| 00:00.12 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:05.23 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 00:09.02 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 00:09.31 | Cuban0 | well |
| 00:09.38 | Cuban0 | 1.6 is not too much different |
| 00:09.50 | Cuban0 | how would you do it ? ;) |
| 00:13.00 | phix | I would first start by banging my head against my desk |
| 00:13.10 | phix | then drinking alot of scotch |
| 00:13.19 | Cuban0 | then ? |
| 00:13.34 | phix | after a day or two of procrasinating I would finally look at the docs and figure it out |
| 00:13.43 | *** join/#asterisk s14ck (n=s14ck@190-76-79-90.dyn.movilnet.com.ve) |
| 00:13.57 | _ShrikE | would drink the scotch before the head banging begins |
| 00:14.04 | phix | total turn around time is about 5 days |
| 00:14.11 | phix | heh |
| 00:16.14 | KavanS | does anyone have experience monitoring sip registration/trunk status for voip providers? |
| 00:26.02 | *** join/#asterisk italorossi (n=italoros@189.124.198.166) |
| 00:30.56 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 00:36.26 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 00:39.51 | cosmicwombat | Who is your DAHDI ? |
| 00:40.12 | cosmicwombat | or does anyone feel like helping me w/DAHDI.... |
| 00:41.26 | JafoJ | warning - crypto library not installed ** SUPER-SEED OPERATION ACTIVE ** please set Max uploads so each peer gets 6-8 kB/s |
| 00:41.29 | JafoJ | Problem connecting to tracker - timeout exceeded (x5) |
| 00:41.38 | JafoJ | wtf |
| 00:42.51 | KavanS | JafoJ, wtf? |
| 00:43.04 | JafoJ | crap, srry, wrong chan |
| 00:53.49 | *** join/#asterisk voipmonk (n=voipmonk@67.204.21.209) |
| 01:00.28 | cosmicwombat | DAHDI.... who is my DAHDI..... |
| 01:00.45 | Mango | cosmicwombat! I am your DAHDI! |
| 01:01.08 | cosmicwombat | Well then feel free to help me with DAHDI my DAHDI |
| 01:01.28 | Mango | Sorry :-/ |
| 01:01.55 | cosmicwombat | You ain't my DAHDI :) |
| 01:02.30 | Mango | :( |
| 01:06.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:26.27 | *** join/#asterisk exothermc (n=miles@74.85.64.49) |
| 01:27.03 | exothermc | Is there anyway to ring one device then after x seconds start ringing a new device without sending a cancel to the first? |
| 01:27.21 | jql | there's a trick to it |
| 01:27.29 | exothermc | jql: which is? |
| 01:28.05 | jql | Dial(Local/123-0@delay&Local/456-10@delay) |
| 01:28.22 | exothermc | ahh sweet |
| 01:28.33 | jql | and in the [delay] context, _X.,1,Wait(${CUT(${DELAY},2)}) or something |
| 01:28.37 | jql | however that function works |
| 01:28.44 | exothermc | so the -X@delay where X is seconds to delay the invite? |
| 01:28.53 | jql | you have to write [delay] yourself |
| 01:28.54 | jql | but, yes |
| 01:29.04 | exothermc | oh interesting. |
| 01:29.15 | exothermc | so that basically forks it, nice |
| 01:29.18 | jql | yep |
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| 01:30.11 | jql | you just use the Local context to handle the waiting, but you launch the dial simultaneously to the Local channels |
| 01:30.56 | exothermc | yup. |
| 01:30.57 | Cuban0 | if i just use asterisk and G729 to route calls between 2 G729 enabled devices....so i dont do transcoding...i dont need license ? |
| 01:31.09 | exothermc | Cuban0: right. |
| 01:31.20 | Cuban0 | but how to enable the codec ? |
| 01:31.29 | Cuban0 | it is enabled by default ? |
| 01:32.30 | exothermc | Cuban0: you have to do media bypass |
| 01:32.36 | Cuban0 | it is module format_g729.so ?? |
| 01:32.56 | Cuban0 | media bypass |
| 01:33.03 | Cuban0 | how ? |
| 01:33.10 | exothermc | hmm sorry not media bypass |
| 01:33.10 | exothermc | no there is no module. |
| 01:33.22 | Cuban0 | then ? |
| 01:34.16 | Cuban0 | i have here a CISCO SIP Phone restricted to G729 but when dialing *43 for echo test, call can not be done |
| 01:34.35 | exothermc | actually not sure how to do it in asterisk. The trick would be getting g.729 in the SDP. Something like the redirect_media (or whatever it is called should work) but you need to get asterisk to pass the SDPs from each leg to each other. |
| 01:35.25 | Cuban0 | i will google |
| 01:35.39 | Cuban0 | i thought it had native support |
| 01:36.10 | exothermc | at -a-boy |
| 01:36.45 | exothermc | Cuban0: You thought free software had native support for a commercial codec? Not sure how that work out logically. |
| 01:38.19 | Cuban0 | i mean for just routing the compressed packages no matter the codec inside ;) |
| 01:39.57 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 01:41.25 | Cuban0 | got it ;) |
| 01:41.26 | Cuban0 | http://www.voip-info.org/wiki/view/Asterisk+G.729+pass-thru |
| 01:41.34 | exothermc | Cuban0: Ya that really has nothing to do with support. It is just routing packets. |
| 01:41.40 | exothermc | cool |
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| 01:51.40 | Cuban0 | even in that mode it needs the opensource implementation |
| 01:51.47 | Cuban0 | dont know why but it does :p |
| 01:56.06 | file | in a scenario where it is one device calling another if both sides negotiate G.729 then Asterisk will simply forward the audio |
| 01:56.34 | file | the issue comes up that each side is negotiated independently, which can cause one side to be g729 and the other another codec forcing a transcoding scenario |
| 02:03.24 | carrar | specify that in sip.conf |
| 02:23.11 | Cuban0 | how many simultaneos calls can suport a powerfull asterisk server with no transcoding ??? lets say a DELL server with two Quad Core Xeons 2.8 GHz ? |
| 02:23.18 | *** join/#asterisk zerohalo (n=zerohalo@66.167.13.201) |
| 02:23.33 | Cuban0 | limit is on bandwidth , memory or CPU power ? |
| 02:23.46 | Cuban0 | and using G729 |
| 02:23.55 | Cuban0 | çbut just routing packets.. |
| 02:36.31 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-21-209.acanac.net) |
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| 02:58.14 | [TK]D-Fender | Cuban0: What is your goal? |
| 03:00.02 | *** join/#asterisk gones (n=gones@121.34.38.69) |
| 03:02.02 | godfreyhk | Hi there - I'm a complete noob to asterisk. I've been using those "digital phone" services for a few years and skype out for several months. I am thinking to ditch the phone service altogether and roll my own asterisk box instead. 1. Is that feasible? 2. Would it translate to actual cost saving or is it more like a project for fun? 3. What kind of hardware investment do I have to make? |
| 03:02.50 | gones | I'm writing a AGI Perl script and use the following lines to initiate a |
| 03:02.52 | Mango | Is this for a home or office system? |
| 03:02.52 | gones | call : |
| 03:02.52 | gones | $res = $AGI->exec("DIAL $dialstr"); |
| 03:02.52 | gones | And then : |
| 03:02.52 | gones | $answeredtime = $AGI->get_variable("DIALSTATUS"); |
| 03:02.52 | gones | But I can't got DIALSTATUS , what is the problem ? Does anyone could give me a little help ? Thanks a lot ! |
| 03:03.57 | godfreyhk | I have a old linux box which is running some flavor of P4 processor and somewhere between 256 - 384 MB ram |
| 03:03.58 | godfreyhk | Mango a home system |
| 03:05.33 | Mango | godfreyhk: There are few features of Asterisk for a home user that a good VoIP provider would not provide. |
| 03:05.33 | Mango | A few things I like Asterisk for are call recording and failover, if my VoIP provider is down. |
| 03:05.45 | godfreyhk | Mango: well, the current "digital phone" services i'm subscribing to aren't even true voip services |
| 03:05.48 | Mango | I would only consider Asterisk if the features you want are not otherwise available, or are too expensive. |
| 03:05.59 | Mango | Ah, VoIP over cable or somesuch? |
| 03:06.04 | godfreyhk | yep |
| 03:06.09 | godfreyhk | voip behind the scene |
| 03:06.32 | Mango | Oddly, here in Vancouver, VoIP over cable is actually cheaper than a great deal of Broadband VoIP providers. |
| 03:06.45 | godfreyhk | but ultimately the end-point is just a voip "box" (modem..?) that plugs into an analog phone |
| 03:06.57 | godfreyhk | Mango: oh hey I'm in Vancouver too :) |
| 03:07.02 | Mango | No way!!! |
| 03:07.04 | godfreyhk | I'm using the shaw digital phone |
| 03:07.06 | Mango | looks out his window and waves |
| 03:07.12 | Mango | Do you see me!??! |
| 03:07.21 | godfreyhk | oh so THAT was you |
| 03:07.22 | godfreyhk | :P |
| 03:07.26 | Mango | LOL |
| 03:07.49 | Mango | I use Shaw Digital Phone at the office and VoIP.ms at home. VoIP.ms is the only Canadian VoIP provider I feel confident using. |
| 03:08.03 | Mango | It's pay-as-you-go, so for light use or long distance, it is a better deal than Shaw. |
| 03:08.27 | Mango | $0.0055 to major centres in Canada/US. |
| 03:08.33 | godfreyhk | ah |
| 03:08.46 | godfreyhk | well. I'm currently subscribing to 3 services |
| 03:08.52 | Mango | with Shaw? |
| 03:09.11 | godfreyhk | 1. SkypeOut unlimited NA --- $3/mo (keeping this anyways) |
| 03:09.29 | godfreyhk | 2. Shaw digital phone --- like $30/mo? not 100% positive |
| 03:10.05 | godfreyhk | 3. A vancouver - hong kong voip phone service -- pretty much same thing as shaw except the box sits behind my router |
| 03:10.13 | Mango | Ah. If you're paying $30/month, you can likely do better if you want to put the time into learning how to use Broadband VoIP. |
| 03:11.45 | godfreyhk | I thought I could just buy one of those cisco/linksys box and subscribe to some sort of "real" voip services which seems like what shaw is doing - but apparently it's not that simple |
| 03:12.33 | Mango | How so? :) |
| 03:12.49 | chuckf | godfreyhk: it should be that easy depending on the provider |
| 03:12.52 | Mango | Add a good QoS router and you've pretty much got it. |
| 03:13.10 | godfreyhk | those "voip" services seems to be pricier than i think |
| 03:13.18 | Mango | Where are you looking? |
| 03:13.20 | godfreyhk | or maybe i am looking at the wrong type of service |
| 03:13.30 | godfreyhk | i don't know... just googled vancouver voip :P |
| 03:14.27 | godfreyhk | I don't use my landline that much - i could pretty much just stick with skype but I guess my family members are not very comfortable with making calls from PCs so being able to use the "old school" phones would be a big plus |
| 03:14.43 | godfreyhk | (and there's no skypein for canada. hmmmm) |
| 03:14.50 | chuckf | godfreyhk: not sure, but see if you can get vitelity.net services up north of the border |
| 03:16.09 | chuckf | the plan I have is a DID for $1.49/mo and 1.1 cents per minute (US) |
| 03:16.17 | Mango | VoIP.ms sounds similar. |
| 03:17.18 | godfreyhk | ah |
| 03:17.28 | godfreyhk | so they are usually billed by minute right? |
| 03:17.49 | Mango | Some VoIP providers have unlimited plans. |
| 03:18.07 | Mango | I do not know of any reputable Canadian VoIP provider with unlimited outgoing though. |
| 03:18.41 | godfreyhk | umhm.. my impression is with a voip line you can make unlimited (simultaneous) calls.... I guess that's for cooperates |
| 03:19.19 | Mango | The unlimited outgoing plans usually have a limit; the pay-as-you-go plans often are unlimited or have high limits. |
| 03:19.32 | godfreyhk | I see |
| 03:19.51 | godfreyhk | so let's say i subscribed to either vitelity or voip.ms. O |
| 03:20.16 | godfreyhk | then I just need one of those cisco/linksys box OR an asterisk box with some sort of analog phone adapter? |
| 03:21.16 | Mango | Start with a PAP2T and then work your way up to Asterisk if and when you need it. |
| 03:21.21 | Mango | You can always upgrade. |
| 03:24.45 | godfreyhk | umhm |
| 03:25.12 | godfreyhk | does it work "with" asterisk? |
| 03:25.24 | Mango | Yes. |
| 03:25.25 | godfreyhk | i am not sure if that question makes sense or not |
| 03:25.30 | chuckf | and if you go with an asterisk box (and possibly without it) you can just use a softphone |
| 03:26.14 | godfreyhk | chuckf - that's what i'm doing with skype now, excpet my family members are not quite comfortable with that |
| 03:26.15 | chuckf | the only hardware I have is an older pc that is running asterisk. Everything else is softphones attached to it |
| 03:26.25 | chuckf | they don't need to be |
| 03:27.03 | godfreyhk | oh. i might be mistaken. my impression is that a softphone == a piece of software, is that correct? |
| 03:27.18 | chuckf | my family and friends pick up their phones and dial me as they would, I answer via a softphone or cel phone if I'm forwarding at the time |
| 03:27.47 | chuckf | godfreyhk: that is correct. a softphone is a piect of software |
| 03:28.25 | godfreyhk | chuckf: ah. I see. you mean mixing "traditional" phones and soft phones |
| 03:28.29 | godfreyhk | that would be pretty cool |
| 03:28.37 | chuckf | and those friends of mine that are comfortable with softphones have extentions going from my asterisk box and can dial each other via 4 digit numbers |
| 03:28.50 | chuckf | I don't mix anything |
| 03:29.20 | godfreyhk | chuckf: oh. oops. I was being unclear - I live with my other family members :) |
| 03:30.26 | godfreyhk | I have one more question - as I've mentioned before, I am also subscribing to a HK voip service (which comes with a locked PAP2T) |
| 03:30.39 | chuckf | well you can get IP phones if you wish or ATAs to convert your existing phones |
| 03:31.10 | godfreyhk | so if I want to replace that service too, I guess I just find some similar service providers in HK and get a local DID? do I need an extra PAP2T for that? |
| 03:31.23 | Mango | A PAP2T supports two providers. |
| 03:31.31 | godfreyhk | interesting. |
| 03:32.11 | godfreyhk | I suppose I'd set up some sort of rules in the PAP2T, like if it's +1 then use provider A, otherwise if it's +852 use provider B? |
| 03:32.36 | Mango | Looks like Hong Kong is two and a bit cents per minute for VoIP.ms |
| 03:32.43 | chuckf | I have a cousin that regularly tours Europe with an opera company. She gives me her itenerary and I get DIDs in those countries so that she can dial locally there to my asterisk box then dial family here from within asterisk |
| 03:32.44 | godfreyhk | any recommendation on the PAP2T btw? seems like everyone is using the linksys one |
| 03:32.46 | Mango | Not quite that simple...you need two phones, or a two-line phone. |
| 03:33.09 | Mango | Yes, the PAP2T is the most popular ATA by far. |
| 03:33.15 | Mango | I have one. |
| 03:33.59 | godfreyhk | very exciting stuff :) |
| 03:34.16 | Mango | :) |
| 03:34.21 | godfreyhk | so if I need two providers, I guess I am looking for the one with two analog "outlets" |
| 03:34.24 | *** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus) |
| 03:34.42 | Mango | Right. The PAP2T does fit that requirement. |
| 03:35.09 | Mango | (We call an outlet an FXS port ;)) |
| 03:35.51 | godfreyhk | I better start memorizing the glossaries so that i won't look like an idiot here :P |
| 03:35.59 | Mango | No problem at all :) |
| 03:36.14 | Mango | Always happy to have another Vancouverite :) |
| 03:36.19 | godfreyhk | where did you get that PAP2T mango? I am willing to bet $10 that bestbuy won't see them :P |
| 03:37.46 | Mango | Craigslist has them from time to time. Of course, you have to be sure they're not from Vonage as they'll probably be locked. Also, avoid very very good deals from Hong Kong as they will likely be counterfeit. |
| 03:38.01 | godfreyhk | :P |
| 03:38.03 | Mango | checking prices of new units for you |
| 03:38.55 | Mango | NCIX has them for $60 |
| 03:39.09 | Mango | Oo, FrontierPC has them for $50 |
| 03:39.18 | Mango | Not often NCIX gets beat on price. |
| 03:40.00 | Mango | godfreyhk: I have to go now, but you may wish to check out http://www.dslreports.com/forum/voip |
| 03:40.17 | Mango | Here's some recommended settings for the PAP2T: http://www.toao.net/25-linksys-pap2t-voip-adapter-review |
| 03:40.46 | godfreyhk | thanks a lot Mango! |
| 03:40.55 | Mango | Good luck :) |
| 03:40.57 | godfreyhk | you are very helpful :D |
| 03:41.11 | godfreyhk | and chuckf too :) |
| 03:41.11 | Mango | I try ;) |
| 03:42.00 | Mango | One last thing...if your router does not support Quality of Service to prioritize VoIP, you should consider that too. Anything that supports Tomato firmware seems to work well with some tweaking. |
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| 04:14.19 | godfreyhk | hmm any good router suggestions? |
| 04:15.12 | ReDNeQ | how large of an enviroment |
| 04:15.28 | godfreyhk | I got fed up with a few sub-$100 routers (drops, range etc) and was gonna pick up a time capsule - but i just realized it doesn't even has QoS built in o_O |
| 04:15.39 | godfreyhk | ReDNeQ a pretty standard home (hose) |
| 04:15.41 | godfreyhk | house |
| 04:15.57 | ReDNeQ | i have found good success with a few sub 100s and a few 300s |
| 04:16.40 | godfreyhk | well... I guess it really depends on your luck = / |
| 04:17.05 | godfreyhk | currently using a dlink one, it sucks so bad that I have to replace it with a linux box and use it as an AP/switch instead |
| 04:17.26 | ReDNeQ | what dlink model |
| 04:17.37 | godfreyhk | a sec |
| 04:17.53 | godfreyhk | WBR 1310 |
| 04:18.19 | godfreyhk | I am thinking about upgrading to a 802.11n router, and seems like i'd need QoS for voip too |
| 04:18.29 | ReDNeQ | wbr are shit |
| 04:18.38 | ReDNeQ | i like the old school ones/.. |
| 04:18.46 | ReDNeQ | like the DIR and the DHV |
| 04:18.53 | ReDNeQ | as for sub pars. |
| 04:19.01 | ReDNeQ | the dlinks then, netgear |
| 04:19.08 | ReDNeQ | i stay away from linksys |
| 04:19.13 | ReDNeQ | cant stand those (&#( |
| 04:19.29 | godfreyhk | i thought linksys are supposedly great |
| 04:19.41 | ReDNeQ | supposedly because cisco bought them? |
| 04:19.42 | ReDNeQ | hah! |
| 04:19.43 | godfreyhk | had one before, my experience wasn't that great either |
| 04:19.59 | ReDNeQ | cisco hasnt dropped on dime of their technology into those POS's |
| 04:20.13 | godfreyhk | :P |
| 04:20.15 | ReDNeQ | they bought them because it was a way to sell their own product after hating the linksys |
| 04:21.10 | godfreyhk | soo |
| 04:21.16 | godfreyhk | any recommendations? |
| 04:21.55 | godfreyhk | requirements are simple... 1. 802.11n 2. QoS 3. I can actually find them in the stores :) |
| 04:22.07 | ReDNeQ | what stores you got near yah |
| 04:22.22 | godfreyhk | bestbuy (canada) |
| 04:22.32 | godfreyhk | future shop... NCIX.. |
| 04:22.45 | godfreyhk | well, anything from the major brands shouldn't be a problem |
| 04:23.00 | ReDNeQ | you can also order from newegg.ca |
| 04:23.06 | ReDNeQ | they have good return and prices |
| 04:23.10 | godfreyhk | oh.. and hopefully somewhat robust |
| 04:23.21 | godfreyhk | i.e. can handle some moderate amount of bit torrent :P |
| 04:23.34 | ReDNeQ | you on rogers? |
| 04:23.47 | godfreyhk | nope.. shaw for now, might switch to telus later |
| 04:23.53 | ReDNeQ | ahh ok |
| 04:24.05 | ReDNeQ | well i would say try some of the cheap cheap ones.. like Airlink |
| 04:24.08 | ReDNeQ | i know i know |
| 04:24.18 | ReDNeQ | but hey at 39US a pop you can buy 3 of them |
| 04:24.34 | ReDNeQ | i use them in my home and have them at a few business just because they are cheap and easy |
| 04:24.47 | godfreyhk | :P |
| 04:24.50 | ReDNeQ | and work well. they have the same chips in them as dlink, netgear nad linksys |
| 04:25.02 | godfreyhk | checking them out now |
| 04:30.42 | ReDNeQ | they are suppa cheap |
| 04:31.21 | ReDNeQ | this is what im using at home right now |
| 04:31.22 | ReDNeQ | http://www.airlink101.com/products/ar690w.php |
| 04:31.34 | ReDNeQ | i have 2 different voip providers |
| 04:31.41 | ReDNeQ | 7 wireless machines |
| 04:31.47 | ReDNeQ | and 10 lan machines in the hous |
| 04:31.51 | ReDNeQ | 4 kids.. |
| 04:31.56 | ReDNeQ | 2ps3 |
| 04:32.01 | ReDNeQ | one downstairs/one up |
| 04:32.05 | ReDNeQ | also media server |
| 04:32.14 | ReDNeQ | and my home office with servers |
| 04:32.15 | godfreyhk | wow |
| 04:32.17 | ReDNeQ | 3 actually |
| 04:32.29 | godfreyhk | that sounds GREAT :) |
| 04:32.34 | ReDNeQ | im able to download @ 3M a sec |
| 04:32.38 | ReDNeQ | without issues |
| 04:32.41 | godfreyhk | just wondering... do you bittorrent :P |
| 04:32.50 | ReDNeQ | umm no i utorrent! |
| 04:32.54 | ReDNeQ | l?) |
| 04:33.09 | godfreyhk | lol |
| 04:33.29 | ReDNeQ | able to see "current" movies in about an hour |
| 04:33.30 | ReDNeQ | ;) |
| 04:34.01 | ReDNeQ | im running complete gigabit |
| 04:34.12 | ReDNeQ | in the home here except for the wireless of course |
| 04:34.21 | godfreyhk | i'm jealous :P |
| 04:45.19 | Corydon76-dig | Because of course, you've got enough network contention and fast enough hard drives for that to make a difference over 100BT |
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| 04:55.03 | drmessano | Gigabit always speeds up my torrent downloads on my 30meg connection |
| 04:55.08 | TJNII | Corydon76-dig: Shenanigans. I have three machines sitting next to me, each with hard-drives faster than 100BT. One has SATA, one PATA, and one SCSI. Granted, it only comes into play when copying files to/from the NFS server, but it does make a difference. |
| 04:55.58 | TJNII | The SCSI machine is not on gigabit because I don't have a gigabit switch, and I do see a difference between it and the box that is tied to the server over gigabit. |
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| 05:30.19 | *** part/#asterisk gones (n=gones@59.40.235.38) |
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| 05:48.34 | *** join/#asterisk |Cybex| (n=John@atwork-26.r-212.178.82.atwork.nl) |
| 05:55.08 | *** join/#asterisk Jenna (n=JJ@unaffiliated/jenna) |
| 05:56.09 | Jenna | hey all |
| 05:56.20 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:56.41 | Jenna | anyone savy with zaptel ? can assist in this erroneous msg /root/zaptel-wcopenpci-1.4.9vt1/kernel/zaptel-base.c:4911: error: implicit declaration of function âclass_device_createâ |
| 05:57.12 | phix | Jenna: I has a zaptel |
| 05:57.49 | Jenna | Im compiling this thing on lenny 2.6.26-2-686 kernel |
| 05:57.57 | phix | I installed the drivers using this --> sudo apt-get install zaptel-source && m-a a-i zaptel-source |
| 05:58.13 | Jenna | I did see this https://issues.asterisk.org/view.php?id=13088#90513 but it fails to point out any solution |
| 05:58.25 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 05:58.32 | box2 | zaptel as in dahdi? |
| 05:58.43 | phix | Jenna: have you tried using the command I pasted above? |
| 05:58.45 | phix | box2: yes |
| 05:58.57 | Jenna | phix, hmm. the card Im using is from voicetronix so. I downloaded their version http://www.voicetronix.com.au/downloads.htm |
| 05:59.03 | phix | ah ok |
| 05:59.03 | Jenna | let me give it a try |
| 05:59.28 | phix | sudo m-a a-i zaptel-source even :) |
| 05:59.34 | phix | don't forget the extra sudo |
| 05:59.42 | phix | unless you are doing that as root |
| 06:00.19 | Jenna | yeah Im doing it the root way |
| 06:01.11 | Jenna | after it finishs I did modprobe ztdummy but it again errors out. FATAL: Module ztdummy not found. |
| 06:03.26 | Jenna | phix, u there ? |
| 06:03.49 | TJNII | If you are using ztdummy why do you need the special source? |
| 06:04.14 | phix | I m here |
| 06:04.31 | TJNII | If you have a card, shouldn't you modprobe the driver built for that card instead? |
| 06:04.42 | phix | Jenna: you dont need ztdummy if you have a zaptel card |
| 06:05.03 | phix | zaptel / voicetronix, whatever |
| 06:05.08 | Jenna | hmm. |
| 06:05.10 | phix | just reboot and it will load it :P |
| 06:05.20 | *** join/#asterisk ketema (n=ketema@turtle.ketema.net) |
| 06:05.22 | Jenna | okay wait |
| 06:05.27 | phix | then refer to dmesg to see if it worked |
| 06:05.36 | phix | or sudo less /var/log/syslog |
| 06:06.33 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 06:07.19 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 06:07.25 | Jenna | btw any ideas how would I know what module has been generated. I mean not sure what to modprobe ..... |
| 06:07.43 | Jenna | hmm. okay waiting for it to boot |
| 06:11.41 | Jenna | hmm. doesnt look nice or is it. http://pastebin.ca/1556812 |
| 06:12.08 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 06:12.33 | Jenna | phix, are thing all rosy or what ? (in that asterisk -c output) |
| 06:18.51 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
| 06:19.29 | phix | ? |
| 06:19.53 | phix | sudo /etc/init.d/zaptel restart |
| 06:20.00 | phix | what does that say? |
| 06:21.14 | phix | dmesg |
| 06:22.18 | Jenna | Zaptel telephony kernel driver: FATAL: Module ztdummy not found. |
| 06:22.30 | Jenna | the restart of zaptel i.e |
| 06:23.07 | phix | hmmm |
| 06:23.25 | phix | did the m-a a-i zaptel-source work correctly |
| 06:23.26 | phix | ? |
| 06:23.34 | phix | or did it report buil errors? |
| 06:23.36 | phix | build even |
| 06:23.39 | *** join/#asterisk xrmx__ (n=rm@host197-226-dynamic.1-79-r.retail.telecomitalia.it) |
| 06:23.54 | Jenna | phix, http://pastebin.ca/1556825 |
| 06:24.00 | Jenna | the dmesg output |
| 06:24.31 | Jenna | m-a a-i zaptel-source finnished without errors |
| 06:25.19 | phix | nice |
| 06:25.24 | phix | and it created a deb package? |
| 06:25.31 | phix | and I am assuming it installed that package too? |
| 06:26.37 | Jenna | not sure about that. how do I find that out |
| 06:26.39 | Jenna | <PROTECTED> |
| 06:27.20 | Jenna | I am getting zapconf zaptel_hardware binaries in my path. so I guess it has indeed installed it |
| 06:28.01 | Jenna | but you guys said that I dont need ztdummy since I have the hardware based card in the system. |
| 06:30.21 | phix | dpkg -l | grep -i zaptel |
| 06:30.52 | phix | have you edited your /etc/zaptel.conf file? |
| 06:31.06 | phix | BRB |
| 06:31.11 | Jenna | yup they are installed |
| 06:31.50 | Jenna | loadzone = us , defaultzone = us & nothing else in zaptel.conf |
| 06:32.52 | Jenna | phix, zaptel_hardware -v displays nothing |
| 06:33.10 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 06:38.10 | *** join/#asterisk oberon (n=oberon@CBL217-132-121-144.bb.netvision.net.il) |
| 06:38.13 | oberon | hi |
| 06:38.30 | oberon | in rtp.conf I set: |
| 06:38.35 | oberon | rtpstart=16348 |
| 06:38.35 | oberon | rtpend=32768 |
| 06:39.14 | oberon | yet, when I dial using SIP I see the following line in "sip set debug on": |
| 06:39.39 | oberon | Peer audio RTP is at port [ip]:64992 |
| 06:40.02 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 06:40.35 | oberon | why doesnt it limit itself to the ports range I gave ? |
| 06:47.12 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:47.42 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:53.37 | phix | Jenna: hmmm |
| 06:53.39 | phix | pwned |
| 06:53.56 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-vlldjbhglcnphjup) |
| 06:54.03 | Jenna | wow what u think ? |
| 06:54.28 | Jenna | vpbscan gives me this CARD1:V4PCI:irq=21 sub=56345654 bus-dev:01:00 |
| 06:54.28 | Jenna | BOARDS:1 |
| 06:55.49 | phix | you know the name of the module it uses? |
| 06:55.54 | phix | try and modprobe i |
| 06:55.55 | phix | it |
| 06:56.18 | phix | see if you can find a ubuntu package for it |
| 06:56.35 | Jenna | thats the problem. I dont know the name of it. otherwise I would have modprobed it |
| 06:56.36 | phix | if it is a card not covered by the ubuntu zaptel-source package |
| 06:56.45 | phix | google |
| 06:56.47 | KyleK | dang |
| 06:56.57 | phix | google name of card module name |
| 06:56.58 | KyleK | odbc voicemail doesn't do an UPDATE when a message is read |
| 06:57.07 | phix | KyleK: oh noes |
| 07:01.29 | Jenna | here is what the vendor specific drivers did. http://pastebin.ca/1556848 |
| 07:01.52 | *** join/#asterisk fnordus (n=dnall@70.70.0.215) |
| 07:02.35 | Jenna | hmm. modprobe vpb seems to have done the trick. Now asterisk recognizes the card |
| 07:03.30 | Jenna | phix, Am I good to go now http://pastebin.ca/1556850 |
| 07:03.44 | phix | nice |
| 07:04.07 | phix | you will have to put a line in /etc/zaptel.conf or run the auto config util for zaptel to generate the file |
| 07:04.42 | *** join/#asterisk Tim_Toady (n=moi@adsl94-108.kln.forthnet.gr) |
| 07:06.19 | Jenna | phix, I think this has done the trick http://pastebin.ca/1556852 |
| 07:10.21 | phix | yay |
| 07:10.24 | phix | gg |
| 07:11.08 | phix | vpb |
| 07:11.08 | phix | ? |
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| 07:14.57 | ultrav1olet | how can I set the language of messages spoken in extenstions.conf? Do I need to define my language in sip.conf or iax.conf? |
| 07:16.20 | KyleK | language for what? |
| 07:17.40 | godfreyhk | hmm.. any suggestions on routers for VoIP? |
| 07:17.54 | godfreyhk | someone here suggested an AIRLINK AR690W |
| 07:18.01 | godfreyhk | but can't really find it here in canada |
| 07:18.27 | KyleK | something with QoS or traffic shaping is good |
| 07:19.00 | godfreyhk | ya that's what i am looking for |
| 07:19.25 | godfreyhk | but I have too many bad experiences with routers so I'd like to see if there are any successful stories here.. |
| 07:20.11 | KyleK | ah |
| 07:20.11 | KyleK | that stuff is on my todo list :( |
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| 07:28.36 | godfreyhk | anyone knows where to get a +852 (hong kong) DID? |
| 07:30.31 | Jenna | phix, thanx for all the help |
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| 07:33.09 | maour_ | is it possible to have something like Read(digito,,1-480) ?! i don't want to put just 3 because 3 is from 0-999 ! |
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| 07:36.01 | *** join/#asterisk SparFux (n=raoul@e182031220.adsl.alicedsl.de) |
| 07:36.11 | SparFux | Hi all. |
| 07:36.41 | SparFux | Has anybody successfully used dahdi-linux-2.2.0.2 with HFC-S bri cards? |
| 07:37.23 | Tim_Toady | SparFux these cards work better with misdn afaik |
| 07:39.52 | SparFux | Hm... that's bad news. |
| 07:41.32 | SparFux | Because mISDN seems to be crappy as of now. sporadic DTMF tones and noise. |
| 07:43.29 | Tim_Toady | i cant say much on this, i had worked with a card like this only once |
| 07:43.44 | Tim_Toady | but that time neither bristuff or dadhi worked for me |
| 07:44.03 | Tim_Toady | on the other hand misdn looked to work fine |
| 07:45.26 | *** join/#asterisk dominic1 (n=Miranda@213.221.82.242) |
| 07:45.27 | SparFux | Aha, thx for the advice. |
| 07:45.32 | dominic1 | morning guys.... |
| 07:45.49 | dominic1 | i have a new error .... never saw it before |
| 07:46.09 | dominic1 | http://pastebin.org/15609 |
| 07:48.25 | dominic1 | okay think the recent changes are the problem in iax. I think i am using asterisk 1.4.19 on my main system .... so ... -> requirecalltoken=no? |
| 07:50.05 | drmessano | 1.4.19 is very old and buggy |
| 07:52.23 | *** join/#asterisk voxter (n=hardcore@174.6.173.231) |
| 07:52.41 | voxter | anyone know of a way to use a wav file natively as MoH but -not- start it from the beginning every time moh is invoked? |
| 07:53.47 | dominic1 | drmessano: I am using a appliance with 1.4.19 |
| 07:53.49 | dominic1 | :-( |
| 07:54.33 | dominic1 | I added requirecalltoken to my gateway with asterisk 1.6.1.6 but still get the error with calltokensupport..... |
| 07:54.35 | dominic1 | any ideas? |
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| 08:19.50 | dominic1 | Why can't i disable calltokensupport? I am using ast 1.6.1.6 on one system and 1.4.19 on another.... |
| 08:20.25 | dominic1 | when i set requirecalltoken=no |
| 08:20.25 | dominic1 | calltokenoptional =0.0.0.0/0.0.0.0 in the iax.conf of my 1.6.1.6 system i still get errors that the calltoken is required |
| 08:22.27 | Tim_Toady | dominic1 read doc/IAX2-security.txt |
| 08:28.12 | *** join/#asterisk lb (n=lb@pd95b8344.dip0.t-ipconnect.de) |
| 08:31.28 | dominic1 | Tim_Toady, i read the guide as pdf |
| 08:31.48 | dominic1 | but there is not any more information, i already did the things explained tehere |
| 08:34.10 | dominic1 | okay |
| 08:34.11 | Tim_Toady | it says 2 things: 1: its not backwards compatible, so ur 1.4.19 server will not work with it, and 2: u have to disable it by using calltokenoptional = 0.0.0.0/0.0.0.0 maxcallnumbers = 16382 on the 1.6.1 server side |
| 08:35.08 | dominic1 | ah i know what was wrong... did that in the definition of the trunk not in global.... |
| 08:35.14 | dominic1 | now it works |
| 08:35.23 | dominic1 | thanks Tim_Toady! |
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| 08:55.55 | dominic1 | is it normal, that i also have to specify username= in a friend context to get into the correct context on the target system? |
| 08:56.20 | dominic1 | what do you guys prefer? separate user, peer contexts for iax or one friend? |
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| 09:18.21 | OrNix | hello, can you show me sample command "Dial" with option "f" (force caller-id) ? |
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| 09:24.04 | *** join/#asterisk SparFux (n=raoul@e182021105.adsl.alicedsl.de) |
| 09:25.11 | SparFux | Hi all. I have a problem getting dahdi-zaphfc running with asterisk 1.6. Everything seems to be ok, but I get the error channel is busy.https://rl2.dnsalias.net/public/DAHDI/asterisk-error-outgoing_call.txt |
| 09:25.34 | SparFux | My installation is described in https://rl2.dnsalias.net/public/DAHDI/installation.txt |
| 09:26.28 | SparFux | sry, not channel Zap. i have it already changed to DAHDI. |
| 09:28.31 | tzafrir_laptop | SparFux, dahdi zaphfc driver from where, exactly? |
| 09:28.46 | tzafrir_laptop | you point to some forum post that seems to discuss misdn |
| 09:29.37 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 09:30.17 | SparFux | the site is in german only, sorry. From http://www.ip-phone-forum.de/attachment.php?attachmentid=38420&d=1249613114 on this site together with original digium dahdi-2.2.0.2 |
| 09:31.20 | SparFux | Or put it this way (my question): I have a zaphfc version installed which does not seem to do hardhdlc and most of all does not work for me though I can install it ok. Is there a different patch against dahdi-linux-2.2.0.2 to use zaphfc? |
| 09:33.10 | tzafrir_laptop | I use the following patch: http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/zaphfc?view=markup |
| 09:33.55 | Tim_Toady | there is also bristuff http://bristuff.org/index.php/Main_Page |
| 09:36.09 | SparFux | Well, and that's a debian patch. Hm... my debian packages does not include zaphfc. :-( |
| 09:37.11 | SparFux | Is there a tutorial how to build debian package form this "trunk" ? The dahdi and the dahdi-linux? |
| 09:38.10 | Tim_Toady | generally u do apt-get source 'package-foo' apply the pacthes u want and build the package |
| 09:43.02 | dominic1 | hi sparfux |
| 09:43.26 | dominic1 | what i am running asterisk 1.6.1.4 with that patches |
| 09:43.53 | SparFux | dominic1: the patches tzafrir_laptop mentioned? Well, this patch does not even build zaphfc module for me. :-( |
| 09:44.39 | tzafrir_laptop | SparFux, what error do you get? On what environment? |
| 09:45.09 | dominic1 | i looked at you output |
| 09:45.11 | SparFux | Debian unstable, kernel 2.6.30-1-amd64, no compile error, there is no zaphfc module build. |
| 09:45.24 | dominic1 | in dahdi show status he found the card..... |
| 09:45.47 | SparFux | dominic: yes. but I get busy line. Anyway, the patch tzafrir mentioned seems to be better, it uses hardhdlc :-) |
| 09:46.23 | dominic1 | i used the patches from ip-phone-forum |
| 09:46.25 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 09:46.32 | SparFux | dominic: ah!ok. |
| 09:46.49 | SparFux | dominic: well, all I get with those is busy line error. The line definitely isn't busy at all. |
| 09:47.03 | SparFux | And I cannot see any other error there. |
| 09:47.18 | dominic1 | the only problems i have with that patches is that i get sometimes zaphfc[0]: b channel buffer overflow: 55, 55 in my dmesg |
| 09:48.46 | dominic1 | the module must be compiled, cause you get informations from the card |
| 09:48.52 | dominic1 | what does dmesg say? |
| 09:49.11 | dominic1 | which dahdi version did you use? |
| 09:49.18 | dominic1 | the latest stable? |
| 09:49.46 | SparFux | No, this gets messy now. The module ip-forum.de gets compiled and installed ok. But the patch tzafrir mentioned does not get the modules compiled, at least not with dahdi-linux-2.2.0.2. |
| 09:49.59 | SparFux | dominic: 2.2.0.2. |
| 09:50.15 | *** part/#asterisk viraptor (n=viraptor@87-194-164-154.bethere.co.uk) |
| 09:50.43 | SparFux | now I am trying this with dahdi-linux_2.2.0~dfsg~rc5 fromr apt-get source. |
| 09:50.57 | SparFux | I put the patch into the debian/patches dir and bild the package? |
| 09:51.19 | dominic1 | okay, i used the forum-patches and dahdi 2.2.0.2 and this system.conf |
| 09:51.51 | dominic1 | you also have to add the modules to the /etc/dahdi/modules (i think you did that!) |
| 09:51.59 | SparFux | yes, did that. |
| 09:52.10 | SparFux | I added zaphfc module. |
| 09:52.15 | dominic1 | If u want i can send you my patched files |
| 09:52.33 | SparFux | patched in how far? |
| 09:53.03 | zeeesh | how many concurrent calls asterisk supports if there is no any transcoding using ulaw or alaw ... by using Intel(R) Xeon(R) CPU with 8 gb of ram ????? |
| 09:53.23 | dominic1 | With the latest patches from the forum (there were different versions) |
| 09:53.48 | Chainsaw | zeeesh: No transcoding makes things very easy. We're talking thousands. |
| 09:54.06 | Chainsaw | zeeesh: Obviously more complicated things like conferencing bring that number back down again. |
| 09:55.25 | SparFux | dominic1: can you give a url with the patches? |
| 09:56.05 | dominic1 | i can check out my whole package from my svn and upload the file for u |
| 09:56.10 | *** join/#asterisk mumtazah (n=mumtazah@60.53.138.48) |
| 09:58.59 | SparFux | How to build this dahdi-linux trunk? |
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| 10:09.43 | iksik | hm, asterisk sends some informations about server somewhere? |
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| 10:34.00 | SparFux | I build the debian svn trunk with dahdi-linux 2.2.0.2 and the zaphfc module still doesn't get built. |
| 10:34.22 | *** join/#asterisk godfreyhk (n=godfreyh@S0106000ae628a2fe.vc.shawcable.net) |
| 10:37.44 | SparFux | Aha, now I build with svn-buildpackage. |
| 10:40.22 | SparFux | I know! The modules are built from dahdi-source of course. So I have to install it and build the modules with m-a. *dong* |
| 10:41.59 | SparFux | OMG the owner bug. Over and over again! |
| 10:58.18 | *** join/#asterisk DrDeth (n=DrDeth@mail.mexuar.net) |
| 10:59.32 | DrDeth | hi |
| 11:01.04 | DrDeth | is anyone around to help with a config issue? |
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| 11:04.38 | synthetic | dont ask a question to ask a question |
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| 11:05.22 | DrDeth | ok then |
| 11:06.05 | DrDeth | I have a Realtime configuration using an IAX softphone |
| 11:07.09 | DrDeth | in my iax table, I've set the callerid field to a valid 'full' caller id - eg. "Name" <number> |
| 11:08.09 | DrDeth | however, when dialing the caller id is being set to: ""Name" <number>" <asterisk> |
| 11:08.33 | DrDeth | any ideas on how to fix this ? |
| 11:11.39 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-dklsmntrmzezunjl) |
| 11:35.02 | *** part/#asterisk DrDeth (n=DrDeth@mail.mexuar.net) |
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| 11:57.13 | garymc | anyone know where my voicemail.conf file is kept? |
| 11:58.07 | garymc | found it |
| 11:58.11 | garymc | etc/asterisk |
| 12:00.19 | garymc | ok i forgot this one - How do i restart asterisk in Terminal? |
| 12:02.18 | *** join/#asterisk mace (n=mace@debian/developer/mace) [NETSPLIT VICTIM] |
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| 12:20.53 | garymc | http://pastebin.ca/1557085 - this is the error i recieve when trying to send to voicemail. It is all setup correctly as im aware. |
| 12:21.18 | garymc | Asterisk CLI |
| 12:21.20 | garymc | :( |
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| 12:50.05 | defswork | any recommendations for a SIP DECT phone ? |
| 12:53.55 | *** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
| 12:55.52 | *** join/#asterisk SparFux (n=raoul@e182030022.adsl.alicedsl.de) |
| 12:55.56 | SparFux | re |
| 12:57.06 | SparFux | tzafrir_laptop: https://rl2.dnsalias.net/public/DAHDI/installation-debian-svn-trunk.txt ;-) |
| 12:59.24 | tzafrir_laptop | hmm... what's this about the "owner"? |
| 12:59.49 | tzafrir_laptop | let me try building this vs. 2.6 git |
| 13:00.32 | SparFux | tzafrir: 2.6.30 has the owner field removed. |
| 13:00.55 | SparFux | tzafrir: the patch for dahdi is to remove this one line there. But be aware, if you remove it, the package won't build on 2.6.29 anymore, I guess. |
| 13:01.43 | SparFux | I think it won't compille against 2.6 git. |
| 13:01.54 | SparFux | the owner field was dropped in 2.6.30 as a design decision. |
| 13:01.57 | tzafrir_laptop | that patch has already been applied to dahdi before 2.2 was released |
| 13:02.43 | SparFux | tzafrir: yes, that's why you need to update the patch perhaps. There might be an #ifdef I'd suggest. it should check kernel version and if kerenl >= 2.6.30 should leave the owner line. |
| 13:03.01 | tzafrir_laptop | SparFux, btw: check the bits about modulestest in the README for a shortcut next time |
| 13:03.03 | SparFux | I compiled it without that line and --svn-ignore and the thing is working here. |
| 13:03.52 | SparFux | yeah, that with the modules was stupid of me. |
| 13:04.22 | SparFux | I have tried so many things with zaptel / dahdi and now got it running finallly, but I got more and more away from reading anything... too bad. |
| 13:08.10 | SparFux | yeah, thx for the hint for modulestest. |
| 13:10.48 | *** join/#asterisk Ickmund (n=magnus@ada-bcn-fw01.adamoeurope.com) |
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| 13:15.00 | tzafrir_laptop | SparFux, oh well, wctc4 fails to build with 2.6.31.-rc9 |
| 13:16.20 | SparFux | what error message? |
| 13:17.04 | tzafrir_laptop | https://issues.asterisk.org/view.php?id=15747 |
| 13:17.18 | SparFux | Damn, and I need 31. That's the whole point. I need dahdi / zaptel for 2.6.31 :-) The nouveau graphics driver will only run well in 2.6.31 and that's what I want :-) |
| 13:17.46 | SparFux | ah, that's some other error even. |
| 13:17.47 | *** join/#asterisk digilink (n=digilink@tn-76-5-159-175.sta.embarqhsd.net) |
| 13:18.51 | SparFux | known problem: http://forums.digium.com/viewtopic.php?t=70800&sid=ab2264e2e530c0bf28203be1c9788aed |
| 13:19.37 | SparFux | https://issues.asterisk.org/view.php?id=15747 |
| 13:20.01 | SparFux | This is already fixed on the head of the dahdi-linux 2.2 branch in revision 6937. |
| 13:20.03 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:20.06 | SparFux | So no need to worry. |
| 13:21.53 | SparFux | Have you tested this zaphfc patch already? I hope it's running stable here. I already got a freeze, but only with dchan=3 in system.conf. With hardhdlc=3 it's running well so far. |
| 13:22.27 | *** join/#asterisk skirmisha (n=ast@90-154-149-243.btc-net.bg) |
| 13:22.31 | skirmisha | hello guys |
| 13:22.39 | skirmisha | can someone tell me why i get this error |
| 13:22.43 | atis_work | could anybody help me troubleshooting meetme sound issue? |
| 13:22.47 | skirmisha | pbx.c no application dbget |
| 13:22.59 | skirmisha | i have 2 servers with same version of ast |
| 13:23.05 | atis_work | i have dahdi_dummy and dahdi modules loaded, and there are no related errors in logs |
| 13:23.08 | skirmisha | the one is giving me above error |
| 13:23.08 | SparFux | atis:work: got a timing source? |
| 13:23.16 | skirmisha | and the other works just perfect |
| 13:23.18 | SparFux | atis: ok. |
| 13:23.37 | skirmisha | any ideas |
| 13:25.02 | atis_work | skirmisha: isn't it now func_db? |
| 13:25.37 | atis_work | skirmisha: so, now you use Set(DB(family/name)=123) and ${DB(family/name)} |
| 13:25.52 | E-bola | Hmmm trixbox or asterisknow |
| 13:26.05 | skirmisha | yes i thought so, but on the other server with same version i do not get this error |
| 13:26.16 | skirmisha | and dilaplan is same |
| 13:26.29 | atis_work | skirmisha: which asterisk version is that? |
| 13:26.31 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:26.37 | skirmisha | 1.4.18.1 |
| 13:27.09 | skirmisha | where is this application located |
| 13:27.12 | skirmisha | in which module |
| 13:27.25 | skirmisha | i think some module could not be loaded |
| 13:27.55 | skirmisha | i am fighting with this 5 hours |
| 13:28.02 | skirmisha | and could not find any reason |
| 13:28.07 | skirmisha | could it be compile issue |
| 13:28.11 | skirmisha | missing libs? |
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| 13:29.36 | skirmisha | let me try something |
| 13:29.38 | atis_work | skirmisha: try "core show application dbget" on server where it's working |
| 13:29.52 | atis_work | skirmisha: perhaps that app magically works from older version |
| 13:30.18 | atis_work | skirmisha: i think it shouldn't be at all in 1.4 versions |
| 13:30.20 | skirmisha | yep |
| 13:30.21 | skirmisha | think so |
| 13:30.24 | skirmisha | let me see that |
| 13:30.57 | skirmisha | be back soon |
| 13:30.59 | atis_work | or do a "make install" and check the list of modules that wasn't replaced |
| 13:31.39 | atis_work | SparFux: do you have any experience with dahdi debugging? |
| 13:31.46 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:32.11 | SparFux | atis_work: nope. I just installed dahdi in a working manner for the first time today :-) |
| 13:32.38 | SparFux | But I have a history of trying DADHI over months or years. |
| 13:32.38 | atis_work | well, i had zaptel working previously |
| 13:33.09 | SparFux | atis: me too. I come from zaptel. But I had it in 2.6.29 and wanted to use more recent kernels, mainly to be able to bet bluetooth running and the nouveau graphics driver. |
| 13:33.52 | atis_work | SparFux: was there something special to get rid of zaptel, or just rmmod and remove from init.d |
| 13:34.14 | SparFux | atis: I use debian. I just did aptitude remove zaptel and that's it. |
| 13:34.25 | *** join/#asterisk dannyp (n=dpapasia@SHANGRILA.NET.CMU.EDU) |
| 13:34.43 | SparFux | basically just remove zaptel from /etc/modules, and the init.d/ script from the /etc/rc?.d directories. |
| 13:35.00 | SparFux | atis: check wether zaptel still gets loaded with lsmod | grep zap |
| 13:35.12 | atis_work | SparFux: sure, done that.. |
| 13:35.22 | SparFux | If they are not loaded, zaptel is gone. :-) |
| 13:35.41 | dannyp | I'm attempting to set up a streaming radio station using VoIP+asterisk for call-in support... the idea is that someone can enter an extension in asterisk, the call goes to the DJ, and at some point the DJ can put both sides of the conversation into the radio stream, if they so desire |
| 13:36.20 | dannyp | I'm thinking the easiest way of doing this is getting the audio into JACK and using JACK to determine which audio (playlist, phone call, both) is being routed to darkice to go off to the icecast server |
| 13:36.33 | tzafrir_laptop | SparFux, fixed in SVN |
| 13:36.43 | dannyp | but it is not clear to me that app_jack is what I actually want for this purpose (I want both sides of the conversation to go to JACK) -- how would people build this? |
| 13:37.07 | atis_work | dannyp: do you really want dj's voice through phone's mic on the air? |
| 13:37.33 | atis_work | dannyp: better let them speak, and then redirect caller to air and let dj to speak normal mic |
| 13:37.33 | Cuban0 | hello , for dialup and IAX what should be the best compression codec ?? ilbc, gsm or g729 ? |
| 13:37.47 | Cuban0 | i mean for dialu slow nternet a t 33.6 Kbps |
| 13:38.21 | dannyp | atis_work: which we can do with app_jack, sure. but I'd be interested in being able to do it either way, because potentially the DJ will not be the person receiving the call |
| 13:39.09 | atis_work | dannyp: then just throw everyone into meetme room :) |
| 13:40.01 | dannyp | and get the audio to JACK via... having a softphone auto-connect? |
| 13:40.25 | atis_work | dannyp: you can do "originate" call and connect one side to meetme, second to jack |
| 13:42.26 | dannyp | using app_jack and the JACK() endpoint? or am I way off here? I'm not familiar with originate, and am reading docs now |
| 13:43.06 | *** join/#asterisk cjk (n=cjk@vodsl-9494.vo.lu) |
| 13:43.49 | dannyp | or are you saying, originate to have a call from the meetme to a softphone set to auto-answer |
| 13:46.06 | cjk | hi, i have to asterisk servers interconnected over the internet via SIP. dtmf on both sides is set to rfc2833. dtmfs from server 2 to server 1, but not the other way round? server 1 is using realtime. any idea what could be the problem? |
| 13:47.43 | synthetic | 33.6k voip call lol |
| 13:49.18 | coppice | he think the ISO 7 layer protocol means voice -> rtp -> ip -> modem -> voice -> isdn -> E1 |
| 13:51.20 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:55.14 | *** join/#asterisk cb122 (n=cb122@adsl-074-184-238-084.sip.asm.bellsouth.net) |
| 13:55.36 | cb122 | hello all. |
| 13:58.12 | *** join/#asterisk wonderworld (n=ww@mue-88-130-116-008.dsl.tropolys.de) |
| 13:58.44 | wonderworld | hi, is there a variable that would give me the SIP-username who is dialing out? |
| 13:58.58 | wonderworld | (in the dialplan) |
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| 13:59.50 | tompaw | Hi. |
| 14:00.38 | tompaw | Is is possible to rely on Asterisk only in terms of handling NAT? I mean - to connect clients w/o STUN and stuff.. |
| 14:01.05 | tompaw | Cause I enabled NAT support in the confs, and RTP still seems to be trying to connect to lan ips. |
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| 14:08.03 | GGD | good morning |
| 14:12.36 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 14:12.50 | SparFux | When trying to receive a fax I get this error: https://rl2.dnsalias.net/public/DAHDI/asterisk-fax-error.txt |
| 14:13.31 | cb122 | so has everyone else pulled their hair out over configuring cisco 7960's for sip and asterisk? |
| 14:13.54 | synthetic | its not that bad cb122 |
| 14:14.25 | SparFux | Perhaps I need a dahdi timing source. |
| 14:14.48 | cb122 | synthetic: I have been working on getting this thing going all morning and still no love. |
| 14:15.04 | SparFux | No, my conference in asterisk is working, so there should be a timing source already. |
| 14:15.27 | cb122 | phone is stuck in the configuring Vlan reset loop |
| 14:15.56 | synthetic | refurb phone? |
| 14:16.01 | synthetic | put the correct firmware? |
| 14:17.29 | cb122 | I have set up the firmware in the tftpboot folder |
| 14:17.40 | atis_work | a quick question. If i have two dialplan extensions: sw-161-ANSWER and sw-161-. which should execute for Goto(sw-161-ANSWER)? |
| 14:17.44 | cb122 | trying to upgrade it to the newest version |
| 14:18.02 | atis_work | i meant _sw-161-. |
| 14:26.08 | GGD | question is there a gui that you can use to configure/ admin asterisk? i saw webmin on the site but when i checked the fourms there was a few bugs in it any sugestions? |
| 14:26.31 | SparFux | all of the sudden, Fax works now. |
| 14:26.50 | Corydon76-dig | atis_work: always the first |
| 14:27.11 | Corydon76-dig | atis_work: and the reason is, exact matches always get matched ahead of patterns |
| 14:28.06 | Corydon76-dig | atis_work: now if you marked the exact match as a pattern, it would depend upon which version you're running as to which will match |
| 14:28.44 | atis_work | Corydon76-dig: i already found issue i'm having - 14615 |
| 14:28.56 | atis_work | Corydon76-dig: i just wonder why it's not fixed in 1.6.1.5 |
| 14:32.56 | *** part/#asterisk SparFux (n=raoul@e182030022.adsl.alicedsl.de) |
| 14:33.15 | *** join/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-9-206.w90-56.abo.wanadoo.fr) |
| 14:33.28 | merlin8282 | Hi all ! Anyone has used TxFax here ? |
| 14:34.20 | merlin8282 | I wonder what it exactly does. When I call my test extension with my softphone, it does almost nothing. In fact, it seems being waiting for a fax machine to talk with it. |
| 14:36.26 | merlin8282 | Is TxFax intended to "forward" a fax from a fax-machine ? |
| 14:39.23 | cjk | hi, my calls comes from pri to asterisk1 and then goes via SIP to asterisk2. if i use a sip device on asterisk2 i receive the dtmfs correctly (checked in the logs), but if i use disa to answer the call i do not get dtmfs at all. any idea? |
| 14:41.39 | *** join/#asterisk italorossi (n=italoros@189.124.198.166) |
| 14:42.44 | wonderworld | hi, is there a variable in the dialplan that would give me the SIP-username who is dialing out? |
| 14:43.48 | garymc | I know this is off channel, but no reply for 3 hours now in other channels. My Fpbx isnt displaying recorded calls. I know the calls are recording to the correct location but my fpbx GUI aint showing them. Any help apprecitaed |
| 14:44.50 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 14:47.49 | obruT | anyone knows what the "WARNING[1278]: chan_dahdi.c:14992 process_dahdi: Ignoring any changes to '...' (on reload) at line X" warrning message means and how to get rid of it ? |
| 14:48.15 | kaldemar | wonderworld: maybe not the exact username parameter, but function SIPPEER might be suitful. |
| 14:49.16 | obruT | ok, I deleted users.conf and no warrnings anymore :) |
| 14:53.47 | *** join/#asterisk scalex000 (n=chatzill@77puntacana02.codetel.net.do) |
| 14:54.16 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 14:54.25 | scalex000 | hello everyone, I need to connecta a Nortel BCM 200 and asterisk which protocol you recommend for do this? |
| 14:57.37 | rene- | well, it depends on what does the nortel 200 supports, if it is old PRI, if it is newish SIP |
| 14:59.02 | *** join/#asterisk mumtazah (n=mumtazah@60.54.34.9) |
| 14:59.41 | scalex000 | h323, sip |
| 15:00.35 | rene- | sip is builtin in asterisk, why bother with h323 |
| 15:00.56 | Chainsaw | rene-: To interface with Microsoft(R) NetMeeting(TM) |
| 15:01.13 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:01.30 | rene- | i know it has its uses, i myself use it here to interface to some GSM celullar gateways |
| 15:01.31 | kaldemar | that's some value! |
| 15:01.54 | rene- | but if i had the option of running sip to those i would in a heartbeat |
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| 15:18.02 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@89.193.28.123) |
| 15:18.58 | box2 | sip is not as fun as h323 |
| 15:20.37 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:20.54 | rene- | ehehe yeah i guess if your idea of fun is being dragged along 2 miles of hot pavement littered with broken glass then yes it is not as fun |
| 15:21.12 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 15:22.26 | coppice | H.323 gets a lousy press through poor implementations |
| 15:23.42 | *** join/#asterisk adorah (n=Administ@87.69.176.87) |
| 15:29.51 | scalex000 | so, you recommend to use SIP |
| 15:29.53 | *** join/#asterisk PoWeRKiLL (n=powerkil@bzq-79-176-64-31.red.bezeqint.net) |
| 15:32.06 | rene- | in asterisk sip has better support than h323 |
| 15:32.16 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 15:35.50 | merlin8282 | I'm not able to receive multiple pages faxes with RxFax. Am I doing something wrong ? |
| 15:36.16 | merlin8282 | Only the first page is received in my .tif file. |
| 15:36.41 | coppice | how many pages are reported as being received? |
| 15:37.51 | merlin8282 | I didn't notice that (now I do, but no new fax came) |
| 15:38.03 | merlin8282 | But in the fax Header it says 1/2 |
| 15:38.27 | merlin8282 | I double checked my scripts, the second page is not even in the tif file. |
| 15:38.40 | coppice | see what tiffdisp says about the file |
| 15:39.36 | coppice | sorry, tiffinfo |
| 15:40.22 | merlin8282 | mmm, it seems that the tiff file has 2 pages. |
| 15:40.36 | coppice | what are you using to view the fax? |
| 15:40.48 | merlin8282 | viewfax, but I just installed it |
| 15:40.56 | merlin8282 | dont't know how to use it |
| 15:41.19 | merlin8282 | but, tiff2pdf should convert the 2 pages, no ? |
| 15:41.37 | merlin8282 | both tif and pdf files don't have a second page. |
| 15:42.05 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:42.58 | coppice | well, tiffinfo tells you the actual sizes of the images. If it says there are two pages of a sane size, there really should be |
| 15:43.31 | merlin8282 | both are Image Width: 1728 Image Length: 1121 |
| 15:43.46 | coppice | sounds good |
| 15:44.13 | merlin8282 | oh |
| 15:44.54 | merlin8282 | strange : the tif file sent by mail contains only 1 page. I just copied the tif file with scp to my workstation, and this one is OK. |
| 15:45.31 | merlin8282 | I think RxFax puts data into the tiff file progressively, and that my script sent the file before it was complete. |
| 15:45.52 | merlin8282 | is going to modify his scrips |
| 15:46.24 | coppice | rxfax writes page by page. it won't report completion of the fax until it has finished writing |
| 15:47.19 | merlin8282 | ok, ok, I see the problem. Thank you very much ! |
| 15:51.29 | cjk | hi, does the qualify option has an effect on the number of retries/retransmissions? |
| 15:54.33 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 15:55.21 | drfreeze | Hello |
| 15:56.01 | drfreeze | I have deployed asterisk to 5 dental offices and have always used polycom phones. |
| 15:56.31 | drfreeze | I have some clients trying to save some money, are there any phones out there that can compete with polycom 550, but at a fraction of the price? |
| 15:56.45 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 15:56.47 | Chainsaw | drfreeze: Can they, or can they not, be awful? |
| 15:56.50 | drfreeze | auto provision is a requirement for me |
| 15:57.22 | drfreeze | Chainsaw: they should compete with polycom in quality. Don't really want a budgetone |
| 15:57.48 | Chainsaw | drfreeze: "Yes, I would like a Mercedes, but it can only cost as much as a Fiat" |
| 15:57.49 | merlin8282 | drfreeze: I don't know polycom's, but I'm using many Snom360's that do fine. |
| 15:58.29 | voxter | anyone know of a way to use a wav file natively as MoH but -not- start it from the beginning every time moh is invoked? |
| 15:58.34 | drfreeze | merlin8282: do they auto provision? |
| 15:58.54 | merlin8282 | drfreeze: sure |
| 15:59.22 | drfreeze | backlit display? Do you only recommend a certain version? |
| 15:59.43 | [TK]D-Fender | drfreeze: IP550's are wasted overkill |
| 16:00.03 | [TK]D-Fender | drIf you want a backlight, IP 450 saves a bunch |
| 16:00.03 | merlin8282 | Not especially. But the display is not so good as it could be |
| 16:00.06 | [TK]D-Fender | drfreeze: If you want a backlight, IP 450 saves a bunch |
| 16:00.08 | merlin8282 | it's too dark |
| 16:00.19 | drfreeze | [TK]D-Fender: what do you recommned? The backlight is a much appreciated option |
| 16:00.42 | [TK]D-Fender | drfreeze: http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-IP450 |
| 16:01.01 | [TK]D-Fender | drfreeze: http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-550 |
| 16:01.05 | [TK]D-Fender | drfreeze: Good bit cheaper |
| 16:01.46 | drfreeze | I see the 450 at $180. Can get the 550 for $185 |
| 16:02.22 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 16:02.24 | *** part/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 16:03.12 | drfreeze | the 450 looks like the same body as the 350 |
| 16:05.29 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 16:05.50 | drfreeze | Let's say I take backlighting out of the requirements list: What about the 320 or 330 polycom? |
| 16:06.12 | [TK]D-Fender | drfreeze: Decent phones |
| 16:13.19 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 16:13.24 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 16:21.10 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 16:22.25 | merlin8282 | mmm, how can I detect a hangup in the dialplan ? |
| 16:22.43 | merlin8282 | ...so that I can launch my script when a faxcall hangs up ? |
| 16:28.50 | [TK]D-Fender | merlin8282: "h" <- Asterisk Standard Extensions. |
| 16:29.30 | merlin8282 | >< sure... i'm tired ! |
| 16:29.57 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-11.static.o1.com) |
| 16:31.07 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-11.static.o1.com) |
| 16:31.16 | GGD | question is there a gui that you can use to configure/ admin asterisk? i saw webmin on the site but when i checked the fourms there was a few bugs in it any sugestions? |
| 16:33.17 | tgunr | ggd: there are several |
| 16:33.19 | tgunr | http://www.google.com/search?client=safari&rls=en&q=asterisk+gui&ie=UTF-8&oe=UTF-8 |
| 16:34.47 | GGD | tgunr |
| 16:34.48 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 16:35.04 | GGD | thanks but is there a specific one? |
| 16:35.09 | GGD | thats better than the rest/ |
| 16:35.10 | drmessano | No |
| 16:35.11 | [TK]D-Fender | GGD: Several |
| 16:35.23 | drmessano | FreePBX is the most popular |
| 16:35.29 | GGD | i saw the webmin there |
| 16:35.30 | drmessano | But not supported in here |
| 16:35.37 | drmessano | heh |
| 16:35.40 | [TK]D-Fender | GGD: FORGET WEBMIN |
| 16:35.48 | GGD | but i saw that they were having issues with it |
| 16:35.50 | [TK]D-Fender | GGD: that one is an ancient flaming piece of shit |
| 16:36.01 | GGD | i was just checking |
| 16:36.05 | GGD | LOL |
| 16:36.14 | drmessano | Asterisk is far too complex for webmin |
| 16:36.19 | GGD | i agree |
| 16:36.19 | [TK]D-Fender | GGD: #freepbx or #asterisk-gui |
| 16:36.23 | GGD | ahh |
| 16:36.35 | [TK]D-Fender | GGD: those are the 2 front-runners in the "free" category |
| 16:36.45 | GGD | ok |
| 16:36.52 | GGD | i will check them |
| 16:37.42 | GGD | thank you |
| 16:38.02 | tgunr | I am having some difficulty with my sip.conf file. I have my localnet domains defined (10.0.0.0/22) but I also have a computer on the WAN side that I want to use. I added it as domain=66.249.166.10 but I still see "Not a local domain" for any registrations. Can someone give me some clues? |
| 16:40.23 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 16:40.45 | [TK]D-Fender | tgunr: Don't use "domain" |
| 16:40.55 | [TK]D-Fender | tgunr: just follow the guide : |
| 16:40.57 | [TK]D-Fender | ~sipnat |
| 16:40.58 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:41.40 | tgunr | looking, thanks |
| 16:42.09 | Maliuta | mo'ning Mr D-Fender |
| 16:47.17 | gr0mit | anyone been using XML configurations for the latest Snom firmware? |
| 16:47.35 | gr0mit | I have seen what I think is a bug in the provisioning of function keys |
| 16:49.27 | gr0mit | relevent extracts and example at http://pastebin.com/m6ccc4959 |
| 16:53.23 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 16:55.20 | merlin8282 | gr0mit: did it work before ? |
| 16:55.36 | merlin8282 | I just did an update of my Snom360's today, and it keeps working |
| 16:55.39 | gr0mit | I don't know. Probably |
| 16:55.58 | merlin8282 | One line from my XML conf: <fkey idx="0" context="active" perm="">blf <sip:212@asterisk.oska.bringe.net;user=phone></fkey> |
| 16:56.28 | gr0mit | <PROTECTED> |
| 16:56.34 | merlin8282 | use < and > instead of < and > |
| 16:56.45 | merlin8282 | no, it's from XML file on the provisioning server. |
| 16:56.58 | gr0mit | <PROTECTED> |
| 16:57.02 | merlin8282 | yes. |
| 16:57.11 | merlin8282 | I had the same problem as you. |
| 16:57.12 | gr0mit | Interesting |
| 16:57.30 | gr0mit | I think it is probably a bug |
| 16:57.38 | gr0mit | but if I can get round it this way so be it |
| 16:58.07 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
| 17:00.57 | *** join/#asterisk DetroitArmz (n=DetroitA@74.204.89.249) |
| 17:02.54 | *** join/#asterisk errotan (n=errotan@84.3.230.36) |
| 17:07.22 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 17:08.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:08.31 | DetroitArmz | Greetings All, I am brand new to Asterisk, books in hand, scratching head on turning up a new SIP trunk. Inbound works great, my outbound gives me Failed to authenticate on INVITE to '"2000" |
| 17:08.45 | DetroitArmz | server is multihomes public/private |
| 17:08.52 | DetroitArmz | phone on private side |
| 17:08.54 | DetroitArmz | no nat |
| 17:08.57 | gr0mit | hmm. does not look like it likes < or > |
| 17:09.31 | gr0mit | merlin8282, any other things i need to set? |
| 17:09.44 | merlin8282 | no, only this solved my problem. |
| 17:10.02 | gr0mit | with version.24? |
| 17:10.42 | DetroitArmz | http://detroitarms.pastebin.com/m5ff49fc1 is output of console when trying to place call |
| 17:10.43 | gr0mit | I think I will reset to factory defaults again |
| 17:10.44 | merlin8282 | with 7.3.14, 7.3.23 and 7.1.33 on snom360 |
| 17:11.10 | gr0mit | which is not tried 7.3.24? |
| 17:11.17 | gr0mit | I mean you have not tried that? |
| 17:11.31 | merlin8282 | no, not tried 7.3.24 |
| 17:12.17 | gr0mit | okay, resetting to factory defaults |
| 17:12.28 | gr0mit | they will try again, just in case anything else got corrupted |
| 17:13.37 | gr0mit | DetroitArmz, can you please paste bin the relevant part of your extensions.conf |
| 17:16.03 | DetroitArmz | http://detroitarms.pastebin.com/d506c5193 |
| 17:16.13 | gr0mit | looks |
| 17:16.27 | *** join/#asterisk cjk (n=cjk@vodsl-9494.vo.lu) |
| 17:18.34 | gr0mit | and is*bound to both network adapters? |
| 17:18.59 | gr0mit | If incoming calls worked, I am assuming so |
| 17:19.13 | DetroitArmz | i have bind address set to 0.0.0.0 in sip.conf |
| 17:19.19 | DetroitArmz | so I believe so yes |
| 17:19.29 | gr0mit | can you paste in your sip.config file, eliminating passwords of course |
| 17:19.32 | DetroitArmz | internal phones can call each other (I have to hard phones) |
| 17:19.35 | DetroitArmz | yep |
| 17:19.36 | DetroitArmz | just a sec |
| 17:19.50 | gr0mit | I suspect your authentication is not set correctly |
| 17:20.10 | gr0mit | ;-) |
| 17:20.40 | gr0mit | do I assume that you are somewhere in Detroit? |
| 17:21.05 | *** join/#asterisk nibu (n=abululul@unaffiliated/nibu) |
| 17:21.56 | DetroitArmz | http://detroitarms.pastebin.com/d43e359d |
| 17:21.56 | raden_work | what would be better dual head 22" wide screens or 1x 28" widescreen ? |
| 17:22.04 | DetroitArmz | Just outside of Detroit, (Chesterfield) |
| 17:22.23 | nibu | hi to all! some one can indicate me a good how-to on how to configure the asterisk to work on home? |
| 17:22.40 | gr0mit | I have been to canton quite a few times |
| 17:22.50 | nibu | just a simple configuration for home. just to start to learning the asterisk features? |
| 17:22.57 | DetroitArmz | got friends there in Canton |
| 17:23.05 | gr0mit | so do we! |
| 17:23.26 | gr0mit | used to live between Hagerty and Cherry Hill somewhere in that area |
| 17:23.29 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 17:23.33 | DetroitArmz | own a gun store here in Chesterfield (detroitarms) this is a system I'm playing with to see if I can use for a phone system at the store when done |
| 17:23.57 | gr0mit | I'm sure it will be fine |
| 17:24.03 | gr0mit | but it takes a bit of setting up |
| 17:24.26 | DetroitArmz | I'm amazed at the features as read through the OReily book |
| 17:25.10 | gr0mit | it is the host= setting correct? |
| 17:25.28 | DetroitArmz | Done a few callmanager installs a few years back, haven't touched the stuff since SIP was adopted so I'm newbie again |
| 17:25.47 | nibu | DetroitArmz, do u know if this book exist in pdf format on the net? |
| 17:25.56 | nibu | can u say the name of the book? |
| 17:25.58 | DetroitArmz | yes, it does |
| 17:26.17 | [TK]D-Fender | ~book |
| 17:26.17 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:26.19 | rene- | hehe |
| 17:26.29 | DetroitArmz | go to the asterisk.org site and click on documentation, it's there. the name of the book is Asterisk The Future of Telephony |
| 17:26.39 | [TK]D-Fender | Or you could just look UP |
| 17:26.57 | DetroitArmz | lol thanks |
| 17:27.04 | nibu | DetroitArmz, thank!! |
| 17:27.48 | DetroitArmz | host= line [bandwidth] is the trunk I'm trying to use |
| 17:27.55 | DetroitArmz | it works for inbound quite nicely |
| 17:28.14 | gr0mit | I think you want to type equals friend |
| 17:28.32 | gr0mit | this allows both incoming and outgoing calls |
| 17:28.37 | gr0mit | if I recall correctly |
| 17:28.44 | DetroitArmz | i signed up for the site via freebx site, they had examples for use with freepbx but I'm not following how to incorporate in Asterisk |
| 17:29.01 | DetroitArmz | k let me try |
| 17:29.59 | raden_work | is there a way to have a default and a fall over host like for example vitelity-inbound i have to swithc between inbound21 and inbound22 all the time because I cannot get a route or other technical issues |
| 17:30.32 | rene- | there are several ways you would achieve redundancy |
| 17:30.44 | raden_work | howso ? |
| 17:30.52 | DetroitArmz | nope, no change, that I can see, (did sip reload) inbound still works, outbound same error |
| 17:30.53 | rene- | one would be having two gateways and using DNS SRV |
| 17:30.53 | [TK]D-Fender | raden_work: Dial out one. Check status, dial out another if failed |
| 17:30.55 | raden_work | im talking with asterisk itself registering to the ITSP |
| 17:31.13 | rene- | hmmm |
| 17:31.19 | rene- | register to both? |
| 17:31.26 | DetroitArmz | it's almost like the other end trunk is seeing a request straight from the phone vs just passing it through |
| 17:31.35 | rene- | what happens if you register to both? |
| 17:31.51 | raden_work | rene-, then wouldnt it try to use both unless i used diffrent contexts |
| 17:32.25 | rene- | do you get the same DNIS on both servers? do they host the same or different numbers |
| 17:32.51 | DetroitArmz | so they gave me an example if I was using freepbx at this url http://www.freepbx.org/freepbx-trunks |
| 17:33.08 | [TK]D-Fender | raden_work: You want redundency for INCOMING calls? |
| 17:33.13 | DetroitArmz | I'm not sure how to take what's provided there and place into my config as they have it broken into 2 sections |
| 17:33.31 | rene- | well incoming should be only one section |
| 17:33.41 | [TK]D-Fender | DetroitArmz: FreePBX is NOT supported here |
| 17:34.01 | [TK]D-Fender | ~freepbx |
| 17:34.02 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:34.02 | DetroitArmz | Hi Fender, understood, I'm not running it |
| 17:34.09 | DetroitArmz | I'm running asterisk alone |
| 17:34.22 | scalex000 | Hi there, someone know about problem of dahdi, I need to disconnect or first receive call before I can dial call out |
| 17:34.32 | raden_work | [TK]D-Fender, yes i keep loosing vitelity and they cant figure it out except to use both inbound 21 and inbound 22 |
| 17:34.44 | DetroitArmz | the issue I have is the trunk config examples for the service provider only showed free-pbx |
| 17:36.06 | nibu | DetroitArmz, I saw in the site of the o'relly that this book have a new edition. U know if the one that is in pdf format is still be usefull with the new versions of asterisk? |
| 17:36.13 | rene- | raden_work have you tried increasing your register frequency? |
| 17:36.26 | raden_work | how > |
| 17:36.52 | DetroitArmz | I'm brand new nibu so couldn't comment, the book I have in front of me is the 2nd edition |
| 17:36.57 | DetroitArmz | covers version 1.4 |
| 17:37.04 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 17:37.05 | rene- | chekout the wiki for sip.conf, also qualify=yes shouldnt hurt |
| 17:37.07 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 17:37.22 | [TK]D-Fender | nibu: Some changes, but a good basic reference |
| 17:37.57 | nibu | DetroitArmz, [TK]D-Fender: I undertand, thanks! |
| 17:37.58 | DetroitArmz | which is what I installed. It's been helpful thusfar from a fresh install perspective, walked me through installation from source files and basic configs |
| 17:38.06 | rene- | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 17:38.12 | rene- | check for registertimeout |
| 17:42.03 | *** part/#asterisk nibu (n=abululul@unaffiliated/nibu) |
| 17:50.36 | DetroitArmz | so if you're setting up a SIP trunk with a provider is it normal to have more than one definitions in the sip.conf file? Like 1 for outbound and 1 for inbound? |
| 17:50.56 | DetroitArmz | currently I just have 1 definintion for the provider in the sip.conf file |
| 17:51.05 | DetroitArmz | and traffic is only working inbound |
| 17:51.17 | DetroitArmz | Just wondering if that is perhaps my problem? |
| 17:52.41 | tzafrir_laptop | OT: Silly patent application of the day: http://www.google.com/patents/about?id=ZUbHAAAAEBAJ&dq=WWW+Addressing |
| 17:54.56 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 17:56.59 | *** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-9-206.w90-56.abo.wanadoo.fr) |
| 17:59.54 | dustybin | what does this mean: Call rejected, CallToken Support required. |
| 18:01.50 | *** join/#asterisk SparFux (n=raoul@e182016088.adsl.alicedsl.de) |
| 18:02.34 | kaldemar | dustybin: IAX2 protocols was changed a little: http://downloads.asterisk.org/pub/security/AST-2009-006.html |
| 18:02.59 | kaldemar | dustybin: see this on how to avoid that: http://downloads.asterisk.org/pub/security/IAX2-security.html |
| 18:03.05 | dustybin | thanks :D |
| 18:03.22 | DetroitArmz | In turning on SIP Debug I see the response back from Invite from the provider is <--- SIP read from 216.82.225.24:5060 ---> |
| 18:03.22 | DetroitArmz | SIP/2.0 407 Proxy Authentication Required |
| 18:04.00 | dustybin | requirecalltoken = auto |
| 18:04.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:04.57 | kaldemar | DetroitArmz: that is the remote end telling you to authenticate the invite. |
| 18:06.29 | DetroitArmz | hmm so if I already have authentication setup for the trunk, is there somewhere else I need to turn this on? Seems like I have to tell my phone that if it's calling outside it needs to get proxy authentication from the trunk config |
| 18:14.38 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 18:14.49 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 18:15.07 | tgunr | I can now see my domains But still getting "not a local domain", I've been reading the docs and thought I had it set correctly but evidently not. Could someone give me a clue or two? http://paste.lisp.org/display/86718 |
| 18:16.38 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:18.02 | *** join/#asterisk SirColin (n=SirColin@83.216.68.241) |
| 18:23.18 | *** join/#asterisk scalex000 (n=chatzill@77puntacana02.codetel.net.do) |
| 18:26.18 | DetroitArmz | Got it guys, had fromuser and username confused. Also next issue was it doesn't appear to be accepteding 7 digit but 10 works |
| 18:26.33 | DetroitArmz | that's easy enough to fix |
| 18:26.59 | DetroitArmz | kaldemar thanks, that made me relook at authentication details |
| 18:27.22 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 18:27.31 | DetroitArmz | I assumed that since inbound was working that authenticaiton was working, that apprently was incorrect |
| 18:29.40 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444747.dsl.bell.ca) |
| 18:48.53 | scalex000 | hello |
| 18:49.27 | *** join/#asterisk cnu (n=cnu@63.80-203-44.nextgentel.com) |
| 18:50.23 | scalex000 | who knows about this error Unable to create channel type DAHDI |
| 18:50.31 | *** join/#asterisk Gugge (n=Guggeman@vlan2.dlxhosting.dk) |
| 18:51.24 | *** join/#asterisk augustf (n=augustf@pool-173-68-58-129.nycmny.fios.verizon.net) |
| 18:54.11 | *** join/#asterisk leon01 (n=leon01@ool-457c12b2.dyn.optonline.net) |
| 18:54.12 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 18:54.58 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 18:58.23 | *** join/#asterisk heit0050 (n=andy@mail2.heitkeconsulting.com) |
| 18:58.28 | manxpower | ~answers |
| 18:58.29 | infobot | well, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 18:58.45 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:19.52 | dustybin | does one need to install asterisk addons, for 'Digital Timeout' ? I am getting this error: |
| 19:19.55 | dustybin | No application 'DigitTimeout,5' for extension (mainmenu, s, 3) |
| 19:20.07 | dustybin | *Digit |
| 19:20.27 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 19:20.53 | *** join/#asterisk af_ (n=getsmart@88-149-240-57.dynamic.ngi.it) |
| 19:21.26 | dustybin | oh |
| 19:21.26 | dustybin | Deprecated in Asterisk v1.2 - use 'Set(TIMEOUT(digit)=timeout)' instead |
| 19:28.34 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
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| 19:41.48 | pugachevcobra | Hi...I'm runing asterisk on a vmware. My asterisk sounds from system sounds and IVR's were playing fine. Just now I had to hard reset the host machine where the vmware runs, and now all asterisk sounds (except MOH) are playing fastly-chopped (not chopped slow, but as in fast forward). How can I fix this? |
| 19:42.26 | rikman87 | Hi, can someone tell me how to disable the display of the caller name for an sip INVITE? I want to use my Asterisk 1.4 as an SIP-Client, but my Provider can only understand the sip-Adresses without the Names. |
| 19:42.26 | rikman87 | The register is in the correct format:: |
| 19:42.26 | rikman87 | .. |
| 19:42.26 | rikman87 | Thats the INVITE: |
| 19:42.26 | rikman87 | the stuff before the Adress shouldn't be there |
| 19:42.28 | rikman87 | From: "4614048821" <sip:4614048821@77.246.119.163>;tag=as13d628c2 |
| 19:42.30 | rikman87 | To: <sip:004917617700120@77.246.113.42>;tag=test_tag_0065166992 |
| 19:42.47 | rikman87 | the format of the register is: |
| 19:42.49 | rikman87 | From: <sip:4614048821@77.246.113.42>;tag=as1f7bdcfa |
| 19:42.49 | rikman87 | To: <sip:4614048821@77.246.113.42> |
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| 19:44.03 | chuckf | ~pastebin |
| 19:44.04 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
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| 19:49.47 | s0lid | anyone tried to interconnect asterisk with mera? |
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| 19:55.01 | s0lid | anyone tried to interconnect asterisk with mera? |
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| 20:10.15 | pavlovski12 | does anyone of you have an idea how to let asterisk dial * and # over zaptel? |
| 20:10.16 | s0lid | pavlovski12: what do you mean just press it |
| 20:10.16 | s0lid | anyone tried to interconnect asterisk with mera? |
| 20:10.16 | rikman87 | in bayonne you have to replace the * with a B |
| 20:10.16 | pavlovski12 | i get hangups at the cli prompt |
| 20:10.16 | rikman87 | and the # with a C |
| 20:10.17 | pavlovski12 | how do you mean rikman |
| 20:10.17 | rikman87 | you mean you want to route to a target with routing protection right? |
| 20:10.17 | pavlovski12 | no this asterisk |
| 20:10.17 | pavlovski12 | is behind another pbx |
| 20:10.17 | pavlovski12 | alcatel 4400 |
| 20:10.17 | rikman87 | in bayonne, which is an old telephony-machine you can to that by replacing the * with a 'B' |
| 20:10.18 | rikman87 | on asterisk i dont have any idea |
| 20:10.18 | pavlovski12 | i definately will try that |
| 20:10.18 | rikman87 | but maybe its the same to do |
| 20:10.18 | pavlovski12 | anyone else an idea |
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| 20:31.25 | KyleK | can I use realtime asterisk for just voicemail? |
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| 20:31.58 | scalex000 | Qwell: are you there? |
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| 20:37.09 | Coder365 | hey gang, I'm finding my asterisk install is only listening on port 5038 and not 5060... the sip.conf says port 5060 and i've restarted the service a few times.. turned off the firewall. etc....any ideas? |
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| 20:43.30 | p3nguin | Is there a way, possibly using a macro, to alter the incoming caller ID that will display on the phone? |
| 20:44.06 | Coder365 | p3nguin, as in... incoming call -> alter CID -> extension ? |
| 20:44.48 | Coder365 | or outgoing call -> alter CID -> whoever you called |
| 20:45.04 | p3nguin | incoming call to an extension |
| 20:45.33 | p3nguin | I'm basically wanting to be able to append characters to the incoming caller ID based on conditions of how the extension was reached. |
| 20:45.39 | Coder365 | Oh |
| 20:45.50 | Coder365 | I'm sure it can be done, but I'm not sure how |
| 20:46.23 | Coder365 | so the display says [NUMBER / T1] for trunk1 or whatever |
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| 20:48.42 | p3nguin | For example, if you dial my DID, you will be given a chance to enter an extension if you know it... otherwise you have to wait out the dialplan until it eventually dials a default extension. Let's say that if you never enter any extension when prompted, you might eventually get to ext 101. The CID on the phone would be that of the incoming call. I might want to make it ${CALLERID(num)}dd if they would have direct-dialed ... |
| 20:48.48 | p3nguin | ... the extension when prompted. |
| 20:48.50 | p3nguin | I hope that makes sense. |
| 20:49.22 | Coder365 | yep |
| 20:49.27 | p3nguin | I just want to alter it based on how the extension is reached. |
| 20:49.51 | Coder365 | yep |
| 20:49.55 | Coder365 | that's making sense |
| 20:50.00 | Coder365 | i just dont know how to do it |
| 20:50.18 | Coder365 | know why my asterisk install wont listen on port 5060? |
| 20:50.30 | Coder365 | it's not that the firewall is blocking it, it's not even /listening/ on the port |
| 20:51.01 | p3nguin | Maybe. Is sip.conf configured to listen on 5060? |
| 20:51.51 | p3nguin | You might have to move sip.conf.default to sip.conf, first. Then make sure there is a bindport and bindaddr configured. |
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| 20:52.20 | Coder365 | http://pastebin.ca/1557668 |
| 20:52.58 | p3nguin | And you've started or restarted asterisk after making any changes? |
| 20:53.45 | Coder365 | yep |
| 20:54.06 | p3nguin | lsof -i udp:5060 |
| 20:54.12 | chuckf | Coder365: on your * box, can you pastebin the output of 'netstat -an|grep 5060' and 'ps -ef|grep [a]sterisk' |
| 20:54.12 | Coder365 | http://pastebin.ca/1557674 |
| 20:54.29 | Coder365 | chuckf, ^^ |
| 20:54.44 | Coder365 | not the exact commands, but you get the idea |
| 20:55.15 | chuckf | can you do the exact commands? |
| 20:55.19 | *** join/#asterisk ultramancool (i=ultra@silenceisdefeat.com) |
| 20:55.27 | Coder365 | sure |
| 20:55.39 | ultramancool | hey, anyone mind giving me a little (offtopic) general h.323 help? |
| 20:55.42 | p3nguin | netstat -napt ... the t means tcp |
| 20:55.53 | Coder365 | oh |
| 20:55.57 | p3nguin | YOu need to check for udp. The lsof command would do that. |
| 20:56.00 | p3nguin | lsof -i udp:5060 |
| 20:56.39 | Coder365 | ...wtf? |
| 20:56.43 | Coder365 | now i feel like an ass |
| 20:56.45 | p3nguin | Or netstat -lnpu |
| 20:57.17 | p3nguin | It was listening on udp:5060 all along? |
| 20:57.24 | Coder365 | no, it wasnt |
| 20:57.26 | p3nguin | oh |
| 20:57.29 | Coder365 | but is now |
| 21:00.29 | chuckf | p3nguin: it was there all along, Coder365 was filtering on tcp |
| 21:01.07 | ultramancool | see, my college has a cisco h.323 network for all the dorms and I'm wondering what information I'd need to use asterisk or a softphone on it? |
| 21:01.09 | Coder365 | not sure. but it works. i think... |
| 21:04.04 | p3nguin | That's a good thing. |
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| 21:04.37 | drmessano | It is now that we're looking for something other than tcp |
| 21:04.37 | drmessano | heh |
| 21:04.37 | drmessano | ultramancool: Are you sure it's H.323? |
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| 21:04.52 | Coder365 | im a noob. sorry guys |
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| 21:05.26 | p3nguin | What do you need to do in order to find out for sure? |
| 21:05.27 | rossand | Need suggestions on debugging. I am setting up SIP trunking with unlimitel.ca. Calls connect, but there's no sound in either direction. I have verbosity set to 5 but the logs don't indicate what might be the problem. Looking for suggestions? Thanks. |
| 21:05.27 | p3nguin | It happens. You'll know next time. |
| 21:05.27 | p3nguin | I think I figured out how to change the CID on incoming calls, too. |
| 21:05.58 | drmessano | man netstat |
| 21:05.59 | ultramancool | drmessano: reasonbly sure, yes, I found my phone on the network, got this on it's web server: CallManager 1 10.80.0.21 Active |
| 21:06.03 | ultramancool | portscanned that IP |
| 21:06.21 | ultramancool | and it showed some h.323 ports ;) |
| 21:06.23 | Coder365 | so the service is confirmed listening on port 5060 udp... and the port is confirmed open with netstat... but my sip registrations time out... |
| 21:06.28 | ultramancool | and that's what I read when I googled the phone name too |
| 21:06.31 | ultramancool | so I think it's h.323 |
| 21:06.50 | Coder365 | port is confirmed open with nmap, rather |
| 21:07.15 | ultramancool | drmessano: 1720/tcp open H.323/Q.931 |
| 21:07.51 | *** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus) |
| 21:08.21 | drmessano | ultramancool: I would wager that the CCM may be listening on H.323, but you're likely SCCP |
| 21:08.42 | scalex000 | drmessano: Do you know how to fix a problem using Dahdi? |
| 21:09.38 | drmessano | scalex000: I guess it depends what sort of problem you expect to fix using DAHDI.. Baldness, impotence, blindness, stupidity.. no.. |
| 21:09.43 | ultramancool | drmessano: sorry, I'm relatively new to this voip stuff, I think CCM is Cisco Call Manager? but what's SCCP? |
| 21:10.00 | scalex000 | drmessano: Ha Ha |
| 21:10.22 | ultramancool | drmessano: my phone here is a cisco 7912 if that helps at all |
| 21:10.29 | drmessano | ultramancool: SCCP, being one letter off from CCCP, is Cisco's little VoIP protocol.. It's made up of dead babies and dead kittens |
| 21:10.29 | scalex000 | drmessano: Well, I don't how to resolve this problem: I can't make a call |
| 21:10.30 | p3nguin | ultramancool: That's a proto used by CCM. |
| 21:11.05 | scalex000 | drmessano: I can receive, and if I want to make a call I need to disconnect and connect the line |
| 21:11.27 | scalex000 | drmessano: Error, unable to create channel Type Dahdi |
| 21:11.29 | drmessano | scalex000: I've heard of that issue with others, not sure what the resolve is.. |
| 21:11.35 | p3nguin | Since I use Asterisk and have SIP trunks, I just use SIP on my Cisco phones. |
| 21:12.17 | drmessano | SIP doesn't address a number of the native features of the Cisco phones, but at least you're not a baby killing, gun toting, pinko commie liberal SCCP user |
| 21:12.28 | p3nguin | heh |
| 21:12.34 | ultramancool | drmessano: lol, alright then... reading wiki, seems asterisk has support for it but I don't see any sort of mention of an open source softphone for it |
| 21:13.12 | drmessano | ultramancool: You won't be interfacing with this easily.. If you got a softphone, it would be the Cisco IP Communicator, and it would require a CCM license to use it |
| 21:13.43 | p3nguin | I'm not sure what additional features I really need. I have directories, services, and messages all configured via SIP*.cnf files. I don't know of anything else that I could need. |
| 21:13.59 | drmessano | ultramancool: Perhaps offer the VoIP admin some donuts, hookers, pizza, or perform the sexual favors yourself and he may allow SIP access for you |
| 21:14.45 | drmessano | p3nguin: I think some of the softkey capabilities, presence, and seems like one other minor detail wasn't present |
| 21:14.50 | ultramancool | drmessano: damn :/ |
| 21:15.22 | p3nguin | I haven't noticed anything missing. Not sure exactly what presence is, though. |
| 21:15.25 | Deeewayne | scalex000, I <3 dahdi tag 2.1.0.4+2.1.0.2 |
| 21:16.16 | scalex000 | deewayne: I have the lastest version |
| 21:16.27 | drmessano | p3nguin: Honestly, I have played with SCCP, but not SIP on Asterisk.. The SIP firmware is supposedly less stable and I had a short list at one time of the feature differences.. Not sure beyind a guess now |
| 21:17.25 | drmessano | p3nguin: I felt "dirty" using a Cisco phone on Asterisk |
| 21:17.28 | ultramancool | drmessano: well, thanks for explaining atleast |
| 21:18.00 | p3nguin | I actually feel much better knowing that I am not forking over thousands of dollars for Call Manager licensing. |
| 21:18.04 | *** join/#asterisk geneticx (n=geneticx@adsl-2-59-65.mia.bellsouth.net) |
| 21:18.16 | *** join/#asterisk leon01 (n=leon01@ool-457c12b2.dyn.optonline.net) |
| 21:19.29 | Deeewayne | scalex000, I had problems with some of the earlier 2.2.0 tags and settled on 2.1.0.4+2.1.0.2. You could try rolling back to the tag I heart and see if you still have the problem. |
| 21:19.35 | drmessano | p3nguin: hopefully the day will come soon when an Asterisk logo on a phone will command as much respect as Cisco.. We have/can do so much better, but here in the states we are generally a bunch of corporate name brand-lovin communists |
| 21:19.47 | Deeewayne | I'm just suggesting that to you... maybe there is something in the more recent tags that you need |
| 21:20.29 | scalex000 | Deewayne: Ok |
| 21:21.26 | Deeewayne | ... and now it is time for a walk ... |
| 21:22.49 | drmessano | Asterisk is like the girlfriend that cooks for you, cleans for you, does ANYTHING you want in bed (and is willing to learn new things on the fly), but she's got all those tattoos and piercings.. Cisco is like that hot preppy chick you met at the mall that wont cook, clean, or screw, drains your wallet, and can't spell the word "spell", but you look better standing next to her |
| 21:23.28 | p3nguin | I guess that's one way to look at it. |
| 21:23.49 | ultramancool | lol, great explanation drmessano :) |
| 21:25.01 | p3nguin | But the way I see it, my phones have Cisco logos on them with a company logo on the screen... beyond that, no one really knows what's on the other end of those blue Ethernet cables. |
| 21:25.09 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 21:25.48 | rossand | I've set up two lines (101, and 102). The client on both is xlite. I can call & connect. But there's no sound either direction. As additional information, leaving a voicemail records & works perfect. I'm looking for suggestions on debugging. I don't see any obvious error in the logs. Thanks! |
| 21:26.07 | drmessano | p3nguin: Yes, you parade the cisco chick around the office, then sneak off to the server room and bang the asterisk chick.. Best of both worlds.. |
| 21:26.12 | [TK]D-Fender | drmessano: my IP 501 ha an Asterisk logo on it,,,, |
| 21:26.25 | p3nguin | exactly |
| 21:26.47 | drmessano | Just don't let her get one of her piercings stuck on your packet sniffer |
| 21:27.00 | drmessano | You'll have to explain why it took you so long |
| 21:27.22 | dustybin | [TK]D-Fender: did you phreak phones back in the day |
| 21:27.41 | drmessano | HA |
| 21:27.42 | [TK]D-Fender | swats dustybin with an old copy of 2600 |
| 21:27.45 | drmessano | [TK]D-Fender: YOURE OLD |
| 21:27.51 | dustybin | heh |
| 21:28.15 | drmessano | How many years you got on me, TK? |
| 21:28.19 | p3nguin | rossand: Are you using NAT between the softphones and the asterisk server? |
| 21:28.33 | drmessano | pastebin.ca's a '76 |
| 21:28.42 | [TK]D-Fender | 75 |
| 21:28.43 | rossand | p3nguin: not in this case. they're both on the same network as asterisk |
| 21:28.46 | dustybin | im 76 |
| 21:30.11 | carrar | Dude if you can't make free calls from a pay phone with a popcan tearoff top you're a newbie |
| 21:30.24 | carrar | heh |
| 21:30.26 | drmessano | can't find a payphone |
| 21:30.30 | drmessano | sucks |
| 21:30.45 | carrar | taps out the the quarter drop payment clicks |
| 21:31.40 | [TK]D-Fender | LAME |
| 21:31.55 | [TK]D-Fender | Thats small-fry, and only gets you free calls... payphones used to make me MONEY |
| 21:32.19 | carrar | sledge hammer to the change box? |
| 21:32.35 | [TK]D-Fender | Cut the coin-return pair and let it queue up a ton, then reconnect, toss in a quarter & eject... FOR GREAT PROFIT! |
| 21:32.58 | [TK]D-Fender | carrar: No, it queues them before dropping so you get your coin back on a cancelled call |
| 21:33.38 | drmessano | drives to McDonalds with a stolen laptop, orders 3 free months of vonage under the name Teekay D. Fender and calls China, Japan, and the phone in the restroom of the ISS |
| 21:34.03 | *** join/#asterisk evil_gordita (n=evilgord@ip24-254-160-77.rn.hr.cox.net) |
| 21:34.32 | carrar | hacks drmessano with the FTP exploit since he left IIS running on his windows laptop |
| 21:34.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:35.12 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 21:35.29 | drmessano | looks puzzled at carrar wondering if he fixed the error that prevented IIS from running on Vista laptop to even exploit it |
| 21:35.57 | drmessano | iisreset's |
| 21:36.02 | drmessano | sobs :( |
| 21:39.40 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 21:46.14 | *** join/#asterisk SparFux (n=raoul@e182018110.adsl.alicedsl.de) |
| 21:46.17 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:8a8:cf6d:2ac0:762d) |
| 21:46.20 | cusco | hi |
| 21:46.34 | carrar | HI!!!!!!!!!!!!!!!!!!!!!! |
| 21:46.48 | cusco | clients calling and pressing keys.. dtmf troubles... sometimes client presses key 7 and it comes up twice |
| 21:46.57 | cusco | how can I limit them DTMF tones? |
| 21:47.08 | carrar | sounds like a old bug |
| 21:47.12 | carrar | double dtmf |
| 21:47.20 | cusco | does, doesn't it? |
| 21:47.20 | p3nguin | What's your asterisk version? |
| 21:47.21 | carrar | where it sends it both inband and rfc |
| 21:47.25 | cusco | hold |
| 21:47.31 | carrar | 2388 |
| 21:47.45 | cusco | Asterisk 1.6.1.1 |
| 21:48.13 | SparFux | Is there a way to show which asterisk modules use which ports? |
| 21:48.13 | p3nguin | Are you using SIP? |
| 21:48.29 | cusco | p3nguin: yes |
| 21:48.42 | p3nguin | In your sip.conf, try actually defining the dtmfmode as either rfc2833 or inband. |
| 21:49.28 | cusco | p3nguin: sorry, it is already defined as dtmfmode=rfc2833 |
| 21:49.40 | carrar | set it to dtmf then |
| 21:49.44 | carrar | err |
| 21:49.44 | p3nguin | Try switching to inband just for testing purposes. |
| 21:49.45 | carrar | inband |
| 21:49.48 | cusco | ok |
| 21:50.14 | cusco | reloaded it |
| 21:50.17 | cusco | let me test it |
| 21:50.25 | *** join/#asterisk exothermc (n=miles@74.85.89.193) |
| 21:50.46 | exothermc | grr trying to get a logo loaded on my polycom 601 isn't going smoothly. |
| 21:50.57 | exothermc | Anyone here done that by hand before? |
| 21:51.13 | p3nguin | Only on Cisco 7900 series for me. |
| 21:51.22 | carrar | by hand? |
| 21:51.27 | carrar | use a ftp config file |
| 21:51.57 | p3nguin | Does your phone grab tftp configs like Ciscos do? |
| 21:52.17 | carrar | they can, but I use ftp |
| 21:52.27 | carrar | so it uploads logs |
| 21:52.54 | p3nguin | my tftp logs, so that's all I really care about. |
| 21:53.11 | p3nguin | But I was wondering how the polycom is supposed to obtain the logo information. |
| 21:53.34 | carrar | from the SIPMAC.cfg |
| 21:54.07 | carrar | read your admin guide |
| 21:54.21 | carrar | sipMAC.cfg |
| 21:54.23 | p3nguin | I don't have a polycom, nor do I have a guide for it. |
| 21:54.34 | carrar | this is for exothermc |
| 21:54.48 | p3nguin | If it's the same as Cisco, just add your "logo_url: " line and reboot the phone. |
| 21:55.01 | carrar | yeah cisco's are easy |
| 21:55.10 | carrar | but so are polycom |
| 21:57.13 | *** join/#asterisk digilink_ (n=digilink@c-76-123-252-182.hsd1.tn.comcast.net) |
| 21:57.50 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 22:00.05 | exothermc | carrar: Ya spent some good time with that guide. |
| 22:00.14 | carrar | yes |
| 22:00.15 | exothermc | running 3.1.3 on that phone. |
| 22:00.23 | carrar | youhave too really |
| 22:00.23 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 22:00.35 | exothermc | carrar: I love how they tell you to edit the sip.cfg. |
| 22:00.43 | carrar | if you want to sent them up nicely |
| 22:01.03 | carrar | thats ok for a one off |
| 22:01.13 | carrar | but any amount of phones you end up writting cripts |
| 22:01.15 | carrar | scripts |
| 22:01.20 | carrar | to generate those files for you |
| 22:01.59 | carrar | polycom has excellent configurable phones |
| 22:03.24 | exothermc | ya I'm just trying to get a one off setup for logo as a proof of concept |
| 22:03.25 | dustybin | i want to set up some kind of way for my asterisk server to make a call at to a number at a certain time of the day, and play a sample. what would be the best way of doing this? |
| 22:03.58 | dustybin | would i need some kind of cron? |
| 22:04.05 | dustybin | or can this be done another way |
| 22:04.14 | carrar | dustybin, call file |
| 22:04.31 | dustybin | call = CLI command? |
| 22:04.35 | carrar | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 22:04.40 | dustybin | ace :D |
| 22:04.57 | exothermc | carrar: any idea what these are actually for IP_600 bitmap.IP_600.1 ? |
| 22:05.19 | carrar | settings |
| 22:05.35 | exothermc | ok and besides the obvious and vague? |
| 22:05.45 | carrar | bitmap.IP_600.65.name="polycomimg" |
| 22:05.56 | carrar | ind.anim.IP_600.29.frame.1.bitmap="polycomimg" |
| 22:06.05 | carrar | ind.anim.IP_600.30.frame.1.bitmap="polycomimg" |
| 22:06.23 | exothermc | I just can't find the significance of those IDs 29, 30, 65 etc |
| 22:06.29 | exothermc | if they have any at all. |
| 22:06.48 | carrar | they references |
| 22:06.50 | carrar | are |
| 22:07.14 | exothermc | arbitrary? Since I see other working configs with different IDs using the same sip version. |
| 22:07.15 | carrar | polycomimg is actually a file polycomimg.bmp |
| 22:07.23 | exothermc | ya that much I got. |
| 22:07.33 | exothermc | I have my phone actually pulling the image down |
| 22:07.37 | exothermc | but doesn't display it. |
| 22:07.46 | carrar | wrong format perhaps |
| 22:08.08 | exothermc | I made it a 1 bit bitmap under the specified size. |
| 22:08.11 | [TK]D-Fender | That tage ID's images for display. Other tags REFER to those img #'s |
| 22:08.31 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 22:09.31 | exothermc | [TK]D-Fender: That is what I thought, but I couldn't find another tag that referenced those #'s, but maybe that is ok for these tags. |
| 22:10.09 | exothermc | because it doesn't seem that bitmap.IP_600.65.name="polycomimg" does anything at all and it isn't referenced by anything. |
| 22:10.18 | *** part/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 22:10.22 | carrar | You see it request that file in your logs? |
| 22:10.32 | exothermc | carrar: yup twice. |
| 22:10.46 | carrar | Lets see your file |
| 22:11.21 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:11.43 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 22:11.44 | exothermc | carrar: http://pastebin.com/d152b7b77 |
| 22:11.56 | carrar | no the image file |
| 22:12.31 | exothermc | ahh jas |
| 22:14.19 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 22:14.20 | exothermc | http://dl.getdropbox.com/u/1805102/logo2.bmp |
| 22:15.11 | carrar | hrmm looks ok |
| 22:15.38 | exothermc | carrar: ya tied to make it dead simple. |
| 22:17.00 | carrar | You may need to use the entire sip cofnig |
| 22:17.10 | carrar | use the template that comes with the sip app source |
| 22:17.19 | carrar | don't remove stuff |
| 22:19.23 | carrar | and since you are gonna do that, also use the phone1.cfg template that comes with it also |
| 22:19.25 | exothermc | The entire sip.cfg ? |
| 22:19.29 | carrar | yes |
| 22:19.36 | carrar | and just change items you want to change |
| 22:19.53 | carrar | TK might have more insight on that |
| 22:20.04 | exothermc | Well that is called by default I just override it with macspecific file. |
| 22:20.18 | carrar | I know polycom support likes to have the phone/sip files match the release version of the software |
| 22:20.41 | jql | yes indeed |
| 22:21.02 | jql | use the sip.conf and phone1.conf that come with the firmware, they're there for good reason |
| 22:21.06 | exothermc | carrar: Ya all files except the overrides are straight from the release. |
| 22:21.25 | carrar | exothermc, I don't use a 'default' |
| 22:21.38 | carrar | each phone has it's entire file config |
| 22:21.46 | carrar | both phone * sip |
| 22:21.51 | carrar | as well as s MAC.cfg |
| 22:22.00 | carrar | just my pref |
| 22:22.10 | exothermc | carrar: that sounds like a bunch of work to maintain between releases. |
| 22:22.23 | carrar | by hand, probably |
| 22:22.36 | exothermc | carrar: I like to just isolate the stuff I muck with and leave the defaults to be called from the release file. |
| 22:23.17 | carrar | we'll you have to get it working 1st |
| 22:23.44 | raden_work | [TK]D-Fender, you have any ideas for my inbound21 and inbound22 . vitelity.net ????? as far as redundancy ? |
| 22:23.50 | exothermc | carrar: even if you are generating by scripts you don't have to merge an entire file on upgrade. This way you just upgrade sip replace release files and double check your overides to ensure nothing changed in the release. |
| 22:23.58 | exothermc | true that. |
| 22:24.16 | exothermc | Everything else works fine except this damn logo |
| 22:24.20 | [TK]D-Fender | raden_work: Not quite sure what is wrong in your case. |
| 22:24.30 | [TK]D-Fender | raden_work: trouble following you |
| 22:24.48 | exothermc | [TK]D-Fender: Any idea between that config file and that logo why this isn't working for me? |
| 22:24.57 | carrar | Overrides really for when people change the ring tones |
| 22:25.11 | carrar | and want those saved for the next reboot |
| 22:25.21 | carrar | or whatever |
| 22:25.22 | dustybin | is this the correct format for me dialing a outside number from my dial plan: |
| 22:25.23 | dustybin | exten => s,1,Dial(IAX2/MYID@voiptalk/{OUTGOINGNUMBER}) |
| 22:25.37 | exothermc | carrar: well ya those types, but you can do the exact same thing with a small file for your own settings. |
| 22:25.50 | augustf | anyone know why on a new asterisk install (on debian with freepbx on top), internal extensions would get a SIP 603 (refused) when trying to call each other? |
| 22:25.53 | exothermc | carrar: or phone and group specific settings. |
| 22:25.58 | carrar | jsut complicates everything |
| 22:26.36 | carrar | I just follow the admin guide |
| 22:26.40 | carrar | seems to work best |
| 22:27.08 | raden_work | [TK]D-Fender, we loose registration with vitelity on our inbound21.vitelity.net host about 2 or 3 times a day have to switch to vitelity22.inbound.net which adds a crap load of latency 20ms + but we always have a route to it ( dont ask me ) i have talked to my isp the next isp on the hop no one can figure it out |
| 22:27.41 | exothermc | carrar: Ya what I'm describing is the suggested method from polycom. |
| 22:27.54 | raden_work | [TK]D-Fender, so i was wondering if somehow i can set asterisk if we loose vitelity21.inbound.net it can try the other server vitelity22.inbound.net |
| 22:28.02 | raden_work | inbound21.vitelity.net |
| 22:28.11 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 22:28.17 | raden_work | sorry having issues with freaking asa 5505 atm too brain a lil scattereed |
| 22:28.21 | [TK]D-Fender | exothermc: mode=3, and s/46/30/ |
| 22:28.29 | [TK]D-Fender | errr 29 |
| 22:28.38 | exothermc | mode=3 where? |
| 22:28.49 | exothermc | [TK]D-Fender: s/46/29 ? |
| 22:29.23 | [TK]D-Fender | raden_work: You mean for making your outbound reg alternate? Get a SIP proxy |
| 22:29.32 | raden_work | inbound reg |
| 22:29.38 | [TK]D-Fender | raden_work: or add both and let them duke it out |
| 22:29.50 | [TK]D-Fender | raden_work: WHT is *inbound* reg? |
| 22:29.52 | exothermc | ahh ind.idleDisplay.mode="3" |
| 22:29.58 | [TK]D-Fender | raden_work: You reg to ITSPs, not the other way around |
| 22:30.24 | raden_work | ok this is messed up but this is what we have |
| 22:30.40 | raden_work | tell yeah what i post a pastebin :P |
| 22:30.43 | [TK]D-Fender | raden_work: ummm, no that sounds quite wrong |
| 22:30.49 | raden_work | or im just going to confuse you |
| 22:30.59 | raden_work | i set it up the way vitelity told me toooo |
| 22:32.09 | [TK]D-Fender | Translation = CLUE[-1] |
| 22:38.18 | exothermc | [TK]D-Fender: ok updated and didn't help |
| 22:38.43 | [TK]D-Fender | exothermc: Show more... |
| 22:43.15 | exothermc | [TK]D-Fender: sip.cfg = http://pastebin.com/d46c8c454 , phone1.cfg = http://pastebin.com/d6c336e4d localsettings.cfg = http://pastebin.com/d480089af shown in increasing order of priority. |
| 22:46.16 | [TK]D-Fender | exothermc: Dump your provisioning folder |
| 22:46.52 | exothermc | k |
| 22:48.23 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 22:50.04 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 22:50.45 | SparFux | You know, when I read about DAHDI on the web, it does not seem to me that it is a reimplementation. |
| 22:51.07 | SparFux | It sounds much more like it's basically a later version of zaptel with a different name. |
| 22:51.13 | [TK]D-Fender | ~dahdi |
| 22:51.14 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 22:51.33 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 22:51.37 | raden_work | [TK]D-Fender, http://pastebin.com/m74549075 |
| 22:51.38 | [TK]D-Fender | SparFux: had to rename so took the chance to make more changes too |
| 22:52.01 | SparFux | Aha. ok. |
| 22:52.28 | raden_work | i dont understand why the vitel-inbound needs to be there |
| 22:52.55 | [TK]D-Fender | raden_work: because the host differes from outbound |
| 22:52.58 | raden_work | besides for the fact to register where to send calls |
| 22:53.07 | raden_work | just odd most places do it as one dont they ? |
| 22:53.12 | raden_work | so thats my issue |
| 22:53.36 | [TK]D-Fender | raden_work: I fail to see an issue. |
| 22:53.38 | raden_work | inbound21 goes down i need inbound22 which is crap quality but the calls come through |
| 22:53.44 | [TK]D-Fender | raden_work: Name auth each separately |
| 22:54.25 | [TK]D-Fender | raden_work: So make another peer for it |
| 22:54.39 | raden_work | wont that screw everyhting up if i register twice ? |
| 22:54.46 | raden_work | thats our inbound |
| 22:55.39 | [TK]D-Fender | raden_work: Who said anything about a separate REGISTER? |
| 22:56.04 | [TK]D-Fender | roll-eyes |
| 22:56.12 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 22:56.33 | raden_work | so i can have 2 peers and be registered in global and itll simply choose the best route ? |
| 22:57.38 | raden_work | what does the peer vitel-inbound tell Vitelity exaclty just how to get ahold of my asterisk box correct ? |
| 22:57.43 | raden_work | or what ip its sitting at ? |
| 22:57.52 | raden_work | im missing a part of the logic here im still learning |
| 22:57.53 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
| 22:58.07 | [TK]D-Fender | raden_work: Yuo don't choose who SENDS you calls |
| 22:58.46 | *** part/#asterisk SparFux (n=raoul@e182018110.adsl.alicedsl.de) |
| 22:58.54 | p3nguin | You do choose where they go once they come in, though. |
| 22:59.06 | raden_work | so vitelity-inbound is just saying hi to vitelity im at this address |
| 22:59.23 | raden_work | so if i do it twice it will just tell it to 2 diffrent servers |
| 22:59.29 | [TK]D-Fender | raden_work: NO |
| 22:59.34 | [TK]D-Fender | gah |
| 22:59.41 | raden_work | well explain it to me then |
| 22:59.49 | [TK]D-Fender | raden_work: the peers MATCH the call that they are ALREADY SENDING YOU. |
| 22:59.58 | [TK]D-Fender | raden_work: It does not inform THEM of jack shit |
| 23:00.29 | [TK]D-Fender | raden_work: Your REGISTER statement tells them your IP. Your PEERS try to match the call so you can separate and process them |
| 23:01.23 | raden_work | ok |
| 23:01.23 | p3nguin | What's he trying to do, a failover on incoming calls? |
| 23:01.41 | raden_work | p3nguin, yes cause i keep loosing my inbound peer |
| 23:02.16 | exothermc | [TK]D-Fender: still packing, not a small dir |
| 23:02.24 | p3nguin | I think I would want to figure out why it's dropping rather than trying to configure a second peer for the same provider. |
| 23:03.10 | raden_work | inbound 18 , 19, 20 @ the moment im getting Unable to look up 64.2.142.218: Temporary failure in name resolution |
| 23:03.18 | raden_work | for all 3 now on inbound 21 |
| 23:03.22 | KyleK | is there any voip phones that'll kick out a lot of volume on the handset for hard of hearing peoples? |
| 23:03.41 | *** join/#asterisk pimpwell (n=domin8@ool-ad03dcac.dyn.optonline.net) |
| 23:04.28 | raden_work | KyleK, i can barely hear on any phone and i have to turn my aastra 9133i's down to about 3/4 volume |
| 23:05.21 | [TK]D-Fender | raden_work: So your issue is DNS. |
| 23:05.40 | [TK]D-Fender | raden_work: So allow anon calls and dump them into the same context that matches your DID's |
| 23:05.56 | raden_work | tried 3 diffrent dns servers |
| 23:07.53 | KyleK | raden_work: just what I need :D |
| 23:08.10 | raden_work | KyleK, you check em out ? |
| 23:08.32 | raden_work | KyleK, thats what i put in here they're louder than the polycoms i tested and there elcheapo |
| 23:09.15 | raden_work | you want a good cheap heaset checkout the VXI's on VOIPsupply.com just as good as our expensive plantronics and 1/4 the price |
| 23:11.07 | *** join/#asterisk propellerhead (n=yogurt2u@190.245.197.216) |
| 23:13.15 | p3nguin | They look like 10x as much as the plantronics I was going to get for my Cisco phones. |
| 23:13.24 | wonderworld | hey, i have got MOH problems.... i have just one wave file in /var/lib/asterisk/moh. when starting MOH the CLI says: [Sep 8 01:12:04] WARNING[3861]: file.c:650 ast_openstream_full: File /var/lib/asterisk/moh/CHANGES-asterisk-moh-opsound-2 does not exist in any format ? |
| 23:14.13 | exothermc | [TK]D-Fender: can I pm you the url to download? |
| 23:20.31 | exothermc | [TK]D-Fender: All packed up for you and ready to go. |
| 23:21.36 | [TK]D-Fender | exothermc: Will follow up on later. Got to head out for a while |
| 23:21.43 | exothermc | ok |
| 23:21.53 | exothermc | can you pull the download so I can hose the file? |
| 23:29.19 | raden_work | p3nguin, ???? |
| 23:29.40 | raden_work | p3nguin, i paid like 55 or something for them |
| 23:35.42 | riddlebox | does anyone work on avaya equip here? I heard a rumor that avaya is dropping the ACS next year? |
| 23:36.11 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 23:44.29 | *** join/#asterisk lost_soul (i=shawn@cpe-67-241-66-205.twcny.res.rr.com) |
| 23:44.31 | *** join/#asterisk s14ck (n=s14ck@190-76-70-229.dyn.movilnet.com.ve) |
| 23:45.15 | KyleK | raden_work: well i dont want a headset as the application is a deaf grandmother, I need a phone thats loud, self powered, and possibly disguisable as a regular phone ;) |
| 23:46.14 | *** join/#asterisk Draegon (n=longng78@ip72-223-24-112.ph.ph.cox.net) |
| 23:46.19 | raden_work | KyleK, so simpler the better ? |
| 23:46.41 | KyleK | yea |
| 23:46.45 | raden_work | <PROTECTED> |
| 23:46.49 | raden_work | single line |
| 23:46.51 | raden_work | very loud |
| 23:47.01 | KyleK | k |
| 23:47.01 | raden_work | the ringer will make you go deaf if you within 30 ft of it |
| 23:47.24 | raden_work | i can hear the one in the back warehouse from 300 ft away in my running diesel truck |
| 23:47.38 | raden_work | and there cheap |
| 23:56.15 | Orbixx | How can I implement group voicemail? |
| 23:57.59 | Orbixx | So one number dialled by anybody gets them into their own voicemail. |
| 23:59.50 | Draegon | hi guys, is there an easier way to bridge 2 call if i know the 2 channel? or is parking/meetme the only way? using trixbox 2.6, i think it has asterisk 1.4 and i read somewhere 1.6 has bridge command? |
