| 00:00.53 | ManxPower | put context=une-incoming in [general] so incoming SIP calls with no entries in sip.conf will land in the right place. Obviously you should be fixing the actual problem. |
| 00:01.21 | ManxPower | And the actual problem is that your ISTP is not connecting to your server as the user "une_peer" |
| 00:01.37 | legis | ManxPower: yeah, they are not sending une_peer |
| 00:01.56 | ManxPower | legis: so it will NOT match the entry [une_peer] |
| 00:02.19 | legis | ManxPower: How can tell what are they sending? |
| 00:02.44 | ManxPower | legis: You would have to look at SIP debug. All I know is that they are not sendin the user you are expecting them to send. |
| 00:02.49 | legis | legis: they don't use asterisk or similiar stuff |
| 00:03.24 | ManxPower | legis: you realize if you had the correct stuff in [general] it might have worked. |
| 00:03.38 | legis | ManxPower: let me try it |
| 00:04.31 | *** join/#asterisk nhuisman_work (n=nhuisman@aeko.ifa.hawaii.edu) |
| 00:04.37 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 00:04.51 | nhuisman_work | hey anyone work for digium in the chan, curious where on the website I would go to download the latest iso release. |
| 00:04.54 | nhuisman_work | for asterisk be |
| 00:05.07 | ManxPower | nhuisman_work: have you tried asterisk.org |
| 00:05.17 | ManxPower | nhuisman_work: You must contact Digium for BE questions |
| 00:05.31 | nhuisman_work | yeah I know but some of them lurk here, figured i'd ask before calling |
| 00:05.53 | ManxPower | nhuisman_work: the ones that lurk here will tell you to contact Digium. |
| 00:06.28 | nhuisman_work | fine.. heh |
| 00:12.47 | legis | ManxPower: 19:03 < ManxPower> legis: you realize if you had the correct stuff in [general] it might have worked. |
| 00:12.59 | legis | ManxPower: you mean in extensions.conf? |
| 00:14.40 | *** join/#asterisk metfan2007 (n=jc@189.135.203.16) |
| 00:16.46 | metfan2007 | Hi all! I having some disposition issues, on a no answer state I get "Call failed to go through, reason (3) Remote end Ringing", and "FAILED" in CDR, using chan_zap |
| 00:16.50 | metfan2007 | any idea? |
| 00:18.42 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 00:19.54 | heedly | metfan2007: no one answered I think. |
| 00:20.18 | *** join/#asterisk snowy_owl (n=snowy_ow@201-048-005-105.static.ctbctelecom.com.br) |
| 00:20.31 | exothermc | If both ends of the call are not behind NAT, but one is. Is it necessary to proxy the media? |
| 00:20.41 | snowy_owl | yes |
| 00:20.55 | metfan2007 | :S |
| 00:21.25 | snowy_owl | both .. are not behind... but one yes..... A little bit weird your situation, no? |
| 00:21.54 | joat | need proxy for inbound |
| 00:21.54 | Qwell | yeah, umm.. might need to clarify that |
| 00:22.02 | joat | or port forwarding |
| 00:22.03 | exothermc | snowy_owl: Isn't that the pretty standard, client -> gateway -> PSTN gateway ? |
| 00:22.29 | exothermc | err rather client -> nat -> proxy -> PSTN gateway |
| 00:22.42 | snowy_owl | yes, it would be |
| 00:22.59 | snowy_owl | but.. just your question was not a happy question :) |
| 00:23.04 | snowy_owl | anyway, it will be necessary |
| 00:23.24 | snowy_owl | using stun, maybe. It can help withou proxying the media |
| 00:23.33 | exothermc | snowy_owl: What is gained with the rtp proxy? The fact that it performs symmetric rtp? |
| 00:24.45 | snowy_owl | gained? It solves your problem to nat. You will not gain anything more than just being a new mirror. Your carriers will receive rtp just from your pc |
| 00:25.01 | exothermc | snowy_owl: It would just seem that as long as the RTP is being handed off to another port (and by implication possibly another IP) it could be any random rtp end point. |
| 00:25.29 | snowy_owl | another thing: will gain more and more packts in your network :) |
| 00:25.43 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583027.dsl.bell.ca) |
| 00:25.58 | exothermc | snowy_owl: Ya I'm just trying to find a way to off load 230mbit of rtp. |
| 00:26.12 | snowy_owl | aeeeeeeeeee |
| 00:26.14 | snowy_owl | :) |
| 00:26.30 | snowy_owl | you must be a happy person :) |
| 00:27.00 | snowy_owl | the most number of your clients are behind nat? |
| 00:27.25 | exothermc | I guess to answer my own question though. The RTP proxy would need to be configured to listen for the UA and not send to the private IP contained in the SDP, and most end points aren't configured like that. |
| 00:27.36 | snowy_owl | of course!!! Because your are here, no? :) of course |
| 00:27.52 | exothermc | lol |
| 00:28.52 | snowy_owl | it starts listen to UA. After receiving the first packet, the rtpproxy knows the IP:port of your nat |
| 00:29.34 | snowy_owl | so, when it receives the packets from the side B, it forwards them to side A for your nat |
| 00:30.07 | exothermc | snowy_owl: And you are saying that most non proxies aren't set to listen but work off the SDP? |
| 00:30.49 | exothermc | snowy_owl: because technically the PSTN media gateway could perform that same function. |
| 00:31.34 | snowy_owl | the proxy dont need to rewrite the SDP message |
| 00:31.48 | exothermc | and save me $3k in bandwidth costs ;) |
| 00:31.59 | snowy_owl | sip proxy will forward your INVITE and the pstn will send rtp packets directly to you |
| 00:32.02 | snowy_owl | and vice-versa |
| 00:33.03 | exothermc | snowy_owl: Yes but what I'm saying is what is the main implementation difference between the rtpproxy and media gateway rtp server? |
| 00:34.52 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:36.21 | legis | ManxPower: I put context=une-incoming in [general] in sip.conf but same error |
| 00:36.40 | snowy_owl | exothermc: do u use how many machines? |
| 00:37.06 | exothermc | snowy_owl: I don't follow. |
| 00:39.13 | snowy_owl | i mean, how many machines to handle all this rtp traffic |
| 00:40.05 | snowy_owl | Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down? |
| 00:40.11 | exothermc | snowy_owl: I'm not sure where the cluster is now, but last I checked that was 8 machines or so. |
| 00:40.36 | snowy_owl | very nice |
| 00:42.26 | snowy_owl | without using stun, i dont see other way to save your bandw. However, it will not work for all them, neither for 20%. |
| 00:42.52 | snowy_owl | the world is a sad place... |
| 00:44.59 | snowy_owl | if you find a carrier that accept an invite with an private IP in sdp, it can make your business ($$) too happy |
| 00:46.28 | exothermc | snowy_owl: Ya it would be nice if there was something that could be sent in the SDP that would just say "hold RTP till we send our first RTP then just reply to that IP and port." |
| 00:48.44 | snowy_owl | but it is true. If you dont want to hanld the rtp, u dont need to force it through your network. Just relay the invite to your carrier. But the major number will deny this request |
| 00:48.57 | snowy_owl | *handle |
| 00:50.42 | snowy_owl | another point: when will buy the service, the interconnection, they will ask: "whats your rtp gw". |
| 00:50.55 | snowy_owl | this way, you must to force rtp. |
| 00:51.38 | snowy_owl | but 3K$ is cheap for your bandwidth |
| 00:51.54 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 00:52.26 | snowy_owl | come to brazil, hohoohh, and you will see what is the hell. |
| 00:54.02 | *** join/#asterisk ldiamond (n=ldiamond@bas15-montrealak-1178060550.dsl.bell.ca) |
| 00:54.28 | ldiamond | In order to use asterisk, I need some sort of adapter to plug a telephone? |
| 00:54.55 | exothermc | snowy_owl: Ya our expensive transit is only $16/mbit. All of our carriers only care about our SIP end points since that is the signaling/auth. It doesn't make much sense to try to check the location of the rtp gw, since that is all controlled via SIP anyway. |
| 00:55.13 | jblack | or you can use a soft phone hooked up to your computer, or you can get an IP phone. |
| 00:55.19 | exothermc | ldiamond: No, but that would all depend on how you want to use it. |
| 00:55.20 | jblack | But obviously, you need something to talk into. |
| 00:56.25 | exothermc | jblack: I prefer a mirror. |
| 00:56.52 | ldiamond | well, right now there are a few phones in here, and I wanted to setup asterisk to be able to have something like extensions |
| 00:57.35 | exothermc | ldiamond: In that case you need to either move to softphones, buy new SIP/IAX hardware, or buy ATA devices to interface asterisk to your hardware. |
| 00:58.08 | exothermc | ldiamond: I guess I left out buying a card that acts like a modem. Never used one of those before though. |
| 00:58.13 | ldiamond | and that ATA interface costs about how much? |
| 00:58.30 | exothermc | ldiamond: you can get them starting at $30/unit. |
| 00:58.50 | ldiamond | and what's the exact name of that, so I can look online for more info? |
| 00:59.18 | legis | linksys pap2 :P |
| 00:59.30 | exothermc | grandsteam handytone |
| 01:00.10 | exothermc | if you are going to drop for a pap2 you may as well get a SPA2102 which will help with faxing if you can get t.38 figured out. |
| 01:00.19 | *** part/#asterisk korihor (n=korihor@201.211.168.130) |
| 01:00.35 | ldiamond | ok, but thats only 2 ports |
| 01:00.51 | ldiamond | there are 10 phones, and I want each to have an extension |
| 01:01.04 | ldiamond | err, maybe just 5 actually |
| 01:01.18 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:01.27 | exothermc | Well that will be cheaper than the alternative |
| 01:02.08 | ldiamond | well, I'd have to buy 3 of those? |
| 01:02.09 | exothermc | Are you looking to build it out right, or do you have a roll of duct tape you plan on using here? |
| 01:02.41 | ldiamond | atm, I'd go with the duct tape thing |
| 01:02.54 | exothermc | Then 3ATAs will be your best bet |
| 01:03.04 | ldiamond | otherwise |
| 01:03.12 | exothermc | you may way to look at the analog cards, but I don't have a clue on their prices. |
| 01:03.47 | exothermc | If you want to do it right then you'll go grab some polycom soundpoint phones and make the users happy. |
| 01:04.09 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a4877e4020769147) |
| 01:04.39 | exothermc | because with extensions comes the expectation of wanting to transfer, put people on hold etc. Yes you technically can do all that with the other but it is more complicated and users are inherently stupid. |
| 01:04.43 | ldiamond | nothing cordless? |
| 01:05.48 | exothermc | ldiamond: There are a few cordless phones out there, but IMHO polycom is the only one that figured those out for businesses. |
| 01:06.06 | ldiamond | well, its not for a business tho |
| 01:06.16 | exothermc | There are many other brands that can make a desk phone, but their wireless phones are crap. |
| 01:06.36 | ldiamond | and the only way to interface a regular phone is to use the pap2 ? |
| 01:07.11 | exothermc | ldiamond: No you can use an analog card that will give you rj11 ports out the back of your computer. |
| 01:07.40 | ldiamond | And what term do I use to google this? |
| 01:08.20 | WimpMan | fxs pci |
| 01:09.18 | ldiamond | ty |
| 01:10.33 | exothermc | When it is listed as FXX does that mean it swings both ways? |
| 01:11.16 | WimpMan | Haven't seen that, but I'd assume the same. |
| 01:11.18 | ldiamond | damn, they seem expensive |
| 01:11.46 | *** join/#asterisk gambolputty (n=BC43599@cpe-24-175-90-49.tx.res.rr.com) |
| 01:11.56 | *** join/#asterisk rasterix (n=IceChat7@host81-155-2-150.range81-155.btcentralplus.com) |
| 01:12.46 | ldiamond | There are some at 100$ and some at 1000$ |
| 01:13.08 | WimpMan | You could get away a lot cheaper by using a bri card and a terminal adaptor, but maybe you'd have to get them in Europe. |
| 01:14.25 | LemensTS | do offices usually use the line buttons, or do they park a call and tell people to pick up that parked number? My office does parked calls... |
| 01:14.55 | rasterix | will isdn2 even live beyond the next year or so |
| 01:15.10 | rasterix | even in europe i doubt it |
| 01:15.14 | WimpMan | rasterix: Is there any alternative? |
| 01:15.34 | rasterix | well it looks to me like they are phasing out |
| 01:15.38 | rasterix | in the uk |
| 01:15.41 | rasterix | at least |
| 01:15.59 | LemensTS | hell most places around here are still on dial up |
| 01:16.04 | rasterix | bri was a failure |
| 01:16.12 | WimpMan | Well, indeed real phone lines will probably disappear within the next five years, but people will still use ISDN terminals, I guess. |
| 01:16.29 | rasterix | not if u cant get an isdn line they wont |
| 01:16.55 | WimpMan | rasterix: It wasn't. In fact it's the best solution so far, unless you need larger volumes. |
| 01:17.05 | LemensTS | if i get a 4 line polycom, someone can place a line on hold, then someone else just hits that line button to answer it+ |
| 01:17.06 | LemensTS | ? |
| 01:17.12 | rasterix | wimpman agree to disagree? |
| 01:17.28 | WimpMan | Well there are IADs that support ISDN. |
| 01:17.47 | rasterix | we carry the argument on in 3 years |
| 01:17.56 | rasterix | when isdn2 is dead |
| 01:17.56 | WimpMan | Well, we're from different centuries :-) |
| 01:18.19 | ldiamond | I just found out that I have some cheap USB phones that support h323 |
| 01:18.56 | rasterix | rasterix=21st centutry... wimpman=.... :P |
| 01:19.03 | ldiamond | but its pretty useless since you must have a computer running and it prolly only works on windows |
| 01:19.11 | WimpMan | There are telcos that don;t offer analogue lines. But you can get an NT with built-in TA, so it looks like one. |
| 01:20.03 | rasterix | omg i had a bad experience today |
| 01:20.20 | rasterix | i was developing with a third party control |
| 01:20.42 | WimpMan | Well, most people still live in the 19th centory of telecommunications, obviousely. That's why they think vpoip is the way forward. But when you didn't miss the 20th century, it's a step back. |
| 01:20.57 | rasterix | i edited their config files and when i reinstalled with the full license it wiped out all the config files |
| 01:21.11 | rasterix | two weeks work gone... how i laughed >< |
| 01:22.53 | rasterix | should have backed everything up... id sack people for being as incompetent as me :| |
| 01:23.58 | rasterix | wimpman: dont be so down... voip is the way forward... just like isdn2 was |
| 01:23.59 | LemensTS | backups of your backups ;) |
| 01:24.25 | WimpMan | I agree in theory. |
| 01:24.42 | WimpMan | But in practice it still needs a lot of work. |
| 01:24.46 | rasterix | longterm... maybe really LONGTERM |
| 01:24.54 | rasterix | voip is the answer |
| 01:25.08 | rasterix | bandwidth will become an inrelevant argument of the past |
| 01:25.27 | outtolunc | not if comcast engulfs the world <G> |
| 01:25.32 | rasterix | just like processor power gradually fades from being a factor for asterisk |
| 01:25.52 | WimpMan | Actually I VERY rarely experience bandwidth issues, and I'm all G711. |
| 01:26.04 | rasterix | i get them |
| 01:26.32 | WimpMan | But people designen voip protocolls should really have take a closer look at what's available in the telco world now. |
| 01:26.44 | WimpMan | designing |
| 01:26.56 | rasterix | wimpman: maybe so |
| 01:27.06 | rasterix | but long term it WILL be voip |
| 01:27.35 | WimpMan | Sure. It has to be, if just for the cost of the equipment. |
| 01:28.06 | rasterix | take the last 5 years |
| 01:29.09 | rasterix | in the uk we went from modems offering 8kbps to isdn to 8mbps on adsl |
| 01:29.29 | rasterix | in 10 years |
| 01:29.52 | rasterix | bandwidth will be something old men in cardigans talk about |
| 01:29.56 | WimpMan | Don't compare telephone technique to internet. That doesn't make sense any more. |
| 01:30.14 | LemensTS | You can do outgoing groups with analog, but not sip right? |
| 01:31.07 | LemensTS | (in case the first line is busy it will grab the next in the group) |
| 01:31.42 | WimpMan | LemensTS: There's no reason you can't place several concurrent calls on a single SIP account. |
| 01:32.04 | WimpMan | Unless your provider disallows, off course. |
| 01:32.27 | mchou | then switch ITSP if they has such limitations |
| 01:32.38 | LemensTS | wimpman: yea if you have multiple trunks, but i mean if it didnt |
| 01:33.15 | WimpMan | Maybe I don't understand the question? |
| 01:33.19 | rasterix | i have a question... and i should just ask it rather than saying i have a question |
| 01:33.59 | rasterix | what is the most common mistake everyone makes in their extensions.conf? |
| 01:34.42 | WimpMan | Getting the priorites mixed up when you change something? |
| 01:34.44 | rasterix | for example an extension with duplicate priorities |
| 01:35.03 | brodiem | ~itsplist-us |
| 01:35.04 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 01:35.13 | WimpMan | double, gaps, or not starting with 1. |
| 01:35.41 | rasterix | i always use n |
| 01:35.47 | rasterix | so i dont get gaps |
| 01:36.01 | rasterix | what does asterisk do if u have priority 1 followed by 3? |
| 01:36.13 | WimpMan | But when using n, you could easily end up without 1. |
| 01:36.21 | rasterix | ok true |
| 01:36.25 | WimpMan | Stop after 1. |
| 01:36.28 | rasterix | but gaps i cant get |
| 01:36.42 | WimpMan | correct |
| 01:36.56 | rasterix | use n and n(label) |
| 01:37.03 | *** join/#asterisk jsoftw (n=Administ@60.234.135.124) |
| 01:37.13 | WimpMan | But you can't always use n. |
| 01:37.20 | rasterix | current priority + 100 is deprecated i think |
| 01:37.25 | WimpMan | You can split up a chain and merge it together again. |
| 01:37.34 | jsoftw | Does anyone know how SIP protocol works, More specifically, how a brute force attempt to find logins/passwords would work? |
| 01:37.50 | Qwell | jsoftw: why? |
| 01:38.04 | jsoftw | I am ponderingi if I can rate limit connections to 5060 to stop brute force attempts |
| 01:38.18 | jblack | Sure, you can do that with your firewall. |
| 01:38.50 | jblack | If you google for "limit ssh connections", you should find examples to limit things to X new connections per minute, which you can adapt for * |
| 01:39.51 | *** join/#asterisk Coringao (n=carlos@unaffiliated/coringao) |
| 01:40.04 | WimpMan | You might end up making a brute force scan into a DoS. |
| 01:40.23 | jsoftw | So after X failed login attempts, the udp connection is reset, or if its just kept open |
| 01:40.33 | jsoftw | ssh is tcp. |
| 01:40.39 | rasterix | wimpman: do you have a complex extensions.conf i can look at? im testing an editor |
| 01:40.45 | WimpMan | udp has no connections. |
| 01:40.50 | jsoftw | I know, thats the thing. |
| 01:41.13 | WimpMan | rasterix: Not exately complex I think. |
| 01:41.24 | rasterix | complex is the wrong word |
| 01:41.41 | rasterix | i want to see how it looks at peoples different styles of coding |
| 01:41.52 | WimpMan | At least I have no fancy stuff with split priorities or such things in it. |
| 01:42.01 | rasterix | there are a lot since asterisk syntax definition is quite "loose" |
| 01:42.34 | Qwell | it has a very strict syntax |
| 01:42.35 | rasterix | ive tried googling for sample extensions.conf |
| 01:42.42 | *** join/#asterisk Levonk (n=lk@adsl-76-237-15-226.dsl.lsan03.sbcglobal.net) |
| 01:42.50 | rasterix | but most i find are from like 2003-2005 |
| 01:42.52 | WimpMan | But things like the following do work very well: |
| 01:42.58 | rasterix | i want more recent ones |
| 01:42.59 | WimpMan | exten => _123[0-4],1,doThis |
| 01:43.04 | WimpMan | exten => _123[5-9],1,doThat |
| 01:43.06 | WimpMan | exten => _123X,2,GoOn |
| 01:43.40 | *** join/#asterisk damasceno (n=damascen@189.71.158.93) |
| 01:44.17 | rasterix | wimpman: that is a good example |
| 01:44.20 | rasterix | thanks |
| 01:44.44 | rasterix | screwed up my current error checking |
| 01:44.46 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 01:44.55 | rasterix | but i guess im still grateful |
| 01:44.56 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 01:44.59 | WimpMan | That's such a point where n priorities won't do. |
| 01:45.22 | Qwell | WimpMan: it should work fine |
| 01:45.56 | WimpMan | Qwell: In that exampl you could use 1,1,n but not 1,n,n. |
| 01:46.11 | Qwell | eh? |
| 01:46.35 | WimpMan | n is always the last +1, but you need two identical ones there. |
| 01:46.43 | LemensTS | hey qwell on polycom phones how do you park someone? i know there is a hold key but what is the best way to park them for someone else to pick themup? |
| 01:47.07 | damasceno | Algum Brasileiro por favor? :/ |
| 01:48.05 | *** part/#asterisk Coringao (n=carlos@unaffiliated/coringao) |
| 01:48.20 | rasterix | < sinks into depression as he realises his current duplicate priority checking is a load of bs >< |
| 01:49.42 | rasterix | wheres drmessano tonight i miss beeing insulted :) |
| 01:49.50 | rasterix | wake up cutie pie :P |
| 01:50.22 | *** join/#asterisk UD (n=suck@unaffiliated/underdawg) |
| 01:50.34 | UD | hi |
| 01:50.47 | UD | Does anyone know if this # drops your ANI? http://www.google.com/goog411/ |
| 01:54.41 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 01:55.18 | *** join/#asterisk l8router (n=me@d122-109-92-165.sbr12.nsw.optusnet.com.au) |
| 01:57.46 | rasterix | wimpman: any other errors / mistakes that in extensions.conf that you make? |
| 01:58.12 | rasterix | lol wow i cant speak |
| 01:59.08 | rasterix | i figure i can check for goto that dont exist fairly easily |
| 01:59.10 | *** join/#asterisk joobie (n=joobie@201.023.dsl.mel.iprimus.net.au) |
| 01:59.13 | WimpMan | Forget the _ in front of a wildcarded extension. |
| 02:00.08 | rasterix | hmmm... that could be hard to check for |
| 02:00.25 | WimpMan | You could also check fo duplicate extensions. But I gess it'll be a tough one to figure out if it's wanted or not. |
| 02:00.31 | rasterix | since an extension can include almost anything in the name |
| 02:00.57 | WimpMan | Nah, if you find XNZ or [] chances are they are wildcards and not literals. |
| 02:01.15 | rasterix | i guess u cant have a duplicate extension within a context... |
| 02:01.30 | WimpMan | Yes, you might have to ask the user and save the anser. |
| 02:01.51 | drmessano | kisses rasterix on the lips |
| 02:02.02 | rasterix | *hugs drmessano |
| 02:02.14 | rasterix | strangely i missed your abuse |
| 02:02.15 | WimpMan | No, * will warn you, but with inclued you can and that one gets tricky with wildcard entries. |
| 02:02.37 | WimpMan | includes |
| 02:02.48 | rasterix | i think your right wimpman |
| 02:02.55 | rasterix | the best i can do is offer warnings |
| 02:02.59 | rasterix | not report errors |
| 02:03.17 | WimpMan | In fact it's rather tricky already to check if several wildcards overlap. |
| 02:04.26 | UD | what causes an attempted phone call on ordinary cell phone to start ringing but after first ring go to a "beep beep beep" and hang up? |
| 02:04.55 | UD | not a busy signal but a quicker noise |
| 02:05.00 | rasterix | is it possible to get asterisk to parse an conf file and report errors but NOT load it |
| 02:05.16 | UD | speed as in pause between beeps |
| 02:05.19 | UD | 3 beeps |
| 02:06.45 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 02:07.24 | jaytee | fast busy usually is an indicator that a trunk or trunks are unavailable |
| 02:09.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:09.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 02:09.41 | lmadsen | evening |
| 02:09.56 | jaytee | hi |
| 02:13.35 | lmadsen | quiet nite |
| 02:13.42 | lmadsen | everyone must have went to bed early after a long weekend |
| 02:13.49 | rasterix | what about this... within a context i should check for an extension pattern match overlap and report a warning |
| 02:14.08 | rasterix | nope... forget that hat could happen all the time |
| 02:14.22 | *** join/#asterisk stencil (n=stencil@206-248-132-55.dsl.teksavvy.com) |
| 02:14.46 | WimpMan | It probably will, unfortunaletly there's now way to find out if it was intentionally. |
| 02:14.54 | WimpMan | -w |
| 02:15.13 | rasterix | yah coz u might have a default extension at the end |
| 02:15.18 | rasterix | that picks up everything |
| 02:15.24 | rasterix | (for example) |
| 02:15.55 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:16.28 | rasterix | what i can do is check for duplicate extension+priority |
| 02:16.47 | rasterix | regardless of whether it is a pattern match or not |
| 02:16.56 | rasterix | i dont think that can ever happen |
| 02:17.28 | lmadsen | no, you can only have a single instance of each extension or pattern match in a context |
| 02:17.49 | WimpMan | Actually there is a very strong point in ending all extensions with a Hangup(16). |
| 02:17.52 | lmadsen | s/no/correct/ |
| 02:18.32 | WimpMan | Otherwise you can have strange effects with specific matches with fewer priorities than a wildcarded one that also matches. |
| 02:18.54 | rasterix | can i check for any extension that isnt finished with a hangup() and issue a warning |
| 02:19.18 | WimpMan | Yes, but I'd make that an option. |
| 02:19.28 | rasterix | im thinking |
| 02:19.34 | WimpMan | I don;t know how many people care to do so. |
| 02:19.49 | rasterix | i will have "warnings" and "errrors" |
| 02:20.09 | rasterix | if they dont care to follow good practise.. the can look at my warnings :) |
| 02:21.13 | rasterix | what i really need... really need is somewhere there are X billion extensions.conf pasted |
| 02:21.16 | rasterix | that i can load up |
| 02:21.23 | rasterix | and find the errors in my editor |
| 02:21.38 | rasterix | (of which at present there are XXXX billion) |
| 02:23.29 | rasterix | < will marry the first person that tells him where to go for sample extension.confs |
| 02:23.37 | WimpMan | I think the usual stuff would already be helpful, like matching brackets, checking variable names, user names and filenames. |
| 02:23.46 | rasterix | < will also offer his sister up for marriage |
| 02:24.02 | WimpMan | Teel more about your sister then :-) |
| 02:24.06 | rasterix | matching brackets = done |
| 02:24.23 | rasterix | variable names are checked for validity |
| 02:24.28 | rasterix | i used |
| 02:24.49 | WimpMan | Existance would be better. |
| 02:24.57 | rasterix | must start in alpha then any number of alphanumeric _ or - |
| 02:25.05 | rasterix | not the asterisk rule |
| 02:25.12 | rasterix | but i think its reasonable |
| 02:25.43 | rasterix | not sure i can do that wimpman |
| 02:26.16 | rasterix | would mean id have to follow every possible logical path through the plan |
| 02:26.18 | WimpMan | The name either has to be built in to * or the has to be a Set() somewhere. |
| 02:26.22 | rasterix | to see if variable set |
| 02:26.27 | rasterix | oh ok |
| 02:26.31 | rasterix | thats simple |
| 02:26.35 | rasterix | good idea |
| 02:26.37 | WimpMan | Off course you'd be screwed with AGIs :-) |
| 02:26.42 | *** join/#asterisk nix8n82 (n=nate@63.162.28.92) |
| 02:26.52 | rasterix | lol im back to a warning then |
| 02:27.14 | rasterix | i think i will end up reporting ZERO errors |
| 02:27.22 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 02:27.22 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:27.24 | rasterix | and a BILLION warnings |
| 02:27.25 | WimpMan | Yes, it's not exactely checking if a variable exists at the point it's used but it should catch typos. |
| 02:27.58 | rasterix | yep i like it |
| 02:28.00 | rasterix | SMART |
| 02:28.20 | WimpMan | You could also do it exactely if you like... |
| 02:28.22 | rasterix | added to my to do list |
| 02:28.38 | rasterix | i have a to do list |
| 02:28.55 | rasterix | that spans over 10 pages |
| 02:28.57 | rasterix | :( |
| 02:29.45 | rasterix | wimpman... keep going |
| 02:29.51 | rasterix | ur doing good XD |
| 02:32.11 | rasterix | this whole thing is turning into such a battle |
| 02:32.28 | rasterix | just getting intellisense working reasonably was a nightmare |
| 02:33.23 | WimpMan | Actually I still think the whole thing is not worth it. |
| 02:33.32 | rasterix | u might be right |
| 02:33.34 | rasterix | sigh |
| 02:33.40 | rasterix | but ive started now |
| 02:33.44 | rasterix | so ill finish |
| 02:33.49 | heedly | quit while you are ahead! |
| 02:33.54 | WimpMan | Especially if it doesn't run on the same platform. |
| 02:33.56 | rasterix | lol perhaps i should |
| 02:33.58 | rasterix | well |
| 02:34.06 | heedly | programs are never finished |
| 02:34.13 | rasterix | that is a BIG problem wimpman |
| 02:34.30 | rasterix | im ignoring that lol |
| 02:35.00 | rasterix | and even if i could make it run on linux gui |
| 02:35.06 | rasterix | that would be pointless |
| 02:35.20 | rasterix | since u dont install asterisk on linux with gui |
| 02:35.46 | rasterix | haha u guys r depressing me |
| 02:35.49 | WimpMan | Not a production system, I hope. |
| 02:36.03 | WimpMan | But you could always use a remote display. |
| 02:36.21 | rasterix | might as well be windows then |
| 02:36.32 | WimpMan | But then I don't know why an editor needs a gui. |
| 02:36.36 | rasterix | my argument is... ASTERISK Iis a box |
| 02:36.58 | rasterix | forget whether it runs on linxux windows or whatevah |
| 02:37.07 | rasterix | its a box that needs configuring |
| 02:37.10 | *** join/#asterisk colinm_ (n=colinmat@VDSL-130-13-101-237.PHNX.QWEST.NET) |
| 02:37.26 | *** part/#asterisk colinm_ (n=colinmat@VDSL-130-13-101-237.PHNX.QWEST.NET) |
| 02:37.57 | rasterix | i wonder how many asterisk configs dont have a windows machine on the same network |
| 02:38.11 | WimpMan | Unfortunaletly there are unlimited ways to do that. |
| 02:38.23 | rasterix | i agree but |
| 02:38.31 | rasterix | the gui's are limited |
| 02:38.37 | rasterix | from what ive read and heard |
| 02:38.39 | WimpMan | 3 of 4 I'm into. |
| 02:38.49 | WimpMan | They surely are. |
| 02:39.07 | rasterix | im looking at providing a way of editing config files that removes NO flexibility |
| 02:39.15 | rasterix | and still catches errors |
| 02:39.20 | WimpMan | But you want to check the configs that have might been done by clever people, don't you? |
| 02:39.31 | rasterix | YES |
| 02:39.39 | WimpMan | Let me phrase that in another way... |
| 02:39.41 | rasterix | i need complex conf files |
| 02:39.51 | rasterix | that crash my editor |
| 02:39.58 | rasterix | highlight all its flaws |
| 02:40.20 | WimpMan | Gongratulations to the receipt of your golden crap card. This card has life long validity and is not transferable. |
| 02:41.21 | rasterix | holy shit |
| 02:41.26 | rasterix | its nearly 4 am |
| 02:41.30 | rasterix | im gonna crash |
| 02:41.34 | rasterix | thanks wimpman |
| 02:42.03 | rasterix | uve been both helpful |
| 02:42.08 | rasterix | and infinitely depressing |
| 02:42.25 | WimpMan | oh, for the configs... Didn;t you want to tell more about your sister? *eg* |
| 02:42.29 | *** join/#asterisk knarfly (n=vladt79@c-75-74-155-198.hsd1.fl.comcast.net) |
| 02:42.41 | rasterix | lol shes 21 and stupid |
| 02:42.46 | rasterix | interested? |
| 02:42.54 | WimpMan | Nah |
| 02:42.58 | *** join/#asterisk waverly360 (n=waverly@c-98-211-5-255.hsd1.tn.comcast.net) |
| 02:43.13 | rasterix | shes probably not actually stupid |
| 02:43.13 | WimpMan | Guess I have to keep siting on my configs then. |
| 02:43.44 | rasterix | haha dont make me send u pictures of her on the beach :P |
| 02:44.54 | rasterix | wimpman: if u dont wanna send me the files.. i could send u an install for my editor next week |
| 02:45.01 | rasterix | u can break it yourself |
| 02:45.08 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088810630.dsl.bell.ca) |
| 02:45.20 | *** join/#asterisk hi365_m (n=hi365@213.151.50.174) |
| 02:45.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:45.40 | WimpMan | I most certainly could. I'm really good at breaking software. But I've never been a Windows user. |
| 02:45.55 | *** join/#asterisk demonist (n=demonist@jstickland.ca) |
| 02:45.59 | *** part/#asterisk demonist (n=demonist@jstickland.ca) |
| 02:46.42 | rasterix | maybe im missing a point... but is there really much difference between publishing config files to an asterisk server and publishing html to apache? |
| 02:47.28 | rasterix | anyway must sleep |
| 02:47.49 | jblack | well, some config files have usernames, passwords, give exposure to corporate infrastructure |
| 02:48.04 | WimpMan | rasterix: Theres a config waiting for you. |
| 02:48.28 | *** join/#asterisk kiefer08 (n=Kiefer@unaffiliated/kiefer08) |
| 02:48.35 | *** part/#asterisk kiefer08 (n=Kiefer@unaffiliated/kiefer08) |
| 02:48.39 | jblack | nobody ever uses the channel conference. :( |
| 02:49.09 | LemensTS | how many pair does a poe phone use in cat5 |
| 02:49.16 | rasterix | there is often confidential information contained within the files published to a web server too... |
| 02:49.44 | jblack | were you asking a question, or looking for a fight? |
| 02:50.03 | rasterix | asking a question |
| 02:50.35 | *** join/#asterisk hi365_m (n=hi365@213.151.50.174) |
| 02:50.37 | rasterix | and what u said is right... but i think it applies to publishing to a web server too |
| 02:51.24 | jblack | I don't think password files are typically served with apache. And they also don't open up risks to 2 cents a minute if exposed. |
| 02:51.35 | *** part/#asterisk knarfly (n=vladt79@c-75-74-155-198.hsd1.fl.comcast.net) |
| 02:51.49 | rasterix | perhaps ur right |
| 02:52.01 | rasterix | i need a secure connection to the asterisk server |
| 02:52.08 | rasterix | to transfer conf files |
| 02:52.47 | jblack | I tried bioshock for the first time tonight. |
| 02:52.50 | rasterix | ill come to that later... since ill never finish the editor no point in worrying to much right now |
| 02:53.35 | rasterix | wimpman: where is my config file! |
| 02:53.47 | WimpMan | Still waiting. |
| 02:54.23 | rasterix | right |
| 02:54.31 | rasterix | definitely time to sleep |
| 02:54.32 | rasterix | its 3 |
| 02:54.34 | rasterix | 4* |
| 02:54.37 | rasterix | bye all |
| 02:54.59 | WimpMan | been there an hour ago :-) |
| 02:55.00 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 02:57.37 | LemensTS | poe can be on the data pairs or the unused pairs, or both via 802.3af |
| 03:08.37 | mchou | anyone have an * setup than can successfully make use of sipbroker peering? |
| 03:08.59 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 03:09.58 | *** join/#asterisk L|L_Pr|nC (i=lilprinc@77.28.183.237) |
| 03:22.16 | Knightfa| | mchou: http://faq.sipbroker.com/tiki-index.php?page=Asterisk+Configuration |
| 03:23.09 | mchou | Knightfa|:been there, done that. Requires Proxy Auth for whatever reason. Dunno where that is coming from |
| 03:23.45 | mchou | Knightfa|: and yes, I've tried 'insecure=invite' |
| 03:23.51 | Knightfa| | lol |
| 03:23.53 | Knightfa| | ok ok |
| 03:25.47 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-23-67.revip2.asianet.co.th) |
| 03:28.32 | nhuisman_work | lol talk about hack, I was installing asterisk BE on rhel 5 and there was a .h file that had a hosed setting, so I started the install, paused it with ctrl + z, went into tmp, edited the .h and restarted the build :P |
| 03:28.35 | nhuisman_work | worked like a charm |
| 03:28.41 | nhuisman_work | now I need to tell them to fix it |
| 03:30.43 | *** join/#asterisk jayrod422 (n=jayrod42@pool-96-236-149-175.pitbpa.fios.verizon.net) |
| 03:30.44 | jayrod422 | anyone know of a provider who supports ani2? |
| 03:39.19 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.7) |
| 03:44.12 | *** join/#asterisk newbie___ (n=Some_guy@60.49.23.190) |
| 03:45.10 | jsoftw | Hmm. |
| 03:45.22 | jsoftw | Whats the best way to detect a brute force attempt at SIP logins? |
| 03:47.00 | newbie___ | hi all, in asterisk, how do u define greater or equal to zero and less than or equal than 2 ? |
| 03:47.02 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9aee3d4ed3de5a3f) |
| 03:47.07 | newbie___ | i know is stupid |
| 03:50.24 | [TK]D-Fender | newbie___: define WHERE? |
| 03:51.18 | newbie___ | exten => s,3, GotoIf($[${GROUP_COUNT(OUTBOUND_GROUP)} > 8]?103) |
| 03:51.32 | newbie___ | instead of >8 , i want somethign else |
| 03:51.46 | newbie___ | >= 0 and <= 2 |
| 03:52.21 | baliktad | just use 2 lines to do it |
| 03:52.48 | baliktad | if < 0, then skip to a named priority 2 lines ahead |
| 03:53.01 | *** join/#asterisk ta^3 (n=tacvbo@189.146.183.121) |
| 03:53.02 | [TK]D-Fender | newbie___: Go read the WIKI page on "Asterisk Expressions" |
| 03:53.09 | baliktad | then in the next line, if <= 2 jump to whatever you want |
| 03:53.21 | newbie___ | baliktad: thanks |
| 03:54.08 | [TK]D-Fender | newbie___: it'd be a good learning experience for you to learn how to do it the way you asked. |
| 03:54.20 | [TK]D-Fender | newbie___: I encourage you to go read that page |
| 03:57.01 | newbie___ | [TK]D-Fender: yup, i am reading it. :) thanks |
| 03:57.08 | jsoftw | Whats the best way to detect SIP brute force login attempts ? |
| 03:57.36 | [TK]D-Fender | jsoftw: scan the log constantly and blacklist IP's |
| 03:58.00 | nhuisman_work | [TK]D-Fender, heh |
| 03:58.28 | jsoftw | [TK]D-Fender: there is no way to do it via firewalls because of the way UDP/SIP works, yes? |
| 03:58.45 | jsoftw | Some testing here shows pf's src-con-rate-limit does not work. |
| 04:00.58 | [TK]D-Fender | jsoftw: When I jsut told you "blacklist by IP" what part of "firewall" did that not trigger for you? |
| 04:01.23 | nhuisman_work | I think he might not want to manually check the logs |
| 04:01.54 | [TK]D-Fender | nhuisman_work: Did I say manually? |
| 04:02.11 | jsoftw | [TK]D-Fender: im talking about pf's rate limiting. |
| 04:02.12 | [TK]D-Fender | Where do people come up with the idea of trying to put words in my mouth? |
| 04:02.35 | [TK]D-Fender | jsoftw: You asked for a way, I gave you a way. If you know of more, then more power to you. |
| 04:03.03 | jsoftw | :| |
| 04:05.40 | jsoftw | [TK]D-Fender: Hey. Get the fuck off your high horse eh? |
| 04:06.14 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 04:07.05 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 04:09.04 | [TK]D-Fender | jsoftw:High horse? No, just trying to let people read what I say and not go writing between the lines. If you have a better way then my suggestion then I sincerely DO hope it works out to the full extent of your hopes |
| 04:10.27 | jsoftw | [TK]D-Fender: no no, your just being a bit of a cock. |
| 04:10.39 | [TK]D-Fender | jsoftw: I certainly never claimed to kow it all, or have attempted to shoot down your ideas. |
| 04:10.48 | jsoftw | Anyway.. |
| 04:12.46 | Knightfa| | Anyone currently using Penalties for agents and queues? I plan on testing how it works tomorrow but wanted some insight. |
| 04:13.11 | Alton2 | hdusdhusduyea uu rueeurueuer r rdjduuuuuuuf |
| 04:13.17 | Alton2 | gyfgfgf |
| 04:13.33 | Alton2 | sorry, baby on the loose |
| 04:13.34 | UD | ? |
| 04:13.41 | jsoftw | Alton2: I dissagree. I think in most cases, Frooeooeoeog adhu ffffunna mmmurrrrr. |
| 04:13.58 | Knightfa| | The example I see is for the ring all strat.. I was planning on using it with leastrecent. |
| 04:15.31 | Knightfa| | Anyone :( |
| 04:16.26 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 04:16.35 | pcrane | anyone used grnvoip before? |
| 04:19.22 | pcrane | I'm getting Got SIP response 500 "Server Internal Error" back from them |
| 04:21.42 | [TK]D-Fender | pcrane: Perhaps you should pastebin the SIP debug of what you're sending and what the are responding so we don't have you guess... |
| 04:21.59 | pcrane | eh |
| 04:22.00 | pcrane | heh* |
| 04:22.01 | pcrane | ok |
| 04:23.09 | pcrane | http://pastebin.com/d129def84 |
| 04:23.59 | pcrane | 154 is the line with the internal server error |
| 04:24.10 | pcrane | I'm probably just sending the wrong stuff to the server |
| 04:28.45 | [TK]D-Fender | pcrane: Ever get those errors just sitting idle? |
| 04:28.52 | pcrane | no |
| 04:28.56 | pcrane | only when I make a call |
| 04:29.49 | pcrane | actually |
| 04:29.52 | pcrane | I might be lying |
| 04:30.03 | pcrane | http://pastebin.com/d678af78b |
| 04:30.13 | pcrane | those have just appeared now |
| 04:30.18 | [TK]D-Fender | pcrane: try "insecure=port,invite" for your peer and retest |
| 04:30.24 | pcrane | ok |
| 04:30.52 | [TK]D-Fender | pcrane: the error message does not appear in your 2nd pastebin. |
| 04:30.57 | pcrane | yeah |
| 04:30.59 | pcrane | ok |
| 04:31.05 | pcrane | so, normal sip debug stuff then |
| 04:31.42 | *** join/#asterisk Levonk (n=lk@adsl-76-230-110-3.dsl.lsan03.sbcglobal.net) |
| 04:32.03 | pcrane | http://pastebin.com/d22a7fe40 |
| 04:32.07 | pcrane | there's the latest attempt |
| 04:32.17 | *** join/#asterisk ManxPower (n=manxpowe@4.sub-75-249-90.myvzw.com) |
| 04:35.04 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 04:35.50 | [TK]D-Fender | pcrane: Do calls work at all? |
| 04:35.57 | pcrane | nope |
| 04:36.01 | pcrane | welll |
| 04:36.13 | pcrane | I'm trying to setup this SIP trunk for one of our clients |
| 04:36.23 | jeev | Fender, pump up the jam |
| 04:36.49 | pcrane | I'm trying everything that makes sense to me... |
| 04:37.01 | pcrane | there are a number of trunks that the client has asked to be setu |
| 04:37.01 | [TK]D-Fender | pcrane: pastebin your [general] and [grnvoip] sections masking only passwords |
| 04:37.02 | pcrane | p |
| 04:37.07 | pcrane | ok |
| 04:37.27 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:37.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:38.11 | *** join/#asterisk deadchicken_ (n=deadchic@64.218.49.154) |
| 04:39.21 | pcrane | http://pastebin.com/d335147dd |
| 04:40.32 | pcrane | http://www.grnvoip.com/faq.html |
| 04:40.36 | pcrane | I've found that |
| 04:40.39 | pcrane | and followed it |
| 04:40.45 | pcrane | but it doesn't seem to help |
| 04:43.38 | [TK]D-Fender | pcrane: didn't follow exactly as they wrote |
| 04:43.58 | pcrane | this looks wrong to me |
| 04:43.58 | pcrane | use this: exten => _Z.,1,Dial(SIP/70.42.72.49/11332400${EXTEN}) |
| 04:44.09 | [TK]D-Fender | pcrane: In their sample you are dialing by IP in extensions.conf, in yours you are going through the peer entry |
| 04:44.39 | pcrane | shouldn't it be SIP/11332400${EXTEN}@70.42.72.49 |
| 04:44.40 | pcrane | ? |
| 04:44.42 | [TK]D-Fender | pcrane: Dial("IAX2/101-12495", "SIP/103444017868880990@grnvoip") |
| 04:44.54 | [TK]D-Fender | pcrane: See, not like their sample |
| 04:46.16 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 04:47.02 | [TK]D-Fender | pcrane: and your "standard routing prefix" doesn't match either. Is there are reason that doesn't lok like their sample either? |
| 04:47.13 | pcrane | I'm just testing a single number |
| 04:47.22 | pcrane | to see if I got the SIP details right |
| 04:47.35 | [TK]D-Fender | pcrane: the prefix doesn't look the same. What tells you that what you used is valid? |
| 04:47.42 | pcrane | the client |
| 04:48.07 | [TK]D-Fender | pcrane: Don't shove random number and wonder why things fail. You have not followed their instructions on 2 counts so far |
| 04:48.41 | [TK]D-Fender | pcrane: read it again and try to follow closely, then call them up if need be. |
| 04:48.59 | pcrane | mmm |
| 04:52.34 | *** join/#asterisk ManxPower (n=manxpowe@4.sub-75-249-90.myvzw.com) |
| 04:56.34 | UD | is there a phone # i can call that will tell me my caller id? |
| 04:58.46 | [TK]D-Fender | UD: probably telco test #'s. Could always just ask someone you know who has it and call them |
| 05:02.29 | pcrane | managed to get it working with a different trunk... |
| 05:02.35 | pcrane | stupid number formats... |
| 05:02.45 | pcrane | some expect a 1 in front of the number being dialed... |
| 05:02.47 | pcrane | others don't |
| 05:02.51 | pcrane | *sigh* |
| 05:02.53 | pcrane | anyway |
| 05:03.51 | [TK]D-Fender | pcrane: next time, think less, follow instructions more :) |
| 05:04.20 | pcrane | well |
| 05:04.42 | pcrane | it was only after messing around with a 1 in the number to dial that I thought about playing with a 1 infront of the number of the other trunks... |
| 05:11.16 | Knightfa| | any good ways to stress test my * setup |
| 05:12.25 | WimpMan | Create an account and post its details on some haxx0r website with the promise of free international calls? |
| 05:12.45 | Knightfa| | lol |
| 05:13.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:14.06 | [TK]D-Fender | Knightfa|: Start with water-boarding. Its legal you know! |
| 05:15.52 | [TK]D-Fender | ok, checkout time. Later all. |
| 05:33.22 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 05:34.16 | CrashSys | Does anyone know if GSM codec has issues with GCC 4.3? I know with GCC 4.2 the codec would have horrible audio/choppyness/etc... |
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| 06:40.41 | jblack | Oh geeze. The republicans muffed it again. |
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| 07:52.33 | qp | morning all |
| 07:59.16 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 07:59.18 | r0land | hi all |
| 07:59.23 | r0land | hi jblack |
| 07:59.46 | jblack | I'm having a horrible time with a machine with a PRI that every once in a while, weeks, all of the zap channels start coming down with congestions, error type 34. |
| 07:59.50 | jblack | r0land: Hi? |
| 07:59.51 | r0land | could anyone help out with something.. its something u might face/solved with experience |
| 08:00.28 | r0land | sometimes when im on a call, theres an interference i get connected to some1 else's conversation and we all hear eachother |
| 08:00.34 | r0land | its like the phone calls gets merged into one.. |
| 08:00.44 | r0land | any1 has an idea on wht might cause such a thing! |
| 08:07.06 | *** join/#asterisk newbie___ (n=Some_guy@60.49.23.190) |
| 08:09.09 | newbie___ | i have given up trying to do it myself and now looking for a pay service. any taker? |
| 08:10.19 | *** join/#asterisk steliosk (n=Stelios@79.107.67.6) |
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| 08:21.15 | thomas | Good morning! |
| 08:22.45 | thomas | If i call to a other user (isdn example), is it posible for specified user the call number hide the number? |
| 08:23.12 | thomas | like this: SetCallerPres(prohib_not_screened) |
| 08:23.22 | thomas | but how I can add this in the dialplan? |
| 08:25.02 | rasterix | jblack: what do you have to do get the machine working again? |
| 08:26.13 | thomas | is it posible for several users, if the user outgoing set the SetCallerPres(prohib_not_screened) ? |
| 08:26.23 | thomas | not for each, only for.. 1..2..3 user? |
| 08:33.24 | *** join/#asterisk fcois93 (n=fcois93@bagnolet.acropolistelecom.net) |
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| 08:38.24 | thomas | is it posible with NoOp this value from the sipuser in my asterisk: callgroup ? |
| 08:41.48 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 08:46.13 | qp | is there a good resource/page for getting the best call quality out of a voip only solution? |
| 08:46.22 | qp | im scanning voip-info |
| 08:56.02 | *** join/#asterisk Levonk (n=lk@adsl-76-230-110-3.dsl.lsan03.sbcglobal.net) |
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| 09:02.54 | ObsidianX | hey folks, for realtime dynamic configuration would i set the => variables the same way in the database? |
| 09:03.08 | ObsidianX | ie: agent => xxx,yyy,zzz |
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| 09:07.06 | *** part/#asterisk zydoon (n=zydoon@41.225.156.101) |
| 09:08.38 | zamba | i'm trying to execute Read() to read digits from the user.. i'm using Read(NUMBER,,4) to read a maximum of 4 digits.. but when i try entering the four digits, i see in console: "User entered nothing" |
| 09:08.43 | zamba | <PROTECTED> |
| 09:08.43 | zamba | <PROTECTED> |
| 09:08.53 | zamba | and then it just goes back in the loop |
| 09:17.03 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:28.26 | thomas | the database from asterisk, is this automatic save, also if i stop asterisk? |
| 09:28.42 | thomas | have version 1.2X |
| 09:30.36 | mchou | you mean astdb? |
| 09:30.38 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 09:30.48 | thomas | mchou: jep |
| 09:30.51 | *** part/#asterisk bamsefar (i=emj@core.serv.emj.se) |
| 09:30.53 | mchou | astdb is persistent |
| 09:31.22 | thomas | persistent ? |
| 09:31.30 | thomas | ah ok |
| 09:35.32 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 09:37.53 | thomas | mchou: is it wrong? exten => s,n,DBGet(RUFNUMMER_UNTERDRUECKEN=rufnummer_unterdruecken_fuer/${CALLERID(num)}) |
| 09:37.56 | thomas | ? |
| 09:41.12 | thomas | [Sep 3 11:40:36] WARNING[2809]: pbx.c:1832 pbx_extension_helper: No application 'DBGet' for extension (macro-ausgehend-isdn, s, 2) |
| 09:41.55 | thomas | try exten => s,n,DBGet(rufnummer_unterdruecken=rufnummer_unterdruecken_fuer/8) |
| 09:41.56 | thomas | .-( |
| 09:42.24 | thomas | ideas? |
| 09:42.59 | kaldemar | you don't have application DBGet. use function DB instead. |
| 09:43.30 | thomas | kaldemar: ups. and in asterisk 1.2 is DBget and 1.4 DB ? |
| 09:44.41 | kaldemar | applications DBGet and DBPut were already deprecated in 1.2, and they were removed in 1.4. |
| 09:44.47 | thomas | ok |
| 09:44.59 | kaldemar | they were replaced by the function. |
| 09:47.24 | kaldemar | your get would be Set(RUFNUMMER_UNTERDRUECKEN=${DB(rufnummer_unterdruecken_fuer/${CALLERID(num)})}) |
| 09:47.37 | thomas | kaldemar: and in the 1.2er version? |
| 09:47.47 | thomas | Set(RUFNUMMER_UNTERDRUECKEN=${DBget(rufnummer_unterdruecken_fuer/${CALLERID(num)})}) |
| 09:47.50 | thomas | ? |
| 09:47.56 | kaldemar | both lines will work in 1.2 |
| 09:48.04 | thomas | kaldemar: oh, yes? |
| 09:48.07 | thomas | muy bien :-) |
| 09:48.39 | kaldemar | your last line was wrong, though. |
| 09:48.47 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 09:48.50 | thomas | ok |
| 09:48.51 | kaldemar | you can't nest applications like that. |
| 09:49.18 | kaldemar | you can only put functions inside applications. |
| 09:49.24 | r0land | hello all |
| 09:49.29 | r0land | could anyone help out with something.. its something u might face/solved with experience |
| 09:49.37 | r0land | sometimes when im on a call, theres an interference i get connected to some1 else's conversation and we all hear eachother |
| 09:49.45 | r0land | any1 has an idea on wht might cause such a thing! |
| 09:50.05 | r0land | its like ppl got invited to a conference even though no one sent an invitation |
| 09:52.58 | thomas | kaldemar: is it also posible a entry in the db to change? or only delete and newadd? |
| 09:57.07 | Odd_Bloke | Hello all. I'm trying to find out how to store SIP registrations into MySQL. Is it possible and, if so, how should I go about finding out more about it? |
| 09:57.29 | kaldemar | thomas: yes, it is possible to change a value. |
| 09:57.51 | thomas | kaldemar: can you say how? with put? if exists then "replace" ? |
| 09:58.50 | kaldemar | Set(DB(family/key)=value) |
| 09:59.06 | mchou | no, a put also does replace |
| 09:59.21 | thomas | kaldemar: no, i mean on the asterisk console, sorry. |
| 09:59.36 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 09:59.48 | kaldemar | thomas: database put |
| 10:00.29 | thomas | kaldemar: put replace a exists entry? |
| 10:00.46 | thomas | Updated database successfully |
| 10:00.47 | thomas | ah ok |
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| 11:24.33 | *** join/#asterisk angryuser (n=angryuse@121.215.20.81.dynamic.adsl.abo.nordnet.fr) |
| 11:29.58 | angryuser | good day |
| 11:36.25 | angryuser | i am searching for a simple script which restart asterisk in case of crash, i can write one and launch it with cron every n minutes, but something better would be nice, i am not a bash pro ;) |
| 11:36.41 | brodiem | angryuser: what OS? |
| 11:36.49 | angryuser | debian |
| 11:37.26 | tzafrir_laptop | angryuser, it will not be simple, if you want to allow actually stopping for various reasons |
| 11:38.09 | tzafrir_laptop | if you don't want to write a cron yourself but like that approach, use e.g. monit |
| 11:38.12 | brodiem | angryuser: there is already one included with the source that launches using safe_asterisk. /path/to/ast/src/contrib/init.d/ |
| 11:39.12 | angryuser | tzafrir_laptop: well i can write something like 'ps axjf | grep asterisk then cut the chain, then compate with one var and restart if test no passed |
| 11:39.19 | brodiem | if the process dies, safe_asterisk will restart it |
| 11:39.22 | angryuser | compare* |
| 11:39.26 | r0land | hello tzafrir_laptop |
| 11:39.36 | r0land | i have a prob tht im facing at the moment.. |
| 11:39.41 | tzafrir_laptop | angryuser, you need to not restart if you want to keep it down, and such |
| 11:39.53 | angryuser | brodiem: ok i will look after it |
| 11:40.08 | tzafrir_laptop | Don't re-invent the wheel, as this specific wheel has many corner cases |
| 11:40.31 | r0land | sometimes ill b having a conversation through sipphone->asterisk->sipphone, and suddenly i start hearing others get into my conversation! its like a conference though no one invited them.. |
| 11:40.42 | angryuser | tzafrir_laptop: can you explain the 'keep it down case' i dont really understand |
| 11:41.00 | tzafrir_laptop | you noticed a problem and want to keep asterisk down |
| 11:41.18 | tzafrir_laptop | e.g: to keep it from panicking the system |
| 11:41.27 | angryuser | tzafrir_laptop: ah, ok |
| 11:41.33 | yang | tzafrir_laptop: which phones are good choice to buy by your opinion? |
| 11:41.41 | brodiem | angryuser: safe_asterisk does everything you want it to do and there is already a debian init script included with the ast source. |
| 11:42.05 | tzafrir_laptop | yang, If you'd ask me I'll tell you that analog ones :-P |
| 11:42.17 | yang | hah, then you ned a good sipura :) |
| 11:42.27 | yang | or a fxs bank |
| 11:42.29 | tzafrir_laptop | seriously: I don't know various SIP phones that well |
| 11:43.12 | yang | Becouse I need to suggest what to buy again, so we won't use grandstream anymore |
| 11:43.29 | tzafrir_laptop | that debian init script has problems of its own. Use the init script included in the debian package |
| 11:43.47 | angryuser | yang: i have a good exp with snom, also siemens dect |
| 11:44.07 | yang | angryuser: can you tell me which snom model, i heard snoms are good |
| 11:44.18 | angryuser | yang: 320 340 |
| 11:44.36 | angryuser | yang: 360 with firmware v6 |
| 11:46.02 | yang | yeah they are in the upper price range |
| 11:46.53 | angryuser | yang: siemens are cheap for dect, bus some people have problems with too many phones and 1 base (3-4) |
| 11:46.59 | angryuser | but* |
| 11:48.07 | mchou | angryuser: which siemens models are you referring to? |
| 11:48.17 | angryuser | not stable, some voice loss, losing base sometimes, but with 1-2 phones and 1 base it is fine |
| 11:48.45 | angryuser | i have 470ip (dont put the last firmware!! ) 675ip |
| 11:48.57 | angryuser | 650ip |
| 11:50.04 | angryuser | mchou: ^^ |
| 11:50.12 | *** join/#asterisk Drasha (n=Drasha@faster11.westcom.cz) |
| 11:50.13 | thomas | is it posible to verify if a variable is empty? |
| 11:50.22 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:51.07 | angryuser | thomas: try gotoif |
| 11:51.09 | thomas | is it allowed? exten => _0.,n,GotoIf($["${rufnummer_unterdruecken}">0]? |
| 11:51.37 | yang | well we use 475 ip and I think that when i defined rtp ports it works better now |
| 11:51.47 | tzanger | try "${foo}" > "0" |
| 11:52.09 | thomas | tzanger: like this? |
| 11:52.10 | thomas | exten => _0.,n,GotoIf($["${rufnummer_unterdruecken}" > "0"]?unterdruecken:setze-eigenen-absender) |
| 11:52.34 | r0land | hey all! in show sip peers 307/307 192.168.0.15 D 5061 Unmonitored what does unmonitored stands for! |
| 11:54.05 | angryuser | r0land: by default qualify=no in sip.conf try to set it to qualify=yes and you will see the difference for that peer |
| 11:54.35 | angryuser | r0land: it is for checking the latency |
| 11:56.06 | angryuser | and if latency is more than the value defined in sip.conf the 'peer' is considered unreachable |
| 11:56.57 | tzanger | thomas: looks good |
| 11:57.50 | Drasha | Hi devs... does anyone know how to get an array of addresses that Asterisk listens on programatically? |
| 12:00.40 | mvanbaak | Drasha: netstat -anp | grep asterisk |
| 12:01.54 | Drasha | mvanbaak: thanks, but i need to do it from within the code. Like when I need to know theese addresses when programming new asterisk module |
| 12:02.11 | r0land | angryuser thank you |
| 12:03.02 | mvanbaak | Drasha: I dont think we have that available |
| 12:05.50 | Drasha | mvanbaak: Oh, I see... do you have any idea if there is an option to turn off auth checking either for specific user or for user on same address as Asterisk is? |
| 12:10.18 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:19.13 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 12:24.04 | thomas | database show numberlist <<if i have mor as 100 entries, is it posible with a like "|less" ? |
| 12:24.17 | thomas | like scroll.. :-) ? |
| 12:25.29 | [TK]D-Fender | thomas: asterisk -rx "database show numberlist"|less |
| 12:25.43 | thomas | [TK]D-Fender: ah, its easy .-) |
| 12:25.51 | thomas | [TK]D-Fender: thank you very much |
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| 12:29.54 | angryuser | if i have 100 clients for example, incoming passing from T1's and outgoung to voip provider, outgoing provider after verifying my identity let me set the DID , of course i want to set every did to each outgoing call propertly, what is teh best way to manage them? mysql datbase? astdb ? or should i use callerid(num) and forget about database ? |
| 12:30.49 | angryuser | all the ways are good, but what is easyer to manage ? |
| 12:31.28 | [TK]D-Fender | angryuser: You already know your options, just make up your mind on your own. |
| 12:31.53 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:33.25 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:33.30 | angryuser | [TK]D-Fender: well i think 10-20 peers no need for db, and upper is astdb/mysql, |
| 12:34.01 | [TK]D-Fender | angryuser: So whatever you're going to do, just go do it. |
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| 12:36.33 | r0land | [TK]D-Fender hey i need ur advice bout something... |
| 12:36.47 | r0land | [TK]D-Fender wht might b the cause of channels interfeering with eachother! |
| 12:36.49 | [TK]D-Fender | r0land: #drphil <- |
| 12:37.04 | [TK]D-Fender | r0land: Gee do you want to be a little MORE vague? |
| 12:37.18 | r0land | [TK]D-Fender for example, i would b talking to some1 sipphone->asterisk->siphone.. |
| 12:37.20 | r0land | and then suddenly |
| 12:37.25 | r0land | i hear some1 elses conversation |
| 12:37.33 | r0land | its like were having a conference.. but without the invitation |
| 12:37.37 | r0land | it just merges the conversations |
| 12:37.43 | angryuser | [TK]D-Fender: just looking for a exp of others |
| 12:37.50 | r0land | for no apparent reason |
| 12:38.01 | [TK]D-Fender | r0land: get some real details and ask later |
| 12:38.12 | r0land | [TK]D-Fender k ill pastebin |
| 12:38.28 | [TK]D-Fender | angryuser: Everything depends on who'll maintain it, when, what kind of access they have, etc. |
| 12:39.18 | angryuser | r0land: have you checked that cable ? you asked about it at mailing lists, i thing it is you cable, can you isolate each client byt it's own cable ? |
| 12:39.41 | *** part/#asterisk zydoon (n=zydoon@41.225.156.101) |
| 12:40.19 | *** join/#asterisk kalel008 (n=tamato@209.203.41.26) |
| 12:40.21 | zamba | can someone give me an example of using Read() in asterisk? |
| 12:40.46 | zamba | or explain to me what's wrong with the following piece of code: |
| 12:41.04 | zamba | http://www.pastebin.ca/1192488 |
| 12:41.29 | zamba | the console says that the user didn't enter any digits |
| 12:41.50 | kalel008 | hi there quick question how do i set so when someone calls on the DID it goes straight to voicemail |
| 12:41.56 | [TK]D-Fender | zamba: Then maybe you should fix your DTMFMOD <- |
| 12:42.03 | [TK]D-Fender | DTMFMODE <- |
| 12:42.07 | angryuser | zamba: check your dtmf params |
| 12:42.16 | zamba | angryuser: err :) |
| 12:43.01 | [TK]D-Fender | kalits your dialplan, so have your extens check to see who's calling and have it do what you want if the user matches the number |
| 12:43.11 | [TK]D-Fender | kalel008: its your dialplan, so have your extens check to see who's calling and have it do what you want if the user matches the number |
| 12:43.47 | kalel008 | but all my numbers will go direct to vmail |
| 12:44.10 | kalel008 | should i put Answer() then goto vmailor something like that |
| 12:44.43 | angryuser | zamba: and how do you call that ext anyway? from analog card ? sip client ? others ? |
| 12:44.49 | [TK]D-Fender | kalel008: they go where YOU tell them to. CHECK who's calling first and DECIDE what to do. |
| 12:44.55 | zamba | i'm calling the zap interface on one asterisk that's then dialing another asterisk over sip |
| 12:45.07 | zamba | angryuser: so the problem could definitely be because of that |
| 12:45.16 | kalel008 | crap i think i didnt explain correctly |
| 12:45.39 | kalel008 | i meant when dialed any number it should go dirctly to voicemail |
| 12:45.54 | zamba | angryuser: basically having an extension in the "border" asterisk that's says Dial(SIP/<other_asterisk>/${EXTEN} |
| 12:45.57 | zamba | +) |
| 12:46.36 | [TK]D-Fender | zamba: And you didn't set your DTMF parameters to match between both sides. |
| 12:46.55 | [TK]D-Fender | kalel008: again, its your dialplan, GO ADD IT. |
| 12:46.57 | angryuser | zamba: i dont see a problem here, its dtmf probaly |
| 12:47.12 | angryuser | probably* |
| 12:47.12 | r0land | [TK]D-Fender http://pastebin.com/d6524d1cb <<-- i did my best to explain clearly.. |
| 12:47.14 | zamba | [TK]D-Fender: so i need to set dtmfmode to the same for both peer declarations in sip.conf? |
| 12:47.50 | *** join/#asterisk barakuda_work (n=barakuda@aliens4.betex.ru) |
| 12:48.18 | [TK]D-Fender | r0land: Keep it for someone else, I'm not dealing with it. |
| 12:49.06 | [TK]D-Fender | r0land: And you still have broken priorities and stupid overlapping extensions. FFS fix your mess |
| 12:49.11 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 12:49.48 | r0land | [TK]D-Fender im just asking for an advice on wht might b causing this.. if its somthing to do with the dial plan ill start fixing it.. |
| 12:50.11 | [TK]D-Fender | r0land: how often does this happen? |
| 12:50.14 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:50.15 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:50.20 | r0land | every now and then |
| 12:50.33 | r0land | but mainly every 1 to 2 days |
| 12:51.31 | r0land | u think it might b the dialplan?! i just want to benefit from ur experience to know wht to look for/wht to fix |
| 12:52.05 | r0land | if its the dialplan ill start with it! if not.. i simply wanna pinpoint the prob |
| 12:52.08 | angryuser | r0land: have you done what i asked you ? |
| 12:52.12 | r0land | angryuser yes |
| 12:52.16 | [TK]D-Fender | r0land: You also failed to describe the environment of the affected callers, the server or anything else of value. |
| 12:52.36 | r0land | angryuser added qualify=yes on all sip pers |
| 12:53.05 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 12:53.13 | angryuser | r0land: No about strange analog cable you are using! i think you have messed some ping |
| 12:53.19 | r0land | angryuser ah yes |
| 12:53.20 | angryuser | pins* |
| 12:53.24 | r0land | angryuser i checked them all |
| 12:54.41 | r0land | angryuser i thought of adding : directrtpsetup=yes <<-- maybe this would help |
| 12:55.01 | r0land | if it opens a channel directly to the destination, tht way any activity on asterisk wont affect tit |
| 12:55.09 | r0land | or at least in concept it wouldnt.. |
| 12:55.38 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:55.45 | *** join/#asterisk tobias (n=tobias@user-0c2hj2f.cable.mindspring.com) |
| 12:55.50 | angryuser | r0land: please relplace thet cable by separated one, and see if this problems still affecting you |
| 12:55.56 | angryuser | one's |
| 12:56.07 | r0land | angryuser k i will :) |
| 12:57.09 | r0land | just checking if it might b something i messed up in dialplans or contexts.. |
| 12:57.09 | angryuser | i dont see what possibly can connect you to other conversation in * but analog cable mess |
| 12:57.30 | r0land | angryuser but why doesit happen every once in a while |
| 12:57.31 | r0land | and not always |
| 12:57.42 | r0land | i tried to replicate the issue so i know exactly whts happening.. it never happened with me |
| 12:57.47 | r0land | but it happens every now and then with others.. |
| 12:58.34 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 12:58.47 | angryuser | r0land: maybe some option in sipura, i am not familiar with them, jsut in case try to set call-limit=1 to desired peers |
| 12:59.10 | r0land | angryuser hmm |
| 12:59.17 | r0land | angryuser tht might proove useful |
| 12:59.22 | r0land | i could set it on the PSTN lines |
| 12:59.27 | r0land | tht way only one caller could use them |
| 13:00.02 | r0land | though one question, if i used my softphone to do a conference, and then i called the pstn line tht i've set "call-limit=1" in.. |
| 13:00.05 | r0land | would it work ? |
| 13:00.29 | angryuser | r0land: ?? before calling analog use ChanIsAvail to be sure |
| 13:00.55 | r0land | angryuser k ill change my dialplan accordingly |
| 13:00.56 | r0land | thank you |
| 13:01.07 | angryuser | r0land: no this option is for sip/iax peers |
| 13:01.37 | r0land | u mean the call limit?! |
| 13:02.17 | angryuser | r0land: no it wont work call-limit sets the call limit, so if you have the second call, it will get the busy signal |
| 13:03.27 | r0land | ah ok |
| 13:03.42 | r0land | but the thing is |
| 13:03.45 | r0land | wht were doing |
| 13:03.51 | r0land | is beating around the bush so to speak |
| 13:03.55 | r0land | and not managing the prob itself |
| 13:03.59 | r0land | wht i mean is |
| 13:04.01 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:04.14 | r0land | call from 100 to 200, gets merged with call from 230 to 201 |
| 13:04.25 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:04.43 | r0land | the ChanIsAvail would only work if im calling them manualy |
| 13:04.47 | r0land | not if the system messed up |
| 13:04.51 | *** join/#asterisk existenz (n=Ameen@unaffiliated/existenz) |
| 13:05.06 | r0land | on the other hand.. call-limit=1 would work in theory |
| 13:05.42 | existenz | When I hangup a call from pstn to line1, it keeps rining until the timeout of the Dial is finished, how can I hangup the call once the pstn call is hanged up? |
| 13:06.42 | r0land | existenz i faced tht issue with sipura, i configured a "detect disconnection tone" on it |
| 13:06.51 | r0land | though u need to know the exact disconnect tone of ur pstn line |
| 13:07.12 | r0land | simply sharing experience, not necessarly a solution in ur situation |
| 13:07.48 | existenz | r0land, I see |
| 13:08.03 | existenz | r0land, there should be a message that says that the pstn is hanged up |
| 13:08.20 | angryuser | r0land: you are messing it all, try all things i told you and come back with results, and read about call-limit and also about of use ChanIsAvail in wikis |
| 13:08.40 | r0land | angryuser k i will |
| 13:08.42 | r0land | thank you :) |
| 13:09.08 | *** part/#asterisk r0land (n=roland@193.227.191.91) |
| 13:09.43 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 13:11.22 | angryuser | i have a client he says that the command 'zaptel_hardware' is not found, what could it be ? cat /proc/zaptel/* is ok |
| 13:11.48 | lmadsen | [09:10] <tzafrir_laptop> anybody here with an X100P and Asterisk SVN to test http://bugs.digium.com/view.php?id=12841#c92014 ? |
| 13:12.24 | tzafrir_laptop | angryuser, what version of zaptel? |
| 13:12.47 | tzafrir_laptop | and is /usr/sbin in the PATH? what about /usr/sbin/zaptel_hardware ? |
| 13:13.14 | angryuser | tzafrir_laptop: good guess |
| 13:13.30 | seanbright | he wasn't guessing... he KNEW |
| 13:15.48 | angryuser | seanbright: what is your point ? |
| 13:19.34 | seanbright | i have none |
| 13:22.09 | existenz | tzafrir_laptop, hey, do you have a dialplan for bezeq? |
| 13:22.22 | *** join/#asterisk datacrusher (n=datacrus@200.145.121.55) |
| 13:24.44 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-148-176.dsl.sfldmi.sbcglobal.net) |
| 13:31.28 | yang | hey lmadsen your link to book @ tfot is not working |
| 13:31.42 | *** join/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 13:32.10 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 13:32.50 | *** part/#asterisk korihor (n=korihor@190.78.32.60) |
| 13:33.42 | lmadsen | yang: aha -- my IP changed... nice... |
| 13:33.49 | lmadsen | that hadn't happened for over a year |
| 13:34.07 | *** join/#asterisk oilinki (n=oil@ppp-124-120-21-50.revip2.asianet.co.th) |
| 13:36.24 | mprebello | hi everybody, i'm receiving a new log that i never received. anyone know what means " chan_iax2.c: Packet arrived out of order (expecting 12, got 11) (frametype = 6, subclass = 12)" and "chan_iax2.c: Acking anyway" |
| 13:37.03 | lmadsen | yang: thanks -- IPs updated. Will take a bit for them to propagate |
| 13:37.25 | existenz | has anyone tried skype with asterisk |
| 13:37.37 | [TK]D-Fender | mprebello: Packets were delayed and received out of order |
| 13:38.05 | jaytee | existenz, many have tried, many have failed. Skype is closed source proprietary protocol |
| 13:38.13 | *** join/#asterisk FinboySlick (n=FinboySl@207.134.8.4) |
| 13:38.17 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 13:38.24 | existenz | jaytee, any good alternative to skype? |
| 13:38.44 | jaytee | for what purpose? free internet calls? |
| 13:38.51 | mprebello | Hi Fender, thanks to answer, but it is mean that there is a problem in my network device or in my digium device? |
| 13:39.17 | [TK]D-Fender | mprebello: Network conditions somewhere along the way |
| 13:39.58 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 13:40.11 | angryuser | existenz: the best is skip2pbx but the price it *too much* and the quality is far from desired |
| 13:40.47 | existenz | angler, erm, I wonder whether there are flatrate prices for israel |
| 13:40.51 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 13:41.06 | gaetronik | hi there |
| 13:41.20 | mprebello | i got, but i have other server in the same network coditions and there isn't this problem |
| 13:41.37 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088810630.dsl.bell.ca) |
| 13:45.27 | iulius | Is it possible to only allow sip extensions to register when the phone is on the local network? Sort of like the PERMIT/DENY in manager.conf |
| 13:45.51 | [TK]D-Fender | iulius: Yes, go read "config sip.conf" on the WIKI |
| 13:46.17 | *** join/#asterisk _ShrikE (n=_ShrikE@64.128.34.118) |
| 13:46.55 | iulius | Aha |
| 13:47.14 | jaytee | so priority + 100 is deprecated in future * releases? |
| 13:47.34 | Qwell | s/future/current and past/ |
| 13:47.52 | jaytee | ? it works in 1.4 |
| 13:48.09 | jblack | deprecated doesn't mean "doesn't work" |
| 13:48.22 | jaytee | just means don't count on it to work going forward |
| 13:48.39 | *** join/#asterisk InHisName (n=rich@c-71-225-221-149.hsd1.pa.comcast.net) |
| 13:48.40 | mvanbaak | heya all |
| 13:48.46 | mvanbaak | exten => 570630555,n,Set(CALLERID(name)=IF($["x${CALLERID(num)}" = "x"]?Anoniem:0${CALLERID(num)})) |
| 13:48.54 | mvanbaak | does that line look correct ? |
| 13:48.58 | InHisName | Tried to do a 'soft hangup' on wayward channel but still there. |
| 13:49.41 | kaldemar | mvanbaak: no |
| 13:49.46 | [TK]D-Fender | mvanbaak: No. You may want to actually reference your function... |
| 13:50.14 | [TK]D-Fender | mvanbaak: and the "x" is unnecessary |
| 13:50.28 | existenz | What may cause the WaitExten to wait for an extension of only one digit? |
| 13:50.41 | mvanbaak | hhmm |
| 13:50.59 | [TK]D-Fender | existenz: only having 1 possible match for the digit pressed or you took too long |
| 13:52.38 | mvanbaak | hhmm |
| 13:53.21 | mvanbaak | ahhah, now I see :) |
| 13:53.29 | mvanbaak | stupid braces madness |
| 13:55.04 | *** join/#asterisk KueWisco (n=Administ@208-69-109-135-rev.redanvil.net) |
| 13:55.09 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 13:55.20 | angryuser | existenz: you can set it also: Set(Timeout(digit)=n) |
| 13:55.43 | *** join/#asterisk mog (n=mog@nat/digium/x-606f9b9c813d32fa) |
| 13:55.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:56.45 | FinboySlick | Is there some sort of magic where, through bluetooth, one could turn a cell phone into an asterisk channel? Pick up on ring, dial, etc. |
| 13:56.56 | *** join/#asterisk Levonk (n=lk@adsl-75-62-137-245.dsl.lsan03.sbcglobal.net) |
| 13:56.58 | jblack | chan_mobile. It's in the 1.6 beta. |
| 13:58.28 | angryuser | FinboySlick: dont consider it into production *yet* |
| 13:58.48 | FinboySlick | It's a gadget to me, but it'd be pretty darn cool. |
| 14:00.38 | KueWisco | Does 1.6-Beta9 honor the periodic-announce-frequency setting in Queues.conf? When I try to use it, the frequency does not seem to work properly |
| 14:03.14 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-dfea88cbd4630ffc) |
| 14:03.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.29 | gaetronik | hi putnopvut |
| 14:03.37 | [TK]D-Fender | KueWisco: pastebin your configs and the CLI output and maybe we can see if you've done something wrong. |
| 14:03.38 | putnopvut | hi gaetronik. |
| 14:09.06 | KueWisco | Here is the queue config |
| 14:09.11 | KueWisco | [general] |
| 14:09.11 | KueWisco | persistentmembers = yes |
| 14:09.11 | KueWisco | keepstats = yes |
| 14:09.11 | KueWisco | autofill = yes |
| 14:09.12 | KueWisco | [callcenter] |
| 14:09.12 | KueWisco | musiconhold=default |
| 14:09.14 | KueWisco | strategy=rrmemory |
| 14:09.16 | KueWisco | timeout=25 |
| 14:09.18 | KueWisco | retry=5 |
| 14:09.20 | KueWisco | wrapuptime=10 |
| 14:09.22 | KueWisco | maxlen=0 |
| 14:09.24 | KueWisco | joinempty=no |
| 14:09.26 | KueWisco | leavewhenempty=yes |
| 14:09.28 | KueWisco | periodic-announce-frequency=10 |
| 14:09.30 | KueWisco | periodic-announce = queue-periodic-announce,your-call-is-important,please-wait |
| 14:09.37 | km- | ~pastebin |
| 14:09.37 | jbot | [~pastebin] 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 |
| 14:09.46 | KueWisco | sorry |
| 14:11.42 | *** join/#asterisk neuwald (n=neuwald@189.61.96.60) |
| 14:11.46 | neuwald | Hi folks |
| 14:12.59 | neuwald | What's wrong on that dialplan? http://pastebin.ca/1192543 |
| 14:13.40 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 14:13.57 | neuwald | In GotoIf, when I got other value than 0, it always send to the extension chat-2,s,1 |
| 14:14.29 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 14:16.06 | neuwald | here is the gotoif line: exten => s,n,GotoIf(${count-1} = ${CONFMAX}?chat-2,s,1) |
| 14:17.13 | jaytee | when using Goto("named_priority") does it have to be from the same context that the named priority is in or from any context? |
| 14:17.42 | dandre | Hello, |
| 14:17.47 | neuwald | jaytee could be any context. u can use goto(context,s,1), for example |
| 14:18.59 | neuwald | jaytee or, if u wanna go to the same context, use goto(s,30) . example |
| 14:19.11 | jaytee | neuwald, I realize that but the book has two examples, Goto(context, extension, priority) and Goto(named_priority) and what I was wondering was if I used the latter syntax in one context whether it would jump to the labeled priority in another context. |
| 14:19.17 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-60a15aa29f60e2de) |
| 14:19.32 | dandre | I am using the manager interface to originate calls. When I issue an Originate action, first the channel is rung waiting for a pickup. Is there any way to have this pickup done automatically? |
| 14:20.16 | jaytee | aw, shit! we lost one of our Amur tigers yesterday. Necropsy showed advanced cancer. |
| 14:20.37 | KueWisco | Alright, here is the queue.conf and console stuff http://pastebin.com/m2a87a25f |
| 14:21.58 | [TK]D-Fender | jaytee: the name IS a priority in as specific a destination as you make as though it were hard-numbered |
| 14:22.44 | [TK]D-Fender | dandre: Depends what you're calling. |
| 14:24.19 | qp | hey all, is there a good recent resource/page for getting the best call quality out of a voip only solution? |
| 14:24.50 | jaytee | qp, yes, www.polycom.com |
| 14:24.55 | [TK]D-Fender | KueWisco: <SIP/7830-0078c6e0> Playing 'queue-periodic-announce.ulaw' (language 'en') <--- |
| 14:25.18 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 14:25.25 | [TK]D-Fender | qp: Use decent phone with a decent provider with decent bandwidth/jitter, etc |
| 14:26.00 | [TK]D-Fender | neuwald: Nice try, not go read the WIKI page on "asterisk expressions" |
| 14:26.03 | [TK]D-Fender | now* |
| 14:28.22 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 14:29.20 | dandre | [TK]D-Fender: ok I have a sip phone at extension 43. I want to issue a command that connects this sipphone to an outbound number that is to be dialed. the behaviour I have after the originate manager command is: |
| 14:29.20 | dandre | 1. sip/43 rings |
| 14:29.20 | dandre | 2. I answer sip/43 |
| 14:29.20 | dandre | 3. the party numer is dialed as if I had dialed it from my phone. |
| 14:29.20 | dandre | I'd like to have steps 1 and 2 done automatically |
| 14:30.14 | blebleble | anyone have any idea what could be the potential problem for this situation, we have about 20 ip phones (all seem to work well) except for one user who is over seas, the phone sounds great and works fine for about 20 seconds and then just cuts him off, however he does have an ip phone to our legacy phone system (intertel) that works perfectly and has for years so i know its not an internet connection issue. Anyone have any ideas? Thanks |
| 14:30.21 | [TK]D-Fender | dandre: Then make sure your originated Channel tells SIP/43 to automatically pick up. |
| 14:30.35 | KueWisco | [TK]D-Fender: From the docs, the frequency dictates how often the announcements occur. I set it at 10 (for testing purposes). The announcements do not occur at 10 second intervals. It varies, and is usually around 30 seconds |
| 14:30.53 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:30.56 | dandre | [TK]D-Fender: I don't know how to do that |
| 14:31.14 | [TK]D-Fender | KueWisco: periodic announcements also weave around your agent ring-time. |
| 14:31.43 | [TK]D-Fender | dandre: With the glorious amount of information you've given me, neither do I. |
| 14:32.17 | [TK]D-Fender | KueWisco: Frankly if I had announcements breaking in every 10 seconds I'd hang up personally :) |
| 14:32.24 | KueWisco | so the actual announce frequency is the frequency setting and the ringtime combined? |
| 14:32.32 | KueWisco | i just set it down that low |
| 14:32.41 | KueWisco | because it did not seem to be working correctly |
| 14:32.51 | existenz | Does Playback() play wav by default? |
| 14:32.53 | dandre | could you give me an example of automatically pick up ? |
| 14:33.11 | [TK]D-Fender | existenz: No, it plays the native codec first, then some arbitrary order |
| 14:33.41 | [TK]D-Fender | dandre: on what device? <------ you think this doesn't MATTER? |
| 14:34.15 | dandre | it is a sip phone st2030 from thomson |
| 14:34.34 | [TK]D-Fender | dandre: Go read its admin guide or get Googleing |
| 14:35.21 | dandre | I know how to pick up all calls but not selectively from originate command |
| 14:35.35 | dandre | it could be a linksys too |
| 14:35.45 | [TK]D-Fender | dandre: the method doesn't change... |
| 14:35.58 | [TK]D-Fender | dandre: If you know how to signal an auto-answer, then go DO it. |
| 14:36.21 | existenz | What is wrong about this rule? I want to route every number that start with 014 to pstn -> exten => _014.,1,Dial(SIP/${EXTEN}@pstn) |
| 14:36.32 | dandre | I don't know, there is a configuration option in the phone to autoanswer the call |
| 14:36.41 | *** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 14:36.52 | [TK]D-Fender | dandre: Go get a clue |
| 14:37.09 | lmadsen | [TK]D-Fender: be nice |
| 14:37.13 | [TK]D-Fender | existenz: Who says there is anything wrong with it? |
| 14:37.25 | existenz | [TK]D-Fender, doesn't work for me |
| 14:37.39 | [TK]D-Fender | dandre: If you want to know how to make device XYZ do something, go read its manual |
| 14:37.50 | [TK]D-Fender | existenz: Perhaps you could show us your failed attempt to use it |
| 14:38.00 | [TK]D-Fender | existenz: PASTEBIN is your friend |
| 14:38.53 | [TK]D-Fender | dandre: http://www.voip-info.org/wiki/view/ST2030+feature+wishlist <-- whree you should have already ended up in your quest |
| 14:38.56 | [TK]D-Fender | where* |
| 14:38.57 | lmadsen | s/PASTEBIN/pastebin/ |
| 14:39.39 | angryuser | is there any 'random' application in asterisk, let's say i want a random number in range 1-10 |
| 14:39.58 | lmadsen | angryuser: see function RANDOM() I believe :) |
| 14:40.12 | existenz | [TK]D-Fender, Is it possible to write and read from files? |
| 14:40.29 | lmadsen | angryuser: core show function RAND |
| 14:40.29 | [TK]D-Fender | existenz: Several ways. |
| 14:40.43 | *** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net) |
| 14:40.47 | existenz | [TK]D-Fender, like? |
| 14:40.51 | angryuser | lmadsen: ;) going to drink a cofee |
| 14:41.06 | [TK]D-Fender | existenz: perhaps you could be more specific about your expectations. |
| 14:41.22 | lmadsen | angryuser: nice :) |
| 14:41.50 | existenz | [TK]D-Fender, I want to be able to block numbers according to those I enlist in a file, separated by a comma. |
| 14:42.22 | dandre | ok thanks [TK]D-Fender |
| 14:42.54 | [TK]D-Fender | existenz: you could try the add-on "backticks" application, or do an AGI for that |
| 14:43.31 | mog | there is also a readfile application |
| 14:43.49 | lmadsen | or you can use #exec to generate dialplan code |
| 14:44.41 | dandre | so there is nothing "standard" for answering an inbound call on a phone |
| 14:44.59 | lmadsen | dandre: that is a function of the phone to auto-answer the line |
| 14:45.01 | [TK]D-Fender | mog: Yeah I suppose you could "exec" a grep then "readfile" it |
| 14:45.12 | [TK]D-Fender | dandre: No. |
| 14:45.27 | mog | ya |
| 14:45.56 | [TK]D-Fender | existenz: See me comment to mog |
| 14:46.09 | [TK]D-Fender | existenz: Probably the easiest way |
| 14:46.09 | existenz | Thanks! |
| 14:46.27 | mog | advantage being that they are built in existenz but backticks would work too |
| 14:46.28 | [TK]D-Fender | waits for race conditions to bomb it :) |
| 14:46.43 | mog | NOO |
| 14:46.50 | existenz | Does anyone know a hangup detection for pstn? |
| 14:46.59 | existenz | is the problem with the ata or asterisk? |
| 14:47.03 | jeev | waits for someone to bomb Fender |
| 14:47.23 | [TK]D-Fender | existenz: ATA |
| 14:47.42 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 14:47.52 | [TK]D-Fender | waits for hell to freeze over |
| 14:47.56 | [TK]D-Fender | jeev: lets have a race! |
| 14:47.57 | mog | existenz, there is busydetect |
| 14:48.06 | [TK]D-Fender | mog: Not applicable. |
| 14:48.07 | lmadsen | if it is an analog line (which it is if its an ata), it's because there is no signaling on an analog line, so it is a best effort of the device |
| 14:48.08 | mog | oh ata |
| 14:48.13 | mog | is a little slow |
| 14:48.18 | lmadsen | mog: !!! |
| 14:48.36 | jeev | where are we going to race to? |
| 14:48.39 | jeev | viking land ? |
| 14:48.39 | [TK]D-Fender | mog : half man, half dog.... he's his own best friend! |
| 14:48.41 | mog | device not as good as a sleek digium tdm410b! ^_^ |
| 14:48.56 | mog | but both have problems, analog busy detect is hard |
| 14:48.59 | lmadsen | jeev: yes sir! |
| 14:49.20 | lmadsen | jeev: race ya to Vinland! |
| 14:49.22 | jeev | scratches self |
| 14:49.25 | jeev | where the hell is vinland |
| 14:49.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:49.34 | jeev | how abut we duel? |
| 14:49.38 | jeev | races to his gun and blasts lmadsen |
| 14:49.51 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 14:49.51 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 14:50.03 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:50.07 | guax | whats the implication of that? "Running DeadAGI on a live channel will cause problem" |
| 14:50.23 | guax | or, the complication that can bring me |
| 14:50.37 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:56.41 | *** join/#asterisk steliosk (n=Stelios@79.107.48.71) |
| 14:57.13 | *** join/#asterisk seanmh (i=seanmh@216.31.101.121) |
| 14:57.23 | *** part/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.mn.comcast.net) |
| 15:00.44 | The-Bat | hello |
| 15:00.48 | The-Bat | Is there a way to determine the next agent who'll b getting the call in the queue |
| 15:01.06 | [TK]D-Fender | The-Bat: not that I can think of |
| 15:01.34 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:01.38 | The-Bat | ok thanks |
| 15:04.48 | blebleble | any clue why a remote sip phone would get disconnected automatically from me when they call (takes exactly 20seconds) however if i call them it works fine (no cut off) |
| 15:05.41 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:08.11 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 15:08.13 | *** join/#asterisk devhen|Work (n=devhen@216.194.118.110) |
| 15:08.13 | teknoprep | hey all |
| 15:08.20 | teknoprep | whats a good click to dial app for windows ? |
| 15:08.35 | teknoprep | that integrates with firefox if possible |
| 15:08.54 | teknoprep | i used to have one but got rid of it becuase it didn't support firefox 3.0 |
| 15:08.59 | teknoprep | but i forget the name of it |
| 15:09.49 | [TK]D-Fender | teknoprep: If its out there I suspect you'd see it on their extensions browser |
| 15:10.06 | teknoprep | it wasn't a firefox extension |
| 15:10.07 | beek | https://addons.mozilla.org/en-US/firefox/addon/5688 |
| 15:10.21 | *** join/#asterisk AlexTO (n=alex@201.228.24.214) |
| 15:11.17 | teknoprep | http://www.snapanumber.com/ |
| 15:11.23 | teknoprep | thats what i was lookign for |
| 15:11.53 | tzafrir_laptop | doesn't that one connect to the manager interface directly? |
| 15:12.31 | tzafrir_laptop | Which means every user who has it installed can run whatever command practically on the server? |
| 15:12.51 | *** join/#asterisk masus (n=masus@78.162.5.45) |
| 15:13.49 | *** join/#asterisk Levonk (n=lk@adsl-76-237-14-158.dsl.lsan03.sbcglobal.net) |
| 15:15.01 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 15:15.08 | masus | hi all , have a problem cant register to my asterisk server from home after add one repeater to my adsl modem (Router) -> here is the debug http://rafb.net/p/fi4w1q32.html Thanks |
| 15:15.56 | masus | i get an registration error sometimes 408 |
| 15:16.11 | [TK]D-Fender | masus: SIP/2.0 401 Unauthorized <- clearly not set up right |
| 15:16.33 | *** join/#asterisk daniev (n=ganbarim@190.144.60.154) |
| 15:16.46 | masus | [TK]D-Fender: but it's setup because i can register from anywhere else |
| 15:17.25 | [TK]D-Fender | masus: <--- Transmitting (no NAT) to 192.168.2.5:47661 ---> <-- and it looks like you havent' set your system up to handle NAT properly either |
| 15:18.07 | *** join/#asterisk bbryant (n=Brett_Br@adsl-212-77-233.chs.bellsouth.net) |
| 15:18.58 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 15:24.03 | *** join/#asterisk ManxPower (n=manxpowe@187.sub-75-248-47.myvzw.com) |
| 15:26.12 | *** join/#asterisk vonkleist (n=gcontrer@189.155.176.253) |
| 15:29.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:31.47 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 15:32.12 | *** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net) |
| 15:32.16 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:32.40 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:33.54 | masus | [TK]D-Fender: 192.168.2.5 is my homelan ip adress |
| 15:34.02 | masus | is this right :/ |
| 15:34.29 | [TK]D-Fender | masus: Look at the IP the request came in on. Its public. Look at the address * is responding to. That one is PRIVATE. Clearly not right |
| 15:35.34 | masus | i dont understand what is wrong and where to fix it :( but i know when i go to my neighbour i can register from there :) |
| 15:35.49 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 15:38.40 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 15:39.07 | jameswf-home | some reason the ubuntu zaptel source is missing the menuselect stuff which really screws with zaptel |
| 15:39.27 | [TK]D-Fender | ~sipnat |
| 15:39.27 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:39.28 | [TK]D-Fender | ^^^^^^^^ |
| 15:39.45 | nny_1 | I have a tdm410P and caller id just quit working a couple of weeks ago. I have swapped the module but still no CID. I have also tested the line directly using a traditional phone and the CID info shows up. I am filling out a support ticket for digium, anything else that i can do? |
| 15:39.51 | masus | i use eyebeam and if i uncheck the register with domain and receive incoming calls then i can register to * but cant do calls |
| 15:40.02 | nny_1 | besides start using another product |
| 15:40.11 | tzafrir_laptop | hands jameswf-home dummy-select from http://bugs.digium.com/13132 |
| 15:40.28 | nny_1 | i swear I am at the point where i have a 75% failure or issue resolution with digium cards,... hope it's just my bad luck |
| 15:40.34 | tzafrir_laptop | Will require some adjustments to work with zaptel, I guess |
| 15:41.05 | tzafrir_laptop | jameswf-home, and anyway, what's the problem with building everything? |
| 15:41.43 | jameswf-home | not going to mess with it pulled real sourc |
| 15:41.45 | jameswf-home | e |
| 15:42.16 | tzafrir_laptop | that tarball is supposed to only build modules |
| 15:42.25 | tzafrir_laptop | not utilities |
| 15:42.39 | mog | nny_1, do you ever get cid or does it fail every time, and did anything else change? |
| 15:42.56 | jameswf-home | it doesnt build at all with a make but i can make modules and make applications |
| 15:43.12 | jameswf-home | cd .. |
| 15:43.16 | jameswf-home | doh |
| 15:43.35 | [TK]D-Fender | nny_1: PASTEBIN YOUR ZAPATA.CONF |
| 15:44.37 | nny_1 | mog hard to tell I'll have to check the cdr, as we have both ZAP incoming as well as an ITSP |
| 15:44.39 | nny_1 | [TK]D-Fender: roger |
| 15:45.18 | nny_1 | http://pastebin.com/m1be01252 |
| 15:45.55 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:46.21 | [TK]D-Fender | nny_1: You are missing "callerid=asreceived" |
| 15:46.35 | nny_1 | [TK]D-Fender: interesting, i'll add it |
| 15:48.51 | ManxPower | [TK]D-Fender: Amazing how fast you can diagnose a problem when people do what you tell them to do. |
| 15:49.06 | [TK]D-Fender | ManxPower: yup |
| 15:49.48 | [TK]D-Fender | nny_1: Don't forget that should only be for your FXO ports |
| 15:50.02 | [TK]D-Fender | nny_1: Wouldn't want your FXS looking funky when it calls out |
| 15:50.44 | nny_1 | [TK]D-Fender: FXO only thanks |
| 15:51.17 | [TK]D-Fender | nny_1: signalling=fxo_ks <- you left some old stuff in |
| 15:52.26 | ManxPower | ~fxofxs |
| 15:52.28 | jbot | it has been said that fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 15:53.04 | ManxPower | Hmm. I thought that factoid mentions that FXO ports use FXS signalling and FXS ports use FXO signalling |
| 15:53.10 | ManxPower | I guess not. |
| 15:53.31 | mog | to complicated ManxPower |
| 15:53.59 | ManxPower | mog: go complain to mark, he's the one that designed it. |
| 15:54.14 | [TK]D-Fender | ManxPower: it will |
| 15:54.42 | mog | believe it or not ManxPower telephony existed in some all be it meager premark |
| 15:54.45 | mog | ^_^ |
| 15:54.51 | blebleble | Anyone have any idea, why after exactly 20 seconds outgoing calls get dropped however incomming calls work perfectly fine, "Executing [s@macro-hangupcall:11] Hangup("SIP/" is in the log |
| 15:55.15 | ManxPower | mog: yeah, but it would have been SO simple to make signalling=fxoks really mean fxsks |
| 15:55.24 | ManxPower | then fxs ports get "fxs signaling" |
| 15:55.27 | mog | but that would be wrong |
| 15:55.32 | mog | and people that new the right thing |
| 15:55.37 | mog | would do the wrong thing |
| 15:55.44 | ManxPower | mog: there are MANY things that are wrong in Asterisk. |
| 15:55.45 | *** join/#asterisk Levonk (n=lk@75.62.134.235) |
| 15:55.48 | mog | same problem would ensue |
| 15:56.02 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:56.04 | mog | ManxPower, we are waiting on your patches ^_^ |
| 15:56.09 | nny_1 | [TK]D-Fender: yeah I need to go through now that my education is a little better and clean up (may even rewrite) out conf files |
| 15:56.10 | ManxPower | but my suggestion of "wrongness" helps with thousands and thousands or users. |
| 15:56.20 | ManxPower | mog: or just have the driver figure it out |
| 15:56.33 | ManxPower | mog: It's FAR too late to change it now. |
| 15:56.41 | mog | it would be nice if drivers just did their thing if they could |
| 15:57.03 | ManxPower | mog: They can do the right thing. They don't. |
| 15:57.21 | [TK]D-Fender | blebleble: pastebin the ENTIRE call at verbose 10, SIP DEBUG enabled |
| 15:57.35 | ManxPower | I think the digium analog drivers figure it out automagically |
| 15:57.37 | mog | not all of them can, and their would still be config, but like i said id love to have your patch that just makes it work |
| 15:57.48 | mog | ztscan does that kind of stuff though |
| 15:57.51 | nny_1 | ManxPower: yeah i remember the whole FXO FXS was the first thing i read. Just not being eagle eyed enough to see that in our old zapata.conf |
| 15:58.14 | ManxPower | mog: I am not even close to qualified to submit a patch like that. |
| 15:58.22 | blebleble | [TK]D-Fender: one second |
| 15:58.32 | ManxPower | nny_1: your problem is all that crap in [general] |
| 15:58.45 | [TK]D-Fender | nny_1: Whats the status? |
| 15:58.48 | mog | well id be willing to do it for a LARGE fee ^_^ |
| 15:58.51 | nny_1 | ManxPower: heh i agree |
| 15:59.10 | ManxPower | nny_1: what country are you in? |
| 15:59.53 | dandre | ok I have found that my SIP ST2030 understand Call-Info: answer-after header. |
| 15:59.53 | dandre | How can I send this header in my originate manager command? |
| 16:00.22 | nny_1 | ManxPower: lemme put this fire out, I already was planning on cleaning up our entire set and/or rewriting it and removing all the unrequired crap one day soon |
| 16:00.25 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 16:00.25 | ManxPower | nny_1: http://pastebin.com/m55c2f1f6 |
| 16:00.29 | nny_1 | [TK]D-Fender: no CID still |
| 16:00.36 | nny_1 | ManxPower: thanks US btw |
| 16:00.39 | ManxPower | nny_1: best of luck |
| 16:01.01 | ManxPower | nny_1: well you set your callerid method to a european standard |
| 16:01.29 | [TK]D-Fender | dandre: You can't. Go originate a channel type that WILL allow you to do that. |
| 16:01.35 | ManxPower | what in the WORLD made you think USA used DTMF callerid? |
| 16:02.03 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:02.08 | ManxPower | dandre: you can originate to a Local/ channel and handle it all in the dialplan |
| 16:02.22 | dandre | ok |
| 16:02.30 | nny_1 | ManxPower: wha? |
| 16:02.40 | *** join/#asterisk arpu (n=arpu@chello080109017114.12.14.vie.surfer.at) |
| 16:02.46 | ManxPower | cidsignalling = dtmf cidstart = polarity |
| 16:02.49 | ManxPower | ^^^ NOT USA |
| 16:03.02 | ManxPower | that was in your zapata.conf file. |
| 16:03.05 | nny_1 | ManxPower: wow wtf |
| 16:03.16 | ManxPower | nny_1: now, use the file I created for you and pastebined |
| 16:03.19 | nny_1 | ManxPower: thanks.. Honestly yeah, I need to go ahead and redo our conf files |
| 16:03.20 | *** join/#asterisk AbsTradELic (n=vldmr@unaffiliated/abstradelic) |
| 16:03.22 | nny_1 | ManxPower: i did |
| 16:03.24 | *** part/#asterisk AbsTradELic (n=vldmr@unaffiliated/abstradelic) |
| 16:03.46 | ManxPower | switchtype= is ONLY for ISDN, for example and you had that too. |
| 16:03.53 | nny_1 | ManxPower: the person we paid back when i was goofing around with Asterisknow set these up for me, and I assumed (<-- MY FAULT) they were proper |
| 16:04.01 | blebleble | [TK]D-Fender: pastebin sent |
| 16:04.17 | ManxPower | nny_1: AsteirskNOW? You poor, poor human. |
| 16:04.31 | ManxPower | nny_1: best to nuke the configs and start from scratch. |
| 16:04.33 | nny_1 | ManxPower: time to set up a VM here and start from scratch, reading each field description |
| 16:04.38 | nny_1 | ManxPower: no this is on a clean box |
| 16:05.18 | nny_1 | ManxPower: I originally (wheni first learned about asterisk last year) tried asterisk now just to see what it was about, and someone here helped me set up my first system.. but yeah these conf files may actually have crap left over |
| 16:05.32 | nny_1 | ManxPower: i fully agree on a clean start |
| 16:05.39 | nny_1 | helll i was thinking about it last night |
| 16:05.43 | ManxPower | most of the crap is harmless (like the switchtype) |
| 16:05.54 | nny_1 | ok testing again |
| 16:06.01 | ManxPower | and callwaiting (not supported on FXO) |
| 16:07.08 | nny_1 | ManxPower: heh |
| 16:07.11 | nny_1 | ManxPower: works |
| 16:07.12 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:07.51 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 16:10.03 | nny_1 | ManxPower: ok cool, well the rest of the week I'll spend re designing our conf files |
| 16:10.16 | nny_1 | I really want to strip it down to only whats needed |
| 16:10.42 | nny_1 | if i dont use a feature (say dundi.conf) can I 86 it? |
| 16:19.34 | Qwell | nny_1: sure, might as well add a noload to modules.conf too |
| 16:19.52 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-105-184.pools.arcor-ip.net) |
| 16:20.24 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 16:22.03 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 16:22.42 | nny_1 | Qwell: ok cool |
| 16:23.04 | Blackvel | hi all. on what basis does asterisk select 1 of 2 peers when the patton gateway is the same IP address? I use this: [patton_ext1], [patton_ext2]. the sip invite call looks like <sip:<external-pstn-number>@<pattongwip>:5062> |
| 16:23.18 | jaytee | I just setup a Valcom PA system with * so I can dial a number and get one-way audio so whatever I say comes blasting out the PA horn. |
| 16:23.26 | jaytee | This is so cool |
| 16:23.29 | Blackvel | do i have to setup [patton_ext2] on a different sip port with port=5062? |
| 16:24.17 | Blackvel | my problem is, that asterisk thinks the call should go to [patton_ext1] besides the fact that [patton_ext2] should be the correct peer |
| 16:24.51 | NoxIn- | Blackvel: choising a peer depends of your dialplan |
| 16:25.34 | Blackvel | for incoming calls? |
| 16:25.51 | Blackvel | each patton peer has its own incoming context |
| 16:25.55 | NoxIn- | ho sorry |
| 16:25.59 | ManxPower | Blackvel: no, for incoming calls it matches on username and other stuff. |
| 16:26.11 | NoxIn- | I understood it was call to them |
| 16:26.17 | Blackvel | sorry no... |
| 16:26.20 | Blackvel | <- bad english :( |
| 16:26.25 | ManxPower | NoxIn-: thats because he was vague |
| 16:26.39 | ManxPower | Blackvel: what is the difference between those two peers? |
| 16:26.41 | jaytee | lotta vague going around |
| 16:26.58 | NoxIn- | Blackvel: if both peer register with usernmae/pass asterisk should not confuse one for another |
| 16:27.08 | ManxPower | are they two different gateways or two accounts on the same gateway? |
| 16:27.18 | NoxIn- | if you authentificate using IP and they have the same IP then ... |
| 16:27.43 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 16:27.45 | ManxPower | NoxIn-: "patton" is a SIP gateway and virtually all SIP gateways do not support user/pass auth for outgoing calls to the server. |
| 16:28.09 | ManxPower | But I'm really just waiting for the answer to my question before continuing. |
| 16:28.28 | ManxPower | Looks like he went AFK |
| 16:29.40 | ManxPower | Oh well. |
| 16:29.51 | Blackvel | ManxPower: different contexts. they are not configured as host=192.168.1.4 but as host=dynamic. the patton box registeres to it with identify/username patton_ext1 or patton_ext2. probably this is totally wrong and I would need to change to host=192.168.1.4 for both peers (and not let patton box register). |
| 16:29.52 | Blackvel | for the patton box it means patton_ext1 is pstn bri port 0 (TE mode) and patton_ext2 is internal bri port 1 (NT mode) which is only connected on pbx |
| 16:30.39 | Blackvel | so you guys tell me the decision is made by host ip? |
| 16:31.13 | Blackvel | I can swear this worked yesterday |
| 16:31.24 | ManxPower | Blackvel: so they are both on differnet IPs? |
| 16:32.00 | Blackvel | no same because the patton box is 1 unit with 2 ports (and therefore two virtual sip gateways). so one patton box - one asterisk |
| 16:32.11 | ManxPower | Blackvel: it used never to be based on host=, but in 1.4 I think it will fall back to host= if everything else fails. |
| 16:32.26 | ManxPower | Having registration and differnet IPs should be all that is required. |
| 16:32.37 | Blackvel | 1.4 segfaulted. I am on latest asterisk 1.2.x right now |
| 16:32.46 | ManxPower | Blackvel: it's pretty obvious it's not acting like two different hosts. |
| 16:32.59 | ManxPower | if it registers with user/pass that should be all that is needed. |
| 16:33.00 | Blackvel | so the algo is random right now? |
| 16:33.17 | ManxPower | I never said anything is random. |
| 16:33.34 | Blackvel | i mean choosing between the 2 peers is random (works or might not work) right now :) |
| 16:33.41 | ManxPower | put a a copy of your sip.conf on pastebin. |
| 16:33.46 | *** join/#asterisk ESCulapio__ (n=elvyn@mail.abaca.com.do) |
| 16:33.48 | ManxPower | Blackvel: NO IT IS NOT RANDOM. |
| 16:33.52 | Blackvel | ok :) |
| 16:34.04 | ManxPower | now I am waiting for sip.conf |
| 16:34.14 | Blackvel | <- kicks into his ass for not knowing how the sip protocol works :( |
| 16:34.46 | Blackvel | whole sip.conf? |
| 16:34.47 | ManxPower | Blackvel: if you are going to start using crap sip hardware you had better lean the sip protocol |
| 16:35.01 | ManxPower | Blackvel: how about just patton1 and patton2 and general? |
| 16:35.05 | Blackvel | ok |
| 16:35.38 | angryuser | one of really new to asterisk guys installed zaptel 1.2 and asterisk 1.4 shoul he clean something before reinstaling zaptel 1.4 ? |
| 16:36.06 | angryuser | maybe some modules to delete |
| 16:36.12 | [TK]D-Fender | angryuser: Normall jsut install over 1.2 |
| 16:36.18 | ManxPower | angryuser: no, that is only for 1.2 -> 1.4 asterisk |
| 16:36.35 | angryuser | ok thanks |
| 16:36.39 | ManxPower | 1.2 zaptel to 1.4 zaptel should work fine, but make SURE you recompile astersk after. |
| 16:37.13 | ManxPower | oh and fire the guy that screwed up. |
| 16:37.14 | angryuser | yes it's what i told him |
| 16:37.58 | ManxPower | Imagine if he screwed up the dialplan and allowed people to route calls thru your server -- you could get phone bills reaching the tens of thousands of dollars. |
| 16:38.10 | ManxPower | and it's very easy to have an unsecure Asterisk server |
| 16:39.47 | angryuser | this is his problem, i am just helping get his * running, and his dialplan is more or less sane ;) |
| 16:41.06 | ManxPower | he is too lazy to ask here? |
| 16:42.11 | *** join/#asterisk existenz (n=Ameen@unaffiliated/existenz) |
| 16:42.24 | angryuser | ManxPower: i dont know, maybe he dont speak english very vell, who cares |
| 16:42.28 | existenz | tzafrir_laptop, do you know what is the disctonnect tone in Israel? |
| 16:43.15 | angryuser | existenz: hm normally you got a file under /etc/asterisk with all tones listed by country |
| 16:43.50 | existenz | angler, what's the name of the file |
| 16:44.03 | tzafrir_laptop | busy is listed in the sample indications.conf |
| 16:44.14 | angryuser | ^^^ |
| 16:44.29 | *** join/#asterisk tkbeat (n=tk@p54B97723.dip.t-dialin.net) |
| 16:44.32 | angryuser | and angler is other guy ;) |
| 16:45.27 | Blackvel | patton sip.conf: http://pastebin.com/d236fc5ba. as I can see the sip from does not contain any authentication user |
| 16:45.41 | tzafrir_laptop | one of those filthy tab-completion-wrecking guys :-) |
| 16:46.15 | Blackvel | do you guys think if the registered/authenticated users is send with the sip from (SIP INVITE) asterisk is able to choose between the 2 patton peers? |
| 16:46.48 | *** join/#asterisk steliosk (n=Stelios@79.107.65.33) |
| 16:46.58 | Blackvel | i can remember I had some contact field in the patton gw which I could configure (but didn't do as noone in the ippf forum did it) |
| 16:49.17 | nny_1 | how would i diagnose this further :chan_sip.c: Bad request protocol Caller@192.168.100.10 SIP/2.0 I can't debug 192.168.100.10 (it's the asterisk boxes IP address) |
| 16:49.27 | angryuser | Blackvel: can you explain/develop more ? |
| 16:49.40 | nny_1 | just noticed a crapload of them in asterisk/messages |
| 16:49.49 | existenz | tzafrir_laptop, and hangup? |
| 16:50.05 | tzafrir_laptop | hangup is not a tone |
| 16:50.29 | *** join/#asterisk EI5GTB (n=Paul@78.16.11.162) |
| 16:50.35 | EI5GTB | evening |
| 16:50.42 | existenz | tzafrir_laptop, disconnect tone? |
| 16:51.34 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 16:51.59 | angryuser | existenz: you can play any tone by using this command Playback(!nnn) where nnn is tone number |
| 16:52.13 | angryuser | existenz: maybe try busy tone ? |
| 16:52.28 | existenz | il isn't in the list anyways |
| 16:53.13 | angryuser | existenz: try nearest country |
| 16:53.34 | ManxPower | in some parts of the world a hangup IS a tone. |
| 16:53.38 | ManxPower | sad, but true. |
| 16:54.25 | EI5GTB | dumb question: is Zaptel a protocall? |
| 16:54.31 | *** join/#asterisk HumpBack (n=sdcdcssc@gentoo/developer/HumpBack) |
| 16:54.44 | tzafrir_laptop | existenz, look for [il] in http://svn.digium.com/svn/asterisk/branches/1.4/configs/indications.conf.sample |
| 16:55.45 | HumpBack | Hello all. I am working on a application that will need to do some changes to the configuration of an asterisk server. I have been looking at the http interface to the manager. |
| 16:55.48 | existenz | tzafrir_laptop, how can I translate it to this format << 400\@-12;10(.5/.5/1) >> ? |
| 16:57.23 | HumpBack | I managed to send login commands ( wget "http://127.0.0.1:8088/asterisk/rawman?action=login&userna .......") and it works fine |
| 16:57.23 | HumpBack | But if i send any other comand afterwards i receive a Authentication Required |
| 16:57.23 | HumpBack | How do I maintain some kind of persistence? |
| 16:57.26 | tzafrir_laptop | EI5GTB, Zaptel is an interface. Not really a well-defined protocol |
| 16:57.47 | ManxPower | HumpBack: Manager is not designed to modify config files. |
| 16:57.48 | tzafrir_laptop | (well, it implies a protocol of some sort, but it is not VoIP) |
| 16:58.05 | EI5GTB | i see.... its to do with actual hardware then? |
| 16:58.10 | tzafrir_laptop | yes |
| 16:58.14 | EI5GTB | grand |
| 16:58.28 | ManxPower | EI5GTB: It's a DRIVER with a specific API. Protocol implies inter process or inter host communication. |
| 16:58.37 | EI5GTB | i see |
| 16:58.57 | EI5GTB | i didnt know it was a driver.. i thought it was a protocall of some sort |
| 16:59.01 | ManxPower | Zaptel is just an API to talk to telephony cards, like a video driver or a printer driver |
| 16:59.03 | EI5GTB | thanks for clearing it up guys |
| 16:59.03 | *** join/#asterisk Psychobilly (n=moi@adsl139-217.kln.forthnet.gr) |
| 16:59.19 | nny_1 | HumpBack: we are looking at a combination of asterisk realtime along with static configs to do what you are doing |
| 16:59.23 | existenz | tzafrir_laptop, do you know how can I translate it? |
| 16:59.40 | nny_1 | HumpBack: basically including the asterisk realtime data on top of the static extensions.conf |
| 17:00.21 | ManxPower | nny_1: and that task is NOT simple. |
| 17:00.26 | tzafrir_laptop | "busy = 414/500,0/500" - we have an infinite cycle of: |
| 17:00.28 | nny_1 | ManxPower: heh no it's not |
| 17:00.57 | tzafrir_laptop | ring of 414Hz for 500 ms, and then silence for 500 ms |
| 17:01.08 | ManxPower | HumpBack: you are trying to do one of the hardest things there is to do with Asterisk. |
| 17:01.53 | nny_1 | ManxPower: contextually we were thinking we could modify the mysql database to add fields (user name, sip peer, auth, "macro used" etc) |
| 17:02.06 | tzafrir_laptop | HumpBack, do you have the astman.js sample working? |
| 17:02.29 | ManxPower | nny_1: The first step is to just move the static configs into Realtime/Database. That'll take a few days at least |
| 17:02.38 | tzafrir_laptop | If so, sniff to it one way or the other to see what your borwser sends |
| 17:02.48 | nny_1 | ManxPower: our [people] section is exten => 100,1,Macro(stdexten,190,SIP/190) |
| 17:02.51 | HumpBack | ManxPower: I worked with asterisk before. I know it will be hard. Right now I just want to send any command to the manager, besides login |
| 17:03.04 | tzafrir_laptop | (or look at the docs. I think there is something about "AJAM" there) |
| 17:03.12 | ManxPower | HumpBack: Can't really help you with that, but I bet tzafrir_laptop can |
| 17:03.12 | nny_1 | ManxPower: I have heard two schools of thought, including being able to do both |
| 17:03.29 | nny_1 | ManxPower: the idea of the static configs is to define things that won't change |
| 17:03.35 | ManxPower | nny_1: My school of thought is "it is almost always easer to not use Realtime" |
| 17:03.40 | HumpBack | tzafrir_laptop: I do not know what astman.js is. |
| 17:03.41 | nny_1 | ManxPower: ha agreed |
| 17:03.52 | ManxPower | With a few exceptions of course. |
| 17:03.53 | existenz | tzafrir_laptop, the other format is quite weird |
| 17:03.56 | nny_1 | ManxPower: sounds like an asterisk manager java script :) |
| 17:04.04 | nny_1 | <--- highly intuitive :P |
| 17:04.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:04.12 | tzafrir_laptop | http://svn.digium.com/svn/asterisk/branches/1.4/static-http/ |
| 17:04.17 | nny_1 | er oops that wasn't meant for just Manxpower |
| 17:04.19 | ManxPower | nny_1: we have exten => lines for every extension, but all it does is set channel variables (to control the macro) and run a macro |
| 17:04.37 | nny_1 | ManxPower: yeah same here more or less, Macro, ARG1 and ARg2 |
| 17:05.04 | ManxPower | gives us ultimate flexibility and we can move to a database VERY easily -- just create an AGI that sets the channel variables. |
| 17:05.40 | tzafrir_laptop | HumpBack, IIRC every request needs to be authenticated. There is a session, but it is really short-lived |
| 17:05.51 | ManxPower | nny_1: http://www.fnords.org/~eric/macro-std-exten-v2.inc |
| 17:06.14 | nny_1 | ManxPower: ha BARF |
| 17:06.42 | *** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 17:06.51 | ManxPower | BTW, there is also http://www.fnords.org/~eric/polycom-config-examples/ for the polycom people as well as macro-dial-result-v2.inc for failover etc call routing |
| 17:07.03 | nny_1 | ManxPower: wow there is a lot more info in you macros than mine ^^ |
| 17:07.22 | ManxPower | nny_1: our mail manager program is called "maim" MAIl Manager |
| 17:07.41 | ManxPower | nny_1: my users are prissy little creatures that want it all and not want to learn anything. |
| 17:07.43 | pta200 | Is it possible to set value for ptime/maxptime in sip message via configuration file? |
| 17:07.48 | ManxPower | we they mostly get what they want |
| 17:07.55 | ManxPower | pta200: no |
| 17:07.58 | nny_1 | ManxPower: heh |
| 17:08.07 | ManxPower | unless ptime is like "register time" |
| 17:08.08 | HumpBack | tzafrir_laptop: wget "http://127.0.0.1:8088/asterisk/rawman?action=login&username=asterisk_http&secret=*****" ; wget "http://127.0.0.1:8088/asterisk/rawman?action=getconfig&filename=users.conf" |
| 17:08.16 | ManxPower | pta200: all options should be shown in sip.conf.sample |
| 17:08.20 | HumpBack | tzafrir_laptop: even this gives an error on the second command |
| 17:08.24 | pta200 | that's what I figured. Thanks |
| 17:08.55 | ManxPower | pta200: patches are welcome, just be prepared to justify why people would find that feature helpful. |
| 17:09.00 | nny_1 | ManxPower: heh http://pastebin.com/m4e12cb10 |
| 17:09.15 | nny_1 | ManxPower: not very impressive I know |
| 17:09.23 | tzafrir_laptop | HumpBack, sorry, never really used it "manually" |
| 17:09.54 | existenz | tzafrir_laptop, the right busy tone? 414\@-12;10(.5/.5/1) ? |
| 17:09.55 | tzafrir_laptop | HumpBack, what are you trying to do with that? Can you use the standard manager interface? |
| 17:10.06 | nny_1 | ManxPower: but the idea is to have an include from [people] to check the realtime DB the extension number, macro and ARg1/ARG2 |
| 17:10.21 | Qwell | realtime...meh. use func_odbc |
| 17:10.35 | nny_1 | Qwell: interesting |
| 17:10.47 | Qwell | func_odbc is teh hotness |
| 17:11.00 | nny_1 | Qwell: reading now |
| 17:11.03 | ManxPower | nny_1: *nod* We just don't have that many adds/moves/changes |
| 17:11.04 | Qwell | perfect for getting random data from a DB |
| 17:11.12 | ManxPower | and the current asterisk admin can add a new phone in like 5 mins. |
| 17:11.15 | HumpBack | tzafrir_laptop: the idea is to use the mxml output and parse it in another application |
| 17:11.45 | nny_1 | Qwell: looks like I can just use it to define the individual bits in exten => 100,1,Macro(stdexten,190,SIP/190) |
| 17:11.59 | *** join/#asterisk mog (n=mog@nat/digium/x-719df25a065cbca5) |
| 17:11.59 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:12.01 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 17:12.10 | EI5GTB | is 1.6 the lates asterisK? |
| 17:12.34 | Qwell | EI5GTB: it really depends on your definition of "latest" |
| 17:12.44 | denon | if you have to ask, you're probably interested in 1.4 |
| 17:12.48 | ManxPower | 1.6 does not exist. |
| 17:12.52 | ManxPower | 1.6 betas exist |
| 17:12.54 | nny_1 | ManxPower: Qwell our over all goal is to allow our clients to add or change a user via a simple* web interface. We already have a front end that includes the CDR stuff, a custom moh upload tool, something to check voicemails, something to listen to recorded conversations, and a php system status thing |
| 17:13.05 | EI5GTB | Qwell, well, the current stable |
| 17:13.09 | nny_1 | I do not want to recreate any of the nasty guis |
| 17:13.18 | ManxPower | nny_1: We don't let users change their own settings. /me shudders at the thought. |
| 17:13.19 | denon | EI5GTB: 1.4 |
| 17:13.20 | Qwell | 1.6.0 is rc. |
| 17:13.32 | ManxPower | We have enough problem getting them to remember to put 9 in their call forwarding number |
| 17:14.35 | nny_1 | ManxPower: yeah. our systems are 5-30 SOHO. We give them the option. If they break it or do something stupid, hopefully our stuff catches it and/or we charge them to unf*ck it. It's not actually meant to be used by anyone other than their local admin |
| 17:14.35 | EI5GTB | ktnx |
| 17:15.09 | nny_1 | ManxPower: and the only functionality I am gonna allow is to change an extension password or User name, or add a new one. |
| 17:15.37 | nny_1 | ManxPower: and right on the web interface in big ass bold letters is "if you change this, you MUST change it in the phone too" |
| 17:15.49 | nny_1 | ManxPower: it's purely a work in progress |
| 17:16.07 | nny_1 | and only will be "released" once I am satisified that the screw up factor is low |
| 17:16.20 | nny_1 | right now I just prefer to do these things myself |
| 17:17.14 | nny_1 | Qwell: gonna look at func_odbc now |
| 17:17.46 | EI5GTB | ~centos52bug |
| 17:17.48 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 17:18.03 | EI5GTB | hmm, 1.4.11 |
| 17:18.37 | ManxPower | EI5GTB: that is the latest Zaptel verson |
| 17:18.42 | EI5GTB | yea |
| 17:18.44 | Qwell | for the moment |
| 17:19.05 | nny_1 | Qwell: do you use MySQL with the odbc stuff? |
| 17:19.12 | Qwell | nny_1: I have |
| 17:19.28 | Blackvel | how must a SIP INVITE like that asterisk can choose between peer [patton_ext1] and [patton_ext2] when the call comes from the same IP? |
| 17:19.37 | Blackvel | look like |
| 17:20.13 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 17:20.35 | nny_1 | Qwell: going to do a lot of testing in the upcoming months. I'll test the func_odbc as well |
| 17:20.42 | EI5GTB | i suppose theres not much point installing zaptell then atm |
| 17:20.50 | *** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 17:20.53 | ManxPower | Blackvel: thousands of people use multiple devices on one IP (NAT) and it works just fine. |
| 17:20.55 | Qwell | EI5GTB: the fix is incredibly simple |
| 17:21.06 | EI5GTB | does it make zaptel work tho.. |
| 17:21.12 | EI5GTB | or soes it diable it? |
| 17:21.25 | EI5GTB | i may be adding some hardware st some stgee you see |
| 17:21.30 | Qwell | it disables parts of it. unless you have Xorcom hardware, you don't need that stuff |
| 17:21.36 | EI5GTB | ah, cool. |
| 17:21.49 | Qwell | but, like it says, it's already fixed in svn. you can install from there if you need |
| 17:22.58 | EI5GTB | oh... |
| 17:22.59 | EI5GTB | hmm |
| 17:23.07 | EI5GTB | that may be a simpler plan to a nub... |
| 17:23.15 | EI5GTB | s/to/for |
| 17:23.35 | clyrrad | Is there a way to show the "full channel name" in the CLI? show channels cuts off the full channel name for formatting reasons, and I need to get the full channel name. Anyone can point me in the right direction? |
| 17:26.51 | EI5GTB | Qwell, so you say i should just install from the 1.6.1 svn? |
| 17:27.03 | Qwell | no |
| 17:27.12 | EI5GTB | 1.4? |
| 17:27.12 | Qwell | I'm saying zaptel 1.4 svn |
| 17:27.16 | EI5GTB | ah |
| 17:27.18 | EI5GTB | so'h |
| 17:27.23 | EI5GTB | do'h* |
| 17:27.27 | Qwell | if you're using asterisk 1.4.. |
| 17:27.31 | EI5GTB | yea |
| 17:27.49 | *** join/#asterisk jjohns4 (n=Jeff@pool-71-180-122-2.tampfl.fios.verizon.net) |
| 17:29.43 | clyrrad | anyone know if its possible to do what I need? The channel name is being cut off and I need to get the full channel name. "show channels" does not do what I need. Any help would be apprecated. |
| 17:30.00 | seanbright | core show channels concise |
| 17:30.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:31.12 | clyrrad | seanbright: trying that |
| 17:31.49 | clyrrad | seanbright: perfect, thats exactly what I needed :) thanks - that was driving me crazy |
| 17:32.18 | seanbright | no sweat |
| 17:32.34 | clyrrad | figured there had to be a way to do it, I just couldnt find it lol........ |
| 17:32.46 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088810630.dsl.bell.ca) |
| 17:32.46 | seanbright | you can get the full channel name via AMI as well |
| 17:33.10 | clyrrad | yea that I knew, but wanted to do it on the CLI |
| 17:33.18 | clyrrad | so I could use soft hangup |
| 17:33.24 | seanbright | gotcha |
| 17:33.31 | seanbright | tab completion wasn't enough? |
| 17:33.35 | Blackvel | ManxPower: as this is only my internal networks its not even an NAT issue. And I can swear it was working yesterday :) as long as two accounts on patton register to asterisk its enough to identify peer 1 or 2? patton only forwards to fix ip 192.168.1.2 for both bri interfaces. I still have no clue how Asterisk chooses the "right" peer when the INVITE does not contain any user/authname from the patton? |
| 17:33.56 | tzafrir_laptop | so this looks like a good time to commit that zaptel fix... |
| 17:34.02 | clyrrad | seanbright: damn, I didnt know the CLI had that - thats even better |
| 17:34.05 | clyrrad | :) |
| 17:34.16 | tzafrir_laptop | err... wrong channel |
| 17:34.20 | clyrrad | makes a permanant menal note on that one |
| 17:34.33 | clyrrad | <PROTECTED> |
| 17:34.42 | seanbright | clyrrad: tab completion saves lives. |
| 17:35.05 | clyrrad | indeed, thats a good one to know, I use it in Linux all the time, but didnt know the CLI had that - this is pretty sweet, gonna come in handy |
| 17:40.35 | clyrrad | hrm, ever get it when soft hangup wont actually hangup the channel? |
| 17:43.31 | [TK]D-Fender | clyrrad: pastebin your channel dump & your attempt |
| 17:43.57 | clyrrad | ok |
| 17:46.17 | *** join/#asterisk Firass-z0r (n=asadf@ead224-222.housing.wwu.edu) |
| 17:48.26 | clyrrad | [TK]D-Fender: http://pastebin.ca/1192759 |
| 17:49.23 | [TK]D-Fender | clyrrad: Setup an exten that only calls "hangup" and issue a redirect to it |
| 17:49.42 | clyrrad | ok sure |
| 17:50.34 | clyrrad | [TK]D-Fender: then I suppose you want me to call it and see if the channel is actually hung up? |
| 17:50.47 | EI5GTB | huh? i try to run 'make menuselect' and i am told to install ncurses...but its installed.. |
| 17:50.49 | [TK]D-Fender | clyrrad: No, just redirect that channel to it |
| 17:50.58 | [TK]D-Fender | EI5GTB: ncurses-devel |
| 17:50.59 | *** join/#asterisk existenz (n=Ameen@unaffiliated/existenz) |
| 17:51.07 | EI5GTB | ah.... |
| 17:51.12 | clyrrad | [TK]D-Fender: hrm, not sure how do I redirect a live channel to it? |
| 17:51.13 | existenz | tzafrir_laptop, after I changed it to busy tone, the line goes busy after hangup =/ |
| 17:51.15 | [TK]D-Fender | EI5GTB: Go back and read *'s full list of dependencies |
| 17:51.18 | EI5GTB | puts on a hat with a D on it.. |
| 17:51.23 | [TK]D-Fender | clyrrad: AMI <- |
| 17:51.26 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-147.moldtelecom.md) |
| 17:51.29 | kotique | yo. |
| 17:51.31 | EI5GTB | the bad thing is, i did"!! |
| 17:51.42 | Qwell | EI5GTB: re-run configure |
| 17:51.50 | Qwell | or just make dist-clean and start over |
| 17:51.52 | [TK]D-Fender | ^^ |
| 17:51.53 | clyrrad | [TK]D-Fender: all I have here is CLI access no AMI |
| 17:52.04 | EI5GTB | Qwell, ktnx |
| 17:52.18 | [TK]D-Fender | clyrrad: If you have access to CLI (OS), then yuo can AMI locally from the machine |
| 17:52.25 | kotique | guys, I'm trying to use existing Asterisk server and I'm stuck on INCOMING sip connections. I want to use different username, not that in [username] line |
| 17:52.47 | existenz | tzafrir_laptop, it stops automatically, but then goes busy. |
| 17:53.22 | clyrrad | [TK]D-Fender: Im not exactly sure on how to do what your saying I have not done this before. I assume I would have to setup AMI then find some command that does what your saying |
| 17:53.43 | tzafrir_laptop | existenz, can you please pastebin a CLI trace? |
| 17:53.44 | [TK]D-Fender | clyrrad: I even gave you the magic word.... now hop to it :) |
| 17:53.50 | existenz | tzafrir_laptop, sure |
| 17:54.24 | clyrrad | [TK]D-Fender: alright let me see what I can google up for this |
| 17:54.51 | [TK]D-Fender | BOOK <- |
| 17:55.40 | clyrrad | better yet, show manager command redirect :P |
| 17:56.28 | clyrrad | [TK]D-Fender: ah, i think i got how to do it, lets give this a shot |
| 17:57.22 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 17:58.13 | clyrrad | [TK]D-Fender: ok got the syntax and what I need to do, only quesiton is how to envoke this command from the CLI |
| 17:58.39 | clyrrad | err CML |
| 17:59.05 | *** join/#asterisk raz (n=y@unaffiliated/raz) |
| 17:59.14 | EI5GTB | hmm, i installed ncurses-devel.. but still no menuselect.. |
| 17:59.15 | raz | hi guys |
| 17:59.24 | raz | researching asterisk security right now. is this fixed: http://www.milw0rm.com/exploits/2597 ? |
| 18:00.57 | [TK]D-Fender | clyrrad: TELNET <- |
| 18:00.57 | Nugget | telnet is eeeeeeevil! |
| 18:00.59 | *** join/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 18:01.07 | clyrrad | [TK]D-Fender: yea i was just doing that |
| 18:01.07 | Blackvel | kotique: for what incoming sip connections? looks like I have problems with it myself. you can try to set username, fromuser, authuser |
| 18:01.31 | [TK]D-Fender | raz: look at the VERSION. |
| 18:01.44 | Blackvel | how can I check if the patton gateway is successfully registered as peer (status unmonitored and qualify=no)? |
| 18:01.53 | *** part/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 18:02.10 | Blackvel | only sip debug? |
| 18:02.15 | [TK]D-Fender | Blackvel: "show peer [peerwithoutbraces]" |
| 18:02.24 | [TK]D-Fender | Blackvel: "sip show peer [peerwithoutbraces]" |
| 18:03.13 | *** join/#asterisk Levonk (n=lk@76.227.117.22) |
| 18:04.34 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 18:05.17 | Blackvel | i just feel too lame... but what property should I look into for the sip show peer patton ? useragent + reg.contact? if its empty its not registered and if its registered both lines are filled? |
| 18:05.59 | [TK]D-Fender | Blackvel: Pretty much. Also, is your "host=dynamic" for them in sip.conf? |
| 18:07.11 | Blackvel | right now yes. as it seems without it the patton can not do authbound registration? I can remember I had it like host=192.168.1.4 and then I got complains that the peer is not meant to register |
| 18:07.18 | kotique | hey, I don't want to use string in braces, I want to use separate string in configuration section |
| 18:07.24 | *** join/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 18:07.51 | kotique | string in braces is used by vop software and can't be changed and I'm not supposed to know it. |
| 18:08.19 | Blackvel | MUST there be insecure=very or nowadays insecure=port,invite for patton sip call forwards? |
| 18:08.57 | [TK]D-Fender | belacDepends how it sends calls on to you |
| 18:08.59 | Blackvel | i removed it and now it seems it has authenticated (as sip show peer <peername> displays it as reg'ed) |
| 18:09.34 | kotique | wha'ts the difference between SIP trunk and non-SIP trunk ? |
| 18:09.39 | [TK]D-Fender | kotique: HUH!? |
| 18:09.45 | raz | [TK]D-Fender, yea sorry, tired |
| 18:10.03 | kotique | i mean SIP turnk and SIP non-trunk |
| 18:10.15 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 18:10.25 | clyrrad | [TK]D-Fender: ok i did exactly what you said, no errors on the command or with authentication, howerver when I run the command via telnet i get a redirect response of "Error" and Message" Redirect failed, channl not up" - I double checked the channel name to ensure no errors were made |
| 18:10.35 | [TK]D-Fender | kotique: You're still making no sense at all. |
| 18:10.45 | clyrrad | <PROTECTED> |
| 18:10.58 | [TK]D-Fender | clyrrad: I trust pastebin. thats about it :) |
| 18:11.19 | clyrrad | ok |
| 18:13.04 | kotique | what's the difference between SIP trunk and SIP non-trunk ? I don't get why these terms are differentiated in documentation. In the end, it's all the same. |
| 18:13.16 | clyrrad | [TK]D-Fender: http://pastebin.ca/1192800 |
| 18:13.27 | [TK]D-Fender | kotique: redundent term. Forget it |
| 18:14.07 | vonkleist | Hi everybody |
| 18:14.28 | vonkleist | I've a TE121 card in a constant blue/recovering alarm |
| 18:14.39 | vonkleist | what could be happening here? |
| 18:14.42 | [TK]D-Fender | clyrrad: Do you stills ee the channel? |
| 18:14.49 | clyrrad | [TK]D-Fender: indeed I do |
| 18:15.00 | mvanbaak | kotique: to answer your question in #asterisk-dev |
| 18:15.03 | [TK]D-Fender | clyrrad: Ok, don't know what to tell you other than to restart. |
| 18:15.03 | mvanbaak | ~siptrunk |
| 18:15.04 | jbot | No such thing, my friend.. Like too much salty plum soda. |
| 18:15.13 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 18:15.19 | clyrrad | [TK]D-Fender: yea I know that works, I just didnt want to drop everyones calls |
| 18:15.30 | clyrrad | [TK]D-Fender: there is no other way to force a hangup huh? |
| 18:15.38 | [TK]D-Fender | clyrrad: Is it actually interefering with anything? |
| 18:15.49 | clyrrad | [TK]D-Fender: Yes, realtime Queue members |
| 18:16.02 | [TK]D-Fender | clyrrad: Ok, I'm all out of advice then. |
| 18:16.02 | clyrrad | it keeps them "busy" so they cant take calls |
| 18:16.14 | *** join/#asterisk tkbeat (n=tk@p54B97723.dip.t-dialin.net) |
| 18:16.20 | kotique | mvanbaak, I know what is trunk in traditional telephony |
| 18:16.25 | clyrrad | I cant figure out why these channels dont die when the person hangs up |
| 18:16.28 | kotique | but with SIP, it just doesn't make any sense |
| 18:16.37 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:16.58 | mvanbaak | kotique: that's why there's no such thing :) |
| 18:17.09 | *** join/#asterisk belendax (n=belendax@unaffiliated/belendax) |
| 18:17.33 | clyrrad | [TK]D-Fender: "(In use)" is actually how it shows them |
| 18:17.48 | flujan | hi guys, I am trying to make asterisk dial-out using .call files, the originate command in the cli and even the AMI. My doubt is that asterisk sometimes do the call and sometimes do nothing... I don't have a way to control and monitor if the call is going or what... for instance, i have nothing in the CLI. |
| 18:18.07 | flujan | someone already tried that and end with the same problem? |
| 18:18.08 | [TK]D-Fender | kotique: Use a bad term often enough and people start believing its legit. Just drop it already. |
| 18:18.42 | [TK]D-Fender | flujan: Show us something useful. |
| 18:18.55 | belendax | I can't configure Aserisk in ubuntu , when I ./configure it says configure: error: *** termcap support not found |
| 18:18.55 | belendax | <PROTECTED> |
| 18:18.57 | clyrrad | flujan: you are using mv to put the .call file and not cp right? |
| 18:19.16 | clyrrad | [TK]D-Fender: I take it your out of ideas too, thanks for trying though :) |
| 18:19.18 | kotique | ok |
| 18:19.53 | [TK]D-Fender | belendax: Go install *'s dependencies. Its all listed in the book, and several other docs |
| 18:20.09 | [TK]D-Fender | belendax: that one in partuicular is libtermcap |
| 18:20.11 | flujan | clyrrad: yeap |
| 18:20.54 | belendax | [TK]D-Fender: yes but libtermcap is not in synaptic for ubuntu |
| 18:21.39 | [TK]D-Fender | belendax: missing a "-" somewhere I'm sure. |
| 18:22.31 | [TK]D-Fender | belendax: http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu |
| 18:22.39 | [TK]D-Fender | 8. Compile Asterisk. You may need to install the libncurses5-dev package if configure complains about 'termcap'. |
| 18:23.16 | [TK]D-Fender | belendax: Go follow the guide for every spot you get stuck on |
| 18:23.48 | rasterix | voip-info needs creation date and last updated date on all its pages |
| 18:24.07 | seanbright | it's at the bottom i thought |
| 18:24.10 | belendax | [TK]D-Fender: ok tnx |
| 18:24.21 | seanbright | the last update date is on the bottom |
| 18:24.24 | InHisName | Tried to do a 'soft hangup' on wayward channel but still there. |
| 18:24.26 | rasterix | wow |
| 18:24.34 | [TK]D-Fender | rasterix: looked REAL hard on this one I see |
| 18:24.37 | rasterix | i never noticed that |
| 18:24.51 | rasterix | < already admitted hes an idiot multiple times |
| 18:24.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:25.07 | clyrrad | InHisName: Join the club soft hangup not working for me either :P |
| 18:25.20 | rasterix | they need to put it in BOLD at the top for idiots like me :) |
| 18:26.29 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 18:26.39 | vonkleist | I have this on zaptel.conf : span = 1,1,1,cas,hdb3,yellow |
| 18:27.04 | vonkleist | but zttool still says "Internally clocked" for this card |
| 18:27.15 | vonkleist | #who gluch |
| 18:27.40 | vonkleist | any clue? |
| 18:30.54 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 18:35.50 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 18:36.21 | *** join/#asterisk kalel008 (n=tamato@41-195-84-21.access.uunet.co.za) |
| 18:36.28 | kalel008 | hi there |
| 18:36.47 | kalel008 | how do i reload the voicemail after i have set it up? |
| 18:37.36 | seanbright | module reload app_voicemail.so |
| 18:37.58 | kalel008 | thx |
| 18:38.21 | EI5GTB | if i have no hardware that zaptel is using, is it imperitive that i put in ztdummy? |
| 18:38.34 | seanbright | if you would like timing, yes. |
| 18:38.42 | EI5GTB | iv just 'make install'ed it without having it in =/ |
| 18:39.03 | EI5GTB | heh, timig would be nice.. |
| 18:39.05 | EI5GTB | dam, |
| 18:39.32 | EI5GTB | can i just make the changes and make clean ; make install again? |
| 18:39.35 | [TK]D-Fender | EI5GTB: Recompile * after doing zaptel |
| 18:39.39 | EI5GTB | or do i have to uninstall the dsam thing first? |
| 18:39.48 | EI5GTB | asterisk is not installed yet |
| 18:39.52 | [TK]D-Fender | EI5GTB: No, jsut compile over from scratch. |
| 18:39.57 | EI5GTB | k |
| 18:40.04 | EI5GTB | tnx |
| 18:40.09 | kalel008 | is this correct in extensions.conf exten => 011,n,VoiceMail(0110@default,b) |
| 18:40.32 | [TK]D-Fender | kalel008: the format is valid |
| 18:40.47 | [TK]D-Fender | kalthe content, well thats another matter |
| 18:41.05 | kalel008 | well when i call that extension it does not go to the voivemail menu |
| 18:41.20 | [TK]D-Fender | kalel008: pastebin the entire context its in. |
| 18:41.22 | [TK]D-Fender | ~pb |
| 18:41.22 | jbot | [~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 |
| 18:41.24 | [TK]D-Fender | ^^^^^^^^^^ |
| 18:41.37 | seanbright | >>>>>>>>>> |
| 18:42.22 | EI5GTB | looks for a down arrow |
| 18:42.38 | seanbright | vvvvvvvvvv |
| 18:42.45 | kalel008 | in my voicemail default context 011 => 1234,jhb1 |
| 18:43.04 | seanbright | kalel008: what was unclear about: <[TK]D-Fender> kalel008: pastebin the entire context its in. |
| 18:43.04 | EI5GTB | laughs at his stupidity =/ |
| 18:43.08 | [TK]D-Fender | kalel008: dialplan context please. |
| 18:43.19 | *** join/#asterisk io_error (n=error@unaffiliated/ioerror/x-384576) |
| 18:43.42 | kalel008 | exten => 011,1,Answer() |
| 18:43.48 | io_error | Hey, what is this rumor I hear about the IAXy being EOL'd? |
| 18:44.10 | [TK]D-Fender | kalel008: pastebin your failed call from beginning to end. |
| 18:44.10 | EI5GTB | i was told in a guide that i would just have to uncomment #ztdummy......but all i can find in the makefile is topdir_modules= bvla bal balalab ztdummy bla |
| 18:44.12 | kalel008 | exten => 011,n,VoiceMail(0110@default,b) |
| 18:44.17 | EI5GTB | that means it has been loaded correctly? |
| 18:44.34 | seanbright | EI5GTB: why are you looking at the makefile? |
| 18:44.40 | seanbright | EI5GTB: run 'make menuselect' |
| 18:44.45 | EI5GTB | to make sure ztdummy is there.. |
| 18:44.49 | kalel008 | then hangup after |
| 18:44.51 | EI5GTB | seanbright, ah... |
| 18:44.54 | EI5GTB | ffs, dumb me |
| 18:45.29 | tzafrir_laptop | EI5GTB, that documentation is obsolete. You no longer need to uncomment anything in thr makefile |
| 18:45.29 | [TK]D-Fender | kalel008: pastebin your failed call from beginning to end. |
| 18:45.29 | kaldemar | kalel008: are you trying to use VoiceMailMain maybe? |
| 18:45.29 | kalel008 | i want it to go straight into voicemail when you dial the extension |
| 18:45.29 | seanbright | EI5GTB: don't beat yourself up too much... that is why we are here. |
| 18:45.35 | EI5GTB | haha |
| 18:45.37 | EI5GTB | spose |
| 18:45.39 | EI5GTB | thanks |
| 18:45.49 | seanbright | kalel008: can you read this message? |
| 18:45.50 | kalel008 | to leave the message |
| 18:45.54 | tzafrir_laptop | EI5GTB, where did you read that? |
| 18:45.56 | seanbright | kalel008: hello, kalel008? |
| 18:46.04 | seanbright | kalel008: are you there kalel008? hello? |
| 18:46.08 | kalel008 | yeh |
| 18:46.14 | EI5GTB | tzafrir_laptop, http://www.automated.it/guidetoasterisk.htm |
| 18:46.21 | EI5GTB | was just using it as a reference |
| 18:46.28 | EI5GTB | im not going by it biblically... |
| 18:46.30 | seanbright | kalel008: great. now stop talking and start reading the messages [TK]D-Fender has been sending you and DO WHAT HE ASKS. |
| 18:46.47 | kalel008 | it went to fast didnt c it sorry |
| 18:46.54 | seanbright | sigh |
| 18:47.04 | seanbright | 14:45 <[TK]D-Fender> kalel008: pastebin your failed call from beginning to end. |
| 18:47.28 | seanbright | but in your defense, he only asked variations of that same question 8 times |
| 18:47.35 | kalel008 | i dont understand that pls explain |
| 18:47.36 | tzafrir_laptop | EI5GTB, literally? I hope you have not installed RH8 or RH9 |
| 18:47.41 | EI5GTB | nope |
| 18:47.42 | EI5GTB | centos |
| 18:47.48 | EI5GTB | 5.2 |
| 18:47.54 | [TK]D-Fender | kalel008: Go to * CLI and pastebin the entire call's CLI output |
| 18:48.06 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 18:48.19 | Assid | hey |
| 18:48.27 | kalel008 | pastebin? |
| 18:48.29 | tzafrir_laptop | That guide is obsolete with quite a few useless points (e.g.: mpg123) |
| 18:48.37 | EI5GTB | ha, yea |
| 18:48.48 | EI5GTB | im using the o'rielly asterisk book too |
| 18:49.05 | [TK]D-Fender | ~pb |
| 18:49.06 | jbot | [~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 |
| 18:49.09 | [TK]D-Fender | ^^^ |
| 18:49.24 | Assid | anyone know know a business grade unlimited outgoing? |
| 18:49.30 | Assid | like 4 channels? |
| 18:49.36 | tzafrir_laptop | EI5GTB, start at http://voip-info.org/ |
| 18:49.38 | kalel008 | open the link right |
| 18:49.49 | tzafrir_laptop | At least non-sense in there can be fixed |
| 18:49.59 | EI5GTB | haha, true |
| 18:50.07 | EI5GTB | iv been reading a bit on that site too |
| 18:50.33 | EI5GTB | i have a good/bad habbit of diving straig in and seeing how much i can break things... before starting again and doing it right :P |
| 18:50.54 | kalel008 | im sorry its my first asterisk box and first time here |
| 18:51.22 | [TK]D-Fender | kalel008: you've been here for several days now. |
| 18:51.32 | kalel008 | this week only |
| 18:51.33 | [TK]D-Fender | kalel008: And you need to provide US with the link to your pastebin |
| 18:53.03 | kalel008 | how do i paste that into a web browser or in the cli |
| 18:53.05 | EI5GTB | compile flag: RADIO_RELAX - relax DTMF for radio applications |
| 18:53.09 | EI5GTB | in what way does it relax? |
| 18:53.23 | EI5GTB | give a bit of room for off frequency? |
| 18:53.31 | [TK]D-Fender | kalel008: copy your CLI output and paste it to one of those service |
| 18:53.57 | kalel008 | the link doesnt open anything |
| 18:54.11 | kalel008 | its a blank page |
| 18:54.11 | [TK]D-Fender | kalel008: Do it over again |
| 18:54.58 | [TK]D-Fender | kalel008: link please |
| 18:55.55 | seanbright | man... this is PAINFUL |
| 18:56.06 | kalel008 | ok its in now |
| 18:56.11 | seanbright | Asterisk - Reach Out And Strangle Someone (tm) |
| 18:56.12 | kalel008 | its in now |
| 18:56.51 | *** join/#asterisk i3inary (n=shaman@cpe-76-88-94-149.san.res.rr.com) |
| 18:57.01 | seanbright | kalel008: super... you need to paste the link into this channel. |
| 18:57.05 | seanbright | kalel008: otherwise we can't see it |
| 18:57.12 | kalel008 | http://pastebin.com/d7db36c0 |
| 18:57.27 | EI5GTB | grasping the concepts of pastebin should be easy compared to installing asterisk for the first time |
| 18:57.37 | kalel008 | ok sorry guys im a bit slo |
| 18:58.13 | EI5GTB | so am i! its all good, you always get there in the end |
| 18:58.37 | kalel008 | was that the correctlink |
| 18:58.45 | seanbright | yes... it was |
| 18:58.51 | seanbright | but i don't even know where to start... |
| 18:58.51 | kalel008 | aah cool |
| 18:59.00 | [TK]D-Fender | kalel008: VoiceMail("SIP/1000-081e1250", "0112872130@default|b") <- this is not at all the dialplan line you showed us, nor does the box # look like the line you pasted from voicemail.conf either |
| 18:59.29 | kalel008 | i just showed you the 011 part |
| 18:59.34 | kaldemar | start with Hanup to warm up. |
| 19:00.05 | seanbright | kalel008: pastebin your extensions.conf and your voicemail.conf file |
| 19:00.20 | kalel008 | ok |
| 19:00.26 | [TK]D-Fender | kalel008: 011 => 1234,jhb1 <- doesn't match |
| 19:00.39 | *** part/#asterisk belendax (n=belendax@unaffiliated/belendax) |
| 19:00.46 | kalel008 | but the voicemail is still the origenal so ill just putthe context in |
| 19:00.52 | [TK]D-Fender | kalel008: [Sep 3 20:55:12] WARNING[3792]: app_voicemail.c:2862 leave_voicemail: No entry in voicemail config file for '0112872130' <- this should be a giant tip-off |
| 19:01.17 | [TK]D-Fender | kalel008: Whats that supposed to mean? |
| 19:03.12 | kalel008 | http://pastebin.com/m74168a95 |
| 19:03.35 | seanbright | kalel008: let me ask you a question.... |
| 19:03.50 | seanbright | are this "0112972130" and this "0112872130" the same thing to you? |
| 19:03.52 | [TK]D-Fender | kalel008: your entire voicemail.conf please. |
| 19:04.01 | kalel008 | ok |
| 19:04.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:04.15 | seanbright | kalel008: the entry in your voicemail.conf file has a 9 in it |
| 19:04.21 | seanbright | kalel008: the one in your extensions.conf has an 8 in it |
| 19:04.28 | [TK]D-Fender | kalel008: And you know, seanbright here has a valid thought... |
| 19:04.42 | EI5GTB | has * installing quite niceley...time to screw in the hard drives and put on the case |
| 19:05.10 | Assid | okay back..so anyone know any unlimited good quality business grade outgoing.. even with a channel limit like 4 channels or so |
| 19:07.06 | Corydon76-dig | A valid thought? <gasp> It's the APOCALYPSE! |
| 19:07.14 | seanbright | i know... i get no credit |
| 19:07.25 | [TK]D-Fender | I was wondering what that burning smell was actually ;) |
| 19:07.52 | kalel008 | oh my work |
| 19:08.08 | [TK]D-Fender | seanbright: I give you full credit for this one. |
| 19:08.24 | seanbright | super |
| 19:08.26 | kalel008 | i am sorry for putting you guys through all that just because 1 digit |
| 19:08.34 | seanbright | kalel008: we are too |
| 19:08.45 | [TK]D-Fender | nobody moreso... |
| 19:08.45 | kalel008 | thanks really |
| 19:08.47 | seanbright | kalel008: and it's Hangup (with a 'g') |
| 19:08.56 | [TK]D-Fender | seanbright: TOO GENEROUS!!! |
| 19:08.59 | seanbright | kalel008: you have Hanup in your extensions.conf |
| 19:09.08 | [TK]D-Fender | seanbright: I was gonna let that one slide! |
| 19:09.12 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 19:09.17 | seanbright | [TK]D-Fender: heh |
| 19:09.28 | kalel008 | ok let me ask a decent question |
| 19:09.33 | seanbright | oh no |
| 19:09.37 | seanbright | goes away |
| 19:09.38 | kalel008 | lol |
| 19:09.40 | [TK]D-Fender | kalel008: Why start now? |
| 19:09.48 | seanbright | [TK]D-Fender: well played |
| 19:11.28 | kalel008 | lol crap forgot what its called |
| 19:11.29 | *** join/#asterisk i3inary (n=shaman@cpe-76-88-94-149.san.res.rr.com) |
| 19:11.41 | kalel008 | you use it lik ethis |
| 19:12.15 | kalel008 | with the exclamation next to the context |
| 19:12.21 | kalel008 | um |
| 19:12.36 | seanbright | templates? |
| 19:12.47 | kalel008 | yeh |
| 19:13.15 | kalel008 | anyway can i only use templates on contexts |
| 19:13.28 | seanbright | i dunno |
| 19:13.45 | kalel008 | because i have 28 DID's doing the same thing |
| 19:14.14 | kalel008 | just answer and then going into voicemail |
| 19:15.00 | kalel008 | the DID to leave all the voicemails |
| 19:15.20 | kalel008 | and one DID to listen to the voicemails with a pin |
| 19:15.40 | Assid | anyone know a good outgoing provider with similar rates to rapidvox |
| 19:15.47 | kalel008 | that whay im asking about the templates |
| 19:16.15 | kalel008 | myself use callwithus and they are not bad |
| 19:16.53 | Assid | kalel008: quality wise? as good ? whats the pricing - US? |
| 19:17.13 | [TK]D-Fender | kalel008: templates only apply to device configs, not dialplan. |
| 19:17.39 | kalel008 | aah ok |
| 19:18.26 | kalel008 | well im in south africa and making calls on a adsl 512kbps line and there is about 2 sec latency on local calls |
| 19:18.55 | kalel008 | and the sp is in the US |
| 19:19.02 | kalel008 | good quality |
| 19:19.39 | Assid | hrmm |
| 19:19.44 | *** part/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 19:19.51 | kalel008 | fender so it is ok to have a huge extensions.conf file |
| 19:20.13 | Assid | kalel008: if you can use macros to cut them down.. you should |
| 19:20.32 | EI5GTB | wtf....why didnt asterisk congradulate me for installing it?? it just told me tomgo read security docs... I WANT TO BE CONGRADULATED |
| 19:20.53 | Assid | i had a crappy dialplan once upon a time.. [TK]D-Fender just told me i better clean it up back when he wasnt angry and green like the hulk; all the time |
| 19:20.54 | mahlon | kalel008: You can also logically organize it into different files with #include. |
| 19:20.54 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:21.17 | [TK]D-Fender | kalel008: For what you described there is no reason why your file should be any more than 30 lines bigger than it is now. |
| 19:21.22 | mahlon | But it's not like the size of the file slows down performance. It's pre-parsed into your dialplan. |
| 19:21.33 | seanbright | EI5GTB: congratulations... you can (barely) follow instructions |
| 19:21.36 | seanbright | ;) |
| 19:21.37 | [TK]D-Fender | kalel008: And in so doing, a whole lot cleaner |
| 19:21.37 | EI5GTB | hahaha] |
| 19:21.55 | [TK]D-Fender | EI5GTB: "congratulated" |
| 19:22.10 | [TK]D-Fender | pushes the "APPLAUSE" button |
| 19:22.14 | kalel008 | ill have a look at the macros just started that chapter 2day :) |
| 19:22.20 | EI5GTB | hahahaha |
| 19:22.29 | EI5GTB | my phone system has to have an applause button |
| 19:22.49 | [TK]D-Fender | EI5GTB: Feel free to add one. |
| 19:22.57 | Assid | EI5GTB: do people appluad you when you take a dump? |
| 19:22.58 | EI5GTB | heh, me wonders how.. |
| 19:23.04 | EI5GTB | Assid, depends.. |
| 19:23.58 | *** join/#asterisk mattx86_ (n=root@static20247.sftncc.ken-tennwireless.com) |
| 19:24.02 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 19:24.19 | *** join/#asterisk stoffell (n=stoffell@d51A4D28C.access.telenet.be) |
| 19:24.35 | kalel008 | thanks guys for ur help im off to bed now |
| 19:24.43 | kalel008 | chow |
| 19:26.36 | mattx86_ | any idea why I'm unable to do a hook-flash and get a dial tone? (Analog phone, into a Channel Bank, into Asterisk via a wcte11xp card) |
| 19:27.08 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 19:27.40 | [TK]D-Fender | mattx86_: Enable transfer, 3-way calling, etc? |
| 19:27.48 | mattx86_ | I'm using fxoks signalling on the FXS ports (zaptel.conf/zapata.conf) and I've tried enabling both transfer=yes and threewaycalling=yes in zapata.conf |
| 19:28.04 | [TK]D-Fender | mattx86_: maybe the CB doesn't support it? |
| 19:28.17 | mattx86_ | Channel Bank is configured for Loop Start signalling on the FXS ports |
| 19:28.39 | whymarkwh | hi all need to build a unit for 250 snom 300 and 2 snom 370 with 2 pri interfase and 8 fxo for mobile phone routers what kind of spec must i look at for the asterisk server? |
| 19:28.44 | mattx86_ | [TK]D-Fender: I thought this one did; it's an Adtran TA750 |
| 19:29.28 | [TK]D-Fender | mattx86_: apstebin your zapata.conf |
| 19:30.02 | mattx86_ | [TK]D-Fender: ok. one moment |
| 19:32.18 | stoffell | i wonder if its possible to set verbose to 3 (or higher) without always seeings the manager.conf parsing stuff and logins.. |
| 19:33.51 | [TK]D-Fender | stoffell: as-is, no. |
| 19:34.14 | [TK]D-Fender | IMO we should have a "manager debug" that is separate. |
| 19:34.19 | stoffell | ok [TK]D-Fender , thanks.. |
| 19:34.25 | rasterix | in the globals context in extensions.conf should i allow varname{whitespace}= or make it strict varname= |
| 19:34.32 | stoffell | yeah, same imo here.... |
| 19:34.58 | stoffell | i'll stick to searching the source files for now .... :) |
| 19:35.00 | rasterix | (in a an editor for extensions.conf) |
| 19:36.44 | mattx86_ | [TK]D-Fender: zapata.conf - http://pastebin.com/m30ca12c0 |
| 19:36.59 | [TK]D-Fender | rastno whichspace |
| 19:37.06 | [TK]D-Fender | rasterix: no whitespace |
| 19:37.22 | rasterix | thats what ive done because thats what u said last night fender |
| 19:37.39 | rasterix | its gonna syntax error a lot of peoples working dialplans though >< |
| 19:37.59 | [TK]D-Fender | mattx86_: FFS all that stuff up top belongs under [channels]! |
| 19:38.16 | rasterix | ahhh balls to them... why r they typing whitespace anyway |
| 19:38.25 | [TK]D-Fender | rasterix: Show me a problem and we'll fix a problem. |
| 19:39.15 | rasterix | fender: ur on my top list of people worth listening to (no flattery intended) so im going with your opinion |
| 19:39.16 | *** join/#asterisk klictel (n=klictel@207.61.107.242) |
| 19:39.21 | rasterix | NO WHITESPACE ON MY EDITOR |
| 19:39.23 | mattx86_ | [TK]D-Fender: heh, that's what I had at one point, but with all the debugging we'd done for various things, we kinda rewrote our zapata.conf to match the original that came with it (Fedora Core 8) |
| 19:39.51 | [TK]D-Fender | mattx86_: just move it all down under [channels] |
| 19:40.35 | mattx86_ | [TK]D-Fender: what about the progzone or anything else? does [default] even matter, really? |
| 19:40.57 | [TK]D-Fender | mattx86_: nope. Leave it, and leave it empty |
| 19:41.09 | rasterix | i kissed a girl... i liked it... this song keeps distracting me from my ide >< |
| 19:41.12 | [TK]D-Fender | mattx86_: progzone is a zaptel opt only IIRC |
| 19:41.36 | mattx86_ | [TK]D-Fender: ah. ok, I'll try it out |
| 19:46.10 | mattx86_ | [TK]D-Fender: ok, so I still have [default] at the top, but also have [channels] on the third line, with everything after it. |
| 19:47.15 | mattx86_ | [TK]D-Fender: I've disabled callwaiting(=no) and still have threewaycalling=yes and transfer=yes. I can hook-flash now and transfer, but if I hangup before the other line picks up, they both hangup |
| 19:48.11 | [TK]D-Fender | mattx86_: Not 100% on the fine points of Zaptel analog transfers, but the core is done. |
| 19:48.44 | [TK]D-Fender | mattx86_: I'll leve you to work the "how-to" bits from here. |
| 19:48.46 | [TK]D-Fender | leave* |
| 19:49.35 | mattx86_ | [TK]D-Fender: of course, I haven't tried the "real thing" (tm) just yet. all I've done is dial a test extensions that does Background(beep), Wait(3), Goto(xx,1) to repeat |
| 19:50.16 | mattx86_ | [TK]D-Fender: that's cool. I appreciate the help. |
| 19:50.32 | *** join/#asterisk jtodd (i=_squid@c-76-27-193-118.hsd1.or.comcast.net) |
| 19:50.46 | stoffell | oaky, got rid of the logins of manager.. now still gotta get rid of the parsing manager.conf: found, etc... |
| 19:50.50 | *** join/#asterisk Levonk (n=lk@adsl-76-238-250-23.dsl.lsan03.sbcglobal.net) |
| 19:51.26 | [TK]D-Fender | stoffell: Source mod? |
| 19:51.45 | *** join/#asterisk darkfires (n=lwhite@72.38.114.246) |
| 19:51.50 | stoffell | yes, but the parsing is a bit harder to find :p |
| 19:52.30 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 19:52.45 | darkfires | im having problems with my TDM400P...it was working fine, upgraded zaptel ...now all incoming calls have no caller id.. and when I call and asterisk rings...it sounds all garbled....and I can't dialout |
| 19:53.17 | *** join/#asterisk Greek-Boy (n=email@41.222.89.77) |
| 19:53.17 | darkfires | i put zaptel back to the old version..and its still doing it |
| 19:54.44 | Blackvel | tried to find out what port=5062 means in sip.conf.sample [patton] type=peer section. can you help me? |
| 19:56.44 | mattx86_ | works great! thanks, [TK]D-Fender |
| 19:56.58 | [TK]D-Fender | mattx86_: You're welcome. |
| 19:57.22 | [TK]D-Fender | Blackvel: thats the port # * will send calls to on it |
| 19:58.21 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:00.44 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 20:02.01 | stoffell | ok,modified to stop whining about those things except when verbosity > 13 ... too bad it's not in variables / config yet :( |
| 20:02.49 | *** part/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 20:04.02 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-114-16-171.satx.res.rr.com) |
| 20:04.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:05.30 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 20:10.14 | *** join/#asterisk angryuser (n=angryuse@121.215.20.81.dynamic.adsl.abo.nordnet.fr) |
| 20:10.56 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-247-111-195.dsl.pltn13.sbcglobal.net) |
| 20:13.12 | outtolunc | borrows manx's bat for a few |
| 20:14.36 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:16.46 | *** join/#asterisk pdc_pdb (n=ccbonill@201.228.24.214) |
| 20:17.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 20:17.29 | pdc_pdb | Hello...can any one help me with phpagi ... ??? |
| 20:19.25 | mvanbaak | ~question |
| 20:19.26 | jbot | hmm... question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 20:24.16 | [TK]D-Fender | CHECKOUT TIME, HEADING HOME. |
| 20:24.23 | pdc_pdb | look, I downloaded the phpagi.php classes, put the files in the correct path, configure de ext to execute some agitest.php and play a sound, but the script doesn't seem to be doing nothing, in the asterisok cli i see "AGI Script newAgi/agiTest.php completed, returning 0" |
| 20:24.38 | pdc_pdb | what can be going wrong here...??? |
| 20:27.26 | flujan | hello guys, i am using asterisk to dial out using .call files and the AMI |
| 20:27.39 | flujan | the cdr is not loggin the failed calls that i make here it goes http://pastie.org/265514 |
| 20:27.50 | *** join/#asterisk envisean (n=envisean@166.129.141.63) |
| 20:28.05 | flujan | any clue about it? I just dunnot why it is not recording the CDR of these calls. |
| 20:30.54 | *** join/#asterisk safa12 (n=safa@92-238-52-92.cable.ubr16.croy.blueyonder.co.uk) |
| 20:32.05 | safa12 | need help /dev/ttyS0 set up with a serial cable so a call centre system can |
| 20:32.06 | safa12 | pick up CallerID. How can I redirect the log output of asterisk to |
| 20:33.23 | codefreeze-lap | flujan: looks like it's it's having trouble with Answer at prio 1 in that exten. It looks like Answer is returning -1, so the pbx aborts at that point. |
| 20:34.31 | flujan | codefreeze-lap: I put the answer command because corydon-dig tell me to do that... but it is also nos working with or without the Answer command. |
| 20:34.35 | flujan | it is not recording at all. |
| 20:34.42 | flujan | nothing is recorded on the CDR |
| 20:36.31 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 20:36.31 | codefreeze-lap | Well, it is a single-channel CDR, and maybe we have to refine the criteria where they are sent off. There is an option in cdr.conf, unanswered=yes, that you could insert, but by now it could be broken. |
| 20:36.31 | r0land | hi all |
| 20:36.59 | r0land | im having trouble calling one of my sip peers.. http://pastebin.com/m1ffc62c9 any advice about wht would b the matter? |
| 20:38.19 | stoffell | r0land, are they both registered? |
| 20:38.31 | r0land | stoffell just 179 |
| 20:38.43 | r0land | 334 is just there to recieve incoming calls |
| 20:38.49 | safa12 | need help how to use syslog ? for ttys0 |
| 20:39.16 | r0land | stoffell check my pastebin! |
| 20:39.38 | stoffell | r0land, i was doing that |
| 20:40.04 | klictel | he still needs to register |
| 20:40.06 | r0land | stoffell thanks :) |
| 20:40.17 | stoffell | r0land, try "sip show peers" |
| 20:40.18 | flujan | codefreeze-lap: will try it |
| 20:40.27 | r0land | stoffell just did :) and doing pastebin |
| 20:43.00 | safa12 | need help /dev/ttyS0 set up with a serial cable so a call centre system can pick up CallerID. How can I redirect the log output of asterisk to ttys0 |
| 20:43.35 | r0land | stoffell http://pastebin.com/d5b3d3df1 |
| 20:44.29 | klictel | 334 is not registerer |
| 20:44.32 | klictel | registered |
| 20:44.35 | r0land | i know |
| 20:44.38 | r0land | i allready told u its not |
| 20:44.49 | r0land | <r0land> 334 is just there to recieve incoming calls |
| 20:45.10 | r0land | i never had it registered for the past year (working perfectly) |
| 20:45.11 | klictel | he still needs to register... Asterisk needs to know its IP |
| 20:45.22 | r0land | untill today where i deicded to clean up my dialplan and added/split contexts |
| 20:45.25 | stoffell | r0land, it needs to be registered.. |
| 20:45.45 | r0land | stoffell well if its not on a physical device nor a osftphone! how am i going to register it |
| 20:45.48 | Blackvel | <ManxPower> "patton" is a SIP gateway and virtually all SIP gateways do not support user/pass auth for outgoing calls to the server. |
| 20:45.49 | stoffell | it really does.. asterisk does not know how to find it .... chan_sip: cause 3 - No route to destination |
| 20:46.07 | stoffell | r0land, sorry, can't help you then.. :( |
| 20:46.07 | klictel | if you put host=dynamic in sip.conf then it has to register |
| 20:46.30 | r0land | stoffell it is |
| 20:46.40 | *** join/#asterisk bkw_ (n=bkw_@freeswitch/developer/bkw) |
| 20:46.50 | r0land | sorry i mean klictel i did.. but the thing is! tht peer.. isnt to b set anywhere! |
| 20:47.09 | jaytee | r0land, if it's not an actual physical device then it won't use a SIP channel. The problem is not registration, it's in your contexts that you mucked with. |
| 20:47.13 | r0land | klictel its just to b used for my incoming call to b directed to its context |
| 20:47.35 | r0land | jaytee ya i agree.. which is why im here |
| 20:47.41 | Blackvel | how can asterisk map two peers on same ip when the gateway would not support user/pass auth for outgoing calls (incoming calls to asterisk)? |
| 20:47.59 | Blackvel | I can see nothing in sip debug |
| 20:48.07 | jaytee | r0land, so are these incoming calls from an ITSP? |
| 20:48.21 | r0land | jaytee from sipura |
| 20:48.46 | r0land | pstn ->sipura-> peer 334-> context-incoming |
| 20:48.53 | jaytee | sipura? that tells me a harware brand, not channel info. Are the calls coming in SIP, FXO, PRI? |
| 20:49.01 | r0land | sip |
| 20:49.02 | jaytee | ahhhh, |
| 20:49.25 | Corydon76-dig | safa12: if you want professional help, I can refer you to several places that would be happy to take your money |
| 20:49.29 | jaytee | so yes, the sipura ATA needs to register to * as a sip peer |
| 20:49.39 | r0land | jaytee it is |
| 20:49.44 | Corydon76-dig | safa12: but please do no message me. |
| 20:50.17 | jaytee | and in sip.conf will have a context= that you set to context-incoming for that SIP account |
| 20:50.31 | r0land | jaytee theres extension 299 (pstn line) and then in dial plan theres extension 334.. so when an incoming call gets to 299.. sipura directs it to 334, where ive set asterisk wht to do when he gets an incoming call on 334 |
| 20:50.37 | safa12 | ok any help |
| 20:51.12 | flujan | codefreeze-lap: Corydon76-dig with the unaswered=yes it is logging in the cdr table... it is sending the disposition NO ANSWER. Since i have available trunks and outgoing lines how can I debug and discover why i am receiving this from asterisk? |
| 20:51.24 | jaytee | r0land, pastebin you sip.conf masking passwords and IPs and pastebin your extensions.conf |
| 20:51.25 | stoffell | safa12, look on google.. something like "redirect text to serial port" .. |
| 20:51.42 | *** join/#asterisk tinkerghost (n=eric@thelocalphonecompany.net) |
| 20:51.45 | safa12 | thnx stoffell |
| 20:51.51 | r0land | jaytee http://pastebin.com/m1ffc62c9 |
| 20:52.02 | stoffell | safa12, you should be able to "pipe" almost any text to ttyS0 .. |
| 20:52.05 | klictel | but it all ends up in a SIP dial to peer 334 |
| 20:52.17 | r0land | klictel true |
| 20:52.30 | *** join/#asterisk jarod14 (n=jarod14@ns1.viatelecom.com) |
| 20:52.41 | r0land | klictel it worked perfectly b4 i tried cleaning up my contexts. since i had everything in "spa" context |
| 20:53.00 | safa12 | but how stoffels |
| 20:53.12 | codefreeze-lap | flujan: as far as I know, the possible dispositions are NO ANSWER, ANSWERED, BUSY, FAILED, and NULL |
| 20:53.36 | tinkerghost | I have an odd problem. I am logged into my asterisk server with both twinkle (SIP local) and KIAX (IAX from a remote server). The SIP channel works fine. The IAX channel has an echo. I saw 1 thread in a newsgroup, with no resolution so anyone have some suggestions? |
| 20:53.49 | stoffell | safa12, i have no idea, but i think it's possible with redirection to /dev/ttyS0, should be able to find that in some "llinux serial port" howto on google |
| 20:54.13 | codefreeze-lap | flujan: and NULL shows up as NOANSWER |
| 20:55.24 | flujan | codefreeze-lap: ok thanks... but I want do discover why asterisk is failing putting out this calls since the phone number is ok and i have available outgoing lines... :( |
| 20:55.51 | r0land | jaytee any advice?! |
| 20:56.28 | jaytee | r0land, yes, on line 38, include = coming? shouldn't that be incoming? |
| 20:56.32 | codefreeze-lap | flujan: you can't depend on CDR's for that. Turn on debug and turn up verbosity, and see if console messages can help. (core set verbose 20) (core set debug 10) |
| 20:57.26 | r0land | jaytee :| |
| 20:57.26 | jaytee | what? |
| 20:57.26 | r0land | jaytee if tht was the solution am gonna go nuts.. |
| 20:57.26 | r0land | thanks jaytee lemme try |
| 20:57.42 | r0land | jaytee nop.. same prob! i fixed it .. and reloaded |
| 20:57.49 | *** join/#asterisk dlynes (n=daniel@d206-116-189-12.bchsia.telus.net) |
| 20:57.52 | tinkerghost | don't feel bad, I just fixed a 3 month problem from using riId instead of riID |
| 20:57.59 | jaytee | r0land, dialplan reload? |
| 20:58.06 | r0land | jaytee just "reload" |
| 20:58.11 | jaytee | ok |
| 20:58.39 | tinkerghost | jaytee, 'dialplan reload' is the same as 'extensions reload' for the newer builds |
| 20:58.52 | r0land | tinkerghost hehe ya happened with me as well |
| 20:59.24 | klictel | coming or incoming will not make a difference since it is picking up line 34 that does the Dial(SIP/) |
| 20:59.42 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:59.52 | klictel | if it was not able to resolve the xetension then it would look into the included contexts |
| 21:00.20 | r0land | good point |
| 21:00.20 | klictel | but it matched on line 34 |
| 21:00.54 | klictel | Asterisk is not able to find the peer 334 |
| 21:00.54 | r0land | but klictel it should match on line 53 since thts its context |
| 21:01.24 | jaytee | r0land, this scenario isn't making sense to me. What is the model # of your Sipura? |
| 21:01.32 | r0land | jaytee 3102 |
| 21:01.39 | *** join/#asterisk Levonk (n=lk@adsl-76-237-14-123.dsl.lsan03.sbcglobal.net) |
| 21:02.29 | jaytee | so you call into the line on the FXO port and it is supposed to route the call to 334, right? |
| 21:02.46 | r0land | jaytee i registered my PSTN on sipura with 299, so when i dial out from it, i dial 299 and get a dial tone to call out.. but when an incoming call occurs it goes to 334 |
| 21:03.07 | klictel | did you make any modif to sip.conf and not reloaded your sip.so? |
| 21:03.26 | r0land | klictel nop, i always do "reload" which is enough to reload both sip and anythig relevant |
| 21:03.27 | jaytee | ok, then you try and turn the call around and dial 334 which makes no sense since it's not a phone, it's a sip ATA |
| 21:03.46 | stoffell | i still don't get it .. why route to 334 if 334 does not exist ... |
| 21:04.00 | r0land | stoffell so the attendant answers |
| 21:04.14 | stoffell | r0land, and what extension or application is the attendant ? |
| 21:04.17 | r0land | stoffell if i set the route to the same peer (299) it will always show as busy, since the call is coming from IT! |
| 21:04.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:04.44 | r0land | stoffell so to go around tht, i made 334 with context incoming, tht will answer the call for me and starts the attendant audio msg |
| 21:05.00 | stoffell | seems fair, but you don't need a sip peer 334 then... |
| 21:05.08 | r0land | stoffell wht then ?! |
| 21:05.10 | r0land | any advice? |
| 21:05.30 | stoffell | well, let your dialplan match 334,x,blabla and let it go that audio message.. |
| 21:05.45 | r0land | stoffell in the same context as spa?! |
| 21:06.30 | stoffell | or in a different, doesn't matter, .. as long as its reachable from the context you arrive on |
| 21:06.43 | r0land | well thts wht i did! |
| 21:06.51 | r0land | and ur telling me tht it wont work cos 334 isnt registered |
| 21:07.56 | nhuisman_work | would _X work as a pattern for any number? |
| 21:07.56 | stoffell | sip/334 exists, so it's probably bothering you somewhere in the dialplan |
| 21:08.00 | nhuisman_work | outgoig |
| 21:08.02 | nhuisman_work | outgoing |
| 21:08.18 | nhuisman_work | !book |
| 21:08.29 | r0land | stoffell hmm |
| 21:08.35 | r0land | stoffell could u elaborate |
| 21:08.40 | [TK]D-Fender | nhuisman_work: No. |
| 21:08.48 | jaytee | r0land, pastebin your "sip show peers" output |
| 21:09.07 | klictel | Dial always needs to know the ip of the peer |
| 21:09.47 | r0land | jaytee http://pastebin.com/m1ffc62c9 (check first 2 lines) |
| 21:11.17 | jaytee | r0land, that's not what i asked for |
| 21:12.20 | r0land | jaytee http://pastebin.com/d5b3d3df1 <<-- sorry pasted the wrong link |
| 21:12.36 | klictel | (Unspecified) |
| 21:12.53 | r0land | klictel i totally understand your point |
| 21:12.55 | r0land | i honestly do |
| 21:13.06 | klictel | :) |
| 21:13.07 | r0land | but the thing is, it always worked before, untill i started cleaning up my dialplan |
| 21:13.13 | [TK]D-Fender | r0land: Your 333 334 335 336 extens in [incoming are broken, you split up "s" within that context , and things are still a mess. |
| 21:13.16 | r0land | and it never registered |
| 21:13.31 | jaytee | r0land, did you happen to make something called a "backup"? |
| 21:13.41 | r0land | jaytee of course:) |
| 21:13.47 | jets | can i run a newer zaptel with an older stable asterisk release, say 1.4.18? |
| 21:13.48 | [TK]D-Fender | r0land: and SIP/334 not being registered has NOTHING to dow ith your dialplan. |
| 21:13.56 | jaytee | well, then revert to the backup and test again to verify |
| 21:14.03 | r0land | [TK]D-Fender k il get back to it again then :) |
| 21:14.08 | r0land | jaytee i did .. and it worked:) |
| 21:14.10 | [TK]D-Fender | r0land: NO. |
| 21:14.25 | [TK]D-Fender | r0land: and SIP/334 not being registered has NOTHING to dow ith your dialplan. <-------- |
| 21:14.26 | r0land | [TK]D-Fender i know it has nothing to do with registering thts wht im trying to explain to klictel |
| 21:14.44 | r0land | its a matter of context |
| 21:14.51 | [TK]D-Fender | r0land: Yes, and the fact it isn't registered is the immediate cause of your Dial failing. |
| 21:15.08 | [TK]D-Fender | r0land: No, it is not a matter of "context". |
| 21:15.09 | jaytee | no, it doesn't but you're dialplan is fubared. You're calling a device that doesn't exist. |
| 21:15.19 | r0land | [TK]D-Fender you saw my previous dial plans.. it never registered though it always worked |
| 21:15.30 | [TK]D-Fender | jaytee: Oh, it "exists", * just has no idea where to call it |
| 21:15.55 | klictel | heh |
| 21:16.05 | r0land | jaytee i understand, which is why im here! im trying to find a way to get sipura to send the incoming call to "somewhere" so it executes the "answer" mechanism |
| 21:16.21 | r0land | jaytee if u have an advice on how thts possible please share it with me.. |
| 21:16.29 | jaytee | I've never dealt with the SPA3102, just the 2102 but that's just 2 FXS ports and a router. |
| 21:16.56 | r0land | jaytee ya have one of those as well.. it works like a charm :) |
| 21:17.13 | nhuisman_work | [TK]D-Fender, ah it would need to be _X! |
| 21:17.15 | r0land | jaytee how did u manage to setup ur incoming calls |
| 21:17.33 | [TK]D-Fender | nhuisman_work: a dangerous pattern.... |
| 21:17.40 | jaytee | r0land, my incoming calls come in via PRI not analog |
| 21:17.47 | r0land | oh ok |
| 21:17.56 | [TK]D-Fender | jaytee: SPA sends calls to a targeted extension. |
| 21:18.01 | nhuisman_work | [TK]D-Fender, is the danger that they could call anything they want? |
| 21:18.03 | [TK]D-Fender | jaytee: the SPA-3102 is a great little box |
| 21:18.17 | nhuisman_work | [TK]D-Fender, or is there another risk in that pattern |
| 21:18.20 | jaytee | [TK]D-Fender, I've ordered one to use with the Valcom PA systems. |
| 21:18.26 | jaytee | but it hasn't gotten here yet. |
| 21:18.33 | r0land | jaytee its worth every penny.. |
| 21:18.40 | [TK]D-Fender | nhuisman_work: use proper patterns for things otherwise stupid overlap issues may occur. Invite trouble and it might just pay you a visit |
| 21:18.48 | r0land | i always used it with my softphone/handy but never with asterisk untill recently |
| 21:19.04 | [TK]D-Fender | jaytee: Valcom does the Viking ones? Or is a a competitor? |
| 21:19.10 | [TK]D-Fender | it* |
| 21:19.37 | nhuisman_work | [TK]D-Fender, yeah currently I have a bunch of outgoing rules for 7, 10, and 11 digit numbers, just looking to see how to simplify it. |
| 21:19.39 | jaytee | Valcom is a competitor and cheaper. |
| 21:19.47 | jets | hrm |
| 21:19.49 | *** join/#asterisk tkbeat (n=tk@p54B97723.dip.t-dialin.net) |
| 21:19.54 | nhuisman_work | and then even more rules for long distance |
| 21:20.02 | [TK]D-Fender | jaytee: I have heard the name before... never personally used. Specs look good? |
| 21:20.25 | [TK]D-Fender | nhuisman_work: leave your 7-10-11, thats the way it should be. |
| 21:20.33 | nhuisman_work | [TK]D-Fender, k |
| 21:20.36 | jaytee | [TK]D-Fender, I looked at a Viking too and a company that makes SIP PA devices but they were much more expensive. |
| 21:21.09 | [TK]D-Fender | nhuisman_work: add : _011XX. | _[3469]11 |
| 21:21.20 | jaytee | [TK]D-Fender, I hooked it up today and tested it. Audio from any phone I dial from sounds great, even Pukestream Buggertone-200's |
| 21:22.22 | jaytee | figure with a spa3102 for one zone PA and 1 fxs port to analog phone it only adds about 30 bucks to the cost for each PA zone and we're gonna do about 10 to 12 zones to cover the whole zoo. |
| 21:25.48 | *** join/#asterisk Levonk (n=lk@adsl-76-243-66-168.dsl.lsan03.sbcglobal.net) |
| 21:28.06 | *** join/#asterisk envisean (n=envisean@65.44.128.2.static.brand-up.com) |
| 21:29.14 | nhuisman_work | [TK]D-Fender, what needs 3, 4, 6, or 9 before the 11 ? I'm not too familiar with international stuff |
| 21:29.40 | [TK]D-Fender | nhuisman_work: Do I seriously have to explain the value of 911 to you? :) |
| 21:29.58 | nhuisman_work | [TK]D-Fender, oh heh, I have a separate entry for that :P |
| 21:30.05 | nhuisman_work | thought that was for some weird internation thing |
| 21:30.45 | nhuisman_work | i get it now |
| 21:30.45 | [TK]D-Fender | nhuisman_work: I gave you an all-in-one entry for it. |
| 21:30.45 | [TK]D-Fender | nhuisman_work: Just do the math next time :p |
| 21:30.54 | jaytee | be back later |
| 21:31.39 | *** join/#asterisk ta^3 (n=tacvbo@189.146.183.121) |
| 21:32.29 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:34.20 | safa12 | need help /dev/ttyS0 set up with a serial cable so a call centre system can pick up CallerID. How can I redirect the log output of asterisk to ttys0 <anybody> |
| 21:35.32 | r0land | [TK]D-Fender since u know tht in spa3102 i should direct the call to a specific extension, and since i want an attendant to answer instead of directly ringing on a specific phone, any advice on how to register a peer tht im not gonna use, just for the sake of my incoming context works ?! |
| 21:35.51 | [TK]D-Fender | safa12: You wouldn't do this with log output, your call a script from your dialplan to do it |
| 21:36.45 | [TK]D-Fender | r0land: the SPA should be registering to *. |
| 21:37.35 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 x86_64 compile troubles:type ~centos52bug -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 21:37.38 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:37.39 | r0land | [TK]D-Fender it is, PSTN (fxo) is regsited but the incoming call gets directed to "something" im trying to figure out wht tht "Something" is since my way didnt work |
| 21:38.09 | [TK]D-Fender | r0land: What do you see coming in? |
| 21:38.36 | r0land | 299 to 334 (pstn peer to unregisteed peer) |
| 21:38.44 | r0land | [TK]D-Fender lemme pastebin |
| 21:39.22 | r0land | [TK]D-Fender http://pastebin.com/m1ffc62c9 |
| 21:39.32 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 x86_64 compile troubles:type ~centos52bug -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 21:39.38 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 x86_64 compile troubles:type ~centos52bug -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 21:39.51 | Qwell | something like that |
| 21:40.18 | [TK]D-Fender | Qwell: How did we go from Beta 9 to RC4? |
| 21:40.24 | Qwell | umm |
| 21:40.32 | Qwell | we skipped some |
| 21:41.16 | safa12 | <[TK]D-Fender> safa12: You wouldn't do this with log output, your call a script from your dialplan to do it...i'v been struggling with it from ages been to lots of site can't find anything,,,Any help |
| 21:41.36 | [TK]D-Fender | safa12: pastebin what you've done. |
| 21:42.41 | [TK]D-Fender | ~pb |
| 21:42.45 | jbot | [~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 |
| 21:42.46 | [TK]D-Fender | ^^^^ |
| 21:43.00 | safa12 | i tried extensions.conf and put System(echo ${CALLERIDNUM} > /dev/ttyS0) |
| 21:43.00 | safa12 | didn't work |
| 21:43.32 | [TK]D-Fender | safa12: and show us what it EXECUTED |
| 21:43.40 | r0land | <[TK]D-Fender> r0land: What do you see coming in? <<--- saw wht am talking about ? |
| 21:43.53 | [TK]D-Fender | r0land: no, you aren't showing anything |
| 21:44.12 | safa12 | nothing |
| 21:44.13 | r0land | [TK]D-Fender thts the output on CLI when a call is directed to 334 |
| 21:44.20 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), DAHDI-tools 2.0.0-rc2 (2008/09/03), DAHDI-linux 2.0.0-rc3 (2008/09/03), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 21:44.35 | safa12 | if i go to com port to check connect nothing is coming after doing that |
| 21:44.36 | Qwell | that's getting a bit out of hand.. |
| 21:44.40 | russellb | :) |
| 21:44.45 | [TK]D-Fender | safa12: If it you see nothing its sure looking like your dialplan is bad and NOTHING is getting executed |
| 21:44.55 | tzafrir_laptop | hi, a slight reshuffeling of the dahdi init script: http://tzafrir.org.il/~tzafrir/dahdi.init http://tzafrir.org.il/~tzafrir/dahdi.init.diff |
| 21:44.56 | Qwell | russellb: we're going to need to come up with a new format for that or something, heh |
| 21:45.11 | russellb | removing zaptel will help a bit |
| 21:45.18 | russellb | and perhaps only having dahdi-linux-complete |
| 21:45.29 | Qwell | lemme try that |
| 21:45.46 | russellb | it won't help much ... but it helps a bit |
| 21:45.52 | [TK]D-Fender | russellb: Zaptel is still relevant for the time being no? |
| 21:46.00 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc3+2.0.0-rc2 (2008/09/03), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 21:46.02 | r0land | [TK]D-Fender http://pastebin.com/d43eeb131 <<-- cli |
| 21:46.21 | Qwell | bit shorter |
| 21:46.22 | safa12 | <[TK]D-Fender> safa12: If it you see nothing its sure looking like your dialplan is bad and NOTHING is getting executed< is there any way to rectify it> cheers |
| 21:46.23 | *** join/#asterisk irieKEN (n=irieken@130.191.205.128) |
| 21:46.31 | Qwell | [TK]D-Fender: yeah, for now |
| 21:46.49 | [TK]D-Fender | safa12: pastebin your dialplan and the related channel config |
| 21:46.50 | Qwell | That is the final Zaptel release though |
| 21:47.11 | safa12 | can i send you pm |
| 21:47.25 | [TK]D-Fender | safa12: No. PASTEBIN <----- |
| 21:47.28 | irieKEN | [TK]D-Fender: G'day:) |
| 21:48.01 | [TK]D-Fender | r0land: what is 300 & what is 333? |
| 21:48.31 | rasterix | fender: ok so what i have done is highlight any unneeded white space with the red squiggly underline... they can turn this on / off |
| 21:48.46 | r0land | [TK]D-Fender 300 = pstn registered peer on sipura 3102, and 333 is an unregistered peer tht has "incoming" as a context so the attendant answers |
| 21:49.05 | [TK]D-Fender | r0land: what "attendant"? |
| 21:49.32 | r0land | [TK]D-Fender recorded message asking to enter extension |
| 21:49.56 | [TK]D-Fender | r0land: recorded message coming from where? |
| 21:50.24 | irieKEN | Is the digium site slow for anyone else? |
| 21:50.31 | r0land | [TK]D-Fender converted it to gsm, saved it in /usr/share/asterisk/sounds/ |
| 21:50.51 | [TK]D-Fender | r0land: you are dialing a stupid FXS port. this has NOTHING to do with an "auto-attendant" |
| 21:51.04 | safa12 | <[TK]D-Fender> how do i copy dial plan any help |
| 21:51.05 | [TK]D-Fender | r0land: Thats like trying to fix a flat tire by MAKING A SANDWICH |
| 21:51.23 | r0land | [TK]D-Fender fxs prt has nothing to do with it.. |
| 21:51.24 | [TK]D-Fender | safa12: If you can't manage to cut & paste to a website, I can't help you |
| 21:51.25 | rasterix | < uses low fat spread on his sandwich :) |
| 21:51.32 | [TK]D-Fender | r0land: then why are you calling it? |
| 21:51.52 | safa12 | <[TK]D-Fender> what file do i need to copy and paste . extensions.conf or ??? |
| 21:51.55 | r0land | safa12 either use winscp and copy ur extensions.conf to ur destop :) or simply highlight and paste somewhere else |
| 21:52.16 | [TK]D-Fender | safa12: extensions.conf & the device config for whatever is sending the call to * in the first place. |
| 21:52.20 | r0land | safa12 ill go with winscp if i were u.. it helps keeps logs one way or the other:) |
| 21:52.30 | Qwell | irieKEN: yes |
| 21:52.41 | Qwell | irieKEN: I assume you mean the downloads site? |
| 21:53.36 | irieKEN | Qwell: I was actually referring to the whole domain (digium.com):) I am trying to see if anyone responded to a comment I posted on the forum, but I can't get in. |
| 21:55.29 | r0land | [TK]D-Fender http://pastebin.com/da2fbaeb |
| 21:56.30 | [TK]D-Fender | r0land: Why are you sending your incoming call to an extension that just dial another SIP device? |
| 21:56.58 | r0land | [TK]D-Fender because i cant set such a dial plan in sipura itself |
| 21:57.12 | r0land | i need the help of asterisk ... and in order to do so! i have to point sipura "somewhere" to go |
| 21:57.46 | [TK]D-Fender | r0land: -- Executing [333@spa:1] Dial("SIP/300-082b8c88", "SIP/333|20") in new stack <--- its already going here. WHY? |
| 21:58.12 | r0land | 300=pstn goes to 333 (so far thts just sipura doing its thing) |
| 21:58.33 | r0land | when it reaches 333, it goes to incoming context.. and answers the call where the auto attendant starts talking |
| 21:58.38 | [TK]D-Fender | r0land: And WHY is it going to 333, and why are you telling it to do THAT? |
| 21:58.47 | r0land | 333= auto attendant |
| 21:59.04 | [TK]D-Fender | r0land: Itr0lNo, 333 leads to that stupid dial! |
| 21:59.22 | r0land | 333 leads to context incoming! |
| 21:59.30 | [TK]D-Fender | r0land: -- Executing [333@spa:1] Dial("SIP/300-082b8c88", "SIP/333|20") in new stack <--- this is not a %^@#$ing autoattendant! What are you calling another SIP DEVICE? |
| 21:59.49 | r0land | and in incoming theres " exten => s,1,Answer() ; Answer inbound calls" |
| 22:00.01 | [TK]D-Fender | r0land: You're calling DIAL! DIAL is not an auto-attendant! What the hell don't you understand? |
| 22:00.10 | [TK]D-Fender | r0land: You're not IN INCOMCING! |
| 22:00.40 | r0land | ok u lost me there.. if 333 has context as "incoming" why wouldnt go there! |
| 22:00.42 | safa12 | <[TK]D-Fender> http://pastebin.com/d3516f169 |
| 22:01.25 | [TK]D-Fender | r0land: http://pastebin.com/da2fbaeb <- 300 points to [spa]. |
| 22:01.53 | [TK]D-Fender | safa12: ... |
| 22:01.56 | [TK]D-Fender | ~freepbx |
| 22:01.57 | jbot | [~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 |
| 22:01.58 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 22:02.17 | r0land | [TK]D-Fender 300 is the pstn line! |
| 22:02.47 | r0land | if i set 300 as the destination for incoming calls, theres going to b a loop (incoming line getting directed to itself) the line would drop! |
| 22:02.50 | [TK]D-Fender | r0land: http://pastebin.com/me132e0e |
| 22:03.34 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 22:03.49 | safa12 | <[TK]D-Fender> any help |
| 22:04.01 | iratik | What does raw ulaw data look like ? |
| 22:04.03 | [TK]D-Fender | safa12: fREEpbx IS NOT SUPPORTED IN HERE. |
| 22:04.38 | iratik | Yeah , freepbx is ... crap |
| 22:05.15 | mvanbaak | all gui's are |
| 22:05.31 | iratik | i like ragui though |
| 22:05.41 | mvanbaak | I prefer vim |
| 22:05.54 | iratik | ragui has a very 'literal' representation of dialplans |
| 22:05.57 | r0land | <[TK]D-Fender> r0land: http://pastebin.com/me132e0e <<== if i set 300 as the destination for incoming calls, theres going to b a loop (incoming line getting directed to itself) the line would drop! |
| 22:06.05 | iratik | its like a direct gui for extensions.conf .... http://www.ragui.net/call-route-outbound.jpg |
| 22:06.40 | [TK]D-Fender | r0land: SIP/300 should not be dialing "300" in the first place. |
| 22:06.49 | [TK]D-Fender | r0land: SIP/300 should not be dialing "333" in the first place. |
| 22:06.53 | mvanbaak | iratik: ah, but it's ruby on rails |
| 22:07.07 | [TK]D-Fender | r0land: 333 just dials another SIP device. This is not what you want. |
| 22:07.10 | iratik | mvanbaak: so... its actually built on realtime asterisk |
| 22:07.25 | mvanbaak | yeah |
| 22:07.46 | iratik | anyone ever heard of orecx/oreka ? |
| 22:07.54 | mvanbaak | I prefer stuff that generates conf files |
| 22:08.11 | mvanbaak | dont want my pbx to depend on sql |
| 22:08.12 | r0land | 333 only dials another sip account IF the caller didnt enter an extension after 8 seconds! did u check the incoming context? http://pastebin.com/da2fbaeb |
| 22:08.13 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:08.13 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:08.16 | iratik | mvanbaak: For nitty gritty stuff I do like adhearsion though... |
| 22:08.33 | iratik | I made a huge interactive IVR with that once, with database integration and everything |
| 22:08.50 | [TK]D-Fender | r0land: Nowhere in your CLI output was the incoming context being USED <- |
| 22:09.32 | r0land | [TK]D-Fender once again, it gets me to the question i asked before which is: "sipura should direct incoming calls somewhere, where to! " |
| 22:09.48 | iratik | [TK]D-Fender: who are you ? You are like some sort of asterisk god ... you've been in here consistently for ... almost a year now .... you're patience is incredible |
| 22:10.01 | Qwell | iratik: well over a year.. |
| 22:10.09 | [TK]D-Fender | r0land: You picked a ^#%$ing extension that dials another phone. What kind of decision is that? YOU aren't sending it to your own damn IVR. |
| 22:10.35 | [TK]D-Fender | r0land: Your IVR is under "s" and you're not sending it THERE <- |
| 22:11.02 | r0land | k :) got it |
| 22:11.41 | iratik | anyone ever heard of oreka/orecx ? |
| 22:11.50 | xuser_ | you? |
| 22:13.24 | *** join/#asterisk haryv (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:14.21 | *** join/#asterisk irieKEN (n=irieken@130.191.205.128) |
| 22:17.19 | *** join/#asterisk EI5GTB-macbook (n=EI5GTB@78.16.11.162) |
| 22:17.59 | EI5GTB-macbook | whats the cheapest way to get 4 fxo ports? |
| 22:18.11 | *** join/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 22:18.24 | [TK]D-Fender | EI5GTB-macbook: have someone give you free hardware |
| 22:18.34 | EI5GTB-macbook | aha |
| 22:18.42 | EI5GTB-macbook | will you give me free hardware? |
| 22:18.49 | [TK]D-Fender | EI5GTB-macbook: NO |
| 22:18.55 | EI5GTB-macbook | dam |
| 22:18.57 | [TK]D-Fender | ~cheap |
| 22:18.58 | jbot | methinks cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 22:18.58 | Blackvel | how do you guys match 2 peers (friend or peer type no matter) when the ip is the same for both? does it work with username and registered peers? |
| 22:19.09 | EI5GTB-macbook | lul |
| 22:19.28 | EI5GTB-macbook | but seriously.. get one of those cisco boxes |
| 22:19.32 | EI5GTB-macbook | or use a pci card? |
| 22:19.48 | r0land | [TK]D-Fender works :) thanks |
| 22:20.11 | r0land | <[TK]D-Fender> r0land: Your IVR is under "s" and you're not sending it THERE <- did it for me.. :) |
| 22:23.52 | [TK]D-Fender | EI5GTB-macbook: I suggest you pick a solution with hardware echo cancellation adn that allows you the kind of control of its ports that you want. |
| 22:24.15 | *** join/#asterisk neurosys (n=neurosys@c-66-229-91-186.hsd1.fl.comcast.net) |
| 22:24.31 | EI5GTB-macbook | [TK]D-Fender, yea, im thinking of one of those little cisco boxes... |
| 22:24.40 | [TK]D-Fender | EI5GTB-macbook: CISCO? |
| 22:24.49 | EI5GTB-macbook | ....? |
| 22:24.58 | [TK]D-Fender | whic? |
| 22:25.08 | EI5GTB-macbook | googles, ill look em up |
| 22:25.30 | [TK]D-Fender | EI5GTB-macbook: Googleing tells me you don't seem to even know what you're thinking |
| 22:27.01 | *** part/#asterisk safa12 (n=safa@92-238-52-92.cable.ubr16.croy.blueyonder.co.uk) |
| 22:29.38 | EI5GTB-macbook | [TK]D-Fender, http://cgi.ebay.ie/LINKSYS-SP2100-UK-VoIP-ADAPTER-UNLOCKED-2-PORTS_W0QQitemZ380060124122QQihZ025QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 22:29.50 | EI5GTB-macbook | sorry, linksys / cisco |
| 22:30.10 | EI5GTB-macbook | my bad |
| 22:30.32 | EI5GTB-macbook | and those are fxs |
| 22:30.35 | [TK]D-Fender | EI5GTB-macbook: And 2 more failures, it is a *2* port device. And #2 : those are FXS ports, not FXO |
| 22:30.45 | EI5GTB-macbook | but there is an fxo version |
| 22:31.11 | [TK]D-Fender | EI5GTB-macbook: I would advise against any Linksys device for business FXO needs. |
| 22:31.25 | EI5GTB-macbook | what about personal fxo business |
| 22:31.35 | EI5GTB-macbook | im not using it for business |
| 22:31.39 | EI5GTB-macbook | just for the house here |
| 22:31.51 | [TK]D-Fender | e1What are you doing with 4 phone lines personally? |
| 22:32.07 | EI5GTB-macbook | well, i have a gsm unit on one... |
| 22:32.09 | *** part/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 22:32.19 | EI5GTB-macbook | a pstn phone line on another |
| 22:32.35 | EI5GTB-macbook | a spare |
| 22:32.40 | EI5GTB-macbook | and a skype unit |
| 22:32.54 | EI5GTB-macbook | although, asterisk will do away with skype |
| 22:32.59 | EI5GTB-macbook | so 3 would do |
| 22:33.48 | [TK]D-Fender | EI5GTB-macbook: not a great choice, but 3 x SPA-3102 might do. |
| 22:34.42 | EI5GTB-macbook | hmm |
| 22:34.50 | EI5GTB-macbook | in what way not a great choice? |
| 22:35.28 | EI5GTB-macbook | hmm, their expencive |
| 22:35.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
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| 22:42.16 | EI5GTB-macbook | [TK]D-Fender, would the 3102 pass caller id from the co to asterisk |
| 22:42.27 | [TK]D-Fender | EI5GTB-macbook: Yes |
| 22:42.32 | EI5GTB-macbook | sweet |
| 22:44.55 | *** join/#asterisk Entr4nced (n=IMG001@cpe-76-188-215-122.neo.res.rr.com) |
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| 22:48.26 | *** join/#asterisk Unlockgod (n=benjamin@195.149.30.143) |
| 22:48.31 | Unlockgod | Hey there everyeone |
| 22:49.41 | Unlockgod | can anyone help mewith installation of asterisk |
| 22:50.09 | [TK]D-Fender | Off biking for a while. Back later |
| 22:51.02 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 22:51.14 | angryuser | Unlockgod: never ask if you can ask, just ask |
| 22:51.24 | Unlockgod | sorry, |
| 22:51.28 | Unlockgod | i have CentOS |
| 22:51.31 | Unlockgod | installed on a system |
| 22:51.40 | Unlockgod | but its saying i havnt got the right ccomplier installed |
| 22:51.46 | Unlockgod | i can isntall some stuff just not all :S |
| 22:51.53 | angryuser | install g++ and gcc |
| 22:52.59 | Unlockgod | ill try that now ty:D |
| 22:53.02 | Unlockgod | i got gcc |
| 22:54.39 | *** join/#asterisk safa12 (n=safa@92-238-52-92.cable.ubr16.croy.blueyonder.co.uk) |
| 22:57.23 | *** part/#asterisk safa12 (n=safa@92-238-52-92.cable.ubr16.croy.blueyonder.co.uk) |
| 22:59.10 | Unlockgod | hi, i now got g++ there, its now saying : "You do not appear to have the sources for the 2.6.18-53.1.6.el5.028stab053.6 kernel installed. |
| 22:59.59 | bkruse | Unlockgod: You on debian? |
| 23:00.00 | angryuser | Unlockgod: install kernel headers fot you kernel version |
| 23:00.09 | bkruse | oh nvm, centos, you need kernel headers, as angryuser said |
| 23:00.09 | Unlockgod | CentOS |
| 23:00.30 | angryuser | for your* |
| 23:04.25 | *** join/#asterisk UD (n=suck@unaffiliated/underdawg) |
| 23:04.27 | *** part/#asterisk Psychobilly (n=moi@adsl139-217.kln.forthnet.gr) |
| 23:05.08 | Unlockgod | Ran yum install kernal-devel |
| 23:05.09 | Unlockgod | kernel-devel i686 2.6.18-92.1.6.el5 updates 4.8 M |
| 23:05.17 | Unlockgod | Transaction Check Error: |
| 23:05.17 | Unlockgod | <PROTECTED> |
| 23:05.38 | *** part/#asterisk UD (n=suck@unaffiliated/underdawg) |
| 23:05.58 | Qwell | Unlockgod: where did you get that kernel? |
| 23:06.05 | Qwell | 2.6.18-53.1.6.el5.028stab053.6 ? |
| 23:06.18 | *** join/#asterisk ldiamond (n=ldiamond@bas15-montrealak-1178060550.dsl.bell.ca) |
| 23:06.29 | Qwell | openvz... |
| 23:06.33 | Unlockgod | Pre installed on a system, its a test enviroment |
| 23:06.46 | Qwell | install kernel headers for *that* kernel |
| 23:06.50 | Unlockgod | i know its a virtual system |
| 23:07.46 | Qwell | ovzkernel-devel |
| 23:11.41 | *** join/#asterisk moy (n=moy@189.169.91.147) |
| 23:14.10 | *** join/#asterisk hfb (n=hfb@96.247.52.127) |
| 23:14.44 | *** join/#asterisk moy (n=moy@189.169.91.147) |
| 23:16.25 | angryuser | im off |
| 23:17.09 | nhuisman_work | boy the gui creates some nasty macros and outbound rules |
| 23:17.19 | Qwell | nhuisman_work: which GUI? |
| 23:17.28 | nhuisman_work | asterisknow / be |
| 23:17.39 | Qwell | you'd cry if you saw the mess freepbx made... |
| 23:17.44 | nhuisman_work | just very complex is all |
| 23:18.05 | nhuisman_work | http://pastebin.com/m7f6db01e |
| 23:18.20 | nhuisman_work | i think it would take me 5 mins to figure out wtf that was doing with the caller id |
| 23:18.28 | Qwell | that's not so bad :p |
| 23:18.45 | nhuisman_work | what'st he CID_() function do? |
| 23:18.55 | Qwell | it isn't a function |
| 23:19.05 | nhuisman_work | oh part of a var |
| 23:19.13 | nhuisman_work | or a data structure? |
| 23:19.17 | Qwell | var |
| 23:19.36 | Qwell | it'll evaluate ${CALLERID(num)} first. so if that's 1111, it would be ${CID_1111} |
| 23:19.49 | nhuisman_work | then checks if the length is > 2 |
| 23:20.08 | nhuisman_work | if so then use the caller id, if not use nothing? |
| 23:20.22 | Qwell | looks like it |
| 23:20.24 | seanbright | if not, use whatever callerid already was |
| 23:20.39 | Qwell | no, there's no else there |
| 23:20.47 | bkruse | exactly seanbright |
| 23:20.48 | nhuisman_work | if it's not greater then 6, then its fake and use a the global outbound cid |
| 23:20.52 | Qwell | oh |
| 23:20.53 | nhuisman_work | isn't that a ternary statement |
| 23:20.55 | bkruse | It uses what was set on the channel _before_ entering the macro |
| 23:21.05 | Qwell | well, bkruse would know :p |
| 23:21.10 | bkruse | nhuisman_work: right, heh, maybe :P |
| 23:21.22 | bkruse | It goes user cid -> trunk cid -> global outbound CID |
| 23:21.29 | seanbright | bkruse: some schmuck named brandon wrote that stuff, you know him? |
| 23:21.30 | seanbright | ;) |
| 23:21.36 | nhuisman_work | hehe |
| 23:21.46 | nhuisman_work | yeah I noticed his name in the macro notes |
| 23:21.56 | bkruse | eg for digium it would be my number (if I do not have a CID then my whole DID) (428-6115) and if that is invalid use the trunk (Digium's main number 256-428-6000) |
| 23:22.03 | bkruse | seanbright: Nah, I killed him :X |
| 23:22.07 | seanbright | hehe |
| 23:22.19 | bkruse | That wasn't my idea, pari just added my name in there when we tagged randomly lol |
| 23:22.24 | nhuisman_work | bkruse, why does it check for < 2? |
| 23:22.28 | bkruse | he thinks he takes to much credit, but I don't think he takes _enough_ |
| 23:22.31 | nhuisman_work | instead of just < 6 |
| 23:22.38 | bkruse | nhuisman_work: Basically if the CID exists at all |
| 23:22.43 | bkruse | _and_ that could be an older macro |
| 23:22.46 | nhuisman_work | bkruse, that's better then the opposite. |
| 23:22.49 | bkruse | are you using the 2.0 gui? |
| 23:22.52 | nhuisman_work | bkruse, yes |
| 23:22.55 | bkruse | Oh ok, nvm :) |
| 23:23.00 | nhuisman_work | bkruse, just upgraded my second box to it so I could try it out. |
| 23:23.15 | nhuisman_work | bkruse, I'm looking to redo my setup on the gui for the less technical people |
| 23:23.18 | nhuisman_work | after I leave. |
| 23:23.25 | bkruse | Cool cool, and that's basically "Does this variable contain anything" but there is no isset() function (pokes seanbright) |
| 23:23.31 | Qwell | after you leave, at the consultant rate? ;) |
| 23:23.40 | bkruse | nhuisman_work: That's nice. Give me their number :P |
| 23:23.41 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:23.47 | bkruse | me and Qwell will split it |
| 23:24.11 | seanbright | bkruse: "x${SOMEVAR}"="x" |
| 23:24.13 | seanbright | :) |
| 23:24.53 | bkruse | seanbright: Ugh, that was before bash had -x (I think that's the switch) hehe |
| 23:25.03 | seanbright | well... it works |
| 23:25.06 | bkruse | I guess we could use that, yse |
| 23:25.07 | bkruse | yeS* |
| 23:25.15 | bkruse | I wonder if it's faster |
| 23:25.16 | Qwell | nah, you never need the x's |
| 23:25.35 | bkruse | if "${SOMEVAR}"="" ? |
| 23:25.36 | Qwell | if test "${SOMEVAR}" == "" |
| 23:25.39 | seanbright | i think it makes the test clearer |
| 23:25.43 | Qwell | = or ==? I forget. whichever |
| 23:25.45 | nhuisman_work | bkruse, I believe they will have digiums support number :P |
| 23:25.46 | seanbright | = |
| 23:25.52 | seanbright | or is there a ==? |
| 23:25.52 | bkruse | nhuisman_work: There you go :P |
| 23:25.58 | seanbright | i never write dialplan |
| 23:26.04 | seanbright | uses AEL |
| 23:26.13 | nhuisman_work | bkruse, it was either re-write docs so they knew how to do all the stuff with the configs, or just use the gui and write much easier docs |
| 23:26.14 | seanbright | AEL is t3h awesome |
| 23:26.16 | bkruse | hehe, nice, I think Qwell was talking about bash |
| 23:26.21 | Qwell | just sh |
| 23:26.24 | bkruse | nhuisman_work: Exactly |
| 23:26.25 | seanbright | ohhh |
| 23:26.29 | Qwell | one is a bashism. |
| 23:26.33 | bkruse | Right, _bash_ has -x |
| 23:26.33 | Qwell | (= vs ==) |
| 23:26.40 | bkruse | Totally |
| 23:27.00 | seanbright | one day we will require quoted strings in dialplan though |
| 23:27.04 | seanbright | and on that day i can rest |
| 23:27.13 | Qwell | == == bashism |
| 23:27.15 | seanbright | lay down my keyboard and mouse for the last time |
| 23:27.22 | nhuisman_work | I'll just need to make sure they backup the box often so if the config gets gui munged it can be restored. |
| 23:27.47 | bkruse | nhuisman_work: Gui has backup builtin |
| 23:27.52 | nhuisman_work | ya |
| 23:28.00 | bkruse | it actually works now |
| 23:28.02 | nhuisman_work | I think I'll also keep some dumped images of the system |
| 23:28.02 | bkruse | lol |
| 23:28.12 | nhuisman_work | for a full restore to a known working state. |
| 23:28.18 | bkruse | Ya, not a bad idea |
| 23:28.26 | nhuisman_work | I think i'm going to end up using the redhat rpms instead of rpath |
| 23:28.34 | nhuisman_work | i'm not a fan of conary |
| 23:28.39 | nhuisman_work | at _all_ |
| 23:28.43 | bkruse | lol |
| 23:28.47 | bkruse | it's a requires taste :] |
| 23:28.52 | bkruse | heh, required too |
| 23:28.55 | nhuisman_work | yeah |
| 23:28.56 | Qwell | acquired :p |
| 23:29.02 | nhuisman_work | acquired and required, mmm |
| 23:29.05 | nhuisman_work | sounds like a woman |
| 23:29.12 | bkruse | wow, all 3 |
| 23:29.16 | nhuisman_work | hmm that could be taken wrong :P |
| 23:29.18 | bkruse | nhuisman_work: woah |
| 23:29.20 | bkruse | lol |
| 23:29.35 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 23:29.38 | nhuisman_work | lmao |
| 23:29.43 | bkruse | and it's my time to leave after that. lol |
| 23:29.50 | *** join/#asterisk jplank (n=GBove@ool-18bb018e.dyn.optonline.net) |
| 23:29.51 | nhuisman_work | yeah no kidding |
| 23:29.53 | bkruse | cya guys |
| 23:30.53 | seanbright | later. |
| 23:31.40 | nhuisman_work | is there a simpler way then doubling all my outbound prefixes to deal with some people who dial 9 and some who dont? |
| 23:32.07 | jplank | dailplan to strip or add the 9? |
| 23:32.18 | [TK]D-Fender | nhuisman_work: Why would you add "9" prefixed ones in the first place? |
| 23:32.36 | nhuisman_work | [TK]D-Fender, thought you were out biking :P |
| 23:32.48 | [TK]D-Fender | nhuisman_work: Just got back in |
| 23:32.50 | nhuisman_work | ah |
| 23:33.01 | [TK]D-Fender | nhuisman_work: Speed biking thought the trails |
| 23:33.12 | nhuisman_work | [TK]D-Fender, must be close to the trails |
| 23:33.22 | [TK]D-Fender | nhuisman_work: 5 min tops |
| 23:33.24 | *** join/#asterisk Unlockgod (n=benjamin@195.149.30.143) |
| 23:33.56 | Unlockgod | hi sorry to be a pain,i use that ovzkernel-devel |
| 23:34.07 | [TK]D-Fender | nhuisman_work: Get your monkeys trained off of that archaic "dial 9" BS. |
| 23:34.19 | Unlockgod | it install a differnt verssion |
| 23:34.20 | nhuisman_work | [TK]D-Fender, yeah the old system used it, but people still use it |
| 23:34.22 | jplank | some people like it |
| 23:34.26 | Unlockgod | now i get this. dev < 3.2-7 conflicts with ovzkernel-2.6.18-53.1.6.el5.028stab053.6.i686 |
| 23:34.27 | nhuisman_work | [TK]D-Fender, but I think some people don't use it too. |
| 23:34.37 | [TK]D-Fender | nhuisman_work: 2 words "COLD TURKEY" |
| 23:35.13 | nhuisman_work | [TK]D-Fender, I can only imagine the number of people complaining if I did that |
| 23:35.17 | nhuisman_work | we are talking lots of old astronomy farts |
| 23:35.20 | nhuisman_work | who don't like change |
| 23:35.27 | nhuisman_work | and control my paycheck |
| 23:35.28 | nhuisman_work | hehe |
| 23:35.28 | [TK]D-Fender | nhuisman_work: Don't waste time catering to people who can't consolidate on a common system. Imagine if half the people train newbs 1 way, and the other the other? CHAOS |
| 23:35.50 | seanbright | chaos? with 9 vs. without? |
| 23:36.04 | seanbright | i've told you a gazillion billion trillion times about exagerating |
| 23:36.14 | [TK]D-Fender | nhuisman_work: Seriously "9" REDUCES productivity, and adds complexity for nothing. They don't dial 9 at home, so why should they at the office? |
| 23:36.15 | nhuisman_work | [TK]D-Fender, also our central office still uses 9 so it would probably be an issue consistency too. |
| 23:36.49 | nhuisman_work | I feel the same way |
| 23:37.00 | jplank | couldn't you just add something to your dialplan that looks for the 9 and removes it, or am I missing something? |
| 23:37.39 | nhuisman_work | is it possible to put an if statement in a pattern? |
| 23:37.43 | [TK]D-Fender | nhuisman_work: this requires 1 line for each of your existing patterns |
| 23:37.49 | [TK]D-Fender | nhuisman_work: Nope |
| 23:37.52 | nhuisman_work | yeah didn't think so |
| 23:38.05 | seanbright | lame |
| 23:38.12 | jplank | couldn't you match for the 9, then use the if statement? |
| 23:38.12 | seanbright | asterisk sucks |
| 23:38.19 | nhuisman_work | laugh |
| 23:39.35 | nhuisman_work | currently I have one with a 9 stripped and one normal for each outgoing rule |
| 23:40.46 | [TK]D-Fender | jplank: "the" if? He has to validate multiple patterns. No, that really isn't viable |
| 23:41.59 | jplank | maybe I'm missing what hes getting at? |
| 23:42.02 | [TK]D-Fender | nhuisman_work: My coworkers were so happy when I went with transparent 7-10-11 digit dialing |
| 23:42.10 | [TK]D-Fender | jplank: Could be.... |
| 23:42.43 | nhuisman_work | [TK]D-Fender, what do you mean by transparent? like you it will add the prefix? |
| 23:42.56 | jplank | I thought he wanted to have one outbound rule with people dialing with or without a 9 |
| 23:42.57 | nhuisman_work | s/you// |
| 23:43.06 | nhuisman_work | laugh didn't work |
| 23:43.35 | jplank | couldn't he just make the rule without the 9, then match the 9 inside the dialplan, and use an if to check if its a number prefixed with 9, or something that really had 9 in front of it |
| 23:43.49 | *** join/#asterisk Illarane (n=heifer@pdpc/supporter/student/Veratien) |
| 23:43.58 | jplank | or am I still missing it? |
| 23:44.08 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:44.08 | nhuisman_work | you can check an outgoing dialed number from the pattern match? |
| 23:44.38 | nhuisman_work | to see what the first digit of it is? |
| 23:44.48 | [TK]D-Fender | nhuisman_work: 10-digit became recently mandatory. For those who like thinking that on-island is still 7 digit I add teh default AC when dialing it out. some calls to 10-digit #'s are not LD, and of course all the other NANPA AC's are LD, so if any of our 2 "potentially local" AC's are dialed as 10-digit, leave them alone as dilaing "1" in front would cause failure, and add a 1 to all others. |
| 23:45.06 | nhuisman_work | [TK]D-Fender, ah i get it. |
| 23:45.39 | jplank | [TK]D-Fender: thats pretty much what I was trying to get at, just with a 9 |
| 23:46.00 | [TK]D-Fender | nhuisman_work: Users shouldn't have to think any more than is necessary to be productive. When they get CID from an outside call they should be able to hit redial on their phone's missed-calls list to call right back with impunity |
| 23:46.22 | jplank | [TK]D-Fender: explain that to avaya please :) |
| 23:46.27 | nhuisman_work | [TK]D-Fender, It doesn't seem like 10 digit numbers are mandatory here yet. |
| 23:46.44 | [TK]D-Fender | jplank: does Avaya FORCE you to use a prefix? I know Nortel key systems sure didn't |
| 23:46.48 | nhuisman_work | [TK]D-Fender, that could be because this whole state has one area code though ;) |
| 23:47.02 | jplank | I'm not talking about key systems |
| 23:47.29 | jplank | 10 digit will be mandatory within all 50 states within a few years |
| 23:47.31 | [TK]D-Fender | jplank: Ok, what system can you not configure to not require a "9" prefix? |
| 23:47.32 | nhuisman_work | so to clear things up, I'll show you the mess I have now for my outgoing rules : http://pastebin.com/m483fdc45 I'm pretty sure that could be greatly shortened, I just threw that in there because it worked 6 months ago and now I'm coming back to it. |
| 23:48.07 | jplank | I'm not saying not configure with a 9, I'm saying in some instances you need to use a 9 to pull trunks out of a pool |
| 23:48.29 | jplank | and in those cases avaya doesn't give any easy straight forward way to call back from caller ID |
| 23:48.32 | nhuisman_work | also I only have one trunk |
| 23:48.33 | [TK]D-Fender | jplank: Need to? No choice to dial out otherwise? |
| 23:48.49 | [TK]D-Fender | nhuisman_work: I'm feeling generous, hold on. |
| 23:48.58 | nhuisman_work | [TK]D-Fender, :) |
| 23:49.19 | Unlockgod | can amyone help wih kenel src probs |
| 23:49.29 | Unlockgod | You do not appear to have the sources for the 2.6.18-53.1.6.el5.028stab053.6 kernel installed. |
| 23:49.30 | nhuisman_work | [TK]D-Fender, like you said earlier I can change the long distance to one line _011XX. |
| 23:49.45 | nhuisman_work | err that was international |
| 23:50.10 | seanbright | Unlockgod: looks pretty clear to me |
| 23:50.19 | nhuisman_work | Unlockgod, what os? |
| 23:50.24 | seanbright | Unlockgod: you need to install the sources for the 2.6.18-53.1.6.el5.028stab053.6 kernel |
| 23:50.25 | nhuisman_work | Unlockgod, sorry, what distro. |
| 23:50.41 | nhuisman_work | Unlockgod, you need kernel-devel or whatever it's called in your distro. |
| 23:50.42 | seanbright | yum install kernel-devel kernel-headers |
| 23:50.51 | nhuisman_work | ah yes and kernel-headers. |
| 23:50.53 | jplank | [TK]D-Fender: take like an IP office for example, you have to put something in the dialplan to let it know your trying to grab an outside trunk |
| 23:50.56 | Unlockgod | same for ovz |
| 23:50.57 | jplank | thats where the 9 comes in |
| 23:51.10 | jplank | I guess you can do it by 1 if your doing 11 digits |
| 23:51.33 | [TK]D-Fender | jplank: retard appliance :) |
| 23:51.37 | jplank | yea |
| 23:51.40 | jplank | thats my point |
| 23:51.42 | jplank | lol |
| 23:51.59 | [TK]D-Fender | jplank: My dumb-fuck head office went with that but I managed to pass on the Kool-Aid :) |
| 23:52.22 | jplank | they could of gone worse |
| 23:52.32 | jplank | have you tried avaya's onex phones |
| 23:52.45 | jplank | the pbx built into the phones |
| 23:52.57 | jplank | talk about ass backwards |
| 23:53.51 | jplank | maybe I'm missing something, but couldn't nhuisman_work match for _9. and then use a if to check ${EXTEN} and parse it? |
| 23:54.35 | nhuisman_work | so make a single pattern, then if statements to each type of 7, 10, 11 digit bit if I actually wanted special rules for each |
| 23:54.59 | nhuisman_work | guess i'm not sure what that syntax would look like. |
| 23:55.31 | [TK]D-Fender | nhuisman_work: http://pastebin.com/m4a406802 <----- MUCH better |
| 23:55.41 | mchou | hey, anyone here use Grandcentral for a DID? |
| 23:56.09 | [TK]D-Fender | nhuisman_work: and I fixed the fact you left LD stuff in the NON LD group |
| 23:56.30 | jplank | I guess thats actually a lot less work then ripping the 9 out :) |
| 23:56.43 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 23:56.47 | nhuisman_work | [TK]D-Fender, oh you mean leaving the 11 digit in the outbound-local? |
| 23:56.53 | [TK]D-Fender | jplank: No, ripping 9 out cuts that PB in HALF |
| 23:56.59 | [TK]D-Fender | nhuisman_work: Yes |
| 23:57.01 | nhuisman_work | kk |
| 23:57.28 | jplank | right, but you'd need like 3 lines to rip the 9 out |
| 23:57.39 | nhuisman_work | [TK]D-Fender, yeah using the macro is much nicer. |
| 23:57.43 | jplank | two lines maybe |
| 23:58.03 | [TK]D-Fender | jplank: Look at my PB, 5 <- |
| 23:58.49 | jplank | I meant before the actual dial, not ${EXTEN:1} |
| 23:58.52 | Yourname | ~callforwarding |
| 23:58.54 | rasterix | anyone no a good place to go to get sample real world extensions.conf |
| 23:59.05 | Yourname | Hi, is there somewhere where I can read more about call forwarding? |
| 23:59.08 | nhuisman_work | rasterix, the book sort of has some |
| 23:59.19 | jplank | rasterix: you try googling extensions.conf? |
| 23:59.29 | rasterix | nhuisman: got them |
| 23:59.31 | rasterix | to simple |
| 23:59.33 | [TK]D-Fender | ~jerjerguide |
| 23:59.33 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 23:59.36 | [TK]D-Fender | rasterix: ^^^^ |
| 23:59.39 | JT | s/no/know/ |
| 23:59.41 | JT | s/to/too/ |
| 23:59.45 | JT | english |
| 23:59.56 | rasterix | jblank: google just gets old extensions.conf |
