| 00:00.14 | drynish | However, when the answering machine started, is it possible that it detects another phone answering the line |
| 00:00.26 | drynish | so that it hangup whenever I take another phone? |
| 00:00.37 | drynish | Or would it be simple to just cancel the call by a key |
| 00:04.33 | drynish | somebody here? :0 |
| 00:11.31 | *** join/#asterisk dailylinux (~test@88.87.48.12) |
| 00:12.41 | drynish | is it possible to exist voicemail fast? |
| 00:23.55 | p3nguin | If I spoke that language, I'd try to answer you. |
| 00:26.24 | drynish | hahaha :P |
| 00:26.25 | drynish | thanks |
| 00:28.06 | drynish | :) |
| 00:28.23 | drynish | My asterisk answer a call after a delay of 18 secs ;) |
| 00:28.34 | drynish | If I take the phone after that delay |
| 00:29.28 | p3nguin | I was with you when you said asterisk answers after 18 seconds. Then you lost me with the next part about taking a phone after a delay. |
| 00:30.23 | drynish | If somebody calls in and I don't have the time to answer a phone in my house... asterisk answer the call and plays the voicemail app |
| 00:30.35 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:30.43 | p3nguin | okay |
| 00:30.46 | drynish | however if I take a phone in my house |
| 00:30.54 | drynish | I hear asterisk talking to the caller |
| 00:31.06 | p3nguin | Oh. |
| 00:31.16 | p3nguin | You wired your phones incorrectly. |
| 00:31.26 | drynish | No it is what I want :) |
| 00:31.27 | p3nguin | You have the phone on the same side of asterisk as the line in. |
| 00:31.52 | p3nguin | It might be what you want, but it's not necessarily the best way. |
| 00:32.08 | drynish | I know but my ATA was producing too much echo |
| 00:32.19 | drynish | and my gf was tired of all the issues ;) |
| 00:32.36 | p3nguin | Ideally, asterisk will be the only thing connected to the line. |
| 00:32.43 | p3nguin | It will take calls. |
| 00:33.03 | p3nguin | Then asterisk will make your phone ring for a time. |
| 00:33.07 | drynish | I know |
| 00:33.18 | p3nguin | Then if you don't answer in time, the call goes to voicemail. |
| 00:33.31 | drynish | Understod |
| 00:34.33 | drynish | for now everything is on the same side ;) |
| 00:34.44 | drynish | so I was searching a way to cancel voicemail fast |
| 00:37.01 | p3nguin | I don't think it works like that. |
| 00:37.20 | p3nguin | I think the only way is to configure it correctly. |
| 00:38.12 | drynish | why |
| 00:38.24 | p3nguin | Because that's how it workd. |
| 00:38.26 | p3nguin | works |
| 00:38.43 | drynish | I know |
| 00:39.55 | *** join/#asterisk radic (~radic@dslb-178-002-233-213.pools.arcor-ip.net) |
| 00:39.56 | drynish | But thanks anyway |
| 00:40.06 | drynish | You were great for everytime you helped me |
| 00:41.53 | p3nguin | WTF?! http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=130472604393 |
| 00:41.56 | p3nguin | What am I missing here? |
| 00:42.55 | drynish | haha :P |
| 00:43.54 | p3nguin | No, really. What am I missing? |
| 00:56.22 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 01:02.04 | ChannelZ | PANTS! |
| 01:02.42 | ChannelZ | It was personally touched by God himself, Steve Jobs. |
| 01:05.33 | p3nguin | So the seller is just insane, I guess. |
| 01:10.03 | ChannelZ | But it has free shipping.. |
| 01:13.39 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 01:22.59 | *** join/#asterisk coppice (~coppice@m121-202-71-243.smartone-vodafone.com) |
| 01:23.15 | *** join/#asterisk james_zhu (~Administr@183.16.219.77) |
| 01:29.45 | p3nguin | Is there anything more detailed and thorough than DumpChan()? I'm looking for more technical information on a channel during a call. |
| 01:31.04 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 01:33.54 | *** join/#asterisk krzee (krzee@openvpn/community/support/krzee) |
| 01:34.14 | krzee | does anyone here run a stun server? |
| 01:35.39 | krzee | im trying stund on freebsd, with nothing running stund tells me that the port is in use, even tho i can verify with netstat / sockstat that nothing is listening on that port |
| 01:37.21 | WIMPy | Because you only supplied one port number it tries to open twice? |
| 01:38.04 | krzee | i did use the same ip for -h and -a, the only doc i could find on running it said to (which i found weird) |
| 01:39.21 | WIMPy | You can, but you have to supply different ports for -p and -o then. |
| 01:39.41 | krzee | oh cool, is there a doc somewhere that you know of? |
| 01:39.52 | krzee | i never even found -o and -p... it did not come with a man page |
| 01:39.54 | WIMPy | no |
| 01:40.01 | WIMPy | only --help |
| 01:40.06 | krzee | ahh nice! |
| 01:40.41 | krzee | why does it want 2 ips and 2 ports? |
| 01:40.59 | WIMPy | It starts 2 listeners. |
| 01:41.18 | krzee | and they are the same thing? is there a point to that? |
| 01:42.04 | WIMPy | I think it to help in determining the type of port address translation. |
| 01:43.40 | krzee | hah you're right, default for -p is 3478 -o defaults to 3479, but each one sets up its own secondary |
| 01:43.48 | sunfone | You can use an alias IP |
| 01:43.56 | sunfone | on a single interface, if that is all you have |
| 01:44.01 | krzee | so the default for -p conflicts with default for -o when using the same ip |
| 01:44.44 | krzee | hrm nope im wrong, still didnt work |
| 01:45.10 | krzee | sunfone, this is a darknet, inside a vpn, no alias ip available |
| 01:45.30 | WIMPy | Doesn't work or moans and runs on only one port? |
| 01:46.03 | krzee | doesnt work, still says its in use |
| 01:46.11 | krzee | although when i use a second ip, all is well |
| 01:46.19 | krzee | thing is, i dont want it running on the external ip |
| 01:46.29 | krzee | ooo i could try 127.0.0.1 tho! |
| 01:46.59 | WIMPy | It might linger. |
| 01:47.47 | krzee | woohoo that at least got it to run! |
| 01:47.52 | krzee | now to test it! |
| 01:48.06 | krzee | WIMPy, thank you very much =] |
| 01:49.06 | file | it wants different IPs/ports so it can use it to determine what type of NAT, having two matters |
| 01:49.19 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 01:49.49 | krzee | ohh |
| 01:50.16 | krzee | well i guess if 1 works all will, every unit will be the same model router running openvpn with the same nat config |
| 01:53.42 | krzee | oh now i put together the linger comment with what file said |
| 01:53.55 | krzee | along with the fact that clients only get 1 stun server ip |
| 01:54.03 | krzee | i guess 127.0.0.1 is a bad choice ;] |
| 01:57.06 | WIMPy | That was about a possible socket problem. |
| 01:59.09 | krzee | i guess i could reserve another vpn ip for this, write a script to add the alias, then call the script when openvpn starts with --up in openvpn |
| 01:59.27 | krzee | with 127.0.0.1 it takes forever to register |
| 02:01.08 | krzee | but it did solve what i wanted, so im making definite progress and am pretty sure i know where to go from here |
| 02:01.19 | krzee | another thank you to WIMPy / file! |
| 02:03.40 | krzee | muahaha it works great that way! |
| 02:03.49 | krzee | problem solved! |
| 02:04.03 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:20.02 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279411300.dsl.bell.ca) |
| 02:22.34 | *** part/#asterisk krzee (krzee@openvpn/community/support/krzee) |
| 02:25.49 | dijib | hello all |
| 02:25.55 | dijib | any MOH experts here? |
| 02:30.18 | p3nguin | ~ask |
| 02:30.18 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:44.45 | *** join/#asterisk willj (~willj__@24-145-40-016-dhcp.aik.sc.atlanticbb.net) |
| 02:44.50 | dijib | hey p3nguin hows it going man |
| 02:45.24 | dijib | just ive configured musiconhold.conf to spec i believe but the mp3s arnt playing |
| 02:45.33 | dijib | and im not an * guru |
| 02:45.50 | dijib | should have the mp3 codec installed though. |
| 02:47.38 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:48.12 | willj | Could anyone make a hardware recommendation for someone who is looking to use Asterisk for a small business? There would probably be only 2 or 3 physical phone, it would need to have the ability to forward calls to a cell phone when "out of the office." |
| 02:48.50 | ChannelZ | Do you need to use existing POTS lines or do you want to do VoIP? |
| 02:50.08 | willj | Existing lines that are provided by my ISP, which is a cable company. I would also use the existing lines that are in the building. |
| 02:54.07 | p3nguin | It's almost as if you've said that your ISP provides telephone lines. |
| 02:55.41 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 02:56.09 | WIMPy | Isn't that te satandard now? |
| 02:56.14 | dijib | willj, any p4 with 512MB will do. |
| 02:56.17 | willj | Yeah I figured that that shouldn't matter, but I wasn't sure if the cable company used different protocols for providing their phone service over the coax. It's not from the telco, where traditional RJ15 is used. |
| 02:56.48 | p3nguin | If it's not a copper pair from the telephone company, we typically don't refer to it as POTS lines. |
| 02:57.29 | dijib | willj, dump your ISP's voip as its probably way to expensive |
| 02:57.41 | dijib | go with a good voip ITSP |
| 02:57.54 | willj | It cost $10 extra a month for the next 2 years |
| 02:57.57 | p3nguin | So the question still remains: do you intend to use lines from a telco or do you want to do VoIP? |
| 02:58.25 | dijib | willj is already using voip he just doesnt know it |
| 02:58.31 | willj | I would like to be able to use the line that the ISP provides |
| 02:58.47 | p3nguin | The ISP does not provide phone lines. |
| 02:59.22 | dijib | your ISP is providing you an overprices VoIP service |
| 03:00.00 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 03:00.15 | p3nguin | They may give you cable services over coax, and they probably will not allow you to use it directly from your own equipment. Are you wanting to run your asterisk with a phone line to your voice-enabled cable modem? |
| 03:01.47 | willj | I think so, my cable company provides a voip connection for $10 more on the bill per month. I wouldn't really be interested in using anything else, mainly because where I live it seems as if it's hard to get local numbers. |
| 03:02.08 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 03:02.47 | willj | More or less I want the phone to ring, asterisk to pick up, and do it's thing. Whether it's just go to a voicemail, route a call, or forward it to an external line. I already have a machine running debian, and the source has already been compiled. |
| 03:03.38 | *** join/#asterisk james_zhu (~Administr@183.16.87.112) |
| 03:03.42 | p3nguin | Let's call you... So we start at the telco where I've called your number, which goes to your cable company who turns it into VoIP, who sends it to your modem, which turns it into analog, where you want to run it to asterisk to turn it back into VoIP. |
| 03:04.08 | willj | I just need to find the devices or cards that are required to begin testing. |
| 03:04.16 | p3nguin | Latency, echo... |
| 03:04.45 | willj | Any suggestions for VoiP providers? |
| 03:04.56 | p3nguin | Will the cable company give you VoIP access to your phone account so that it does not have to run through your cable modem? |
| 03:04.57 | WIMPy | willj: Better try to find out how you can get your phone accouts login details. |
| 03:05.33 | dijib | willj, would you settle for a toll-free number instead???? |
| 03:05.38 | willj | In the past they said that what they were using wasn't VoIP, but I always thought that it was. |
| 03:05.48 | p3nguin | They all say that. |
| 03:05.49 | willj | Yeah toll free would be fine. |
| 03:05.56 | willj | Ya, I figured they were full of it |
| 03:06.12 | dijib | then there is your option. ditch your ISP's voip and get a toll free DID |
| 03:06.16 | p3nguin | I'd be surprised if the cable modem isn't a SIP device. |
| 03:06.19 | dijib | run your asterisk box.... be happy |
| 03:06.35 | willj | okay, but that still leaves the question |
| 03:06.41 | willj | What hardware do I need to set up my lines |
| 03:06.50 | p3nguin | With VoIP, there are no lines. |
| 03:07.01 | dijib | now youve got me thinking p3nguin, i think i have a voip cable modem i should see if i can make into a sip client |
| 03:07.14 | CoffeeIV | I'm using 1.8.5.0 and I'm trying to compile it to produce backtraces, as per the wiki, because I am having stability issues - however, in make menuconfig I cannot find the "BETTER_BACKTRACE" option. Do I have to enable that by hand-editting a file ? |
| 03:07.21 | willj | So how will the phones ring w/o lines? |
| 03:07.23 | WIMPy | I've been told the cable guys are often still using H.323. |
| 03:07.38 | p3nguin | IP phones most commonly are connected via Ethernet. |
| 03:07.56 | dijib | back to my issue with MOH, what could my issue be. moh show files lists all files, i think ive got musiconhold.conf correct, in CLI it says it starts... but nothing. |
| 03:08.19 | WIMPy | willj: You don't have a line, so you know the answer. |
| 03:08.53 | p3nguin | This is a telephony channel. "Line" means a PHONE LINE. We don't have those in VoIP. |
| 03:09.38 | WIMPy | So what do you call the part that comes out of the IADs? |
| 03:10.19 | p3nguin | I guess that depends on what is coming out of it. |
| 03:10.26 | p3nguin | If it's a phone line, I'd call it a line. |
| 03:10.43 | p3nguin | If it's an Ethernet cable carrying IP, I wouldn't call it a line. |
| 03:10.52 | WIMPy | That's where the fun begins. |
| 03:11.02 | p3nguin | fuzzy terminology? |
| 03:11.19 | WIMPy | I think it has become fuzzy somehow. |
| 03:12.01 | willj | So the voice gateways, such as the PAP2T-NA or internal telephony such as the ones offered from Digium are useless if you drop the copper lines and go with VoIP/Ethernet phones? |
| 03:12.47 | WIMPy | If you're using VOIp directly, yes. |
| 03:12.48 | p3nguin | Actually, a PAP2T ATA turns IP/SIP into a phone line. |
| 03:12.55 | p3nguin | and vice versa. |
| 03:12.59 | WIMPy | If you get your VOIP on a line, no. |
| 03:13.02 | willj | I appreciate the patience and info very much btw.. It goes without saying that it's been awhile since I dabbled with Asterisk. |
| 03:13.44 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:13.49 | willj | So using the gateway is useful for using the existing lines? |
| 03:14.23 | WIMPy | >>willj: Better try to find out how you can get your phone accouts login details. |
| 03:14.26 | p3nguin | If you have a phone line, which you do not, you could use a device like the SPA-3102 to turn your line (which you don't have) to VoIP. |
| 03:14.46 | willj | I have phone lines |
| 03:14.51 | p3nguin | You said you don't. |
| 03:15.01 | p3nguin | The PAP2T is an FXS device, meaning it only connects to a phone, never a phone line. |
| 03:15.02 | WIMPy | p3nguin: Didn't you just say you call that a line? |
| 03:16.32 | p3nguin | Erm, yes. I thought we were talking about a phone line from a telco, which he does not have. |
| 03:16.32 | dijib | my moh does squat.... would love a little guidance |
| 03:16.42 | *** join/#asterisk penguin (greenwolf@cpe-74-77-221-5.buffalo.res.rr.com) |
| 03:16.46 | p3nguin | dijib: I haven't seen your pastebin yet. |
| 03:16.46 | penguin | join #trixbox |
| 03:17.18 | dijib | willj, im telling you. ditch your ISP Phone solution, signup for a voip.ms toll free DID and away you go with configing your * box |
| 03:17.22 | willj | it comes into the office with RJ15, and out of the modem it plugs into the pre-existing phone line. Which I understand is nothing more than VoIP, but I was just thinking that it would be more convenient to use that since I know I can get a dial tone. |
| 03:17.26 | dijib | oh and buy a PAP2T NA |
| 03:17.28 | WIMPy | So there goes the terminology. |
| 03:18.12 | p3nguin | The PAP2T will only be useful if you want to use analog telephones with Asterisk. |
| 03:18.33 | dijib | i have conf's ready made for you. with autoimixmon working on all in and outbound calls |
| 03:18.35 | p3nguin | Get the phone account information from the cableco. |
| 03:18.44 | dijib | screw cableco |
| 03:18.51 | dijib | drop them like bricks. |
| 03:18.55 | willj | I am not opposed to ditching them at all, I am just trying to buy the equipment to begin setting this up tonight. When I saw many different types of telephony cards offered by Digium, I got somewhat confused. I wasn't sure if I need the card, the card + the gateway, or something else entirely. |
| 03:19.22 | dijib | couldnt he wireshark that p3nguin ? thos creds |
| 03:19.37 | dijib | wait no nvmd |
| 03:19.39 | dijib | docsis3 |
| 03:19.44 | p3nguin | probably not. |
| 03:20.05 | dijib | so pastebin eh... arrg |
| 03:20.17 | dijib | im thinking its a framerate issue |
| 03:20.23 | dijib | i get no errors is why |
| 03:20.37 | willj | So going with third party voip providere with a DID would circumvent the Cablco all teogether? |
| 03:20.52 | p3nguin | If you want to keep using the phone "line" that your cable modem provides, you'll need either a $300 card or a $70 SPA-3102. |
| 03:20.53 | dijib | willj, yes. |
| 03:21.12 | dijib | dont need a $10\mo fee, when $1.49 + usage will suffice |
| 03:21.26 | WIMPy | Or get te login details from the cableco or the modem and use it directly. |
| 03:21.35 | p3nguin | Using an ITSP, they provide the phone number and all phone services. |
| 03:21.53 | dijib | ive had mine for 1.5 months now. bit of usage. $9 total charges & the money hasnt left my paypal yet |
| 03:22.13 | willj | Got ya, so basically I could go the voip route. And if I wanted to use the existing wiring in the office w/ analog phones, I could just get a gateway, such as the PAP2t? |
| 03:22.32 | dijib | oui! |
| 03:22.38 | p3nguin | The PAP2T is not a gateway, it is only an FXS device. |
| 03:22.42 | willj | nice.. Yeah, you guys have convinced me to go VoIP. |
| 03:22.52 | dijib | naming conventions p3nguin |
| 03:23.01 | dijib | they get me errryday |
| 03:23.04 | p3nguin | I don't know what that comment means. |
| 03:23.13 | willj | But a FXS is the port that converts digitial into analog? |
| 03:23.22 | willj | So the internal cards that have FXS ports basically are the same thing, correct? |
| 03:23.23 | p3nguin | The SPA-3102 is a gateway. |
| 03:23.25 | dijib | it means, one person calls one thing something. another knows it as something else |
| 03:23.31 | dijib | common communication issue |
| 03:23.37 | p3nguin | FXS ports provide a jack for telephones. |
| 03:23.56 | p3nguin | FXO ports connect to your telco lines. |
| 03:23.56 | WIMPy | I'd call an ATA a gateway as well. |
| 03:24.04 | p3nguin | Loosely, maybe. |
| 03:24.15 | willj | I actually have an old PAP2T, but I think it's locked for vonage :( |
| 03:24.36 | dijib | ahh gateway.. check. |
| 03:24.52 | dijib | you can unlock those i believe |
| 03:24.55 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 03:25.36 | willj | So if I had that, an analog phone, configured Asterisk, and picked up a VoIP account with a DID. I would be good to go for calls? |
| 03:25.41 | willj | Or at least enough to begin testing? |
| 03:25.56 | dijib | willj, http://www.voip-info.org/wiki/view/Linksys+PAP2+Unlocking+Methods |
| 03:26.15 | dijib | http://www.voip.ms has been great so far |
| 03:26.31 | dijib | i had an issue and they were all over it first day after ticket open |
| 03:27.06 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 03:28.35 | p3nguin | If you have a DID with an ITSP, and you have either Asterisk with IP phones, Asterisk with an ATA and a regular phone, or just an ATA with a regular phone, you can start testing. |
| 03:31.26 | willj | That's kind of exciting, I may try to unlock this phone and pick up a DID.. I am still having a hell of a time finding a number for my town though so it looks like toll-free will be the way to go. |
| 03:31.48 | dijib | if you have a paypal account, voip.ms was instantaneous to get setup. |
| 03:32.20 | dijib | i read something about getting your number through voip.ms |
| 03:32.23 | dijib | let me see |
| 03:32.32 | p3nguin | You can port in. |
| 03:32.47 | p3nguin | It'll cost you $25. |
| 03:35.10 | p3nguin | But that's 2.5 months worth of service if you would just get your phone credentials from the cableco. |
| 03:35.45 | willj | So I would need to get the creds either way? |
| 03:35.52 | p3nguin | No. |
| 03:36.22 | p3nguin | But if you get the information from them, you could just go ahead and use the service that you already have and use the number you already have. |
| 03:37.11 | willj | Will they usually give that out? |
| 03:37.31 | p3nguin | I don't know about usually, but it happens sometimes. |
| 03:37.43 | willj | Ahh, I guess it would depend on who you talk to and what company you're dealing with. |
| 03:38.06 | WIMPy | And you may have to call many times until you fond someone who understands the question. |
| 03:38.10 | p3nguin | You probably won't get it by calling the support phone number and talking to tier 1. |
| 03:38.10 | willj | The ISP that I use is pretty small and local, so they may not even know what I am talking about. |
| 03:38.56 | willj | Yeah that will probably be the case. I agree though, converting voip to analog, then back to voip, and back to analog is kind of ridiculous |
| 03:39.03 | p3nguin | If you know how to reach an engineer, they might be able to tell you how to get the info. |
| 03:44.00 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:45.48 | dijib | now in CLI its not showing any files doing a 'moh show files' |
| 03:45.51 | dijib | it was |
| 03:46.08 | p3nguin | Did I miss the pastebin? |
| 03:46.23 | dijib | no im trying to get that together for you |
| 03:46.25 | dijib | but i suck |
| 03:46.31 | p3nguin | apparently |
| 03:46.33 | dijib | my 3hr = your 20min |
| 03:46.50 | p3nguin | I want to see musiconhold.conf. |
| 03:46.58 | p3nguin | We'll go from there. |
| 03:46.58 | willj | Thanks again for the info everyone, you guys rock! |
| 03:47.09 | p3nguin | Good luck. |
| 03:47.23 | willj | ty :) |
| 03:47.55 | dijib | music on holf.conf http://pastebin.com/fhMc98xe |
| 03:48.31 | p3nguin | Now would you like to see what it should look like? |
| 03:48.40 | dijib | lol |
| 03:48.43 | dijib | as per usual. |
| 03:49.55 | dijib | oh i missed the [general] that was defined |
| 03:49.59 | dijib | in the .conf |
| 03:50.23 | p3nguin | So you have a general section, and it is blank? |
| 03:50.31 | dijib | yep |
| 03:51.24 | p3nguin | Remote the format= line. |
| 03:51.37 | p3nguin | Make sure you have valid mp3 files in /var/lib/asterisk/mohmp3 |
| 03:52.04 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 03:52.06 | p3nguin | Make sure the directory and files are able to be read by asterisk. |
| 03:52.14 | CoffeeIV | Is there a recommended way to make a script that will automatically restart asterisk if the daemon crashes ? |
| 03:52.17 | p3nguin | Then in the cli, run moh reload. |
| 03:52.30 | dijib | did |
| 03:52.31 | p3nguin | coffeeiv: I do it with a single line in crontab using pidof. |
| 03:52.36 | dijib | also format = ??? |
| 03:52.50 | p3nguin | * * * * * [[ -z `/bin/pidof -s asterisk` ]] && /etc/rc.d/asterisk restart>/dev/null 2>&1 |
| 03:53.24 | CoffeeIV | p3nguin: thanks man, I will use that ! |
| 03:53.32 | dijib | killall? |
| 03:53.36 | dijib | kill |
| 03:53.40 | p3nguin | Make sure you notice when it is crashing, and then fix it. |
| 03:53.55 | p3nguin | Hmm? killall/kill? |
| 03:54.04 | dijib | i dont get what you just pasted |
| 03:54.39 | p3nguin | if there are no PIDs for asterisk, use the rc script to restart asterisk. |
| 03:55.02 | p3nguin | Check every minute of every hour of every day of every month, every day of the week. |
| 03:55.48 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 03:55.58 | p3nguin | coffeeiv: Oh, make sure you have pidof. It won't work well if you don't have it. :) |
| 03:56.22 | SunTsu | pgrep would be a viable alternative |
| 03:56.34 | p3nguin | Yeah, it could be done with pgrep. |
| 03:57.56 | p3nguin | Something like * * * * * /bin/sh -c '[[ -z `/usr/bin/pgrep -f asterisk` ]] && /etc/rc.d/asterisk restart>/dev/null 2>&1 |
| 03:58.29 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 03:58.50 | p3nguin | Well, the -f might not be a good idea for "asterisk" alone. |
| 03:59.10 | p3nguin | * * * * * /bin/sh -c '[[ -z `/usr/bin/pgrep asterisk` ]] && /etc/rc.d/asterisk restart>/dev/null 2>&1 |
| 03:59.27 | SunTsu | er, pgrep ... && ... |
| 03:59.28 | p3nguin | I think I like the pidof method just as much, though. |
| 04:01.07 | dijib | ok i have a linux nuub question that pertains to *, how do i run asterisk daemon on boot. /etc/init.d/asterisk is there and linked to rc.d |
| 04:01.31 | CoffeeIV | p3nguin: of course . . . and thanks ! |
| 04:01.52 | p3nguin | dijib: How will depend on your distro. |
| 04:02.03 | dijib | centos |
| 04:02.08 | dijib | know? |
| 04:02.26 | p3nguin | chkconfig asterisk on, I think |
| 04:02.53 | dijib | i thought i had done that then. |
| 04:03.02 | dijib | did it again. how do i check that ? |
| 04:03.05 | p3nguin | chkconfig --list |
| 04:03.09 | dijib | k |
| 04:03.27 | p3nguin | I'm not a CentOS user, so my commands may be slightly off. |
| 04:03.35 | dijib | your the man! |
| 04:03.42 | p3nguin | Learn tab completion if they aren't spot on. |
| 04:03.54 | dijib | i only used it cuz it had a minimal server @240MB iso |
| 04:04.09 | dijib | what do you use? |
| 04:04.27 | p3nguin | primarily Arch Linux |
| 04:05.02 | dijib | i like with cent that i can use redhat rpm's |
| 04:05.10 | p3nguin | But I'm not afraid of others; I just sometimes don't remember all commands' syntax. |
| 04:05.15 | dijib | althought i only use svn |
| 04:05.37 | dijib | and yum |
| 04:06.28 | kaldemar | CoffeeIV: there is a script called safe_asterisk in the contrib/scripts dir that does what you want. |
| 04:07.05 | p3nguin | What method does safe_asterisk use to detect the failure and try to restart it? |
| 04:07.06 | dijib | who was that for? |
| 04:10.09 | dijib | anyone know how i stop this from flooding CLI? > doing dnsmgr_lookup for 'toronto2.voip.ms' |
| 04:11.42 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:13.07 | kaldemar | p3nguin: see the run_asterisk function in the script |
| 04:14.07 | p3nguin | dijib: Turn core verbose to something less than 4. |
| 04:15.14 | dijib | seems to be looking up an awful lot. so core verbose 379 is too high eh? |
| 04:15.52 | p3nguin | It doesn't get any more verbose as you go above 4. |
| 04:17.53 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:17.57 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 04:20.08 | CoffeeIV | kaldemar: thanks, I'll check that out as well |
| 04:24.07 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
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| 04:43.14 | CoffeeIV | by the way, I got my * compiled with debug options and produced a core and backtrace and bug report, for the problems I complained about in here earlier today: https://issues.asterisk.org/jira/browse/ASTERISK-18349 |
| 04:52.39 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 04:58.56 | dijib | anybody still up? |
| 04:59.51 | Maliuta | Still? it's only just gone 3pm |
| 05:00.02 | dijib | 1am here |
| 05:00.15 | Maliuta | well then, you're in the wrong tz |
| 05:00.17 | Maliuta | ;P |
| 05:00.20 | dijib | must be |
| 05:00.45 | dijib | i want to know about MOH & streamplayer |
| 05:00.48 | dijib | how do? |
| 05:01.17 | Maliuta | I haven't done that. I just use MP3's |
| 05:02.22 | dijib | ok.. then mp3 question. how do i get it to switch songs hitting a digit |
| 05:02.23 | dijib | ? |
| 05:02.47 | dijib | digit=8 doesnt seem to work |
| 05:07.26 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:08.12 | kaldemar | dijib: the digit option only changes to the class, it does not switch a file that is plays. you'd have to change to a different class to change a song. |
| 05:08.37 | dijib | hmmm |
| 05:08.40 | dijib | ty |
| 05:09.37 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 05:09.38 | kaldemar | which means that using the digit option, you'd have to define two classes with the same dir and different digit and switch between them. quite a poor solution from the users' point of view. |
| 05:10.37 | kaldemar | i've made my own jukebox extension to get a new random song on a digit. :P |
| 05:12.50 | *** join/#asterisk StaRetji (~BigAll@46.12.157.16) |
| 05:14.15 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 05:14.54 | *** join/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 05:21.08 | dijib | kaldemar, hows that jukebox work? |
| 05:23.01 | Maliuta | you put a coin in a then select the appropriate number of songs you want to here? ;P |
| 05:25.05 | kaldemar | dijib: it takes a random file from a dir and plays it with Background. one digit in the context returns and others get a new random file. |
| 05:26.21 | kaldemar | i used applications Set, Background, While, EndWhile and Goto and functions RAND and SHELL. While and EndWhile are for a loop to prevent getting the same file two times in a row. |
| 05:27.24 | kaldemar | i can pastebin it if you think you'd find use for it. |
| 05:29.24 | dijib | YES PLEASE |
| 05:29.25 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 05:32.41 | kaldemar | dijib: http://pastebin.com/dPPUhbNd |
| 05:33.52 | dijib | dude thanks |
| 05:34.31 | dijib | i just got mpg123 to stream online radio and i have defined digits 1-9&0 to switch radio stations |
| 05:34.35 | dijib | asterisk you rock |
| 05:45.02 | f2knight | I have been tweaking out a FastAGI script.. but as I am finishing it up I am realizing something... FastAGI has no methods for encrypting the data... so If I am doing a FastAGI script to process a credit card.. How can I assure that information is at the verly least encoded until it hits my server? |
| 05:54.51 | *** join/#asterisk irroot (~irroot@41.122.163.3) |
| 05:57.01 | f2knight | well if anyone was wondering .. looks like the SVN trunk has AES_ENCRYPT/DECRYPT and BASE_64 ENCODE/DECODE |
| 06:01.39 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:11.09 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
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| 06:11.45 | *** join/#asterisk StaRetji (~BigAll@46.12.157.16.dsl.dyn.forthnet.gr) |
| 06:13.07 | kaldemar | f2knight: they are in releases already. |
| 06:16.38 | kaldemar | f2knight: AES functions since 1.6.2 and BASE64 functions since 1.4.0. |
| 06:17.56 | f2knight | kaldemar, I had the modules not loaded. :-( Fail on me. |
| 06:21.41 | f2knight | kaldemar, Do I need to format the KEY first? |
| 06:22.07 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:22.13 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 06:26.10 | kaldemar | f2knight: format it? just feed the function 16 characters of your choice and it'll be happy. |
| 06:27.53 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 06:33.55 | f2knight | kaldemar, I think thats what it was .. very picky about the 16char key |
| 06:40.04 | Karen_m | I want to track who is calling, maybe in the 'name' field, have it show : office - 403.123.1233 |
| 06:40.08 | Karen_m | When I dial someone from exten => 123,n,Dial(SIP/voipms/1403...), is there a way to send a dynamic caller-id to myself? When I receive calls, and forward to myself it sends me the caller id of voip phone, but no idea who is calling |
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| 08:42.05 | mmbash | hi all |
| 08:43.25 | ChannelZ | mmmmultipass |
| 08:46.29 | mmbash | i have question about timing test |
| 08:47.42 | mmbash | customer wants to see that the server (vm) passes the dadhdi test |
| 08:48.59 | mmbash | right now it does not using centos 6 and latest dahdi/asterisk |
| 08:49.17 | Chainsaw | mmbash: Sub-second timer stability and VM generally do not mix. |
| 08:49.38 | Chainsaw | mmbash: Your customer is right to insist. Do not virtualise Asterisk. |
| 08:51.38 | ChannelZ | unless you don't care about conferences |
| 08:52.55 | Chainsaw | Or stability under high load. |
| 08:53.08 | mmbash | yes i did read about this problems but he wants a vm. |
| 08:53.35 | Chainsaw | mmbash: He wants stable timers or he wants a VM. Have him make a choice. |
| 08:56.20 | mmbash | is it really impossible? i found some interesting threads like http://pbxinaflash.com/forum/showthread.php?t=9084&page=10 |
| 08:56.57 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-203-129.w83-203.abo.wanadoo.fr) |
| 08:57.04 | merlin8282 | Hiho ! |
| 08:58.04 | merlin8282 | What if I set the dial timeout to 0 ? Does Dial immediately exit, or does it wait indefinitely ? |
| 08:58.19 | kaldemar | merlin8282: did you try it? |
| 08:58.36 | merlin8282 | kaldemar: maybe it's worth I try it, if you say it :) |
| 08:58.42 | WIMPy | Or 136 years? |
| 08:59.17 | merlin8282 | WIMPy: if the server doesn't crash :p |
| 08:59.47 | mmbash | or this post http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/this-is-scaring-the-hell-out-of-me#comment-39648 |
| 09:04.19 | mmbash | but my actually question is about this https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces |
| 09:04.31 | merlin8282 | kaldemar: WIMPy: WARNING[20771]: app_dial.c:1849 dial_exec_full: Invalid timeout specified: '0'. Setting timeout to infinite |
| 09:04.58 | mmbash | from my understanding asterisk by default is not even using dahdi when it comes to timing |
| 09:07.14 | kaldemar | mmbash: default timing resource depends on the used version. timerfd or dahdi. |
| 09:07.38 | mmbash | CLI says res_timing_timerfd.so 1 but the dahdi_test is using the dahdidummy, right? |
| 09:07.44 | kaldemar | mmbash: "timing test" will show which timing module your asterisk uses. |
| 09:09.06 | mmbash | kaldemar: timerfd |
| 09:10.24 | kaldemar | dahdi_test uses the dahdi timer. |
| 09:11.58 | kaldemar | it doesn't care about which resource asterisk uses. |
| 09:12.09 | mmbash | so bad results with dahdi_test does not mean asterisk will perform bad because it uses timerfd |
| 09:14.38 | mmbash | is there a way to test the timerfd? |
| 09:17.40 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 09:17.53 | *** part/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-203-129.w83-203.abo.wanadoo.fr) |
| 09:19.59 | *** join/#asterisk timahvo1 (~rogue@41.215.82.210) |
| 09:35.22 | catphish_ | is it possible for a modem to work through an ATA using PCM? |
| 09:38.21 | Chainsaw | catphish_: You generally want T38 for that. |
| 09:41.21 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 09:42.15 | *** join/#asterisk slash213|wrk2 (slash213@80.90.117.111) |
| 09:42.18 | slash213|wrk2 | hi guys |
| 09:42.33 | slash213|wrk2 | how can i disallow 302 moved temporarily globally? |
| 09:42.39 | slash213|wrk2 | i want * to ignore it completely |
| 09:42.55 | WIMPy | catphish_: In theory it could, but even on a LAN I wouldn;t expect it to work. |
| 09:44.05 | catphish_ | that's a shame, does t38 support any modem conversations? or just t30 fax? |
| 09:44.59 | catphish_ | my problem is that i have a device with an analog modem that needs to phone home sometimes |
| 09:45.02 | WIMPy | Well, fax is modem, unless it's G4 fax. |
| 09:45.21 | WIMPy | But I haven't heard of any non-fax useage. |
| 09:46.17 | *** join/#asterisk zkn (~zkn@193.90.201.218) |
| 09:46.34 | catphish_ | is it easy to enable t38 passthrough on asterisk 1.8? |
| 09:47.48 | zkn | HI, Could someone explain briefly what does the following CLI message mean: SIP/<username> connected line has changed. Saving it until answer for SIP/<trunkname> |
| 09:48.23 | zkn | and what are the implications? |
| 09:50.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 09:50.32 | *** join/#asterisk Atriks (~Hatrix@nid67-1-78-229-20-103.fbx.proxad.net) |
| 09:50.40 | WIMPy | zkn: Look for COLP |
| 09:50.44 | kaldemar | a new connected line identification. if a caller A calls B and B for example transfers the call to C, A gets new connected line information (C) and sees who is on the other side. |
| 09:54.26 | slash213|wrk2 | okay, "i" option in Dial procedure did it for me |
| 09:56.28 | *** join/#asterisk trhodes (tom@cardinal.lizella.net) |
| 10:01.22 | zkn | ok, thanks WIMPy and kaldemar |
| 10:07.00 | *** join/#asterisk Atriks (~Hatrix@2a01:e34:ee51:4670:7cba:30f1:b741:d30b) |
| 10:09.39 | StaRetji | Folk, is there a way to edit: The person at extension 123 is on the phone, please leave your message after the tone. When done, hang up or press the pound key. |
| 10:09.53 | StaRetji | from not pronouncing extension number |
| 10:11.33 | StaRetji | maybe text is not precise, but I would just like to edit that script not to pronounce some words. I have did numbers and it doesn't look okay when asterisk pronounces this when some calls and I'm not available |
| 10:12.10 | StaRetji | for, it is enough Please leave your message after the tone. When done, hang up or press the pound key :) |
| 10:15.07 | *** join/#asterisk mutex7c (~mutex7c@212.184.118.18) |
| 10:16.46 | StaRetji | any help would be highly appreciated |
| 10:20.11 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 10:29.59 | Faustov | is there any way to limit verbosity for a specific module only? |
| 10:31.50 | kaldemar | StaRetji: if you have a non-numeric name for the voicemail box, it only says "the persion at extension is at the phone" |
| 10:32.03 | kaldemar | so no extension number is announced. |
| 10:32.27 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 10:32.43 | irroot | finally getting round to asterisk bussiness today some fixes in t38 gateway coming up |
| 10:40.50 | StaRetji | kaldemar: thx so much!!! |
| 10:43.14 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-203-129.w83-203.abo.wanadoo.fr) |
| 10:43.20 | StaRetji | kaldemar: do you know, by any chance, if is it possible to make following scenario: of 5 calls, 4 goes to voicemail directly while 5th listen to a Playback. Or in other word 2 of 10, 20% of calls listens to Playback and 80% are redirected to Voicemail? |
| 10:43.24 | StaRetji | thx :) |
| 10:44.27 | timahvo1 | anyone know where I can download x-lite 3.0 for linux from ? |
| 10:44.41 | Faustov | is there any way to limit verbosity for a specific module only? |
| 10:44.43 | Faustov | erm |
| 10:44.44 | Faustov | sorry |
| 10:44.53 | kaldemar | StaRetji: sure. use GROUP and GROUP_COUNT functions to count the calls to the extension and to check the number of calls to it, then send anything above 4 to Playback instead of VoiceMail. |
| 10:44.54 | Faustov | http://www.counterpath.net/index.php |
| 10:44.58 | Faustov | timahvo1: ^ |
| 10:45.15 | kaldemar | StaRetji: assuming you meant 5 simultaneous calls. |
| 10:45.39 | timahvo1 | Faustov: thanks |
| 10:48.11 | StaRetji | kaldemar: No, but this isuseful too, Thx man, your PRO!!! |
| 10:48.51 | Faustov | you're |
| 10:49.36 | StaRetji | you're |
| 10:49.51 | StaRetji | yep, sorry, holding baby |
| 10:50.09 | StaRetji | and he likes the keyboard very much |
| 10:50.20 | Chainsaw | An engineer in the making. |
| 10:50.45 | StaRetji | :) |
| 10:56.31 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 11:06.01 | merlin8282 | I have a project where the customer wants to be able to connect a softphone through the internet to the asterisk server, and he wants the connection to be encrypted. Which would be the best way to do this ? Asterisk 1.6.2.9, using only SIP. |
| 11:07.45 | kaldemar | merlin8282: 1.6.2 does not support STRP (unless you backport) but only signaling encryption with TLS. so, if you're stuck with that version, try some VPN. |
| 11:08.19 | merlin8282 | kaldemar: Ok. Stunnel would also be ok, I suppose. |
| 11:27.08 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:34.58 | *** join/#asterisk tiuman (~ya-online@95.167.25.38) |
| 11:35.12 | tiuman | hi all |
| 11:36.36 | tiuman | my asterisk under nat connect to sip provider, dial all phones all very well. But not dial to GSM phones... |
| 11:38.35 | tiuman | from my phone call goes, as if summoned. But to a GSM phone call from a * does not come. |
| 11:40.15 | kaldemar | tiuman: ask your provider. asterisk does not care what the destination number is. |
| 11:44.29 | tiuman | may i published my log? |
| 11:44.34 | tiuman | kaldemar |
| 11:44.41 | tiuman | you look? |
| 11:44.51 | tiuman | please |
| 11:45.14 | kaldemar | go ahead and pastebin it. |
| 11:54.22 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 11:54.53 | tiuman | http://pastebin.ru/320350 |
| 11:54.58 | tiuman | kaldemar |
| 12:02.06 | tiuman | or http://pastebin.ru/320352 |
| 12:07.13 | kaldemar | all i see is a call to something that responds with busy. |
| 12:08.33 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:09.42 | tiuman | why, then, on my phone does not come Vulture BuSy? |
| 12:11.24 | awk | lol |
| 12:11.26 | awk | sorry |
| 12:11.34 | tiuman | ringing all gsm phones if gsm phone turned off too ringing. |
| 12:12.09 | *** join/#asterisk Atriks (~Hatrix@2a01:e34:ee51:4670:7cba:30f1:b741:d30b) |
| 12:12.11 | tiuman | why my sip not notice me about gsm phone busy |
| 12:13.02 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 12:13.31 | singler | you are using r parameter, so asterisk generates ringing, provider at that time probably anounces that phone is out of reach |
| 12:13.45 | tiuman | it time my gsm phone silent |
| 12:14.08 | singler | you said that you phone is off |
| 12:14.29 | singler | *your |
| 12:15.18 | tiuman | no matter if enabled or not enabled, calls will not come to the GSM |
| 12:15.38 | singler | remove r parameter from Dial() and then check |
| 12:16.08 | tiuman | <singler> where it parametr find? |
| 12:16.54 | singler | it is in this paste http://pastebin.ru/320352 line 233 |
| 12:17.03 | singler | find that line in etensions.conf |
| 12:17.08 | singler | *extensions.conf |
| 12:18.35 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 12:22.49 | tiuman | singler http://pastebin.ru/320353 in any of these files, what to look for? |
| 12:24.36 | singler | rung "grep macro-dialout-trunk" /etc/asterisk/* |
| 12:24.45 | singler | run "grep macro-dialout-trunk /etc/asterisk/*" |
| 12:25.33 | tiuman | run: command not found |
| 12:25.39 | tiuman | CentOS |
| 12:25.52 | singler | run is not part of the command :) |
| 12:26.29 | tiuman | OK ^)) |
| 12:27.42 | tiuman | find [macro-dialout-trunk] |
| 12:27.57 | tiuman | what i do? |
| 12:28.04 | singler | in which file? |
| 12:28.22 | tiuman | extension_additional.conf |
| 12:28.54 | tiuman | it is TrixBox |
| 12:29.05 | singler | open that file, find [macro-dialout-trunk] and somewhere below of that should be Dial(<..>) |
| 12:29.29 | *** join/#asterisk james_zhu (~Administr@113.97.180.46) |
| 12:29.35 | tiuman | ok |
| 12:30.26 | tiuman | exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM},300,${DIAL_TRUNK_OPTIONS}) |
| 12:31.30 | singler | backup that line and remove ${DIAL_TRUNK_OPTIONS} |
| 12:32.13 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 12:32.55 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 12:35.39 | tiuman | exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM},300,) |
| 12:36.12 | tiuman | ok |
| 12:36.16 | tiuman | do it |
| 12:37.15 | tiuman | restarting asterisk and dialling |
| 12:37.45 | tiuman | Nothing has changed |
| 12:38.30 | singler | you still hear ringing? |
| 12:38.44 | tiuman | yes |
| 12:38.53 | tiuman | bot gsm phone silent |
| 12:38.57 | tiuman | but gsm phone silent |
| 12:39.04 | singler | can you pastebin log without sip debug? |
| 12:39.11 | tiuman | ok |
| 12:42.21 | tiuman | http://pastebin.ru/320354 |
| 12:43.23 | singler | what phone do you use? |
| 12:43.50 | tiuman | 3CXPhone |
| 12:44.09 | tiuman | write Destination Busy |
| 12:45.01 | tiuman | ringing after 30-40 seconds |
| 12:45.55 | singler | I think that during ringing provider announces reason of the problem, but you phone or asterisk generates ringing, so you do not hear it, Linksys SPA phones have "sticky 183"(?) option, which help to hear early media |
| 12:46.01 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 12:46.13 | singler | you could try capturing packets and replaying with wireshark |
| 12:47.18 | tiuman | ok |
| 12:47.30 | tiuman | mo my gsm phone not BUsy |
| 12:47.48 | tiuman | But my gsm phone not busy |
| 12:49.28 | singler | maybe you are not allowed to dial mobile phones or maybe number is incorrect |
| 12:49.30 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-cflbpligeshhfczn) |
| 12:49.36 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:51.54 | *** part/#asterisk james_zhu (~Administr@113.97.180.46) |
| 12:52.13 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 12:52.39 | tiuman | ok i am be dial +79142738971 |
| 12:53.03 | BenC[UK] | guys how do I kick a sip user? |
| 12:53.28 | singler | as any other user? :) |
| 12:54.15 | BenC[UK] | from cli |
| 12:54.31 | WIMPy | But for legal reasons I'd advise, not to wear shoes when doing so. |
| 12:54.43 | singler | :D |
| 12:54.51 | singler | BenC[UK]: I guess you should rephrase your question |
| 12:55.01 | BenC[UK] | lol |
| 12:55.08 | BenC[UK] | I have a user 2010 signed in.. |
| 12:55.14 | BenC[UK] | the queue thinks hes online, but hes not |
| 12:55.21 | BenC[UK] | and keeps trying to send calls to it |
| 12:55.22 | BenC[UK] | and failing |
| 12:55.29 | WIMPy | BenC[UK]: You are aware, that SIP is stateless? |
| 12:55.34 | BenC[UK] | yes |
| 12:55.51 | BenC[UK] | but the queue is still trying to send it to 2010 |
| 12:55.55 | BenC[UK] | when I try.. |
| 12:55.59 | BenC[UK] | queue remove member 2010 from 7001 |
| 12:56.07 | BenC[UK] | I get an error saying the users not in the queue |
| 12:56.15 | BenC[UK] | but I can see clearly in the logs its trying SIP/2010 |
| 12:56.36 | singler | "<..> member SIP/2010 <..>" maybe? |
| 12:56.45 | singler | use tab-complete |
| 12:57.20 | coppice | yeah, SIP is tasteless... oh, stateless |
| 12:59.23 | *** join/#asterisk JCCash (JCCash@200.205.212.102) |
| 13:01.23 | *** join/#asterisk VoipForces (~Adium@69.165.199.195) |
| 13:02.47 | VoipForces | Hi all, anyone knows if DAL/LDT1 circuits have a telco side setting where it hangsup after 30s if no answered? I send a Dial with a 40s timeout and they seem to timeout at 30s... |
| 13:03.22 | *** join/#asterisk SunTsu_ (miyamoto@unaffiliated/suntsu) |
| 13:05.11 | *** join/#asterisk serafie (~erin@nat/digium/x-cgkrdunwhxrtcqbi) |
| 13:09.20 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:09.20 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:09.49 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:10.14 | BenC[UK] | [Aug 26 09:08:50] WARNING[18070] chan_sip.c: sip_xmit of 0x7f178434c8e0 (len 809) to 0.0.7.218:5060 returned -1: Invalid argument |
| 13:10.22 | BenC[UK] | I keep seeing this where it should be trying to call agents |
| 13:12.11 | wdoekes2 | you're trying to call account 2010 |
| 13:12.55 | BenC[UK] | how can you see that? |
| 13:13.09 | wdoekes2 | but asterisk doesn't know it's an account (for whatever reason) |
| 13:13.27 | wdoekes2 | quick guess tells me that 7*256+218 is 2010 |
| 13:13.40 | VoipForces | BenC[UK]: to 0.0.7.218:5060 <â this looks like an invalid IP... |
| 13:13.46 | *** join/#asterisk irroot (~irroot@41.122.176.139) |
| 13:14.00 | BenC[UK] | lo, nice |
| 13:14.01 | BenC[UK] | lol |
| 13:14.09 | BenC[UK] | thats my fault then |
| 13:14.37 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:14.42 | BenC[UK] | I removed the account from users.conf |
| 13:14.49 | BenC[UK] | but didnt remove it from the queue |
| 13:16.52 | *** join/#asterisk JCCash (~JCash@200.205.212.101) |
| 13:17.26 | *** join/#asterisk robertj (~Rob@dumbledore.athenshousing.org) |
| 13:21.29 | robertj | If I want to move my 706-xxx-xxxx numbers to a sip trunk, does that mean I need to have the sip trunking done by someone in my local calling area? |
| 13:23.07 | _Corey_ | roberyj: Nope |
| 13:23.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:23.46 | robertj | am I going to end up paying additional fees for incoming calls then or is the price the same no matter where the trunking is done at |
| 13:24.04 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:24.06 | _Corey_ | generally, it doesn't matter... |
| 13:24.34 | robertj | and is there a go-to list of services that are big players, cheap, asterisk-friendly, and enterprise ready? |
| 13:24.58 | _Corey_ | Define "enterrprise ready" |
| 13:25.13 | _Corey_ | Here is a general list: |
| 13:25.16 | _Corey_ | ~itsplist-us |
| 13:25.17 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 13:25.29 | BenC[UK] | grr, this queue lark is realling annoying me |
| 13:25.30 | BenC[UK] | http://pastebin.com/XJS9NWmn |
| 13:25.31 | robertj | _Corey_, spending all the extra crash for that extra 9 on the -uptime, knows how to send an invoice and answer the phone ,etc |
| 13:25.47 | BenC[UK] | this one here,.. I have 10 agents waiting for calsl (out of 10) - and it doesnt try any of them!? |
| 13:26.41 | _Corey_ | robertj: Well, your experience may vary but most of those listed have a fairly good reputation. When I need "enterprise" class service I usually engage Level3 or Global Crossing however |
| 13:26.55 | BenC[UK] | http://pastebin.com/raw.php?i=XJS9NWmn < - can see it better there |
| 13:30.08 | robertj | _Corey_, yeah, you can tell those are enterprise solutions because they have worthless web pages :P |
| 13:30.22 | _Corey_ | ? |
| 13:30.32 | tiuman | how it is http://pastebin.ru/320355 ??? |
| 13:30.48 | BenC[UK] | any asterisk consultants here want to sort my queue issue? |
| 13:31.26 | robertj | _Corey_, medium and large-business websites are almost universally useless, they basically are $200k ads that say "Call us and if you pay us we will give you a logon to our real website" |
| 13:31.27 | tiuman | BenC[UK] what? |
| 13:31.42 | BenC[UK] | tiuman: what I just pasted |
| 13:32.06 | tiuman | unknown |
| 13:32.16 | _Corey_ | robertj: I don't know if "useless" is the word... they're not going for a retail experience. You need to talk to an account rep. |
| 13:32.20 | wdoekes2 | BenC[UK]: you have lots of agi stuff |
| 13:32.24 | tiuman | what my dialling declined??? |
| 13:32.29 | tiuman | <PROTECTED> |
| 13:32.36 | BenC[UK] | ignore the agi stuff though, its only writing logs |
| 13:32.57 | tiuman | ok |
| 13:33.02 | tiuman | sip debug? |
| 13:33.05 | BenC[UK] | its not calling any uses in the queue |
| 13:33.17 | robertj | _Corey_, local government account reps are bottom of the barrel :P |
| 13:34.48 | tiuman | http://pastebin.ru/320356 |
| 13:34.53 | _Corey_ | :) good luck |
| 13:35.04 | wdoekes2 | queue show 7001 ? |
| 13:35.49 | tiuman | what does this mean? |
| 13:36.04 | wdoekes2 | tiuman: your dahdi refused the call |
| 13:36.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:36.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:36.35 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 13:37.28 | tiuman | wdoekes2 sip provider used dahdi??? |
| 13:37.38 | tiuman | 3g modems? |
| 13:37.47 | wdoekes2 | your pastebin, line 101 |
| 13:38.11 | BenC[UK] | http://pastebin.com/azD366kz |
| 13:38.18 | BenC[UK] | wdoekes2: thats the queue show 7001 |
| 13:38.26 | BenC[UK] | the call you see at the bottom there |
| 13:38.30 | BenC[UK] | timed out in the queue |
| 13:39.13 | wdoekes2 | hm.. you would say that those "not in use" people should get the call |
| 13:39.45 | tiuman | 7001 queue in my provider??? |
| 13:40.24 | BenC[UK] | wdoekes2: hmm, yeah - that call said it ran 2011, but it didnt ring |
| 13:41.15 | wdoekes2 | BenC[UK]: that might be a different issue, sip set debug peer 2011 |
| 13:41.29 | wdoekes2 | tiuman: 7001 is BenC[UK]'s queue name |
| 13:41.32 | BenC[UK] | wdoekes2: it happens to any agent though |
| 13:41.36 | robertj | ok, las question for the day. If you were going to requisition yourself a 3G SIP handset I take it you would want a fancy-pants android phone? |
| 13:41.46 | BenC[UK] | I've see a few times it rings an agent for 10s then fails |
| 13:41.53 | tiuman | <wdoekes2>: What my dahdi refused yhe call??? |
| 13:41.59 | BenC[UK] | how do I set it so it rings an agent for 2 seconds, and then moves to the next? |
| 13:42.00 | wdoekes2 | BenC[UK]: yes.. that's the 'n' options |
| 13:42.07 | tiuman | <wdoekes2>: Why my dahdi refused yhe call??? |
| 13:42.16 | wdoekes2 | remove the 'n' option |
| 13:42.29 | wdoekes2 | tiuman: I know 0 about non-sip channel drivers |
| 13:42.31 | BenC[UK] | n is to show the caller its ringing |
| 13:42.37 | wdoekes2 | no it's not |
| 13:42.45 | BenC[UK] | hmm |
| 13:42.47 | wdoekes2 | # asterisk -nrx 'core show application Queue' | grep "'n'" 'n' -- no retries on the timeout; will exit this application and |
| 13:43.11 | BenC[UK] | oh man |
| 13:43.53 | BenC[UK] | whats the option to not let caller here ringing |
| 13:43.58 | BenC[UK] | or is there no ringing by defeault? |
| 13:44.16 | wdoekes2 | there is musiconhold by default |
| 13:44.21 | wdoekes2 | unless you don't Answer() before Queue() |
| 13:45.01 | wdoekes2 | in which case the call will probably drop in a minute |
| 13:45.04 | wdoekes2 | I think.. |
| 13:46.36 | BenC[UK] | ok cool |
| 13:46.51 | BenC[UK] | right so.. |
| 13:46.59 | BenC[UK] | which param do I set for 2 seconds per agent ? |
| 13:47.08 | BenC[UK] | (they have a client with auto answer, so should answer pretty instantly) |
| 13:47.26 | BenC[UK] | timeout=2 seems to be total timeout |
| 13:48.03 | wdoekes2 | ; How long do we let the phone ring before we consider this a timeout... |
| 13:48.03 | wdoekes2 | ; |
| 13:48.03 | wdoekes2 | ;timeout = 15 |
| 13:48.17 | wdoekes2 | you might want to look at the queues.conf.sample |
| 13:48.53 | BenC[UK] | hmm |
| 13:49.01 | BenC[UK] | I set it to 2 seconds just now |
| 13:49.10 | BenC[UK] | but, it rang for 2 seconds then failed the queue |
| 13:49.22 | BenC[UK] | exited the queue even |
| 13:49.42 | BenC[UK] | I want it to try agents for a total of 10 seconds, but 2 seconds each |
| 13:51.12 | wdoekes2 | if it exited the queue, you haven't removed the 'n' yet |
| 13:52.03 | irroot | BenC[UK] you will be loading the system quite a bit with that timing r |
| 13:54.50 | *** join/#asterisk DelphiWorld (~VoIpGuy@openvpn/user/DelphiWorld) |
| 13:54.51 | DelphiWorld | hi |
| 13:54.53 | DelphiWorld | see my pb: |
| 13:54.53 | DelphiWorld | http://dpaste.com/603186/ |
| 13:54.57 | DelphiWorld | is an issue about timing/moh |
| 13:55.00 | DelphiWorld | any help is welcome;) |
| 13:57.34 | irroot | DelphiWorld can perhaps help with that one |
| 13:57.51 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 13:57.56 | irroot | there is a ticket open where i have loaded a patch |
| 13:57.57 | DelphiWorld | irroot: please;) |
| 13:58.53 | DelphiWorld | irroot: what we can do ? |
| 13:59.24 | DelphiWorld | irroot: i'm getting lot of output that can't let me read at all |
| 13:59.48 | irroot | what version you using |
| 14:00.16 | DelphiWorld | irroot: let me verify |
| 14:00.49 | ferdna | why DelphiWorld as nick name? as in dolphin? |
| 14:01.01 | DelphiWorld | ferdna: why no |
| 14:01.16 | DelphiWorld | irroot: i think is still 1.4 |
| 14:01.29 | *** join/#asterisk LittleFool (~LittleFoo@over-dozed.com) |
| 14:01.29 | irroot | yeah figgures |
| 14:01.54 | *** join/#asterisk Nasga_ (~Nasga@186.212.10.93.rev.sfr.net) |
| 14:01.57 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 14:01.58 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 14:02.09 | ferdna | DelphiWorld, thought it had a special meaning or something |
| 14:02.12 | DelphiWorld | slaps Nasga around a bit with a large trout |
| 14:02.21 | irroot | have a patch for 1.8 |
| 14:02.21 | DelphiWorld | ferdna: do you know delphi ? |
| 14:02.36 | ferdna | DelphiWorld, the city? |
| 14:02.47 | DelphiWorld | ferdna: you're not a programer then. |
| 14:02.56 | *** join/#asterisk nW44b_ (~Schnitzel@unaffiliated/benwa) |
| 14:03.02 | ferdna | DelphiWorld, that was my 2 programming language |
| 14:03.04 | irroot | Delphi as in pascal :P |
| 14:03.06 | ferdna | and my favorite of all |
| 14:03.12 | ferdna | i have all versions =) |
| 14:03.16 | ferdna | since 3.0 |
| 14:03.59 | ferdna | from delphi i moved to C# and now to Qt =) |
| 14:04.14 | DelphiWorld | ferdna: QT... the shit world |
| 14:04.28 | ferdna | DelphiWorld, why the shit world? |
| 14:04.41 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 14:04.53 | ferdna | DelphiWorld, also i remember being active in a lot of delphi forums... look for ferdna =D |
| 14:04.58 | ferdna | that might be me |
| 14:05.00 | ferdna | :) |
| 14:05.44 | ijpalmer | Hi all, I'm trying to set up a phone system with 'forced calling' where an incoming gets presented to a handset, beeps in the agents ear then automatically connects, does anyone know how to do this thanks |
| 14:05.55 | VoipForces | Hi all, anyone knows if DAL/LDT1 circuits have a telco side setting where it hangs up after 30s if no answered? I send a Dial with a 40s timeout and they seem to timeout at 30s... |
| 14:08.06 | *** join/#asterisk serafie (~erin@nat/digium/x-lqhnnyfgphmttnbd) |
| 14:08.25 | irroot | DelphiWorld if you want the fix for 1.8 shout |
| 14:08.35 | DelphiWorld | irroot: ok |
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| 14:14.22 | p3nguin | ijpalmer: To me, that sounds like you'll want to use queues and agents. When an agent is logged in and a call comes into a queue, the call is giving to one of the agents who is standing by. |
| 14:14.36 | p3nguin | s/giving/given/ |
| 14:15.49 | ijpalmer | p3nguin: That's exactly it, I have the queues and agents setup ok, and the call goes to a specific handset. What I now want is the call to beep in the agents ear and be forcibly answered without them deciding |
| 14:15.54 | DelphiWorld | irroot: 1.6.2.18 |
| 14:16.09 | p3nguin | ijpalmer: It should already be doing that. |
| 14:16.32 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:17.09 | ijpalmer | np3nguin: maybe I have it wrong but the handset rings so the agent can decide to let it ring or answer it, I don't want them to have that choice, I just want it to beep in their ear and then the call is connected |
| 14:18.06 | p3nguin | ijpalmer: Can you explain to me how there is the possibility of any ringing when the agent is logged in? When the agent is logged in, the phone is constantly "on a call." Phones don't ring when they are on a call. |
| 14:18.44 | ijpalmer | p3nguin: this is probably the part I'm doing incorrectly then apologies |
| 14:19.12 | p3nguin | ijpalmer: Can you tell me how you are having agents log in? |
| 14:20.31 | ijpalmer | p3nguin: I don't think I'm having agents login, I just have queues and their members, so what I need to do is have agents logging in rather than adding members to queue |
| 14:20.41 | ijpalmer | ? |
| 14:21.41 | p3nguin | ijpalmer: You still have to add members, but I think that's where you went wrong. You're probably adding members that are either SIP devices or Local channels. You should be adding agents as members, then having the phones login to agentlogin(). |
| 14:22.33 | ijpalmer | p3nguin: ahh I think that is where I'm going wrong, my members are SIP devices |
| 14:23.25 | ijpalmer | p3nguin: does agentlogin work ok with realrime queues do you know? |
| 14:23.38 | p3nguin | ijpalmer: I have no idea about that. |
| 14:24.51 | ijpalmer | ok no worries, cheers for your help a bit more research me thinks :) |
| 14:25.54 | p3nguin | ijpalmer: Assign agent IDs to either people or phones. Then decide how a person will login. You can make it so a person picks a seat at any phone, picks up and dials a special code (such as *17), when prompted, enter his agent ID... or you can make it so a person picks up the phone and dials the code and the phone identifier logs into the agent. |
| 14:26.30 | p3nguin | ijpalmer: If you are keeping stats on agents, you'll probably want the former method as opposed to letting the phone do the login. |
| 14:27.16 | ijpalmer | p3nguin: yes the former method sounds best, as I do need stats |
| 14:27.30 | p3nguin | ijpalmer: When the agent logs in, he will be listening to musiconhold. When a call comes in, the queue will determine which member gets the call, and the agent will get that call. |
| 14:27.45 | *** join/#asterisk Dovid (~Dovid@float-gw.core-fw-t.mypbxmanager.net) |
| 14:27.55 | p3nguin | ijpalmer: When the call ends, agent is back to musiconhold again while he waits. |
| 14:28.00 | Dovid | hi. i have G729 liscences. i added a nic and now it wont work |
| 14:28.04 | Dovid | how do i fix this? |
| 14:28.14 | p3nguin | dovid: Reregister your licenses. |
| 14:28.35 | Dovid | where is the reg util? |
| 14:28.38 | *** join/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 14:28.54 | catphish | is there any way to verify what codecs asterisk is allowing for a sip call? |
| 14:29.10 | p3nguin | dovid: http://downloads.digium.com/pub/register |
| 14:29.10 | catphish | i have set disallow=all, allow=alaw but my calls are being rejected |
| 14:29.30 | p3nguin | catphish: sip set debug on |
| 14:29.31 | ijpalmer | p3nguin: ok, so you're saying that the agent has a channel open to asterisk permanently? |
| 14:30.03 | p3nguin | ijpalmer: When he is logged in, yes. The agent is just listening to music waiting on the call to be given to him. |
| 14:30.09 | *** join/#asterisk MarqueI (~Marquel@static.132.171.47.78.clients.your-server.de) |
| 14:30.20 | ijpalmer | p3nguin: ok thanks |
| 14:30.42 | Dovid | it says: Server response: 469 - Maximum Activations Reached. Please contact Digium Support |
| 14:30.53 | Dovid | anyone here that can help or i need to make a call? |
| 14:30.57 | p3nguin | dovid: Sounds pretty clear to me. |
| 14:31.08 | *** join/#asterisk fIorz (nobody@2001:1a50:503c::1) |
| 14:31.15 | p3nguin | You can only reregister twice, I think. |
| 14:31.36 | p3nguin | for a total of three regs before you have to call. |
| 14:31.54 | p3nguin | I could be off, and you can only rereg once before having to call. |
| 14:32.16 | p3nguin | Ideally, you won't change your NICs often and won't have to rereg. |
| 14:33.51 | catphish | p3nguin: thanks |
| 14:33.58 | catphish | Capabilities: us - 0x0 (nothing), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) |
| 14:34.04 | catphish | that's quite clearly not going to work |
| 14:34.32 | p3nguin | I would disallow=all, allow=ulaw, save, sip reload, make calls. |
| 14:35.00 | catphish | that's what i did |
| 14:35.07 | *** part/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 14:35.38 | catphish | p3nguin: fixed it, the problem was the order |
| 14:35.48 | catphish | (allow=alaw,disallow=all) |
| 14:35.48 | p3nguin | order of what? |
| 14:35.51 | p3nguin | oh |
| 14:35.56 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 14:36.02 | p3nguin | Yeah, you have to disallow=all first, then allow only the codec you want to use. |
| 14:36.28 | catphish | because i use config_mysql it didn't occur to me that the order of the columns would matter |
| 14:36.32 | catphish | clearly it does :) |
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| 14:40.07 | adyn | Is there a method to debug timing issues? Some command that can show where a timing issue might be happening? |
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| 14:53.01 | navaismo | morning |
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| 14:53.44 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:56.54 | BenC[UK] | I hey navaismo |
| 14:59.15 | navaismo | hi |
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| 15:00.08 | pigpen | Hi all. |
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| 15:11.56 | psilikon | Any reasons why I should choose astmanproxy over astproxy.pl? |
| 15:14.55 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
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| 15:23.39 | catphish | one exists and the other doesn't? |
| 15:25.04 | anonymouz666 | at 200 active calls, strange things happens |
| 15:25.10 | anonymouz666 | that's a lot for asterisk |
| 15:25.24 | anonymouz666 | even with 8 cores |
| 15:25.30 | idespinner | is astmanproxy still necessary? I though Asterisk fixed many of the load issues relating to the AMI |
| 15:27.21 | pigpen | anonymouz666, transcoding? |
| 15:27.26 | anonymouz666 | no sir |
| 15:27.26 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 15:27.55 | pigpen | compile from scratch (entire sys and asterisk) ? |
| 15:27.56 | engrxyz | does anyone has any idea on how to make some money with premium rate calls? |
| 15:28.11 | anonymouz666 | yes |
| 15:28.28 | pigpen | so what happens? |
| 15:28.42 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 15:29.18 | anonymouz666 | pigpen: it seems that channel core can't read as fast as channel driver writes in |
| 15:29.33 | anonymouz666 | so the system takes time to interpret dtmf on ivr, for example |
| 15:29.48 | anonymouz666 | and warning of excessive queue length... |
| 15:30.00 | *** join/#asterisk arstanj (~arstanj@175.144.49.176) |
| 15:30.58 | anonymouz666 | asterisk 1.4 - I wonder if 1.8 can handle that better |
| 15:35.19 | idespinner | Hey Digium guys, you should double check DTMF is working on your corporate PBX... :| |
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| 16:07.25 | treborsux | anyone know the tolerance on these polycom 560s if i have to use a power cord it has a niple for one. It says 48 volts at 200 miliamps |
| 16:07.56 | treborsux | with 30 at 600 work or am i being an idiot and looking to burn stuff up |
| 16:08.07 | treborsux | i guess ill have to ebay them |
| 16:12.34 | Chainsaw | treborsux: 30 is a bit on the low side, but it might work. |
| 16:13.14 | Chainsaw | treborsux: It is good that your power supply can supply more than 200 mA; that won't damage anything. |
| 16:14.48 | keith4 | it won't burn, but it just might not work |
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| 16:18.17 | Carlos_PHX_ | treborsux: Under-volting a device is usually more damaging than over-voltage. The total watts consumed must stay the same, so amperage necessarily goes up. Current is what destroys devices. It may or may not work, or may work for a few months and phones start frying. |
| 16:20.19 | psilikon | So, what is the official word is an AMI proxy still necessary? |
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| 16:35.23 | VoipForces | When a device says it required 48 volts at 200 miliamps, it means it will consume 200mA max @ 48v. If you use a 48v adapter that is able to provide only 100mA, your power adapter will burn up, not the device you are powering up. You can safly use a 1000mA 48v power supply in this contition. IT is the voltage that counts, the current ratting on a power supply is the maximum current that power supply is able to provide. |
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| 16:55.36 | p3nguin | voipforces: And what happens if you try to power it with 24V and the current drawn is 400mA? You don't think that extra amount of amperage could cause damage through the circuits? |
| 16:56.45 | p3nguin | Or try to power it with 12V and it draws 800mA through the circuitry. That can't be good if the ampacity is much lower than the current the device is attempting to draw. |
| 16:56.53 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 16:57.03 | VoipForces | 24v is just not sufficient if the device requires 48v. If it's a switching power supply inside the device it will simply not power up. If it's a linear power supply it will not be able to supply the correct voltage to the chips inside and the result will be the same. |
| 16:58.04 | VoipForces | I have never blown up devices by applying voltages lower that the spec. Applying voltage above the device spec voltage, there is 99.9% change you'll blow it up. |
| 16:59.36 | VoipForces | usually devices will allow +-5% of the specified voltage. |
| 17:00.23 | VoipForces | Note that most devices have internal fuses that protect against overvoltage. So usually it's the fuse that blow and not the electronics. |
| 17:01.02 | VoipForces | If that happens it's a simple mather of replacing the fuse (sometimes it's those glass one, sometimes it's a small 0603 SMT component. |
| 17:01.13 | Qwell | who uses fuses anymore? |
| 17:01.17 | Qwell | short it like a champ |
| 17:02.07 | VoipForces | Qwell: bought a lot of Aastra 9112 that were dead, indeed short the fuse works like a charm. BUT after it is shorted you have to make really sure you use the right adapter ! |
| 17:02.38 | p3nguin | I've almost never encountered a blown fuse from overvoltage. Maybe that's just due to the types of things I mess with. But it's usually something causing too much current draw to blow the fuse. |
| 17:04.05 | VoipForces | p3nguin: Yes a fuse is made to blow at a set A rating. But remember I = P/V so voltage is a factor |
| 17:04.43 | p3nguin | Too low of voltage could cause a fuse to blow. |
| 17:07.07 | VoipForces | p3nguin: Not really, it will blow the fuse of the adapter, not the device. Usually⦠But anyway, rule of thumb is keep the adapter voltage within +-5% of the device rated voltage and the current rating of the adapter equal or above the device required current. |
| 17:07.48 | p3nguin | If you lower the voltage, the current drawn goes up. If it goes up above the fuse's rating for long enough, why wouldn't it blow? |
| 17:14.00 | VoipForces | p3nguin: Well, depends a lot on the condition and design of the adapter and device. Remember mode adapter are linear power supply not switching type. |
| 17:22.40 | *** join/#asterisk Alborracho (~chatzilla@200.25.216.222) |
| 17:22.49 | Alborracho | Hi everyone |
| 17:23.28 | Alborracho | does anyone know how to get the addicional parameters in a Perl Agi script? from this in extensions.conf exten => s,n(dial1),DeadAGI(tts/service.pl,${id1},${id2},${id3}) |
| 17:23.34 | Alborracho | im using asterisk 1.4 |
| 17:23.38 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 17:24.02 | Alborracho | i can get all the agi_ variables but this addional parametrs i cant |
| 17:24.56 | navaismo | nice nick Alborracho and nope i dont know how |
| 17:26.05 | ChannelZ | I think they come in as app arguments, not in the AGI variables list |
| 17:26.58 | ChannelZ | so in perl, I don't remember what cryptic variable name those comes in as.. $@ or something silly |
| 17:27.01 | Alborracho | i tried $_[0] |
| 17:27.06 | Alborracho | but no luck :( |
| 17:27.24 | *** join/#asterisk wesphillips (~wphill04@216.130.252.100) |
| 17:27.54 | Alborracho | ill try this $argc = @ARGV; |
| 17:27.54 | VoipForces | Might help |
| 17:27.55 | rawburt | so Asterisk will start, but does not listen on port 5060 (even though I do bindport in sip.conf). any thoughts? |
| 17:27.55 | VoipForces | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-9-SECT-2.html |
| 17:28.19 | VoipForces | rawburt: netstat -nlp ? |
| 17:28.29 | rawburt | any only if I start it from the init.d script, not from asterisk -vvvgc |
| 17:28.52 | ChannelZ | well 'argc' would be argument COUNT, but yeah @ARGV should be the argument vectors |
| 17:29.03 | rawburt | VoipForces: nothing when I | grep 5060 |
| 17:29.21 | ChannelZ | rawburt: is chan_sip even loaded? |
| 17:30.00 | rawburt | ChannelZ: apparently not, I greped for chan_sip in my config directory... so I need to do this in modules.conf? |
| 17:30.09 | rawburt | I have autoload=yes |
| 17:30.11 | rawburt | that's it.. |
| 17:30.15 | p3nguin | module show like sip |
| 17:30.33 | ChannelZ | rawburt: the config file is sip.conf but chan_sip.so is the actual module |
| 17:30.37 | p3nguin | If it isn't loaded, try loading it with module load chan_sip.so |
| 17:30.42 | rawburt | 0 modules loaded |
| 17:30.46 | ChannelZ | do you have "sip" commands listed in "help" on the console? |
| 17:30.57 | rawburt | Unable to load module chan_sip.so |
| 17:30.58 | rawburt | Command 'module load chan_sip.so' failed. |
| 17:30.58 | rawburt | SIP channel loading... |
| 17:31.23 | ChannelZ | check your config |
| 17:31.26 | rawburt | core show help; nothing in there relating to SIP |
| 17:31.45 | rawburt | so it *should* be loading? should I pastebin my sip.conf? |
| 17:31.49 | ChannelZ | either it's parsing something and not loading or something else catastrophic is going on |
| 17:31.58 | VoipForces | rawburt: tail /var/log/asterisk/full when you load chan_sip |
| 17:32.03 | rawburt | I'm running 1.6 BTW |
| 17:32.08 | ChannelZ | s/something/something incorrectly/ |
| 17:32.17 | rawburt | VoipForces: no such file |
| 17:32.24 | VoipForces | rawburt: In 1.8 if you include an non-existant file in any config file it will do that |
| 17:32.37 | VoipForces | rawburt: chengr logger.conf so you have full log |
| 17:32.53 | rawburt | so add debug? |
| 17:32.55 | rawburt | or full |
| 17:33.03 | rawburt | I have warning,notice,error |
| 17:33.04 | VoipForces | rawburt: just uncomment full. |
| 17:33.32 | VoipForces | I usually use: |
| 17:33.32 | VoipForces | full => notice,warning,error,verbose |
| 17:34.20 | p3nguin | While you're doing that, if you'd pastebin your sip.conf we could take a look at it. |
| 17:34.54 | ChannelZ | Any references to burritos or hookers are a clue, those are not valid config options. |
| 17:34.57 | rawburt | http://pastebin.com/tb3AXsHh |
| 17:34.58 | Alborracho | @ARGV is the solutions ty very much guys |
| 17:35.38 | ChannelZ | make sure sip.conf is readable by your asterisk user if you don't run asterisk as root |
| 17:35.47 | rawburt | hmm, full still isn't created... |
| 17:35.57 | rawburt | maybe I just need to start over |
| 17:35.58 | ChannelZ | actually that goes for the modules too and lots of other things |
| 17:36.08 | rawburt | I'm running as root |
| 17:36.12 | rawburt | for nw.. |
| 17:36.22 | rawburt | no, I'm not :D |
| 17:36.30 | rawburt | haha |
| 17:36.38 | rawburt | and none of these files are readable by the user running Asterisk :) |
| 17:36.47 | ChannelZ | exciting |
| 17:36.52 | p3nguin | I guess that's a clue. |
| 17:38.03 | p3nguin | chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/log/asterisk /var/spool/asterisk /var/run/asterisk |
| 17:38.07 | rawburt | http://pastebin.com/rKiVn552 |
| 17:38.44 | ChannelZ | assumes he's running as an 'asterisk' user |
| 17:38.50 | *** part/#asterisk wesphillips (~wphill04@216.130.252.100) |
| 17:38.51 | p3nguin | As he should be. |
| 17:39.05 | rawburt | yeah, I is |
| 17:41.02 | rawburt | still no 5060 thogh :/ |
| 17:41.03 | ChannelZ | hmm well your extensions.conf is a little wrong, though I'm not sure that can prevent sip from loading. |
| 17:41.04 | rawburt | though(* |
| 17:41.09 | ChannelZ | Or maybe you've progressed on to a new problem |
| 17:41.16 | rawburt | maybe I need to RTFM |
| 17:41.25 | p3nguin | ~book |
| 17:41.25 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 17:41.49 | rawburt | ~buybookandteleporttomenow |
| 17:42.02 | VoipForces | rawburt: you should try installing from digium's RPM or dep. it works great |
| 17:42.02 | rawburt | your bot sucks! |
| 17:42.15 | rawburt | I apt-get installed it |
| 17:42.19 | p3nguin | You can buy it as an ebook, I think. |
| 17:42.25 | rawburt | so you're suggesting I get their deb? |
| 17:42.55 | VoipForces | rawburt: I'm a CentOS guy LOL If the Digium .deb work as good as the RPM via yum yes. |
| 17:43.30 | carrar | compile from source |
| 17:43.47 | p3nguin | That's how they made the deb and rpm. |
| 17:43.51 | rawburt | it looks like I'm going to have to compile from source if I want anything newer |
| 17:44.07 | rawburt | or use the source package, assuming it's in the archives |
| 17:45.00 | carrar | get it from the source |
| 17:46.01 | carrar | http://www.asterisk.org/downloads |
| 17:46.05 | ChannelZ | It's not as traumatic as it sounds |
| 17:46.14 | rawburt | OMG WHAT WILL I DO |
| 17:46.15 | carrar | and you'll have a better understanding |
| 17:46.17 | rawburt | make install. |
| 17:46.47 | rawburt | why are there so many damn configuration files? |
| 17:47.05 | p3nguin | Because asterisk is modular. |
| 17:47.08 | carrar | You just want to click on START? |
| 17:47.21 | VoipForces | rawburt: from source for learning/test/experiment/debug is great. For production, I think released RPM is the way to go. |
| 17:47.34 | rawburt | no, I don't want to even have to do that much carrar |
| 17:47.39 | carrar | even more so for production compile from source |
| 17:47.47 | rawburt | VoipForces: alright, think I'll do that.. |
| 17:47.47 | carrar | or make a rpm from your compiled source |
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| 18:11.45 | ectospasm | ~thebook |
| 18:11.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
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| 18:21.08 | [sr] | hi |
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| 18:33.54 | ChannelZ | ahoy |
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| 19:17.56 | mmbash | when it comes to dahdi_test. it is said that below 99.97% is bad. is it bad for the "average" result or for the "worst"? |
| 19:19.20 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 19:20.58 | keith4 | mmbash: average, i believe |
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| 19:23.18 | mmbash | so when i have wort 99,66% and average 99,99 i can go with this server? |
| 19:24.51 | *** join/#asterisk serafie (~erin@nat/digium/x-fvngundnehfyraik) |
| 19:25.16 | mmbash | seems so, thx |
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| 19:32.59 | blizzow1 | My asterisk services got rebooted yesterday and dropped every call in our call center. I'm trying to verify that the guys who provide our support did the rebooting of the service. It doesn't look like asterisk was restarted from /etc/init.d./asterisk or as a service restart command. I'm left thinking it's been done with the asterisk CLI/console. Is there a way to see the history on the asterisk console and/or is there a log of what has been accessing t |
| 19:33.00 | _Corey_ | I had like 12% once on a server with a driver issue... 99+ is fine |
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| 19:49.22 | VoipForces | blizzow1: asterisk -r upper arrow gives you the last commands that were performed. |
| 19:49.34 | VoipForces | blizzow1: what's theuptime of the server itself? |
| 19:50.44 | blizzow1 | VoipForces: CentOS server has been up for 69 days. /usr/sbin/safe_asterisk has been running since June18. |
| 19:52.37 | VoipForces | blizzow1: ok⦠asterisk -rx "core show uptime" |
| 19:53.01 | blizzow1 | 22 hours, 11 minutes, 55 seconds. |
| 19:53.10 | blizzow1 | Says it as reloaded at that time as well. |
| 19:53.17 | VoipForces | ok |
| 19:53.34 | VoipForces | cat ~/.asterisk_history |
| 19:53.44 | VoipForces | or in the user account root .asterisk_history |
| 19:54.18 | VoipForces | 22 hours does that correspond to the down period ? |
| 19:55.40 | blizzow1 | yeah, 22 hours ago was when service dropped. |
| 19:56.49 | blizzow1 | I see some messages in the .asterisk history saying "pri set debug on span 1" and "pri set debug off span1" |
| 19:56.57 | VoipForces | ok, then they probably did 'core restart' command |
| 19:57.23 | VoipForces | blizzow1: those only tuen on and off pri d-channel debuggin messages |
| 19:57.23 | blizzow1 | I see core show channels near the end, and shchs, , but nothing that says restart. |
| 19:57.55 | blizzow1 | I see a dahdi reload. or and up higher I see core restart now. |
| 19:58.02 | VoipForces | they could simply have donne a kill -s SIGKILL from back |
| 19:58.18 | blizzow1 | Is it possible to add timestamps to the asterisk cli history? |
| 19:58.22 | VoipForces | "core restart now" will do it |
| 19:58.41 | VoipForces | no. |
| 19:58.57 | VoipForces | well, there may be a trace in /var/log/asterisk/full |
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| 20:05.11 | afink | hello everyone, I am trying to troubleshoot outgoing calls via a SIP ITSP and I am getting the response -- Got SIP response 604 "Does not exist anywhere" back from 10.250.0.5 any one know what that means or how I can fix it? |
| 20:05.16 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 20:06.43 | mchou | so it looks like google talk started working again. Had to revert a patch I applied a few days ago |
| 20:06.44 | p3nguin | Are you registered to them? |
| 20:06.48 | p3nguin | afink: ^ |
| 20:08.06 | mchou | the patch worked for a few days and I guess google fixed or reverted to whatever config they had (which in turn broke the patch) |
| 20:08.30 | p3nguin | It was a brilliant plan, wasn't it?! |
| 20:08.48 | mchou | what plan? |
| 20:09.18 | p3nguin | To break it, make someone write a patch for it, allow almost everyone to apply the patch, then revert back to the way it was before everyone patched. |
| 20:09.45 | mchou | if you think google did that intentionally you're crazy |
| 20:10.01 | p3nguin | You're underestimating the power of teh googlze. |
| 20:10.15 | mchou | fwiw only asterisk didn't work. other gtalk clients worked just fine |
| 20:10.19 | p3nguin | oh |
| 20:10.24 | p3nguin | I did not know that. |
| 20:10.34 | mchou | right |
| 20:10.46 | p3nguin | I was just being facetious anyway. |
| 20:10.48 | mchou | more like chan_gtalk has "issues" |
| 20:12.21 | p3nguin | I had one client bugging me yesterday to patch for them, but I was just too busy to mess with it. Today, I was basically told, "nevermind, it works now." :/ |
| 20:12.59 | mchou | you can tell them you're a miracle worker |
| 20:15.48 | ChannelZ | Jesus Fixed My Asterisk |
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| 20:21.52 | VoipForces | reading on arp poisonning and is having a head ake |
| 20:23.18 | ChannelZ | Yeah, what with all the stomach pumping and mopping up |
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| 20:32.24 | carrar | poisonning via poison oak is much worse |
| 20:33.05 | VoipForces | lol |
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| 21:16.18 | BenC[UK] | hi guys, me again |
| 21:16.24 | BenC[UK] | still having troubles with my queue |
| 21:16.28 | navaismo | hi |
| 21:16.30 | navaismo | grrr |
| 21:16.36 | navaismo | same problem? |
| 21:16.46 | BenC[UK] | well, I think I've got further |
| 21:17.07 | BenC[UK] | leastrecent |
| 21:17.20 | BenC[UK] | in theory should call teh agent who was off a call longest ago, right/ |
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| 21:20.07 | navaismo | <PROTECTED> |
| 21:22.03 | BenC[UK] | ok, similar then, but it doesnt do that |
| 21:22.07 | BenC[UK] | earlier, I had an agent come on |
| 21:22.16 | BenC[UK] | and he got the next 10 calls |
| 21:23.49 | navaismo | mmm are you tried linear strategy |
| 21:25.47 | BenC[UK] | I havent yet |
| 21:25.55 | BenC[UK] | I have tried leastrecent and roundrobin |
| 21:26.04 | BenC[UK] | another issue I am having is.. |
| 21:26.08 | BenC[UK] | let me pastebin an example |
| 21:27.23 | BenC[UK] | http://pastebin.com/HNw1e3gh |
| 21:27.43 | BenC[UK] | if you ignore where the customer was sending DTMF during a recorded message |
| 21:27.50 | BenC[UK] | look where its trying to allocate the agent |
| 21:27.54 | BenC[UK] | its trying to call the same agent twice |
| 21:29.31 | BenC[UK] | why wont it move on |
| 21:29.57 | navaismo | no idea |
| 21:31.43 | navaismo | try with llinear: Rings interfaces in the order they are listed in the configuration file. Dynamic members will be rung in the order in which they were added. (new in 1.6) |
| 21:32.14 | BenC[UK] | am I wrong in thinking it should move on? |
| 21:32.19 | BenC[UK] | rather than keep trying the same agent? |
| 21:39.59 | navaismo | really dont know, i supossed it dhould move on too |
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| 22:09.39 | dijib | <PROTECTED> |
| 22:09.39 | dijib | <PROTECTED> |
| 22:10.07 | ChannelZ | <PROTECTED> |
| 22:11.09 | dijib | tinfoil hat seated. |
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| 22:47.42 | FreakWent | Hello everyone. I'm struggling a bit trying to get my head around asterisk. |
| 22:47.54 | FreakWent | Can I use it to control a 'slave' PABX? |
| 22:48.16 | p3nguin | Probably. What do you mean by control it? |
| 22:48.37 | FreakWent | Well there's a site with a PABX and handsets on desks. |
| 22:49.40 | FreakWent | It works well enough, except managing and controlling the PABX is a pain, it needs special software and USB or serial link. It's designed for telephony professionals, so they need to call a consultant in every time they want to change the menus or phone numbers. |
| 22:50.08 | jmwpc | Does anyone know of a firefox dialer plugin? I have tried Noojee but can't seem to get it working. I can dial using the originate command through the web inteface (http://asteriskserver/asterisk/manager) with the same credentials I am using with noojee. |
| 22:50.32 | FreakWent | But with all the hardware in place, I thought asterisk might be able to act as a drop-in replacement for the management software only. |
| 22:51.14 | p3nguin | It depends on what your management software does to manage the pbx. |
| 22:51.48 | FreakWent | MM.. It's a PAnasonic TDA30, accepts control by USB or serial link. |
| 22:52.21 | p3nguin | Right, but I'm saying that it depends on what you mean by MANAGE. Manage is such a vague term. |
| 22:52.27 | FreakWent | Ah okay. |
| 22:52.31 | FreakWent | use cases! |
| 22:52.39 | p3nguin | If it can be controlled by a phone, asterisk can probably do it. |
| 22:53.01 | p3nguin | If it requires voice and DTMF, a phone is a good interface. |
| 22:53.22 | FreakWent | 1) Someone goes on holiday so option 3 in the caller's menu needs to dial Roger instead of Andrew, so we need to change the mobile number it dials and connects the caller to. |
| 22:53.58 | *** part/#asterisk jmwpc (~jeremy@c-24-5-58-60.hsd1.ca.comcast.net) |
| 22:54.02 | FreakWent | 2) There's a big sale so we want to add a new option 5 in the menu and a new number. |
| 22:54.08 | p3nguin | If it can be done from a phone, asterisk can do it. Or just use asterisk, which is capable of handling that scenario quite well all by itself. |
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| 22:54.37 | FreakWent | 3) After a power cut, the PABX falls over and won't dial out on the voice menu any more, the alarm light comes on, what's wrong with it? |
| 22:55.01 | p3nguin | I realize you probably don't want to replace the PBX, but someday it will come down to that anyway. |
| 22:55.42 | FreakWent | Right, so I think I understand how asterisk talk to POTS, (at first I naively thought it would use modems in voice mode), but I'm not sure how it talks to the internal phones. |
| 22:56.30 | FreakWent | I'm dissapointingly ignorant on this one. Are the handsets typically vendor-specfic? The phones have a gazillion buttons, are these all part of nice standards that asterisk knows about? |
| 22:57.05 | p3nguin | The most common type of phone for use with Asterisk is a basic IP phone which talks SIP (or sometimes IAX2). |
| 22:57.34 | p3nguin | Your PBX probably uses what most people call "digital" phones. |
| 22:57.42 | FreakWent | Yeah I'm fine with that, but what if you're not interested in VOIP? All the existing hardware's in place, you see. |
| 22:57.48 | FreakWent | yes exactly. |
| 22:57.53 | p3nguin | These digital phones are not compatible with Asterisk at all. |
| 22:58.24 | p3nguin | Does your PBX talk SIP at all? Many do. |
| 22:59.10 | BenC[UK] | hey p3nguin , do you know much about queues? |
| 22:59.20 | p3nguin | I know some. |
| 22:59.32 | FreakWent | Right, that's what I Was getting at. So Asterisk really is VOIP/POTS only, is that the story? |
| 23:00.52 | p3nguin | Pretty much, yes. It has the ability to deal with a few media types, but VoIP and PSTN connectivty via special analog interface are the most common among people who come around this channel. |
| 23:01.18 | p3nguin | It has capabilties beyond what I use it for. |
| 23:01.53 | FreakWent | Okay, well I don't think a replacement of the desk handsets will fly at this stage. |
| 23:01.57 | FreakWent | Thanks a lot for your help! |
| 23:02.25 | p3nguin | A lot of people who have PBXs that do SIP will use Asterisk in conjunction with their legacy PBX. |
| 23:02.40 | BenC[UK] | Im having troubles getting queues to work properly |
| 23:02.58 | BenC[UK] | if I set timeout=3 and then Queue(7001,,,,10) |
| 23:03.02 | p3nguin | Basically let the pbx deal only with the phones, and let asterisk deal with call routing. |
| 23:03.29 | BenC[UK] | should that try the same, agent for 3 seconds at a time up to 10 seconds |
| 23:03.36 | BenC[UK] | or try a differnt agent for 3 seconds each time? |
| 23:03.45 | p3nguin | Any incoming calls would touch asterisk first, and then asterisk could send the call over to the PBX if a person is required. |
| 23:06.15 | p3nguin | That's probably how I'd deal with it if possible. That way the legacy PBX can run all those silly phone that don't work with anything else. |
| 23:06.38 | FreakWent | thanks P3nguin. |
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