| 00:03.14 | jozu | thanks for all, the problem is from my voip provider |
| 00:04.39 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 00:04.55 | grandpapadot | Hi all. In Asterisk 1.2, is it possible to log queues to a database similar to the CDR? |
| 00:05.33 | barhom | Strom_C: you said Z to match 1-9, what about 0-9? |
| 00:06.11 | *** join/#asterisk galeras (n=Martin@Dynamic-IP-cr20011843102.cable.net.co) |
| 00:07.18 | galeras | Dear sirs, anyone can give me some references about Linksys SPA901 phone?. Thanks in advance. |
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| 00:10.25 | grandpapadot | galeras: crap? |
| 00:10.34 | grandpapadot | Try Polycom or Aastra. |
| 00:12.23 | JT | grandpapadot: i wouldn't be that harsh |
| 00:13.33 | grandpapadot | Ok, fair. galeras - you'll have the best luck with Polycom phones with Asterisk since a ton of people use them and there is a lot of info online. |
| 00:13.47 | grandpapadot | The 330 is a great phone for the money. |
| 00:14.14 | Nukemizer | is there a way I can prevent the caller ID being re written with the system Caller ID when ACD sends a call out to a Cell phone agent ? |
| 00:15.03 | JT | grandpapadot: yet that does nothing to help him if he already has a linksys and is just wanting some tech info |
| 00:15.11 | grandpapadot | Ahh, then nevermind. |
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| 00:21.14 | galeras | you're right my first option is Polycom SoundPoint 320, Linksys is just to give to my client a cheap option. |
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| 00:54.21 | unbeatabl | hi |
| 00:54.43 | unbeatabl | if i want to create a service like skype, what would i need? can someone give me some pointers? |
| 00:55.21 | JT | millions of dollars |
| 00:55.29 | JT | market capitalisation - wait, that's priceless |
| 00:56.58 | unbeatabl | lets just say money is not the issue |
| 00:57.33 | unbeatabl | id like to have a service where you can call anyone in the world via skype type software |
| 00:57.37 | JT | the issue is if you want to be just like skype, it's too late |
| 00:57.41 | JT | skype is already here |
| 00:57.57 | unbeatabl | its okay |
| 00:58.13 | unbeatabl | we just doing this for experiment |
| 00:58.24 | JT | what do you actually want to do |
| 00:58.41 | unbeatabl | a free phone call to anyone in the world |
| 00:58.54 | rob0 | like FWD |
| 00:59.07 | JT | i don't think you actually mean anyone |
| 00:59.07 | rob0 | (anyone who is using that service) |
| 00:59.20 | unbeatabl | yes |
| 00:59.59 | unbeatabl | i mean to call landline phone |
| 01:00.18 | JT | then you are crazy |
| 01:00.25 | JT | that costs money to terminate |
| 01:00.33 | rob0 | Let's see, there's FWD, Gizmoproject ... um, to call from a softphone to PSTN? |
| 01:00.37 | JT | losing money sounds like a crap business plan |
| 01:00.47 | rob0 | yup, don't know of anyone who does that for free |
| 01:00.51 | unbeatabl | i already said money is not the issue |
| 01:01.05 | JT | even if you lose it all? |
| 01:01.26 | unbeatabl | yes |
| 01:01.57 | unbeatabl | this isn't a business purpose |
| 01:02.11 | unbeatabl | its for an inside corporate company purpose |
| 01:02.51 | JT | then i guess you will need a bunch of pstn gateways, sip or h.323 proxies, and maybe some softswitches/pbxes |
| 01:03.08 | rob0 | pretty basic stuff, no? |
| 01:03.47 | *** join/#asterisk icel (n=icel@63.78.162.82) |
| 01:03.50 | rob0 | not even that dreadfully expensive, depending of course on the scale. |
| 01:03.54 | unbeatabl | is pstn gateway , a service? |
| 01:04.05 | unbeatabl | where you subscribe to? |
| 01:04.08 | JT | no |
| 01:04.12 | JT | it's a piece of hardware |
| 01:04.26 | JT | whether you use someone else's or not is a business decision |
| 01:04.33 | JT | same goes for any part of the network |
| 01:04.52 | unbeatabl | okay so we can outsource this as a service if i wanted to? |
| 01:05.02 | km- | are we sure that unbeatabl is really asking the right question? |
| 01:05.17 | icel | Hey could some1 help me out for like 2 minutes, I switched * servers and I can't get zaptel/pri action on the new one. Same working configs and all. |
| 01:05.28 | km- | unbeatabl: are you wanting to link remote offices together via voip, or are you trying to do something like, have indian contractors that you can call through local indian phone lines via voip in the US? |
| 01:05.47 | km- | icel: check to make sure your zap hardware isn't sharing an IRQ with anything else |
| 01:06.09 | icel | km: it shouldn't be, but do you have to start zap manually or somethign? |
| 01:06.09 | km- | icel: check /proc/interrupts and make sure wctdm, wcfxo, or any of the wc modules aren't occupying an irq with something else. |
| 01:06.13 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 01:06.15 | rob0 | zaptel.conf ? |
| 01:06.16 | JT | yeah digium hardware is useless with irq sharing |
| 01:06.29 | km- | icel: you'd need to first modprobe the right module then run ztcfg to config the channels |
| 01:06.36 | km- | icel: if you haven't done that, it won't fire up. |
| 01:06.52 | icel | <PROTECTED> |
| 01:06.58 | icel | that is what the line looks like |
| 01:06.58 | JT | lol |
| 01:07.02 | icel | modules are loaded |
| 01:07.06 | JT | triple irq sharing |
| 01:07.06 | CrazyTux | Hi guys |
| 01:07.06 | JT | terrible |
| 01:07.10 | unbeatabl | our company have thousands of salesmen. we want them to be able to call anywhere in the world for free. we need the cheapeset mechanism to do so. thats how we found out about asterisk and need more info. |
| 01:07.19 | *** join/#asterisk klictel (n=klictel@modemcable159.7-200-24.mc.videotron.ca) |
| 01:07.25 | icel | JT: what should I do about it? |
| 01:07.27 | CrazyTux | TIMEOUT(digit)=# OF digits, hit, before it hits "invalid block" correct? |
| 01:07.27 | file | klictel: I emailed you! |
| 01:07.36 | Qwell | unbeatabl: asterisk doesn't let you "call anywhere in the world for free" |
| 01:07.36 | icel | is that the problem you think? |
| 01:07.42 | JT | unbeatabl: all they need to do is log in to your voip servers, not make another skype |
| 01:07.50 | unbeatabl | basically, we want to setup something like skype, or some type of phone cards system |
| 01:07.50 | CrazyTux | SET(TIMEOUT(digit)=#) rather |
| 01:07.52 | JT | and it can be free for the salesman |
| 01:07.59 | JT | as the company will pay termination costs |
| 01:08.01 | km- | icel: most bioses allow you to reserve the IRQ for a particular slot. Check that out. |
| 01:08.08 | unbeatabl | what do you mean by termination cost |
| 01:08.11 | JT | unbeatabl: it's no big deal, remote voip extensions |
| 01:08.12 | icel | km: thx, i appreciate it |
| 01:08.15 | JT | either that or callback |
| 01:08.23 | JT | unbeatabl: the cost to terminate a pstn call |
| 01:08.28 | km- | unbeatabl: our remote salesmen have sip phones on their desks |
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| 01:08.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:08.44 | km- | but if they're mobile, then the callback idea is legitimate |
| 01:08.55 | unbeatabl | hmm |
| 01:08.55 | km- | you want to keep your infrastructure centralized where possible so it's easier to manage |
| 01:09.15 | km- | get a huge wholesale voip provider and a huge fat internet pipe |
| 01:09.33 | JT | meh |
| 01:09.37 | km- | if the salesmen already have mobile phones, which they likely do, program their phone numbers into asterisk and make them extensions. |
| 01:09.42 | JT | screw wholesale voip, get some PRIs |
| 01:09.52 | km- | that is true |
| 01:09.56 | km- | lots of PRI's would also do the trick |
| 01:10.37 | km- | I really need to setup a chatporn service and get some 1-900 number termination |
| 01:10.37 | km- | hehe |
| 01:10.49 | km- | asterisk + porn, who wouldn't do it? |
| 01:10.57 | Qwell | who *hasn't* done it? |
| 01:11.01 | Qwell | err, I mean |
| 01:11.01 | km- | get some ugly girls online to use voip to talk dirty. |
| 01:11.12 | km- | heh, I thought this was a quasi-original idea |
| 01:11.20 | km- | qwell, if you've already implemented, for god sakes give me the number! |
| 01:11.20 | km- | hehe |
| 01:11.29 | km- | "What are you wearing, Qwell?" |
| 01:14.30 | *** join/#asterisk Strom_M (n=strom@m770e36d0.tmodns.net) |
| 01:14.44 | km- | fine, it was internally funny to me at least |
| 01:17.30 | polerin | http://pastebin.ca/800025 anyone willing to take a look at my first attempt at a custom macro? intended to be used to manage IVR prompts, not for general use. |
| 01:18.01 | polerin | no specific problem with it (yet) just looking for general thoughts if anyone sees something thats bad form or not going to work ;p |
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| 01:20.15 | unbeatabl | what is a PRI |
| 01:20.31 | unbeatabl | and how much do they cost |
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| 01:21.23 | JT | primary rate interface |
| 01:21.27 | JT | digital phone circuit |
| 01:21.54 | JT | most businesses that have more than a dozen employees use them |
| 01:24.03 | JT | how much they cost depends on your location and the telco |
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| 01:27.34 | km- | pri's contain 23 telephone lines on 2 pairs. Around here (philly suburbs) you can get a PRI for $300-400 a month. |
| 01:28.15 | unbeatabl | okay so basically, i dont even need Asterisk to do what I need is what you are saying? |
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| 01:30.00 | km- | you would need something to terminate those PRI's, like asterisk or a professional PBX at your company. |
| 01:30.11 | km- | asterisk + a PRI card, rather. |
| 01:30.39 | unbeatabl | hmm what about VOIP service |
| 01:30.53 | JT | err |
| 01:31.09 | unbeatabl | wont i need those too? |
| 01:31.14 | JT | km-: T1 PRIs are 23 channels for voice |
| 01:31.27 | JT | and they can be delivered over 1 or 2 pair |
| 01:31.48 | unbeatabl | can we run this over fiber as well? |
| 01:32.00 | unbeatabl | currently we have oc192 |
| 01:32.31 | JT | sometimes, depends on the telco |
| 01:32.43 | JT | it's not your problem what it's delivered over |
| 01:32.46 | JT | it's the telco's |
| 01:33.03 | JT | km-: E1 PRIs are 30 voice channels |
| 01:33.15 | unbeatabl | well, we are multi-homed, have over 5 carriers |
| 01:33.26 | JT | data or voice? |
| 01:34.05 | unbeatabl | not sure, but if its data only, we can still somehow split the voice right? |
| 01:34.53 | unbeatabl | we have at&t level3 mzima xo savvis cogent |
| 01:35.08 | JT | depends how it's delivered |
| 01:35.10 | unbeatabl | surely one of them has got to serve voice |
| 01:35.16 | JT | sure |
| 01:35.24 | JT | but i have no idea what services you have |
| 01:36.43 | unbeatabl | but isnt the beauty of VOIP the ability to use internet connection for both outgoing and incoming calls? |
| 01:36.48 | unbeatabl | thast where i am confused about |
| 01:37.17 | *** join/#asterisk Strom_C (n=strom@m5a0e36d0.tmodns.net) |
| 01:37.38 | unbeatabl | i assumed, a server connected to a fiber, would be able to supply such services |
| 01:38.01 | unbeatabl | i even heard people using their own cable internet connection |
| 01:38.24 | JT | who cares if something is fibre or not |
| 01:38.27 | JT | it's just glass |
| 01:38.31 | JT | as opposed to copper |
| 01:38.41 | JT | PRIs are more reliable than phone calls over the Internet |
| 01:38.54 | rob0 | But is the glass half empty or half full? |
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| 01:39.23 | unbeatabl | yeah i dont care if its PRI or Fibre or even DSL |
| 01:40.11 | unbeatabl | i just want to know what I need to make outside calls via VOIP, but first i need some authentication system like when you use Phone Cards over the phone |
| 01:40.29 | unbeatabl | so i assume asterisk will handle that task |
| 01:40.39 | JT | yeah what we are suggesting is if you use voip to connect the remote staff |
| 01:40.44 | JT | like the SIP protocol |
| 01:40.48 | JT | that goes over the Internet |
| 01:40.56 | JT | but you terminate the calls via PRIs |
| 01:41.11 | JT | sip has authentication |
| 01:41.19 | unbeatabl | terminate meaning like, hangup the call? |
| 01:41.36 | km- | terminate meaning, the equipment that is on the other end. |
| 01:41.46 | km- | it's a telephony term |
| 01:41.57 | unbeatabl | oh |
| 01:42.00 | km- | getting SIP termination means someone is giving you sip connectivity. |
| 01:42.05 | km- | for instance, Vonage provides SIP termination. |
| 01:42.10 | unbeatabl | such a scary word for something so trivial |
| 01:42.21 | km- | yeah, it kinda is. |
| 01:42.26 | *** join/#asterisk zenspider (n=fooby@greed.zenspider.com) |
| 01:42.33 | km- | sorta like scsi requires terminators to finish the line, so does phones. |
| 01:42.43 | unbeatabl | ok so, this SIP thing |
| 01:42.49 | unbeatabl | is this a service i sign up to? |
| 01:43.02 | zenspider | hey... where is the copyright assignment contract for asterisk contributors? |
| 01:43.06 | nestAr | lol |
| 01:43.14 | JT | unbeatabl: it's just a protocol. |
| 01:43.19 | JT | how you use it is up to you |
| 01:43.29 | JT | skype uses a proprietary protocol |
| 01:43.29 | unbeatabl | i mean, some thing and somewhere, its gotta make calls right |
| 01:43.42 | JT | yes so you have sip servers, and sip clients |
| 01:43.55 | unbeatabl | dont i need real phone lines? |
| 01:44.08 | JT | sip is a voip protocol |
| 01:44.18 | JT | whether you have real phone lines or not is a different issue |
| 01:44.34 | unbeatabl | ok so, if i sign up with one of the sip carrier |
| 01:44.48 | unbeatabl | that is all i need to make and receive calls ? |
| 01:45.25 | km- | in your situation, do you want your salesmen to use their existing phones? |
| 01:45.31 | JT | i guess, but it would be better to use PRIs |
| 01:45.37 | JT | for the pstn side |
| 01:45.46 | unbeatabl | i want them to dial into a specific number, punch in their password, and dial the number they want to call |
| 01:45.58 | km- | I keep thinking there's a really simple solution for what unbeatabl wants that we just aren't getting the synapses connected |
| 01:46.03 | unbeatabl | so they can use their home phone/cell / anything |
| 01:46.17 | unbeatabl | yes that is what i feel too |
| 01:46.19 | km- | ok. What does this buy you, as a business? |
| 01:46.19 | unbeatabl | =/ |
| 01:46.26 | zenspider | anyone? where is the license? |
| 01:46.28 | km- | are you looking at this for masking their caller ID? |
| 01:46.40 | JT | ~thebook |
| 01:46.41 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 01:46.41 | km- | zenspider: I dunno, when I signed up it was a couple years ago and Mark just sent it to me directly. |
| 01:46.49 | unbeatabl | gives us more control and allows our developers to track usage of each salesmen's phone activity |
| 01:47.22 | unbeatabl | and gives us cheaper rates than to just use our at&t |
| 01:47.36 | km- | ok, so what you're really looking for is to setup a private calling-card system. |
| 01:47.40 | JT | if you only want them to call you first |
| 01:47.44 | unbeatabl | and we want them to even call from their cellphones |
| 01:47.45 | JT | then all you need is PRIs and servers |
| 01:47.53 | citats | zenspider: if you go to bugs.digium.com and submit a patch you can sign the license there |
| 01:47.56 | unbeatabl | km-: yeah, exactly |
| 01:48.02 | km- | unbeatabl: you want to have a calling card system that only your people use. |
| 01:48.10 | km- | OK. What's your predicted usage? |
| 01:48.27 | unbeatabl | high |
| 01:48.29 | km- | in other words, how many simultaneous calls and how long will they be on? |
| 01:48.34 | unbeatabl | 1000's |
| 01:48.48 | zenspider | citats: I don't have a patch to file yet, I want to read the language first... |
| 01:48.53 | km- | ok, I'm going to just say this, and I know I'm going to get lambasted. |
| 01:49.23 | km- | you're going to want to get a crapton of DS3's |
| 01:49.26 | km- | lots of cisco AS5400's |
| 01:50.01 | km- | you're talking initial investment of like... half a million dollars maybe? |
| 01:50.08 | citats | zenspider: i think if you login to the bug tracker and havent signed a license theres an option to do it somewhere |
| 01:50.45 | unbeatabl | yes |
| 01:50.53 | km- | ZenBSDi: it's kinda funny, I've had to negotiate two employment contracts so far because of the disclaimer |
| 01:50.54 | unbeatabl | in the millions to be precise |
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| 01:51.28 | km- | unbeatabl: yeah, you're talking a lot of telephony there. You'd need a cluster of asterisk servers and lots of terminating equipment |
| 01:51.54 | km- | unbeatabl: this would be an incredibly large project for you to implement alone. |
| 01:51.57 | JT | and sip proxies |
| 01:52.08 | km- | cisco as5400's take DS3's and terminate them as sip |
| 01:52.18 | km- | they're lovely pieces of technology, but at 50k a pop they're expensive toys |
| 01:52.44 | km- | we have them where I work to terminate all of our DS3's and I've heard from the vonage guys that they use them there to terminate their pots as well |
| 01:53.16 | JT | they terminate directly? |
| 01:53.31 | JT | there are other brands of pri gateways too |
| 01:53.36 | km- | unbeatabl: at this point I would suggest a few things. 1, find someone to hire who's better at telephony than you. 2, Figure out exactly how much usage you're expecting |
| 01:53.37 | JT | nothing that special about cisco |
| 01:53.44 | km- | no, but cisco's support is fantastic |
| 01:54.05 | JT | their sip firmware always leaves something to be desired |
| 01:54.07 | unbeatabl | yeah but i need to know before i start hiring |
| 01:54.17 | JT | there are other gateway manufacturers |
| 01:54.23 | km- | unbeatabl: you need more conclusive numbers asides from "thousands" |
| 01:54.41 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-47b94263c63ae125) |
| 01:54.45 | JT | defining the requirement would be the first thing to know |
| 01:54.51 | km- | yeah. |
| 01:54.55 | unbeatabl | 100 concurrent simultaneous user is a good guess |
| 01:55.27 | km- | with the 100 people calling in only able to call out to one other party? |
| 01:55.33 | km- | would you be utilizing conferencing? |
| 01:55.41 | km- | that increases the amount of telephony. |
| 01:55.45 | unbeatabl | they are calling to their clients |
| 01:56.08 | km- | asterisk can do telephone conference calls with multiple parties |
| 01:56.08 | *** join/#asterisk Strom_M (n=strom@m1f0e36d0.tmodns.net) |
| 01:56.12 | km- | salesmen usually love that. |
| 01:56.34 | km- | If you were anticipating 100 salesmen making 100 calls, that's 200 simultaneous telephone connections |
| 01:56.35 | unbeatabl | what do you mean by conference call |
| 01:56.46 | km- | multiple people all in the same telephone call |
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| 01:56.53 | unbeatabl | oh |
| 01:56.54 | Hadi- | hello everyone |
| 01:56.55 | km- | 3 or more parties all conversing in the same call. |
| 01:56.58 | unbeatabl | no, we dont need that |
| 01:57.13 | Hadi- | any Cisco voip experts here? (a little off topic :() |
| 01:57.23 | km- | ok, i'd say you could implement it for probably about $100k |
| 01:57.26 | unbeatabl | ok so who is the most knowledgeable guy in this channel |
| 01:57.29 | km- | plus per month |
| 01:57.35 | km- | hahaha, that's a difficult question |
| 01:57.48 | JT | are you joking, you don't even know what a conference is? |
| 01:57.50 | unbeatabl | ? |
| 01:58.06 | km- | we all have our areas of expertise, I guess. |
| 01:58.15 | JT | well this is just silly |
| 01:58.28 | JT | perhaps google stuff to stop looking so silly |
| 01:58.29 | km- | I take myself out of the running already because I work for a company where I can't do consulting on the side |
| 01:58.34 | Hadi- | The question is if there is any Cisco VoIP experts here :) |
| 01:58.35 | km- | JT: no need to be rude. |
| 01:58.45 | JT | km-: he wants everything spoonfed |
| 01:58.47 | km- | JT: the guy is obviously not a techie |
| 01:59.13 | JT | most businesspeople know what conferencing is. |
| 01:59.21 | km- | JT: I'd expect you to tell me to google if I was asking questions like that, but given his inability to even explain what it is he needed adequately I'm not sure he's at that level. |
| 01:59.22 | unbeatabl | ofcourse i know what it is |
| 01:59.37 | km- | not to sound like a prick inadvertantly myself :) |
| 01:59.39 | unbeatabl | i thought it was some asterisk program |
| 01:59.53 | km- | Hadi-: what's your issue? |
| 02:00.43 | Hadi- | km-: we have a cisco 2811 and I would like to be able to transcode codec's |
| 02:00.46 | JT | i suggest reading the book that i have pointed out the url to |
| 02:00.50 | JT | or hiring a consultant |
| 02:01.00 | Hadi- | im just wondering if a PVDM2-48 module will work without the Cisco Call Manager |
| 02:01.21 | km- | Hadi-: ah, callmanager stuff I'm definitely not comforatble with. we have asterisk for pbx stuff at my company |
| 02:01.52 | Hadi- | I see |
| 02:01.56 | *** join/#asterisk MikeJ (n=MikeJ@adsl-69-209-153-61.dsl.sfldmi.ameritech.net) |
| 02:02.00 | Hadi- | Cisco has to be so complicated |
| 02:02.03 | Hadi- | not like asterisk |
| 02:02.13 | Hadi- | buy a $700 module to be able to transcode |
| 02:02.16 | Hadi- | :) |
| 02:02.29 | *** part/#asterisk zenspider (n=fooby@greed.zenspider.com) |
| 02:02.34 | km- | it's an upside though |
| 02:02.49 | km- | you get a bunch of good hardware dsps, suddenly your machine doesnt need to do nearly as much CPU to transcode. |
| 02:04.08 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:05.37 | km- | heh, every time I see kumbang's nick I misread it as Kumbag and I think to myself "wow, what an incredibly rude nickname..." |
| 02:06.37 | JT | well it'd be mispelt for that context you had in mind |
| 02:07.02 | km- | I didn't think there was really a defined "websters dictionary"-esque spelling. |
| 02:07.10 | km- | I just know it exists in several variations :) |
| 02:11.51 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) |
| 02:13.45 | *** join/#asterisk BadHorsie (n=sebas@ip254-10.ct.co.cr) |
| 02:15.29 | *** join/#asterisk EnigmaCurry (n=user@67.166.72.245) |
| 02:18.38 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 02:19.34 | ZX81 | hi all I have two identical machines - same kernel, cpu, box etc - one show 8192 zaptel samples transferred in 8192 system clock the other 8192 in 8188.631 system clock - any ideas? |
| 02:20.05 | ZX81 | both have 1000HZ clock ticks |
| 02:20.52 | ZX81 | both same tdm400p cards |
| 02:21.35 | MikeJ | same zaptel and config too? |
| 02:21.45 | ZX81 | yep :) copied from one machine to the other |
| 02:21.55 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-8c1f3b3631c16d4c) |
| 02:22.04 | MikeJ | same absolutely everything |
| 02:22.14 | ZX81 | well something's not! :) |
| 02:22.28 | ZX81 | one has fractions of a clock and the other whole numbers |
| 02:22.34 | MikeJ | darn.. I was hoping to follow with... SAME IP's !! ? |
| 02:22.38 | ZX81 | :) would have expected kernel ticks etc |
| 02:22.40 | ZX81 | lol |
| 02:22.46 | ZX81 | actually |
| 02:22.52 | ZX81 | eth1 are the same |
| 02:22.57 | ZX81 | eth0 not though :) |
| 02:22.59 | MikeJ | one card is faster or one system clock is slower..... |
| 02:23.08 | MikeJ | :D |
| 02:23.10 | ZX81 | :D |
| 02:23.24 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 02:23.29 | ZX81 | hrmmm - were different zap versions |
| 02:23.36 | ZX81 | now got whole numbers |
| 02:23.52 | MikeJ | ?? |
| 02:23.55 | ZX81 | but transferred 8192 samples in 8188 sample intervals! |
| 02:24.05 | ZX81 | running too fast lol |
| 02:24.13 | ZX81 | the opposite of missing interrupts |
| 02:24.14 | ZX81 | :) |
| 02:24.37 | ZX81 | might try turning on multimedia timer in bios and see what happens |
| 02:24.49 | ZX81 | reboot |
| 02:24.50 | ZX81 | hah |
| 02:24.54 | ZX81 | wrong keyboard |
| 02:24.54 | ZX81 | :) |
| 02:26.10 | MikeJ | I don't think that will effect zaptel at all when using a card.. its timing off the card itself. |
| 02:26.28 | MikeJ | I would talk to digium support on this one.. something is not right for sure? |
| 02:26.32 | ZX81 | yeah same - here's hoping - installing it tomorrow |
| 02:26.39 | ZX81 | yeah will do |
| 02:27.40 | *** join/#asterisk BeeBuu (n=chatzill@61.145.76.132) |
| 02:27.45 | BeeBuu | hello,all |
| 02:27.51 | ZX81 | hi |
| 02:28.27 | BeeBuu | anyone give me a iax.conf to help me setup a user in IAX,please? |
| 02:28.39 | ZX81 | make samples? |
| 02:28.51 | BeeBuu | yes,would you? |
| 02:28.59 | ZX81 | or cat /usr/src/asterisk/configs/iax.conf.sample |
| 02:29.21 | ZX81 | gimme a sec |
| 02:29.26 | ZX81 | I'll pastebin you something |
| 02:29.26 | BeeBuu | not found |
| 02:29.38 | BeeBuu | thank you very much! |
| 02:30.08 | *** join/#asterisk tzafrir_laptop (i=tzafrir@192.117.42.208.static.012.net.il) |
| 02:30.31 | ZX81 | ooh xt :) |
| 02:30.32 | ZX81 | http://pastebin.ca/800086 |
| 02:30.50 | ZX81 | thats for a user with the username 101 and password mypassword |
| 02:31.06 | ZX81 | and their context is internal (i.e. thats where the calls go) |
| 02:32.16 | BeeBuu | thanks again |
| 02:32.50 | ZX81 | you might also want to check: |
| 02:32.51 | ZX81 | http://www.voip-info.org/wiki-Asterisk+config+iax.conf |
| 02:33.14 | ZX81 | then read TFOT |
| 02:33.15 | ZX81 | sec |
| 02:33.30 | ZX81 | http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip |
| 02:34.55 | ZX81 | might go offline - just checking dhcp serving on this new box - have fun |
| 02:34.57 | ZX81 | ~adn |
| 02:34.57 | jbot | rumour has it, adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 02:35.03 | BeeBuu | ZX81: no general context? |
| 02:35.20 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 02:35.26 | UserReg_CL | mmm |
| 02:35.35 | ZX81 | yeah just copy from that voip-info site |
| 02:35.52 | UserReg_CL | ~book |
| 02:35.52 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 02:36.35 | BeeBuu | ok,let me try |
| 02:38.50 | MikeJ | hey.. thats jim/life/jarrod's book.. |
| 02:42.41 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 02:42.41 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:43.08 | BadHorsie | i'm having this same issue while compiling libpri: http://asterisk.pastebin.ca/745585 gcc 4.2.1-6; glibc 2.7-2; |
| 02:44.13 | MikeJ | BadHorsie: what is on pri_internal.h:263 ? |
| 02:44.30 | tzafrir_laptop | BadHorsie, trunk? |
| 02:45.00 | MikeJ | greetings tzafrir_laptop |
| 02:45.13 | BadHorsie | it's libpri-1.4.2 downloaded as libpri-1.4-current, lemme see that line |
| 02:45.23 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 02:45.34 | MikeJ | BadHorsie: what os/distro? |
| 02:45.37 | BadHorsie | void libpri_copy_string(char *dst, const char *src, size_t size); |
| 02:45.57 | BadHorsie | debian 2.6.22-3-amd64 |
| 02:46.08 | MikeJ | the error is that it doesn't know what size_t is |
| 02:46.58 | UserReg_CL | what version debian recommend ? |
| 02:47.00 | MikeJ | going to be a missing header file.. weird that your seeing that in debian tho.. is that etch? |
| 02:47.18 | BadHorsie | i had this sid machine around and i'm just trying it :P |
| 02:47.29 | BadHorsie | wish i could dist-downgrade... |
| 02:47.48 | MikeJ | trying to remember what header size_t is from |
| 02:48.06 | MikeJ | try adding includes in q931.c |
| 02:48.15 | MikeJ | above the pri_internal.h |
| 02:48.41 | MikeJ | its going to be stdio, stdlib, unistd... somthing like that |
| 02:48.58 | BadHorsie | sure thing |
| 02:49.06 | MikeJ | curious what its missing |
| 02:49.27 | BadHorsie | ok, i't was cstring.h |
| 02:49.31 | BadHorsie | now it works |
| 02:50.02 | MikeJ | stdio.h |
| 02:50.15 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 02:50.18 | MrTelephone | hey hows it goin guys |
| 02:50.26 | MikeJ | gcc has a habit of including that "magically" ... |
| 02:50.33 | MikeJ | maybe they finally stopped that.. |
| 02:50.42 | MikeJ | that would be nice. |
| 02:50.46 | BadHorsie | haha, gotta love those automagic operations |
| 02:51.10 | tzafrir_laptop | MikeJ, that's the recent bug report? |
| 02:51.32 | MrTelephone | what happens if you indicate ringing to a sip channel and then the remote sip agent responds with ringing.. what does asterisk do? |
| 02:51.55 | MikeJ | tzafrir_laptop: huh? |
| 02:52.23 | MikeJ | I ran in to that on the iphone toolchain.. i know gcc includes that silently for you... not sure of any bug report |
| 02:54.15 | MikeJ | BadHorsie: did adding that include help |
| 02:54.41 | BadHorsie | i first did an include of cstring.h and it worked, same with your stdio.h... |
| 02:55.19 | MikeJ | ok.. looks like new gcc .. at least debian's newer gcc doesn't include that for ya.. |
| 02:55.33 | BadHorsie | pretty bad |
| 02:55.40 | MikeJ | no.. good |
| 02:56.07 | BadHorsie | i wonder why |
| 02:56.08 | MikeJ | silently including stuff is bad for cross platform... assumes everyone uses a compiler that behaves in this non standard waay |
| 02:56.20 | BadHorsie | oh, i see |
| 02:57.04 | MikeJ | it's just like gcc enabling all the gcc extensions by default... people generally don't realize when things are gcc specific.. |
| 02:59.22 | MikeJ | BadHorsie: short story.. can you submit a bug on that so someone can fix it in tree |
| 03:05.38 | BadHorsie | not familiar with submitting bugs for libpri, let's give it a try |
| 03:06.30 | tzafrir_laptop | MikeJ, sorry, got to go |
| 03:06.39 | MikeJ | you can even submit a patch :D |
| 03:07.15 | *** join/#asterisk b_d (n=brian@209.240.42.151) |
| 03:07.29 | MikeJ | take a quick look.. make sure its not already there |
| 03:07.48 | BadHorsie | yeah, lemme try the svn repository and do an update |
| 03:08.03 | b_d | I think Im going to puke |
| 03:08.35 | b_d | I just dug up usenet postings from myself, which are almost a decade old, debating about motif and lesstif |
| 03:09.18 | b_d | I'm not even 25 yet |
| 03:10.00 | b_d | sorry, wrong window |
| 03:11.55 | MikeJ | b_d: your old! |
| 03:12.43 | BadHorsie | ok, instead of stdio.h they added stddef.h, that's the only change from the current 1.4.2/pri_internal.h i just downloaded like 10 minutes ago... |
| 03:13.11 | BadHorsie | and yep, it's fixed |
| 03:13.43 | b_d | MikeJ: certainly feels like it...realizing that you spent soo much time on USENET....also realizing your first linux kernel was pre-2.0.... |
| 03:14.12 | BadHorsie | and release date from http://downloads.digium.com/pub/libpri doesn't help saying it was from 1969, hehe |
| 03:14.26 | MikeJ | heh... |
| 03:14.48 | MikeJ | I find that unlikely given that pri didn't exist in 1969 :P |
| 03:17.04 | MikeJ | opengroup.org confirms.. stddef.h is the right one.. |
| 03:20.27 | BadHorsie | so the only thing left to do is to update the libpri-1.4-current...the file was changed on october 22, how often are this files updated? |
| 03:21.33 | MikeJ | I honestly have been out of the loop for over a year.. not sure what the release and fix backporting procedures are now |
| 03:21.50 | BadHorsie | ok |
| 03:22.20 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:22.20 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:22.29 | MikeJ | russellb: might know |
| 03:22.39 | russellb | orly? |
| 03:22.43 | MikeJ | oyeah |
| 03:22.56 | russellb | about? |
| 03:22.58 | MikeJ | how do you guys time minor releases of stuff now adays? |
| 03:23.11 | russellb | magic |
| 03:23.22 | MikeJ | ok.. so about the same as previous |
| 03:23.28 | MikeJ | :P |
| 03:23.45 | MikeJ | you still doing backporting stuff? or who does that now ? |
| 03:24.00 | Qwell | nobody, we have a sane process now :p |
| 03:24.03 | *** join/#asterisk salzh (n=salzh@124.77.9.255) |
| 03:24.09 | russellb | we always fix things in the release branches first |
| 03:24.14 | MikeJ | was just helping BadHorsie chase down a compile error in libpri that it looks like was fixed in trunk already. |
| 03:24.23 | MikeJ | but not in the marked release |
| 03:24.33 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 03:24.45 | MikeJ | so you forward port... fun |
| 03:24.46 | Qwell | hasn't been a libpri release in a while |
| 03:24.54 | MikeJ | 6 weeks :P |
| 03:24.55 | CrazyTux | Hi guys, anyone here have a successful setup, of click to dial type stuff using manager.conf / ami etc? |
| 03:25.09 | Qwell | glibc 2.7 hasn't been in any distros for long |
| 03:26.01 | Qwell | it's in like Fedora 9 alpha, or whatever it's called now |
| 03:27.15 | MikeJ | Fedora 627 RC 3 |
| 03:28.01 | MikeJ | Fedora is such a great idea.. redhat gets to try out all the new shit on people for free and release every 10 days... |
| 03:28.15 | MikeJ | and then once it gets beat to hell, put it in redhat |
| 03:30.52 | MikeJ | anyhow.. gotta run.. |
| 03:30.56 | *** part/#asterisk MikeJ (n=MikeJ@adsl-69-209-153-61.dsl.sfldmi.ameritech.net) |
| 03:42.32 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-08e05744abd15938) |
| 03:42.45 | perf3kt | i was thumbing through the new book |
| 03:43.05 | perf3kt | what do you hardcore cli'ers think of the part in the book about the gui |
| 03:43.20 | perf3kt | necessary? too much? |
| 03:47.01 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 03:47.17 | [TK]D-Fender | perf3kt, are we really going through all this AGAIN? |
| 03:48.48 | JT | [TK]D-Fender: troll troll troll your boat gently down the stream ;) |
| 03:49.19 | [TK]D-Fender | JT, "I think we're going to need a bigger boat." |
| 03:49.34 | hijacked | that's a good answer. |
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| 04:04.38 | salzh | during asterisk background tip sound, if i dial number via ip phone, there is no action at all, but if i use analog phone, it can get working. why does ip phone fail to dial number? |
| 04:06.02 | JT | tip sound? |
| 04:06.23 | [TK]D-Fender | salzh, betting you didn't set the right DTMFMODE for you SIP phone |
| 04:07.10 | [TK]D-Fender | salzh, most phones should use rfc2833 and you'll hae to make sure its sip.conf entrey matches |
| 04:13.33 | Qwell | hey, could somebody in the US do me a favor, and try dialing 888-454-8366? I think my cellphone might be going crazy |
| 04:13.45 | Qwell | just see if the call completes |
| 04:14.44 | Qwell | ring, call failed...no inband indication...quite strange |
| 04:14.44 | mog | ring |
| 04:15.26 | mog | disconnect |
| 04:15.31 | Qwell | weird, so it's them |
| 04:17.13 | UserReg_CL | mmm helpme |
| 04:17.14 | UserReg_CL | please |
| 04:18.11 | JT | help help omg so much info to go on |
| 04:18.22 | [TK]D-Fender | UserReg_CL, #drphil |
| 04:18.36 | UserReg_CL | mmm hi Fender |
| 04:19.13 | [TK]D-Fender | UserReg_CL, pwned :D |
| 04:19.53 | [TK]D-Fender | NEXT!@@!one@!one (c) BKW |
| 04:22.26 | UserReg_CL | please... helpme with one AGI |
| 04:23.03 | JT | ~question |
| 04:23.04 | jbot | extra, extra, read all about it, 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 |
| 04:23.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:25.37 | [TK]D-Fender | ~ask |
| 04:25.37 | jbot | 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. |
| 04:26.16 | JT | haha |
| 04:29.21 | [TK]D-Fender | "help! help! Oh woe is me! waaaahhhhhh" /quit |
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| 04:32.43 | nestAr | lol |
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| 04:37.57 | *** join/#asterisk n0cturnal_ (n=n0c@cl1-203-82-208-197.45ru.net.au) |
| 04:39.17 | n0cturnal_ | anyone know if running caching squid for an office on a production asterisk box would be asking too much? ( single quad core xeon 2gig, 4gb ram, 4 x 15k rpm disks R5) |
| 04:40.02 | [TK]D-Fender | n0cturnal_, I doubt it |
| 04:40.19 | [TK]D-Fender | n0cturnal_, describe your * usage |
| 04:41.38 | n0cturnal_ | single E1 sangoma, roughly 10 normal XTN's for everyday usage, up to 10 agents in the CC, 2 queues, both recorded |
| 04:42.31 | [TK]D-Fender | n0cturnal_, All G711 straight recording? |
| 04:43.25 | n0cturnal_ | mp3 |
| 04:46.55 | [TK]D-Fender | n0cturnal_, what are they pre-conversion? |
| 04:47.17 | [TK]D-Fender | n0cturnal_, * doesn't record MP3 direct and usually a batch conversion is a lot less load than on the fly |
| 04:48.40 | n0cturnal_ | to tell the truth I haven't actually configured the recording yet, so i'm not sure exactly how i'm going to do it, but i've got nothing against using straight g711 then batching it i guess |
| 04:48.48 | n0cturnal_ | the CC goes live next week |
| 04:51.11 | [TK]D-Fender | n0cturnal_, you should be fine I'm guessing. how my cahcing BW do you figure will be passing through? |
| 04:54.36 | russellb | <PROTECTED> |
| 04:55.09 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 04:55.19 | russellb | just keep an eye on the load at first until you're sure |
| 04:55.46 | CrazyTux | Hey guys how do I pass a variable with AMI |
| 04:55.53 | CrazyTux | Variables: variable = value ? |
| 04:56.24 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:00.47 | Corydon76-dig | CrazyTux: what do you mean? |
| 05:01.16 | CrazyTux | Corydon76-dig, if I want to pass, like, for example a "filename digits" being 1000, so I can do a record(VAR:gsm) |
| 05:01.31 | CrazyTux | Corydon76-dig, how would I access / even pass that variable with AMI |
| 05:02.44 | Corydon76-dig | CrazyTux: use the SetVar event |
| 05:04.08 | CrazyTux | Corydon76-dig, whats an example of how that would be passed? |
| 05:04.11 | CrazyTux | Corydon76-dig, like I'm doing action: originate |
| 05:04.30 | CrazyTux | Corydon76-dig, would it be within that "bit" written to ami, or would I have to do a whole thing before it? action: setvar? |
| 05:04.47 | Corydon76-dig | Action: SetVar Channel: SIP/123-456 Variable: bar Value: baz |
| 05:08.17 | CrazyTux | Corydon76-dig, thank you, that works :) |
| 05:12.42 | MrTelephone | anyone have trouble with ata's ringing the phone right after onhook event? |
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| 05:51.56 | variable_office | is there a way to jump to Voicemailmain from voicemail() by hitting a special key? |
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| 05:53.01 | [TK]D-Fender | variable_office, go read up on *'s Standard Extensions. |
| 05:54.04 | variable_office | looks like o is that right? |
| 05:54.08 | variable_office | 0=o i mean |
| 05:54.32 | [TK]D-Fender | variable_office, Read up.... |
| 05:54.41 | fetcher | anyone worked with a Siemens Hicom 2000 ? |
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| 06:16.53 | obnauticus | How do I change the Caller ID name for an incomming call? |
| 06:17.12 | obnauticus | like any incomming call on this sip line then change the caller ID |
| 06:17.15 | obnauticus | because SetCAllerID isn't working :| |
| 06:22.05 | [TK]D-Fender | obnauticus :because like so much else its DEPRECATED. "show function CALLERID", and read up on UPGRADE.TXT and all the other wonderful docs that came with the source... |
| 06:22.07 | [TK]D-Fender | night all |
| 06:22.11 | obnauticus | <3 |
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| 06:34.46 | jaike | were using agentcallbacklogin so agents can sit on any phone and calls to their "agent id extension" will go to that phone. with agentcallbacklogin deprecated, are there other applications we can use to do this? |
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| 06:54.11 | philth | I am trying to use WaitExten instead of Background, but it doesn't get any DTMF #'s Works fine with Bckground but nogo with WaitExten. |
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| 07:01.38 | Corydon76-home | philth: did you Answer the channel already? |
| 07:02.15 | philth | Corydon76-home, Yes. Answer() Wait(2) then WaitExten(10) |
| 07:02.35 | philth | when I call it it times out after 10 seconds weither or not I press digits. |
| 07:03.07 | Corydon76-home | Well, it's doing exactly the same thing as Background |
| 07:05.13 | philth | I know, thats whats so weird. I've been digging at this all day. I am probably just going to make a silent sound file and use Background. Bit of a kludge but I am out of ideas. |
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| 07:18.01 | philth | Corydon76-home, Not sure if your interested or not, but putting Background(beep) then WaitExten(10) works. |
| 07:18.12 | philth | dtmf is rfc2833 |
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| 07:24.17 | yangvnc | Hello I am trying to call an extension and my phone number - I got this setup in extensions.conf -> exten => 2970,1,Dial(SIP/2970&Phone/041710598) but I get error Nov 28 08:21:28 NOTICE[23188]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Phone' (cause 0 - Unknown) |
| 07:25.05 | yangvnc | I want dual ring, a SIP extension + a phone number which is 041710598 |
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| 07:38.55 | BlackSlik | any know any tool i can use for asterisk for mass call (astrisk mass Dialer ) |
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| 07:44.00 | zerocod3r | could someone explain me the difference between Goto(supportmenu,s,1) and Goto(supportmenu, s, 1) reading a book and dont understand what the author imply |
| 07:44.52 | JT | one is correct syntax, one is not |
| 07:45.26 | zerocod3r | JT but the second one with blank lines also works sometimes works sometimes not works I mean |
| 07:45.51 | JT | so the idea is not to use it |
| 07:47.36 | zerocod3r | ok, I dont know why but sometimes want to understand why it is doing that :) I mean why it is working in some situation and not in others but doesnt matter |
| 07:49.46 | JT | because asterisk does not have a proper parsing engine |
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| 08:05.11 | yangvnc | Hello I am trying to call an extension and my phone number - I got this setup in extensions.conf -> exten => 2970,1,Dial(SIP/2970&Phone/041710598) but I get error Nov 28 08:21:28 NOTICE[23188]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Phone' (cause 0 - Unknown) |
| 08:05.18 | yangvnc | I want dual ring, a SIP extension + a phone number which is 041710598 |
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| 08:23.26 | endre | phone is not a technology |
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| 08:39.58 | obnauticus | YAy |
| 08:39.58 | obnauticus | NETSPLIT! |
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| 08:45.19 | salzh | have sb been using appliance. |
| 08:46.46 | salzh | i cann't reset the password by the instruction in its user manual |
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| 09:09.20 | yangvnc | Hello I am trying to call an extension and my phone number - I got this setup in extensions.conf -> exten => 2970,1,Dial(SIP/2970&Phone/041710598) but I get error Nov 28 08:21:28 NOTICE[23188]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Phone' (cause 0 - Unknown) |
| 09:09.29 | yangvnc | I want dual ring, a SIP extension + a phone number which is 041710598 |
| 09:11.04 | kaldemar | what kind of connection do you have to PSTN? |
| 09:11.45 | yangvnc | its going out over the trunk |
| 09:11.47 | yangvnc | liker this |
| 09:11.51 | kaldemar | what trunk? |
| 09:11.59 | kaldemar | what technology? |
| 09:12.16 | yangvnc | exten => _X.,1,Dial(SIP/pritrunk/${EXTEN}) |
| 09:12.38 | kaldemar | where did you come up with the Phone then? |
| 09:13.11 | kaldemar | try SIP/2970&SIP/pritrunk/041710598 |
| 09:16.22 | yangvnc | thanks works |
| 09:17.08 | yangvnc | how would i make it to call 041710598 after 10 sec. ? |
| 09:17.49 | kaldemar | after calling 10 seconds to 2970 first? |
| 09:17.57 | yangvnc | yes |
| 09:18.46 | kaldemar | you make a line that dials 2970 with a 10 second timeout and then a following line that dials the other number. |
| 09:19.31 | yangvnc | yes, but that can be specified in the Dial command? |
| 09:19.47 | kaldemar | "core show application dial" in CLI will give you help on the dial command. |
| 09:19.53 | yangvnc | ok |
| 09:21.04 | yangvnc | hm |
| 09:21.04 | yangvnc | it won't show that |
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| 09:21.05 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 09:21.07 | yangvnc | 1.2.24 |
| 09:21.17 | kaldemar | show application dial then. |
| 09:21.36 | mort_gib | What would cause Asterisk to be instable?? |
| 09:22.37 | mort_gib | Or in other word, has anyone had stability problems?? |
| 09:23.29 | mort_gib | Extensions suddenly not working?? Forwards suddenly not working, but no line issues?? |
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| 09:40.34 | R1ck | whats the definition of a "channel" ? |
| 09:46.50 | mort_gib | R1ck: In zaptel.conf it's like a physical lien, or bundle of lines |
| 09:46.57 | mort_gib | lien=lies |
| 09:47.02 | mort_gib | :-) |
| 09:47.23 | mort_gib | Eating snacks and typing == bad spelling |
| 09:47.27 | mort_gib | Lines of course |
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| 09:54.41 | yangvnc | I do get a busy line error http://openpaste.org/en/4078/ |
| 09:54.57 | Uatec | hi, is there an easy way that I can turn on DTMF notifications in the CLI without turning all a load more bumf? |
| 09:55.10 | yangvnc | when dialing 400 |
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| 10:17.47 | yangvnc | what does this mean Nov 28 11:17:19 NOTICE[21869]: chan_sip.c:9750 handle_response_invite: Failed to authenticate on INVITE to '"2970" <sip:2970@212.103.133.6>;tag=as6e325366' |
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| 10:36.38 | zapp-branigan | hello i'm looking for the g723.1 in digium but i can't find please where can i buy a license for g723.1 ? |
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| 10:46.23 | UserReg_CL | Hi !!! good day ! |
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| 11:40.12 | vale-ICS | morning all |
| 11:44.07 | zeeesh | voicemail.conf problem "This content is stored as http://sial.org/pbot/28906. |
| 11:44.08 | zeeesh | From: "zeeesh" at 124.29.214.46 |
| 11:44.08 | zeeesh | "? |
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| 11:47.11 | zeeesh | where are the asterisk champions ? i m in this problem from last week .. could not resolve ... |
| 11:48.54 | RoyK | http://xkcd.com/350/ |
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| 11:52.30 | buscon | Hi, I'm trying to use ekiga with asterisk but I can't get to authenticate |
| 11:52.35 | buscon | I followed this guide |
| 11:52.42 | buscon | http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client |
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| 11:53.02 | buscon | and still get this error |
| 11:53.05 | buscon | chan_sip.c:10545 handle_request_invite: Failed to authenticate user "marcello kdlasj" <sip:101@10.0.1.10>; |
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| 11:58.13 | yangvnc | what does this mean Nov 28 11:17:19 NOTICE[21869]: chan_sip.c:9750 handle_response_invite: Failed to authenticate on INVITE to '"2970" <sip:2970@212.103.133.6>;tag=as6e325366' |
| 11:58.32 | yangvnc | I do get a busy line error http://openpaste.org/en/4078/ when dialing 400 extension |
| 11:59.48 | *** part/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 12:01.47 | coppice | has anyone here tried the phones from Zed-3? |
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| 12:09.38 | puzzled | hi |
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| 12:21.55 | Belgarath | i have a problem with linksys router/ata and the regular phone |
| 12:22.01 | Belgarath | it is connected to my asterisk |
| 12:22.11 | Belgarath | and when i call outsiede everything is ok |
| 12:22.26 | Belgarath | but when i clall locally io do not hear the voice in the headphone |
| 12:22.34 | Belgarath | on most conversations |
| 12:22.39 | Belgarath | but echo is working ok |
| 12:22.46 | Belgarath | and i can hear a dial tone |
| 12:22.53 | Belgarath | and ringing |
| 12:23.55 | ai-a | locally ? |
| 12:24.10 | ai-a | you mean another sip phone on the local lan, connected to an outside pbx ? |
| 12:25.04 | Belgarath | ai-a: anather phone on the lan connected to asterisk not linksys |
| 12:25.31 | ai-a | and the asterisk is outside your lan. |
| 12:25.37 | Belgarath | it is inside |
| 12:25.44 | ai-a | so everything is inside your lan ? |
| 12:25.53 | Belgarath | the linksys is only wroking asa ata not as a router |
| 12:25.57 | Belgarath | yes |
| 12:26.10 | ai-a | so, lets ignore all the words 'router' in your question, as its on your lan. |
| 12:26.19 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.134.26) |
| 12:26.19 | Belgarath | yes |
| 12:26.36 | ai-a | show the sip.conf / extension.conf for the system. |
| 12:26.50 | ai-a | please dont be using the sample conf files... |
| 12:26.54 | ai-a | and use pastebin.ca for pasting. |
| 12:27.29 | zeeesh | voicemail.conf problem "This content is stored as http://sial.org/pbot/28906. ? |
| 12:27.50 | Belgarath | ai-a: i have users listed in users.conf |
| 12:27.52 | Belgarath | not sip.conf |
| 12:28.02 | ai-a | ok, then users.conf ;) |
| 12:28.12 | ai-a | again, not samples and minimal comments please. |
| 12:28.24 | ai-a | zeeesh: and using sendmail yourself works ? |
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| 12:31.20 | Belgarath | ai-a: http://pastebin.ca/800392 |
| 12:31.24 | Belgarath | that is users.conf |
| 12:31.38 | Belgarath | which part of extensions.conf are you interested in |
| 12:31.39 | Belgarath | ? |
| 12:31.49 | Belgarath | the calls between other phones on the lan works ok |
| 12:32.19 | Belgarath | it is only that the calls from the ata on the lan when they are not echo calls go mute |
| 12:32.25 | Belgarath | and it is only one direction |
| 12:32.32 | Belgarath | to the ata |
| 12:33.26 | ai-a | why does pastebin crash my browser ;) |
| 12:33.40 | Belgarath | evil js |
| 12:33.51 | mosty | it just looks blank in my browser |
| 12:33.57 | Belgarath | i 0wn y0ur machine n0w |
| 12:34.01 | Belgarath | :) |
| 12:34.19 | ai-a | Belgarath: wait,, everything works apart from ata boxes ? |
| 12:34.24 | Belgarath | yes |
| 12:34.37 | Belgarath | well the ata box works perfectly to |
| 12:34.49 | Belgarath | when i call from ata through asterisk to soem external phone |
| 12:34.52 | Belgarath | it is working ok |
| 12:35.04 | Belgarath | if i call echo service on my asterisk it is working |
| 12:35.13 | Belgarath | but if i call any other phone on lan |
| 12:35.20 | Belgarath | or voicemaile on asterisk |
| 12:35.28 | Belgarath | i do not hear what the other end is saying |
| 12:35.32 | ai-a | ok, then show us the cli output. |
| 12:35.46 | ai-a | 1 way audio issue ? - try unplugging the ata box and pluggin it back in. |
| 12:36.04 | Belgarath | ai-a: i unplugged it 100 times already |
| 12:36.17 | Belgarath | and the one way autdio only happens inside the lan |
| 12:36.26 | Belgarath | and cli is not reporting anything |
| 12:36.34 | Belgarath | and with sip debug sip transaction is ok |
| 12:36.38 | ai-a | do you have nat=yes in the users.conf ? |
| 12:36.53 | Belgarath | no |
| 12:36.57 | Belgarath | but why should i ? |
| 12:37.03 | Belgarath | no device is behind a nat |
| 12:37.07 | ai-a | nope, thats the point :) |
| 12:37.14 | ai-a | not sure, seems strange to me. |
| 12:37.24 | Belgarath | well seems strange to me to |
| 12:37.29 | Belgarath | i can even live with it |
| 12:37.35 | Belgarath | becasue i relly call myself |
| 12:37.36 | Belgarath | :) |
| 12:37.41 | Belgarath | rarly* |
| 12:38.01 | Belgarath | just interested if anybody had similar problem |
| 12:38.26 | Belgarath | the strange thing is it is working if it goes through the asterisk to the outside world |
| 12:38.35 | Belgarath | and not working if it goes back to lan |
| 12:39.00 | Belgarath | the even stranger thing is it is working with echo and ringing and nothing else |
| 12:39.23 | Belgarath | if i put a playback service just after the rings i would not hear what playback is saying |
| 12:39.27 | Belgarath | on the ata device |
| 12:39.38 | Belgarath | it will work on other phones in the lan |
| 12:45.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:55.30 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 12:59.32 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 13:01.42 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 13:02.12 | Aurs | off topic: does anyone know a sip client that works in osx leopard? |
| 13:02.18 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 13:07.15 | *** join/#asterisk KeltusEx (i=Keltus@c-76-21-119-166.hsd1.ca.comcast.net) |
| 13:13.11 | deever | re |
| 13:14.20 | deever | i'm trying to cross-compile asterisk on i686 for armel with optware for debian... |
| 13:14.36 | deever | ...and get the following error: "dlfcn.c:133: error: storage size of `mainStatus' isn't known" |
| 13:14.55 | deever | how can i fix that? |
| 13:17.47 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 13:18.39 | hi365 | my zaptel has a pri configured on span 1, but if i do "pri show spans" asterisk is seeing it on span 2. how could that be? |
| 13:19.39 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:20.19 | hi365 | my zaptel has a pri configured on span 1, but if i do "pri show spans" asterisk is seeing it on span 2. why would that be? |
| 13:20.55 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 13:21.12 | mosty | is that a sangoma card or digium? |
| 13:21.42 | hi365 | sangoma |
| 13:22.27 | [TK]D-Fender | hi365: pastebin your zaptel & zapata |
| 13:22.49 | mosty | pastebin your wanpipe1.conf and wanpipe2.conf also |
| 13:23.05 | *** join/#asterisk foobar-- (n=foo@p13324-adslbkksp4.C.csloxinfo.net) |
| 13:23.07 | hi365 | ops, im not on site. ill pasebin later |
| 13:23.12 | [TK]D-Fender | hi365: Oh and never ask something like that wiout already having the pastebin ready and a full list of all cards in that system |
| 13:23.28 | [TK]D-Fender | hi365: Even BETTER. Ask us why and have nothing to show us! Congratulations! |
| 13:23.44 | mosty | i think with wanpipe, you can select which span a particular port should be on. that's where i'd look first |
| 13:23.56 | hi365 | sir, yes sir! |
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| 13:25.52 | *** join/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) [NETSPLIT VICTIM] |
| 13:26.25 | Siya | how would I go about substituting + for 00? outbound calls to various trunks |
| 13:26.36 | Siya | exten => _+NX!,1,Goto(00${EXTEN:1}) |
| 13:26.39 | Siya | doesn't work |
| 13:27.01 | hi365 | Siya: skip the + |
| 13:27.03 | Siya | as it tries to find the resulting number internally rather than matching one of the trunks |
| 13:27.25 | Siya | hi365: how do you mean? |
| 13:27.29 | hi365 | can ur phone actualy dial a +? |
| 13:27.37 | hi365 | exten => _NX!,1,Goto(00${EXTEN:1}) |
| 13:27.41 | foobar-- | Hi, is it possible to use Asterisk with existing ordinary PBX? If so, how does it work? |
| 13:27.53 | hi365 | great! |
| 13:28.09 | Siya | I'd like to be able to dial +31123456789, as it would a nice integration of softphone with say outlook |
| 13:28.23 | hi365 | Siya: can your phone dial a "+"? |
| 13:28.32 | Siya | hi365: my 7960's can dial urls fine |
| 13:28.39 | hi365 | hmm |
| 13:28.52 | Siya | as can Xlite and other softphones |
| 13:29.14 | yangvnc | hello [TK]D-Fender |
| 13:29.14 | yangvnc | Could you help me around the INVITE error ? http://www.ip-phone-forum.de/showthread.php?t=154099 |
| 13:29.34 | Siya | you're right in that on my handsets I program 0031 0044 etc but in my PIM software (eg outlook) I use +31 +44 etc |
| 13:29.46 | mosty | foobar--, depends what interfaces the old world pbx has |
| 13:30.12 | foobar-- | How do I find out? |
| 13:30.13 | [TK]D-Fender | foobar--: what exactly do you want to do? |
| 13:31.02 | [TK]D-Fender | yangvnc: Nov 28 13:58:25 NOTICE[21869]: chan_sip.c:9750 handle_response_invite: Failed to authenticate on INVITE to '"2970" <sip:2970@212.103.133.6>;tag=as19b535fd' |
| 13:31.07 | foobar-- | I want to implement Rule-based call forwarding in a small office environment. We have 15 phone outlet |
| 13:31.32 | [TK]D-Fender | yangvnc: tells me you probaby didn't sey your auth right. And of course you didn't provide SIP debug with taht so there is nothing more to say. |
| 13:31.36 | deever | i'm trying to cross-compile asterisk on i686 for armel with optware for debian... |
| 13:31.47 | deever | ...and get the following error: "dlfcn.c:133: error: storage size of `mainStatus' isn't known" |
| 13:31.48 | mosty | foobar--, what type of outgoing telephone lines does it use? |
| 13:31.50 | deever | how can i fix that? |
| 13:31.59 | foobar-- | mosty, PSTN |
| 13:31.59 | [TK]D-Fender | foobar--: what exactly would you be forwarding? |
| 13:32.12 | foobar-- | TDK, normal voice calls |
| 13:32.33 | yangvnc | [TK]D-Fender: ok debugging it now....can i save debug into a file? |
| 13:32.57 | mosty | foobar--, well you could put fxs ports in an asterisk box, and connect that to the old pbx |
| 13:33.03 | [TK]D-Fender | foobar--: Ok, that is WAY to vague. I need you to descibe a COMPLETE specific scenario. |
| 13:33.17 | [TK]D-Fender | yangvnc: copy& paste into a PB |
| 13:33.19 | [TK]D-Fender | ~pb |
| 13:33.21 | 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 |
| 13:34.43 | foobar-- | mosty, and from there, I can set up the PBX to forward all incoming calls to Asterisk, right? |
| 13:35.31 | yangvnc | [TK]D-Fender: http://www.ip-phone-forum.de/showthread.php?t=154099 at the bottom |
| 13:35.36 | mosty | foobar--, i would assume so |
| 13:35.52 | foobar-- | TK, okay -- A caller calls in over PSTN, Asterisk picks up and route the call to either internal extension or mobile phone based on conditions |
| 13:36.35 | [TK]D-Fender | foobar--: How would it send it to a specific internal phone? |
| 13:37.13 | foobar-- | TK, we already have a PBX so we'll keep it for internal routing |
| 13:37.42 | yangvnc | [TK]D-Fender: I don't know why it's saying From: "2970" <sip:2970@212.103.133.6>;tag=as29496e42 |
| 13:37.48 | yangvnc | should be another ip there... |
| 13:38.28 | yangvnc | just the opposite actually |
| 13:38.43 | [TK]D-Fender | yangvnc: pastebin the ENTIRE call with SIP debug. |
| 13:38.44 | *** join/#asterisk xDie (n=xdie@3-31.dedicado.com.uy) |
| 13:39.10 | [TK]D-Fender | foobar--: that does not answer my question |
| 13:39.38 | yangvnc | [TK]D-Fender: from which side? the 2970 is calling to 400 |
| 13:40.08 | foobar-- | TK, can you clarify your question? |
| 13:40.10 | yangvnc | the log is pasted from the 400 side server |
| 13:40.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:41.16 | [TK]D-Fender | yangvnc: Where do I see SIP/2970 defined as a device in the first place? |
| 13:45.10 | yangvnc | this is the 2970 side |
| 13:45.13 | yangvnc | http://openpaste.org/en/4082/ |
| 13:46.29 | [TK]D-Fender | yangvnc: ok, I'm not giong to work on this in pieces. put EACH system into its own pastebin in its entirely. |
| 13:46.43 | Siya | how would I go about substituting + for 00? outbound calls to various trunks in extensions.conf? |
| 13:46.57 | [TK]D-Fender | yangvnc: And I did NOT even get the full invite in your SIP debug. Do it AGAIN, and stop cutting corners |
| 13:47.38 | yangvnc | which side of SIP DEBUG the 400 side i have allready pasted |
| 13:47.39 | [TK]D-Fender | Siya: You don't "substitute", you just accept a different pattern and do what you have to with the number that was dialed |
| 13:47.53 | yangvnc | i am going to paste you 2700 side now |
| 13:47.59 | [TK]D-Fender | yangvnc: It doesn't have the FULL invite header. its clearly chopped off. |
| 13:48.00 | yangvnc | 2970 even |
| 13:48.19 | yangvnc | [TK]D-Fender: well i get that by sip debug |
| 13:48.25 | yangvnc | i havent chopped it off |
| 13:48.30 | yangvnc | maybe i ned an extra -v |
| 13:49.12 | [TK]D-Fender | yangvnc: <-- SIP read from 212.103.134.179:5060: <--- see this? I don't have one and I don't see the INVITE line. I know its missing now get off your assand do the job right. |
| 13:49.23 | *** join/#asterisk _x86_ (n=x86@i.am.leet.org) |
| 13:50.43 | *** join/#asterisk Kigh (n=kai@ciphron.de) |
| 13:50.54 | [TK]D-Fender | <PROTECTED> |
| 13:51.28 | *** part/#asterisk xDie (n=xdie@3-31.dedicado.com.uy) |
| 13:51.34 | [TK]D-Fender | foobar--: I wasnt you to include the WIRING and PORTS yuo expect on both sides that will allow for this. |
| 13:51.51 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 13:53.32 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 13:55.01 | deever | i'm trying to cross-compile asterisk on i686 for armel with optware for debian... |
| 13:55.04 | deever | ...and get the following error: "dlfcn.c:133: error: storage size of `mainStatus' isn't known" |
| 13:55.08 | deever | http://rafb.net/p/ISWgoZ28.html |
| 13:55.10 | deever | how can i fix that? |
| 13:56.07 | foobar-- | TK, I'm not sure how * will send the call, that's essentially what I'm trying to find out |
| 13:58.27 | *** part/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 13:59.49 | Siya | [TK]D-Fender: well that would mean doubling all my international rules, I was hoping to avoid that with a single dialplan rule |
| 14:00.30 | [TK]D-Fender | foobar--: If you can't even guess how your old PBX could take any kind of call in and know where to send it then this is a key sign that this entre idea is a mistake. |
| 14:00.50 | [TK]D-Fender | foobar--: How many phones do you have, and how many lines on your existing PBX? |
| 14:01.56 | *** part/#asterisk PepOSX (n=pepOSX@190.78.220.149) |
| 14:02.10 | *** join/#asterisk PepOSX (n=pepOSX@190.78.220.149) |
| 14:02.28 | foobar-- | tk, as i said before, we have 15 phones |
| 14:02.34 | [TK]D-Fender | Siya: >exten => _+NX!,1,Goto(00${EXTEN:1}) <- looks close, but you've got your aprameters wrong. Go read its instructions again. |
| 14:03.12 | [TK]D-Fender | Siya: And mind you that is a dangerous looking pattern. I hoe you aren't clashing with things like extens for internal phones, etc |
| 14:03.17 | [TK]D-Fender | hope* |
| 14:03.20 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:03.26 | [TK]D-Fender | foobar--: And lines? |
| 14:04.11 | foobar-- | maybe 1 or 2. i'm not sure how many goes through the PBX |
| 14:04.52 | foobar-- | for the sake of this discussion, lets say 1. |
| 14:05.07 | [TK]D-Fender | foobar--: For the pain this would cause even if its possible would be a glorious pain in the ass. You' better off selling your old setup and just using * straight |
| 14:06.00 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:06.10 | foobar-- | tk, to do that, I'm looking for a TDM card with 15 or more FSO outlet, correct? |
| 14:06.15 | Siya | [TK]D-Fender: tx |
| 14:06.25 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 14:07.07 | [TK]D-Fender | foobar--: I'd advise SIP hardphones. |
| 14:08.16 | foobar-- | tk, thanks for your input |
| 14:10.33 | *** join/#asterisk Somebee (n=Somebee@80.232.5.97) |
| 14:10.49 | Somebee | Anyone got IAX2 working on OS X Leopard ? |
| 14:11.38 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:12.47 | Nugget | I used to run asterisk all the time on my MBP in Tiger, but I haven't touched it in a while. Does it not work in Leopard? that's weird. |
| 14:13.57 | *** join/#asterisk bantu (n=Miranda@p54A32A4C.dip0.t-ipconnect.de) |
| 14:15.44 | *** join/#asterisk Aeudian (n=somewher@75.148.21.113) |
| 14:17.53 | Aeudian | Are there any SIP based hardware phones that support Cisco VPN natively. I'm looking for something for someone to take on the road that ties into the corp network via vpn, but the client doesn't want a software based phone, but rather a physical phone. I know i can do a cisco 800 series router and a phone combo, but is there a phone that supports vpn connections? |
| 14:19.10 | Siya | doh! |
| 14:19.25 | Siya | exten => _+NX!,1,Goto(00${EXTEN:1},1) <<< note the ,1 at the end |
| 14:19.38 | Siya | Goto(extension,priority) Goto(priority) |
| 14:19.59 | Siya | I was setting the priority not changing the extension |
| 14:20.01 | Siya | :) |
| 14:20.05 | [TK]D-Fender | Siya: yup :) |
| 14:21.41 | hi365 | what does "Auto fallthrough" mean? |
| 14:21.44 | hi365 | (as in: Auto fallthrough, channel 'IAX2/ipconnect-1' status is 'UNKNOWN') |
| 14:22.58 | ai-a | Aeudian: hmmm, problem is the router needs to have vpn. |
| 14:23.03 | [TK]D-Fender | hi365: means you hi the end of your exten without disabling autofallthrough and execution just stops |
| 14:23.30 | hi365 | but i have this: exten => *151,n(badpass),Playback(vm-mismatch) |
| 14:23.30 | hi365 | extne => *151,n,Goto(s,1) |
| 14:23.43 | [TK]D-Fender | Aeudian: I can imagine a CISCO phone supporting CISCO VPN, but why would anybody else? |
| 14:24.04 | hi365 | in theory if the goto heads back to the top you can never hit "the end" |
| 14:24.06 | [TK]D-Fender | hi365: Gee does the spelling on #2 look right to YOU? |
| 14:24.24 | hi365 | [TK]D-Fender: yeh, thanks |
| 14:25.09 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:25.18 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:25.55 | ai-a | Fender: we need a 'auto-spellcheck' module :) |
| 14:26.09 | hi365 | or just error reporting for misspeled words |
| 14:26.26 | hi365 | Channel 'IAX2/iax-1' sent into invalid extension 's' in context 'a2billing', but no invalid handler |
| 14:26.30 | [TK]D-Fender | hi365>or just error reporting for misspeled words <- IRONIC |
| 14:26.36 | hi365 | whats THAT? |
| 14:26.48 | [TK]D-Fender | hi365: OMG, do you have a clue at all? |
| 14:26.52 | ai-a | hi365: do you have "The Book" ? |
| 14:26.55 | [TK]D-Fender | hi365: No exten "s". |
| 14:27.17 | [TK]D-Fender | hi365: You don't have one and thats where a call is landing. End of story |
| 14:27.34 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-57-3.dsl.tul2ok.sbcglobal.net) |
| 14:27.41 | hi365 | [TK]D-Fender: take it easy. just a bad day i guess |
| 14:28.16 | [TK]D-Fender | hi365: Stop jumping at this "problems", grab a coffee (or TWELVE) and stare at them for a while before jsut pasting them all here. |
| 14:28.33 | ai-a | use your own coffee,, not ours ;) |
| 14:28.40 | hi365 | lol |
| 14:32.38 | [TK]D-Fender | these* |
| 14:33.00 | [TK]D-Fender | See.. a spell-check would have let that one slide! |
| 14:34.50 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:39.58 | *** part/#asterisk Aeudian (n=somewher@75.148.21.113) |
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| 14:44.54 | deever | i'm trying to cross-compile asterisk on i686 for armel with optware for debian... |
| 14:44.57 | deever | ...and get the following error: "dlfcn.c:133: error: storage size of `mainStatus' isn't known" |
| 14:45.01 | deever | how can i fix that? |
| 14:45.03 | deever | http://rafb.net/p/ISWgoZ28.html |
| 14:46.05 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 14:46.22 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-ce1b8d8de58a7005) |
| 14:48.47 | bkw_ | deever: if you're going to cross compile it would be expected that you would know a little something about how to program. I would recommend you look at the astfin project to see if maybe that can help. |
| 14:49.28 | *** join/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 14:49.55 | BCS-Satori | Good Morning, we have setup and installed asterisk in 4 locations. All 4 sites are complaining that the music on hold is too loud from the end user who is calling in. I have gone in with audacity and lowered the db rating to attempt to quiet the audio with no prevail. I believe it is only really loud on cell phones because it is digital to digital. Any suggestions on quieting the music on hold? |
| 14:50.38 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:50.40 | bkw_ | BCS-Satori: yes you didn't have to do it like that |
| 14:50.52 | bkw_ | BCS-Satori: you using mp3? |
| 14:50.54 | bkw_ | or native? |
| 14:51.18 | BCS-Satori | bkw_: i have tried both mp3 and native, right now all sites are using default music with no adjustments |
| 14:51.22 | deever | bkw_: i know a little bit about how to program and want to use the optware package... |
| 14:51.34 | bkw_ | btw hold music shouldn't be very loud in the first place |
| 14:51.46 | deever | bkw_: do you know sth about this error? |
| 14:52.04 | bkw_ | deever: Not in Asterisk because I refuse to even look at the code anymore. |
| 14:52.09 | Belgarath | bkw_: it depends what kind of music you are playing |
| 14:52.12 | bkw_ | deever: but google can help |
| 14:52.14 | Belgarath | and who do you work for |
| 14:52.21 | Belgarath | :) |
| 14:52.24 | coppice | perhaps its barry manilow. anything audible would then be too loud |
| 14:52.35 | bkw_ | Belgarath: phone etiquette says not very loud. |
| 14:52.38 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 14:52.58 | Belgarath | coppice bkw but you can work as Balck Sabbath fan call center |
| 14:53.01 | Belgarath | f.e. |
| 14:53.11 | bkw_ | Belgarath: did you pay for a license to play that music on hold? |
| 14:53.12 | deever | bkw_: it didn't, else i wouldn't have asked here...;) |
| 14:53.13 | BCS-Satori | bkw_: even the default music that comes with asterisk they say is to loud, i have even tried using mpg123 to quiet the audio with the quietmp3 command and audacity but it is very loud and crakly |
| 14:53.31 | bkw_ | mpg123 has a quiet mode that works great |
| 14:53.48 | BCS-Satori | bkw_: i saw no difference =/ |
| 14:53.49 | bkw_ | sounds like someone broke something in the quietmp3 |
| 14:53.54 | Belgarath | bkw_: i'm not playing it but if i was their 'fan call center' i would definitly tell them that they shpoudl plya one of their songs |
| 14:54.13 | bkw_ | Belgarath: hehe |
| 14:54.22 | BCS-Satori | bkw_: that doesn't make sense to me, having 4 sites all donig the same thing |
| 14:54.25 | [TK]D-Fender | BCS-Satori: What are you using to terminate to the PSTN? |
| 14:54.43 | BCS-Satori | TK: we are using a voip carrier called nexVortex |
| 14:55.22 | [TK]D-Fender | BCS-Satori: Then they are running their gains "hot". The MoH files taht come with * are jsut fine for standard gain lines. Its your provider |
| 14:56.30 | BCS-Satori | TK: sorry for my ignorance but what do you mean by hot? Does the carriers system notice a change between moh and normal rtp talk? |
| 14:56.48 | bkw_ | BCS-Satori: cranked up |
| 14:57.01 | BCS-Satori | TK: i guess what i'm asking is what do I need to say to the carrier to see what they can do for me |
| 14:57.14 | bkw_ | wait a min while I laugh my ass off |
| 14:57.25 | bkw_ | BCS-Satori: you been in telcom long? |
| 14:57.31 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 14:57.55 | BCS-Satori | bkw_: PSTN and POTS, yes but VoIP no only about 1 year |
| 14:58.00 | jameswf | hideeee hoooo |
| 14:58.03 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 14:58.16 | BCS-Satori | bkw_: the carrier does listen to my requests though |
| 14:58.16 | bkw_ | BCS-Satori: you can't get anyone in telcom do anything they don't wanna do.. an chances are they don't care. |
| 14:59.59 | BCS-Satori | bkw_: I guess i don't understand how "normal audio" such as talking sounds fine, but music is too loud? Does the carrier adjust the audio once the moh comes on or is it just luck that it happens |
| 15:00.34 | bkw_ | BCS-Satori: because youre mouth is away from the receiver and not tied directly to your vocal chords |
| 15:00.51 | BCS-Satori | bkw_: ok that makes sense |
| 15:01.16 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 15:01.18 | casix | hello |
| 15:02.18 | PTorres | hello, any experts on chanspy around ? |
| 15:02.39 | bkw_ | PTorres: yes |
| 15:02.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:04.21 | PTorres | bkw_: hi, I am having a problem with spies on a quintum registered to my * , it is dropping too many frames mostly from the remote end of the call being spied on |
| 15:04.50 | bkw_ | PTorres: sounds like you have silence supp. on the quintum |
| 15:05.12 | *** join/#asterisk NovceGuru (i=shelby@ballmung.easymac.org) |
| 15:05.12 | PTorres | bkw_: sorry, the agents on the quintum , the spy on polycom/eyebeam |
| 15:05.45 | bkw_ | what do the calls come in on? |
| 15:06.05 | PTorres | bkw_: calls come into the agents on the quintum |
| 15:06.27 | bkw_ | no the end the caller is on |
| 15:06.29 | bkw_ | not the agent |
| 15:07.16 | PTorres | bkw_: both zap channels and voip providers |
| 15:07.36 | bkw_ | can you tell if the calls you are listening to are via voip or tdm when it does this? |
| 15:07.41 | bkw_ | that needs to be known |
| 15:08.06 | PTorres | bkw_: yes... it happens on both, but I believe its worse on tdm calls |
| 15:09.48 | Somebee | Hi. I try to dial out to 5 different sips with Dial, and have a macro that fires when someone picks up the phone. How can I (in that macro) find out who it was that actually answered? |
| 15:09.52 | [TK]D-Fender | wow... a coherant technical question with decent description of circumstances.... yup... there goes our daily quota! :p |
| 15:13.07 | PTorres | bkw_: we tried changing codecs on the quintum and the spies but the problem persisted |
| 15:13.34 | bkw_ | PTorres: take the quintum out of the picture and see if it goes away |
| 15:15.13 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:15.13 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:16.07 | *** join/#asterisk saftsack (n=saftsack@pD9E04797.dip.t-dialin.net) |
| 15:19.01 | *** join/#asterisk fbuilesv (n=fbuilesv@unaffiliated/febuiles) |
| 15:19.26 | fbuilesv | Hi, I'm trying to get started with Asterisk but I'm having some issues. The Asterisk CLI is "asterisk -r" right? |
| 15:19.54 | PTorres | bkw_: we did that already, we listened to other calls softphone/tdm and softphone/voip and the sound comes clear from both ends , I can check again just in case |
| 15:19.56 | fbuilesv | I'm trying to run "dialplan reload" but it says "No such command dialplan" |
| 15:20.00 | [TK]D-Fender | fbuilesv: Once * is running as a daemon, ytes |
| 15:20.09 | [TK]D-Fender | fbuilesv: try "reload" |
| 15:21.00 | fbuilesv | That doesn't give me any error but it does show some warnings in chan_zap.c |
| 15:21.23 | fbuilesv | [TK]D-Fender: "reload" does the same as "dialplan reload"? Trying to follow O'Reilly's book here. |
| 15:21.28 | [TK]D-Fender | fbuilesv: Ok, well thats another issue. |
| 15:21.41 | [TK]D-Fender | fbuilesv: Depends which version of * vs which ver of the book. |
| 15:21.45 | [TK]D-Fender | fbuilesv: Things change... |
| 15:22.04 | [TK]D-Fender | fbuilesv: So what versions? |
| 15:22.11 | fbuilesv | Asterisk: 1.2.13 |
| 15:22.37 | fbuilesv | Oh, the book covers 1.4, problem might be there? |
| 15:22.54 | [TK]D-Fender | fbuilesv: if it doesn't match. but thats jsut for your CLI misha. |
| 15:22.58 | [TK]D-Fender | mishap |
| 15:23.32 | fbuilesv | So the "dialplan" command was available before 1.4? |
| 15:23.33 | [TK]D-Fender | fbuilesv: But no matter. If you'd like to address your Zaptel error, pastbin your zaptel.conf & zapata.conf and tell us what card(s) yuo have) |
| 15:23.35 | [TK]D-Fender | ~pb |
| 15:23.36 | 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 |
| 15:23.37 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:24.04 | [TK]D-Fender | fbuilesv: I'm not sure what the syntax is between 1.2 and 1.4 anymore :) I just try both till one works :) |
| 15:24.28 | fbuilesv | [TK]D-Fender thanks for the help but I'm trying it out on a friend's server, so I'd rather not play with his settings. I'll try to contact him first :D |
| 15:24.42 | [TK]D-Fender | fbuilesv: Your call. |
| 15:26.40 | fbuilesv | The "module" command _should_ be available too to do a "module show", right? |
| 15:27.46 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:29.15 | *** join/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 15:29.48 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:32.18 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 15:32.30 | PTorres | bkw_: if it is of any help, we are getting the "queue too long, dropping frames" warning on * cli |
| 15:33.20 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:33.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:34.04 | Somebee | In an inbound call: ... ,n,Dial(SIP/dev1&SIP/dev2|60|M(macro_link_that_will_fire_when_taken)) <-- How can I found out who (sip1 or sip2) answered? |
| 15:39.11 | Corydon76-dig | In the Macro, check your ${CHANNEL} variable |
| 15:39.32 | Somebee | Corydon76-dig: But in the macro I can't access any variables except the ones I pass along |
| 15:39.48 | Corydon76-dig | ${CHANNEL} isn't really a variable |
| 15:39.50 | Somebee | and if I pass along the channel, it is the channel from before the lines were answered |
| 15:40.01 | Corydon76-dig | It's a builtin |
| 15:41.28 | Corydon76-dig | but you will need to CUT it, to remove the call identifier at the end |
| 15:41.34 | Somebee | I know |
| 15:41.36 | Somebee | :) |
| 15:41.53 | *** join/#asterisk arsirc (n=arsirc@unaffiliated/arsirc) |
| 15:41.54 | Somebee | But, I reckon I've tried calling channel, I'll test again now |
| 15:42.38 | *** part/#asterisk arsirc (n=arsirc@unaffiliated/arsirc) |
| 15:43.04 | km- | hey, does anyone here know if there's a cisco 79xx config file reference somewhere? |
| 15:43.22 | Nivex | yes there is |
| 15:43.31 | km- | I wanna set all the phones to Full100 media negotiation since we're seeing some handshaking errors, but I'm coming up zeros trying to find it, must not be using the right search terms. |
| 15:44.17 | *** join/#asterisk afritz (n=jhr@adsl-69-151-116-233.dsl.fyvlar.swbell.net) |
| 15:44.59 | afritz | good morning |
| 15:45.31 | Corydon76-dig | What's so good about it? |
| 15:46.41 | km- | not much |
| 15:48.59 | buscon | Hi, I need a good cli sip client |
| 15:49.08 | buscon | any advice ? |
| 15:49.40 | afritz | Corydon76-dig: hmmm ... the sun rose again |
| 15:50.21 | Corydon76-dig | Yay skin cancer |
| 15:50.31 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 15:50.42 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 15:50.54 | JunK-Y | any has seen that weird issue before: |
| 15:50.56 | JunK-Y | AGI Rx << STREAM FILE tt-monkeys "" -- Playing 'tt-monkeys' (escape_digits=) (sample_offset 0) -- AGI Script /home/junky/dyn_menu2.agi completed, returning 0 |
| 15:50.57 | [TK]D-Fender | Corydon76-dig: Good Morning Starshine, the Earth says HELLO! |
| 15:50.58 | afritz | gee ... what an optimist :) |
| 15:51.02 | iratik | I am not receiving inbound calls.... is there anyway to monitor inbound requests using the asterisk CLI ? |
| 15:51.02 | JunK-Y | and it doesnt wait for the stream. |
| 15:51.18 | *** join/#asterisk r3zon8 (n=hg@64.80.101.162) |
| 15:51.38 | [TK]D-Fender | iratik: "set verbose 10" and enable debug on whatever channel type the call should be coming in on |
| 15:52.03 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 15:52.45 | [TK]D-Fender | afritz: there are GSM gateways out there... |
| 15:54.09 | afritz | [TK]D-Fender: so i'd have to buy hardware, eh? |
| 15:54.11 | *** part/#asterisk matt_c (n=mcroydon@137.147.45.66.cm.sunflower.com) |
| 15:55.14 | afritz | it would be nice to be able to use an existing cell phone |
| 15:55.56 | [TK]D-Fender | afritz: It'll require hardware one way or another. |
| 16:02.45 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 16:03.33 | iratik | "enable debug on whatever channel type" ... whats a channel type? |
| 16:07.32 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:07.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:07.53 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 16:08.02 | [TK]D-Fender | iratik: sip, iax, mgcp, zap, sccp, h323, SMOKE SIGNAL |
| 16:08.26 | iratik | sip channel type.... so how to enable debug on sip channel type ? |
| 16:08.36 | [TK]D-Fender | iratik: "sip debug" |
| 16:14.42 | *** join/#asterisk masus (n=ethemc@88.248.14.186) |
| 16:15.02 | masus | can anyone give me ta tipp howto install rxfax |
| 16:15.27 | jameswf | I get an od pleasure out of using the word bug(s) around tweakers |
| 16:16.15 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:16.32 | *** join/#asterisk russellb_ (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:16.32 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 16:20.41 | *** join/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 16:30.41 | iratik | I'm doing something wrong --- i won't even attempt to guess at what it is though.... whats wrong with this picture? http://pastie.caboo.se/123095? |
| 16:32.09 | [TK]D-Fender | iratik: I'll give you a clue..... "Looking for 4175534249 in icall_in (domain 70.248.216.14)" SIP/2.0 404 Not Found <------------- |
| 16:32.16 | iratik | i saw that |
| 16:32.23 | iratik | but what do i have improperly configured |
| 16:32.29 | [TK]D-Fender | iratik: Your dialplan of course |
| 16:32.41 | iratik | i thought dialplans only had to do with outgoing |
| 16:32.46 | [TK]D-Fender | iratik: You don't have an exten in that context to match the exten coming in. |
| 16:33.01 | [TK]D-Fender | iratik: no, exactly the OPPOSITE |
| 16:33.17 | nestAr | iratik: asterisk doesn't know the differnce between a outbound and a inbound call |
| 16:33.22 | [TK]D-Fender | iratik: a call is coming IN to * and needs to be preocessed. |
| 16:33.29 | [TK]D-Fender | processed* |
| 16:34.53 | ai-a | iratik: read the book ? |
| 16:37.52 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:38.10 | *** join/#asterisk NirS (n=chatzill@87.68.79.176.cable.012.net.il) |
| 16:38.26 | NirS | hi all |
| 16:38.41 | NirS | anybody home ? |
| 16:40.09 | blitzrage | nope |
| 16:40.17 | blitzrage | everyone went out to the party |
| 16:40.29 | Qwell | except you |
| 16:40.31 | Qwell | and me, apparently |
| 16:40.47 | blitzrage | This has been an auto-message by blitzbot. |
| 16:40.51 | Qwell | fail |
| 16:43.08 | *** join/#asterisk wothinn (n=Allfathe@vs1.svartalfheim.net) |
| 16:43.55 | NirS | anyone ever had problems with DTMF's coming in from COMCAST ? |
| 16:46.19 | bkw_ | NirS: no but I have heard people with problems with p2p packets not coming in! |
| 16:46.29 | nestAr | bastards |
| 16:47.34 | NirS | I have a customer that is having some really funky DTMF issues |
| 16:47.49 | NirS | they have an Asterisk server, connected to DID providers via SIP |
| 16:48.20 | NirS | no matter where the call is coming from, DTMF is picked up nicely, apart from calls coming in from COMCAST, which always end up with double digits in the DTMF sequence |
| 16:48.21 | *** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net) |
| 16:48.23 | bkw_ | 2833 or sip info? |
| 16:48.42 | bkw_ | NirS: sounds like someone has a misunderstanding on 2833 |
| 16:48.47 | bkw_ | does the problem happen if you dial slowly |
| 16:48.54 | bkw_ | with short delays between pressing digits? |
| 16:50.44 | NirS | rfc2833 is in use, it happens be it slow or fast |
| 16:52.04 | bkw_ | someone isn't marking the end bit or honoring it |
| 16:52.12 | bkw_ | you might wanna hack asterisk to not send the end marker bit |
| 16:52.14 | bkw_ | works with Cisco |
| 16:52.24 | bkw_ | some ciscos do the exact same thing |
| 16:52.49 | NirS | Brian, care to run that by me again? I've never hacked asterisk at that level before |
| 16:54.00 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 17:02.06 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:05.49 | *** join/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 17:05.52 | *** part/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 17:07.22 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 17:13.48 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
| 17:14.49 | *** join/#asterisk dinesh___ (n=dinesh@vpn-epfl-b128.epfl.ch) |
| 17:14.51 | kand | Hey all, when I place callers on hold asterisk is resetting the SEQ number back to 0 without sending a new INVITE SSRC. Anybody have any ideas? |
| 17:15.14 | *** join/#asterisk s0lid (n=_freq@210.213.198.25) |
| 17:15.46 | dinesh___ | hi all - I would like to send a "missed call" to a given number using either asterisk, or SIP directly, from a PHP script, someone knows where i could start ? |
| 17:17.30 | Belgarath | dinesh___: asterisk can put data into db2 |
| 17:17.36 | Belgarath | from dialp[lans |
| 17:17.42 | Belgarath | you can extract it from there |
| 17:18.14 | [TK]D-Fender | dinesh___: can you more fully explain what it is you're looking to do. That description is far to vague |
| 17:19.50 | dinesh___ | from a php script, dial a number XXXXXX and hangup immediatly , using a given SIP configuration in asterisk |
| 17:20.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:21.47 | [TK]D-Fender | dinesh___: please describe the complete process. |
| 17:22.19 | [TK]D-Fender | dinesh___: What is this "missed call" and what is it you're "sending"? |
| 17:22.31 | dinesh___ | a missed call is a call not answered |
| 17:22.44 | dinesh___ | if I dial and hang up immediatly, the receiver dont have the time to answer it |
| 17:23.22 | dinesh___ | (basicly, I want to use this as a "signal" , and there is no need to pay anything, because there is no real communication, only the ringing time which is free) |
| 17:24.33 | [TK]D-Fender | dinesh___: So if you dial and hang up immediately (I presume this is a normal call palced through *), what do you want to have happen next? |
| 17:24.56 | dinesh___ | nothing |
| 17:24.57 | dinesh___ | done |
| 17:25.02 | [TK]D-Fender | ... |
| 17:25.37 | dinesh___ | i just need to call a given number and hang up just after the first ring or something similar |
| 17:25.51 | dinesh___ | so in order to do that |
| 17:25.52 | [TK]D-Fender | dinesh___: how is this call to be placed? |
| 17:25.58 | dinesh___ | I'll need to execute an extension |
| 17:26.24 | dinesh___ | from my PHP script, I guess I could run asterisk -r , and then write "run extension XXX" or i don't know how it works |
| 17:28.44 | [TK]D-Fender | dinesh___: lookup "call files" on the WIKI for this |
| 17:29.10 | [TK]D-Fender | dinesh___: you can have * place an outbound call which you can terminate in say 3 seconds (enough to ring once or so I guess) |
| 17:29.32 | [TK]D-Fender | dinesh___: and if by some chance they answer you have it pass them into your dialplan on an exten that will simply hangup |
| 17:31.39 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 17:32.26 | dinesh___ | ok thanks |
| 17:32.37 | variable_office | if i wanted any call that comes in on the incoming contact to do something what would I make its extension be? |
| 17:34.04 | [TK]D-Fender | variable_office: What exten is the call coming in under? |
| 17:34.29 | variable_office | could any integer, i am just assigning usernames integers |
| 17:34.40 | variable_office | *could be any integer |
| 17:35.34 | variable_office | If I needed it to a be certain length I could give it fix lengthed by padding it with 0s in front or make it sip_00001 |
| 17:36.30 | *** join/#asterisk mrempire (n=trefpunt@mrempire.demon.nl) |
| 17:36.42 | [TK]D-Fender | variable_office: please do not mix USER NAMES with EXTENSIONS. |
| 17:37.03 | [TK]D-Fender | variable_office: for an EXTENSION dialed just pick a pattern that matches the kind you're expecting. |
| 17:37.57 | variable_office | this is for incoming calls and for the system I build extensions and usernames are one in the same |
| 17:38.22 | variable_office | the calls coming in at 1@asterisk.xxx.xxx |
| 17:38.26 | variable_office | for example |
| 17:38.37 | [TK]D-Fender | variable_office: then make an exten taht can match that. |
| 17:39.12 | variable_office | but the EXTEN could be anything from 1-max |
| 17:39.27 | [TK]D-Fender | variable_office: thats why you pick a PATTERN <----- http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 17:39.32 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:39.55 | [TK]D-Fender | variable_office: this is Dialplan 101 and the absolutely most important part of Asterisk. |
| 17:41.19 | variable_office | so _. would do |
| 17:42.03 | NirS | btw, anyone ever encountered here a problem with AGI, where the first GET VARIABLE in the script will always return a 510 error, and all other GET VARIABLE commands are offseted or something like that ? |
| 17:42.52 | [TK]D-Fender | variable_office: _. is very dangerous because it also matches Asterisk Standard Extensions like "h" and can caue problems. This is to be avoided. |
| 17:43.16 | *** join/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 17:43.54 | variable_office | ah, good point, i will add a sip prefix to that |
| 17:43.57 | variable_office | so it is sip1 |
| 17:43.59 | variable_office | for example |
| 17:44.22 | variable_office | so _sip. |
| 17:44.43 | [TK]D-Fender | variable_office: how are you planning on dialing letters in front? |
| 17:45.06 | variable_office | the number -> username mapping is all done through enum |
| 17:45.12 | variable_office | completely seperate server |
| 17:45.15 | variable_office | that is running openser |
| 17:45.26 | [TK]D-Fender | variable_office: stop mixing up the USERNAME and a DIALED EXTENSION. |
| 17:45.59 | variable_office | as i said for me its the same. all i do is dial(SIP/${EXTEN}) eventually |
| 17:46.10 | *** join/#asterisk red9012 (n=marc3234@76-10-149-62.dsl.teksavvy.com) |
| 17:46.11 | [TK]D-Fender | variable_office: a PHONE is not an EXTENSION. An extension is NUMBER you DIAL. |
| 17:48.01 | variable_office | yes, well when the user dials a number it goes to enum which translates it into the exten(or username same thing for me) |
| 17:49.51 | [TK]D-Fender | variable_office: well if that is forced to match by some other process, fine. |
| 17:49.58 | variable_office | it is |
| 17:52.21 | *** join/#asterisk mascool (n=george@adsl-75-46-27-119.dsl.sfldmi.sbcglobal.net) |
| 17:52.59 | mascool | does anyone know how to fix the Got SIP response 415 "Unacceptable Content-Type" " error on a GXP2000, fw 1.1.4.22 and asterisk 1.4.4 ? |
| 17:53.40 | mascool | i checked the forums, bugs but they're all from different versions of asterisk and gxp fw |
| 17:53.55 | *** join/#asterisk robeph (n=robf@24.214.206.254) |
| 17:54.03 | [TK]D-Fender | variable_office: for sanity's sake I would make them similar but not identical like "exten => _U.,1,Dial(SIP/USER${EXTEN:1},30) |
| 17:54.48 | [TK]D-Fender | mascool: pastebin a complete call where that occurs so we can see what it is. |
| 17:54.51 | [TK]D-Fender | ~pb |
| 17:54.52 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:54.53 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 17:55.06 | [TK]D-Fender | mascool: SIP debug enabled |
| 17:55.55 | mascool | coming |
| 17:56.18 | *** join/#asterisk saftsack (n=saftsack@pD9E04797.dip.t-dialin.net) |
| 17:58.38 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:01.19 | mascool | [TK]D-Fender, http://pastebin.com/m1c0b45d9 |
| 18:04.17 | [TK]D-Fender | mascool: Hrm...not sure, but you are several revisions out of date... |
| 18:04.53 | mascool | i know but supposedly this was fixed before 1.4.4 |
| 18:05.14 | [TK]D-Fender | mascool: well go try the current release and see. |
| 18:05.20 | mascool | I will |
| 18:05.21 | mascool | thanks |
| 18:08.14 | dinesh___ | great thanks [TK]D-Fender the call files is exactly what I need ;) |
| 18:08.54 | variable_office | [TK]D-Fender, any problem you foresee with prefixing everything with 000 for uniqueness in the asterisk dialplan? |
| 18:09.10 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:09.23 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:09.39 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:10.06 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:11.04 | [TK]D-Fender | variable_office: actaully its best if the exten were numberic and the SIP user mixed - exten => _XXXX,1,Dial(SIP/USER${EXTEN},30) |
| 18:12.25 | variable_office | for logical reason i could see that, but i also needed the voicemailbox to be the same as the user, so i need everything to be numeric |
| 18:12.57 | variable_office | I cant think of anything wrong with a 000 prefix, but you obviously have considerably more experience than me |
| 18:13.51 | [TK]D-Fender | variable_office: You can add/strip prefixes as needed as you call your dialplan apps, so its no worry |
| 18:14.10 | variable_office | true |
| 18:14.32 | [TK]D-Fender | variable_office: make it as simple and straightforward as you can to math the logic of how they are named. |
| 18:16.37 | *** join/#asterisk RoyK (n=roy@ip-75-8-149-91.dialup.ice.no) |
| 18:18.35 | *** join/#asterisk anonymouz333 (n=voce@201.19.134.26) |
| 18:22.09 | *** join/#asterisk shtoom (n=shtoom@59.93.123.86) |
| 18:23.12 | blitzrage | anyone know of a working IAX2 softphone for OSX? |
| 18:23.19 | blitzrage | (yes, I've tried Zoiper) |
| 18:23.45 | shtoom | hi I am writing a small billing application. For this is it ok to depend on ANSWERED time variable set by Dial for getting accurate call duration ? |
| 18:23.47 | blitzrage | and I just tried jackeniax with no luck |
| 18:24.37 | twisted | blitzrage: uhh |
| 18:24.49 | twisted | voix? |
| 18:25.10 | blitzrage | going to update my zoiper and see if my bugs were fixed, and if not, then I'll look for that one |
| 18:25.12 | blitzrage | thx twisted |
| 18:25.35 | twisted | np |
| 18:25.43 | blitzrage | wow... maybe I should up the temperature past 18C in here... brrrrrrrrr! |
| 18:27.01 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 18:27.05 | shtoom | twisted: can you help me on this ? is it ok to depend on ANSWEREDTIME variable for getting accurate call duration? |
| 18:27.52 | twisted | shtoom: dunno |
| 18:28.23 | shtoom | twisted:Ok |
| 18:28.36 | _x86_ | what do you guys normally recommend for receiving faxes from the PSTN and making a PDF and emailing it to someone? |
| 18:29.34 | _x86_ | spandsp? hylafax+iaxmodem? |
| 18:30.35 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 18:31.10 | muiro | wow, setting up this sip trunk is so easy it blew my mind |
| 18:31.59 | [TK]D-Fender | _x86_: Hylafax + good modem/fax board + Analog line |
| 18:35.15 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
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| 18:35.48 | *** join/#asterisk Hadi-- (n=AlanS@208.113.18.155) |
| 18:35.59 | Hadi-- | hello everyone.. I have an issue and I hope someone can help me |
| 18:36.38 | Hadi-- | for some reason.. everytime I send calls from our Cisco router to any of our termination partners that are using asterisk.. the call drops after 10 seconds. |
| 18:36.47 | Hadi-- | every single partner is the same |
| 18:37.01 | Hadi-- | and it only happens for those using asterisk |
| 18:39.51 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 18:48.01 | *** join/#asterisk [N00B] (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 18:48.32 | [N00B] | if i am in queue, is it possible to make it so that I can hit 0 and be tranferred to an extension instead of waiting on hold? |
| 18:49.47 | *** join/#asterisk tru_`z24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 18:50.01 | *** join/#asterisk macros73 (n=cs@dsl093-063-226.pit1.dsl.speakeasy.net) |
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| 18:50.46 | [TK]D-Fender | [N00B]: yes, est "context=yourcontextname" to give it a plce to check for single-digits dials while waiting. |
| 18:50.54 | [TK]D-Fender | s/est/set/ |
| 18:52.13 | *** part/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 18:52.26 | *** join/#asterisk trevarthan (n=jesse@c-71-59-54-137.hsd1.ga.comcast.net) |
| 18:52.41 | [N00B] | [TK]D-Fender: i set that in the queues.conf? |
| 18:52.50 | trevarthan | Hello, is anyone here running Cepstral on a recent asterisk and OS, like Centos 5? |
| 18:53.05 | [TK]D-Fender | [N00B]: Yes, taht is for your queue's entry in queues.conf |
| 18:53.11 | [N00B] | ok thanks |
| 18:53.12 | [N00B] | will try it |
| 18:53.26 | blitzrage | [N00B]: FYI -- you would have found that in queues.conf.sample |
| 18:53.42 | trevarthan | I'm trying to get an idea of how difficult Cepstral will be to install and use. Looks kinda difficult from the howtos I've read. |
| 18:55.28 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 18:56.14 | *** join/#asterisk VoIP_Troll (n=VoIP_Tro@S010600e07d8a3740.cg.shawcable.net) |
| 18:56.33 | VoIP_Troll | Anyone here familiar with a SIP provider in Edmonton, Alberta called KBT? |
| 18:59.38 | *** join/#asterisk Corydon76-dig (i=eleven@pdpc/supporter/bronze/Corydon76-home) |
| 18:59.38 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 19:05.38 | *** join/#asterisk TheBiggestGnome (n=TbG@74-135-245-83.dhcp.insightbb.com) |
| 19:06.58 | _x86_ | in zapata.conf, do i absolutely have to make a given single channel part of a span? |
| 19:07.49 | _x86_ | I have a single channel that i want out on it's own.... dedicated context and everything... and I dont want it to be part of the channel group where all my other channels are |
| 19:09.48 | outtolunc | give it a go, i do not see why it would |
| 19:09.56 | outtolunc | matter |
| 19:11.58 | outtolunc | note: unless the signalling is different you should 'group' them separate in zapata.cnf |
| 19:12.00 | outtolunc | er conf |
| 19:12.33 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 19:12.48 | *** join/#asterisk mog (i=mog@nat/digium/x-2656a5ae20429abd) |
| 19:12.49 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:13.17 | funxion | is there anyway to play audio while doing a dial? |
| 19:13.32 | *** part/#asterisk trevarthan (n=jesse@c-71-59-54-137.hsd1.ga.comcast.net) |
| 19:13.41 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 19:13.42 | funxion | so the caller hears the audio prompt instead of ringing for one particular channel? |
| 19:14.16 | funxion | or should I rather look into queues and making a phone permanantly available |
| 19:14.20 | mog | yes |
| 19:14.25 | funxion | which one |
| 19:14.28 | mog | show application dial |
| 19:14.29 | mog | both |
| 19:14.34 | mog | or either |
| 19:14.36 | mog | they both work |
| 19:14.46 | funxion | hmm |
| 19:15.36 | *** join/#asterisk zerocod3r (n=Z3R3CoD3@b206d191.dorm.bilkent.edu.tr) |
| 19:15.43 | funxion | its weird I was asked to dial then play one file and at the end loop another file |
| 19:15.52 | funxion | know of a way to do that |
| 19:16.04 | funxion | I was thinking of just cat'ing the files into eachother |
| 19:18.06 | _x86_ | [TK]D-Fender: what's the solution if you have an analog line and don't have a faxmodem? |
| 19:18.12 | outtolunc | why wouldn't you just use teh dial G and have multiple playback's |
| 19:18.25 | _x86_ | [TK]D-Fender: for example, I just want to be able to receive inbound faxes to a PDF file, then email the PDF |
| 19:21.26 | outtolunc | oopsie, i remembered G as called, not calling |
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| 19:23.30 | *** part/#asterisk [N00B] (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 19:24.41 | *** join/#asterisk karleeto (n=karl@207.191.91.242) |
| 19:26.32 | karleeto | OK, i'm having a problem with one of my customer's call parking.. the calls get parked for a max of 1:30, then ring back the original parker; is there ANY way to make abbandoned parked calls ringback the whole callgroup instead of the origial parker ? |
| 19:27.21 | [TK]D-Fender | _x86_: GET a Faxmodem :) |
| 19:28.27 | nestAr | karleeto: don't bitch, my parked calls ring back to themselves and get disconnected |
| 19:29.41 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 19:31.11 | nohup_ | quick question... has the DigitTimeout command changed ? |
| 19:31.12 | karleeto | nestAr: meaning that your customers get disconnected if noone answers the parked call?? |
| 19:31.24 | nestAr | yea |
| 19:31.42 | [TK]D-Fender | nohup_: no longer exists. You are reading ancient stuff. |
| 19:31.50 | [TK]D-Fender | nohup_: "show function TIMEOUT" |
| 19:32.01 | nohup_ | yeah, it's hard to know if the stuff is old or not :) |
| 19:32.07 | [TK]D-Fender | nohup_: And go read all the docs in your source /docs folder |
| 19:32.16 | karleeto | nestAr: allthough, if i set the parking limit really high, that wouldn temporarily solve the problem |
| 19:32.22 | nohup_ | half of the examples on voip-info.org use ancient stuff :) |
| 19:32.25 | *** join/#asterisk Blackhold (n=Blackhol@33.Red-213-98-143.staticIP.rima-tde.net) |
| 19:32.27 | Blackhold | hello |
| 19:32.34 | [TK]D-Fender | nohup_: quite true |
| 19:32.45 | nestAr | my people just like to put people in park and forget them |
| 19:32.47 | Blackhold | how I could close or send to background the asterisk console |
| 19:32.48 | Blackhold | ? |
| 19:33.01 | nohup_ | ok, let's see if the debian people were kind enough to ship those docs with their package :) |
| 19:33.04 | nestAr | Blackhold: ctrl+c |
| 19:33.17 | [TK]D-Fender | Blackhold: How did you get to the console in the first place? |
| 19:33.29 | [TK]D-Fender | nestAr: BAD answer |
| 19:33.38 | nestAr | [TK]D-Fender: works for me. ;) |
| 19:33.50 | nestAr | TIMTOWTDI |
| 19:33.57 | [TK]D-Fender | nestAr: Yes, I've heard how well your system "works" before :) |
| 19:34.05 | Blackhold | nestAr: sure? |
| 19:34.12 | Blackhold | with that I kill asterisk |
| 19:34.13 | nestAr | Blackhold: apparently not. |
| 19:34.20 | [TK]D-Fender | Blackhold: answer my question |
| 19:34.51 | Blackhold | asterisk -gvvvvvvvc |
| 19:34.57 | Blackhold | or something like this |
| 19:35.00 | nestAr | [TK]D-Fender: Works great, cept for that whole "hanging up on people" thing... |
| 19:35.28 | [TK]D-Fender | Blackhold: You are going to have to completely stop * and run it as a daemon |
| 19:35.39 | Blackhold | wtf |
| 19:35.48 | [TK]D-Fender | Blackhold: "stop now" , then at CLI "safe_asterisk &" |
| 19:36.01 | [TK]D-Fender | Blackhold: you are running the * process INTERACTIVE |
| 19:36.04 | zerocod3r | can I get fax messages and send them in mail format to our users?I didnt find any guide or document |
| 19:36.13 | [TK]D-Fender | Blackhold: yuo cann't force it to the background now. |
| 19:36.38 | Blackhold | ok |
| 19:36.44 | Blackhold | yes I'll do |
| 19:37.10 | nestAr | in retrospect, i wish i had put the office traffic through iptables |
| 19:37.20 | nestAr | so i could block my boss's file sharing |
| 19:37.36 | *** join/#asterisk debiano778 (n=nana@213-140-19-123.fastres.net) |
| 19:37.40 | debiano778 | hi all |
| 19:37.50 | Blackhold | tk thanks |
| 19:37.54 | debiano778 | i need a little help |
| 19:37.59 | *** join/#asterisk moprilo (n=jjohn@201.198.78.23) |
| 19:38.41 | barhom | can somebody name a well-tested ata box with asterisk which is not a router as well? |
| 19:39.08 | *** join/#asterisk arguile (i=user224@KTNRON06-1242488957.sdsl.bell.ca) |
| 19:39.14 | nestAr | barhom: sipura stuff is great, but i guess they got bought by linksys. |
| 19:39.29 | nestAr | so that may be "was great" now.. |
| 19:39.33 | debiano778 | when i transfer a call on internal extension after the transfer asterisk recall the first extension and say me thi message |
| 19:39.42 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:39.43 | debiano778 | <PROTECTED> |
| 19:40.01 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:40.16 | debiano778 | is there a procedure to destroy zombie channel? |
| 19:40.17 | [TK]D-Fender | barhom: SPA-1001 |
| 19:40.53 | barhom | heh, the spa lines are linksys now yeh? |
| 19:41.18 | [TK]D-Fender | barhom: Been over a year |
| 19:41.26 | [TK]D-Fender | barhom: 2 actually. |
| 19:41.43 | barhom | no reason to get the newer spa ones fender? |
| 19:47.31 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 19:48.44 | twisted | I AM A LARGE PINEAPPLE |
| 19:48.57 | Qwell | twisted: Hello large pineapple. |
| 19:50.40 | debiano778 | anyone can help me ? |
| 19:51.21 | Corydon76-dig | twisted: I've heard things about pineapple juice..... |
| 19:52.09 | blitzrage | lol... I heard that too on Keys to the VIP |
| 19:53.50 | *** join/#asterisk RoyK (n=roy@ip-75-8-149-91.dialup.ice.no) |
| 19:54.30 | yangvnc | [TK]D-Fender: I have these parameters http://openpaste.org/en/4084/ , but I don't see anything in sip show registry |
| 19:54.35 | barhom | Im trying to get the password from my old ISP which have locked their ATA box, I tried setting up my IP address to act as the weblogin to the box and also made a telnet listen where they can write in "Login:" and "Password:". Then I tried calling them and told them to change volume on the ata box, thought they would need to login to do this |
| 19:54.35 | Nugget | telnet is eeeeeeevil! |
| 19:54.51 | barhom | but instead he said he changed some config file which the ata box would get on restart :/ |
| 19:55.48 | barhom | i.e. Im trying to steal the password by pretending my ata box is connected and simulate the login for them to write the user/pass, but, he changed the config and my work is kinda for nothing - isnt there anything where he would have to login to the box to change maybe? |
| 19:55.56 | [TK]D-Fender | yangbecause you magled where everything is locatd. your register has to be AFTER [general} and BEFORE any other section. |
| 19:56.13 | _Sam-- | barhom: quit wasting your time and get an unlocked ATA? |
| 19:56.23 | _Sam-- | you've probably wasted more dollars of your time, than a new ATA would cost. |
| 19:56.24 | barhom | _Sam--: its worth a try ;) |
| 19:56.40 | barhom | Ive wasted 1hour to do this, it was fun and educating for making a tcp listen on port 23 |
| 19:56.41 | _Sam-- | or, search google for the ATA model / SIP provider and find the default pass |
| 19:56.57 | barhom | they have changed it _Sam--, Ive tried that |
| 19:57.51 | _Sam-- | or, skip an ATA altogether, and just use xlite or a cheap SIP phone |
| 19:58.23 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:59.03 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:59.37 | barhom | using cheap sip phones means I need to put new ethernet cables to the place I want the phone, or buy wlan sip phones = more money |
| 19:59.43 | barhom | Ill just go buy an ATA I guess |
| 20:00.04 | barhom | xlite is fine for me, but there is other ppl who cant use xlite here |
| 20:00.36 | nestAr | wlan phones suck anyway |
| 20:00.39 | *** join/#asterisk andrebarbosa (n=andrebar@83.240.149.156) |
| 20:01.45 | [TK]D-Fender | barhom: Or you do the smart thing and plug it in at your home's demarc and you whole plce shares it. |
| 20:02.02 | barhom | demarc? |
| 20:02.08 | Qwell | ~demarc |
| 20:02.12 | Qwell | stupid bot |
| 20:02.20 | yangvnc | [TK]D-Fender: as always, thanks a lot seems to work....now I have sip accounts registered on asterisk which connect actually to another asterisk....but I can use the same login/pass and my asterisk IP with my ekiga phone? |
| 20:02.21 | [TK]D-Fender | barhom: the point where you home's wiring comes in from the exterior |
| 20:02.22 | Qwell | demarcation point - where the cable comes in from your telco |
| 20:02.38 | barhom | [TK]D-Fender: yeh thats what I have now with the current locked ata |
| 20:02.43 | barhom | internet -> ata -> dmarc |
| 20:02.57 | barhom | its working fine, but the provider stinks |
| 20:03.02 | [TK]D-Fender | barhom: then well... DUH <-- |
| 20:03.06 | barhom | yeh |
| 20:03.27 | *** join/#asterisk klasstek (n=karl@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 20:04.00 | mvanbaak | any good ref doc to setup res_jabber etc ? |
| 20:04.12 | Qwell | mvanbaak: RTFSC :) |
| 20:04.18 | mvanbaak | lol |
| 20:04.21 | mvanbaak | ok |
| 20:04.29 | Qwell | no, but the sample configs have a lot in them |
| 20:05.07 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 20:05.11 | l0verb0y | hey has anyone used hylafax? |
| 20:05.13 | mvanbaak | k |
| 20:05.41 | Qwell | I should update my gtalk+jingle merge branch... |
| 20:06.23 | Qwell | afaik, gtalk actually conforms to the jingle spec now |
| 20:08.33 | yangvnc | [TK]D-Fender: i mean how do I forward calls then from asterisk to my ekiga client? |
| 20:08.52 | [TK]D-Fender | yangvnc: Just work on this a while. |
| 20:10.04 | _Sam-- | nestAr: who said anything (before) about a wlan phone? |
| 20:10.10 | _Sam-- | <nestAr> wlan phones suck anyway |
| 20:10.25 | _Sam-- | oh sorry i didnt read up far enough. |
| 20:10.26 | _Sam-- | my fault |
| 20:10.57 | nestAr | k |
| 20:12.55 | yangvnc | [TK]D-Fender: must there always be also host=dynamic beside host=IP ? |
| 20:13.14 | [TK]D-Fender | yangvnc: No, its one or the other |
| 20:13.34 | yangvnc | <PROTECTED> |
| 20:15.03 | karleeto | OK, i'm having a problem with one of my customer's call parking.. the calls get parked for a max of 1:30, then ring back the original parker; is there ANY way to make abbandoned parked calls ringback the whole callgroup instead of the origial parker ? |
| 20:15.29 | *** join/#asterisk bmg505 (n=leon@196.209.176.47) |
| 20:15.53 | *** join/#asterisk Bl0w_M0nk (n=ak@adsl-69-215-128-147.dsl.milwwi.ameritech.net) |
| 20:19.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 20:27.28 | [TK]D-Fender | yangvnc: When you set a FIXED address with HOST you shouldn't be registereing |
| 20:32.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:32.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:35.01 | *** join/#asterisk rene- (n=rene@c75-109-208-108.bcstcmta01.clsttx.tl.dh.suddenlink.net) |
| 20:35.15 | rene- | hello, i am having problems getting zaptel networking to work |
| 20:35.46 | rene- | i have compiled the kernel according to information floating on the net but the compilation of zaptel fails |
| 20:36.01 | r0d3nt | error ? |
| 20:36.04 | _Sam-- | what distribution? |
| 20:36.04 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 20:36.07 | _Sam-- | debian? |
| 20:36.10 | [TK]D-Fender | networking? |
| 20:36.23 | [TK]D-Fender | rene-: and need I say...PASTEBIN!?? |
| 20:36.51 | rene- | iheheh |
| 20:37.03 | rene- | D-Fender: HDLC |
| 20:37.13 | Blackhold | I also wants to activate zaptel |
| 20:37.17 | Blackhold | I'm gonna start now |
| 20:37.21 | Blackhold | I have asterisk running |
| 20:37.34 | Blackhold | and I could access to my user and make a dial call to 500 |
| 20:37.34 | [TK]D-Fender | rene-: you mean mixed voice/data over T1? |
| 20:37.58 | rene- | mostly one t1 port for data and another for voice |
| 20:39.23 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:40.05 | rene- | D-Fender: http://www.pastebin.ca/800878, |
| 20:40.27 | rene- | compilatin stops at zaptel-base.c |
| 20:40.39 | rene- | s/compilatin/compilation |
| 20:41.37 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:42.24 | rene- | i can compile the sethdlc-new utility and i believe i am correctly compiling the kernel with raw hdlc support if i go to the source now i found that make menuconfig shows me that hdlc raw is enabled |
| 20:43.33 | rene- | Sam: Centos 5. kernel 2.6.23 zaptel 1.4.5.1 or 1.4.7 |
| 20:45.18 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582979.dsl.bell.ca) |
| 20:46.56 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 20:47.26 | zapp-branigan | hello i'm looking for the g723.1 in digium but i can't find please where can i buy a license for g723.1 ? |
| 20:48.09 | file | Digium does not sell licenses for it. |
| 20:50.20 | zapp-branigan | :( is someone who shell licenses for asterisk or diguium will sell licenses in the future ? |
| 20:50.51 | file | the transcoder card does G723 |
| 20:51.00 | Blackhold | how I could know if my X100P is working? |
| 20:51.05 | Blackhold | or asterisk is detecting it? |
| 20:51.21 | [TK]D-Fender | Blackhold: Have you considered actaully placing or receiveing calls on it yet? |
| 20:51.42 | rene- | Blackhold: you can run lsmod and see if you find the module for it |
| 20:51.55 | hmmhesays | dialstatus should be chanunavail if there are no open lines in your zap group right? |
| 20:52.09 | [TK]D-Fender | hmmhesays: Should |
| 20:52.52 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 20:52.53 | hmmhesays | i need a sound file that says all channels are in use |
| 20:54.17 | [TK]D-Fender | hmmhesays: all-circuits-busy-now.gsm |
| 20:54.22 | Blackhold | mmm, if I use that url http://urbanita:8088/asterisk/static/config/setup/install.html |
| 20:54.26 | [TK]D-Fender | all-outgoing-lines-unavailable.gsm |
| 20:54.28 | rene- | i compiled a kernel with hdlc support, or so i believe. if i go again to menuconfig i see hdlc support checked, however the .config file knows nothing about it? why? |
| 20:54.42 | Blackhold | in verify analog ports it says me that: No Analog ports have been detected on your system |
| 20:55.01 | rene- | oh no actually they are in there |
| 20:55.08 | [TK]D-Fender | rene-: flush your folder and re-extract and redo ./configure |
| 20:55.17 | rene- | but as modules |
| 20:55.17 | Blackhold | I'm writing this manual, and what I do is there: http://blackhold.blogspot.com/2007/11/asterisk-asterisk-gui-x100p.html |
| 20:55.26 | hmmhesays | where is the file in the source that lists all the sounds |
| 20:55.34 | rene- | does not have a y in it but an m? should i modprobe them ? |
| 20:56.11 | [TK]D-Fender | hmmhesays: "ls -l /var/lib/asterisk/sounds" <--- |
| 20:57.55 | rene- | here is my relevant .config section http://www.pastebin.ca/800906 |
| 20:59.17 | [TK]D-Fender | rene-: trash your libPRI & zaptel, download the latest of each and rebuild them both in order |
| 21:00.15 | rene- | ok |
| 21:00.20 | funxion | any reason for "is not valid for automatic login of agent " |
| 21:00.46 | [TK]D-Fender | funxion: Static agent |
| 21:00.58 | funxion | something like that |
| 21:01.05 | [TK]D-Fender | funxion: or better phrased "member" |
| 21:01.12 | funxion | Im using a queue for the frontdesk phone |
| 21:01.26 | [TK]D-Fender | funxion: My condolences to your receptionist :) |
| 21:01.34 | funxion | thnx |
| 21:01.42 | funxion | at least its abe 1-2 |
| 21:01.48 | funxion | and not 1.0 |
| 21:02.22 | Blackhold | zaptel detects me the X100P, but asterisk no :( |
| 21:02.23 | funxion | I'm doing the exact same type of queue on a 1.4 box and it works perfectly |
| 21:02.29 | Blackhold | could someone help me please? |
| 21:02.32 | funxion | not sure why this isnt working |
| 21:02.59 | [TK]D-Fender | Blackhold: After compiling Zaptel you have to do Asterisk all over again so that chan_zap compiles |
| 21:03.52 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 21:04.10 | _Sam-- | soudns familiar. |
| 21:04.13 | funxion | [TK]D-Fender http://pastebin.ca/800920 *53 should log user in and out |
| 21:06.28 | [TK]D-Fender | funxion: exten => *53,3,GotoIf(${autoattendant} = "1"?frontdesk_logoff,s,1) <-- missing $[] |
| 21:06.38 | funxion | that part works |
| 21:08.03 | [TK]D-Fender | funxion: your logoff : exten => s,2,Dial(Local/*99@frontdesk_logoff/n,,D(${CALLERID(num)}#${CALLERID(num)}##)) |
| 21:08.07 | *** join/#asterisk BCS-Satori (n=BCS-Sato@68.34.28.26) |
| 21:08.16 | [TK]D-Fender | funxion: should be : exten => s,2,Dial(Local/*99@frontdesk_logoff/n,,D(${CALLERID(num)}##)) |
| 21:09.03 | [TK]D-Fender | funxion: you have to enter the box, then "#" instead of telling it where to log you in so as to log you out |
| 21:09.44 | rene- | my compilation breaks again at the same spot |
| 21:10.01 | funxion | ahh |
| 21:10.02 | funxion | thnx |
| 21:10.29 | yangvnc | 21:27 < [TK]D-Fender> yangvnc: When you set a FIXED address with HOST you shouldn't be registereing , what exactly did you mean ? I can somehow use the same user/pass in my SIP client, than it is used on asterisk. |
| 21:11.52 | funxion | [TK]D-Fender the first thing you found was the problem it wont logoff |
| 21:11.54 | rene- | i am going try to downgrade to the kernel version the guide say it was made on |
| 21:12.08 | funxion | i changed it to exten => *53,3,GotoIf($["${autoattendant}" = "1"]?frontdesk_logoff,s,1) |
| 21:12.10 | *** join/#asterisk BCS-Satori (n=BCS-Sato@68.34.28.26) |
| 21:12.40 | *** join/#asterisk icewaterman (n=immagine@i577B8B67.versanet.de) |
| 21:12.50 | rene- | because |
| 21:12.52 | rene- | #if LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,3) |
| 21:12.52 | rene- | #define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT |
| 21:12.52 | rene- | #endif |
| 21:13.09 | rene- | according to zaptel/zconfig.h |
| 21:13.48 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 21:14.54 | hmmhesays | now that is fscking retarded, if there are no channels available in your zap group it returns congestions |
| 21:14.58 | hmmhesays | *congestion |
| 21:15.03 | *** join/#asterisk UserReg_CL (n=dede@200.113.137.78) |
| 21:15.09 | UserReg_CL | Helpme with AGI |
| 21:15.10 | *** join/#asterisk illc0mm (n=bill@rrcs-67-78-138-116.se.biz.rr.com) |
| 21:16.03 | [TK]D-Fender | UserReg_CL: Try asking a SPECIFIC question with DETAILS this time without simply quitting like last time and maybe we'll see... |
| 21:16.21 | UserReg_CL | hi Fender :) |
| 21:18.34 | UserReg_CL | here.. is very easy: http://pastebin.com/m7424111c .. but not work :( |
| 21:19.56 | illc0mm | Hey, I'm trying to track down a strange SIP issue I'm seeing where when I have a call established on a SIP "trunk" (bandwidth.com) and I receive an inbound call from bandwidth.com, the previous call is terminated as the new call is created, and it appears to be terminated by Asterisk (the Bye command looks to originate from asterisk) according to the SIP debug, but I can't find out why. |
| 21:20.15 | [TK]D-Fender | UserReg_CL: What ver of *? |
| 21:20.46 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:21.27 | UserReg_CL | what line ? |
| 21:22.20 | BCS-Satori | core show version |
| 21:22.56 | [TK]D-Fender | UserReg_CL: WHAT VERSION OF ASTERISK <---- |
| 21:23.49 | UserReg_CL | sorry... |
| 21:24.10 | UserReg_CL | 1.2.9.1 |
| 21:24.25 | UserReg_CL | not remember what * == asterisk :) lol |
| 21:24.26 | hmmhesays | now I need a phone number that returns busy |
| 21:24.44 | TheBiggestGnome | You guys haven't hear of Virtual VoIP phones or anything have you? I'm more or less trying to educate myself in PBX systems but I dont have the room or money to setup multiple phones in a lab and whatnot. I suppose what I'm looking for is some kind of Virtual Phone. |
| 21:26.20 | BCS-Satori | TheBiggestGnome: setup a VM machine and software based phones on the system, no extra hardare or software needed. |
| 21:26.20 | [TK]D-Fender | UserReg_CL: Pastebin an attempt to use this. |
| 21:26.20 | [TK]D-Fender | UserReg_CL: a COMPLETE CALL, with AGI & core debug on max |
| 21:26.20 | TheBiggestGnome | BCS-Satori: I have a vm machine already |
| 21:26.33 | TheBiggestGnome | any recomendations on software based phones? |
| 21:26.46 | illc0mm | TheBiggestGnome: x-ten lite is a commonly used one |
| 21:26.47 | BCS-Satori | TheBiggestGnome: I use Counterpaths X-Lite personally its free |
| 21:26.53 | illc0mm | or x-lite, heh |
| 21:26.56 | BCS-Satori | lol |
| 21:26.59 | [TK]D-Fender | ~softphone |
| 21:26.59 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam |
| 21:27.01 | BCS-Satori | ya same thing |
| 21:27.07 | Qwell | ~bria |
| 21:27.08 | jbot | [~bria] Bria is a NON-free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ This includes licensed audio & video codecs and is full-featured. |
| 21:27.08 | illc0mm | sorry, got HA and phones on the brain I guess |
| 21:27.20 | Qwell | bria != eyebeam? |
| 21:27.32 | [TK]D-Fender | Qwell: Nope |
| 21:27.33 | Qwell | ~eyebeam |
| 21:27.34 | jbot | [~eyebeam] eyeBeam is a NON-free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ This includes licensed audio & video codecs and is full-featured. |
| 21:27.41 | UserReg_CL | Fender: what ? |
| 21:27.43 | Qwell | well...that seems pointless, don't ya think? |
| 21:27.49 | illc0mm | hyah |
| 21:27.53 | BCS-Satori | Qwell: /chuckle |
| 21:28.01 | [TK]D-Fender | UserReg_CL: What part of my request is not blatantly obvious? |
| 21:28.12 | TheBiggestGnome | Wow, thanks guys. Here I was thinking I was going to have to go out and buy 4 SIP phones or something. |
| 21:28.20 | UserReg_CL | { a COMPLETE CALL, with AGI & core debug on max } ? |
| 21:28.41 | PTorres | ~twinkle |
| 21:28.41 | jbot | [~twinkle] Twinkle is an OSS SIP phone for KDE/Qt systems and can be downloaded at http://www.twinklephone.com |
| 21:28.43 | BCS-Satori | TheBiggestGnome: np there are a lot of "premade" vm images out there as well custom asterisk ones as well as asterisknow and trixbox |
| 21:28.53 | [TK]D-Fender | UserReg_CL: I want to see the entire call where this setup of yours FAILS. What don't you understand? |
| 21:29.36 | UserReg_CL | { a COMPLETE CALL, with AGI & core debug on max } How? |
| 21:29.42 | [TK]D-Fender | TheBiggestGnome: You probably SHOULD. Softphones usually lack a lot of features, you'd have to buy a headset anyways, and quality = iffy. |
| 21:29.46 | TheBiggestGnome | BCS-Satori: I just installed AseriskNow in VMware |
| 21:30.54 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 21:31.05 | illc0mm | TheBiggestGnome: CentPBX is a new one that uses freepbx as a front end. it looks pretty promising from a easy to setup / less fluff perspective |
| 21:31.10 | BCS-Satori | BCS-Satori: I used Asterisknow to learn about the system in vm same way |
| 21:31.21 | BCS-Satori | err |
| 21:31.22 | BCS-Satori | ... |
| 21:31.27 | BCS-Satori | why message myself? |
| 21:31.36 | illc0mm | BCS-Satori: hah |
| 21:31.46 | iratik | how do i turn verbosity from 10 to 3? |
| 21:31.48 | BCS-Satori | TheBiggestGnome: The new trixbox beta has centos 5 and asterisk 1.4 installed as well |
| 21:32.26 | TheBiggestGnome | ok, good to know |
| 21:32.38 | TheBiggestGnome | I think this is one of the more helpful channels I've been in for a while |
| 21:32.39 | TheBiggestGnome | heh' |
| 21:32.59 | yangvnc | Tell me, If I have account on VOIP provider and i set it up also on my asterisk, how do i use it also on my SIP client ? |
| 21:33.15 | yangvnc | SIP client =softphone |
| 21:33.34 | illc0mm | yangvnc: It's possible, but it might be up to your VOIP provider if it's supported |
| 21:33.53 | BCS-Satori | yangvnc: are you asking if its a voip trunk provider? or an extension based voip provider? |
| 21:34.05 | illc0mm | BCS-Satori: yeah, good point |
| 21:34.24 | yangvnc | BCS-Satori: no, normal VOIP provider which gives out one sip account |
| 21:34.42 | BCS-Satori | yangvnc: Have you made a user/extension for the softphone? |
| 21:34.51 | yangvnc | BCS-Satori: as I understood, my asterisk in the middle can serve as the answering machine, when I am not connect4d with a SIP client = softphone |
| 21:35.16 | yangvnc | Yes, I made everything, and I get registration failed with ekiga |
| 21:35.48 | BCS-Satori | yangvnc: the soft phone setup is really no different then setting up a phsyical phone, if the software phone isnt "registerd" meaning running then asterisk will move the call to the next location typically voicemail if you have it set that way unless you use followme which then can go to a cell phone or lan line |
| 21:36.25 | UserReg_CL | now ... |
| 21:36.35 | UserReg_CL | full debug: http://pastebin.com/m57c25b1e |
| 21:36.41 | UserReg_CL | please helpme with agi perl |
| 21:37.26 | BCS-Satori | yangvnc: take a working user, mimik the settings with a new extension/password and tell the software phone to use that extn/pass with the proxy being asterisk server, |
| 21:37.36 | yangvnc | BCS-Satori: so I need to point my ekiga phone to my asterisk, even that my asterisk is connected with that account to VOIP provider allready |
| 21:38.37 | BCS-Satori | yangvnc: think of it this way, extension 1 through 10 registers to asterisk then registers to voip provider. all 10 extensions use the same provider |
| 21:39.02 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:39.56 | BCS-Satori | yangvnc: trying to think of better wording |
| 21:41.08 | BCS-Satori | yangvnc: what you think is correct, asterisk acts as the middle man, allowing the phones to know how to communicate with the provider |
| 21:41.30 | *** join/#asterisk salviadud (i=ralfalfa@voss.dreamhost.com) |
| 21:42.00 | salviadud | guys, I'm having trouble registering a sip client, because the usual 5060 udp port is blocked at my school |
| 21:42.05 | salviadud | i've heard that skype works |
| 21:42.12 | salviadud | do you know which udp port skype uses? |
| 21:43.05 | hmmhesays | what is the easiest way to record all phone calls placed through asterisk? |
| 21:43.12 | BCS-Satori | salviadud: http://www.skype.com/help/guides/firewall.html |
| 21:43.24 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 21:43.25 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 21:43.25 | bkw_ | hmmhesays: I would have expected you of all people to already know that one |
| 21:43.26 | TheBiggestGnome | now I suppose its a matter of setting up x-lite correctly heh |
| 21:43.50 | hmmhesays | bkw_: i've always just used mixmonitor in the dialplan |
| 21:43.56 | BCS-Satori | TheBiggestGnome: before setting up xlite you need to make extensions to register too in asterisk |
| 21:43.57 | *** join/#asterisk Schumie (n=Steve@212.183.136.194) |
| 21:43.57 | hmmhesays | I was wondering if there was something new and fancy |
| 21:43.58 | bkw_ | thats about the best way |
| 21:44.33 | bkw_ | hmmhesays: we have something in freeswitch that does that |
| 21:44.45 | bkw_ | puts the callers in each ear..so you can really tell who is speaking |
| 21:45.02 | hmmhesays | I haven't had a chance to play with fs in awhile |
| 21:45.31 | zapp-branigan | hi asterisk can send dtmf defore some time ? for send the number to call in a prepaid system ? |
| 21:45.49 | salviadud | thanx |
| 21:46.35 | rene- | ok i have advanced a little bit with my zaptel-wan thing, it turns out that i did have compiled hdlc support in the kernel but they were modules waiting to be loaded |
| 21:47.38 | rene- | now, i am geting an error when i try to compile hdlc (WAN) support in zaptel, if i dont try to compile hdlc zaptel will compile fine but if i try then it wont http://www.pastebin.ca/800969 |
| 21:49.22 | rene- | help please? |
| 21:49.53 | zapp-branigan | hi asterisk can send dtmf defore some time ? for send the number to call in a prepaid system ? |
| 21:50.15 | BCS-Satori | ^^ |
| 21:50.57 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 21:51.06 | teknoprep | does anyone here recomend a softphone that works with video ? |
| 21:51.08 | teknoprep | and asterisk |
| 21:51.13 | teknoprep | and say a windows pc with a webcam |
| 21:52.33 | *** join/#asterisk santor_ (n=xidarian@pool-71-180-154-74.tampfl.fios.verizon.net) |
| 21:52.36 | bkw_ | teknoprep: eyebeam |
| 21:53.00 | BCS-Satori | ~eyebeam |
| 21:53.01 | jbot | [~eyebeam] eyeBeam is a NON-free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ This includes licensed audio & video codecs and is full-featured. |
| 21:54.01 | funxion | does anyone know if the announceoverride feature in the command queue plays a message once then continues to play the specified moh? |
| 21:54.09 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 21:54.18 | santor_ | exit |
| 21:54.21 | santor_ | oops |
| 21:54.52 | teknoprep | ahh so x-lite can doit? |
| 21:55.08 | teknoprep | nvm i see eyebeam |
| 21:55.12 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:56.08 | illc0mm | I saw a debug of a sip conversation a while ago that was a ASCII art representation of the call setup. Anyone know what produces that? |
| 21:56.49 | teknoprep | for video to work... do you have to be connected to the same asterisk box ? |
| 21:58.53 | *** join/#asterisk Schumie (n=Steve@cpc1-oxfd2-0-0-cust25.oxfd.cable.ntl.com) |
| 21:59.39 | *** part/#asterisk BCS-Satori (n=BCS-Sato@68.34.28.26) |
| 21:59.41 | *** join/#asterisk BCS-Satori (n=BCS-Sato@68.34.28.26) |
| 22:00.06 | funxion | does anyone know how to get * to continue platying moh instead of ringing when dialing an agent? |
| 22:01.13 | nestAr | funxion: isn't that a function in the queues.conf file? |
| 22:01.31 | funxion | thas what i thought but its not working |
| 22:01.37 | [TK]D-Fender | funxion, pastebin your setup and I'll tell you what you're doing wrong :) |
| 22:01.55 | Corydon76-dig | funxion: drop the r flag from your call of the Queue app |
| 22:02.14 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:03.21 | *** join/#asterisk Aughey (n=jha@64.219.54.125) |
| 22:03.31 | *** join/#asterisk ForSaleSticker (i=chatzill@nat/yahoo/x-e534ab27d1178e6c) |
| 22:04.13 | Aughey | When I turn on DTM logged, I get a message saying "DTMF end accepted without begin...." This doesn't seem right. Before that I get "DTMF END'1' received on...... duration 0ms" |
| 22:04.18 | funxion | Corydon76-dig Im not using the r flag |
| 22:04.22 | Aughey | DTMF logging that is |
| 22:04.24 | funxion | using tn |
| 22:04.29 | funxion | [TK]D-Fender http://pastebin.ca/800986 |
| 22:05.13 | Yourname`` | Hi. What kind of settings in sip.conf can affect the inner workings of DTMF such that DTMF doesn't work? |
| 22:05.42 | funxion | I think indications.conf is where that is defined |
| 22:05.50 | Corydon76-dig | funxion: you're not using the Local channel to log in Queue members, are you? |
| 22:06.05 | funxion | ? |
| 22:06.11 | [TK]D-Fender | funxion, pastebin a complete call |
| 22:06.15 | funxion | I think I may be |
| 22:06.24 | funxion | complete call of caller going to queue? |
| 22:06.39 | ForSaleSticker | Hi - |
| 22:07.02 | ForSaleSticker | anyone used freepbx gui |
| 22:07.12 | funxion | crap |
| 22:07.14 | Corydon76-dig | ~freepbx |
| 22:07.15 | 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 |
| 22:07.36 | ForSaleSticker | thanks |
| 22:08.07 | funxion | [TK]D-Fender I updated it |
| 22:08.10 | funxion | its at the bottom |
| 22:08.16 | [TK]D-Fender | funxion, LINK |
| 22:08.23 | funxion | : http://pastebin.ca/800992 |
| 22:08.30 | Yourname`` | Hi. What kind of settings in sip.conf can affect the inner workings of DTMF such that DTMF doesn't work? http://pastebin.ca/800993 -> SIP.conf |
| 22:08.40 | funxion | although i found that the announce overide plays to the agent |
| 22:10.10 | funxion | guess I didnt read |
| 22:10.20 | [TK]D-Fender | funxion, :) |
| 22:10.25 | funxion | thats one |
| 22:10.30 | funxion | but the moh never plays |
| 22:10.42 | funxion | if I use the agent phone to place a call |
| 22:11.08 | funxion | then place a call into the queue from another phone it rings the agent phone regardless if its on a call or not |
| 22:11.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:12.17 | [TK]D-Fender | funxion, Ah yes, well a callback agent is not something * will know is on the phone or not. |
| 22:12.18 | Navion | Has anyone ever gotten a sound to play on a paging speaker? I've tried most fo the WiKi suggested methods and no joy... |
| 22:12.45 | [TK]D-Fender | funxion, thats the problem with them. "show application chanisavail" <-- ;) |
| 22:13.05 | funxion | ahh |
| 22:14.00 | [TK]D-Fender | funxion, give it a good read and a few swings and let me know if you need a hand after :) |
| 22:14.31 | funxion | ok |
| 22:14.40 | funxion | but do I scrap the queue idea? |
| 22:14.48 | funxion | and write a make shift queue |
| 22:14.53 | ForSaleSticker | i run cat /etc/asterisk/agents.conf and it gives gobldy gook - like a corrupt file. Is there a way to regenerate? |
| 22:15.05 | [TK]D-Fender | funxion, nope, you'll want to maintain priority.... |
| 22:15.22 | [TK]D-Fender | funxion, and its the only way to maintain steady MoH w/o driving your caller nuts |
| 22:15.43 | funxion | not sure i understand what you mean |
| 22:16.02 | [TK]D-Fender | funxion, Forget the DIY queue and jsut mod up your callback :) |
| 22:16.21 | [TK]D-Fender | funxion, I dealt with this years ago |
| 22:17.07 | funxion | mod the callbacklogin? |
| 22:17.16 | funxion | i think I see what your saying |
| 22:17.34 | *** join/#asterisk RoyK (n=roy@ip-75-8-149-91.dialup.ice.no) |
| 22:17.56 | *** join/#asterisk Hadi-- (n=AlanS@208.113.18.155) |
| 22:17.58 | Hadi-- | hello |
| 22:18.02 | *** join/#asterisk troy- (i=20b4e9e7@bgp4.us) |
| 22:18.13 | troy- | are there any sip/iax clients for BlackBerry phones? |
| 22:18.37 | [TK]D-Fender | funxion, change the exten that gets dialed to check if the device is on a call or not |
| 22:18.48 | funxion | Im doign that |
| 22:18.50 | funxion | thnx |
| 22:18.55 | [TK]D-Fender | funxion, which means don't point your loging to a context used for anything else. |
| 22:19.08 | *** part/#asterisk rene- (n=rene@c75-109-208-108.bcstcmta01.clsttx.tl.dh.suddenlink.net) |
| 22:19.09 | [TK]D-Fender | funxion, Which yeah, will involve a tiny amount of duplication |
| 22:19.09 | funxion | got ya |
| 22:19.34 | [TK]D-Fender | ForSaleSticker, copy over the sample from your source configs folder |
| 22:20.24 | troy- | any thoughts [TK]D-Fender ? |
| 22:20.39 | [TK]D-Fender | troy-, nope |
| 22:20.40 | ForSaleSticker | it's just gunk - ÓECÝÁÉt TL²t1oqÜxµÿîrðî²w´ÍÔøôãìf¢0.~¸ï#î1éC ÓÞ`ÿücÛ8~óúu«þñåÉÛÞ`Ð?Ä8Ó,d |
| 22:20.41 | ForSaleSticker | FN³tU!p®æ÷Kqw+qÏ»¯¸k{FÙýxîvü×¥¥/ |
| 22:20.43 | ForSaleSticker | sZ;-ÈØí |
| 22:20.44 | ForSaleSticker | <PROTECTED> |
| 22:20.46 | ForSaleSticker | [andy@asterisk1 asterisk]$ PuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTYPuTTY |
| 22:20.52 | [TK]D-Fender | ForSaleSticker, I jsut told you what to do. REPLACE IT. |
| 22:21.30 | ForSaleSticker | gotcha |
| 22:22.44 | twisted | uhm |
| 22:23.03 | *** part/#asterisk killfill (n=killfill@200.55.220.3) |
| 22:23.09 | ForSaleSticker | what would cause the corrupt file - should I take steps to prevent it. |
| 22:24.32 | funxion | http://pastebin.ca/801012 |
| 22:24.37 | funxion | [TK]D-Fender |
| 22:24.46 | funxion | I dont think I did it right |
| 22:24.48 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:25.54 | [TK]D-Fender | funxion, where do you see it checking if they're on the phone? |
| 22:26.04 | funxion | at the bottom |
| 22:26.11 | funxion | callback context? |
| 22:26.23 | funxion | although i changed hangup to congestion |
| 22:26.53 | [TK]D-Fender | exten => *99,1,AgentCallbackLogin(||${CALLERID(num)}@frontdesk) <------------------ |
| 22:27.03 | funxion | I changed that too |
| 22:27.04 | funxion | sry |
| 22:27.07 | funxion | forgot to update |
| 22:27.11 | funxion | taht part |
| 22:27.24 | funxion | its now exten => *99,1,AgentCallbackLogin(||${CALLERID(num)}@callback) |
| 22:27.26 | [TK]D-Fender | funxion, you can also very clearly see that the first app it calls is DIAL. |
| 22:27.37 | [TK]D-Fender | funxion, You have to logout and back in |
| 22:27.44 | funxion | did that |
| 22:27.50 | [TK]D-Fender | funxion, FAILURE |
| 22:27.54 | [TK]D-Fender | somethings not right |
| 22:28.02 | funxion | first app it calls is dial? |
| 22:28.03 | funxion | where |
| 22:28.15 | funxion | exten => _3083,1,ChanIsAvail(SIP/3083,s) |
| 22:28.15 | funxion | exten => _3083,2,Dial(${availchan},30,t) |
| 22:28.22 | [TK]D-Fender | line #91 |
| 22:28.31 | [TK]D-Fender | read your own pastebin :) |
| 22:28.46 | funxion | I have to redo the pastebin |
| 22:28.48 | funxion | I've updated |
| 22:28.49 | funxion | sry |
| 22:28.52 | funxion | give me a sec |
| 22:28.54 | [TK]D-Fender | ......... |
| 22:29.16 | Yourname`` | DTMF help please :( |
| 22:29.23 | Yourname`` | What kind of settings in sip.conf can affect the inner workings of DTMF such that DTMF doesn't work? http://pastebin.ca/800993 -> SIP.conf |
| 22:29.24 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 22:31.03 | funxion | [TK]D-Fenderhttp://pastebin.ca/801024 |
| 22:32.57 | [TK]D-Fender | funxion, Executing Dial("Local/3083@callback-f290,2", "SIP/3083-082d7498|30|t") in new stack |
| 22:33.04 | [TK]D-Fender | umm... no comment :) |
| 22:33.57 | funxion | I guess Im not using the chanisavail properly |
| 22:34.06 | hmmhesays | and the chord of the day is Eb in standard tuning, open D shape |
| 22:34.12 | [TK]D-Fender | funxion, No, your DIAL is bad |
| 22:34.58 | *** part/#asterisk Blackhold (n=Blackhol@33.Red-213-98-143.staticIP.rima-tde.net) |
| 22:35.03 | [TK]D-Fender | hmmhesays, starts on 6th fret on A |
| 22:35.17 | hmmhesays | yeah thats easy though |
| 22:35.42 | funxion | Dial(${CUT(AVAILCHAN||1)}) |
| 22:35.43 | funxion | ? |
| 22:35.57 | [TK]D-Fender | funxion, don't use the var, just use the normal device |
| 22:36.18 | muiro | here I'm going to ask for a little advice from my experienced telephony people. I'm looking to host an IVR that's probably going to get, at peak, 10 concurrent calls. All incoming. For testing and demo purposes I've gotten a sip service to call in from a local number and it's working great. I was originally going to buy one of digium's T1 cards, but I'm thinking now that it might be more economical to just use the sip service. Considering |
| 22:36.50 | [TK]D-Fender | hmmhesays, only place you can play it.... |
| 22:37.03 | hmmhesays | [TK]D-Fender: that is not true |
| 22:37.11 | [TK]D-Fender | hmmhesays, sorry.. 18th? |
| 22:37.11 | funxion | [TK]D-Fender I dont get you |
| 22:37.15 | [TK]D-Fender | hmmhesays, ouch |
| 22:37.33 | hmmhesays | muiro: it is going to cost considerably more to have a t1 come in for just 10 calls |
| 22:37.55 | [TK]D-Fender | funxion, Dial(SIP/3083|30|t) |
| 22:38.07 | Corydon76-dig | hmmhesays: frac T1 PRI |
| 22:38.08 | muiro | hmmhesays: yeah, that's what I'm thinking. I was just wondering if there might be any downsides I'm not visualizing |
| 22:38.29 | [TK]D-Fender | funxion, your dial itself should be normal and you should be checking the CONTENT of ${AVAILCHAN} to see if its clear to go |
| 22:38.39 | *** part/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 22:38.53 | hmmhesays | http://www.xs4all.nl/~hippy/chords/data/c757.htm |
| 22:38.56 | hmmhesays | there is another way |
| 22:38.58 | Corydon76-dig | The only major downside is that your SIP provider is using a shared medium, so you're at the mercy of your ISPs uplink |
| 22:39.19 | funxion | thnx [TK]D-Fender |
| 22:39.21 | [TK]D-Fender | hmmhesays, umm... I am just slow today :) |
| 22:39.29 | Corydon76-dig | If your ISP's uplink goes down, has trouble, or is flooded, your calls will not work |
| 22:39.47 | hmmhesays | its ok I forgive you |
| 22:39.50 | [TK]D-Fender | hmmhesays, must be the fact my arm is completely shot. |
| 22:40.24 | Corydon76-dig | And while you could still have the problem of your T1 going down, at least it's your T1 that you can get the telco to fix |
| 22:40.56 | Corydon76-dig | If some script kiddie is hell bent on knocking over your ISP, there isn't a thing you can do until the onslaught stops |
| 22:41.22 | [TK]D-Fender | hmmhesays, I love 4-5 string power chords over D-shape. since I play a LOT in E I use 02245X a lot and the top 4 on anything transposed |
| 22:41.35 | Corydon76-dig | muiro: see why you might want a T1? |
| 22:41.53 | bkruse | Corydon76-dig: anyone with a botnet can take down anything |
| 22:42.10 | Corydon76-dig | bkruse: botnets can't take down PSTN T1s |
| 22:42.40 | muiro | Corydon76-dig: it's hosted at a data center with some redundancy. I couldn't expect better. |
| 22:42.57 | Corydon76-dig | muiro: what's your connection to that data center? |
| 22:43.03 | *** part/#asterisk salviadud (i=ralfalfa@voss.dreamhost.com) |
| 22:43.14 | `Sean | Corydon76-dig with enough bots you could ;) |
| 22:43.23 | `Sean | remember they dont need to be crappy home boxes either |
| 22:43.33 | `Sean | there is always bunch of dedi's out there aswell |
| 22:43.39 | `Sean | on gbit links |
| 22:43.42 | Corydon76-dig | `Sean: you might as well expect to take out a water main by pushing water down it |
| 22:43.42 | `Sean | then there is DrDOS |
| 22:43.51 | hmmhesays | yeah it is nice you can get a more full sound playing them in the d shape |
| 22:43.52 | `Sean | no |
| 22:44.22 | Corydon76-dig | `Sean: how does a botnet take out a T1 that isn't connected to the internet? |
| 22:44.54 | muiro | Corydon76-dig: right now it's just my cable connection to the datacenter. The box running asterisk is going to be hosted there. There needs be no constant connection between this office and the box |
| 22:45.18 | Corydon76-dig | muiro: so your box is hosted at the same location as the SIP provider? |
| 22:45.53 | Yourname`` | Funny thing is, same phone is able to do perfect DTMF with same settings for one Asterisk server. But not able to do good DTMF with another Asterisk server. SAME settings too! |
| 22:45.56 | muiro | no, the box is hosted at a datacenter, the sip provider is hosted wherever they host it |
| 22:46.04 | muiro | presumabely another data center |
| 22:46.22 | Corydon76-dig | muiro: right, so you're dependent on every leg in between |
| 22:46.45 | muiro | Corydon76-dig: not any more so than our web applications |
| 22:46.46 | _Sam-- | Yourname``: i dont know who your SIP provider is, or if this has any bearing, but i do know that teliax has dtmf issues. |
| 22:47.04 | _Sam-- | they are aware of them, though, and a have fix, if you know you hae a problem. |
| 22:47.08 | Yourname`` | _Sam--: Both the asterisk servers are on Teliax :( |
| 22:47.14 | Corydon76-dig | muiro: yes, but voice is not tolerant of delays. Web apps are. |
| 22:47.17 | _Sam-- | talk to david |
| 22:47.31 | _Sam-- | your dtmf will be fixed in 5 mins. |
| 22:47.50 | muiro | Corydon76-dig: I think I'm going to go with the SIP service, the price difference is immense |
| 22:47.51 | Yourname`` | _Sam--: Thing is, the server 1 where DTMF works everytime is on teliax, and server 2 where DTMF doesn't work also is on teliax :S |
| 22:47.57 | _Sam-- | the teliax dtmf issue is also related to asterisk version. |
| 22:48.02 | _Sam-- | 1.2.* have problems with teliax |
| 22:48.03 | _Sam-- | 1.4 dont |
| 22:48.25 | Yourname`` | _Sam--: Now we're getting somewhere! You're right.. the one that doesn't work is 1.2 |
| 22:48.34 | Yourname`` | _Sam--: Do they have a fix for 1.2? |
| 22:48.35 | _Sam-- | just call them, and explain. |
| 22:48.41 | _Sam-- | they know the problem and will fix it on their side, for you. |
| 22:48.47 | _Sam-- | they have a fix. |
| 22:48.49 | _Sam-- | for 1.2 |
| 22:48.52 | Corydon76-dig | muiro: as long as you see that there is a possible disadvantage, then I've answered your question |
| 22:48.59 | Yourname`` | _Sam--: Gotcha, thank you, _Sam-- :) Appreciate it. Let me call them up. |
| 22:49.13 | _Sam-- | good luck. i would have fixed it for you, but david isnt on IM |
| 22:49.27 | Yourname`` | _Sam-- : ok |
| 22:49.34 | muiro | Corydon76-dig: thanks |
| 22:49.42 | Yourname`` | You work for them, _Sam--? |
| 22:49.49 | _Sam-- | no. but ive been a long time customer. |
| 22:50.01 | Yourname`` | ah |
| 22:50.06 | Hadi-- | Hi guys |
| 22:50.12 | Hadi-- | I have an issue sending calls from Cisco |
| 22:50.15 | Hadi-- | to Asterisk |
| 22:50.20 | Hadi-- | calls drop right after 31 seconds |
| 22:50.39 | Hadi-- | any suggestions? |
| 22:51.57 | _Sam-- | Yourname`` : the reason i know all this, is because i went through the same exact problems in august, and yesterday. ymmv, but i did not have any luck with asterisk 1.4, and went back to 1.2 yesterday. doing so broke my dtmf. the reason i went to 1.4 to begin with, was because my dtmf was broke in 1.2 with teliax. |
| 22:52.28 | _Sam-- | id like to hear if your 1.4 works well for you, and with teliax. |
| 22:53.33 | bkruse | Corydon76-dig: my botnet can :D |
| 22:55.02 | twisted | OMFGuHAX0r |
| 22:55.24 | _Sam-- | what legimate purposes is any botnet used for? |
| 22:55.31 | Corydon76-dig | Testing |
| 22:55.40 | _Sam-- | testing what? (just curious) |
| 22:55.45 | Yourname`` | _Sam--: Seems to work bang on with 1.4. |
| 22:55.53 | Yourname`` | _Sam--: Did they fix you up for 1.2 |
| 22:55.54 | Yourname`` | ? |
| 22:55.58 | Corydon76-dig | Well, you can use it to simulate call volume, for example |
| 22:56.01 | _Sam-- | dtmf worked fine with 1.4, but asterisk 1.4 didnt work fine in general. |
| 22:56.15 | _Sam-- | why would you use a botnet to simulate call volume? there are dedicated apps to that |
| 22:56.18 | Yourname`` | _Sam--: aahhh, ok |
| 22:56.19 | _Sam-- | for doing that, rather. |
| 22:56.38 | [TK]D-Fender | _Sam--, "because it was there" |
| 22:56.51 | *** join/#asterisk booray (n=ray@150.118.ultimate-int.uia.net) |
| 22:57.25 | Corydon76-dig | _Sam--: because sometimes you need to test it in a more real world situation |
| 22:57.41 | _Sam-- | i wasnt trying to be an ass, i have nothing against botnets, or people who use them. i was just curious what non-evil things people may do with them. |
| 22:57.57 | Corydon76-dig | and then having many different connections connecting to the central server is helpful |
| 22:58.14 | _Sam-- | i dont believe that you, or anyone, uses a botnet in that fashion, but thats just my opinion. |
| 22:58.47 | booray | hi all. I'm looking for a 4-port fxo to sip gateway.. I'm looking at the linksys spa400 and the grandstream gxw4104. I need to be able to specify which line I want when dialing out through asterisk, rather than first available. does anyone know if either of those or a different unit is capable of that? |
| 22:58.52 | Corydon76-dig | Don't blame the technology, blame the goons using the technology to do bad things |
| 23:01.13 | _Sam-- | kind of like those that spoof caller id? |
| 23:01.14 | [TK]D-Fender | booray, Not the SPA. The GS I would avoid period regardless. This leaves Mediatrix / AudioCodes. |
| 23:02.51 | _Sam-- | Yourname`` : you are getting the same stability and reliability from your 1.4 asterisk, as from your 1.2 box? |
| 23:02.51 | [TK]D-Fender | Corydon76-dig, http://www.founditemclothing.com/t-shirts/guns.html |
| 23:03.18 | Yourname`` | _Sam--: Yessir. |
| 23:03.38 | _Sam-- | thanks for the info |
| 23:03.49 | Yourname`` | _Sam--: They want me to upgrade to 1.4 :( |
| 23:03.51 | Corydon76-dig | [TK]D-Fender: same idea |
| 23:04.08 | _Sam-- | Yourname``: you have AOL IM? |
| 23:04.10 | booray | [TK]D-Fender: I'm leary of GS myself. I'll look at mediatrix and audiocodecs. I'm also looking at aastra's ventureip 4fxo, just pulled up the page and was reading |
| 23:04.20 | Yourname`` | _Sam--: nope, why? |
| 23:04.32 | _Sam-- | because if you did, i would have given you an IM contact to add, who would have fixed it for you. |
| 23:04.36 | [TK]D-Fender | Corydon76-dig, I'm just waiting for someone to stab another in the neck 17 times with a ball-point pen so we can see the US ban Bic & PaperMate :p |
| 23:04.43 | booray | [TK]D-Fender: looks like it may also do more than i want, however |
| 23:04.47 | *** part/#asterisk Schumie (n=Steve@cpc1-oxfd2-0-0-cust25.oxfd.cable.ntl.com) |
| 23:05.18 | _Sam-- | (doesnt matter, he's 'away' on IM at the moment) |
| 23:05.19 | [TK]D-Fender | booray, The P2P Aastra stuff is only usable as P2P, not straight SIP IIRC. I'm leary of anything non-standard |
| 23:05.27 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 23:06.28 | Corydon76-dig | [TK]D-Fender: banning weapons was never on the agenda, just keeping guns out of the hands of bad people |
| 23:06.43 | booray | ok |
| 23:06.57 | Yourname`` | _Sam--: Who, josh? |
| 23:07.14 | _Sam-- | no. the guy i know, is the owner |
| 23:07.27 | _Sam-- | there is a josh here, file? |
| 23:07.34 | _Sam-- | i think he's a josh |
| 23:07.41 | [TK]D-Fender | Corydon76-dig, yup, because we all know that criminals won't want to obtain a gun illegally or nothin' ;) |
| 23:07.50 | Yourname`` | _Sam--: : oh |
| 23:08.00 | Corydon76-dig | [TK]D-Fender: key word being illegally |
| 23:08.04 | *** join/#asterisk rpm (n=russell@d154-5-130-82.bchsia.telus.net) |
| 23:08.07 | *** join/#asterisk mog (i=mog@nat/digium/x-334fb47f195c15e8) |
| 23:08.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:09.04 | Corydon76-dig | [TK]D-Fender: if a criminal has to break the law twice to hold up a store with a gun, that's half the chance he'll get off on a technicality |
| 23:09.06 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 23:09.06 | *** mode/#asterisk [+o twisted] by ChanServ |
| 23:09.07 | twisted | fun! |
| 23:10.07 | _Sam-- | Yourname`` : i sent him a message. i'll let you know if i get anywhere for you. |
| 23:10.12 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 23:10.15 | _Sam-- | like i mentioned, his status is 'away' |
| 23:10.21 | [TK]D-Fender | Corydon76-dig, and odds are that anyone willing to hold up a store with a gun won't have too many qualms about where it comes from either. Not so for the reverse. |
| 23:10.33 | _Sam-- | it took me about 3 minutes to have my DTMF fixed yesterday for my 1.2 |
| 23:10.38 | Yourname`` | _Sam--: Thank you.. :) |
| 23:11.01 | Yourname`` | _Sam--: Oh, nice. Was that David you mentioned earlier? |
| 23:11.02 | mosty | is it possible to reload the zapata.conf file without dropping all zap calls? |
| 23:11.17 | *** join/#asterisk digimania (n=none@24-119-242-84.cpe.cableone.net) |
| 23:11.43 | _Sam-- | i cant disclose my sources :) |
| 23:12.21 | _Sam-- | i feel lucky enought that i have the ability to instant message when there are problems, dont want that rescinded cause i gave up too much info :) |
| 23:12.24 | [TK]D-Fender | _Sam--, Except his name and that he's on IM ;) |
| 23:12.43 | Yourname`` | lol ok . . I just verifying, because you mentioned "talk to david, it'll be fixed in minutes" :D |
| 23:12.55 | digimania | Is it possible to send a test voicemail to an extension from the linux cli? |
| 23:14.12 | _Sam-- | the only further info is that the david i refer to, is the same david who replied to this message : http://www.voip-info.org/tiki-index.php?page=TelIAX :) |
| 23:18.37 | _Sam-- | Yourname`` : if your 1.4 is fine and stable, then i dont know what your downside to upgrading your 1.2 to 1.4 would be. |
| 23:18.48 | _Sam-- | and just fixing the teliax problem that way. |
| 23:20.14 | _Sam-- | i wish i was able to do that , but my 1.4's crash multiple times per week on my stuff. |
| 23:22.12 | Yourname`` | _Sam--: That'd be a band-aid to the real problem at hand. ;) |
| 23:22.31 | _Sam-- | not really. if 1.4 works for you, why WOULDNT you upgrade the 1.2 asterisk? |
| 23:22.54 | Yourname`` | Because this 1.2 depends on something else. |
| 23:23.20 | _Sam-- | i dont know what that means, or why 1.4 wouldnt fit right in its place. |
| 23:23.25 | _Sam-- | but im no expert either. |
| 23:27.00 | _Sam-- | just my own opinion, but calling the upgrade from 1.2 to 1.4 a bandaid is far off base. 1.2 is no longer supported, and no further development. upgrading to 1.4 would be a good choice, if it worked well for you, in my own opinion. |
| 23:27.50 | twisted | jbot: wtf? |
| 23:27.51 | jbot | wtf is probably what that's fine? |
| 23:28.04 | twisted | hah. |
| 23:39.00 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585913.dsl.bell.ca) |
| 23:41.23 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 23:43.14 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 23:44.30 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 23:45.52 | nohup_ | hello peoples... |
| 23:46.30 | nohup_ | the "callerchan" in the /var/spool/asterisk/voicemail/*/*/msg*.txt files... |
| 23:46.57 | nohup_ | i have one here with "SIP/nohup-081771e8" in it... that suffix after the username, is that a uniq ID ? |
| 23:49.28 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 23:51.10 | *** join/#asterisk dec (n=tom@unaffiliated/dec) |
| 23:56.43 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |