:20.31 | CunningPike | Register with nonce ---> |
| 21:20.33 | CunningPike | <---- OK |
| 21:21.05 | wunderkin | just taking a poll here, outside of asterisk, a normal phone tech should know the term fxo/fxs right? and don't normal phone systems have an option for a t1 card to interface to maybe another pbx? |
| 21:21.29 | CunningPike | wunderkin: You bet |
| 21:21.37 | wunderkin | CunningPike, both, right? |
| 21:21.42 | CunningPike | Absolutely |
| 21:21.55 | wunderkin | CunningPike, this is sad, i know more than their phone tech.. ok, i thought so, thanks |
| 21:22.04 | wunderkin | ;) |
| 21:22.10 | CunningPike | arse != elbow |
| 21:22.14 | wunderkin | heh |
| 21:22.19 | *** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl) |
| 21:22.33 | asterboy | CunningPike, ok thanks for that, helps to debug. |
| 21:22.44 | vader-- | does anyone know how to create a null byte in vi? |
| 21:23.10 | ManxPower | wunderkin, smaller systems may not have options for T-1 interface. Most PBX T-1 interfaces are EXPENSIVE. |
| 21:23.10 | wunderkin | how many people here have worked with the tdm2400 cards? i know bweschke has and he says it works ok, anyone else? |
| 21:23.22 | ManxPower | Not all of them support Inter-PBX type of T-1 trunks |
| 21:23.33 | ManxPower | wunderkin, I never have and I never will. |
| 21:23.34 | wunderkin | ManxPower, yes, but this guy says that NO PHONE SYSTEMS have a way to interface via a T1 to another pbx |
| 21:23.45 | ManxPower | wunderkin, Nortels do. |
| 21:24.04 | wunderkin | i could sware nortel was one of the logos on their card |
| 21:24.05 | ManxPower | wunderkin, I will use a Adtran TA750 or TA850 with a T-1 card in the Asterisk server. |
| 21:25.42 | wunderkin | maybe there is another way to skin the cat here, they have some kind of key system, can you use some kind of adran box to split the t1 to the key system and a t1 to *? |
| 21:25.42 | asterboy | ~unzip |
| 21:25.44 | jbot | Unpacks .zip files such as those made by pkzip under DOS. URL: http://www.cdrom.com/pub/infozip/ |
| 21:25.44 | ManxPower | wunderkin, The ability to use T-1 -> Telco .vs. T-1 -> Other PBX are different software options for Nortel. |
| 21:25.44 | [TK]D-Fender | asterboy : Install MC. |
| 21:25.44 | wunderkin | yeah, well, no way and expensive/optional are different hehe |
| 21:25.44 | ManxPower | wunderkin, so you want to take some channels off the T-1 to the telco and send them to Asterisk? |
| 21:25.44 | jake1932 | they may call it a tie line or digital interface |
| 21:25.45 | *** join/#asterisk darby_t (i=darby_t@aaov230.neoplus.adsl.tpnet.pl) |
| 21:25.45 | wunderkin | ManxPower, yes |
| 21:25.45 | ManxPower | yes, Tie line would be a term the tech should know. |
| 21:25.47 | ManxPower | wunderkin, Adtran has several boxes that do that. |
| 21:25.47 | asterboy | jbot, unzip is an zip decompression utility that guys like asterboy need cause they run lfs and don't have it by default, so go here to get it http://easynews.dl.sourceforge.net/sourceforge/infozip/unzip552.tar.gz |
| 21:25.48 | jbot | ...but unzip is already something else... |
| 21:25.59 | ManxPower | I believe the term is "drop and insert" |
| 21:26.00 | wunderkin | i was going to try that term but he didn't know fxs, yeah, i thought so, just didn't know the models |
| 21:26.35 | asterboy | jbot, unzip is also an zip decompression utility that guys like asterboy need cause they run lfs and don't have it by default, so go here to get it http://easynews.dl.sourceforge.net/sourceforge/infozip/unzip552.tar.gz |
| 21:26.36 | jbot | asterboy: okay |
| 21:26.40 | *** part/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 21:26.40 | wunderkin | ManxPower, drop and insert would be something can be assigned on the fly? so it can dynamically assign open channels? not predetermined, say i allow 1-16 to port 1 and 17-24 on port 2? |
| 21:26.53 | ManxPower | wunderkin, no. |
| 21:27.08 | ManxPower | wunderkin, want to know how we "integrate" Asterisk with our Nortel PBX. |
| 21:27.29 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 21:27.37 | ManxPower | Before Asterisk: Telco -> CT1 -> Adtran Channel Bank -> Nortel |
| 21:27.51 | wunderkin | alright, i'll bite, i know there are a lot of ways to do it, it will require other hardware that they don't have which means they probably won't do it anyway, but i would like to tell them at least what their options are |
| 21:27.56 | ManxPower | After Asterisk: Telco -> CT1 -> Asterisk -> Adtran Channel Bank -> Nortel |
| 21:28.19 | ManxPower | Asterisk has 4-port T-1 card, two to the telco, two to the adtrans. |
| 21:28.21 | wunderkin | yeah... i didn't really want to put asterisk in front, but.. |
| 21:28.57 | ManxPower | Now, we ALSO use the Zaptel DACS features to just patch some channels from the telco T-1 into the same channels on the Adtran. DACS does not require Asterisk |
| 21:29.00 | wunderkin | what about the telco -> ct1 -> adtran channel bank -> asterisk/notel |
| 21:29.13 | ManxPower | wunderkin, that would just be silly. |
| 21:29.16 | wunderkin | or something else i mean instead of a channel bank |
| 21:29.37 | ManxPower | Why not Telco -> CT1 -> Asterisk -> Your PBX |
| 21:29.54 | ManxPower | You can stop and start Asterisk all day and not impact the DACS channels |
| 21:30.05 | ManxPower | just don't run ztcfg or reboot or unload the zaptel drivers. |
| 21:30.36 | wunderkin | well, i'm not their phone person, i'm just coming in as a tech guy, trying to 'add on', their phone guy would be responsible for anything, so that rules out putting asterisk in front |
| 21:31.25 | wunderkin | sure i could handle asterisk, but since it is in front effecting their normal pbx, i don't want that responsibilty |
| 21:31.36 | wunderkin | and that would get kinda hairy |
| 21:32.35 | wunderkin | if i could use some box in front of the pbx (something they would know how to configure) that would split out the channels between the pbx and asterisk, that would be cool |
| 21:33.37 | wunderkin | that would be a lot better than getting a tdm card and using fxs ports off of their pbx, blah! |
| 21:33.52 | asterboy | ok, I upgraded to 1.6.2, still same ... |
| 21:34.00 | asterboy | wonder if bootrom update would help |
| 21:34.06 | [TK]D-Fender | asterboy : nopt |
| 21:34.09 | [TK]D-Fender | nope |
| 21:34.21 | asterboy | has to be something in config. |
| 21:34.36 | asterboy | I'll talk to Polycom now...done all I can think of. |
| 21:34.55 | [TK]D-Fender | asterboy : I'll give you a few mins by SSH if you like. |
| 21:35.50 | asterboy | ok, may take you up on that |
| 21:36.03 | asterboy | lol, polycom's tech support menu is broken |
| 21:36.05 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 21:36.13 | SexyKen | Max Calls 15 -- Calls 0 -- Hold Time 245 -- Completed 118 -- Abandoned 74 -- Service Level 0 -- Service Level Perf 0.0 -- Weight 0 |
| 21:36.24 | SexyKen | Asterisk reports that information -- now what I want to know is for Hold time, Completed & Abandoned....how is this actually calculated? |
| 21:36.24 | dlynes | cool |
| 21:36.30 | [TK]D-Fender | btw : if they hear "ASTERISK" THEY MAY RUN FOR THE HILLS... |
| 21:36.36 | SexyKen | IE (What period of time is this based on? Since Asterisk was last started??) |
| 21:37.00 | jake1932 | service level - 0 :) Wow! |
| 21:37.13 | SexyKen | Well -- that doesn't change..I dont know what the purpose of that is |
| 21:37.13 | *** part/#asterisk apr (n=apr@r200-40-234-53-dialup.adsl.anteldata.net.uy) |
| 21:37.40 | dlynes | jake1932, I think asterisk is trying to tell him that he might soon take the title of worst call center away from Telus :) |
| 21:37.44 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 21:37.47 | jake1932 | haha |
| 21:38.09 | SexyKen | jake: show me what your queues service level is? |
| 21:38.12 | dlynes | but for that to happen, i think service level would have to be -100 or something |
| 21:38.21 | *** join/#asterisk critch (i=critch@steven.basesys.com) |
| 21:38.23 | jake1932 | SexyKen: 100% |
| 21:38.30 | asterboy | dlynes, Telus will be hard to take that title away from |
| 21:38.36 | SexyKen | Right...and how do you get that calculated? |
| 21:38.41 | dlynes | asterboy, lol...you live in BC or Alberta, I take it? |
| 21:38.44 | jake1932 | no queue - we answer all our calls |
| 21:38.50 | jake1932 | (right away) |
| 21:38.54 | dlynes | oh yeah...edmonchuck |
| 21:38.58 | asterboy | yep, tax paid monopoly land |
| 21:39.06 | SexyKen | LoL Jake, you're funny..you must be pueny, right? |
| 21:39.09 | critch | been out of new asterisk code for too long. Is there a easy way to look at what formats are recognized by asterisk for use in agi record file command? |
| 21:39.34 | jake1932 | sexy ken - we are a small shop - but I don't know of any big shop with a 0% service level |
| 21:39.45 | jake1932 | that takes a lot |
| 21:39.49 | dlynes | critch, show file formats |
| 21:39.49 | SexyKen | Jake, you obviously aren't paying attention. |
| 21:39.57 | SexyKen | It's a statistic that isn't being calculated. |
| 21:39.59 | SexyKen | Smart ass. |
| 21:40.04 | critch | dlynes: thanks |
| 21:40.12 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:40.33 | asterboy | so, polycom won't provide direct support to end-users |
| 21:40.40 | dlynes | nope |
| 21:40.53 | dlynes | You have to go the company you bought it from |
| 21:42.26 | [TK]D-Fender | jake1932 : is that during moments of brilliance raising you above "dumb ass"? :) |
| 21:42.40 | jake1932 | indeed |
| 21:42.43 | tainted- | ouch! |
| 21:42.44 | [TK]D-Fender | heh |
| 21:42.55 | jake1932 | but it's all relative |
| 21:43.04 | tainted- | haha |
| 21:43.13 | [TK]D-Fender | heh |
| 21:43.31 | tainted- | [TK]D-Fender learn SER!! |
| 21:43.41 | SexyKen | No one knows how that information is calculated or from what period of time it is calculated? |
| 21:43.43 | [TK]D-Fender | tainted- : I may one day... so far never needed to.... |
| 21:44.09 | [TK]D-Fender | and by the time its worthwhile something else will have taken its place. |
| 21:44.23 | [TK]D-Fender | I should learn LINUX first.... |
| 21:44.23 | tainted- | yea no kidding |
| 21:47.06 | mikefoo | whats SER? |
| 21:47.14 | tainted- | sip express router |
| 21:47.38 | asterboy | well no polycom support |
| 21:49.55 | C4T3l | SER allows asterisk to scale |
| 21:50.10 | *** part/#asterisk critch (i=critch@steven.basesys.com) |
| 21:50.41 | C4T3l | you can load balance with it or on special occasions it'll make you breakfast in bed! |
| 21:50.47 | tessier | It's funny that there are 19 people in #ser and 323 people in #asterisk when it should be the opposite. |
| 21:50.55 | C4T3l | i agree |
| 21:51.02 | tessier | You can't do more than an office pbx with asterisk without SER. |
| 21:51.03 | tainted- | C4T3l hmm i must be using the wrong build |
| 21:51.06 | *** join/#asterisk codebreaker (n=codebrea@xserver.flexserv.de) |
| 21:51.22 | tessier | You can try but it won't work well if it works at all (depends on how big you go). |
| 21:51.24 | C4T3l | how so |
| 21:51.28 | tainted- | this is the most i've heard from SER users EVER |
| 21:51.48 | tainted- | C4T3l my SER just pushes SIP packets around.. no breakfast :( |
| 21:51.48 | tessier | The biggest difference is that SER only handles the SIP packets. |
| 21:52.08 | tainted- | tessier C4T3l what scale / # of * boxes are u guys at? |
| 21:52.14 | tessier | Asterisk handles the media also. If you have calls going from VOIP phone to PSTN gateway you don't want that media going through asterisk. |
| 21:52.19 | C4T3l | tainted-: sorry... |
| 21:52.39 | tainted- | C4T3l do u think u could interconnect some breakfast this way? |
| 21:53.07 | C4T3l | i prefer my breakfast with alittle bit of g729 sauce |
| 21:53.23 | tainted- | i wish it would do IAX |
| 21:53.24 | C4T3l | a little ulaw for taste |
| 21:53.28 | tainted- | not that i care |
| 21:53.41 | tainted- | but sometimes i get the stray who insists on IAX |
| 21:53.41 | C4T3l | oh man iwish it would do iax as well |
| 21:54.01 | codebreaker | i would like to call 12345 but i dont know if he is connected to server1,server2 or server3(all have the same iax.conf). how can i connect to server1 and he will look on wich server[123] 12345 has registered? |
| 21:54.16 | C4T3l | dundi? |
| 21:54.16 | tainted- | but i think freeswitch will bitchslap asterisk and SER |
| 21:54.35 | codebreaker | i think this was possible long befor dundi get created |
| 21:54.36 | C4T3l | ques que se "freeswitch" |
| 21:54.36 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 21:55.45 | tainted- | it is sweetness |
| 21:55.50 | tainted- | frenchie |
| 21:56.19 | C4T3l | how does freeswitch compare to * ? |
| 21:56.26 | dlynes | C4T3l, shenme yi ci? c'est la telephoner systeme de la #freeswitch |
| 21:56.28 | tainted- | it's media, it's ivr, it's proxy, it's sip, it's iax |
| 21:56.31 | codebreaker | like having rrd in dns. all iax clienst registering to iax.maydomain wich is a rrd to 5 different server. now i need to coppel them so together that they will route the calls between the servers. |
| 21:56.53 | dlynes | tainted-, how can it be ivr? it doesn't have a pbx module yet |
| 21:57.00 | CunningPike | asterboy: Fellow Canuck here |
| 21:57.03 | tainted- | dlynes just wait :D |
| 21:57.14 | VoicePulse | tainted-: Just received an email that those API functions you were looking for are being moved into production right now. |
| 21:57.34 | tainted- | VoicePulse bout f'in time.. asked for that stuff almost two years ago |
| 21:57.53 | swytch | anyone has experience/thought on using OpenRadius to fork radius accouting start/stop requests to two other radius servers. |
| 21:58.00 | dlynes | See? That's how you get stuff done...swear at people that help you :) |
| 21:58.06 | VoicePulse | tainted-: Can't change that, but they're there now :) |
| 21:58.53 | tainted- | dlynes simple API to provision DIDs.. i was promised that late 2004 |
| 21:59.18 | dlynes | tainted-, damn...and that message from voicepulse just seemed to come out of the blue somewhere |
| 21:59.37 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:59.40 | tainted- | well he pvt msged me some stuff earlier |
| 21:59.45 | dlynes | ah |
| 21:59.51 | tainted- | this is probably his marketing pitch to the channel |
| 22:00.47 | CunningPike | asterboy: Do you want to email me your configs? |
| 22:01.10 | dlynes | ah...pike...you're in vancouver, too? |
| 22:01.31 | CunningPike | Yup :D |
| 22:01.33 | CunningPike | Small world |
| 22:01.46 | dlynes | So how well do you know Asterisk? |
| 22:02.05 | CunningPike | Adequately - why |
| 22:02.14 | file | adequately, cute |
| 22:02.22 | dlynes | Just curious...my boss is looking for some other Asterisk people in town here |
| 22:02.34 | CunningPike | Ah |
| 22:02.47 | dlynes | ah |
| 22:02.48 | dlynes | lol |
| 22:02.59 | dlynes | So I guess you're not available for consulting then :) |
| 22:03.02 | tainted- | dlynes [hC], harryvv, firestrm are in your area |
| 22:03.05 | DoktorGreg | whats the variable that holds the channel the call is from the way exten holds the dialed digits? |
| 22:03.13 | CunningPike | Only outside office hours, I'm afraid |
| 22:03.24 | dlynes | CunningPike, That's still fine |
| 22:03.31 | CunningPike | Really? |
| 22:03.34 | dlynes | CunningPike, if you want some extra work, feel free to drop me an email then |
| 22:03.38 | CunningPike | k |
| 22:03.44 | asterboy | CunningPike, what configs do you have? |
| 22:03.53 | dlynes | cunningpike: dlynes@247communications.com |
| 22:04.02 | CunningPike | OK - thanks dlynes |
| 22:04.27 | denon | CunningPike: so you're the guy who wrote the Homeland Security sniffer module for IAX2? |
| 22:04.28 | CunningPike | asterboy: Come again? |
| 22:04.33 | dlynes | DoktorGreg, ${CALLERID(num)} |
| 22:04.40 | asterboy | interesting that the calls can be parked and picked up again. |
| 22:04.43 | CunningPike | Rats, I've been rumbled |
| 22:04.49 | DoktorGreg | thx |
| 22:04.51 | asterboy | 16:00 < CunningPike> asterboy: Do you want to email me your configs? |
| 22:04.57 | Dr-Linux | can we assign multiple context in zapata.conf if using dual T1 cards? |
| 22:05.09 | CunningPike | asterboy: I was offering to take a peek at yours |
| 22:05.38 | asterboy | ah ok, ya just saw the other window. |
| 22:05.44 | asterboy | sometimes miss that in irssi |
| 22:06.03 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:06.10 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 22:06.13 | CunningPike | Me too - Colloquy does the same |
| 22:06.23 | Dr-Linux | can we assign multiple context in zapata.conf if using dual T1 cards? |
| 22:07.40 | *** join/#asterisk tparcina (n=tparcina@83.131.134.105) |
| 22:09.09 | dlynes | Dr-Linux, yeah...you can even assign contexts on a channel by channel basis if you want |
| 22:10.23 | Dr-Linux | dlynes: i'll DIDs, so i'd like to route each DID to particular context |
| 22:11.44 | [TK]D-Fender | Dr-Linux : Typically you use 1 context in zapata for all of your channels, and they call an exten based on the DID and from there you can do whatever you want. |
| 22:13.35 | Dr-Linux | [TK]D-Fender: yeah, that what i was thinking, exten will sense callerid and will go to specific context? |
| 22:14.21 | asterboy | CunningPike, here is my pasty: http://pastebin.ca/52109 |
| 22:14.51 | [TK]D-Fender | Dr-Linux - no, incoming calls LAND on an exten. |
| 22:15.16 | asterboy | strange though that I can park a call and pickup at the polycom. |
| 22:15.35 | Dr-Linux | like i have 2 DIDs, zapata.conf has context=incoming but i want one DID on context "agi" and 2nd DID to "ivr" |
| 22:16.12 | asterboy | just can't pickup a transfered specific 1-877 type number even though the phone rings. |
| 22:16.23 | *** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 22:17.02 | bzbw | anyone here? |
| 22:17.10 | mog_work | no |
| 22:17.19 | brookshire | i agree |
| 22:17.21 | asterboy | nope all snapped up by women |
| 22:17.30 | bzbw | ha, just got a quick question |
| 22:17.34 | Dr-Linux | [TK]D-Fender: both contexts "agi" and "ivr" are in the extensions.conf |
| 22:17.38 | brookshire | but nobody is here |
| 22:18.18 | bzbw | don't know why my following context make the sip call to a unknow IP address, appreciate it for help:) |
| 22:18.19 | bzbw | <PROTECTED> |
| 22:18.19 | bzbw | <PROTECTED> |
| 22:18.25 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:18.58 | bzbw | the BroadVoice context supposedly should go to sip.broadvoice.com, but it goes to a unknow IP |
| 22:19.08 | bzbw | for sip invite, |
| 22:19.38 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:19.42 | *** join/#asterisk nortex (n=nortex@snapper.titanspecialties.com) |
| 22:20.12 | [TK]D-Fender | Dr-Linux : inside [incoming] you'll make an exten like ---- exten => 8005551234,1,Goto(ivr1,s,1) and so on |
| 22:20.17 | bzbw | nslookup will resolve to the right Ip address for broadvoice sip server |
| 22:20.29 | CunningPike | asterboy: Plowing through it now...... |
| 22:20.43 | bzbw | this is for outgoing calls. |
| 22:21.08 | asterboy | ok thanks |
| 22:22.11 | Dr-Linux | [TK]D-Fender: i see, Thanks it's the same method that i'm already using i thought there will be some difference in DID/T1 case |
| 22:22.14 | bzbw | pretty much I define a peer(using Broadvoice account) and any extension can use this peer to dial out for PSTN call, but it simply keep failing |
| 22:23.08 | *** part/#asterisk tparcina (n=tparcina@83.131.134.105) |
| 22:23.13 | Dr-Linux | bzbw: where you defined broadvoice's host/ip ? |
| 22:23.22 | asterboy | CunningPike, you might also want to look at sip debug peer: http://pastebin.ca/52090 |
| 22:23.39 | CunningPike | asterboy: Check your pm window |
| 22:24.01 | *** join/#asterisk tparcina (n=tparcina@83.131.134.105) |
| 22:24.15 | vader-- | have any of you guys done or seen anyone link asterisk with active directory? |
| 22:24.21 | bzbw | here is the configuration: |
| 22:24.37 | bzbw | host=sip.broadvoice.com |
| 22:24.37 | bzbw | fromdomain=sip.broadvoice.com |
| 22:24.53 | bzbw | it's defined as "friend". |
| 22:25.03 | Dr-Linux | vader--: windows? |
| 22:25.42 | vader-- | ya |
| 22:25.45 | Dr-Linux | bzbw: try IP instead, otherwise call their support |
| 22:25.56 | saftsack | is it possible to change the CID number depentend from the MSN of the callers telephone? |
| 22:26.22 | bzbw | I did, not working: ;host=147.135.0.128 |
| 22:26.22 | bzbw | ;host=147.135.12.128 |
| 22:26.37 | bzbw | tried both ips. |
| 22:26.57 | Aurs | with ; ? |
| 22:27.14 | Dr-Linux | Aurs: he tried. |
| 22:27.23 | *** part/#asterisk tparcina (n=tparcina@83.131.134.105) |
| 22:27.24 | bzbw | nope, just commented it out once it's not working |
| 22:27.25 | Dr-Linux | bzbw: did you call their support? |
| 22:27.33 | Aurs | anyway.. I'm off to bed ;) |
| 22:27.40 | bzbw | they sucks, no one answered:( |
| 22:28.15 | *** join/#asterisk cp5 (n=cp5@rtr3-t1.fonality.com) |
| 22:28.18 | bzbw | btw, I also use following to register: |
| 22:28.19 | bzbw | register => 8885551212:password@147.135.12.128/8885551212 |
| 22:28.57 | vader-- | any ever have a problem where when you first call from sip to sip there is a moment of silence on one end even if a person is talking |
| 22:28.57 | bzbw | bottom line, it used to works, but suddently not working anymore:( |
| 22:29.22 | vader-- | like i called another phone i have here and a person on the one end was talking and nothing was coming over then all of a sudden i heard them |
| 22:29.23 | vader-- | find |
| 22:29.23 | vader-- | fine |
| 22:30.06 | cp5 | hi guys, does anyone know how to track back a channel to the original channel that called into asterisk? for example, a call comes into asterisk over Zap/1, goes to a queue, queue goes to Local/exten@somewhere, that ends up doing a Dial(...M(some-macro))...now there's a bunch of channels, but from within [macro-some-macro] I want to know the original channel (e.g., Zap/1) |
| 22:30.35 | vader-- | cp5 pass it in as a variable to your macros |
| 22:30.40 | vader-- | and then just carry it through |
| 22:31.11 | bzbw | emm, looks like this question maybe a tough one, no one had a clue. |
| 22:31.11 | cp5 | hmm, so i would have to set SOMEVAR to ${CHANNEL} at the beginning of the call before it gets routed anywhere, correct? |
| 22:31.26 | vader-- | ya |
| 22:32.09 | vader-- | and then when you call your macro just pass that in as an arguement |
| 22:32.16 | vader-- | and make sure to keep passing along |
| 22:32.45 | cp5 | will the variable even get passed once it hits Local/extn@context? |
| 22:32.53 | vader-- | i dunno |
| 22:32.57 | cp5 | k, i'll try |
| 22:33.05 | vader-- | im not that fluent in asterisk dial plans just yet |
| 22:33.17 | [TK]D-Fender | cp5 : not out of a queue... there are ways around this if you're creative. |
| 22:34.08 | cp5 | vader, it seems to get dropped after it goes through Queue() -> Local/exten@context |
| 22:34.13 | bzbw | anyone saw this warning before, what is it? |
| 22:34.14 | bzbw | Apr 28 18:33:45 WARNING[9642]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '' |
| 22:34.15 | cp5 | D-Fender, any suggestions? |
| 22:34.44 | cp5 | even calling an AGI to perform a `show channels`, i have no way to track it back (if there are multiple calls) |
| 22:35.14 | Dr-Linux | bzbw: the remote host DNS didn't reloved |
| 22:35.19 | [TK]D-Fender | cp5 : Change the caller ID to a unique value and use that as a DB key, changing it back temporarily whenever necessary. |
| 22:35.25 | Dr-Linux | resovled |
| 22:35.39 | cp5 | [TK]D-Fender -- ahh interesting idea |
| 22:36.04 | CunningPike | cp5: Each leg of a call is really a separate call, so the variable idea is your best bet |
| 22:36.17 | cp5 | CunningPike, the variable gets lost when the call hits the Queue |
| 22:36.48 | bzbw | Dr-Linux: I did not define using DNS, why does it try to do so? And btw, all the FQDN name can be resolved in my sip.conf file for A Record. |
| 22:36.50 | cp5 | the only queue member is a Local/s@context, and within context, it's already gone by priority 1 |
| 22:36.56 | cp5 | i'll try with callerid.. |
| 22:37.13 | [TK]D-Fender | <- (successful) 2-bit hack |
| 22:37.31 | CunningPike | Right - may because you're switching channels? I think variables have channel scope |
| 22:38.32 | bzbw | are following 2 lines achieve the same thing? |
| 22:38.32 | bzbw | exten => _2.,3,Dial(sip/${EXTEN:1}@BroadVoice) |
| 22:38.32 | bzbw | exten => _2.,3,Dial(sip/BroadVoice/${EXTEN:1}) |
| 22:41.20 | tainted- | bzbw depends on provider |
| 22:41.26 | Dr-Linux | bzbw: i heard asterisk has srv bug as well. but not sure |
| 22:41.31 | tainted- | bzbw why are u using broadvoice |
| 22:41.42 | bzbw | because it's cheap:( |
| 22:42.09 | bzbw | i'm using 1.2.7.1, supposedly fix the bugs:0 |
| 22:42.14 | Dr-Linux | bzbw: what don't you call their support team? |
| 22:42.41 | [TK]D-Fender | Broadvoice will quickly divorce themselves from you upon hearing the name "Asterisk" |
| 22:42.51 | bzbw | honestly, I did a test with my sip phone with the account, it works!:( |
| 22:43.03 | tainted- | bzbw which plan are you using |
| 22:43.11 | bzbw | 24.99 |
| 22:43.13 | tainted- | broadvoice hates asterisk |
| 22:43.24 | Dr-Linux | [TK]D-Fender: oo why? Broadvoice doesn't like asterisk? :) |
| 22:43.25 | mog_work | ?!? |
| 22:43.27 | tainted- | bzbw how many minutes did u use last month |
| 22:43.43 | bzbw | not a lot, just some normal calls. |
| 22:43.46 | tainted- | Dr-Linux b/c their customer service people have silly putty for brains |
| 22:43.47 | bzbw | why? |
| 22:44.28 | bzbw | tainted: hope BV does not block asterisk base on SIP user agent:( |
| 22:44.42 | tainted- | bzbw u can always hide useragent |
| 22:44.49 | tainted- | make it 'broadvoice blows' |
| 22:44.50 | bzbw | yape |
| 22:45.13 | Dr-Linux | i used to Vonics sip provders |
| 22:45.18 | bzbw | hmm, strange things is it was working a few more days ago. |
| 22:45.20 | Dr-Linux | providers |
| 22:45.30 | tainted- | bzbw not surprising |
| 22:45.35 | cp5 | [TK]D-Fender, great idea regarding the cid |
| 22:45.45 | tainted- | bzbw they change configuration on u all the time |
| 22:45.48 | tainted- | w/o telling anyone |
| 22:45.57 | nahirean | broadvoice is a parent company of g-naps, no? or is it vice versa? or are they unrelated? |
| 22:46.11 | bzbw | this time seems to be my * issue, as when I use tcpdump, the invite went to a unknown IP:( |
| 22:46.27 | tainted- | nahirean my nufone account doesn't work.. any ideas? |
| 22:47.30 | nahirean | tainted, hmm.. have you considered http://www.dslreports.com/r0/download/985083~eb5b1adf17cc7f4063683940d86efe7d/jerkjerk.jpg .. it's a flowchart of the current situation :) |
| 22:47.48 | bzbw | and I tried another service provider, seem to have similar issue, somehow in the host=FQDN defined in the context, the FQDN is not resolved correclty by * |
| 22:48.00 | tainted- | INCONCEIVABLE! |
| 22:48.20 | tainted- | that's going on my dart board |
| 22:49.54 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 22:50.09 | *** join/#asterisk tomcontr3 (n=gcontrer@68-78-246-201.adsl.terra.cl) |
| 22:50.33 | tomcontr3 | can any one pastebin me a good and working SIP TRUNK config? |
| 22:51.54 | tomcontr3 | ?? |
| 22:54.26 | *** join/#asterisk startled (i=startled@d58-105-31-172.dsl.vic.optusnet.com.au) |
| 22:55.03 | znoG | question: i have setup Asterisk on one server, and over the Internet to another server, which is plugged (by means of a TDM2400 card) to a Lucent PBX. Every now and then calls from/to Asterisk between the Lucent and the * box seem to "hang" (ie. hangup is never detected). Can anything be done about this? |
| 22:55.14 | znoG | busydetect maybe for SIP? |
| 22:58.13 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:59.00 | *** join/#asterisk RoyK (n=roy@ti211310a080-13647.bb.online.no) |
| 23:01.34 | *** join/#asterisk Eggplant (i=No@dsl-664.cascadeaccess.com) |
| 23:06.17 | Dr-Linux | any UK dude around? |
| 23:17.21 | cp5 | [TK]D-Fender or anyone else, what are some variables that i can set that will be kept after Queue() is called? for example, SetCIDName and SetCIDNum will do it |
| 23:17.56 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.191.Dial1.SanJose1.Level3.net) |
| 23:19.47 | asterboy | anyone report a bzzziitttt sound on a freezing GXP-2000 |
| 23:20.12 | asterboy | occasionally and for no reason the gxp will freeze. |
| 23:21.03 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 23:21.35 | [TK]D-Fender | cp5 : I suggest you just set that and as I said, use is as a KEY for a database in which you can whatever the heck you want. |
| 23:23.15 | CunningPike | asterboy: In our somewhat limited experience with grandstream, that's expected behaviour :) |
| 23:23.32 | asterboy | crap, now they are all doing that. |
| 23:24.38 | [TK]D-Fender | I think poor design and shoddy workmanship are perfectly valid reasons :) |
| 23:25.44 | Dr-Linux | asterboy: what is bzzziitttt? |
| 23:28.21 | [TK]D-Fender | Dr-Linux : A noise |
| 23:28.51 | asterboy | it just freezes in the middle of the call or whenever just ringing sometimes. |
| 23:29.07 | asterboy | that is the sound it makes over the speaker. |
| 23:29.12 | asterboy | loud. |
| 23:29.25 | asterboy | only way to fix it is to pull out the fucking power chord. |
| 23:29.41 | asterboy | I'm going to dump these phones big time. |
| 23:29.45 | asterboy | pieces of shit. |
| 23:30.19 | CunningPike | Polycom - you're only man |
| 23:30.31 | Dr-Linux | asterboy: relax and replace them with some polycom or cisco |
| 23:30.50 | [TK]D-Fender | asterboy : "power chord" major or minor? :) |
| 23:31.00 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
| 23:32.04 | asterboy | lol |
| 23:32.09 | ManxPower | asterboy, did you get the polycoms working? |
| 23:32.17 | ghost99 | if i dial the line and it just answers with a bad tone .. what have i not done ? |
| 23:32.19 | asterboy | ya, I'm going to suggest polycom...what I suggested in the first place |
| 23:32.36 | asterboy | ManxPower, no...but I did a work around using Parking |
| 23:32.53 | ManxPower | asterboy, something bizzare is going on. They work just fine for us. |
| 23:33.02 | *** join/#asterisk Vahram (n=VX@83.139.4.112) |
| 23:33.04 | asterboy | ya, I'm not exactly sure |
| 23:33.07 | cp5 | [TK]D-Fender, i'll do that if necessary, just rather use a variable and not require disk access (writing to astdb) |
| 23:33.17 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:33.24 | asterboy | CunningPike and I have thouroughly combed the configs |
| 23:34.07 | asterboy | what about firmware upgrade on the gxps |
| 23:34.09 | asterboy | > |
| 23:34.22 | asterboy | like like 1.1.0.1 |
| 23:34.36 | asterboy | dam echo followed me in here too :P |
| 23:36.46 | riddlebox | Strom_C: are you around? |
| 23:40.02 | poisoner | hmm |
| 23:40.24 | ghost99 | if i dial up and my card x100p card picks up the trunk line and plays a dialtone at me .... what have i done wrong can anyone figure ? |
| 23:40.50 | ghost99 | casue once it does than i can't hang up without upplugging the line |
| 23:40.52 | poisoner | has anyone run into problems with segfaulting app_meetme when user is joining? |
| 23:41.29 | Dr-Linux | ghost99: imediate=yes/no ? |
| 23:42.11 | ManxPower | ghost99, I have never ever heard of that problem |
| 23:42.30 | ManxPower | ghost99, what does the Asterisk console say? |
| 23:44.35 | *** join/#asterisk mog_home (n=achika54@68.62.237.103) |
| 23:44.59 | ghost99 | <PROTECTED> |
| 23:44.59 | ghost99 | <PROTECTED> |
| 23:44.59 | ghost99 | <PROTECTED> |
| 23:45.48 | ManxPower | ghost99, um, Milliwat generates a 1000Hz @ 1db, which sounds quite annoying. That's it's sole reason for existance. |
| 23:45.52 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 23:46.07 | Ariel_ | buzzzzzzzzzzzzz |
| 23:46.18 | asterboy | do you have to upgrade the gxp-2000 in steps? |
| 23:46.37 | ghost99 | mmanxpower: when i hang up i can hear the real signal under that annoying tone |
| 23:46.41 | asterboy | trying to upgrade from 1.0.1.2 to 1.1.0.1 |
| 23:46.44 | ManxPower | asterboy, The best bet to upgrade is to throw it down some stairs |
| 23:46.46 | asterboy | but just ignores |
| 23:46.50 | *** join/#asterisk nortex (n=breeves@adsl-69-149-172-106.dsl.amrltx.swbell.net) |
| 23:46.51 | asterboy | lol |
| 23:46.52 | asterboy | yep |
| 23:47.08 | asterboy | I wouldn't even use this crap for a door stop |
| 23:47.08 | ManxPower | asterboy, you have more problems with Asterisk than anyone I've seen in at least a year |
| 23:47.16 | asterboy | lol |
| 23:47.17 | bzbw | asterboy: 1.1.0.11 is about available, I've heard from GS:) |
| 23:47.31 | asterboy | yep, if I can trip up * with every problem, I will. |
| 23:47.32 | [TK]D-Fender | ManxPower : No, I think you've seen more from Dr-Linux here :) |
| 23:47.34 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:47.44 | asterboy | how do you do the upgrade? |
| 23:47.49 | ManxPower | [TK]D-Fender, Ah. He was on /ignore for a long time, I think. |
| 23:48.00 | asterboy | I put the file in my tftp root and pointed to my server |
| 23:48.07 | ghost99 | dr_linux: immediate = yes |
| 23:48.26 | ManxPower | ghost99, you never ever want immediate=yes |
| 23:48.47 | ghost99 | so i can change that to a no and reload ? |
| 23:49.10 | ManxPower | ghost99, correct. |
| 23:49.25 | ManxPower | it's not going to fix your current problem, however, but it will avoid future problems |
| 23:50.14 | ManxPower | Asterisk is doing EXACTLY what you have told it to do. Pick up the line and run Milliwatt |
| 23:52.44 | cp5 | anyone know of other variables that i can set before Queue() and still have it from within the queue? e.g., the callerid is the only variable i know of that i can set beforehand and have it afterwards |
| 23:53.18 | bzbw | guys, this has to be my * bug, but i have no clue:( |
| 23:53.18 | bzbw | INVITE sip:18885551212@BroadVoice SIP/2.0 |
| 23:53.19 | bzbw | Via: SIP/2.0/UDP 111.111.111.222:5060;branch=z9hG4bK05a58947;rport |
| 23:53.19 | bzbw | From: "GS" <sip:8885551211@111.111.111.222>;tag=as72553000 |
| 23:53.19 | bzbw | To: <sip:8885551212@BroadVoice> |
| 23:53.19 | bzbw | Contact: <sip:8885551211@111.111.111.222> |
| 23:54.07 | bzbw | The to line supposedly should resolve to the host="sip proxy" defined in sip.conf for BroadVoice Context |
| 23:54.44 | bzbw | what could be wrong? |
| 23:55.07 | bzbw | I was using "sip debug" to capture this. |
| 23:56.03 | [TK]D-Fender | bzbw : next time PASTEBIN, and while you're at it pastebin you sip.conf |
| 23:56.30 | [TK]D-Fender | ~pb |
| 23:56.31 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 23:56.37 | bzbw | u need to see the whole sip.conf? |
| 23:56.49 | asterboy | what am I missing to upgrade the GXP? |
| 23:56.58 | [TK]D-Fender | bzbw : all the pertinent parts |
| 23:57.15 | asterboy | put file in tftp server root, update phone to point at server and reboot. |
| 23:57.21 | asterboy | did I miss something? |
| 23:57.30 | bzbw | k |
| 23:57.52 | asterboy | has anyone else actually upgraded a GXP? |
| 23:58.04 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-140-189.dyn.sprint-hsd.net) |
| 23:58.25 | asterboy | that too, but if you want the latest beta stuff. |
| 23:58.34 | asterboy | you need to point at your own server. |
| 23:58.43 | cp5 | anyone know of a character that i can put into the callerid string (similar to a null terminator) and put data after that, but only the data before that character will be displayed on phones? |
