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| 01:03.10 | *** join/#asterisk Rob3Rt (n=admin@87.37.96.58.static.exetel.com.au) |
| 01:03.12 | Rob3Rt | hey guys |
| 01:03.30 | Rob3Rt | im getting an auth name/pass mismatch but i set it myself in sip.conf when I set up the extension |
| 01:03.34 | Rob3Rt | is this a common issue ? |
| 01:03.39 | Rob3Rt | all codecs are supported etc |
| 01:03.51 | Rob3Rt | no other extensions of similar format naming wise. |
| 01:04.44 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
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| 01:43.02 | [TK]D-Fender | Rob3Rt: If you say so. |
| 01:46.38 | Rob3Rt | I asked a question [TK]D-Fender, are you always this obtuse? At times you seem helpful, most other times you're objective. |
| 01:46.48 | Rob3Rt | What's the deal? |
| 01:47.16 | Rob3Rt | I'm reading what I need to read. Some of doesn't make sense until I can find my way around. Just asking questions about those points . |
| 01:48.51 | Alfio | Rob3Rt why dont you paste to us a cli output and your config files to see what you wrote to us? |
| 01:49.38 | Rob3Rt | Alfio, ok, won't be too long, just remade my confs I may have removed something I shouldn't have, brb. |
| 01:49.39 | [TK]D-Fender | Rob3Rt: Well... I'm not disagreeing with you... this was an opportunity for you to show what you're doing. So far you should know all we can do is take your word for it. |
| 01:50.14 | [TK]D-Fender | Rob3Rt: You should ask us what's wrong and then make us ask you to show us what you're doing. You've got things backwards. |
| 01:50.54 | [TK]D-Fender | Rob3Rt: Just like you tell your mechanic about the grinding noise when braking BEFORE asking him to "fix it" so he doesn't take apart your motor first wondering if thats whats wrong. |
| 01:50.58 | Rob3Rt | [TK]D-Fender, no problem, sorry mate, this thing stesses me out, Im not new to configuration or coding by any means lol, but this is a foreign concept to me. Apologies, not trying to rouse on you. |
| 01:51.02 | Rob3Rt | brb |
| 01:52.21 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
| 01:53.10 | [TK]D-Fender | Rob3Rt: Foreign is fine. When I have OS issues my first statement is like "I'working with XYZ version 123. I'm trying to get FGH working, and getting the error in pastebin JKL. I've included my configs. It says that QRS is wrong, but it looked like I set it right in the PB." |
| 01:53.17 | *** join/#asterisk vegbox (n=kevinle@adsl-64-173-83-182.dsl.lsan03.pacbell.net) |
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| 01:53.29 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:53.46 | vegbox | How do I make a x100p card answer calls with an IVR? |
| 01:53.50 | [TK]D-Fender | Rob3Rt: Clarify, and always show the dead body before asking for an autopsy :) |
| 01:54.09 | [TK]D-Fender | vegbox: answer, play a prompt, ask for input. |
| 01:55.10 | Rob3Rt | Ahh, No body, No Murder type thing :D |
| 01:55.18 | Rob3Rt | No probs, be back in a bit . |
| 01:56.16 | *** join/#asterisk rosscoh (n=rosscoh@212.sub-75-220-194.myvzw.com) |
| 01:56.26 | vegbox | [TK]D-Fender: hmmm, how much more complicated would be to set in special hours? |
| 01:57.12 | rosscoh | did a backup and restore on a new asterisknow system...only problem is that it seems voicemails aren't working anymore and when I go to record a new message it gives about less then a second to record |
| 01:57.18 | rosscoh | does anyone have any ideas what could be wrong? |
| 01:57.43 | Alfio | vegbox you can use GotoIfTime but you need to makesome research |
| 01:58.06 | *** join/#asterisk ozziffied (n=ozziffie@173-9-137-118-Miami.FL.hfc.comcastbusiness.net) |
| 01:58.20 | [TK]D-Fender | vegbox: Depending.... a line or two |
| 01:58.52 | [TK]D-Fender | rosscoh: No. Pastebin a failed call at verbose 10 |
| 01:59.11 | [TK]D-Fender | ~pb |
| 01:59.12 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 01:59.12 | rosscoh | not sure what you mean? |
| 01:59.15 | [TK]D-Fender | ^^^^^^^^ |
| 02:00.14 | rosscoh | did a backup and restore on a new asterisknow system |
| 02:00.15 | *** join/#asterisk Kumbang (n=khauri@167.205.24.69) |
| 02:00.20 | rosscoh | voicemail doesn't seem to be working now |
| 02:01.01 | rosscoh | can't record a message...it just beeps and then immediately prompts to listen to the vmail |
| 02:01.26 | [TK]D-Fender | rosscoh: No. Pastebin a failed call at verbose 10 <-------------------- |
| 02:01.33 | vegbox | In the logs it says, Detected Alarm on Channel 1: Red ALarm |
| 02:01.50 | rosscoh | no idea what any of this means |
| 02:01.56 | [TK]D-Fender | vegbox: perhaps the line isn't plugged into the right port... |
| 02:05.36 | rosscoh | Is there no one that can help me? |
| 02:06.16 | ozziffied | rosscoh: [TK]D-Fender is the man |
| 02:06.23 | vegbox | whats the command to ask for input? |
| 02:06.26 | ozziffied | if he can't help not many can |
| 02:06.35 | rosscoh | I believe he is |
| 02:06.44 | rosscoh | I posted in pastebin.com |
| 02:06.49 | ozziffied | I can try but I am no genius |
| 02:06.50 | rosscoh | now how do I display it in the channel? |
| 02:06.58 | ozziffied | what the pastebin link? |
| 02:07.19 | ozziffied | copy and pate the link.... one line at a time |
| 02:07.22 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 02:07.28 | rosscoh | http://pastebin.com/m6b29cd08 |
| 02:07.54 | rosscoh | Thank you I'm greatly appreciative |
| 02:08.09 | rosscoh | sorry about my long post in the channel |
| 02:08.16 | rosscoh | I've never heard of postbin b4 |
| 02:08.16 | ozziffied | umm |
| 02:08.26 | ozziffied | you imported you settings from the old system? |
| 02:08.36 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 02:08.36 | ozziffied | is this a live system? |
| 02:08.44 | [TK]D-Fender | rosscoh: thats why I had the bot link you the explanation and left the giant line of arrows to point it out :) |
| 02:08.45 | maxagaz | is there a log of this channel somewhere ? |
| 02:09.04 | rosscoh | Fender, yes sorry about that |
| 02:09.09 | [TK]D-Fender | rosscoh: and I asked you to pastebin the CLI OUTPUT of a failed call so we can see what is HAPPENING. |
| 02:09.13 | maxagaz | i lost the answer to a question 14 hours ago |
| 02:09.25 | [TK]D-Fender | maxagaz: What was the question? |
| 02:09.34 | rosscoh | Fender the calls are working fine |
| 02:09.36 | [TK]D-Fender | maxagaz: Logs only update once daily |
| 02:09.37 | maxagaz | [TK]D-Fender, when i call an external number using an SIP phone, and that they ask me to choose for a language, the digits don't work, can someone help me to solve this problem ? |
| 02:09.38 | rosscoh | voicemails are not |
| 02:09.48 | [TK]D-Fender | rosscoh: .... I want to SEE IT FAIL |
| 02:10.08 | vegbox | if i made a custom recording do i just do Playback(Recording-Name) |
| 02:10.18 | ozziffied | [TK]D-Fender: I agree with you, but it sounds like an import old settings problem |
| 02:10.40 | rosscoh | sorry can you give me a short explanation of what you want me to post regarding the failure? Where would I get a log of a failed voicemail? |
| 02:11.02 | [TK]D-Fender | ozziffied: Sorry, I have no trustworthy details and I have no idea how its configured so I'm not going to just to conclusions until I see actual CLI output |
| 02:11.04 | ozziffied | asterisk -vvvr I think |
| 02:11.26 | [TK]D-Fender | rosscoh: * CLI. I've said ti a few times. You know... where you actually see your calls being processed <--- |
| 02:11.56 | [TK]D-Fender | * CLI = Open the hood and look at the damn engine turning :) |
| 02:12.00 | ozziffied | rosscoh: what distro are you running? |
| 02:12.06 | rosscoh | asterisknow 1.5 |
| 02:12.22 | ozziffied | then you should be in #asterisknow |
| 02:12.47 | [TK]D-Fender | ozziffied: Not yet |
| 02:13.04 | [TK]D-Fender | ozziffied: Its a config issue, not a distro issue so far |
| 02:13.05 | ozziffied | [TK]D-Fender: ok sorry, I will shut up |
| 02:13.13 | [TK]D-Fender | ozziffied: Colelct evidence, then defer if needed |
| 02:13.26 | rosscoh | appreciate that fender...working on the output |
| 02:14.12 | [TK]D-Fender | ozziffied: Don't need to be too defensive on it. I understand you want to help so I'm helping you understand where that line gets drawn. |
| 02:14.39 | [TK]D-Fender | ozziffied: Always properly identify what side of the fence the issue is on, then move forward |
| 02:14.50 | ozziffied | [TK]D-Fender: kk |
| 02:14.56 | ozziffied | * nods |
| 02:15.04 | vegbox | [TK]D-Fender: i am looking for the dialplan command to ask for a prompt, so that i can direct people to the intended voicemail box |
| 02:15.04 | [TK]D-Fender | ozziffied: He's shown nothign, so there isn't more to say than "Hey, I asked you for backup of your proble twice!" |
| 02:15.07 | vegbox | but i cant find it |
| 02:16.01 | [TK]D-Fender | vegbox: You should set TIMEOUTS first ("core show function TIMEOUT") and then use WaitExten ("core show application WaitExten") |
| 02:16.28 | rosscoh | to be honest I'm still trying to figure out how to output the errors |
| 02:17.01 | ozziffied | rosscoh: can you telnet/ssh into the machine? |
| 02:17.12 | rosscoh | yes |
| 02:17.14 | ozziffied | ok |
| 02:17.20 | ozziffied | once in their |
| 02:17.24 | vegbox | Hmmmm |
| 02:17.33 | ozziffied | asterisk -vvvvr |
| 02:17.36 | rosscoh | got it |
| 02:17.40 | ozziffied | and then do you vm screwup thingy |
| 02:17.54 | ozziffied | and copy the crap that flies across the window to patebin |
| 02:17.59 | ozziffied | and then give the link to the pastebin |
| 02:18.46 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 02:20.46 | ozziffied | rosscoh: ok? |
| 02:22.39 | rosscoh | yes trying to ssh in I'm not in the office right now |
| 02:22.46 | vegbox | when i do waitext, and people type in a number |
| 02:22.50 | rosscoh | configuring the router accordingly to dmz to the ip |
| 02:22.52 | vegbox | what variable does that get stored in to? |
| 02:22.54 | rosscoh | of the box |
| 02:22.58 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:24.44 | [TK]D-Fender | vegbox: it jumps to an extension in that context that matches what is dialed. |
| 02:25.06 | vegbox | hmm i want a user to be able to dial 120, and go straight to 120s vm box |
| 02:25.29 | [TK]D-Fender | vegbox: then make a "120" exten that does jsut that |
| 02:25.38 | vegbox | oh |
| 02:25.39 | vegbox | OHH |
| 02:25.48 | vegbox | thats genius |
| 02:25.48 | vegbox | lol |
| 02:26.37 | vegbox | it be just like, exten => 120,1,VoiceMail()? |
| 02:26.51 | [TK]D-Fender | vegbox: largely, yes |
| 02:26.59 | Rob3Rt | Does anyone here run a Linksys WRT54GP2-NA |
| 02:27.03 | Rob3Rt | with Aster >? |
| 02:27.24 | vegbox | would it be VoiceMail(300)? |
| 02:27.26 | vegbox | right? |
| 02:29.27 | vegbox | Now if I want to be able to have them enter in a extension at any time, i use the background and not playback? |
| 02:30.09 | [TK]D-Fender | vegbox: Correct, doing good... |
| 02:30.17 | vegbox | ohh shit! hahaha |
| 02:30.44 | vegbox | how come i need quicktime to play back voice mails |
| 02:31.26 | [TK]D-Fender | vegbox: O RLY? Playback from where? How? |
| 02:31.34 | vegbox | like on the web portal |
| 02:31.39 | vegbox | i log in with my ext. number and secret |
| 02:31.40 | [TK]D-Fender | vegbox: See I listen to my VM's.... on my PHONE |
| 02:31.48 | vegbox | oh |
| 02:32.06 | [TK]D-Fender | vegbox: Words can hardly express how much GUI's are NOT supported here <- |
| 02:32.14 | vegbox | yeah |
| 02:32.21 | vegbox | but i want these emails vms sent to me |
| 02:32.21 | vegbox | hehe |
| 02:32.29 | [TK]D-Fender | vegbox: And if Quicktime thinks its the boss of things on your system.... its because you LET IT :p |
| 02:32.35 | vegbox | hahah true |
| 02:32.40 | [TK]D-Fender | VLC <3 |
| 02:35.22 | vegbox | Can I somehow debug what extension i am calling from |
| 02:35.24 | vegbox | in the dial plan? |
| 02:37.47 | [TK]D-Fender | vegbox: Couple of things to consider : caller-id & the channel-name |
| 02:38.03 | [TK]D-Fender | vegbox: and that's "device I'm calling from". |
| 02:38.15 | vegbox | well these are old NEC analog phone systems |
| 02:38.22 | [TK]D-Fender | vegbox: An extension is jsut a number you dial, not the action it happens to take. |
| 02:38.26 | vegbox | i am just trying to replace our old dos voicemail box |
| 02:38.46 | [TK]D-Fender | vegbox: Ah yes, I remember now... |
| 02:39.09 | vegbox | yup thats me :) |
| 02:39.40 | vegbox | there is a bug in the dos voicemail system |
| 02:39.57 | [TK]D-Fender | vegbox: Your call is coming in a dumb analog line IIRC for which all you have is the callerID (if the NEC even provides it), and any digits the PBX might happen to pass before bridging. The latter is unlikely, but would ahve been nice. |
| 02:40.22 | vegbox | so how can i check if it produces a callerID? |
| 02:40.57 | [TK]D-Fender | vegbox: Look at it <- |
| 02:41.20 | vegbox | the logs you mena? |
| 02:41.37 | vegbox | or are you telling to put in a function in my conf file |
| 02:41.42 | vegbox | to see whats goign on? |
| 02:43.17 | [TK]D-Fender | vegbox: Ok, just because your current plans are small I'll jsut hand this one to you : NoOp(the CID is "${CALLERID(name)}" <${CALLERID(num)}>) |
| 02:43.31 | [TK]D-Fender | vegbox: Make that the 1st priority you process on your incoming call. |
| 02:43.52 | [TK]D-Fender | vegbox: You'll be able to see in CLI what * receives. |
| 02:45.11 | vegbox | trying now |
| 02:45.39 | vegbox | haha man its blank |
| 02:45.41 | vegbox | for both |
| 02:46.17 | *** join/#asterisk [Outcast] (n=asdf@ip202-27-193-197.cust.kci.net.nz) |
| 02:47.04 | [TK]D-Fender | vegbox: You need to be checking for CID in the first place. Does it seem to wait 2 rings before * picks up? |
| 02:47.13 | vegbox | yeah |
| 02:47.27 | vegbox | three rings then i get the answer |
| 02:48.02 | vegbox | i have it as s,1,noop then s,2,answer |
| 02:48.20 | [TK]D-Fender | vegbox: Ok, then that answers that. * waits, it never gets sent |
| 02:48.33 | [TK]D-Fender | vegbox: Describe the size of your NEC setup |
| 02:48.42 | vegbox | Its a NEC Electra System |
| 02:48.46 | vegbox | about 16 phones here |
| 02:48.56 | vegbox | it has 8 cards for 8 calls in/out at the same time |
| 02:48.58 | [TK]D-Fender | vegbox: How many phones, what type, how many lines, what type |
| 02:49.13 | [TK]D-Fender | vegbox: So 8 lines, 16 phones? |
| 02:49.20 | vegbox | yeah, they all share |
| 02:49.26 | vegbox | if all 8 are filled up you get no dial tone |
| 02:49.27 | vegbox | lol |
| 02:49.39 | vegbox | outgoing dial tone |
| 02:49.48 | vegbox | you know after you dial 9, you can still do inter office calls |
| 02:50.03 | [TK]D-Fender | vegbox: You know... this would cost so very little to completely replace with * :) |
| 02:50.31 | vegbox | thats what everyone says, get a pri and boom |
| 02:50.48 | vegbox | but the owners here are feeling the Souther California Crunch |
| 02:50.58 | vegbox | ie, 10 percent tax, hollywood isnt filming here anymore |
| 02:51.20 | vegbox | Basically what I want to do is, This |
| 02:51.44 | vegbox | When people outside call our number after 530pm, it goes to a closed message and you can enter in a vmail box any time to leave a voicemail |
| 02:52.07 | *** part/#asterisk ozziffied (n=ozziffie@173-9-137-118-Miami.FL.hfc.comcastbusiness.net) |
| 02:52.16 | vegbox | During the day, when employees dial 250, it should ask for a voicemail box then their password |
| 02:52.20 | vegbox | to check their mail |
| 02:52.50 | maxagaz | where can one get the logs of this channel ? |
| 02:52.52 | [TK]D-Fender | vegbox: Sure why not... |
| 02:53.03 | vegbox | but thats only for calls made inside the building , to check vmails hehe |
| 02:53.14 | vegbox | maxagaz: no clue |
| 02:53.15 | vegbox | bash? |
| 02:53.16 | vegbox | heheh |
| 02:53.44 | *** join/#asterisk ctp (n=quassel@brsg-d9beec9e.pool.mediaWays.net) |
| 02:55.03 | [TK]D-Fender | maxagaz: http://ibot.rikers.org/%23asterisk/ |
| 02:55.47 | [TK]D-Fender | maxagaz: And for your problem, you have a wrong DTMF mode set |
| 02:56.13 | maxagaz | [TK]D-Fender, thanks! |
| 02:59.45 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 03:00.59 | maxagaz | [TK]D-Fender, in which file should it be changed ? |
| 03:01.05 | vegbox | if i use background to play my pseudo ivr |
| 03:01.12 | vegbox | do i still need to use waitexten? |
| 03:02.13 | [TK]D-Fender | vegbox: Yes, unless you disable autofallthrough at which point the typical "run out of priorities in "s" takes effect |
| 03:02.50 | [TK]D-Fender | maxIf you are dealing with SIP, its in there. Either your device's config, or your providers if using an ITSP, etc |
| 03:02.50 | *** join/#asterisk [Outcast] (n=asdf@ip202-27-193-197.cust.kci.net.nz) |
| 03:02.53 | vegbox | oh shit! |
| 03:02.54 | vegbox | this is dope |
| 03:02.56 | vegbox | too dope |
| 03:02.56 | vegbox | hahaha |
| 03:03.01 | vegbox | i need more x100p cards |
| 03:03.10 | vegbox | is there anything out there that has 3 or 4 ports? |
| 03:03.19 | jaytee | yep |
| 03:03.26 | jaytee | or 8 even |
| 03:03.31 | jaytee | or 24 |
| 03:03.39 | vegbox | thats on the cheap? |
| 03:03.40 | vegbox | lol |
| 03:03.47 | jaytee | no, not cheap |
| 03:03.52 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
| 03:04.00 | vegbox | x100p cards are 30 bucks |
| 03:04.05 | vegbox | from that x100p website |
| 03:04.18 | jaytee | and worth less than 5 bucks |
| 03:04.23 | vegbox | haha |
| 03:04.24 | vegbox | fine |
| 03:04.27 | jaytee | and perform like shit |
| 03:04.31 | [TK]D-Fender | offers $2. |
| 03:04.37 | *** join/#asterisk Sargun_screen (n=sargun@atarack/Staff/Sargun) |
| 03:04.38 | Sargun_screen | Key: "Event", Value: "Hangup" |
| 03:04.40 | jaytee | wow, big spender! |
| 03:04.41 | vegbox | any recs for good 4 ports? |
| 03:04.42 | Sargun_screen | Key: "Event", Value: "StatusComplete" |
| 03:04.54 | Sargun_screen | Which one of these two events should one shutdown handling of a phone call. |
| 03:04.56 | vegbox | now i just need to get the Timing of the day correct |
| 03:04.58 | [TK]D-Fender | vegbox: vegbox It's scary you |
| 03:04.59 | jaytee | Digium or Sangoma, don't waste time on other brands |
| 03:05.09 | vegbox | what does its scary you mean? |
| 03:05.19 | [TK]D-Fender | vegbox: It'd scare you* |
| 03:05.24 | [TK]D-Fender | (the price) |
| 03:05.24 | vegbox | oh 250? |
| 03:05.31 | [TK]D-Fender | ger450 ~ |
| 03:05.37 | vegbox | ger? |
| 03:05.38 | [TK]D-Fender | vegbox: $450~~ |
| 03:05.40 | vegbox | hmmm |
| 03:05.43 | vegbox | what brands? |
| 03:05.47 | [TK]D-Fender | holy crap my typing has gone completely AWOL |
| 03:05.58 | [TK]D-Fender | vegbox: Sangoma B600d |
| 03:06.26 | jaytee | looks like someone isn't going to be snatching any pebbles from Master Po's hand tonight for sure :-) |
| 03:06.51 | vegbox | so dont get stuff like the tdm400p? |
| 03:07.15 | jaytee | I've used a TDM04B card with satisfactory results |
| 03:07.31 | jaytee | which is the TDM400p card with 4 FXO modules |
| 03:07.37 | vegbox | i mean im going from dos voicemail box |
| 03:07.48 | vegbox | so im sure these ISA FXO cards are not that steller |
| 03:08.01 | jaytee | ISA? WTF? |
| 03:08.21 | jaytee | you mean PCI don't you? please tell me you mean PCI |
| 03:08.36 | vegbox | no |
| 03:08.37 | vegbox | ISA man |
| 03:08.46 | vegbox | its a really really old voicemail computer |
| 03:08.49 | vegbox | like 1994 i believe |
| 03:08.52 | jaytee | what kind of antique system you running on? |
| 03:08.56 | vegbox | i still still lurking around in junior high |
| 03:09.02 | vegbox | Replay Voice Mail 1.6! |
| 03:09.07 | vegbox | the monitor is GREEN |
| 03:09.11 | vegbox | not even in color |
| 03:09.12 | vegbox | hahaha |
| 03:09.21 | jaytee | got a shotgun? a sledgehammer? |
| 03:09.28 | vegbox | i got a machete |
| 03:09.29 | vegbox | haha |
| 03:09.37 | vegbox | but yeah i am thinking the tdma400p |
| 03:09.43 | vegbox | will be a UPGRADE upgrade |
| 03:09.47 | vegbox | compared to whats rocking right now |
| 03:09.49 | jaytee | which is PCI so you are SOL |
| 03:09.54 | [TK]D-Fender | "Leave it to Cleaver" |
| 03:10.00 | vegbox | jaytee: |
| 03:10.10 | vegbox | no cause i am in the process of getting this asterisk box going |
| 03:10.19 | vegbox | so taht i can process voicemails on asterisk |
| 03:10.20 | vegbox | soon |
| 03:10.51 | vegbox | i just need to figure out how to do the time schedule on the dial plan thing |
| 03:10.52 | jaytee | and the * box is a newer computer with PCI slots I hope? |
| 03:10.58 | vegbox | yeah |
| 03:11.02 | vegbox | pentium 4 HT! |
| 03:11.02 | vegbox | lol |
| 03:11.02 | jaytee | whew! |
| 03:11.07 | *** join/#asterisk MindTheGap_ (n=MindTheG@189.59.177.210.dynamic.adsl.gvt.net.br) |
| 03:11.28 | jaytee | thought you were going to try to load linux and asterisk on that old dinosaur. |
| 03:11.33 | *** join/#asterisk [Outcast] (n=asdf@ip202-27-193-197.cust.kci.net.nz) |
| 03:11.44 | vegbox | i believe its a 486sx |
| 03:11.45 | jaytee | which would have been pointless |
| 03:11.46 | vegbox | cpu |
| 03:11.49 | vegbox | so it could run linux |
| 03:11.51 | vegbox | lol |
| 03:12.19 | jaytee | I don't think anyone makes ISA fxo cards. if they do they should be shot or shut down by the FTC |
| 03:13.38 | vegbox | im sure some kid in china |
| 03:13.42 | KavanS | saw the words "ISA" |
| 03:13.44 | vegbox | is making them and has not gotten the update |
| 03:13.46 | KavanS | *word |
| 03:13.56 | KavanS | someone mentioned ISA fxo cards? |
| 03:14.07 | jaytee | yeah |
| 03:14.09 | jaytee | someone |
| 03:14.15 | KavanS | hehehe...I lol'd when I saw that |
| 03:14.41 | jaytee | I was cringing when I typed it |
| 03:14.45 | vegbox | dont lol |
| 03:14.47 | vegbox | help me |
| 03:14.50 | vegbox | figure out this dial plan |
| 03:14.51 | vegbox | lol |
| 03:14.56 | jaytee | help you do what? |
| 03:14.58 | KavanS | vegbox, you can get a pentium for like 30$ these days |
| 03:15.16 | vegbox | man thats steep haha |
| 03:15.18 | KavanS | hell the whole box should be cheaper than your FXO (correct me if I'm wrong) |
| 03:15.21 | KavanS | steep? lol |
| 03:15.25 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 03:15.26 | KavanS | are you in uganda or something? |
| 03:15.29 | vegbox | no |
| 03:15.32 | vegbox | hollywood, ca |
| 03:15.33 | vegbox | haha |
| 03:15.40 | KavanS | oh shit |
| 03:15.57 | vegbox | i got pay 3 dollars in taxes |
| 03:16.00 | vegbox | for this so called pentium |
| 03:16.01 | vegbox | lol |
| 03:16.03 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 03:16.33 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 03:16.46 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.97) |
| 03:16.56 | KavanS | heh vegbox is this the first time you've played with asterisk? |
| 03:17.00 | KavanS | it is a lot of fun... |
| 03:17.04 | vegbox | yes |
| 03:18.50 | vegbox | well thanks everyone for the hlep |
| 03:19.01 | vegbox | ill be back thursday to harrass you guys some more |
| 03:19.05 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 03:19.09 | KavanS | nice, see ya then! |
| 03:19.44 | KavanS | time to call it a day for me... |
| 03:21.50 | xpot-mobile | Question: I am still having issues with call queues. When I press '0' to get to voicemail I see a script 'dialparties.agi' execute. This happens for as many times as I punch the '0' button. Anyone know why this agi script would execute instead of the context= value that I have listed in queues.conf ?? |
| 03:23.24 | [TK]D-Fender | xpotPASTEBIN<_ |
| 03:24.53 | *** join/#asterisk Rob3Rt (n=admin@87.37.96.58.static.exetel.com.au) |
| 03:25.50 | xpot-mobile | [TK]D-Fender: http://pastebin.com/d5da8bdce |
| 03:27.23 | [TK]D-Fender | xpot-mobile: I don't see Queue() being caleld in there at all. |
| 03:27.37 | [TK]D-Fender | xpot-mobile: Nor your queue config |
| 03:29.33 | [TK]D-Fender | xpot-mobile: I also don't see a single-digit exten being executed in there |
| 03:30.48 | xpot-mobile | [TK]D-Fender: it looks like my terminal buffer wasn't high enough to get the initial incoming call. The pastbin I uploaded is what floods the cli while in queue listening to MOH. I notice that when I do not hit anything this 'dialparties.agi' executes on its own in a slower fashion. When I press a number on the key pad of my phone, it seems to re-execute this script. |
| 03:31.25 | *** join/#asterisk empiric (n=empiric@124.29.242.138) |
| 03:31.49 | xpot-mobile | [TK]D-Fender: going to get extensions.conf and queues.conf in pastebin for you if your interested |
| 03:32.31 | [TK]D-Fender | I want CLI output.... |
| 03:32.48 | [TK]D-Fender | I'll ask for dialplan when I'm feeling masochistic |
| 03:33.17 | xpot-mobile | [TK]D-Fender: :) ok |
| 03:34.36 | *** join/#asterisk R0b3Rt (n=admin@87.37.96.58.static.exetel.com.au) |
| 03:34.47 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:38.28 | xpot-mobile | [TK]D-Fender: here is a pastebin that should include all cli from call start to call end. http://pastebin.com/d21de4b7c |
| 03:38.55 | *** join/#asterisk s14ck (n=s14ck@190-76-108-33.dyn.movilnet.com.ve) |
| 03:42.00 | [TK]D-Fender | xpot-mobile: -- Nobody picked up in 0 ms <-- doesn't look normal. Nor do I see any indication of a signe digit escape attempt in there |
| 03:42.13 | [TK]D-Fender | xpot-mobile: And you still didn't provider your config. |
| 03:43.07 | empiric | Fender i have 4 PSTN lines and 6 extension |
| 03:43.11 | xpot-mobile | [TK]D-Fender: you said you only wanted cli until you felt masochistic?? |
| 03:43.36 | empiric | for 1 extension i want to fix that this ext will dial from a fixed PSTN from those 4 lines |
| 03:43.46 | empiric | how it would be done in asterisk any idea? |
| 03:43.57 | [TK]D-Fender | xpot-mobile: I didn't ask for EXTENSIONS.CONF. I asked for your QUEUE CONFIG. Repeatedly |
| 03:44.16 | [TK]D-Fender | empiric: Dial that channel specifically |
| 03:44.21 | xpot-mobile | [TK]D-Fender: oh, I must have missed that. ;) getting it... |
| 03:45.23 | empiric | how |
| 03:45.35 | empiric | how i wrtie my dial plan |
| 03:47.05 | [TK]D-Fender | empiric: Go look at how you dial out the interface already |
| 03:54.11 | empiric | what u mean |
| 03:54.35 | empiric | # |
| 03:54.36 | empiric | exten => _XXX.,1,Answer() |
| 03:54.36 | empiric | # |
| 03:54.36 | empiric | exten => _XXX.,2,MixMonitor(${TIMESTAMP}-${EXTEN}-${UNIQUEID}.wav) |
| 03:54.36 | empiric | # |
| 03:54.36 | empiric | exten => _XXX.,3,Dial(SIP/200/${EXTEN}) |
| 03:54.43 | empiric | this is my dial plan to dial out |
| 03:54.55 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 03:54.56 | empiric | its Dlink FXO configured with SIp accounts |
| 03:55.01 | xpot-mobile | [TK]D-Fender: Here is the pastebin for queues http://pastebin.com/d741ddd47 |
| 03:55.24 | empiric | my dial plan http://pastebin.com/m794105c2 |
| 03:57.40 | *** join/#asterisk Kumbang (n=groove@rusnas.paume.itb.ac.id) |
| 04:00.24 | [TK]D-Fender | empiric: ... you are just dialing a SIP peer. What gives you the impression you have a COICE where the remote PBX sends it? |
| 04:00.29 | [TK]D-Fender | CHOICE* |
| 04:01.10 | [TK]D-Fender | xpot-mobile: I still don't see a single digit escape attempt, and I'd want to see that IVR dialplan... |
| 04:01.10 | empiric | my FXO just needs a sip account |
| 04:01.23 | empiric | i have 4 lines and 4 sip acounts are configured |
| 04:01.28 | empiric | for dial out |
| 04:01.59 | [TK]D-Fender | empiric: Go read your device's manual |
| 04:02.28 | empiric | ii have |
| 04:02.34 | empiric | its Dlink DVS-300S |
| 04:02.46 | empiric | what u r trying to say |
| 04:04.45 | [TK]D-Fender | empiric: What I am saying is no-one here uses that device. The Linksys equivalent of it doesn't even let you pick a specific port. I do not know for sure if it supports that option. |
| 04:05.32 | empiric | agreed |
| 04:05.50 | empiric | see each Dlink line is configured with a SIP account |
| 04:05.58 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:06.07 | empiric | let say 200,201 ,202,203 are for Dlikn |
| 04:06.20 | empiric | now i want 204 to dial out only from 200 |
| 04:08.19 | xpot-mobile | [TK]D-Fender: Here is the ivr dialplan and only the pieces it refers to. http://pastebin.com/d215849a1 |
| 04:09.30 | xpot-mobile | [TK]D-Fender: I am pressing a digit, what test do you suggest I perform to get the digit escape attempt? It seems to only run that crazy agi script when I press any digit. |
| 04:12.41 | [TK]D-Fender | xpot-mobile: Right now I see the call enter the queue, and call out Agent '100" twice. That is all |
| 04:12.50 | [TK]D-Fender | xpot-mobile: nothign about it trying to exit |
| 04:13.14 | [TK]D-Fender | empiric: then put it in a context that only has extens that dial out it |
| 04:13.32 | xpot-mobile | [TK]D-Fender: I will dial in again and press '0' a few times... |
| 04:13.33 | empiric | well yes exactly |
| 04:13.38 | empiric | am thinkning |
| 04:13.44 | empiric | can u guide me how |
| 04:14.03 | empiric | my current dial plan http://pastebin.com/m794105c2 |
| 04:15.07 | xpot-mobile | empiric: you could try creating a context called [dialout] with exten => s,1,Dial() and wherever you set your context for 204 you can do a exten => s,1,GotoIf($[${EXTEN}=200]?dialout,s,1) or similar *NOTE: you may want to check syntax |
| 04:15.47 | empiric | ok wait let me check |
| 04:16.19 | [TK]D-Fender | empiric: You are shoving everything into one messy context and pointing everything to it. amke SEPARATE contexts that only do what each should be allowed to |
| 04:16.31 | empiric | ok |
| 04:17.13 | empiric | ok |
| 04:20.58 | *** join/#asterisk ariel_ (n=chatzill@96.49.119.119) |
| 04:22.46 | xpot-mobile | [TK]D-Fender: here is another call attempt, I pressed 0 multiple times and I still only see a script execute. FYI: while in the IVR playing custom/StatupNight etc. When I press a value to select my choice those digits do not show up in cli, however I do end up going to my intended location. ?? http://pastebin.com/d76050a80 |
| 04:24.17 | [TK]D-Fender | xpot-mobile: Nowhere in there do I see it jumping out. |
| 04:24.40 | [TK]D-Fender | xpot-mobile: Where do YOu see it doing so? |
| 04:25.37 | xpot-mobile | [TK]D-Fender: for example see lines 42-48... I press 1 to hear StartUpPrice (line 47) and it goes to my indicated option and plays custom/StartUpPrice |
| 04:26.59 | [TK]D-Fender | xpot-mobile: -- Executing [s@ivr-4:1] Set("IAX2/voip-2280", "MSG=custom/StartUpPrice") in new stack |
| 04:27.09 | [TK]D-Fender | xpot-mobile: I don't see that call ENTERING THE QUEUE |
| 04:27.29 | [TK]D-Fender | xpot-mobile: So far I just see that call navigating an IVR. |
| 04:27.47 | xpot-mobile | [TK]D-Fender: that is after, call enters queue when I hit 0 after hearing StartUpPrice... let me see if I can find the queue entry |
| 04:28.09 | [TK]D-Fender | xpot-mobile: it DIDN'T |
| 04:29.02 | xpot-mobile | [TK]D-Fender: lines 61-65 is where I hit 0 to enter queue... it play MOH... maybe it's not hitting queue for some reason?? |
| 04:29.35 | [TK]D-Fender | xpot-mobile: http://pastebin.com/d76050a80 <-- WHERE? |
| 04:32.18 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 04:33.00 | xpot-mobile | [TK]D-Fender: I missed a block when I copied. Found it --> http://pastebin.com/d2c45df7 |
| 04:34.03 | xpot-mobile | [TK]D-Fender: lines 14-20 looks like the queue process |
| 04:34.07 | [TK]D-Fender | xpot-mobile: Missed? that is a completely diffferent call |
| 04:34.35 | xpot-mobile | well yeah, my buffer didn't go back far enough for the other call, but I did the exact same thing |
| 04:34.59 | xpot-mobile | is that a cli >> file.txt option is asterisk? |
| 04:35.03 | [TK]D-Fender | xpot-mobile: And in that last PB it sure as shit didn't ewxit the queue |
| 04:36.21 | xpot-mobile | [TK]D-Fender: my apologies for the inconsistency. can I do some sort of # asterisk -rx " " >> file.txt ?? so my copy paste errors don't screw things up |
| 04:36.58 | [TK]D-Fender | xpot-mobile: Come back when you're competant enough to copy & paste |
| 04:37.33 | xpot-mobile | [TK]D-Fender: thanks smart-ass :) |
| 04:38.28 | [TK]D-Fender | xpot-mobile: You've wasted a phenomenal amount of my time now with these wild goose chases already. There is nothing "smart-ass" about it |
| 04:38.39 | ariel_ | xpot-mobile: I think he is being nice since he was trying to help you here with an Freepbx issue and from your last post the queue sent the call to agent 100 |
| 04:39.26 | [TK]D-Fender | Indeed this isn't #freepbx but I was willing to look if you were at least capable of C&P-ing a stupid little bit of CLI output. |
| 04:39.43 | xpot-mobile | [TK]D-Fender and ariel_: I completely understand the help and it is much appreciated. the "smart-ass" was denoted with a --> :) to show a joking remark. |
| 04:40.28 | [TK]D-Fender | This should have been a *#$&^ing 1 minute search. |
| 04:41.26 | xpot-mobile | [TK]D-Fender: you have to at least realize that I was not trying to waste your time.. it's a pain in the rear trying to copy a slew of cli output without knowing how to cat it from asterisk to a file (which would have been the easiest and fastest had I known how to do it).. My apologies for wasting your time. |
| 04:42.27 | [TK]D-Fender | xpot-mobile: SSH in with a real friggen client witha big buffer. Connect. place call from beginning to end. exit CLI immdeatly after. Copy entire buffer. End of story |
| 04:45.40 | ariel_ | xpot-mobile: with putty you can send all items to a file, secure crt also, That is if your using windows. |
| 04:46.09 | ariel_ | [TK]D-Fender: don't you sleep? or get any time off? |
| 04:46.30 | xpot-mobile | ariel_: not in windows but I do have putty via wine |
| 04:46.34 | [TK]D-Fender | ariel_: Or Linux. |
| 04:46.39 | [TK]D-Fender | ariel_: Or BSD. |
| 04:46.42 | [TK]D-Fender | ariel_: Or whatever |
| 04:46.45 | ariel_ | there is a putty for linux |
| 04:47.01 | ariel_ | lol |
| 04:47.17 | WindowsUser | xpot-mobile: what term are you using? |
| 04:47.44 | xpot-mobile | WindowsUser: GnomeTerm 2.24 |
| 04:48.36 | WindowsUser | thats got configurable scrollback |
| 04:49.10 | xpot-mobile | WindowsUser: I am trying to find it now |
| 04:49.21 | ariel_ | can't watch my TV programs through the web up here in Canada...argh |
| 04:49.23 | WindowsUser | Edit -> Profile Preferences |
| 04:49.47 | xpot-mobile | WindowsUser: I found it, thank you |
| 04:49.57 | WindowsUser | ariel_: yup, gotta proxy via a us host |
| 04:50.36 | ariel_ | missed his Syfy shows... |
| 04:53.51 | ariel_ | ultra-vnc to my home ubuntu is not sending sound, sucks I can watch but can't hear the show... let me see if I can do a openvpn through my home box.. |
| 04:54.10 | WindowsUser | squid might be better |
| 04:54.41 | ariel_ | don't have that much time. I will be going back to sea tomorrow for a week... |
| 04:54.47 | WindowsUser | ah |
| 04:55.01 | xpot-mobile | [TK]D-Fender: If I have not frustrated you too much here is my cli output without buffer pebkac error ;) http://pastebin.com/d76647109 |
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| 04:58.50 | [TK]D-Fender | xpot-mobile: Nowhere does IAX2/voip-3974 ever exit the queue short of the hangup |
| 04:59.59 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 05:00.58 | WindowsUser | ariel_: http://pastebin.ca/1495439 |
| 05:02.51 | ariel_ | WindowsUser: Nice t/y, I got openvpn up and I am watching Eureka.....via my home pc.. |
| 05:04.06 | xpot-mobile | [TK]D-Fender: that is the problem... would you have any suggestions as to why, when I press a digit, it does not exit to the intended context, ie: ivr-6 ? It responds to the ivr flow, just will not exit the queue if I decide not to sit there on hold/MOH |
| 05:07.00 | [TK]D-Fender | xpotYou can only exit while it is not dialing an agent |
| 05:07.26 | WindowsUser | sweet |
| 05:08.23 | xpot-mobile | [TK]D-Fender: ah, so perhaps the time for agent retry is too fast? |
| 05:08.44 | [TK]D-Fender | xpot-mobile: Perhaps |
| 05:09.51 | xpot-mobile | [TK]D-Fender: ok, I will see if I can tweak that. I do appreciate your help and services. Please forgive me for the torment I put you through. It is out of love, I promise. ;) |
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| 05:32.12 | [TK]D-Fender | checkout time, later all |
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| 05:39.14 | PhunTelTek | . |
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| 06:40.12 | Rob3Rt | no offence |
| 06:40.24 | Rob3Rt | but this softwares getting the bin if i dont this bs sorted out |
| 06:40.38 | Rob3Rt | ffs been on it for 3 days and shit wont route no matter how i set it up |
| 06:40.40 | Rob3Rt | and now this, |
| 06:40.42 | Rob3Rt | Registration from '"2000"<sip:2000@10.0.0.5:5060>' failed for '10.0.0.5' - Username/auth name mismatch |
| 06:41.00 | Rob3Rt | wtf it thinks I cant assign it a stupid uname and pasS? |
| 06:41.29 | Rob3Rt | what next? random unauthorised calls to detroit? |
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| 07:39.04 | Rob3Rt | shoulda called it SPASTIX |
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| 08:47.59 | astswan | y as quelqu un ?? |
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| 08:52.07 | astswan | no one here ?? i need help!! |
| 08:58.26 | kaii | astswan: hello. this is an english channel and you need to ask something to get a response. |
| 08:58.55 | kaii | typing "/names" in your irc client will show that there are many people here. |
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| 09:01.47 | kaii | astswan: before you describe what you try, maybe you want to tell us something about your goal. what is it you want to achieve when simulating a queue member? |
| 09:03.35 | astswan | kaii: What i want to do is to give to agent the option to answer call witout pressing # |
| 09:04.30 | kaii | why do you want this? how should the agent answer then, if not using #? |
| 09:04.37 | astswan | agent mus not use softphone |
| 09:05.12 | astswan | <PROTECTED> |
| 09:05.24 | astswan | connected by asterisk manager interface |
| 09:05.32 | kaii | ah isee |
| 09:05.55 | kaii | but you need to send DTMF on the agents channel then |
| 09:06.01 | astswan | yes |
| 09:06.19 | astswan | i see asterisk code |
| 09:06.24 | kaii | modifying application senddtmf() seems not appropriate to me |
| 09:06.46 | astswan | it s was my first idea |
| 09:06.48 | kaii | you should implement an AMI command "AgentAnswer" |
| 09:06.59 | astswan | that i want to do |
| 09:07.14 | astswan | to do this |
| 09:07.33 | astswan | i think that i must put # in this strcut |
| 09:07.41 | kaii | no |
| 09:07.53 | astswan | like this f = ast_read(p->chan); |
| 09:07.53 | astswan | f->frametype = AST_FRAME_DTMF; |
| 09:07.53 | astswan | f->subclass='#'; |
| 09:08.05 | astswan | Why ?? |
| 09:08.19 | kaii | well maybe this will work |
| 09:08.20 | astswan | it like i simulate the # |
| 09:08.29 | kaii | but its not a direct solution |
| 09:08.35 | astswan | ok |
| 09:08.44 | astswan | what is your idea ? |
| 09:09.19 | kaii | do what pressing "#" does directly, without simulating '#' press |
| 09:09.32 | kaii | look after the code where '#' is handled and the agent answer is executed |
| 09:09.56 | kaii | do the same in your code |
| 09:10.54 | astswan | i think that it s here : function agent_ack_sleep |
| 09:11.03 | astswan | in chan_agent.c |
| 09:13.18 | astswan | have you see the cpde kaii ? |
| 09:14.12 | astswan | cod |
| 09:14.14 | astswan | code |
| 09:14.45 | kaii | im working on a massive sql statement today, no time for asterisk code |
| 09:15.29 | kaii | btw: this discussion is better for #asterisk-dev :-) sry for sending you here. your start sounded like it was a typical user question |
| 09:16.10 | astswan | sorry but i ant to descrip the situation and my english is not good |
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| 09:53.00 | ronr | hi, I'm trying to figure out how to make a .call file in outgoing to send and sms, I did find a SendText function but I can't really figure out what to do next, I'm probably reading the docs wrong here; could anyone help me by maybe pointing me to a step by step howto or providing an example .call file that uses SendText to send an sms? |
| 09:55.11 | kaii | ronr: SendMessage is for SIP messages, not SMS. there is an application SMS() for this. |
| 09:55.19 | kaii | ronr: you may want to have a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms |
| 09:55.29 | kaii | possibly "smsq" is what your are looking for. |
| 09:56.00 | ronr | ok, thx, I'll check those docs |
| 09:57.20 | ronr | kaii: any clue what options I got for someapptohandlelocalsms in the example dialplan for smsmorx? |
| 09:57.47 | kaii | ronr: you may want to start reading at the beginning of a page and ask questions when you reached the bottom. thx. |
| 09:58.00 | kaii | have a deeper look at "smsq" or even search for it on that page. |
| 10:02.31 | ronr | kaii: I'll go back to reading (and this time I won't stop at the first how to send an sms example ;) ) |
| 10:02.49 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) |
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| 10:30.20 | a2_ | Is it possible to spoof or change a DID number type from Landine/Prepaid/Wireless/VOIP etc..? |
| 10:33.04 | creativx | a2_: thats up to your provider |
| 10:37.03 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 10:37.03 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.25.1 (2009/06/05), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.2.0.1 (2009/06/30), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 10:37.28 | a2_ | creativx: Do you know how widely accepted XOR.com is? |
| 10:37.37 | a2_ | Do telco's use them in local areas? |
| 10:40.37 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 10:43.16 | astswan | hi how can i register new AMI Action command ? |
| 10:43.46 | astswan | Loadmodule is enough ?? |
| 10:49.01 | tzafrir_laptop | astswan, is it implemented in a separate module? |
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| 11:01.40 | astswan | no |
| 11:01.54 | astswan | it s implemented in chan_agent |
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| 11:07.10 | kombi | http://pastebin.se/198510 outgoing callerid truncated? |
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| 11:26.25 | astswan | http://pastebin.se/198512 warning when agent establish call |
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| 11:41.45 | shazaum | hi |
| 11:42.17 | beek | hi |
| 11:42.48 | shazaum | how can I clean the counters of a queue? |
| 11:43.17 | beek | module reload app_queue |
| 11:43.45 | shazaum | imagined it, but did not work |
| 11:43.50 | beek | of a single queue, I don't know. |
| 11:44.13 | shazaum | all queue |
| 11:44.53 | shazaum | well |
| 11:44.55 | shazaum | thks |
| 11:44.59 | shazaum | see ya |
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| 11:51.41 | mvanbaak | hhmm, 4.4% packetloss and best ping at 2.3ms and worst ping at 507.4 ms |
| 11:51.59 | mvanbaak | that's not the best line to do voip on correct ? |
| 11:53.04 | leifmadsen | hoi hoi! |
| 11:53.14 | leifmadsen | mvanbaak: not so much |
| 11:53.17 | mvanbaak | goede middag ! |
| 11:53.49 | mvanbaak | leifmadsen: it's the figures I have when running mtr on the asterisk box: mtr pbx.provider.net |
| 11:54.02 | leifmadsen | god morgen! |
| 11:54.11 | leifmadsen | mvanbaak: booo.... |
| 11:54.28 | mvanbaak | the line is rocksolid till the last hop |
| 11:54.54 | leifmadsen | ouch... bad router? |
| 11:54.57 | mvanbaak | 0.01% packetloss and best ping at 2.1 and worst ping at 2.2ms |
| 11:55.11 | leifmadsen | sounds like something is incorrectly configured at the last hop |
| 11:55.14 | leifmadsen | or bad wiring... ? |
| 11:55.19 | mvanbaak | I only see loss and variations on the pbx.provider.net line |
| 11:55.24 | leifmadsen | right, that's what I mean |
| 11:55.33 | leifmadsen | it must be localized |
| 11:55.40 | mvanbaak | and they tell me everything is fine and it's our problem :) |
| 11:55.42 | leifmadsen | (on their end) |
| 11:55.42 | mvanbaak | morons |
| 11:55.46 | leifmadsen | no kidding |
| 11:55.50 | leifmadsen | everyone is a moron these days |
| 11:56.00 | leifmadsen | back in a few, gotta go make some breakfast! |
| 11:56.07 | a2_ | moron |
| 11:56.08 | a2_ | lol |
| 11:56.09 | mvanbaak | have a red vino with it |
| 12:01.20 | a2_ | with some thc |
| 12:01.56 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:04.08 | mvanbaak | amen |
| 12:04.29 | beek | good morning [TK]D-Fender |
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| 12:07.59 | [TK]D-Fender | *yawn* |
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| 12:12.31 | [TK]D-Fender | *BOING* ! |
| 12:12.54 | mvanbaak | :) |
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| 12:22.18 | kombi | good morning everyone, may I present you with a wonderfull piece of phone trouble, I minor issue but a thing of beauty! http://pastebin.se/198510 |
| 12:28.57 | kombi | I'll try with a different number now to see, whether it depends on length or digits.. |
| 12:31.48 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 12:33.44 | [TK]D-Fender | kombi: What tech are you dialing out on? |
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| 12:50.55 | jes-o-mat | hi |
| 12:51.19 | jes-o-mat | is there some CLI command like core show modules loaded |
| 12:51.30 | jes-o-mat | to see which modules are loaded? |
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| 12:53.03 | JunK-Y | jes-o-mat: module show |
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| 12:58.54 | jes-o-mat | JunK-Y: thnx |
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| 13:02.10 | Rob3Rt | congratulations #asterisk, you've made my autojoin-4-life list. |
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| 13:15.12 | cousin_luigi | hello |
| 13:16.12 | cousin_luigi | any recommendations for a USB FXO card? |
| 13:17.22 | [TK]D-Fender | cousin_luigi: So far Sangoma is the only small scale model choice for this with any hope of reputability. |
| 13:18.13 | mort_gib | Why USB and not an ATA |
| 13:18.13 | cousin_luigi | [TK]D-Fender: thanks |
| 13:18.55 | [TK]D-Fender | Indeed these devices should only be considered if absolutely necessary |
| 13:19.11 | cousin_luigi | mort_gib: maybe I'm mixing up FXO with FXS, but can I use an ATA to link my POTS line to asterisk? |
| 13:19.30 | [TK]D-Fender | cousin_luigi: ATA no. |
| 13:19.38 | [TK]D-Fender | cousin_luigi: What are your actual needs? |
| 13:19.56 | mort_gib | I have used a zyxel 2002L like that |
| 13:19.58 | [TK]D-Fender | cousin_luigi: How many ports, what kind of usage, where will it be installed, etc |
| 13:19.59 | cousin_luigi | I just need to get rid of anonymous calls. Switching to VoIP is not a possibility. |
| 13:20.07 | cousin_luigi | single POTS line |
| 13:20.20 | cousin_luigi | I admit it's overkill, but they're doing my head in |
| 13:20.38 | [TK]D-Fender | cousin_luigi: Linksys SPA -3102 <--- |
| 13:20.50 | cousin_luigi | <googles> |
| 13:21.03 | [TK]D-Fender | cousin_luigi: http://www.telephonydepot.com/Catalog/Cisco-Analog-Adapters/Linksys-SPA3102 |
| 13:21.54 | kombi | fender: sorry, on the phone for an ages... dialing out on dahdi/bri/b410p |
| 13:22.02 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:22.25 | cousin_luigi | [TK]D-Fender: cheap, even |
| 13:22.39 | mort_gib | kombi: HEY! point-to-multipoint?? |
| 13:22.49 | cousin_luigi | mort_gib: you used that zyxel model the same way? |
| 13:23.06 | [TK]D-Fender | kombi: enable actual debug on the card. This is not an * issue, its down the the channel-driver |
| 13:23.29 | mort_gib | Yeah, it will take a phone line |
| 13:23.43 | kombi | wort_gib:ptp |
| 13:24.16 | kombi | wort_gib: funny you should ask..;) |
| 13:24.38 | cousin_luigi | [TK]D-Fender: will the stock firmware suffice? |
| 13:24.40 | mort_gib | kombi: I need point-to-multipoint working |
| 13:25.37 | kombi | on b410p? |
| 13:25.37 | kombi | have you set the jumpers right? |
| 13:25.37 | mort_gib | do I need jumper setting for point-to-multipoint |
| 13:25.37 | kombi | fender: must do that some other time, I'm in NY, box is over in Germany.. |
| 13:25.53 | cousin_luigi | [TK]D-Fender: sounds great though |
| 13:26.00 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:26.02 | kombi | mort_gib: I faintly recall that (is that english?), not near the box right now |
| 13:26.38 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:27.15 | mort_gib | kombi: Not sure maybe American or Canadian. I haven't seen anything indicating that a jumper could be involved... |
| 13:28.26 | kombi | I believe the card can do both, you'll need to set all the configs right to ptmp. dahdi-tool might come in handy |
| 13:29.09 | mort_gib | I did all that, but come up saying "illegal signalling" |
| 13:30.05 | cousin_luigi | bbl |
| 13:30.07 | *** part/#asterisk cousin_luigi (n=luigi@unaffiliated/cousinluigi/x-395723) |
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| 13:37.03 | anonymouz666 | anyone aware of any patch regards to the DNS async support? |
| 13:38.28 | Rob3Rt | can someone check my 5060 is open with sipsak or somethin pleae ? |
| 13:38.31 | Rob3Rt | please* |
| 13:38.34 | Rob3Rt | xs.iq.id.au |
| 13:39.16 | Rob3Rt | im running aster, ive done some checks, 50% checks says its open and %50 says its now, and I cang seem to register my voip provider, thanks in advance ? |
| 13:41.07 | [TK]D-Fender | Rob3Rt: Just because you say you can't register doesn't mean anything |
| 13:41.23 | Rob3Rt | lol |
| 13:41.34 | Rob3Rt | im running aster and grc says the port is closed, shouldnt it be open? |
| 13:41.49 | [TK]D-Fender | Rob3Rt: imagebin.ca <- |
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| 13:47.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:49.16 | Rob3Rt | [TK]D-Fender, http://imagebin.ca/view/re6Nzka.html |
| 13:49.18 | *** join/#asterisk ehsjoar (n=ehsjoar@24.9.91.203) |
| 13:49.49 | [TK]D-Fender | robFull-page the GRC one |
| 13:50.16 | [TK]D-Fender | Rob3Rt: And SIP SHOW REGistrY being empty tells me that you aren't even CONFIGURED TO REGISTer |
| 13:51.15 | Rob3Rt | http://imagebin.ca/view/GP7QibSC.html |
| 13:51.58 | Rob3Rt | register => 0756416068:PASSWORD@58.96.1.2/0756416068 |
| 13:51.59 | [TK]D-Fender | robrobNow back up one screep where you told it what to scan and PB that |
| 13:52.11 | Rob3Rt | told it to scan 5060 dude |
| 13:52.16 | [TK]D-Fender | Rob3Rt: and PB your sip.conf masking only passwords |
| 13:52.22 | [TK]D-Fender | robrobNow back up one screep where you told it what to scan and PB that <--- DO IT |
| 13:52.42 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 13:53.38 | [TK]D-Fender | Rob3Rt: and "CLOSED" says that the router isn't blocking it. |
| 13:53.42 | Rob3Rt | [TK]D-Fender, i dont understand what you mean, you want me to pastebin where i entered the port? |
| 13:53.50 | [TK]D-Fender | Rob3Rt: Yes |
| 13:53.54 | Rob3Rt | yeah but aster is running, shouldnt it be open to accept connections ? |
| 13:54.01 | [TK]D-Fender | Rob3Rt: DO IT |
| 13:54.37 | Rob3Rt | I put 5060 into the box and clicked "customer port probe" at http://www.grc.com/x/ne.dll?rh1dkyd2 |
| 13:54.46 | Rob3Rt | cant pastebin that |
| 13:54.50 | Rob3Rt | its a webapp |
| 13:55.15 | [TK]D-Fender | robAnd you can take an F-ing screen shot of you filled in before hitting the "go" button |
| 13:55.33 | Rob3Rt | err |
| 13:55.35 | Rob3Rt | yeah. |
| 13:55.37 | Rob3Rt | ok. |
| 13:55.47 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 13:56.05 | Kobaz | so i've been transfered 10 times by dell... trying to resolve a billing problem... still not at the right department |
| 13:56.33 | Rob3Rt | [TK]D-Fender, http://imagebin.ca/view/0B1aiL.html |
| 13:56.49 | Rob3Rt | theres the shot of "before" i hit the button to scan |
| 13:57.01 | Rob3Rt | nfi why u want that, but there it is |
| 13:57.08 | [TK]D-Fender | Rob3Rt: Now for the quick lesson : The port was listed as "CLOSED". |
| 13:57.15 | Rob3Rt | ok |
| 13:57.41 | [TK]D-Fender | Rob3Rt: * SIP is **UDP**. UDP is STATELESS. That scan was a **TCP** scan and is fucking worthless |
| 13:57.50 | [TK]D-Fender | starts swinging |
| 13:57.53 | jaytee | wow! this asterisk thingy is cool! you can actually make calls with it.....and stuff :-) |
| 13:57.54 | Rob3Rt | fucking nice poiint smart one :p |
| 13:57.59 | kombi | mort_gib: are you sure the telco sends ptmp? maybe try ptp just for kicks.. |
| 13:58.03 | Rob3Rt | woops, soz, swearing |
| 13:58.16 | mort_gib | kombi: I did, failing.... |
| 13:58.18 | [TK]D-Fender | Rob3Rt: Now pull your head out of your ass and PB your sip.conf |
| 13:58.18 | kombi | jaytee: how about that?;) |
| 13:58.27 | Rob3Rt | :( |
| 13:58.29 | Rob3Rt | k :p |
| 13:58.40 | mort_gib | kombi: I use ptmp with Sangoma cards |
| 13:58.51 | [TK]D-Fender | Rob3Rt: The fact * CLI doesn't show that it has anything to register TO suggestions you screwed up your config |
| 13:59.02 | kombi | hmm, lights are green where plugs are? |
| 13:59.08 | anonymouz666 | disconnects [TK]D-Fender's asterisk boxes from Internet to lock up chan_sip |
| 13:59.11 | anonymouz666 | heh. |
| 14:00.16 | Rob3Rt | [TK]D-Fender, http://pastebin.com/m6f2b0709 |
| 14:00.25 | Rob3Rt | there 'tis |
| 14:00.54 | [TK]D-Fender | Rob3Rt: Try again, that isn't all of it |
| 14:01.25 | [TK]D-Fender | Rob3Rt: and wtf is * doing registering to itself? |
| 14:02.44 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:02.54 | Rob3Rt | [TK]D-Fender, http://pastebin.com/d505f7818 |
| 14:02.58 | Rob3Rt | umm |
| 14:03.22 | Rob3Rt | what do you mean to itself |
| 14:03.25 | Rob3Rt | um |
| 14:03.30 | [TK]D-Fender | Rob3Rt: register => 1000:1000@10.0.0.5 |
| 14:03.33 | Rob3Rt | dont tell me im registering tomy ip |
| 14:03.35 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 14:03.38 | Rob3Rt | yeah thats a 3cx server |
| 14:03.42 | Rob3Rt | running on 5061 |
| 14:03.45 | Rob3Rt | which registers good |
| 14:03.53 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:03.57 | [TK]D-Fender | Rob3Rt: Well you port forwarded your router to your 3CX, NOT F-ING ASTERiSK |
| 14:04.13 | Rob3Rt | no |
| 14:04.17 | Rob3Rt | 3cx = 5061 |
| 14:04.23 | Rob3Rt | asterisk=5060 |
| 14:04.36 | [TK]D-Fender | Rob3Rt: Your 3CX = PC w/ * on it as well? |
| 14:04.47 | Rob3Rt | yea |
| 14:04.49 | [TK]D-Fender | GAH |
| 14:04.53 | [TK]D-Fender | wretches |
| 14:04.55 | Rob3Rt | but it was working fine last night |
| 14:04.58 | leifmadsen | anyone know of an online script that will test dialing your SIP URI ? |
| 14:04.59 | Rob3Rt | its a test server |
| 14:05.06 | [TK]D-Fender | Rob3Rt: Ok, that's at least MARGINALLY sane |
| 14:05.08 | leifmadsen | I need a test, but needs to come from outside my network :) |
| 14:05.22 | lirakis | leifmadsen: ?? you mean some thing like sipp on a webpage |
| 14:05.35 | [TK]D-Fender | Rob3Rt: Next lesson : REGiSTER's have to come after everything else under [general] |
| 14:05.50 | [TK]D-Fender | Rob3Rt: So your NAT config is cut off by them. |
| 14:05.51 | leifmadsen | uh... kinda. I wanna test my sip:user@pbx.hostname.com setup |
| 14:05.53 | lirakis | leifmadsen: I never thought of that... that would be a nice tool for people to test stuff ... just visit a site and make a test call |
| 14:06.32 | [TK]D-Fender | leifmadsen: You you at that address now? |
| 14:06.39 | [TK]D-Fender | Are* |
| 14:06.39 | leifmadsen | [TK]D-Fender: no :) |
| 14:06.48 | [TK]D-Fender | leifmadsen: then jsut grab a softphone :) |
| 14:07.01 | leifmadsen | [TK]D-Fender: like I said above -- needs to be a test outside of my network |
| 14:07.28 | [TK]D-Fender | leifmadsen: I meant if you were on that subnet / behind its NAT, etc |
| 14:07.53 | leifmadsen | I still don't know what you meant |
| 14:07.56 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 14:08.06 | leifmadsen | you mean pbx.hostname.com? no |
| 14:08.14 | Rob3Rt | [TK]D-Fender, it was the registry thing needing to come after everything else in sip, and errr, THANKS PAL =] |
| 14:08.33 | Rob3Rt | THX BUDDYZ =) |
| 14:08.34 | [TK]D-Fender | Rob3Rt: No, after everything else in [general] <- |
| 14:08.37 | Rob3Rt | hehehe |
| 14:08.48 | Rob3Rt | k sorted that freakin leet, cheers (H) |
| 14:08.52 | Rob3Rt | that's* |
| 14:08.59 | lirakis | leifmadsen: ive got a ruby scrip that will send invites.. but its not webbased |
| 14:09.12 | lirakis | i can originate some thing if you want, pm |
| 14:09.23 | [TK]D-Fender | Rob3Rt: Functional? |
| 14:10.05 | lirakis | leifmadsen: or you can use it from some shell out side of your network |
| 14:10.24 | leifmadsen | lirakis: that works -- I just need to test this SIP URI I'm setting up, but can't test it locally :) |
| 14:10.39 | Rob3Rt | [TK]D-Fender, Registered. Let's put it that way :p |
| 14:10.44 | Rob3Rt | cheers ;) |
| 14:10.47 | [TK]D-Fender | Rob3Rt: 1 down... |
| 14:10.51 | leifmadsen | lirakis: can you msg me the login info and place a copy of the script in my home dir? |
| 14:10.57 | Rob3Rt | grabs face like home alone guy |
| 14:11.06 | leifmadsen | lirakis: if you were offering such a setup :) |
| 14:11.41 | lirakis | leifmadsen: do you have a shell to run it from outside your network? if so ill just give you the ruby script |
| 14:12.01 | lirakis | leifmadsen: ill definitely setup some thing web based though b/c thats a really useful tool for a lot of people. |
| 14:12.10 | leifmadsen | lirakis: hmmmm.... actually, I don't think I do right now |
| 14:12.21 | leifmadsen | I have to get that server moved from under my bed to downtown :) |
| 14:12.28 | lirakis | leifmadsen: if you dont have a shell just pm me where you want the invite sent, and what you want the dn etc. to appear as |
| 14:12.31 | lirakis | and i will originate it |
| 14:12.45 | leifmadsen | lirakis: coolio, will send you a msg in a moment |
| 14:14.57 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.215) |
| 14:16.48 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:16.48 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:17.04 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 14:23.09 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 14:23.25 | *** join/#asterisk benedikt (n=benedikt@gestir.utrad.is) |
| 14:24.43 | benedikt | I have written a script that looks up (icelandic) phone numbers online and returns the owner. I want to integrate this script with asterisk. But i dont know how i should go about doing this |
| 14:27.02 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 14:27.15 | *** join/#asterisk maverick (n=maverick@83.240.229.38) |
| 14:27.41 | benedikt | ive seen this done and i know its possible |
| 14:32.22 | Dovid | it will prob. be put in to asterisk add-ons |
| 14:32.25 | *** join/#asterisk x86 (n=porteb1@p3m/member/x86) |
| 14:32.27 | Dovid | you can ask in #asterisk-dev |
| 14:32.33 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 14:32.53 | benedikt | Ah, another channel for dev. Thanks. |
| 14:33.10 | [TK]D-Fender | benedikt: I don't see how that's a dev issue |
| 14:33.35 | [TK]D-Fender | benedikt: Perhaps you could elaborate on what "integration" is supposed to mean in your case |
| 14:33.37 | Dovid | i was just asking myself. thought he wanted to add to the ast code* |
| 14:34.08 | benedikt | [TK]D-Fender: developing a addon for asterisk maybe? Doesnt matter to me in what channel i speak. I want to make it a addon for asterisk |
| 14:34.11 | Dovid | unless he wants to create an AGI |
| 14:34.36 | [TK]D-Fender | Dovid: I don't see a need for even that yet |
| 14:35.12 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 14:35.24 | benedikt | Ive never written anything for asterisk before so i dont konw where to start. Couldnt find anything relevant by googling so I came here. |
| 14:35.44 | lirakis | leifmadsen: i sent that invite to my asterisk test server, i think it "understands" it b/c it sends options requests back - which i dont respond to. I guess it needs a more full featured signaling suite for test calls. |
| 14:36.03 | leifmadsen | ah perhaps |
| 14:36.38 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 14:38.08 | *** join/#asterisk aurax (n=aurax@bzq-179-76-199.static.bezeqint.net) |
| 14:38.52 | [TK]D-Fender | benedikt: Do far this isn't really "writing for *". You need to come up with a practical application of the result of your script |
| 14:38.53 | aurax | hello, I'm having issue with random one-way audio, new calls suddently have only incoming audio. |
| 14:39.07 | aurax | i'm using 1.4.25-1 |
| 14:41.33 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 14:41.36 | lirakis | leifmadsen: i sent a couple of tries with sipp instead - not sure if you got anything |
| 14:41.44 | benedikt | [TK]D-Fender: I have and I also have written it here. Ill give you a practical example. the ophone number 5876655 calls and the asterisk machine recies the call and it goes to the approriate phone wich then displays something like "Call from 5876655". But i want it to execute my script and return "5876655: John Smith" and send that to the phone wich displays a somewhat more useful information |
| 14:41.48 | leifmadsen | lirakis: yep, that one went through |
| 14:41.55 | lirakis | leifmadsen: i definitely got options messages back |
| 14:41.56 | leifmadsen | lirakis: my phone started to do some odd rings :) |
| 14:41.59 | lirakis | lol |
| 14:42.25 | lirakis | leifmadsen: let me know if you need anything different, or another test to watch the debug |
| 14:42.39 | [TK]D-Fender | benedikt: then mod your script as an AGI and update the callerid name for that call. |
| 14:43.04 | leifmadsen | lirakis: sure thing -- did that go to the pbx.-----.com or -----.com ? |
| 14:43.12 | benedikt | [TK]D-Fender: that sounds like what i want to do. But as i have said, ive never done any work with asterisk and didnt even know what AGI was until a fe wminutes ago |
| 14:43.20 | lirakis | leifmadsen: pbx. |
| 14:43.34 | lirakis | leifmadsen: but the uri specified ---.com |
| 14:43.41 | leifmadsen | lirakis: ok, so that works fine -- I still need to sort out the SIP URI stuff in DNS I think |
| 14:43.42 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 14:44.14 | [TK]D-Fender | ~book |
| 14:44.14 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:44.34 | [TK]D-Fender | benedikt: Well you've got a lot of learning to do. Go read the BOOK as listed above |
| 14:44.36 | afink | Opinion question, I have 5 companies that reside in the same building they all the users do business together and call each other regulary would it be necessary or preferred to split them up into different contexts? |
| 14:45.01 | [TK]D-Fender | afink: Of course. What each context has access to is another matter |
| 14:45.21 | afink | ok thanks |
| 14:45.23 | leifmadsen | afink: different contexts to control what they can dial etc, but you can do "include => company_B_phones" etc.... |
| 14:45.24 | [TK]D-Fender | afink: Depends what points to that context. |
| 14:45.27 | *** join/#asterisk kn0x (n=pinochle@67.159.48.101) |
| 14:45.36 | [TK]D-Fender | (s) |
| 14:45.37 | *** join/#asterisk rosscoh (n=rosscoh@87.sub-75-220-114.myvzw.com) |
| 14:45.57 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:47.02 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 14:47.14 | aurax | can anyone help |
| 14:47.43 | lirakis | aurax: YES!!! let me dedicate time to a completely unknown question! |
| 14:47.51 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 14:51.22 | rosscoh | I need some help as well |
| 14:51.23 | rosscoh | http://pastebin.com/m6b29cd08 |
| 14:52.44 | beek | ~freepbx |
| 14:52.45 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:52.59 | beek | rosscoh: ^^^^^^^^ |
| 14:53.11 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 14:54.00 | [TK]D-Fender | "click to hear".. huh? |
| 14:54.03 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 14:54.53 | afink | wow. I am just now breaking free from the bondage of a GUI and everything seems so simple. Wish I would have read the book first... |
| 14:56.55 | beek | hears heavenly chorus |
| 14:57.37 | [TK]D-Fender | adds some reverb, delay and a filter-pan |
| 14:58.04 | beek | queues the fireworks |
| 14:58.15 | aurax | lirakis :) |
| 14:58.16 | thehar | lights em up |
| 14:58.20 | aurax | I'm having random one way audio |
| 14:58.59 | *** join/#asterisk ariel_ (n=chatzill@96.49.119.119) |
| 14:59.02 | [TK]D-Fender | aurax: And chance of getting any useful details about what you're doing and maybe... I dunno... debug of a failed call while you're at it? |
| 14:59.15 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 14:59.41 | ariel_ | Morning everyone |
| 14:59.51 | leifmadsen | adds some auto-tune |
| 15:00.28 | [TK]D-Fender | ARGHHHHHHHH!!!!!!!! |
| 15:00.41 | [TK]D-Fender | axe-murders leifmadsen |
| 15:00.47 | [TK]D-Fender | leifmadsen: HERESY! |
| 15:00.57 | leifmadsen | [TK]D-Fender: you die now |
| 15:01.04 | [TK]D-Fender | leifmadsen: A little every day.... |
| 15:02.58 | *** join/#asterisk AussieGuy (n=AussieGu@r220-101-170-252.cpe.unwired.net.au) |
| 15:04.31 | AussieGuy | if I have a linux machine, what kind of hardware would I need to take a phone call from a ptsn network, then route it over the internet to a sip server? |
| 15:05.37 | AussieGuy | in other words I want to take a phone call on my landline and have it routed to an overseas mobile number via a voip provider |
| 15:06.19 | [TK]D-Fender | AussieGuy: You wouldn't need a voip provider.. you would BECOME your own. |
| 15:06.42 | [TK]D-Fender | AussieGuy: And if you're looking for something cheap for personal use, get the Linksys SPA-3102 |
| 15:07.01 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 15:08.02 | AussieGuy | to route it from the computer back to an overseas mobile I would probably use a voip provider to avoid standard international rates |
| 15:08.36 | AussieGuy | unless I had a box set up in that country, but were not at that stage yet |
| 15:09.36 | Joel | AussieGuy what is on the other end that you're calling? |
| 15:09.52 | AussieGuy | an overseas mobile number |
| 15:09.59 | Joel | oh, duh, you said that, sorry. |
| 15:10.05 | AussieGuy | clients will call in, then get routed to the overseas mobile |
| 15:10.13 | ickmund | Anyone know what the keys on a Aastra 536EM are called when used with a 55i? :P |
| 15:11.03 | [TK]D-Fender | AussieGuy: Ok, that makes sense then |
| 15:11.16 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:12.01 | ickmund | Note to self, expmod1 keyX |
| 15:12.21 | ariel_ | If it's only one pstn line the SPA-3102 would be a good pick. |
| 15:21.46 | *** join/#asterisk NemesisX11 (i=480c5081@gateway/web/freenode/x-35a7b310950a1fd8) |
| 15:23.05 | NemesisX11 | Hi all - has anywon out there gotten Asterisknow and a Magicjack to play nice? I am having trouble setting it up I have all my info but I am struggling when it comes to doing my incoming and outgoing routes. |
| 15:23.48 | Joel | NemesisX11 wrong channel. |
| 15:24.21 | NemesisX11 | ? website said to come here ? |
| 15:24.39 | bmoraca | this is asterisk general chat...you want #asterisknow |
| 15:24.49 | NemesisX11 | Ok - Thank you! |
| 15:25.01 | NemesisX11 | Sorry about the mixup |
| 15:25.05 | bmoraca | no problem |
| 15:25.19 | ariel_ | actually has anyone really gotten MagicJack working with plain asterisk? |
| 15:25.35 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:25.36 | aurax | [TK]D-Fender sure.. what info can be helpful ? |
| 15:25.36 | Joel | ariel_ I've known people who have, but it's a tos violation |
| 15:26.01 | ariel_ | well I know most that have set it up get blocked after a week or so. |
| 15:26.08 | [TK]D-Fender | auxYou've described noting about your calls, your configuration, etc. |
| 15:28.15 | NemesisX11 | If anywon is interessted I do have the links for plain asterisk config - this is where my confusion comes from because I have sample configs but I cannot just edit sip.conf with asterisk now i have to do it through the GUI |
| 15:28.21 | NemesisX11 | anyone* |
| 15:29.14 | *** join/#asterisk errotan (n=errotan@62.201.122.123) |
| 15:29.34 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 15:32.00 | [TK]D-Fender | NemesisX11: the GUI doesn't sue sip.conf, it configures users.conf <- |
| 15:32.04 | [TK]D-Fender | use* |
| 15:32.15 | ariel_ | NemesisX11: if you have the basic setup put into your trunk settings. Registration line is at the bottom. |
| 15:32.28 | ariel_ | actually asteriskNow 1.5 uses freepbx last I saw |
| 15:32.54 | [TK]D-Fender | (Asterisk GUI that is) |
| 15:33.18 | [TK]D-Fender | NemesisX11: And if you're using FreePBX well.... you're completely in the wrong channel... |
| 15:34.23 | NemesisX11 | So in other words life would he a hell of a lot easier if I just build a regular asterisk box like I was going to do from the get go haha |
| 15:34.25 | ariel_ | there's the [TK]D-Fender I know... |
| 15:34.38 | beek | hears the heavenly chorus again |
| 15:34.48 | ariel_ | rofl |
| 15:35.12 | ariel_ | they all have there use |
| 15:35.38 | [TK]D-Fender | NemesisX11: Yeah.. like you were going to... |
| 15:35.50 | [TK]D-Fender | NemesisX11: And that changed when/why? |
| 15:36.01 | NemesisX11 | well i am good in a linux shell i just throught i would take the easy way out |
| 15:36.41 | [TK]D-Fender | NemesisX11: Well move along.... you're in their playground now... |
| 15:37.07 | bmoraca | NemesisX11: if you understand the limitations and scope and purpose of a GUI, it can be just fine. but if you don't have at least a basic understanding of what's going on behind the scenes, then don't even bother. |
| 15:38.39 | NemesisX11 | ive set up 3 regular asterisk boxes for small companies before asterisknow and when I came back and saw it i figured it was worth a shot |
| 15:39.52 | ariel_ | it is worth a shot, you can setup your settings in the sip_custom.conf or learn how freepbx does it's dialing and set it up as a trunk |
| 15:41.25 | aurax | so can anyone help me out with my random one way audio issue ? |
| 15:41.55 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 15:42.03 | [TK]D-Fender | aurax: Found a clue and some details to share? |
| 15:43.12 | ariel_ | hint nat |
| 15:43.36 | aurax | sure, |
| 15:44.21 | aurax | i'm using local sip provider, with g729, sometimes totaly random, users experience new calls with one way audio, then suddently it's back again... what info would i need to pastebin ? |
| 15:46.13 | ariel_ | do you have enough license for the amount of channels your using? Do you have canreinvite=yes? There are so many things that can do that, but really a good debug post via pastebin when it happens is needed. |
| 15:46.21 | aurax | i have canreinvite=no |
| 15:48.02 | ronr | hi, I'm following the .call example in the book, but my asterisk says Playback is not an application :S it's working just fine from extensions.conf, am I missing something here? |
| 15:49.56 | *** join/#asterisk dymaxion (n=dymaxion@79.123.71.144) |
| 15:50.19 | *** join/#asterisk twitchnln (n=twitchnl@loopback-dktc.atl.oneringnetworks.net) |
| 15:51.03 | twitchnln | morning, i was wondering if anyone else has experienced caller id getting mangled when arriving on iax trunk and being passed out sip? |
| 15:51.49 | ronr | show applications in the console does show Playback btw |
| 15:52.03 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:52.42 | [TK]D-Fender | aurax: pastebin the CLI output of a failed call with SIP debug enabled |
| 15:52.43 | ronr | got it... seems to be a bug in the book, it says Application: Playback() where it should say Application: Playback |
| 15:52.44 | [TK]D-Fender | ~pb |
| 15:52.45 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 15:52.47 | [TK]D-Fender | ^^^^^^^ |
| 15:52.49 | [TK]D-Fender | ^^^^^^^ |
| 15:53.17 | [TK]D-Fender | twitchnln: Nope, but you're welcome to show us a failed call with full debug as well... |
| 15:53.53 | [TK]D-Fender | ronr: PASteBIN is your friend as well... |
| 15:53.53 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 15:53.53 | twitchnln | TK - http://pastebin.ca/1495917 |
| 15:54.29 | ronr | [TK]D-Fender: really? what did I 'paste' in the channel that I shouldn't have? |
| 15:54.57 | [TK]D-Fender | twitchnln: Full debug please.... |
| 15:55.07 | [TK]D-Fender | twitchnln: Don't start chopping this up. |
| 15:55.30 | [TK]D-Fender | ronr: Nothing. You SHOULD pastebin useful backup for your problem so we can look at it to help you <- |
| 15:56.21 | ronr | [TK]D-Fender: ah like that, I started to worry you considered my posts flooding :) I'll paste the problem next time |
| 15:57.15 | twitchnln | TK - that is a call that completed, i can see correct caller id coming into asterisk, but in the invite header it's changed to unknown, i'm pulling this from production system with 68 concurrent calls, but it's only happening on 1 trunk |
| 15:59.49 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 16:00.06 | ariel_ | ok I am off to catch a ship. See you in about 4 days..... |
| 16:01.32 | *** join/#asterisk proute (n=AnthonyC@LMontsouris-152-63-16-150.w217-128.abo.wanadoo.fr) |
| 16:01.37 | proute | Hello |
| 16:02.01 | proute | I use * 1.4.25 with B410P BRI card. So I installed dahdi package.... |
| 16:02.37 | proute | If I put in dahdi.conf file the parmeter : signalling= bri_cpe_ptmp, I've got an error... |
| 16:03.07 | proute | Does * 1.4.25 with dahdi package and B410P are compatibles? |
| 16:03.18 | [TK]D-Fender | twitchnln: Unknown can also happen due to privacy flag settings <- |
| 16:03.28 | [TK]D-Fender | proute: Yes |
| 16:03.57 | twitchnln | TK - where would i find that setting? I'm not familiar with it |
| 16:04.09 | [TK]D-Fender | twitchnln: "core show application setcallerpres" |
| 16:04.53 | [TK]D-Fender | lunch time, BBIAB |
| 16:05.03 | timeshell | Does fop work with * 1.6.0? |
| 16:05.39 | proute | [TK]D-Fender: thanks for your answer. but in my log I have this: chan_dahdi.c: Unknown signalling method 'bri_cpe_ptmp' |
| 16:06.07 | tzafrir_laptop | proute, what version of asterisk do you use? |
| 16:06.19 | proute | tzafrir_laptop: 1.4.25.1 |
| 16:06.23 | tzafrir_laptop | hmm... that code is not included in asterisk 1.4 |
| 16:06.36 | tzafrir_laptop | though I know of a backport... |
| 16:07.09 | proute | So, if this code is not included in * 1.4, How can I "fix" this problem (without upgraded to * 1.6)? |
| 16:07.13 | tzafrir_laptop | timeshell, latest fop (0.30 IIRC) should |
| 16:07.31 | timeshell | aw, fop2 isn't free |
| 16:07.48 | tzafrir_laptop | the latest version of the GPLed FOP |
| 16:09.22 | leifmadsen | so who here uses AgentCallbackLogin()? If you do, why have you not switched to using the AddQueueMember() method? What are the "gotchas" or what has been complicated for you? |
| 16:09.52 | leifmadsen | I'm putting together a "Moving from AgentCallbackLogin() to other methods" article, and looking to figure out why people get stuck, because it turns out this really isn't very hard... |
| 16:09.58 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
| 16:11.27 | bmoraca | i never really saw the allure of the agentcallbacklogin method of joining queues anyway...just seemed clunky and wrong |
| 16:12.38 | timeshell | Not wrong |
| 16:12.41 | timeshell | preferred. |
| 16:12.48 | leifmadsen | why preferred? |
| 16:12.59 | timeshell | Works better for us. |
| 16:13.01 | timeshell | :D |
| 16:13.58 | leifmadsen | ok, define "works better for us" -- because Asterisk 1.6.x does not have AgentCallbackLogin() -- I'm trying to write the article that will get you away from using it |
| 16:13.59 | timeshell | AddQueueMember / AgentCallBackLogin preferred. I don't care which. |
| 16:14.33 | timeshell | I'm not saying I prefer AgentCallBackLogin to AddQueueMember. |
| 16:14.36 | leifmadsen | they are totally different functionalities... perhaps you don't understand what I'm trying to do |
| 16:14.40 | leifmadsen | oh :) |
| 16:14.45 | leifmadsen | that's how I read it |
| 16:15.07 | timeshell | To me, they do the same thing so they're the same. |
| 16:15.26 | leifmadsen | heh |
| 16:15.31 | leifmadsen | you're be surprised how many people don't see it that way |
| 16:15.41 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:15.50 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 16:17.32 | *** join/#asterisk Verz (n=asldkj@dhcp-199-17-149-24.cf-res.cfu.net) |
| 16:18.37 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 16:25.02 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:25.18 | *** join/#asterisk rosscoh (n=rosscoh@72.37.203.61) |
| 16:25.27 | rosscoh | Fender I have that printout that you asked for |
| 16:26.50 | rosscoh | http://pastebin.com/m336e75d7 |
| 16:26.51 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-ba22c067cf20df6b) |
| 16:26.58 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 16:26.58 | rosscoh | I need some help if anyone is available |
| 16:27.09 | *** join/#asterisk voipguy7777 (n=greekman@75-63-110-57.uvs.hstntx.sbcglobal.net) |
| 16:27.43 | voipguy7777 | <PROTECTED> |
| 16:28.38 | *** join/#asterisk Carlos_PHX (n=carlos@108.sub-75-210-203.myvzw.com) |
| 16:28.56 | *** part/#asterisk Carlos_PHX (n=carlos@108.sub-75-210-203.myvzw.com) |
| 16:31.14 | [TK]D-Fender | leifmadsen: PB 1.4's "core show application AgentCallbackLogin" |
| 16:31.39 | leifmadsen | why? you have just as easy access to it as I do |
| 16:31.50 | proute | ok, I found a solution: http://groups.google.es/group/asterisk-es/browse_thread/thread/d808350ed0e66708/73b105ac0d022f16?lnk=gst&q=dahdi+b200p#73b105ac0d022f16 |
| 16:32.18 | [TK]D-Fender | leifmadsen: I don't ahve a 1.4 system handy |
| 16:32.27 | *** join/#asterisk Verz (n=asldkj@24.149.5.28) |
| 16:33.03 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 16:34.33 | *** join/#asterisk voxter (n=voxter@190.241.15.217) |
| 16:35.25 | [TK]D-Fender | leifmadsen: Ok, well first i'd wonder about persistence with AQM. Second is having to write your own auth. |
| 16:35.38 | leifmadsen | VMauthenticate() |
| 16:35.54 | leifmadsen | you don't have to write your own auth, you can still use built in auth methods |
| 16:36.11 | leifmadsen | what do you mean "persistence" ? |
| 16:36.16 | rosscoh | Fender do you have any suggestions for me? |
| 16:36.18 | [TK]D-Fender | leifmadsen: I suppose the "membername" option makes it viable for ID-ing which agent got called regardless of the interface used... |
| 16:36.28 | rosscoh | http://pastebin.com/m336e75d7 |
| 16:37.04 | [TK]D-Fender | rosscoh: "When I click to hear the split second recording" <- CLICK? what ... "click"? |
| 16:37.25 | rosscoh | Fender it says "Press the number one to listen to your message" |
| 16:37.35 | [TK]D-Fender | rosscoh: and what does "can't get any of the voicemails to work" mean? What aspect is specifically broken? |
| 16:37.38 | rosscoh | basically it gives me only a split second to record a message |
| 16:37.54 | rosscoh | i press to record my voicemail |
| 16:38.18 | rosscoh | says begin recording, it beeps for me to record then INSTANTLY |
| 16:38.24 | rosscoh | says would you like to hear your message |
| 16:38.28 | [TK]D-Fender | rosscoh: Check your permissions <- |
| 16:38.37 | rosscoh | in what folder do you think it is? |
| 16:38.43 | rosscoh | to fix the permissions |
| 16:39.02 | [TK]D-Fender | rosscoh: =0, open writing: /var/spool/asterisk/voicemail/default/200/unavail.tmp format: wav49, (nil) <------ CLUE |
| 16:39.29 | [TK]D-Fender | rosscoh: And PB a COMPLETE call next time |
| 16:40.03 | rosscoh | PB? |
| 16:40.05 | rosscoh | what's PB |
| 16:40.22 | leifmadsen | pastebin |
| 16:40.30 | leifmadsen | or... peanut butter |
| 16:40.57 | rosscoh | love peanut butter |
| 16:40.58 | [TK]D-Fender | leifmadsen: AQM I would wonder how it survives a restart, etc. Then a question of controlling who can join a queue. Also that queues using chan_agent only has to log the agent in, and they are already memebers of multiple queues. You'd have to AQM each queue they are a member of (or should be) and determine that list through even more scripting in the first place. |
| 16:41.08 | [TK]D-Fender | I love pastebin |
| 16:43.02 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 16:43.12 | *** join/#asterisk kombi (n=kombi@cpe-68-175-101-211.nyc.res.rr.com) |
| 16:43.22 | *** join/#asterisk theGoat (n=none@75-151-177-173-Philadelphia.hfc.comcastbusiness.net) |
| 16:44.27 | theGoat | My upstream provider passes the callerid and name info in the packets, how can I get asterisk to pull the info. I can't find any good documentation on it |
| 16:44.37 | fister | what |
| 16:45.15 | theGoat | My upstream SIP provider is passing me the callerid info, and when a call comes in, asterisk doesn't seem to display that info on my phone when it rings |
| 16:45.17 | fister | what is it pulling now |
| 16:45.19 | fister | oh |
| 16:45.23 | fister | what phone are you using |
| 16:45.32 | redax | hi |
| 16:45.34 | theGoat | Looks like the did they dialed, and the number that called. |
| 16:45.37 | fister | are they on zaptel cards? |
| 16:45.42 | theGoat | I have a cisco 7960 |
| 16:45.46 | fister | oh |
| 16:45.47 | theGoat | No it's coming in via a sip trunk |
| 16:45.58 | fister | i could fix that w/ zaptel |
| 16:46.00 | fister | sry |
| 16:46.22 | fister | unless its in some extensions.conf macro |
| 16:46.38 | redax | I tried to interconnect a Siemens HiCom 300 vs asterisk using Digium Wildcard TE122 |
| 16:46.46 | theGoat | That's what I thought, but I can't find anything that will give me a definite answer. |
| 16:46.46 | redax | calls from siemens OK. |
| 16:46.59 | [TK]D-Fender | theGoat: NoOp the CID and include this in a PB with SIP debug for a failed call. |
| 16:47.03 | [TK]D-Fender | ~pb |
| 16:47.04 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 16:47.05 | [TK]D-Fender | ^^^^^^^^^^ |
| 16:47.05 | redax | calls from asterisk to siemens hicom == no voice |
| 16:47.10 | redax | any ideas? :o |
| 16:47.18 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 16:47.35 | [TK]D-Fender | redax: Where is the hicom located relative to *? |
| 16:47.57 | theGoat | Any certain level of verbosity you need? |
| 16:48.01 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:48.03 | redax | 100m far (relative next to it) over the E1 link |
| 16:48.51 | redax | but it happened even when it was interconnected with a 2m long utp cable |
| 16:48.54 | [TK]D-Fender | redax: Ah, its a PBX you're connecting? |
| 16:49.06 | redax | yes. hicom 300 |
| 16:49.17 | redax | the university's old pbx |
| 16:49.59 | rosscoh | Fender calls are working fine |
| 16:50.00 | [TK]D-Fender | redax: gotcha. What signalling, sure you're using the right kind of cable? (cross-over VS not. checked your timing as well? Is your test with * playing audio * receiving all by itself without other devices coming in play? |
| 16:50.03 | redax | actually I'm tried to use euroisdn, and the asterisk side is the pri_net |
| 16:50.06 | rosscoh | only thing that isn't working is the voicemails |
| 16:50.14 | rosscoh | and when someone calls it goes to a busy signal |
| 16:50.18 | rosscoh | not to any voicemail |
| 16:50.26 | [TK]D-Fender | rosscoh: ..... You are CALLING VOICEMAIL. You PB'd only a small portion of that call. |
| 16:50.44 | rosscoh | ok I will do it all again |
| 16:50.45 | [TK]D-Fender | rosscoh: Busy signal? PAstebin THAT failed call as well. |
| 16:50.54 | rosscoh | ok no problem |
| 16:50.55 | rosscoh | doing it now |
| 16:51.14 | redax | [TK]D-Fender: the strange thing is anybody can call the * extensions well from the hicom extensions, everything's fine voice and no problem at all |
| 16:51.45 | [TK]D-Fender | redax: PB a good call, and a failed call. |
| 16:52.01 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.70) |
| 16:52.18 | redax | meanwhile if I try to call any HiCOM extension from the *, the channels are up. but no voice in/out at all |
| 16:52.59 | redax | shoudl be the switchtype=euroisdn cause the whole problem ? |
| 16:53.17 | redax | should we use qsig, maybe ? |
| 16:55.06 | redax | [TK]D-Fender: the cabling's OK. the spans are up, and active, signalling is pri_net, switchtype is euroisdn |
| 16:55.10 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 16:55.19 | rosscoh | http://pastebin.com/m3d138e0b |
| 16:55.26 | rosscoh | Fender I've updated it |
| 16:55.37 | rosscoh | the first part is when I call from the outside trying to get to voicemail |
| 16:55.46 | rosscoh | the second part is when I try to record the voicemail again |
| 16:55.49 | rosscoh | I try twice |
| 16:55.59 | rosscoh | then press '2' to listen to it |
| 16:56.13 | rosscoh | and nothing happens and it just exits out |
| 16:56.14 | rosscoh | automatically |
| 16:57.14 | [TK]D-Fender | rosscoh: delte the recording, and try to leave a VM |
| 16:57.18 | [TK]D-Fender | delete* |
| 16:57.33 | rosscoh | nothing is recording |
| 16:57.35 | rosscoh | it looks like |
| 16:57.36 | bmoraca | rosscoh: in the future, don't sit on freepbx's dashboard page while getting htese logs. |
| 16:57.38 | rosscoh | at all |
| 16:58.37 | rosscoh | ok sorry about that |
| 16:58.41 | rosscoh | i shut it down now |
| 16:58.47 | rosscoh | or killed the browser rather |
| 16:58.52 | bmoraca | rosscoh: it looks like you have a file permissions error in your spool directory. |
| 16:59.10 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:59.43 | [TK]D-Fender | bmoraca: Which is something I suggested looking at some time ago |
| 16:59.51 | rosscoh | looking now at permssions |
| 16:59.56 | bmoraca | yes, i remember :) |
| 17:00.00 | rosscoh | i changed all of them |
| 17:00.06 | rosscoh | looking back over |
| 17:00.49 | [TK]D-Fender | theGoat: verbose 10, sip debug enabled. |
| 17:01.31 | rosscoh | what permissions do I have to give to what files? |
| 17:01.33 | rosscoh | exactly |
| 17:02.30 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 17:02.36 | bmoraca | rosscoh: you shouldn't have changed them at all...asterisknow had them set exactly as they needed to be for that particular distribution of applications |
| 17:02.50 | rosscoh | I never changed them |
| 17:03.03 | rosscoh | this is all default installation |
| 17:04.15 | rosscoh | I just need to know exactly what to look at to fix the file permssions |
| 17:05.12 | redax | [TK]D-Fender: don't you have any good tip? :) |
| 17:05.35 | [TK]D-Fender | redax: Got a good PAStebIN? |
| 17:06.04 | redax | ah PB ;x got it |
| 17:10.59 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:12.07 | *** join/#asterisk sjobeck (n=Adium@69-30-99-139.dv1sn.easystreet.com) |
| 17:12.24 | *** part/#asterisk sjobeck (n=Adium@69-30-99-139.dv1sn.easystreet.com) |
| 17:12.29 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
| 17:12.30 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 17:13.26 | redax | [TK]D-Fender: here's the call from * to HiCOM. called party answered, no voice at all... : http://pastebin.ca/1495985 |
| 17:14.02 | redax | [TK]D-Fender: actually I can't make a trace where HiCom calls *... maybe half an hour |
| 17:14.03 | *** join/#asterisk bhodder (n=blake@142.166.111.219) |
| 17:14.19 | redax | but does the above trace shows anything |
| 17:14.47 | [TK]D-Fender | redax: What about those SIP devices? I asked for a test only between * and HiCom |
| 17:14.59 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 17:15.07 | bhodder | Hi does anyone know much about using the Page() application |
| 17:15.28 | afink | when creating ring groups do you guys create a seperate context for them or just use extensions? |
| 17:16.10 | [TK]D-Fender | bhodder: Plenty. Now try a more specific question :) |
| 17:16.13 | redax | [TK]D-Fender hm. how can I do that, ... using originate? |
| 17:16.19 | theGoat | Is there a quick way to sanitize the output before I put it into pastebin. Don't want all my real did info, ips, posted publically |
| 17:16.29 | [TK]D-Fender | afink: What is a "ring group" and what does it have to do with "contexts'? |
| 17:16.34 | [TK]D-Fender | ReDYes |
| 17:16.41 | redax | ah |
| 17:16.42 | [TK]D-Fender | redax: Yes |
| 17:17.00 | bhodder | I am trying to set up a page extension however when i dial the page extension it also dials the originator. Is there a way to prevent that? |
| 17:17.18 | [TK]D-Fender | theGoat: Don't Nobody cares about your #'s and you'll likely cover up a screwup. |
| 17:18.09 | [TK]D-Fender | bhodder: You are the one telling Page() what to dial. If you didn't want it calling the originator, then you need to exclude them in the parameters you're passing |
| 17:18.25 | bhodder | ok |
| 17:18.46 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 17:18.51 | afink | [TK]D-Fender: an extension that rings a bunch of phones. Just trying to find the best way. Just asking b/c that is the way the gui had it setup before |
| 17:18.57 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 17:19.23 | [TK]D-Fender | afink: What does a single Dial command have to do with contexts? |
| 17:19.41 | [TK]D-Fender | afink: Dial will call whatever techs you tell it to. |
| 17:20.43 | afink | this is the way it was before. http://pastebin.com/m51594162. ok makes sense no need for seperate contexts really..just more smegma produced by the gui. |
| 17:20.49 | bhodder | If you, for example want extensions that are also in the Page() parameters can an extension there also be able to dial the Page() extension without it page them as well? |
| 17:21.45 | [TK]D-Fender | bhodder: Its your dialplan, you can do whatever you want. |
| 17:22.34 | theGoat | http://paste.lisp.org/+1SJB |
| 17:22.57 | [TK]D-Fender | afink: Does make a tiny amount of sense being a cascaded incrementally growing group of targets... of course that spams calls... |
| 17:23.28 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 17:24.22 | redax | [TK]D-Fender: http://pastebin.ca/1495997 should be a background music vs HiCom extension |
| 17:24.45 | bhodder | Ok. I understand that, but when I set my dialplan up and use the Page() application if one of the extensions in the parameter tries to page the group, that extension also recieves a call from itself and I am unable to determine how to stop that. is it possible? |
| 17:25.32 | theGoat | Ok, pate posted. |
| 17:26.01 | bhodder | Or is it necassary to have separate Page() extensions? |
| 17:27.02 | [TK]D-Fender | bhodder: You are the one passing that list of aprameters. If there is a device in there you didn't want, its YOUR FAULT. |
| 17:27.29 | theGoat | I see the caller name in the from field |
| 17:27.52 | [TK]D-Fender | bhodder: bhodder there is no "check" in age(). It will do exactly what you tell it. So go build a better list of devices to page. |
| 17:28.14 | [TK]D-Fender | theGoat: I see 2 numbers (1 in name field, 1 in number) |
| 17:28.36 | theGoat | But in one of the debug statements there is a name |
| 17:30.00 | bhodder | It is impossible then to part of the page list as well as one the extensions calling the page without it dialling back to yourself? |
| 17:31.18 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
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| 17:33.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:33.25 | bhodder | The "ChanIsAvial" will not work for page() |
| 17:34.17 | *** join/#asterisk apdg (n=grey@64.18.101.106) |
| 17:34.26 | [TK]D-Fender | bhodder: You don't seem to get it. You are passing that list of devices to call. I fyou included the device that is doing the page, it's your faul. Why the hell are you adding them? What makes you think that youa ren't in control of what YOU put into your Page()? * doesn't come up with all of that itself. |
| 17:34.34 | [TK]D-Fender | bhodder: Go looka t what you're doing. |
| 17:35.39 | [TK]D-Fender | bhodder: And what would ChanIsAvail() have to do with Page()? the app by itself works. |
| 17:36.13 | bhodder | Ok, thanks. |
| 17:37.31 | bhodder | I understand what you are saying, it was just what I was trying to do is an incorrect way of doing things |
| 17:37.49 | afink | [TK]D-Fender: is there any simple way to just have the first phones keep ringing then add the others? |
| 17:38.40 | [TK]D-Fender | afink: Dial the ones on a delay via a Local channel that wait()'s first |
| 17:38.51 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:38.56 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:39.22 | [TK]D-Fender | afink: And yes, the GUI makes a very messy bowl of spaghetti |
| 17:39.40 | afink | ^^ this looks better: http://pastebin.com/m5292f0bd |
| 17:39.46 | coppice | if spaghetti is really gui, you overcooked it |
| 17:40.28 | [TK]D-Fender | coppice: Roger... his goose is cooked. |
| 17:40.57 | *** join/#asterisk Sweeper (n=ppp@74.51.109.60) |
| 17:41.00 | coppice | BBQ goose with spaghetti sounds good. |
| 17:41.20 | Sweeper | blah. anyone got a working config with asterisk on a public IP and polycom 301 behind NAT? |
| 17:41.50 | [TK]D-Fender | Sweeper: "canreinvite=no", "nat=yes", "qualify=yes" for the SIP peer. that is all. |
| 17:42.02 | Sweeper | hmm |
| 17:42.12 | twitchnln | TK - got it figured out, thanks for your help :-) |
| 17:42.28 | theGoat | Fender: I see that some people are using a perl script to extract the info..... |
| 17:42.32 | [TK]D-Fender | afink: Yes, clearly much more compact. Now go make the tiered dial. |
| 17:42.46 | [TK]D-Fender | twitchnln: You're welcome |
| 17:43.04 | [TK]D-Fender | theGoat: From where? |
| 17:43.10 | redax | hm. if I try to call the city over the hicom300 I got isdn cause no.63 |
| 17:43.52 | redax | Cause No. 63 - service or option not available. unspecified. |
| 17:43.57 | theGoat | Here is one example http://blog.tmcnet.com/blog/tom-keating/asterisk/cnam-callerid-with-name-on-asterisk-using-reverse-phone-number-lookup.asp |
| 17:44.25 | Sweeper | man I hate that I have to freaking reboot the phone every time I change one param |
| 17:44.58 | *** join/#asterisk vlt (n=dm@suez.activ-job.com) |
| 17:45.33 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 17:46.33 | *** join/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fedc:10e) |
| 17:46.47 | *** join/#asterisk Airozi (n=user@200-171-41-207.dsl.telesp.net.br) |
| 17:46.52 | vlt | Hello. How can I suppress caller ID when dialling via a SIP-to-ISDN gateway (Patton)? "Set(CALLERID(name)=anonymous)" doesn't work here. |
| 17:47.01 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 17:47.18 | Sweeper | [TK]D-Fender: so when I set that up, and I dial, I get a bust from the pcom, and "Really destroying SIP dialog '22ec821d-f8f31aec-d1493843@192.168.2.233' Method: SUBSCRIBE" from asterisk |
| 17:47.23 | Sweeper | *busy |
| 17:47.40 | [TK]D-Fender | Sweeper: PB <-------- |
| 17:47.50 | [TK]D-Fender | ~pb |
| 17:47.51 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 17:47.56 | afink | [TK]D-Fender: Do I need to do something to be able to use a local channel since I am using 1.4? |
| 17:48.20 | [TK]D-Fender | afink: No. It is a channel type the same as it has been since the beginning |
| 17:48.35 | Sweeper | [TK]D-Fender: pb what tho? |
| 17:50.57 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 17:54.29 | bhodder | [TK]D-Fender: There is no way then to set an extension to be on the Page() parameter as well as be an extension to dial the same page() extension without receiving the call? |
| 17:55.23 | afink | anyone around who can shed a little more light on what TK was talking about with the tiered dialing using a local channel? My brain just isn't comprehending |
| 17:56.28 | bhodder | does anyone else know much about using the Page() application |
| 17:58.47 | Katty | wah! |
| 17:58.54 | Katty | bhodder: what about it, specifically |
| 17:59.55 | Katty | eppigy: symantec couldn't spell my first name to open a ticket :< |
| 18:00.16 | Katty | eppigy: so not only do i not have a ticket open, ir eally have to pee |
| 18:00.48 | redax | is there a way to send the dialed number by pieces over the E1 link ? |
| 18:02.13 | *** join/#asterisk AsteriskDom (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 18:03.01 | eppigy | Katty: :< |
| 18:03.26 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 18:03.47 | [TK]D-Fender | ^&%#$ing Windows Update Ambush |
| 18:03.53 | Katty | eppigy: here, hold onto this call for me. |
| 18:04.05 | *** part/#asterisk rosscoh (n=rosscoh@72.37.203.61) |
| 18:04.12 | Katty | deposits symantec troubleshooting problem on eppigy's lap |
| 18:04.53 | bhodder | I am trying to set up a Page() that includes some extensions that would also dial the page extension but exlude the originator from being called |
| 18:05.52 | eppigy | D: |
| 18:06.44 | Katty | bhodder: do what? |
| 18:07.01 | Katty | bhodder: english, please. |
| 18:07.28 | eppigy | i am sooooo tired |
| 18:07.30 | eppigy | holy cow |
| 18:07.46 | *** join/#asterisk empiric (n=empiric@116.71.56.108) |
| 18:09.06 | [TK]D-Fender | bhodder: Go build up the string of devices to call in the dialplan and check before you add to see if its for the calling device |
| 18:09.58 | Katty | eppigy: let's nap. |
| 18:10.50 | afink | [TK]D-Fender: Could you please give me a few more hints on the tiered dialing thing? I just don't get it yet. |
| 18:10.52 | bhodder | Ok |
| 18:11.09 | [TK]D-Fender | afink: PB it again, I lost in in the reboot |
| 18:11.16 | [TK]D-Fender | (or relink) |
| 18:11.32 | afink | http://pastebin.com/m5292f0bd |
| 18:11.40 | [TK]D-Fender | afink: And you sure yuo want to extra people to only ring for 5 s?\ |
| 18:11.53 | elguero | Deeewayne: I tried out the B channel service messaging with the switchtype being NI2. I ended up going into chan_dahdi and changing the conditional so that it would allow the use of service messaging with that switch type. I don't think that it worked even though I could see asterisk sending the maintenance message to the switch. I tried to disable all the channels on the one PRI. But when I called in, it would still come in on that PRI. I con |
| 18:12.00 | elguero | firmed that the database was showing it as being disabled by running database show at the cli. |
| 18:12.10 | elguero | Deeewayne: here is a pri trace http://pastebin.ca/1496047 |
| 18:12.15 | afink | [TK]D-Fender: they keep adding so some ring for 10 some for 20 |
| 18:12.32 | afink | but I'll try to figure that out once I get the syntax |
| 18:12.41 | [TK]D-Fender | afink: You've got typo's in your dial. Fix those and reorder so identical devices align on-screen (without whitespace) |
| 18:13.08 | [TK]D-Fender | afink: I se 10,15,20 logicals in there. thats more sane. |
| 18:13.23 | eppigy | Katty: yes please |
| 18:13.23 | [TK]D-Fender | afink: So clean it up and you win today's "freebie answer" |
| 18:13.25 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:13.33 | eppigy | Katty: what do you want to watch whilst we nap? |
| 18:13.38 | afink | lol k will do |
| 18:14.03 | redax | how can I _Call_ anything using overlapped dial? |
| 18:14.56 | empiric | guys i have this this dial plan http://pastebin.com/m15df1817 |
| 18:15.20 | empiric | 204 ext is configured to dial 200 |
| 18:16.04 | empiric | how i setup that when some one call from outside on this PSTN line that call should be transfer to 204 ext |
| 18:16.43 | [TK]D-Fender | empiric: exten => s,1,GotoIf($[${EXTEN}=200]?dialout,s,1) <--- will never = 200, and as far as we see [dialout] does not exist |
| 18:17.02 | empiric | it works |
| 18:17.04 | empiric | i did |
| 18:17.17 | [TK]D-Fender | empiric: that is a BROKEN dialplan line. |
| 18:17.29 | [TK]D-Fender | empiric: it will always return FALSE |
| 18:17.44 | [TK]D-Fender | (0) |
| 18:18.11 | empiric | see logs http://pastebin.com/mdc65d6c |
| 18:18.21 | empiric | Fenfer then waht to do |
| 18:18.35 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 18:19.10 | *** join/#asterisk keulin (n=cray@bne75-6-82-229-246-155.fbx.proxad.net) |
| 18:19.29 | empiric | ok true |
| 18:19.30 | empiric | [Jul 15 10:49:52] WARNING[2910]: chan_sip.c:15630 handle_response: Host '192.168.1.51' does not implement 'NOTIFY' |
| 18:19.38 | empiric | i reload asterisk |
| 18:21.11 | empiric | Fender can u guide |
| 18:22.38 | [TK]D-Fender | empiric: You are very unclear about what you want to change about what you're doing. |
| 18:23.08 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 18:23.30 | afink | [TK]D-Fender: http://pastebin.com/m1bbfe9ef think I got everything. helps to have them in line |
| 18:24.09 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 18:30.02 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:31.11 | *** join/#asterisk sproket (n=chatzill@199.85.124.154) |
| 18:32.35 | empiric | Fender |
| 18:32.45 | empiric | ? |
| 18:32.50 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:34.02 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:34.32 | [TK]D-Fender | afink: http://pastebin.com/ma4a7394 |
| 18:34.36 | [TK]D-Fender | afink: Merry christmas |
| 18:34.47 | empiric | Fender ? |
| 18:34.51 | empiric | plz guide me |
| 18:34.53 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:35.10 | [TK]D-Fender | empiric: Your request does not make sense. You are not beaing clear about what you want. |
| 18:35.13 | afink | [TK]D-Fender: wow thank you! |
| 18:35.22 | *** join/#asterisk sjobeck (n=Adium@69-30-99-139.dv1sn.easystreet.com) |
| 18:35.40 | empiric | i have 4 lines PSTN and 5 extensions |
| 18:35.47 | sproket | hi, i currently run zaptel + asterisk 1.4 and I am migrating to a new server using DAHDI and asterisk 1.4. I have no digium hardware (SIP only - this is a meetme server) and use the Sangoma voicetime USB for clocking. My question is, since ECHO CANCELLATION is configured per channel with DAHDI, what do I need to configure |
| 18:35.53 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 18:35.55 | empiric | i want 1 ext 204 is fixed for 1 PSTN line |
| 18:36.04 | empiric | no other ext can call from that PSTN |
| 18:36.10 | [TK]D-Fender | afink: The outer dial has the master timeout to cancel everything. all 4 local channels get called immediately, however they have varying delays before starting their own Dial's |
| 18:36.13 | sproket | with zaptel echo cancellation, I didnt specify channel numbers, etc. and just set the echo training, etc. options i needed. thanks for your help in advance. |
| 18:36.57 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 18:37.21 | [TK]D-Fender | empiric: pThe call comes FROM SIP/204. What od you want to restrict about it? |
| 18:37.58 | empiric | 204 call outbound calls from 1 particular PSTN line |
| 18:38.01 | theGoat | If I ever decide to build a new asterisk version, what files could I copy from the old src folder to the new one so I didn't have to go through ./configure and make menuselect again? |
| 18:38.10 | empiric | all 3 lines used my other ext |
| 18:38.40 | jplank | if asterisk only records the source CID of an outbound call by the outbound CID, how can I figure out what extension made the call? |
| 18:38.47 | *** part/#asterisk sjobeck (n=Adium@69-30-99-139.dv1sn.easystreet.com) |
| 18:38.47 | afink | [TK]D-Fender: thats awesome thanks again. |
| 18:38.47 | jplank | it must be possible of ARI can do it |
| 18:39.27 | *** join/#asterisk tdg911 (n=tdg911@75-131-246-217.static.slid.la.charter.com) |
| 18:39.39 | theGoat | Tk, did u see the name in the one sip debug event I had posted earlier? |
| 18:40.22 | empiric | Fender? |
| 18:41.08 | [TK]D-Fender | empiric: What "ext"? What other lines? |
| 18:41.49 | [TK]D-Fender | theGoat: If you are compiling a new * you HAVE to do ./configure IIRC and so forth |
| 18:43.21 | empiric | Fender |
| 18:43.55 | afink | empiric: not the way to get an answer |
| 18:44.55 | *** join/#asterisk ManxPower (n=manxpowe@141.sub-70-223-36.myvzw.com) |
| 18:47.06 | empiric | I have a Dlink FXO with 4 PSTN lines and i have 6 sip accounts , i want a PSTN line fixed for ext 204 when 204 dial out it picks that specific PSTN line |
| 18:47.41 | lirakis | neither asterisk 1.4 or 1.6 will follow multiple choice redirects correct? only the first contact choice? |
| 18:49.29 | [TK]D-Fender | empiric: What represents a "line" in there? |
| 18:49.55 | empiric | PSTN from which i dial to other networks like my cell |
| 18:50.43 | [TK]D-Fender | empiric: I'm sorry you seem incapable of clearly describing where the ins&outs are. I cannot help you. |
| 18:50.48 | empiric | only sip account dial each other is like a intrcom i need to dial on other networks , cellenor networks |
| 18:51.37 | ManxPower | empiric: That is what key systems do. PBXs pick the first available line. You should be able to fake it in the dialplan. Put the channel for that line and for exten 204 (I'm assuming analog extension) into a separate context |
| 18:52.07 | empiric | is did |
| 18:52.16 | ManxPower | "is did"???? |
| 18:52.37 | ManxPower | analog lines can't be DID (direct inward dial) |
| 18:52.47 | ManxPower | At least in Asterisk. |
| 18:53.01 | [TK]D-Fender | ManxPower: RUN FORREST RUN!!!! |
| 18:53.28 | ManxPower | [TK]D-Fender: Oh, I'm not going to waste very muchof my time on him |
| 18:53.54 | empiric | guys am sorry to disturb you |
| 18:54.09 | *** join/#asterisk Breyer (n=Breyer@ool-43540592.dyn.optonline.net) |
| 18:54.14 | ManxPower | If someone does not understand what they are being told then they should read the Asterisk Book 8-) |
| 18:54.36 | ManxPower | empiric: We usually only get disturbed when we can't get enough info to help someone. |
| 18:54.50 | empiric | ill explain |
| 18:55.14 | ManxPower | telecom is very hard to get right. One typo and you could open up your system to being used by random people and you get to pay their phone bill. accuracy matters. |
| 18:55.17 | empiric | see my dialplan |
| 18:55.18 | empiric | http://pastebin.com/m15df1817 |
| 18:55.26 | Corydon76-dig | You actually can do did with analog, just keep in mind that it's not oob |
| 18:55.30 | Breyer | can anyone recommmend any sip trunking providers with servers in the NE US? |
| 18:55.54 | Corydon76-dig | Just send the DTMF right after answer |
| 18:56.42 | ManxPower | empiric: you do not have a specific extension 204. You'll need to put extension 204 and the pstn line in their own context. That is the answer. |
| 18:56.51 | Corydon76-dig | Breyer: why would it matter where the servers are physically located? |
| 18:57.08 | Breyer | latency I suppose? |
| 18:57.09 | empiric | how |
| 18:57.10 | empiric | ? |
| 18:57.38 | ManxPower | empiric: It would take too much time for me to explain it to you. Read the Asterisk book and you should be able to do that fairly easily. |
| 18:57.40 | empiric | i make this context [204context] |
| 18:57.40 | empiric | exten => s,1,GotoIf($[${EXTEN}=200]?dialout,s,1) |
| 18:58.01 | Corydon76-dig | Breyer: then choose one using your provider. Physical location is irrelevant to network latency |
| 18:58.33 | ManxPower | empiric: why do you need a gotoif when the only extension is 203. You would not use a goto. [204context] exten => _NXXNXX,1,Dial(Zap/pstnlineport/${EXTEN}) |
| 18:58.41 | *** join/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746) |
| 18:58.50 | ManxPower | empiric: Now GO READ the book. |
| 18:59.10 | Breyer | well you get my meaning, I certainly have higher latency to servers in california |
| 18:59.11 | ManxPower | ~book |
| 18:59.12 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:59.23 | _Sam-- | does anyone have a recommendation for hardware phones w/ dual gigabit network ports? |
| 18:59.30 | Corydon76-dig | Breyer: I have machines in the same town that run 20 hops to me because the providers aren't directly connected. Likewise, there are servers halfway across the country within 20ms, because they're on the same backbone as me |
| 19:00.08 | Corydon76-dig | Breyer: again, physical location is irrelevant to latency |
| 19:00.28 | Corydon76-dig | Breyer: you want low network hops, not physical closeness |
| 19:00.39 | Breyer | okay well then do you know of any popular providers that I should look into |
| 19:00.52 | Breyer | you're right, thanks |
| 19:01.00 | ManxPower | Breyer: who provides your internet? |
| 19:01.15 | Breyer | NetAccess |
| 19:01.32 | Corydon76-dig | naxs.net ? |
| 19:01.34 | ManxPower | run traceroutes to the various providers and see which are closest |
| 19:01.50 | Breyer | nac.net |
| 19:01.56 | _Sam-- | (not that it matters, ive been happy with teliax. they have servers in ATL, NYC, DEN, LAX with pretty good connectivity.) |
| 19:02.17 | sproket | can anyone help me with how to configure DAHDI echo cancellation when I have no physical digium cards / channels (SIP only), I am trying to move from zaptel to dahdi. thanks |
| 19:02.19 | _Sam-- | check your routes to: atl.teliax.net or nyc.teliax.net (i remember NAC.net used to be in NJ) |
| 19:02.23 | ManxPower | Teliax is good. I use vitelity |
| 19:02.41 | leifmadsen | I've been using voiptrunk lately, and they seem pretty good too |
| 19:02.58 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 19:03.04 | Breyer | It's just that there are an OCEAN of providers out there and I have to have phone conversations with every one of them to get their proxy IPs :) |
| 19:03.08 | leifmadsen | (voiptrunk.com) |
| 19:03.09 | ManxPower | leifmadsen: I used like 60 mins /month of mins |
| 19:03.09 | Breyer | are/is |
| 19:03.27 | ManxPower | Breyer: Welcome to OUR world. We have to do the same thing. |
| 19:03.46 | Breyer | Well that's why I'm here asking, been doing it for the past couple days |
| 19:03.56 | Breyer | thanks for the welcome |
| 19:04.02 | Corydon76-dig | Breyer: then you want a provider with potentially a cross-connect with nac.net |
| 19:04.18 | Corydon76-dig | Ask for that specifically |
| 19:04.33 | Corydon76-dig | That's a simple yes/no |
| 19:04.54 | Corydon76-dig | Failing that, try a cross connect with tinet.net |
| 19:04.56 | ManxPower | Breyer: most of us don't care so much about this |
| 19:04.59 | _Sam-- | asterlink used to be on NAC.net |
| 19:05.37 | ManxPower | All providers suck. The trick is finding the ones that suck less that most. |
| 19:06.06 | _Sam-- | i like teliax -- their main guys is on my instant messenger is always helpful. i wouldnt say they suck. |
| 19:06.14 | leifmadsen | I like Unlimitel.ca because they don't resell SIP services from Global Crossing or Level 3 -- they have physical connections to the phone company. However they only offer Canadian DIDs |
| 19:06.15 | Breyer | Thanks all |
| 19:06.23 | leifmadsen | they are super rock solid |
| 19:06.30 | jplank | ManxPower: Telecom in general sucks, its not always the providers faults |
| 19:06.45 | ManxPower | jplank: it's sometimes the customer's fault? |
| 19:06.56 | _Sam-- | all of the VOIP providers did USE to suck -- they have gotten muchbetter in the last 5 years. |
| 19:07.10 | _Sam-- | more redundant, more servers, better connectivity. |
| 19:07.10 | jplank | I would say over 50% of the time its the customers fault |
| 19:07.14 | bmoraca | ManxPower: sometimes it's the people who designed the system's fault |
| 19:07.36 | jplank | I would say in general though, its the customers not understanding what they are getting, and the provider not understanding the customers needs |
| 19:08.12 | ManxPower | jplank: I would say in general that it is the lack of good support from a provider that is a major part of suckage. |
| 19:08.20 | bmoraca | jplank: yep...run in to that all the time...telco rep sells customer something that the customer doesn't understand/need and then they call us to try and make heads or tails of it |
| 19:08.40 | *** join/#asterisk tdg911 (n=tdg911@75-131-246-217.static.slid.la.charter.com) |
| 19:08.42 | thehar | leifmadsen: how is XO's sip trunking? happen to know? |
| 19:08.45 | ManxPower | If I wanted to be ignored I'd be using AT&T instead of a CLEC. |
| 19:08.51 | leifmadsen | no idea |
| 19:08.55 | thehar | hrmm |
| 19:09.45 | jplank | ManxPower: I can't disagree with you there |
| 19:13.15 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:15.57 | redax | basicly what's the difference btween 1.6.0.10 and 1.6.1.1 ? and why do we have 2 kind of 1.6 versions? |
| 19:16.12 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 19:20.07 | ManxPower | redax: 1.6.0.x gets no new features, only bugfixes. 1.6.1.x gets no new features, only bug fixes. |
| 19:20.20 | ManxPower | Digium made an announcement about it last year, but I don't remember where it might be. |
| 19:21.28 | Deeewayne | elguero, I'll check that out and get back to you |
| 19:21.49 | elguero | Deeewayne: excellent! it is greatly appreciated |
| 19:22.07 | jplank | spent 3 hours last night troubleshooting why I couldn't get g729 to passthrough, only to realize allow=g729 wasn't set inside sip.conf. fun fun. I guess thats what I get troubleshooting at 3am |
| 19:22.15 | TheOpenSourcerer | I have just discovered a small problem - I think due to my dialplan and use of a Macro. I am working on an AMI script and the Destination gets set to "s" in the Cdr Event even though I had a successful call up for 30secs or so. |
| 19:22.22 | TheOpenSourcerer | Any obvious suggestions? |
| 19:22.35 | redax | ManxPower: thx, will google for it |
| 19:22.40 | elguero | redax: not sure if this will help answer your question http://leifmadsen.wordpress.com/2009/06/06/about-the-new-asterisk-versioning-method/ |
| 19:22.47 | *** join/#asterisk korcan (n=korcan@99.23.50.73) |
| 19:23.32 | ManxPower | TheOpenSourcerer: that is the way macros work. I worked around the problem by having exten => s,1,Goto(${MACRO_EXTEN},1) and then have exten => _whateverpatternwouldmatch, the real stuff you want to do. |
| 19:23.33 | Breyer | ManxPower: do you know what vitelity's IPs are? |
| 19:23.42 | Breyer | or rather, can you tell me, please |
| 19:23.50 | ManxPower | Let me check |
| 19:24.51 | redax | elguero: thanks, |
| 19:25.17 | *** join/#asterisk [Rated-R] (n=rated-r@216.186.131.152) |
| 19:25.24 | ManxPower | inbound18.vitelity.net and outbound.vitelity.net is what is set on my account |
| 19:25.25 | TheOpenSourcerer | ManxPower: Thanks. I am not sure I understand though. |
| 19:25.31 | [TK]D-Fender | redax: Read: |
| 19:25.33 | Breyer | pretty solid connection to teliax they are on level3 which is directly connected to nac |
| 19:25.34 | [TK]D-Fender | ~astersikversioning |
| 19:25.35 | TheOpenSourcerer | I call the Macro so: exten => s,n,Macro(belllord,${ALANL}&${ALANB},303,TOLC) |
| 19:25.38 | [TK]D-Fender | ~asteriskversioning |
| 19:25.39 | infobot | it has been said that asteriskversioning is Information about the new Asterisk versioning method with the 1.6.x series is available here: http://www.asterisk.org/node/48602 |
| 19:25.41 | leifmadsen | elguero: additionally, I setup ~asteriskversioning for a quick key to that link |
| 19:25.47 | leifmadsen | heh :) |
| 19:25.55 | ManxPower | TheOpenSourcerer: The CDRs log the extensions the call goes to. If the call goes into a macro it goes to exten "s". So that is what the CDR logs. |
| 19:25.57 | Breyer | Thank you ManxPower |
| 19:26.29 | TheOpenSourcerer | ManxPower: But the Macro calls a "proper" extension. |
| 19:26.36 | ManxPower | ${MACRO_EXTEN} contains the original extension before you went into the macro. |
| 19:26.43 | TheOpenSourcerer | exten => s,n,Dial(${ARG1},20,t) |
| 19:26.53 | ManxPower | TheOpenSourcerer: there your extension is "s" |
| 19:27.06 | TheOpenSourcerer | Ahhh. Got you. |
| 19:27.07 | jplank | ManxPower: would there be anything wrong with using a desciptive word as the extension, and then using a goto to trigger it? Would that work? |
| 19:27.10 | TheOpenSourcerer | Thanks! |
| 19:27.31 | elguero | leifmadsen: Okay, thanks... I thought that I saw it on asterisk's blog but google brought me to yours |
| 19:27.40 | jplank | something like exten => sampleapp,1,NoOp(sample app!) |
| 19:27.42 | ManxPower | jplank: extensions don't have to be numbers. |
| 19:27.55 | leifmadsen | elguero: you're right -- it was on my blog first, and then I was asked to put it on the asterisk.org site too |
| 19:28.18 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 19:28.23 | jplank | I know they don't have to be numbers, but I dont have to stick with just hangup,s, ect, I can put whatever I want, without messing anything up? |
| 19:28.53 | ManxPower | jplank: other than messing with your CDRs, no. |
| 19:29.05 | [TK]D-Fender | jplank: exten =>becauseifeellikeit,1,NoOp(\o/) |
| 19:29.33 | redax | thanks D-fender |
| 19:29.34 | jplank | I mean to do that to help your CDR. using s as the goto to an extension thats a descriptive word |
| 19:29.36 | ManxPower | http://www.fnords.org/~eric/macro-std-exten-v2.inc |
| 19:30.07 | jplank | so you |
| 19:30.36 | ManxPower | Don't worry about the details of that link, just notice I send the call to a pattern inside the extension so the CDRs will be more correct. |
| 19:30.38 | jplank | so you would see the s extension in the CDR, then directly after that so the someapp extension |
| 19:31.48 | ManxPower | I need to rewrite that macro for 1.6 and maybe convert it to AEL |
| 19:32.11 | *** join/#asterisk psilikon (n=psilikon@rrcs-24-227-87-238.se.biz.rr.com) |
| 19:32.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:32.51 | jplank | "I send the call to a pattern inside the extension " can you point out what your talking about? I don't understand. |
| 19:34.26 | psilikon | I am trying to get more information as to what "overruns" and "underruns" are on a span of a Sangoma a104d. Can anyone point me in the right direction? |
| 19:34.27 | leifmadsen | ewwww AEL :) |
| 19:34.57 | leifmadsen | ManxPower: you might want to look into doing it with a GoSub(), just for fun :) |
| 19:35.32 | ManxPower | leifmadsen: I think a total rewrite would be the correct way. |
| 19:35.37 | ManxPower | And I LIKE AEL. |
| 19:35.46 | leifmadsen | ManxPower: you're weird :) |
| 19:35.55 | ManxPower | leifmadsen: you have no idea how weird. |
| 19:36.03 | leifmadsen | I've met you... :) |
| 19:36.15 | ManxPower | 8-) |
| 19:36.26 | ManxPower | Madrid doesn't count. I was drunk. |
| 19:36.53 | leifmadsen | who wasn't? |
| 19:37.48 | ManxPower | 8-0 |
| 19:38.16 | ManxPower | for anyone with polycoms: http://www.fnords.org/~eric/polycom-config-examples/ |
| 19:39.58 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 19:41.57 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:42.47 | carrar | who's Charlotte |
| 19:42.50 | carrar | she hot? |
| 19:42.53 | carrar | :) |
| 19:43.02 | carrar | she a spider? |
| 19:48.18 | *** join/#asterisk xp_prg (n=xp_prg3@99.2.31.217) |
| 19:48.34 | xp_prg | anyone familiar with tc for qos traffic shaping on linux? |
| 19:58.17 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 19:58.23 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 20:06.18 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 20:07.34 | errr | ManxPower: ping |
| 20:08.25 | *** join/#asterisk blebleble (n=godie@s7.n194.n222.n216.static.myhostcenter.com) |
| 20:08.36 | ManxPower | errr: pong |
| 20:08.49 | errr | ManxPower: would you mind if I pm you a question? |
| 20:09.52 | ManxPower | BTW everyone, I'm looking for employment. My Resume http://www.fnords.org/ewieling-resume.pdf |
| 20:09.55 | blebleble | maybe someone will be able to help with this, I have asterisk setup with openfire+asterisk-IM, (jabber im system that integrates with asterisk), everything is working well except when a user calls in the im notification is displaying the unique id instead of the caller id, i can see in the code their getting the value via 'request.getParameter("callerID");' anyone have any clue how i can make sure that value is set to the call |
| 20:10.23 | ManxPower | errr: if you don't want the question posted publicly |
| 20:10.31 | errr | thanks |
| 20:13.00 | rob0 | ManxPower: pm a non-tech question? |
| 20:14.27 | jplank | ManxPower: where do you live? |
| 20:14.38 | psilikon | ManxPower, you got a CCNA and/or Secret clearance? |
| 20:14.58 | ManxPower | psilikon: You are funny. |
| 20:15.10 | ManxPower | jplank: Huntsville, AL. |
| 20:15.22 | *** join/#asterisk ingenius (n=alektro@190.247.156.247) |
| 20:15.22 | psilikon | ManxPower, not trying to be funny. Just a a simple question. |
| 20:15.31 | jplank | CCNA makes me laught |
| 20:15.48 | psilikon | CCNA doesn't make employers laugh |
| 20:15.53 | ManxPower | I do not have a CCNA. as for Secret clearance I don't have that either. |
| 20:16.02 | ManxPower | CCNA could be useful. |
| 20:16.08 | jplank | CCNA is a joke |
| 20:16.29 | jplank | any job that *requires* a CCNA, is not a place that I want to work |
| 20:16.34 | jplank | same with A+ |
| 20:16.38 | psilikon | "CCNA is a joke" is often said from someone without it |
| 20:16.48 | jplank | <-CCNP before it expired |
| 20:16.59 | psilikon | A+ is a joke tho |
| 20:17.23 | ManxPower | A govt job would be my LAST resort, BTW. |
| 20:17.42 | Qwell | "'CCNA is a joke' is often said from someone without it" is often said from someone who realized after they got theirs that it's a joke and wants to save face. ;) |
| 20:17.55 | blebleble | heh |
| 20:18.10 | jplank | ManxPower: we might be needing someone soon for asterisk with light T1 (data) work, I don't think working remotely would be a problem (I do), I'll def keep you in mind |
| 20:18.16 | blebleble | mine expired about 7 years ago, and would agree with that statement |
| 20:18.19 | ManxPower | At least the Cisco certs actually require you to understand some stuff, iunlike many other certs |
| 20:18.33 | jplank | Qwell: LOL |
| 20:18.40 | DarthPointer | xp_prg? Which traffic shaping engine are you using in linux? |
| 20:18.45 | ManxPower | jplank: I'm mainly looking for full time work, but will consider one-off jobs |
| 20:18.51 | jplank | not one off |
| 20:18.54 | jplank | I'm talking full time |
| 20:19.03 | ManxPower | Oh! |
| 20:19.53 | ManxPower | jplank: I managed a data network for about 8 years, helped manage the data network for 2 years before. |
| 20:20.02 | DarthPointer | Anyone have any experience with Astribank USB attached "Channel Banks"? |
| 20:20.14 | jplank | I'll give you more info once I get the OK to start looking (and if your still looking), but the job will mainly consist of supporting, remotely configuring asterisk boxes, and turning up T1s |
| 20:20.24 | jplank | this is wan stuff though |
| 20:20.28 | ManxPower | jplank: right up my alley |
| 20:20.30 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 20:20.37 | ManxPower | I love WAN stuff. |
| 20:20.43 | jplank | as do I |
| 20:21.02 | ManxPower | I did far more WAN stuff than Asterisk stuff for many years |
| 20:21.28 | xp_prg | DarthPointer I am trying to use tc |
| 20:21.49 | xp_prg | for some reason my brain is just not understanding how it works |
| 20:22.14 | ManxPower | jplank: I'm looking for something before the last day of this month. |
| 20:22.21 | jplank | basically the WAN stuff consists of testing T1's inside juniper and cisco equipment, turning it up, helping tech configure the CPE we install, and helping out support if it gets over their head |
| 20:22.33 | DarthPointer | what problem are you having with it? |
| 20:22.48 | ManxPower | jplank: As long as nobody expects me to be a programmer or desktop support. |
| 20:23.02 | xp_prg | well the first problem I am having is that I am using openvpn, this sets up a tunnel interface tun0 |
| 20:23.03 | jplank | I don't know if I'll have the OK before the 31st, but either way, I'll give you the heads up as soon as I get the green light |
| 20:23.14 | ManxPower | jplank: Thanks. |
| 20:23.14 | jplank | nah screw that, we have IT for that, programming, we outsource |
| 20:23.15 | xp_prg | that is what I want to pass to as the tc dev parameter right? |
| 20:23.26 | DarthPointer | k; what device are you running it on, btw? DD-WRT? |
| 20:23.32 | DarthPointer | distribution, etc. |
| 20:24.15 | xp_prg | I am on centos which is a xen vm |
| 20:24.20 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 20:24.22 | xp_prg | I don't understand the question regarding dd-wrt |
| 20:24.55 | xp_prg | ifconfig shows tun0 and that is the interface for the openvpn |
| 20:26.08 | ManxPower | I'm outta here. |
| 20:26.28 | DarthPointer | (a common deployment is to use DD-WRT on commodity hardware, like linksys, what ever, it doesn't matter) |
| 20:27.15 | xp_prg | ok so I should use tun0 as the dev parameter to tc right? |
| 20:28.05 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 20:28.23 | voxter | have any of you guys evaluated any USB connected handsets for use with a softphone? |
| 20:28.35 | xp_prg | DarthPointer? |
| 20:28.38 | DarthPointer | that depends on what you're trying to do |
| 20:28.46 | DarthPointer | sorry, multitasking here |
| 20:28.49 | xp_prg | I am trying to limit bandwidth usage |
| 20:28.50 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
| 20:28.57 | xp_prg | to prevent isp overage charges |
| 20:29.09 | DarthPointer | you can see what's happing using: tc class show dev tun0 (for example) |
| 20:29.12 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 20:29.35 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 20:29.41 | xp_prg | DarthPointer nothing shows up when I do that |
| 20:29.56 | [TK]D-Fender | xp_prg: Why are you asking about that in here? |
| 20:30.22 | xp_prg | because traffic quality of service is integral to voip |
| 20:30.23 | [TK]D-Fender | xp_prg: You should be in ##linux , ##networking, or your distro channel... |
| 20:30.39 | xp_prg | I have tried but nobody helps me, they just don't have the experience |
| 20:30.58 | DarthPointer | I'll just pm you |
| 20:31.03 | xp_prg | ok thanks |
| 20:31.44 | [TK]D-Fender | xp_prg: You want to cut back BW to prevent overages. This clearly has nothng to with with *. And regardless of what you want to prioritize, that just a few ports. Config of that app has nothing to do with * beyond that |
| 20:31.48 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 20:32.16 | xp_prg | *sigh* yes, just trying to get some help |
| 20:34.13 | Kobaz | notworking |
| 20:37.59 | *** join/#asterisk alonzo (n=Miranda@79.111.160.56) |
| 20:38.40 | alonzo | registration server for use with multiple Asterisk servers - it's possible? |
| 20:38.40 | alonzo | how to do? |
| 20:42.21 | tzafrir_laptop | alonzo, some SER, I presume (opensips, kamailio) |
| 20:43.14 | alonzo | ok.. and how asterisk knows about registered users? |
| 20:46.47 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:46.52 | alonzo | tzafrir_laptop, do you use asterisk with opensips? |
| 20:47.14 | tzafrir_laptop | alonzo, not me. But I figure others here do |
| 20:47.26 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 20:47.47 | alonzo | but how? |
| 20:50.31 | beek | What are the requirements for DEVICE_STATE() to work properly in 1.6? In sip.conf I have call-lmit=32 and busylevel=1 yet I can tie up the device with a phone call and a DEVICE_STATUS returns 'NOT_INUSE' every time. |
| 20:50.46 | beek | This is driving me INSANE. |
| 20:58.48 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 21:01.53 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 21:02.18 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:02.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:05.20 | *** join/#asterisk OmiKrOn (i=x@unaffiliated/omikron) |
| 21:06.11 | OmiKrOn | hey.. do you guys know any softphone other than x-lite that allows you to add multiple sip accounts? |
| 21:07.14 | juanIMP | OmiKrOn: zoiper |
| 21:08.42 | DarthPointer | anyone seen this error before in asterisk: |
| 21:08.46 | DarthPointer | [2009-07-15 17:05:16] ERROR[10376] codec_dahdi.c: Failed to open /dev/zap/transcode: No such file or directory |
| 21:09.15 | DarthPointer | resulted in crash; the events leading up to it were |
| 21:09.16 | DarthPointer | [2009-07-15 17:05:16] WARNING[10376] chan_dahdi.c: Detected alarm on channel 91: Red Alarm |
| 21:09.17 | DarthPointer | [2009-07-15 17:05:16] WARNING[10376] chan_dahdi.c: Detected alarm on channel 92: Red Alarm |
| 21:09.45 | *** join/#asterisk sumasuma (n=root@64.92.218.196) |
| 21:09.48 | DarthPointer | on 73-95; |
| 21:10.01 | tzafrir_laptop | DarthPointer, you don't need codec_dahdi unless you have a transcoder card |
| 21:10.42 | tzafrir_laptop | as for the second message: that port seems to have red alarm: nothing connected to it? |
| 21:11.02 | DarthPointer | correct; 3 out of 4 ports in use |
| 21:11.46 | DarthPointer | so codec_daahdi is only used for transcoder cards? |
| 21:13.23 | DarthPointer | I'm using a Digium TE420 on the box (PCI_Express quad port) |
| 21:15.00 | DarthPointer | I am using; can I safely unload that module and then specify a noload? |
| 21:15.12 | *** part/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 21:15.20 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 21:15.24 | DarthPointer | s/using;/using zaptel;/ |
| 21:16.25 | *** join/#asterisk voxter (n=voxter@190.241.15.217) |
| 21:21.49 | *** join/#asterisk rue_mohr (n=Dennis@24.207.122.10) |
| 21:22.22 | rue_mohr | is the dahdi tx gain reversed, so that higher is actually louder? |
| 21:22.38 | rue_mohr | my channelbank does that, I think its just an oops |
| 21:23.23 | *** join/#asterisk Xetrov` (n=xetrov@c-98-244-130-238.hsd1.ga.comcast.net) |
| 21:24.29 | *** join/#asterisk [netman] (n=netman@158.Red-88-24-108.staticIP.rima-tde.net) |
| 21:29.31 | *** part/#asterisk Sweeper (n=ppp@74.51.109.60) |
| 21:31.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:31.36 | jplank | customers are weird, they don't want voicemail, they don't want their extension to fall back to another route or extensions, but get mad after X amount of rings, it gets a busy signal |
| 21:33.37 | *** join/#asterisk cusco (n=tralala@2001:0:53aa:64c:2491:d64f:2ac0:762d) |
| 21:33.42 | cusco | hi |
| 21:33.49 | cusco | how can I place a call on hold via cli? |
| 21:33.53 | rue_mohr | jplank, well duh :) |
| 21:34.09 | jplank | rue_mohr: isn't a + gain always louder, and a - gain lower? |
| 21:34.34 | jplank | thats always been my understanding with not only dahdi, but zaptel, and anything else I've ever had to set a gain on |
| 21:34.35 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:35.14 | rue_mohr | k |
| 21:35.36 | rue_mohr | I'm not set up for testing outgoing gains, you cant transfer to a milliwatt app |
| 21:36.11 | cusco | is that possible? |
| 21:36.21 | jplank | just put it at a crazy high level and call your cellphone |
| 21:37.02 | jplank | worst case you blow your module, but you know its working :P |
| 21:37.05 | cusco | hahaha |
| 21:37.08 | rue_mohr | yea, you make all your lines go stright to a milliwatt app, dial in, and check your voltage and dahdi_monitor levels |
| 21:37.32 | rue_mohr | no if you have a db meter its easy, a lot of ac voltmeters have dbm |
| 21:37.40 | jplank | you can just do a dahdi_monitor -vv or whatever, the higher the gain, the higher the bar should go |
| 21:38.02 | rue_mohr | let me ask you this, WHAT NUMBER IS 0dbm in the dahdi_monitor? |
| 21:38.15 | jplank | I can't answer that |
| 21:38.20 | rue_mohr | heh |
| 21:38.24 | jplank | I've never actually seen a number in dahdi_monitor |
| 21:38.26 | rue_mohr | I have before, I forgot the answer |
| 21:38.27 | jplank | I only get that bar |
| 21:38.40 | rue_mohr | and do you know where you shoudl set that bar? |
| 21:41.52 | rue_mohr | I'm the only person who actaully know the answer |
| 21:42.00 | rue_mohr | and I fogot it at home |
| 21:42.03 | rue_mohr | :/ |
| 21:42.23 | rue_mohr | kb1's answer is really close |
| 21:43.57 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 21:44.28 | rue_mohr | however it dosn't help much if half your system is sip and you cant monitor it |
| 21:45.06 | jplank | right |
| 21:45.42 | jplank | I think that bar half way would be 0db, but don't quote me on that |
| 21:45.58 | jplank | if I could find my miliwatt test number, I could tell you for sure |
| 21:46.24 | rue_mohr | its not half |
| 21:46.37 | rue_mohr | its basically full scale |
| 21:47.10 | jplank | found it (317-614-9960) |
| 21:47.27 | rue_mohr | no, 604 958 2011 |
| 21:47.29 | rue_mohr | :) |
| 21:47.31 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 21:47.35 | rue_mohr | what do you get? |
| 21:47.52 | jplank | about to test it |
| 21:47.52 | rob0 | 867-5309 |
| 21:48.00 | rue_mohr | if you have a pri, you get a real number from dahdi_monitor |
| 21:48.14 | rue_mohr | hah, we all know our 1mw eh? |
| 21:48.23 | jplank | I'm about to do it over a PRI |
| 21:48.29 | rue_mohr | cool |
| 21:48.37 | jplank | I've actually never ran dahdi_monitor over a PRI, |
| 21:48.51 | rue_mohr | thats how kb1 got his number |
| 21:48.52 | jplank | grrr, I'm going to have to push the call over a specific channel |
| 21:49.00 | rue_mohr | hah |
| 21:49.14 | rue_mohr | you can trace the call to find the channel |
| 21:49.20 | jplank | that 604 number doesn't work for me |
| 21:49.25 | jplank | yea, thats what I'm going to do |
| 21:49.28 | rue_mohr | its canada |
| 21:49.36 | rue_mohr | 1-604-958-2011 |
| 21:49.38 | jplank | ahhhh |
| 21:49.44 | rue_mohr | use yours :) |
| 21:49.46 | jplank | I am |
| 21:52.23 | jplank | the box I have with a PRI is zaptel not dahdi, do you still want the results |
| 21:52.31 | jplank | I don't think I have any dahdi boxes with PRI, just POTS |
| 21:53.11 | jplank | your right, full bar |
| 21:55.15 | jplank | going to eat dinner, wow, I don't have any work to do tonight, nice! later all |
| 21:56.52 | *** part/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fedc:10e) |
| 21:57.53 | rue_mohr | heh, you need to do it with a pri to get a true 0db, pots lines have drop in them |
| 21:57.56 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 21:58.03 | rue_mohr | my house has -4.3db |
| 22:03.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:12.40 | vlt | Hello. Which dialplan function creates a SIP header that tells a SIP/ISDN gateway (like Patton SmartNode) to suppress the caller ID? |
| 22:14.26 | jplank | privacy |
| 22:15.05 | jplank | exten => _X.,n,SipAddHeader(Privacy: none) caller ID |
| 22:15.15 | jplank | exten => _X.,n,SipAddHeader(Privacy: id) no caller id |
| 22:16.22 | vlt | jplank: Thank you. |
| 22:16.23 | jplank | I *think* that needs to be used with a p-asserted-ID (RFC3325) but I could be wrong |
| 22:24.34 | *** join/#asterisk Alfio (n=Amunoz@190.6.138.119) |
| 22:26.07 | rue_mohr | k, I'm lost again, where is the config in a polycom that says that when the handset is picked up to light the line lamp and fake a dialtone as opposed to doing nothing |
| 22:32.38 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
| 22:33.43 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:34.36 | *** join/#asterisk theGoat (n=jircii@c-71-224-180-83.hsd1.pa.comcast.net) |
| 22:43.32 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 22:45.23 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 22:53.36 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
| 22:55.45 | bmoraca | rue_mohr: far as I know, that's default behavior |
| 22:56.41 | *** join/#asterisk Rob3Rt (n=admin@87.37.96.58.static.exetel.com.au) |
| 22:56.55 | Rob3Rt | heloo |
| 22:57.14 | Rob3Rt | i cant hear callers, is that usually a port issue? |
| 22:57.16 | jasonwoot | Does zap try to use the D channel on a T1 unless told not to? chan_zap.c:2401 pri_find_dchan: No D-channels available! Using Primary channel 48 as D-channel anyway! |
| 22:57.35 | Rob3Rt | i have, allow=g729,ulaw,alaw,gsm in conf |
| 22:57.57 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 23:00.36 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 23:02.27 | *** join/#asterisk srbaker (n=srbaker@24.222.125.139) |
| 23:02.29 | srbaker | heya folks. |
| 23:02.43 | srbaker | any recommendations for a provider that has canadian DIDs? |
| 23:04.02 | [TK]D-Fender | srbaker: voip.ms has remarkably good rates |
| 23:04.12 | *** join/#asterisk sjobeck (n=Adium@63.227.212.111) |
| 23:04.17 | srbaker | i signed up for a link2voip account |
| 23:04.24 | [TK]D-Fender | srbaker: les.net is pretty solid, unlimitel is great quality, but costs a bit more |
| 23:04.30 | srbaker | but they want me to photocopy my driver's license and credit card before they'll take my money |
| 23:04.44 | *** part/#asterisk sjobeck (n=Adium@63.227.212.111) |
| 23:04.50 | Rob3Rt | omg [TK]D-Fender do u ever sleep ? |
| 23:04.58 | *** join/#asterisk citywok (n=chatzill@67-148-102-3.dia.static.qwest.net) |
| 23:05.01 | Rob3Rt | ur undead arent |
| 23:05.02 | Rob3Rt | u |
| 23:05.11 | Rob3Rt | i knew it. |
| 23:05.15 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:05.58 | srbaker | whoa |
| 23:06.01 | srbaker | voip.ms seems awesome |
| 23:07.38 | srbaker | blah |
| 23:07.50 | srbaker | voip.ms doesn't have DIDs in the small town i'm in :P |
| 23:08.38 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
| 23:09.16 | [TK]D-Fender | srbaker: Its not that you're small or inimportant... |
| 23:09.22 | [TK]D-Fender | srbaker: Umm... what was your name again? |
| 23:09.30 | srbaker | [TK]D-Fender: Steven |
| 23:09.38 | [TK]D-Fender | srbaker: Whoever :p |
| 23:09.54 | eppigy | oh boy |
| 23:14.55 | srbaker | ? |
| 23:17.39 | Rob3Rt | is there a free win32 implementation of the g729 codec? |
| 23:22.44 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 23:29.53 | drmessano | There is no free implementation of G729 |
| 23:30.17 | iflux | there's an implementation that's free in europe but not in the us, right? |
| 23:30.17 | Sargun_screen | free as in beer, or free as in speech |
| 23:30.56 | drmessano | Free in countries where the patent holders of G729 have no claim.. |
| 23:31.22 | Rob3Rt | http://asterisk.hosting.lv/ |
| 23:31.28 | drmessano | Which is like sealand, parts of Kampuchea, and ..... |
| 23:31.33 | drmessano | Please dont post that link here |
| 23:31.39 | Rob3Rt | free versions, not for commercial use or distribution is my understanding |
| 23:31.54 | drmessano | Thats not a "free implementation" no more than a link to Pirate Bay is free software |
| 23:32.18 | Rob3Rt | unless ur an op |
| 23:32.20 | drmessano | "Not for commercial use" is not correct |
| 23:33.26 | drmessano | http://www.voip-info.org/wiki/view/Asterisk+G.729+Licensing |
| 23:33.51 | drmessano | You can google for the use of the codec as easily as the download.. somehow that part of the process always gets missed |
| 23:33.53 | Rob3Rt | please dont post that link here |
| 23:34.01 | Rob3Rt | that's blatant advertising |
| 23:34.12 | drmessano | Advertising of what? |
| 23:36.05 | ehsjoar | Reading that link it would be free to use the binaries in a non-commercial setting |
| 23:36.19 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 23:36.20 | drmessano | Negative |
| 23:36.26 | Rob3Rt | prove it |
| 23:36.40 | Rob3Rt | dont go stating "NEGATIVE" without firther proof |
| 23:36.59 | drmessano | Its all there in the link I posted.. unless you cant fucking read |
| 23:37.02 | Rob3Rt | just so you know, when you provide your proof, im still going to disagree with you |
| 23:37.23 | Rob3Rt | nah go fuck yourself bitch :) |
| 23:37.42 | ehsjoar | Okay, very creative stuff guys. I am off |
| 23:37.46 | drmessano | Of course you are.. So to answer your question, no there is no G729 DLL so you can run AsteriskWin32 on your Vista box, you jackass |
| 23:37.54 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 23:38.12 | Rob3Rt | there is, and ive got it, and its a test situation newb |
| 23:38.14 | drmessano | Get a real system and pay the $10 per channel for the codec, or one of the other 20 or so asterisk supports natively |
| 23:38.24 | Rob3Rt | ive got a gentoo system ready for it |
| 23:38.36 | drmessano | But youre testing on Windows? |
| 23:38.44 | Rob3Rt | dont tell me what to spend my money on who the fuck do u think u are? |
| 23:38.47 | drmessano | You sound like the newb |
| 23:38.59 | Rob3Rt | at least im not a wanna be doctor :) |
| 23:39.15 | drmessano | Shows how stupid you are |
| 23:39.28 | drmessano | Wait, already proven |
| 23:39.40 | drmessano | Good luck with Vista.. hope it works for ya |
| 23:39.42 | Rob3Rt | u wanna be a doctor so u can justify ur poor treatment of others with a cloak of pretend superiority |
| 23:39.51 | Rob3Rt | its working fine, for testing :) |
| 23:39.53 | Rob3Rt | THANKS |
| 23:39.55 | drmessano | I don't have to pretend to be superior |
| 23:40.00 | drmessano | I KNOW I am |
| 23:40.02 | *** mode/#asterisk [+b *!*n=admin@*.37.96.58.static.exetel.com.au] by Qwell |
| 23:40.02 | *** kick/#asterisk [Rob3Rt!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 23:40.12 | rob0 | whew |
| 23:40.14 | Chainsaw | Thanks. |
| 23:40.25 | Qwell | rob0: lucky I hit tab twice :P |
| 23:40.32 | drmessano | heh |
| 23:40.34 | rob0 | It was getting painful with that guy here for days on end. HAHA |
| 23:40.57 | Qwell | now, to clear things up |
| 23:41.05 | Qwell | That codec *CANNOT* be used legally. Period. |
| 23:41.07 | bmoraca | dude...that was better than cable. |
| 23:41.16 | Qwell | You either violate the patents, or you violate the code licensing. |
| 23:41.40 | drmessano | Unless you're on Sealand |
| 23:41.47 | drmessano | ITS LEGAL ON SEALAND |
| 23:41.52 | Qwell | Doesn't Sealand respect copyright? |
| 23:41.55 | bmoraca | everything's legal on sealand |
| 23:42.28 | drmessano | Qwell: They respect nothing on Sealand.. all 2 of them |
| 23:43.35 | Qwell | it's like 30 |
| 23:44.20 | bmoraca | it'd have been cool if Pirate Bay had bought them a couple years ago...turned it into a giant off-shore datacenter...that'd be neat |
| 23:44.43 | Qwell | it is a big DC :p |
| 23:45.22 | bmoraca | wonder how much it'd cost to get a server there... :P |
| 23:48.37 | drmessano | Rob3Rt is pretty angry |
| 23:48.43 | Chainsaw | bmoraca: If people get annoyed enough they'll unplug the fibre or the power. |
| 23:48.58 | Chainsaw | bmoraca: I'm not convinced it's sustainable. |
| 23:49.25 | bmoraca | Chainsaw: as long as they pay the companies they connect to, there's nothing wrong |
| 23:50.00 | bmoraca | drmessano: what channel is he in? |
| 23:50.12 | Chainsaw | bmoraca: If you consistently generate abuse@ reports and don't act on them, your upstream will disconnect you. |
| 23:50.21 | drmessano | I PM'ed him to insult him.. and I think I made him angrier |
| 23:50.28 | drmessano | He's typing all caps now |
| 23:50.28 | bmoraca | oh, lol |
| 23:50.35 | bmoraca | piss on the fire |
| 23:50.37 | rob0 | haha, just leave it be |
| 23:50.51 | rob0 | you meanie :) |
| 23:51.01 | drmessano | He insists he's only testing on Windows and really is going to use GENTOO |
| 23:51.05 | drmessano | REALLY |
| 23:51.06 | drmessano | HE IS |
| 23:51.06 | Sargun_screen | Chainsaw: you can always get more upstreams |
| 23:51.13 | Sargun_screen | people will just drop your advertisements. |
| 23:51.48 | drmessano | I need to go tell him his mother uses BeOS.. BRB |
| 23:52.40 | Chainsaw | Sargun_screen: *nod* In the end you'll run out of upstreams or money though. |
| 23:55.29 | drmessano | [19:55] <Rob3Rt> i didnt know aster was made by digium (is it) ? <-- Some people just need a bigger cluebat |
| 23:57.16 | bmoraca | havenco.com is expired :( |
| 23:57.18 | ManxPower | spammers might put up with changing upsteams all the time, people tend not to be as forgiving (no spammers are not people) |
| 23:57.54 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:57.58 | Slade- | hey is there a good open source tts out there for asterisk? |
| 23:58.02 | KavanS | lol |
| 23:58.04 | KavanS | sealand |
| 23:58.08 | KavanS | I thought they abandoned it... |
| 23:58.15 | Chainsaw | Slade-: Good or open source. Pick one. |
| 23:58.22 | Slade- | how about one of each? |
| 23:58.26 | KavanS | and sealand won't work anyways....they have to get upstream internet from somewhere |
| 23:58.29 | Chainsaw | Slade-: Open source: Festival. |
| 23:58.37 | Chainsaw | Slade-: I'll wait for Fender to come up with a good one. |
| 23:59.33 | Slade- | Cepstral sounds semi interseting |
| 23:59.49 | ManxPower | Slade-: there is an inexpensive TTS out there. Cepstral. |