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01:24.45 | staffmember | anyone awake ? |
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01:26.07 | p3nguin | Is that all you want to know, or did you have a question about Asterisk? |
01:26.58 | carrar | I'm your huckleberry |
01:27.38 | paulc | I'm awake, but fading fast |
01:27.45 | staffmember | just awkward silence, but yeah there is a question.. anyone know where i can find some docs on nortel and asterisk |
01:27.48 | staffmember | whitepapesr |
01:27.54 | staffmember | *whitepapers |
01:27.58 | carrar | google |
01:28.03 | staffmember | did that and got what i watned |
01:28.04 | voipmonk | staffmember: t1? |
01:28.08 | paulc | What kind of docs? like Nortel using Asterisk? Integration between BCM or DMS and Asterisk? |
01:28.11 | staffmember | but wanted to know if anyone else had some sources i dont know about |
01:28.15 | carrar | like the unistm driver? |
01:28.17 | staffmember | paulc: yeah |
01:28.34 | staffmember | voipmonk: AEX800 |
01:28.52 | carrar | I'm installed asterisk between a nortel and the PSTN |
01:28.59 | carrar | I've |
01:29.13 | paulc | I've got a friend who's done that too.. PSTN <--> Asterisk <--> Option 11C |
01:29.29 | paulc | I recall the Opt11 was a bitch to configure, but once done, all was fine and happy |
01:29.42 | staffmember | paulc: how about PSTN -> Asterisk -> Nortel Compact ICS + callpilot |
01:29.46 | carrar | This was PRI on both sides |
01:34.16 | paulc | staffmember: Compact ICS is like "Norstar" right? |
01:34.20 | paulc | that should be easy |
01:34.23 | paulc | I love those systems :) |
01:36.49 | paulc | staffmember: What do you want to do with it exactly? Can you give me the 1 or 2 sentence "It should work like this..." ? |
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01:53.39 | simplydrew_ | I've been wanting to get a DID for my asterisk box that I can have forward to my cell phone, just as a "work only" number essentially to reach me on the go. Although I know I can configure this without a problem, is it also possible to forward text messages to my cell if sent to the DID? |
01:54.13 | *** join/#asterisk cadmium (i=mike@217.194.139.22) |
01:54.41 | cadmium | hi what channel variable can I use to get the peername of the phone which is placing the call |
01:56.00 | paulc | simplydrew_ It's been talked about but SMS forwarding is still a bit up in the air it seems. Hard to find a DID provider that supports it. |
01:56.08 | paulc | That said, if you find one, let me know |
01:56.28 | carrar | SIPCHANINFO(peerip) |
01:56.28 | simplydrew_ | paulc: ah gotcha, so the dependency is all reliant on the DID provider as well then? |
01:56.35 | carrar | oh |
01:56.37 | carrar | peername |
01:56.38 | paulc | my understanding is that most will ping a URL when you get sent a text message (but maybe some provide it as SIP messaging?) |
01:56.59 | simplydrew_ | hmm. I see. I'm with vitelity. I'll have to ask them. |
01:57.00 | paulc | simplydrew_ yes - The DID provider has to be a recognised "end point" for SMSs to be sent to |
01:57.01 | carrar | SIPCHANINFO(peername) |
01:57.17 | paulc | I'm with them too - they're pretty good (but don't do SMS as far as I know?) |
01:57.42 | p3nguin | Maybe the SIPPEER() function would be better? |
01:58.27 | cadmium | aghh got it myself ${CALLERID(num)} |
01:58.54 | p3nguin | That's not the PEER INFO |
01:59.37 | p3nguin | The CALLERID() function is only as reliable as the person who set his caller ID info. It is easily spoofed. |
01:59.57 | p3nguin | carrar is right -- SIPCHANINFO(peername) is probably the best. |
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02:00.23 | random_mike | greetings p3nguin :) |
02:00.25 | carrar | well that this point the user doesn't even know what he's looking for |
02:00.41 | p3nguin | random_mike: hi |
02:02.15 | random_mike | thanks for your ongoing assistance |
02:02.15 | *** part/#asterisk paulc (n=paulc@unaffiliated/paulc) |
02:02.16 | *** join/#asterisk Kevin10258 (n=airmadne@173-112-45-135.pools.spcsdns.net) |
02:02.22 | Kevin10258 | Hello all.. |
02:02.25 | random_mike | my asterisk box is nearly complete :) |
02:02.32 | Kevin10258 | Is SipFoundry Asterisk based? |
02:02.41 | Kevin10258 | Does anyone know |
02:03.04 | p3nguin | random_mike: Are you confident enough in your doings to reveal your works to the world? Specifically, to #Asterisk? |
02:03.38 | random_mike | perhaps once its 100% |
02:03.51 | random_mike | just ironing out the last few bugs, adding the last features |
02:04.08 | random_mike | ive no doubt i'll be hassling for more assistance when I get stuck again :P |
02:04.15 | Kevin10258 | random_mike : What did you use to create your PBX? |
02:04.33 | carrar | rpm -i |
02:04.41 | carrar | heh |
02:05.03 | random_mike | Kevin10258, I'm running Asterisk 1.4.28 under CentOS 5.4 |
02:05.03 | Kevin10258 | I am trying to decide on using a distro like Elastic or creating my own on Defora |
02:05.07 | Kevin10258 | Fedora |
02:05.22 | Kevin10258 | Did you install all the RPMs or was it an ISO install? |
02:05.35 | random_mike | I enabled the atrpm's repo, and did a yum install |
02:05.35 | p3nguin | neither |
02:05.45 | Kevin10258 | I c |
02:06.02 | Kevin10258 | I used to used TrixBox, and I loved it. |
02:06.20 | random_mike | although to get everything correct, I had to enable testing and bleeding edge to the base repo. |
02:06.24 | random_mike | I've not used TrixBox |
02:06.27 | Kevin10258 | But the server was to loud, drowning out the TV, so it's down until I find a new one that doesn't sound like a 747 taking off in my tv room. |
02:06.37 | p3nguin | I run two asterisk servers -- one is ArchLinux, the other is FreeBSD. |
02:06.55 | Kevin10258 | Just another layer over the top of Asterisk to make it easy. |
02:07.29 | p3nguin | Maybe you want an Atom system so you can run fanless. |
02:07.35 | *** part/#asterisk cadmium (i=mike@217.194.139.22) |
02:07.48 | Kevin10258 | I have thought about that. |
02:07.49 | random_mike | My server is virtualised on a Dell m1000 blade chasis :P |
02:08.19 | Kevin10258 | I have also thought about a small p3 based system without a fan. Cheaper, and the same performance as the Atomless. |
02:08.37 | Kevin10258 | But I really want a RACK mounted solution. |
02:08.49 | Kevin10258 | To go in my rack with my HP Procurve 2948. :D |
02:09.14 | random_mike | any rack server is going to be noisy, blame the industry for that. |
02:09.29 | random_mike | perhaps you can Atom it, and mount some super big ears to it, and rack it up :P |
02:09.40 | carrar | Kevin10258, you have our permission. Please make it so |
02:10.06 | Kevin10258 | huh? |
02:10.07 | Kevin10258 | lol |
02:10.18 | Kevin10258 | Yeah, I am working on paying off the credit cards first. |
02:10.29 | carrar | Thats not very American |
02:10.40 | Kevin10258 | Yeah, well my wife and kids are cherokee |
02:10.46 | Kevin10258 | so we don't americans very much. |
02:10.47 | carrar | Free money! |
02:10.47 | Kevin10258 | :D |
02:10.52 | Kevin10258 | yeah right |
02:10.55 | Kevin10258 | that's a MYTH dood. |
02:10.59 | carrar | haha |
02:11.05 | carrar | you 1/4 or more? |
02:11.08 | Kevin10258 | My wife is certified, as are our kids. |
02:11.11 | carrar | err 1/8 |
02:11.19 | Kevin10258 | My wife is, and my kids are, I am not certified. |
02:11.31 | carrar | Do they call you 'Running with Asterisk' |
02:11.31 | random_mike | you americans are weird :P |
02:11.32 | Kevin10258 | They have their RES cards and all, there is no free money. |
02:11.44 | Kevin10258 | I am not American, I am a leftist. |
02:11.45 | carrar | heh |
02:11.49 | Kevin10258 | Sorta |
02:12.01 | Kevin10258 | Well mate.. |
02:12.19 | Kevin10258 | We could say the same about Ausie, and what was done to the natives there. |
02:12.30 | random_mike | Lets not walk down that alley |
02:12.34 | Kevin10258 | LOL |
02:12.48 | random_mike | it's despicable. And continues to be. |
02:12.59 | Kevin10258 | We don't "celebrate" thanks giving. |
02:13.30 | Kevin10258 | My oldest daughter and I talk about how Columbus trickd the indians, enslaved them, and killed those who would not comply. |
02:13.44 | Kevin10258 | Never forget. |
02:13.59 | carrar | Indians had weapons of mass desctruction |
02:14.05 | Kevin10258 | ROFL |
02:14.07 | carrar | they had to be shut down |
02:14.18 | Kevin10258 | <PROTECTED> |
02:14.22 | Kevin10258 | Wait, I am white.. ;) |
02:15.02 | carrar | be proud of you who you are |
02:15.06 | random_mike | im sorry I dont know the history of the Indian's in the US |
02:15.13 | random_mike | Other than there's not many left |
02:15.25 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
02:15.34 | Kevin10258 | Yeah, it's TERRIBLE. Watch "A peoples history" and you will find the truth. |
02:15.44 | Kevin10258 | And that's only some of it. |
02:15.52 | random_mike | Terrible as in Aboriginal Australians terrible? |
02:15.55 | Kevin10258 | Kill de white people... He he he o. |
02:15.57 | Kevin10258 | lol |
02:16.09 | Kevin10258 | I know nothing of what has happened to them. |
02:16.36 | Kevin10258 | But the indians were mass murdered, forced into slavery in some cases, and then forced onto reservations because they were hard to look at. |
02:16.50 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
02:16.58 | Kevin10258 | They were forced to abandoned their beleifs, language etc. |
02:17.19 | Kevin10258 | I get pretty pissed when I think about it, not much different that what is happening to us as a people today. |
02:17.29 | Kevin10258 | Minus most of the murdering. |
02:18.03 | random_mike | right - well the Aboriginal Australians were mass murdered, chased off into the bush, and then in the 50's and 60's the Australian Gov had a great idea to go and take every Aboriginal Child and place them in Foster Care, to teach them white australian ethics. |
02:18.15 | random_mike | this ofcourse didnt work |
02:18.44 | Kevin10258 | LOL |
02:18.51 | Kevin10258 | Asterisk is hard man. |
02:18.55 | Kevin10258 | but awsome when it works. |
02:19.17 | random_mike | agreed - all this damn history - wait... asterisk? I thought I was in #history for a second then.,... |
02:19.24 | random_mike | Asterisk IS awesome when it works |
02:20.59 | random_mike | Anyone know a good wakeup call script/scripts/service to stick on my asterisk box? :D |
02:21.19 | random_mike | I've experimented with a few but didnt really find them that appealing. |
02:23.54 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
02:26.39 | simplydrew_ | paulc: I got a response from vitelity about the DID SMS. They said they are in the final stages of SMS platform completion. So sounds like it'll be available son |
02:26.40 | simplydrew_ | *soon |
02:29.54 | Kevin10258 | Does anyone know of a GOOD cost effective SIP provider? |
02:30.50 | carrar | ~itsp |
02:30.50 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
02:30.55 | random_mike | heh - too bad you're not in Australia or I'd suggest my company |
02:31.18 | carrar | What state are in? |
02:33.15 | Kevin10258 | Missouri |
02:33.19 | Kevin10258 | AKA MYsery |
02:34.39 | p3nguin | kevin10258: VoIP.ms |
02:35.05 | p3nguin | kevin10258: I would say Flowroute, but I am told they are declining new sign-ups right now. |
02:45.01 | random_mike | query regarding extensions.conf |
02:45.27 | random_mike | I have my extension set up like: exten => *99,1,Answer |
02:45.27 | p3nguin | !ask |
02:45.46 | random_mike | if the user then press's 1, their directed to that area of the conf |
02:46.09 | random_mike | exten => 1,1,Set(SUCCESSCOUNT=$[${SUCCESSCOUNT} + 1]) |
02:46.13 | p3nguin | "area of the conf" |
02:46.48 | random_mike | i think this call's for a paste bin |
02:47.06 | p3nguin | What's with the random apostrophes? |
02:48.40 | random_mike | im random |
02:48.45 | random_mike | i sec I broke something |
02:48.50 | random_mike | *tears out hair again* |
02:51.23 | jaytee | ever notice that most older I.T. professionals are balding or have very thin hair? ever wonder why? |
02:51.28 | *** join/#asterisk RobH_ (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
02:54.34 | Kevin10258 | They don't have "unlimited" plans. |
02:55.13 | Kevin10258 | Yeah, "F" that.. |
02:55.20 | Kevin10258 | Callcentric is at least unlimite |
02:55.21 | Kevin10258 | Callcentric is at least unlimited |
02:55.53 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
02:56.27 | Kevin10258 | jaytee : Because we have been watching it ALL FALL APART for years. |
02:56.28 | Kevin10258 | :D |
02:58.38 | p3nguin | kevin10258: CallCentric also isn't that good. |
02:59.27 | p3nguin | kevin10258: Well, flowroute is letting me sign up, so I don't know what's up with them. |
03:00.33 | p3nguin | kevin10258: Flowroute's Flat Rate Unlimited plan - Bundles a DID with two channels for unlimited free incoming calls to that number, but no more than two simultaneous calls. |
03:00.44 | p3nguin | kevin10258: $6.95/mo |
03:01.53 | *** join/#asterisk netpro25_ (n=mmanning@64-238-176-248.ksg.apt.gru.net) |
03:04.02 | netpro25_ | Hello, I am having issues with hold music. Asterisk is saying the hold music is started but I hear nothing. I am using mp3s. |
03:04.12 | netpro25_ | Asterisk says: Started music on hold, class 'default', on channel 'SIP/desktop-ec3b3628' |
03:04.41 | netpro25_ | I have asterisk 1.6 |
03:06.59 | *** join/#asterisk hobbes615 (n=ryan@c-68-48-8-21.hsd1.dc.comcast.net) |
03:07.15 | hobbes615 | join #freepbx |
03:13.36 | *** join/#asterisk _ShrikE (n=_ShrikE-@adsl-074-185-215-060.sip.msy.bellsouth.net) |
03:15.18 | _ShrikE | In 1.4, will asterisk match the username or ip address first if type=friend for sip inbound call? |
03:31.00 | Kevin10258 | 6.95 per months? |
03:31.01 | Kevin10258 | What? |
03:32.05 | p3nguin | kevin10258: what? |
03:33.15 | Kevin10258 | <p3nguin> kevin10258: Flowroute's Flat Rate Unlimited plan - Bundles a DID with two channels for unlimited free incoming calls to that number, but no more than two simultaneous calls. |
03:33.16 | Kevin10258 | <p3nguin> kevin10258: $6.95/mo |
03:33.32 | p3nguin | Okay, so what's the problem? |
03:34.08 | Kevin10258 | I don't beleive it |
03:34.13 | Kevin10258 | They don't state it on their website |
03:34.22 | Kevin10258 | I am paying nearly 4 times that for 1 line |
03:34.27 | p3nguin | Actually they do. That's how I was able to give you the info. |
03:34.27 | Kevin10258 | as it stands right nowe |
03:34.36 | Kevin10258 | Where are you seeing this? |
03:34.40 | p3nguin | VoIP.ms is about the same rate as Flowroute, too. |
03:34.44 | Kevin10258 | I see a point click run around nightmare |
03:35.01 | p3nguin | https://www.flowroute.com/services/inbound/ |
03:35.43 | Kevin10258 | That's inbound, what about outbound? |
03:35.55 | p3nguin | http://voip.ms/dids.php?action=statesearch&state=MO |
03:36.20 | p3nguin | Outbound with flowroute is $0.0098/minute |
03:36.40 | p3nguin | or $19.95/mo. with CallCentric. |
03:37.12 | Kevin10258 | Yeah, I will stick with CC, I use the phone WAY to much to have that over my head. |
03:37.20 | Kevin10258 | Their support is pretty good as well. |
03:37.43 | p3nguin | You use more than 2000 outbound minutes each month? |
03:37.49 | Kevin10258 | WAY |
03:37.57 | Kevin10258 | I have 5 kids, a wife and myself. |
03:38.15 | Kevin10258 | I do A-LOT of side work and testing for other people that reuires me to be on the phone. |
03:39.02 | Kevin10258 | Even if I didn't I am not big on "limited" plans anyway, call it the "American" in me. |
03:39.16 | Kevin10258 | I ALWAYS get screwed at some point. |
03:40.39 | *** join/#asterisk d00gster (n=doughant@77.31.197.112) |
03:42.56 | dlynes | Kevin10258, vitelity has an all you can eat plan |
03:43.12 | p3nguin | for termination? |
03:43.27 | p3nguin | It should be 1.4c/minute for termination. |
03:43.29 | ChannelZ | How many minutes do you use? .0098/min is nothing |
03:44.16 | p3nguin | He uses far more than 2000 minutes, so the $19.95/mo. plan with CallCentric seems to be more cost effective. |
03:44.49 | p3nguin | That's $19.95/mo. for unlimited outbound calling. |
03:45.13 | p3nguin | Most ITSPs don't do unlimited outbound rates. |
03:45.57 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
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04:25.08 | titter | Anyone know of a good IAX provider? |
04:36.14 | random_mike | anyone know what RFC asterisk prefers? |
04:42.39 | p3nguin | titter: VoIP.ms |
04:43.50 | *** join/#asterisk FirstSgt (n=cheney@cpe-72-181-174-190.tx.res.rr.com) |
04:44.33 | FirstSgt | Does anyone have a network topology map for how SIP Clients, Softphones, PBX, SIP, DID, and DOD work? |
04:44.55 | p3nguin | I don't get it. |
04:46.38 | FirstSgt | Neither do I... it's a complex subject. |
04:46.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
04:46.52 | FirstSgt | i dont even know what to google for |
04:51.35 | p3nguin | Can someone verify the stock price on Charter Communications for me, please? |
04:51.44 | p3nguin | If it is what I think it is, I think I am rich. |
04:52.09 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
04:53.59 | ChannelZ | depends what market and counyty, seems they are listed in a bunch of places |
04:54.30 | p3nguin | USA, NASDAQ |
04:57.19 | daidoji | 31.75 |
04:57.21 | daidoji | you rich? |
04:58.16 | daidoji | although its been in decline for the past couple of days |
04:58.25 | daidoji | and didn't you get wiped out in the bankruptcy? |
04:58.34 | daidoji | or did you buy like 5 days ago? |
04:59.03 | p3nguin | I lost $500 on it just today alone. |
04:59.22 | p3nguin | I bought months ago when it was under $0.25 per share. |
04:59.35 | daidoji | o well then you are rich |
04:59.47 | daidoji | cause 31.75 is a lot higher than .25 |
05:00.04 | daidoji | although I thought you would have gotten wiped out in the bankruptcy |
05:00.38 | ChannelZ | only rich if you sell it |
05:00.49 | daidoji | ehhh I'd keep it |
05:01.09 | daidoji | Charter's a good company, they re'orged, and they've turned nice profits in the past |
05:01.22 | daidoji | now that cheap credit is over and they don't have to try and expand everywhere they'll do all right |
05:01.29 | ChannelZ | we're shooting an internal 'rah-rah' video for them |
05:01.33 | daidoji | especially if you bought at .25 |
05:03.04 | daidoji | actually if you wanna go really crazy on it and make some serious money |
05:03.13 | dlynes | random_mike, your question is totally random, and makes absolutely no sense |
05:03.15 | daidoji | short it for about 3 days on margin |
05:05.10 | random_mike | dlynes, I know that - but where I sit... We have a SIP server running (slipper) and an upstream SIP provider (isphone) and they are very odd about which SIP RFC they support |
05:05.48 | random_mike | for a call to be marked as a private number, they ignore the P-Asserted-ID tag, and the Privacy: id tag, and only rely on the FROM field being re-written to Anonymous |
05:06.18 | random_mike | it's my understanding that re-writing the from field is an older standard, although good practice now, it's not the only method for making a call as private. |
05:08.07 | random_mike | you'll have to forgive me, after being thrown in the VoIP deep-end I'm struggling to float ;) |
05:08.34 | dlynes | random_mike, it's not a matter of which rfc, but rather which version of sip, and how fully they support that version |
05:08.48 | dlynes | random_mike, asterisk supports all of sip v2 (afaik), and part of sip v3 |
05:08.56 | dlynes | afaik, again |
05:09.01 | random_mike | thankyou dlynes |
05:09.03 | dlynes | that being said |
05:09.40 | dlynes | oej is working on a complete sip v3 stack for asterisk...I heard rumors it was going into asterisk 1.6.x, but I'm not so sure that ever happened |
05:10.43 | dlynes | random_mike, however, a quick way to see which method it's using for the particular behaviour you're talking about is to enable sip debug during a call where you expect to see that, so you can see how asterisk does it |
05:10.55 | dlynes | random_mike, or use pcap or pcapsipdump |
05:11.09 | dlynes | ~pcapsipdump |
05:11.21 | dlynes | hrm...i guess the bot doesn't know about pcapsipdump |
05:11.38 | dlynes | random_mike, pcapsipdump.sf.net, iirc |
05:11.58 | dlynes | it'll dump every single sip conversation into a separate pcap file |
05:12.10 | dlynes | great for debugging sip traffic |
05:12.12 | [TK]D-Fender | throws a pineapple at dlynes |
05:12.17 | dlynes | [TK]D-Fender, ? |
05:12.39 | [TK]D-Fender | dlynes: Check your history... |
05:12.47 | dlynes | [TK]D-Fender, which history is that? |
05:12.54 | [TK]D-Fender | * |
05:13.04 | dlynes | [TK]D-Fender, regarding sip version numbers? |
05:13.40 | dlynes | [TK]D-Fender, is the sip v3 stack going into asterisk 1.8, then? |
05:15.21 | dlynes | random_mike, anyways...here's a sip compliance document for your perusal: http://svn.asterisk.org/svn/asterisk/team/oej/sip-compliance/asterisk-sip.txt |
05:16.23 | dlynes | [TK]D-Fender, hrm...you throw a pineapple at me, and then run off like a little girl.... :( |
05:16.46 | random_mike | sorry was afk |
05:16.50 | ChannelZ | maybe he went to get more pineapples |
05:17.00 | dlynes | ChannelZ, or maybe some mangoes |
05:17.19 | dlynes | or maybe he's just not fruity enough |
05:17.51 | dlynes | random_mike, is that file what you were looking for, then? |
05:18.18 | random_mike | yes dlynes, this looks deliciously juicy. |
05:18.38 | random_mike | whilst you're here, and you seem to know your stuff |
05:18.48 | dlynes | random_mike, i don't know my stuff that well :) |
05:19.04 | random_mike | is it possible, using extensions.conf to find if a "Privacy: id" tag is present? |
05:19.07 | dlynes | random_mike, at least not the super technical protocol level stuff |
05:19.18 | [TK]D-Fender | dlynes: http://www.google.ca/#hl=en&source=hp&q=asterisk+SIP+pineapple&btnG=Google+Search&meta=&aq=f&oq=asterisk+SIP+pineapple&fp=ada65163c2197ad6 |
05:19.44 | [TK]D-Fender | "Psst. Top Secret... Codename Pineapple | Asterisk 24 Nov 2006 ... So what is the chan_sip3 project (codename pineapple) about? " |
05:20.07 | [TK]D-Fender | dlynes: There have been 3-4 simultaneous complete rewrites in the roks for almost half a decade |
05:20.12 | [TK]D-Fender | works* |
05:20.33 | dlynes | [TK]D-Fender, so what you're saying in a roundabout way, is that you have absolutely no confidence it'll get completed in our lifetime? |
05:20.49 | random_mike | I've found this regarding privacy headers: exten => _9.,1,SIPAddHeader(P-Asserted-Identity: <sip:+1${CALLERID(num)}\;user=phone>) |
05:20.56 | random_mike | but what I seek is Privacy: id header |
05:21.16 | dlynes | random_mike, why not use sipaddheader for it, too? |
05:21.30 | random_mike | I dont know what sipaddheader is :) |
05:21.40 | random_mike | although i've now consulted the google bible |
05:21.47 | random_mike | so hopefully this paints a pretty picture for me |
05:22.19 | random_mike | ok - clarification - sipaddheader only add's a header? |
05:22.35 | random_mike | it cant be used in a GotoIf, if the header is present? |
05:23.22 | dlynes | random_mike, how about the function, 'CALLERPRES'? |
05:23.55 | *** part/#asterisk nny (n=scott@64.203.239.83) |
05:24.01 | dlynes | random_mike, type 'core show function CALLERPRES' at the asterisk cli |
05:24.21 | random_mike | bf1*CLI> core show function CALLERPRES |
05:24.21 | random_mike | No function by that name registered. |
05:24.26 | random_mike | that's not fun |
05:24.28 | dlynes | random_mike, yes...sipaddheader only adds a header |
05:24.40 | random_mike | dlynes, perhaps I should further clarify |
05:25.03 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) |
05:25.13 | dlynes | random_mike, which version of asterisk are you using? |
05:25.30 | dlynes | random_mike, CALLERPRES as a function only exists in asterisk 1.6 |
05:25.53 | dlynes | random_mike, in earlier version it was SetCallerPres() (a dialplan application) |
05:26.03 | random_mike | my server accepts incoming calls only. Based upon the existance of the Privacy: id, I can tell whether or not the call has callerid suppression enabled or not. And depending on whether or not callerid suppression is enabled, I wish to read "CALLER ID SUPPRESSION ENABLED" or CALLER ID SUPPRESSION DISABLED" as such., to the incoming call. |
05:26.12 | random_mike | I'm using 1.4.28 |
05:26.26 | *** join/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
05:27.18 | *** part/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
05:28.10 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-fmthotrzkjqmajmx) |
05:28.16 | dlynes | random_mike, i can't remember...refresh my memory...does 1.4.28 have SIP_HEADER function? |
05:28.22 | dlynes | random_mike, core show function SIP_HEADER |
05:28.51 | random_mike | bf1*CLI> |
05:28.51 | random_mike | <PROTECTED> |
05:28.51 | random_mike | [Syntax] |
05:28.52 | random_mike | SIP_HEADER(<name>[,<number>]) |
05:28.52 | random_mike | [Synopsis] |
05:28.52 | random_mike | Gets the specified SIP header |
05:28.54 | random_mike | [Description] |
05:28.56 | random_mike | Since there are several headers (such as Via) which can occur multiple |
05:28.57 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-zdrnotdnckwzqruo) |
05:28.58 | random_mike | times, SIP_HEADER takes an optional second argument to specify which header with |
05:29.01 | dlynes | random_mike, PID=${SIP_HEADER(Privacy)} |
05:29.02 | random_mike | that name to retrieve. Headers start at offset 1. |
05:29.04 | random_mike | bf1*CLI> |
05:29.14 | dlynes | random_mike, wouldn't that do it? |
05:29.31 | random_mike | Yes, that appears to do what I'm after |
05:29.36 | dlynes | random_mike, also, if you're pasting 3 or more lines, please use pastebin in the future |
05:29.37 | dlynes | ~pb |
05:29.38 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
05:29.38 | random_mike | is that not valid in 1.6.x? |
05:29.45 | dlynes | random_mike, it is |
05:29.58 | dlynes | random_mike, it is valid, that is |
05:30.39 | dlynes | random_mike, so ${PID} will now contain the value of your privacy id |
05:30.51 | *** join/#asterisk voipmonk (n=shido6@CPE00222d72fc72-CM00222d72fc6e.cpe.net.cable.rogers.com) |
05:30.59 | dlynes | random_mike, that being said, i would use a different name than 'PID', because PID usually means process ID |
05:30.59 | random_mike | sorry - have an outage at work |
05:31.00 | random_mike | afk |
05:33.10 | drmessano | You may have more friends on Facebook than I, but what about enemies? |
05:33.32 | drmessano | </thought of the day> |
05:33.49 | *** part/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
05:34.46 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
05:36.20 | random_mike | thanks dlynes |
05:36.21 | random_mike | sorry |
05:36.30 | random_mike | our upstrean VoIP provider just had an outage |
05:36.34 | random_mike | talk about good timing |
05:37.03 | random_mike | so dlynes, in your example: PID=${SIP_HEADER(Privacy)} |
05:37.23 | random_mike | what would happen if the sip header: "Privacy" didnt exist? |
05:38.56 | random_mike | would it be like dividing by 0? |
05:39.41 | random_mike | I cant test this current, as our SIP server which trunks calls to my Asterisk box is not correctly sending the invites. This is under investigation by our vendor atm. |
05:44.14 | random_mike | can anyone assist? |
05:45.16 | Corydon76-dig | random_mike: nope, just blank |
05:45.23 | random_mike | thanks Corydon76-dig |
05:45.33 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
05:45.48 | Corydon76-dig | There should not be anything that you can do in the dialplan to make Asterisk crash, short of consuming all memory |
05:46.20 | random_mike | Yes, but then again, I have all sorts of magical powers! |
05:46.30 | random_mike | Especially crashing things that shouldnt crash :P |
05:46.38 | Corydon76-dig | and there's a few other things, like turning off safety checks in Macro, that enables you to overflow the stack |
05:47.27 | Corydon76-dig | but really, you shouldn't be able to crash Asterisk from the dialplan |
05:47.37 | random_mike | that's reassuring :) |
05:47.52 | *** join/#asterisk sjobeck_ (n=sjobeck@valdisere.sjobeck.com) |
05:48.11 | Corydon76-dig | Dividing by zero might do it, though. Dunno if there's a safety check for that in the expression parser or not |
05:48.47 | *** part/#asterisk sjobeck_ (n=sjobeck@valdisere.sjobeck.com) |
05:49.26 | Corydon76-dig | Yep, there is. Division by zero returns INT_MAX. |
05:50.01 | random_mike | Brilliant! |
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06:47.33 | sun28 | moin \o/ |
06:51.51 | *** join/#asterisk simplydrew___ (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
06:55.27 | ChannelZ | yawns |
06:58.46 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
07:03.58 | *** join/#asterisk CheBuzz_Home (n=CheBuzz_@88-202-45-98.ip.skylogicnet.com) |
07:04.28 | CheBuzz_Home | Is there a way to specify what verbosity asterisk should use instead of issuing "core set verbose 3" every time on the CLI? |
07:05.14 | reptiles | change the default in the source and recompile? |
07:05.19 | reptiles | 8^) |
07:06.00 | ChannelZ | well you can run asterisk initially with many v's if you want it always to be like that |
07:06.05 | *** join/#asterisk e-jones (n=jkastner@nat/redhat/x-ecegbxyfdeofokys) |
07:06.12 | ChannelZ | If you are reconnecting to the console, you can just do it there to: asterisk -rvvvvv |
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07:32.14 | ChannelZ | does the safety dance |
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08:54.46 | Polysics | hello |
08:55.04 | Polysics | how do i know if * is registering correctly with a provider? incoming calls only |
08:56.36 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
08:59.23 | Polysics | please? :-) |
09:01.14 | angryuser_ | Polysics, 'sip show registry" |
09:01.30 | Polysics | nice, thanks |
09:05.37 | Polysics | one more thing... what would be a quick and dirty way to hook up the number to a SIP account that is already registered? |
09:05.47 | Polysics | for an incoming call |
09:05.54 | Polysics | do i need to use the "s" extension? |
09:07.16 | *** join/#asterisk TimeRider (n=steve@nat73.mia.three.co.uk) |
09:11.22 | Polysics | ok, * is registering correctly with te provider, and i have a simple s extension configured, yet calling the number gets me an "user is unreachable" message from the provider |
09:11.31 | Polysics | is there anything i can do in * to diagnose the problem? |
09:13.43 | *** join/#asterisk titter` (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
09:18.09 | Polysics | i can't tell if it is something about * or the provider |
09:19.17 | Polysics | i suppose that since i hear the message from the provider, it's the provider |
09:19.20 | Polysics | any pointers? |
09:25.55 | *** join/#asterisk tigerdyr2 (n=bach@217.195.185.143) |
09:26.13 | tigerdyr2 | Hi all. Nice to be here :-) |
09:32.33 | *** join/#asterisk DelphiWorld (n=Miranda@41.110.0.199) |
09:32.37 | DelphiWorld | hi |
09:32.52 | DelphiWorld | any asterisk business channel? i know that this is a support channel... no business talk here |
09:33.55 | tigerdyr2 | Im looking for a way, to easy disable the MOH function in an asterisk system. All things i could find relates to the speciffic dial command or user. Any hints ? |
09:35.14 | *** join/#asterisk ashd (n=ashleyd@83-148-137-168.class.managedbroadband.co.uk) |
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09:52.59 | klochan | hi, is there some way to use followme app using sip/realtime-mysql? =) |
09:53.15 | fiddur | Hi. I have a problem using my new sip trunk; it's the first time for me where login-name differs from the number I'm calling from. I have set it up with fromuser = 0317195500 and authname = MyLogin, but in the INVITES asterisk sends, the authname isn't used. I've tried with auth = MyLogin@trunkserver but still nothing mentioned of "MyLogin" in the sent INVITE, and the trunk server responds with SIP/2.0 401 Unauthorized... Anyone versed in sip that |
09:55.20 | fiddur | Btw, I'm using 1.6.1.4... |
09:56.41 | *** join/#asterisk TSM (n=the_soft@fw-lon1.wenn.com) |
09:57.05 | TSM | what is the best way to dial people into a conference room |
09:57.27 | TSM | i want someone to call into the system and then go into the room, but they need to be able to pull people into the room |
10:05.02 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
10:08.06 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
10:17.21 | *** join/#asterisk oktay (n=oktay@81.215.202.193) |
10:17.26 | oktay | hi |
10:27.55 | *** part/#asterisk oktay (n=oktay@81.215.202.193) |
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10:37.34 | benngard | what's the default value for "x" in allow=alaw:x statement, i use :40 in my ooh323.conf but cant remember why :( think it was left when may and i did som ooh323 tests... |
10:42.54 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
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10:43.47 | Polysics | hello |
10:43.56 | Polysics | has anyone ever used red5phone with their asterisk? |
10:44.08 | Polysics | i have a weird problem and can't seem to solve it |
10:44.20 | Polysics | calls go through but only stay up for 2 seconds |
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11:39.18 | benngard | damn, i did receive the fax but asterisk crashed :( |
11:40.33 | benngard | best attempt so far |
11:43.20 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
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12:02.59 | benngard | guys, can u give me some help plz... i try to receive fax, (i actually receives the fax) but i stop a step 4 in dialplan, look here: |
12:03.08 | benngard | exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) |
12:03.08 | benngard | exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) |
12:03.08 | benngard | exten => 101,5,System(mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf magnus.b@inputinterior.se < /dev/null) |
12:03.23 | benngard | <PROTECTED> |
12:03.23 | benngard | <PROTECTED> |
12:03.23 | benngard | sip*CLI> |
12:03.39 | benngard | why am i not executing step 5? |
12:04.07 | benngard | if i run the system command from linux prompt, it send the fax to me as an attachment |
12:12.23 | klochan | maybe someone can advise the best way to use followme with realtime sip-users? =) |
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13:03.19 | DelphiWorld | hi, do asterisk support H.248? |
13:12.43 | coppice | the megabucks protocol |
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13:13.06 | DelphiWorld | coppice: ;) |
13:13.56 | DelphiWorld | [TK]D-Fender: hi, do asterisk support h248? |
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13:15.29 | Omorika | hi guys |
13:15.45 | Omorika | is asterisk capable of detecting pressed pulse digits? |
13:15.51 | Omorika | in IVR? :D |
13:16.23 | DelphiWorld | Omorika: you mean dtmf? |
13:16.37 | Omorika | nope |
13:16.43 | Omorika | i mean pulse dialing |
13:16.44 | Omorika | :D |
13:17.03 | coppice | wow, someone still uses that? :-\ |
13:17.11 | Chainsaw | coppice: The red phone in the office does. |
13:17.22 | Chainsaw | coppice: My FXS gateway port is set up for pulse dialling there. Works fine. |
13:17.36 | Chainsaw | coppice: http://www.vroon.org/ringring.jpg |
13:17.41 | [TK]D-Fender | DelphiWorld: What does "core show codecs" or "show translation" tell you? |
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13:17.53 | DelphiWorld | Omorika: lol, got it/................ |
13:18.14 | DelphiWorld | [TK]D-Fender: i can't read it |
13:18.29 | Omorika | so... is there a way to detect if some1 turns the dial? |
13:18.34 | coppice | Chainsaw: that's an old BT phone, isn't it? |
13:18.35 | Omorika | on that phone |
13:18.44 | Chainsaw | coppice: Correct :) |
13:18.47 | Omorika | like "press 0 for menu" |
13:18.51 | Omorika | and u turn 0 :) |
13:19.03 | [TK]D-Fender | DelphiWorld: Because? |
13:19.33 | DelphiWorld | [TK]D-Fender: console is plin of text... and my screen reader can read only 1 line at time and can't navigate troug it |
13:20.40 | coppice | [TK]D-Fender: how would show codecs help him? |
13:21.17 | [TK]D-Fender | Ah, protocol. |
13:21.23 | [TK]D-Fender | DelphiWorld: Sure looks like mgcp to me. |
13:21.45 | [TK]D-Fender | DelphiWorld: For which yes, * does, but only as a central, not as an endpoint |
13:21.59 | coppice | does asterisk support MGCP these days? it used to only support a tiny subset |
13:21.59 | DelphiWorld | [TK]D-Fender: hehehe;) |
13:22.11 | [TK]D-Fender | coppice: Hasn't grown |
13:22.17 | dmast | g'morning |
13:22.42 | DelphiWorld | [TK]D-Fender: i want to use it a a media gateway controler, is it pocible? |
13:22.45 | DelphiWorld | from sip to MGCP |
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13:24.34 | [TK]D-Fender | DelphiWorld: Again depends whats ont he OTHER side of that MGCP connection |
13:26.39 | DelphiWorld | [TK]D-Fender: user dial from sip to asterisk, asterisk pass / contril it troug the media gateway, media gateway to pstn |
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13:40.14 | hluesea | anyone know the hotdesk pins are the sip configured secret ? |
13:40.14 | [TK]D-Fender | grrr |
13:40.15 | [TK]D-Fender | hluesea: huh? |
13:40.17 | hluesea | i am trying to configure hotdesk feautera and i am wondering to hotdesk users pin ? where can i configure the pins for agents ? |
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13:40.17 | [TK]D-Fender | hluesea: What agents? |
13:40.17 | [TK]D-Fender | hluesea: Configured where? |
13:40.28 | DelphiWorld | [TK]D-Fender: http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/VoIP/Q_22882892.html |
13:40.28 | hluesea | my used agents mean phone users using hot desking. |
13:40.28 | DelphiWorld | [TK]D-Fender: is mgcp... no h248 is difere;) |
13:40.28 | hluesea | my real question is where i can configure the hot desking users pin ? |
13:40.28 | [TK]D-Fender | hluesea: What PIN? Where in * do you configure PINs? |
13:40.40 | [TK]D-Fender | hluesea: There is no such thing as "hot-desking". It is a concept you have to implement |
13:41.05 | [TK]D-Fender | hluesea: What you use to auth state-changes for this is upt o you |
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13:43.11 | Polysics | hello |
13:43.19 | Polysics | anyone is using red5phone or some web phone? |
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13:55.25 | benngard | SendFAX(filename[,a]) - Send a given TIFF file to the channel as a FAX. how do u define/set the channel? |
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13:58.42 | [TK]D-Fender | benngard: What channel? |
13:58.59 | [TK]D-Fender | benngard: You're IN ONE when you call it. |
14:00.08 | benngard | oki, think i got u |
14:00.21 | dmast | Are there any good CDR reporting apps for * out there that support MSSQL? |
14:00.49 | Gido-E | mssql /me is going to bark |
14:02.57 | benngard | if i have a fax attached to an ata (with extension 033211101), then exten => 033211001,1,SendFAX(test.tiff) will convert test.tiff to a fax an send it to the ata/fax? |
14:05.07 | Amorsen | benngard: Nope. You need to Dial() first. |
14:05.27 | Amorsen | Or am I confusing myself? |
14:05.32 | Gido-E | yep, or use callfiles. |
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14:05.49 | Amorsen | Sorry, I'm just causing confusion. Forget what I said. |
14:06.53 | [TK]D-Fender | benngard: You don't seem to be thinking about where the call is starting from. |
14:07.19 | Holos | dmast: Are you writing directly into mssql or parsing the logs through an insert script? |
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14:11.31 | dmast | Holos: Writing directly to SQL |
14:13.09 | dmast | I'm not going to die on the mssql hill, either... I would just rather not put up another db server if it can be avoided. |
14:13.38 | Polysics | agh, this web-based phone thing is killing me |
14:13.51 | Polysics | no one is using red5phone or another web-based phone? |
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14:14.11 | [TK]D-Fender | Polysics: Why would we? |
14:14.30 | Polysics | [TK]D-Fender, maybe your boss wants one like mine does :-) |
14:14.37 | [TK]D-Fender | Polysics: Only the smallest subset of * users could really care for such a thing. |
14:14.54 | Polysics | [TK]D-Fender, i was hoping to meet someone from such subset here :-) |
14:14.58 | [TK]D-Fender | Polysics: So whats his reason to? |
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14:16.02 | Polysics | project is to have translators provide services by voice, and the potential callers often have network/system setups where they can't use/install a softphone |
14:16.41 | Polysics | Leif Madsen himself? sign my book :-) |
14:16.49 | leifmadsen | :) |
14:17.03 | Polysics | [TK]D-Fender, and the reasoning was, let's look for a browser-based alternative |
14:17.24 | leifmadsen | leifmadsen: if you check my website, you can select the signature you want. Signatures range from $5-$100 plus shipping and handling. |
14:17.25 | [TK]D-Fender | Polysics: Again, what makes this the preferred solution? |
14:17.39 | [TK]D-Fender | Polysics: Sorry, just got the "why". |
14:17.47 | [TK]D-Fender | Polysics: Good luck with that :) |
14:17.54 | leifmadsen | errr.... s/leifmadsen/Polysics/ |
14:18.29 | coppice | leifmadsen: does that include you providing a cheque from your current account for the signature to go on? |
14:18.39 | Polysics | [TK]D-Fender, i don't know if it is a good "why", but it does not look that bad of an idea |
14:18.49 | Polysics | too bad for it being nigh-unimplementable |
14:18.56 | leifmadsen | coppice: that's why I don't sign cheques, because they tend to end up on eBay |
14:19.08 | Polysics | due to ANY clients i found having some sort of problem |
14:19.48 | [TK]D-Fender | Polysics: have you considered having people CALL in with...oh I dunno.... PHONES? |
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14:20.28 | Polysics | [TK]D-Fender, what you are saying SOUNDS silly, but in fact, it is not... what can i say :-) |
14:20.45 | leifmadsen | what was the original question? |
14:21.06 | [TK]D-Fender | leifmadsen: he's got woes with 2 web-soft-phones |
14:21.12 | Polysics | if anyone knows about a browser-based SIP client that actually works |
14:21.28 | Polysics | possibly open source |
14:21.43 | Polysics | that's the skeleton of the question, then there are some others that are related |
14:22.05 | Polysics | red5phone works... for 2 seconds each call, it's a reported and not solved issue |
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14:22.45 | leifmadsen | Polysics: have you looked at the web phone from Zoiper? |
14:22.56 | leifmadsen | I haven't tried it, but Zoiper in general is a very good softphone |
14:23.04 | leifmadsen | and they have a web phone now too |
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14:23.34 | Polysics | wow |
14:23.41 | Polysics | see? i knew that someone would know |
14:23.50 | Polysics | what i find interesting is the "no interface at all" option |
14:24.01 | Polysics | so i can brand and integrate it whenever |
14:28.26 | russellb | check with phonefromhere.com, too |
14:28.57 | russellb | if SIP isn't a hard requirement, anyway, as I believe their main product is IAX2 based (which is a better choice for a browser client IMO) |
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14:29.21 | leifmadsen | and the Zoiper softphone will likely do IAX2 as well, so now you have 2 options :) |
14:29.45 | russellb | yup, and good people behind both of those options |
14:30.44 | leifmadsen | http://moziax.mozdev.org/ <-- never used, but a third option |
14:30.44 | leifmadsen | damn you net split! |
14:30.44 | russellb | leifmadsen: fail |
14:30.44 | leifmadsen | russellb: your fail |
14:30.44 | leifmadsen | russellb: and your face |
14:30.44 | russellb | your face's fail |
14:30.44 | russellb | damn |
14:30.44 | leifmadsen | HA! |
14:30.44 | leifmadsen | I need a beer |
14:30.44 | russellb | same here |
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14:43.58 | Polysics | i suppose it's because the other option is cumbersome stuff like the Flash thing i've beent testing |
14:43.58 | leifmadsen | ok, so freenode is useless again -- time to go work on something else |
14:43.58 | Polysics | leifmadsen, the fact that i have no idea how to configure IAX2 access and SIP works is a problem :-) |
14:43.59 | leifmadsen | Polysics: you configure IAX2 just like SIP in Asterisk -- and IAX2 only uses a single port |
14:43.59 | leifmadsen | Polysics: but if SIP works, then enjoy :) |
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14:44.49 | [TK]D-Fender | Polysics: Skype web client < - |
14:44.49 | leifmadsen | the options are too numerous! |
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14:44.50 | phasip | I don't think that's what I want, I want something like a java application that translates calls to 222 to 5551123 and then sends the tones 222 so that the asterisk server on 5551123 calls it's internal number 222 |
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14:44.51 | [TK]D-Fender | phasip: huh? |
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14:44.51 | phasip | So that i could use a cellphone as a local extension |
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14:46.58 | phasip | I have a cellphone number 777333, then I set something like GotoIf($[${CALLERID(num)} = 777333]?local) and in local i have something like WaitExten(10), but instead of having to type 777333 on my phone, wait for asterisk to answer and then type 222 i want a java application to do this for me so I only have to type 222 |
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14:47.42 | Polysics | ok, while i mull over the web client options, how does one troubleshoot a SIP provider connection? |
14:47.53 | Polysics | registration is ok (sip show registry shows it) |
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14:48.20 | Polysics | but i get the "customer is unreachable" message from the provider (not from my *) |
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14:48.52 | Polysics | might be a problem with the firewall? |
14:48.58 | [TK]D-Fender | phasip: You are missing part of your description. Just because you say you have a cell phone doesn't tell us where the call STARTS, what actions you want * to take, and what you what it to do from there |
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14:49.35 | [TK]D-Fender | Polysics: You've given no description as to what that is a response TO. |
14:49.58 | Polysics | to dialing the number from a PSTN phone |
14:50.12 | [TK]D-Fender | Go look in SIP debug |
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14:52.05 | phasip | [TK]D-Fender: What i want is a cellphone application that instead of dialing 222 on the phone network it will dial my home number and then make asterisk understand I want to dial a local extension, just as if it was one of the sip phones connected to the internal network. |
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14:52.42 | Polysics | [TK]D-Fender, SIP debug doesn't budge, which i equate to something else not working |
14:53.07 | [TK]D-Fender | phasip: Ok, so far this is nothing VoIP related, or * related. |
14:53.26 | [TK]D-Fender | Polysics: Bad firewall, registeration, etc |
14:53.34 | [TK]D-Fender | ~sipnat |
14:53.35 | 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 |
14:53.36 | [TK]D-Fender | ^^^^^^^^^ |
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14:53.57 | Polysics | chan_sip.c:19043 handle_request_invite: Call from '' to extension '0159526021' rejected because extension not found |
14:54.09 | Polysics | might it simply be a bad extension? |
14:54.31 | [TK]D-Fender | phasip: Just "I want a generic java app for a phone platform I didn't even bother to specify to interpret my dialed numbers and script locally on the phone" |
14:54.46 | Polysics | i have an "s" extension defined, but apparently i get the number |
14:54.47 | *** join/#asterisk vrban (n=chatzill@p50997436.dip0.t-ipconnect.de) |
14:54.49 | [TK]D-Fender | Polysics: How much more does it have to tell you right to your face? :p |
14:54.58 | [TK]D-Fender | Polysics: "s" is not magic. |
14:55.05 | [TK]D-Fender | Polysics: Nor a catch-all |
14:55.07 | *** join/#asterisk TimeRider (n=steve@nat72.mia.three.co.uk) |
14:55.10 | [TK]D-Fender | ~stdextens |
14:55.11 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
14:55.11 | Polysics | neither are my * skills, it seems :-) |
14:55.32 | [TK]D-Fender | Polysics: Sure they are... they've disappeared ;) |
14:55.42 | [TK]D-Fender | pwned |
14:55.49 | Polysics | you win :-) |
14:56.02 | *** part/#asterisk benngard (n=benngard@213.88.138.230) |
14:56.14 | Polysics | but someday you will need something about PHP, Ruby, CSS or JS, then i'll get my revenge .-) |
14:57.00 | phasip | [TK]D-Fender: I am sorry, i don't understand what you ment by that, but yes what I am looking for is a generic java app that works for most java enabled phones. I was hoping that someone here had used something like that since it would be a very powerful thing to combine with asteirsk. I am sorry if I upset you. |
14:57.26 | phasip | asterisk* |
14:57.31 | [TK]D-Fender | phasip: As an attempt to be lazy this one falls short or being worthwhile. |
14:57.53 | Polysics | phasip, i would say that is not much very related to *, but as i said my skills with * aren't that great :-) |
14:58.12 | [TK]D-Fender | phasip: Most phones have 1-though dialing capability to which you could point to a number to reach your *. So basically its an app to remove dialing a single digit (or quick sequence) |
14:58.57 | *** part/#asterisk Andrew_Zimmern (n=dmast@exchange.newpointe.org) |
15:00.05 | *** join/#asterisk dmast (n=dmast@exchange.newpointe.org) |
15:00.11 | phasip | [TK]D-Fender: That's a solution I haven't tought about, I look if it'll work, I hope you don't think I came here before trying to search for all terms and combinations I found relevant to this on google. |
15:00.19 | *** join/#asterisk smooth_penguin (n=smoove@59.95.58.17) |
15:01.56 | [TK]D-Fender | phasip: There is no "term" for this. You gave no indication on the phone platform, made an assertion that java can actually cause the phone to dialout like that. That it would also be a RUNNING app all the time (otherwise what's the convenience?), and provided all this as a BROKEN description. |
15:03.59 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
15:04.16 | *** join/#asterisk fogrider (n=fog@wsip-70-165-16-71.ok.ok.cox.net) |
15:04.20 | [TK]D-Fender | phasip: On my HTC Touch phone I grab my phone, swipe my thumb up, and press on the middle button of a 3x3 grid and it dials my home answering with a menu and awaiting input. Swipe + push. Is that worth replacing with a Java app that I'd have to leave running all the time? What about exiting it so I could do something else? |
15:04.31 | fogrider | anybody here worked with superfecta CID? |
15:04.49 | *** join/#asterisk MAbbas (i=Jinbaba@115.186.22.254) |
15:04.55 | [TK]D-Fender | fogrider: Wrong channel ---> #freepbx |
15:04.57 | [TK]D-Fender | ~freepbx |
15:04.58 | 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, or fpbx |
15:05.12 | fogrider | ok thanks |
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15:05.48 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
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15:07.18 | MAbbas | Hi everyone, my WinSip generated sip calls are disconnected by Asterisk every 20 sec. |
15:07.19 | MAbbas | http://www.voip-info.org/boards/index.php?t=19148 |
15:07.56 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
15:08.26 | [TK]D-Fender | MAbbas: You've provided NO details about where these 2 systems are relative to each other, provided CLI & SIP debug or much else. |
15:09.07 | MAbbas | These two system lie in the same network .. |
15:09.18 | Polysics | tricky but interesting question: is the ringing sound ITSELF part of the Dial() app, or do i need to call something else? |
15:09.24 | [TK]D-Fender | MAbbas: Where is the failed call to look at? |
15:09.39 | [TK]D-Fender | Polysics: Depends |
15:09.48 | MAbbas | let me dump the sip debug .. if its not against forum rules? |
15:09.50 | Polysics | the call from cellphone to * goes through fine, only the caller does not hear any ringing sound till it is answered |
15:09.59 | Polysics | MAbbas, pastebin and we'll love you mroe |
15:10.02 | Polysics | *more |
15:10.10 | [TK]D-Fender | MAbbas: PASTEBIN <_ wtf are are you bringing forums up in here for? |
15:10.22 | [TK]D-Fender | MAbbas: If you want our help just PB stuff like normal |
15:10.57 | [TK]D-Fender | Polysics: I also don't see a call I can help debug. |
15:11.06 | Polysics | there is nothing to debug |
15:11.09 | Polysics | no errors at all |
15:11.13 | [TK]D-Fender | Polysics: And I'm quite certain there is no call going from cellphone to cellphone |
15:11.28 | [TK]D-Fender | Polysics: it has issues. I'd call that something to "debug" |
15:11.30 | MAbbas | I have updated the message with SIP debug, can you chk there? |
15:11.34 | Polysics | [TK]D-Fender, not from cellphone to cellphone |
15:11.48 | Polysics | i am simply dialing a dial-in number i got from my provider |
15:12.00 | Polysics | which is hooked up to a simple Dial() extension |
15:12.12 | Polysics | and EVERYTHING works, but the caller does not hear ringing sound |
15:12.28 | [TK]D-Fender | Polysics: and I still don't see the call. |
15:12.45 | [TK]D-Fender | MAbbas: Retransmitting #3 (no NAT) to 10.110.32.44:5161: <--- packets aren't getting back. check your FIREWALLS |
15:13.19 | MAbbas | <[TK]D-Fender>: I have updated the sip debug in http://www.voip-info.org/boards/index.php?t=19148 |
15:13.57 | [TK]D-Fender | MAbbas: Your other side is likely FIREWALLED |
15:14.28 | MAbbas | WinSip is generating calls on Windows, which is not firewalled .. |
15:18.26 | Polysics | one thing i have always wondered: is there a way to log all and only one call's SIP debug? |
15:18.33 | [TK]D-Fender | MAbbas: Whatever the reason, packets aren't getting answered by their side |
15:18.49 | [TK]D-Fender | Polysics: One call's? No. 1 IP's, sure |
15:19.53 | *** join/#asterisk donnib (n=aaa@0x555281d0.adsl.cybercity.dk) |
15:19.56 | donnib | hi |
15:20.17 | donnib | have a stupid question :) What's the difference between CLI termination and non-CLI termination ? |
15:20.39 | donnib | is it callerid where the call comes from ? |
15:20.58 | donnib | i mean when it's CLI termination the callerid get's sent to the end caller ? |
15:21.31 | [TK]D-Fender | donnib: that makes no sense. |
15:21.46 | *** join/#asterisk Akiraaa (n=Akiraaaa@79.112.27.76) |
15:21.47 | donnib | [TK]D-Fender: you tell me :) |
15:22.00 | [TK]D-Fender | donnib: No, you come up with a better phrased question. |
15:22.26 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:1079:1e49:a077:acde) |
15:22.38 | donnib | i just looked up some prices for VoIP termination in Romania and the pricelist has two difference prices. One CLI Terminaion and one non-CLI so i am asking what's the difference ? |
15:22.52 | MAbbas | <[TK]D-Fender>: I have tried without WinSip, using SipPhone and its works fine .. "BTW same environment" |
15:23.22 | donnib | [TK]D-Fender: http://bit.ly/7B3O7L |
15:23.24 | cusco | hi |
15:23.37 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
15:24.01 | [TK]D-Fender | MAbbas: Maybe their software is broken. Either way, its not * |
15:24.36 | MAbbas | <[TK]D-Fender>: Any other tools you might have in mind that will do bulk call generation like WinSip? |
15:25.12 | Polysics | ok, i was just missing the "r" option |
15:25.21 | [TK]D-Fender | donnib: Maybe you should ask THEM what their wording means. |
15:25.57 | donnib | [TK]D-Fender: ok i just thought that this is something i should have known before i talked with them. i thought this was a normal term. |
15:26.32 | cusco | We hae a prefix for outbounds like: _7878XXXXXXXXX and that Dial(DAHDI/g1/${EXTEN:4}); |
15:26.53 | [TK]D-Fender | donnib: Acronyms thrown around here tend to be ENGLISH-origin ones. Coming frmo a clearly non-english language page it could mean ANYTHING, and right now doesn't seem to say anything |
15:27.06 | cusco | all good, now I would like to add a 78782XXXXXXXX because most landline numbers start with 2 |
15:27.17 | cusco | tho there are some that strt with 3 |
15:27.33 | cusco | should I add 78783XXXXXXXX and 78782XXXXXXXX separatelly? |
15:27.50 | cusco | or can I do it any other way? |
15:28.05 | [TK]D-Fender | cusco: Whats wrong with the X instead? |
15:28.23 | donnib | [TK]D-Fender: ok thanx |
15:28.25 | cusco | [TK]D-Fender: they can dial mobile numbers |
15:28.31 | cusco | and we want to prevent that |
15:28.41 | cusco | mobilenumbers start with a 9 |
15:28.44 | [TK]D-Fender | cusco: 78782 + 8 X ---vs---- 7878 + 9X |
15:28.52 | [TK]D-Fender | cusco: What is the processing difference? |
15:29.25 | keith4 | so... polycom phones come with wall brackets. right? |
15:29.31 | cusco | 78782 the number start with a 2 + 8 digits |
15:29.40 | Naikrovek | keith4: yes |
15:29.46 | cusco | all phone numbers have 9 digits in here |
15:29.51 | cusco | including mobile |
15:29.54 | Naikrovek | my 321s did, anyway |
15:30.02 | cusco | fixed numbers start either with a 2 or with a 3 |
15:30.02 | keith4 | and they're designed to hang on those faceplates with the studs for standard phones, right? |
15:30.42 | cusco | so if operator dials on his softphone 78789XXXXXXXX it won't dial... or somehting |
15:30.43 | keith4 | but... i can't find anyone that sells keystone faceplates with phone studs |
15:30.55 | [TK]D-Fender | keith4: many are inhreently wall-mountable |
15:31.01 | cusco | else I need an IF evaluating if the number start with a 9 |
15:31.17 | cusco | I thought there would be a way to do 78782-3XXXXXXXX |
15:31.19 | cusco | or soemthing |
15:31.32 | [TK]D-Fender | cusco: there is.... go read your chaper on dialplan patterns again |
15:31.36 | keith4 | [TK]D-Fender: yah, looks like the 320 is. as is its replacement |
15:32.13 | [TK]D-Fender | cusco: exten => _78789[23]XXXXXXXX,1,Dial(BLAH) |
15:32.23 | cusco | thanks |
15:32.53 | keith4 | [TK]D-Fender: but it seems like nobody makes a faceplate with phone studs that takes a regular keystone |
15:33.02 | keith4 | so what good is the wall bracket? |
15:33.28 | [TK]D-Fender | keyPardon? I'd wall-mounted IP 301's, 430's, etc |
15:33.35 | [TK]D-Fender | keith4: Pardon? I'd wall-mounted IP 301's, 430's, etc |
15:33.57 | keith4 | right. i want to mount them on the wall boxes |
15:34.01 | MAbbas | Any tools to generate SIP calls other than WinSip? |
15:34.04 | [TK]D-Fender | keith4: keith4 and I have faceplates with them. |
15:34.12 | keith4 | [TK]D-Fender: where did you get them? |
15:34.27 | [TK]D-Fender | keith4: Local interconnector who did this place's wiring |
15:34.29 | Naikrovek | http://qdb.us/301260 |
15:34.34 | [TK]D-Fender | keith4: this is common stuff... |
15:34.42 | Naikrovek | yeah |
15:34.49 | keith4 | [TK]D-Fender: that's what i thought! but none of the usual places that i get stuff from have them |
15:34.51 | [TK]D-Fender | Naikrovek: 403 |
15:34.56 | keith4 | [TK]D-Fender: something like this? http://www.twacomm.com/catalog/dept_id_1138/model_IC107FWP.htm |
15:35.03 | [TK]D-Fender | keith4: Maybe your usualy places suck :p |
15:35.23 | [TK]D-Fender | keith4: Looks like. |
15:35.34 | keith4 | [TK]D-Fender: but $2.77 each is armed robbery |
15:36.56 | [TK]D-Fender | keith4: http://tinyurl.com/sulfm |
15:37.05 | keith4 | nice |
15:37.13 | [TK]D-Fender | :D |
15:37.20 | keith4 | for "common stuff", even google can't find 'em |
15:38.12 | keith4 | ... and the Polycom documentation suggests wall anchors for wall mounting |
15:41.32 | phasip | exit |
15:41.52 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
15:42.24 | [TK]D-Fender | keith4: http://www.tigerdirect.ca/applications/SearchTools/item-details.asp?EdpNo=348893&CatId=745 |
15:42.36 | [TK]D-Fender | keith4: No, $100 is robbery. $2.77 looks like a BARGAIN |
15:43.03 | keith4 | [TK]D-Fender: how would I wall mount on that plate? |
15:43.42 | keith4 | http://www.monoprice.com/products/product.asp?c_id=105&cp_id=10517&cs_id=1051703&p_id=6263&seq=1&format=2 |
15:43.51 | keith4 | I would say.... $0.40 is a bargain |
15:44.12 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
15:44.14 | keith4 | but no studs |
15:44.23 | keith4 | or maybe I just need special "stud screws"? hmmm |
15:44.40 | Naikrovek | you mount a polycom the same way you mount a traditional analog phone |
15:44.55 | Naikrovek | to go lowe's, find the mounting plate, install it, plop the phone on there |
15:45.00 | Naikrovek | done |
15:45.12 | keith4 | Naikrovek: right. but Lowe's doesn't have keystone mounting plates with phone studs |
15:45.32 | Naikrovek | with phone studs? yes they do |
15:45.39 | Naikrovek | i don't know wtf you mean by keystone |
15:45.48 | keith4 | they have little kits that have RJ11 jacks, and face plates that screw to them |
15:46.32 | keith4 | keystone: http://images.google.com/images?q=keystone jack |
15:46.51 | Naikrovek | my lowe's has wall-mount phone plates, with studs, with holes for rj45 jacks and the metal studs that phones hang on |
15:47.28 | keith4 | yah? where is it? |
15:47.34 | Naikrovek | and they have the jacks, too |
15:47.41 | Naikrovek | east peoria, il |
15:47.48 | keith4 | ooh, that's a bit far |
15:47.53 | Naikrovek | yeah i figured |
15:48.07 | keith4 | man. their website's search function is absolutely worthless |
15:48.17 | Naikrovek | you should be able to call any union electrician and find out where they get their stuff. they do wiring like that for new offices, electricians do |
15:48.20 | Naikrovek | ours do anyway |
15:50.39 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
15:51.24 | keith4 | what the hell? CDW seems to have them... for $24 each? |
15:51.30 | keith4 | http://www.cdw.com/shop/products/default.aspx?edc=1525596 |
15:51.36 | keith4 | that'd better be a 10 pack |
15:51.38 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
15:52.03 | high-freq | has anyone successfully used fonosip to ipkall to your pbx and able to receive calls? |
15:52.37 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
15:55.52 | *** join/#asterisk mbranca (n=matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
15:58.06 | *** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil) |
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15:58.38 | [TK]D-Fender | high-freq: Whats the point of having them together? |
15:59.05 | high-freq | sip connection to ipkall setup then ipkall gives number to use |
15:59.24 | high-freq | unless if i'm doing somethin wrong here |
16:00.06 | [TK]D-Fender | high-freq: No, these are 2 separate services. Why are you mentionging them TOGETHER? |
16:00.24 | [TK]D-Fender | [10:59]<high-freq>sip connection to ipkall setup then ipkall gives number to use <- this makes no mention of fonosip |
16:00.53 | high-freq | oooook..um ipkall website said to create a sip account then use that sip account registering it with ipkall |
16:01.08 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
16:01.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
16:01.10 | [TK]D-Fender | high-freq: No need. |
16:01.18 | high-freq | kk so just use ipkall settings then |
16:01.34 | Polysics | is there such a feature as "emergency call" that would disconnect a busy SIP account and connect it to the entrant call? |
16:01.54 | high-freq | i'll try that then..hell i'm trying everything over here ;) |
16:02.18 | Polysics | ie. my 1000 user is talking, a call comes that has the precedence, he gets disconnected from what he's doing and connected to that call - with or without warnings |
16:04.02 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
16:06.00 | high-freq | nope i don't understand. ipkall setup is straight forward, has to have sip phone number, sip proxy, setup via once logged in. This service uses SIP config which then ipkall gives out a free number for others to call so using this one service is not gonna work |
16:09.53 | high-freq | omg if i had hair i'd pull it out already lol this just aint right |
16:10.02 | *** join/#asterisk dahl (i=dahl@4.84-234-222.customer.lyse.net) |
16:10.05 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
16:14.47 | [TK]D-Fender | high-freq: number = exten on your system. proxy = IP or hostname for your server |
16:14.56 | [TK]D-Fender | high-freq: this ain't Raw-Cat Sigh Hence |
16:15.17 | high-freq | k i'll try that |
16:16.10 | dahl | is it possible to check if outgoing calls can be made every 5 minutes or so with a simple cronjob, just to see that things work properly? |
16:16.27 | [TK]D-Fender | dahl: Sure |
16:16.39 | *** join/#asterisk jo8330 (i=d04149c9@gateway/web/freenode/x-xmhqymmgiokbigzp) |
16:16.45 | jo8330 | hello! |
16:17.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
16:20.33 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
16:25.00 | high-freq | [TK]D-Fender, for the proxy do i put my external ip because i'm behind a vlan i setup and putting that ip in doesn't seem right |
16:26.49 | chilicuil | Hi there, I'm totally new to asterisk and I'd like to know what're the softphone you're currently using, right now I've tried ekiga, SJPhone and Xlite but I'd know the opinion of the experts |
16:27.29 | Polysics | Xlite is good |
16:27.33 | [TK]D-Fender | high-freq: Of course your external IP. |
16:27.57 | [TK]D-Fender | Chithey all suck (as all softphones do). Some suck a bit less than others |
16:29.03 | high-freq | k |
16:29.34 | UQlev | chilicuil: depends on protocols you want use SIP or IAX |
16:29.57 | chilicuil | I'm using currently only SIP, UQlev |
16:30.37 | *** join/#asterisk bmoraca_work (n=bmoraca@66-242-174-254.ceres.bvn.net) |
16:30.45 | UQlev | chilicuil: for SIP you may have a way more options |
16:32.35 | UQlev | chilicuil: twinkle (for Linux) and Phonelite (for Win) |
16:32.46 | *** join/#asterisk youngproguru (n=youngpro@74.10.229.45) |
16:32.50 | UQlev | Phonerlite |
16:32.53 | *** join/#asterisk markwaters (n=mark@weloveit.info) |
16:33.45 | chilicuil | Ok, I'm gonna give them a try, thx UQlev |
16:36.01 | *** join/#asterisk manxpower (n=ewieling@226.sub-70-214-193.myvzw.com) |
16:38.49 | *** join/#asterisk mrchrisadams (n=mrchrisa@87-194-125-43.bethere.co.uk) |
16:40.01 | mrchrisadams | hi guys - where would I find stats on how popular the various commercial and noncommercial asterisk based distros like SwitchVox, FreePBX, Trixbox et al are in relation to each other? |
16:40.31 | bmoraca_work | you probably wont |
16:40.48 | bmoraca_work | is this for a study or is it because you're interested in getting in to asterisk? |
16:40.53 | manxpower | They will all claim they have the largest "market" share, |
16:41.17 | manxpower | mrchrisadams: just remember we don't support Asterisk distros or Asterisk GUIs here. |
16:41.19 | mrchrisadams | bmoraca_work: I've been tasked with sorting out a decent phone setup for the company I work for |
16:41.39 | p3nguin | mrchrisadams: Don't forget non-asterisk ones, such as FreeSWITCH. |
16:41.48 | mrchrisadams | we're using a badly configured setup of elastix right now |
16:42.06 | bmoraca_work | mrchrisadams, that's all well and good. asterisk is a very capable platform. i would recommend vanilla asterisk. |
16:42.12 | mrchrisadams | p3nguin - freewitch does interesting |
16:42.28 | bmoraca_work | mrchrisadams, you're going to be sadly disappointed if you think that switching platforms is going to solve your planning/configuration woes |
16:42.29 | mrchrisadams | but I'm nowhere familiar enough with voip to be able to dive into it |
16:43.39 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
16:43.41 | manxpower | mrchrisadams: All Asterisk GUIs are "badly configured" No way to avoid that when you let something else write your dialplan and configs |
16:43.54 | Nugget | The best asterisk gui is gvim. |
16:44.10 | mrchrisadams | Nugget: try telling the receptionist that! |
16:44.37 | p3nguin | She shouldn't be writing dialplans in the first place. |
16:45.31 | Nugget | or even adding phones or extensions or anything like that. |
16:45.34 | dmast | If she is, she needs more to do. |
16:46.47 | *** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it) |
16:47.04 | bmoraca_work | there's nothing wrong with letting users modify their own find-me-follow-me settings or call forward or things like that...but they only need a very minimal GUI (or even no GUI at all) to do that |
16:48.26 | mrchrisadams | bmoraca_work: that's more in line with the use case I was thinking about |
16:49.19 | *** part/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
16:51.32 | Polysics | hmm |
16:51.48 | Polysics | what if a user is behind a firewall that does not let him connect on port 5060? |
16:51.56 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
16:51.57 | Polysics | i mean, he has outbound connnections blocked |
16:52.05 | Polysics | nat=yes isn't enough, i'd say |
16:52.20 | p3nguin | Then you had better look into a VPN or something where he CAN connect out. |
16:52.42 | p3nguin | Is he allowed any ports at all? |
16:52.48 | p3nguin | You can change which port to use. |
16:53.37 | Polysics | basically only port 80 and 113 |
16:53.43 | Polysics | Skype somehow works |
16:53.55 | Polysics | i think i will have to look into a STUN proxy, do I? |
16:55.12 | Chainsaw | Polysics: STUN would be your best chance. But if the outbound firewalling is too aggressive it might be intermittent it or might not work at all. |
16:55.50 | Polysics | in case I decide to investigate STUN, what is the nmost widely used server? |
16:55.54 | manxpower | Asterisk is a PBX, not a firewall busting voice system |
16:55.56 | *** join/#asterisk fenrus (i=fenrus@oklart.com) |
16:55.59 | fenrus | o/ |
16:57.14 | Chainsaw | Polysics: I believe ekiga has a STUN server or two. I remember stun.xten.com working for us. |
16:57.21 | voipmonk | add a Damnit Jim in front, manxpower |
16:57.30 | Polysics | so you mean i can use someone else's stun server? |
16:57.40 | Polysics | i thought i had to install one :-) |
16:57.42 | Chainsaw | Polysics: I've tried hosting my own, I could never get it to work. |
16:57.59 | Chainsaw | Polysics: There's only one public codebase and it seemed... rather abandoned. |
16:58.40 | high-freq | [TK]D-Fender, thx for help, managed to make it ring lol but i picked up and it quit on me haha...just another thing to work on |
16:59.09 | *** join/#asterisk daemon (n=paul@cpc1-linc11-2-0-cust594.12-1.cable.virginmedia.com) |
16:59.14 | *** join/#asterisk mrchrisadams (n=mrchrisa@78-105-1-158.zone3.bethere.co.uk) |
16:59.22 | daemon | hey all im having a bit of a problem with asterisk, it seems that it is getting registration time outs from my provider |
16:59.32 | daemon | but I cannot see anything wrong with my setup |
16:59.39 | daemon | I verified the username and password etc |
16:59.41 | daemon | and im not behind nat |
16:59.49 | daemon | apparently |
17:00.00 | daemon | [Jan 21 16:59:56] NOTICE[54443]: chan_sip.c:12997 handle_response_register: Failed to authenticate on REGISTER to '__SOMEUSER__@voiptalk.org' (Tries 3) |
17:00.19 | daemon | followed by: [Jan 21 17:00:16] NOTICE[54443]: chan_sip.c:7849 sip_reg_timeout: -- Registration for 'USER@voiptalk.org' timed out, trying again (Attempt #24) |
17:03.40 | manxpower | daemon: set srvlookup=no in sip.conf [general] and "module reload chan_sip.so" |
17:03.49 | *** join/#asterisk k4tana777 (n=roni@190.196.71.206) |
17:04.04 | k4tana777 | hi everybody ! |
17:04.09 | daemon | manxpower, thanks ill give it a whirl |
17:04.24 | p3nguin | "sip reload" isn't enough? |
17:04.46 | manxpower | Polysics: a STUN server must be OUTSIDE the network |
17:04.49 | manxpower | p3nguin: it should be |
17:04.51 | Kobaz | aww |
17:04.54 | Kobaz | katty's not here |
17:04.56 | Kobaz | http://www.whowantsabalti.com/funpics/mild/squirrel-surprise.jpg |
17:05.04 | p3nguin | Just wondered if there was a difference. |
17:05.06 | Naikrovek | lol |
17:05.34 | daemon | manxpower, ah ha now im getting failure to authenticate no timeout though |
17:05.43 | daemon | strange though I know im using the right username and password |
17:05.43 | daemon | hmm |
17:06.09 | k4tana777 | somebody can give me a little help ? i need that a call, rings in all extensions that i have here .. |
17:06.28 | manxpower | daemon: srvlookup=yes is the default and your provider must have screwed up their SRV records. |
17:06.40 | k4tana777 | i now that many of you gonna send me to google :) , but i just wanna know how can i look for that |
17:06.52 | manxpower | k4tana777: Dial(SIP/123&SIP/124&SIP/125) |
17:07.03 | k4tana777 | really ? |
17:07.04 | manxpower | you don't dial extensions, you dial DEVICES |
17:07.06 | daemon | manxpower, actually just got reg timeout |
17:07.10 | k4tana777 | so simple ? |
17:07.21 | manxpower | k4tana777: read the Asterisk book |
17:07.22 | manxpower | ~book |
17:07.23 | 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 |
17:07.25 | k4tana777 | damn .. i gonna try that .. |
17:07.29 | k4tana777 | thanks |
17:08.05 | manxpower | daemon: timeout means "no response from server, not even a "sod off" message" |
17:08.23 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
17:08.36 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
17:08.36 | manxpower | Which means 1) provider is dead 2) your firewall is blocking 3) you put in the wrong server host/IP |
17:08.50 | daemon | manxpower, logged into it via telnet not a single thing in the way |
17:08.50 | Nugget | telnet is eeeeeeevil! |
17:09.01 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
17:10.18 | manxpower | daemon: you can't connect to a sip server over telnet |
17:10.34 | Kobaz | nc -u host 5060 |
17:10.36 | daemon | manxpower, my fault I meant I could connect to their http port over telnet, so the host is up |
17:10.46 | manxpower | SIP (normally) uses UDP. Telnet uses TCP. You did not actually test ANYTHING when you try telnetting to a SIP port |
17:10.46 | daemon | I dont have anything to conenct via udp, unless I quickly hack something together in perl |
17:10.54 | Kobaz | sip is generally udp, but sip tcp is coming along in asterisk slowly i heat |
17:11.00 | Kobaz | daemon: netcat |
17:11.06 | Kobaz | heat/hear |
17:11.10 | Naikrovek | sip over tcp is in 1.6 and up |
17:11.20 | Naikrovek | 1.6.0 i should say |
17:11.23 | manxpower | daemon: remember ITSP provided configs are almost ALWAYS wrong. |
17:11.25 | daemon | cool ill try that |
17:11.46 | Kobaz | heh |
17:12.07 | daemon | manxpower, yeah I tried there config, it did not work so I installed the ajax api |
17:12.18 | daemon | adderd it in my self, getting same problem though |
17:13.23 | daemon | fixed it |
17:13.30 | *** join/#asterisk comradeb14ck (n=comradeb@72.37.252.50) |
17:13.34 | manxpower | what the hell does an ajax API have to do with registration? |
17:13.49 | Naikrovek | wow ajax |
17:13.54 | daemon | it suddenly occured to me, that if I connect to my mail server I use mail/pop/smtp . host, so I wondered why they was using 'voiptalk.org' as the host |
17:14.00 | daemon | I changed it to sip.voiptalk.org and wallah |
17:14.01 | daemon | it works |
17:14.06 | daemon | I should really email them about that |
17:14.09 | Naikrovek | s/wallah/viola/ |
17:14.21 | comradeb14ck | sup |
17:14.32 | Qwell | Naikrovek: viola is a musical instrument |
17:14.38 | Naikrovek | it is |
17:15.02 | Naikrovek | it's also how you spell the word that sounds like "wallah" |
17:15.13 | Qwell | No it's not |
17:15.18 | Naikrovek | then tell us |
17:15.20 | Naikrovek | how do you spell it |
17:15.23 | Qwell | voilà|
17:15.35 | Naikrovek | well aren't you just the biggest nitpicker of all |
17:15.47 | Naikrovek | my keyboard doesn't have that, and à doesn't work |
17:15.49 | Qwell | remove the accented a. It's still different. |
17:15.53 | beek | applauds Qwell's fast fingers. |
17:16.07 | Naikrovek | or á |
17:16.15 | Qwell | oi != io |
17:16.26 | Naikrovek | fine |
17:16.30 | k4tana777 | <manxpower> thanks man ... works perfect... and was so easy .. |
17:16.35 | Naikrovek | you could just correct me instead of being a dick |
17:16.44 | Qwell | I did correct you. |
17:16.46 | Naikrovek | but the @ infront of your name apparently gives you any rights you want |
17:16.55 | *** mode/#asterisk [-o Qwell] by Qwell |
17:16.56 | Qwell | clearly |
17:17.05 | *** join/#asterisk mrchrisadams_ (n=mrchrisa@78-105-1-158.zone3.bethere.co.uk) |
17:17.08 | Qwell | no @, and I'm still merely correcting you. |
17:17.13 | Naikrovek | okay |
17:17.28 | Naikrovek | everyone is so fucking fast to correct me |
17:17.33 | manxpower | Naikrovek: You are correct. @ means he can do anything he wants. Deal with it. |
17:17.37 | Naikrovek | but never anyone else |
17:18.38 | manxpower | Naikrovek: many people here have the ability to get ops (@) any time they want, Most of them don't unless there is a need. |
17:23.32 | Qwell | Naikrovek: I seem to recall you correcting somebody (incorrectly..). Saying I'm fast to correct you isn't really going to fly. |
17:23.59 | Qwell | and seriously, you're getting upset over this? |
17:30.23 | comradeb14ck | drama llamas |
17:30.25 | comradeb14ck | :s |
17:34.00 | *** join/#asterisk Yedidya (n=chatzill@host86-142-22-34.range86-142.btcentralplus.com) |
17:34.38 | drmessano | I dare say he's the one showing bias against someone for being an OP |
17:34.51 | drmessano | Like Qwell can't correct someone or have an opinion because of the @? |
17:34.54 | drmessano | Hello? |
17:35.00 | hardwire | this thing isn't on. |
17:35.04 | hardwire | f.y.i. |
17:37.49 | *** join/#asterisk securevoip (n=securevo@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
17:41.00 | EmleyMoor | has finally added a feature his partner has been asking for to the dialplan, without compromising existing facilities |
17:41.38 | *** join/#asterisk carrar (i=tim@osburn.com) |
17:42.47 | comradeb14ck | Question: How many good asterisk developers do you think are out there in the wild? |
17:43.05 | comradeb14ck | And by good I mean someone who knows and can write dialplan, asterisk modules, agi, and ami code. |
17:43.13 | Qwell | That isn't an "Asterisk developer" |
17:43.19 | Naikrovek | right |
17:43.21 | Naikrovek | but there are a lot |
17:43.28 | Naikrovek | a lot more than you'll find in here i'd think |
17:43.35 | comradeb14ck | I don't really see many. |
17:43.39 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
17:43.42 | Qwell | you haven't looked |
17:43.43 | Naikrovek | doesn't mean they're not there |
17:43.45 | comradeb14ck | I see a lot of people who work with Asterisk, but who aren't very good at it. |
17:43.46 | carrar | because they are too busy to be in here |
17:43.47 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
17:44.06 | Naikrovek | the really good people come in here once in a while with a specific, well worded question |
17:44.27 | Naikrovek | one guy came in a couple weeks back, had an asterisk system with 20,000 end points and 5,000 simultaneous calls |
17:45.07 | Naikrovek | i asked him about all that then he left, and hasn't been back |
17:45.21 | Qwell | comradeb14ck: "I've been searching around this rehab clinic for days, and I haven't found any upstanding citizens!" It's a place people (usually..) only go when they have problems. |
17:45.45 | Naikrovek | if you hang out in here long enough you'll find many very very smart people running very very large, complex systems |
17:45.45 | drmessano | Asterisk and or mental problems |
17:45.46 | Qwell | I suspect you aren't on any of the Asterisk-related mailing lists |
17:45.46 | comradeb14ck | Qwell, I didn't ask how many were in here. My question was: "How many good asterisk developers do you think are out there in the wild?" |
17:45.58 | Naikrovek | comradeb14ck: many |
17:45.58 | comradeb14ck | As in, everywhere. |
17:45.59 | drmessano | 3,500 |
17:46.09 | comradeb14ck | Just an opinion question =p |
17:46.11 | Naikrovek | a number? ... dunno. but a lot |
17:46.13 | Qwell | My point is - you're looking in the wrong place. |
17:46.27 | comradeb14ck | I'm not looking anywhere, I asked for an opinion. |
17:46.31 | Qwell | whether it be here on IRC, trixbox forums, etc |
17:46.35 | comradeb14ck | It's just for my own curiosity. |
17:46.39 | Qwell | You said "I don't see a lot" |
17:46.54 | drmessano | Give him a frickin number, @master |
17:46.56 | comradeb14ck | Right, that's why I was asking you! |
17:47.01 | drmessano | 6,786 |
17:47.07 | comradeb14ck | I figured some people in here may have seen more than myself =p |
17:47.15 | Qwell | comradeb14ck: it's unknowable |
17:47.35 | comradeb14ck | Is it less than a million? |
17:47.40 | Naikrovek | probably |
17:47.45 | comradeb14ck | Is it less than 999,999? |
17:47.45 | drmessano | Its less than 6 billion |
17:47.51 | seanbright | just shy of |
17:48.05 | comradeb14ck | fair enough :) |
17:48.06 | drmessano | Id say 50,000, +/- 50,000 |
17:48.10 | Naikrovek | heh |
17:48.17 | Naikrovek | they're everywhere |
17:48.24 | seanbright | just pedophiles |
17:48.30 | comradeb14ck | it is always possible to approximate something. Regardless of whether or not it is correct =/ |
17:48.30 | seanbright | like pedophiles |
17:48.41 | seanbright | 'just' = 'like' in na'vi |
17:48.44 | comradeb14ck | It's like getting the limits in calculus =p |
17:48.50 | drmessano | comradeb14ck: Thats not approximating, thats inventing |
17:49.11 | seanbright | 70% of statistics are made up |
17:49.16 | seanbright | 35% of people know that |
17:51.55 | *** join/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
17:51.55 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
17:53.18 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
17:54.58 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
17:59.35 | *** join/#asterisk stones_ (i=stones@linuxpro.org) |
18:00.24 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
18:09.22 | *** mode/#asterisk [+o Qwell] by ChanServ |
18:09.56 | FinboySlick | Anyone has a good example for calling dialplan apps from a shell using asterisk -rx ? |
18:10.30 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
18:10.44 | [TK]D-Fender | FinboySlick: There is no such thing |
18:11.30 | FinboySlick | [TK]D-Fender: That settles it. What should I look into as an alternative? AGI? |
18:12.05 | seanbright | asterisk -rx "originate SIP/1234 application MeetMe 1234,d" |
18:13.12 | FinboySlick | seanbright: Interesting, that should be able to do just what I want. |
18:13.42 | seanbright | that will call SIP/1234 and when it is answered send it into MeetMe |
18:14.40 | [TK]D-Fender | FinboySlick: make an extensions that answers, and plays silence for a while like 5 min or something |
18:14.47 | [TK]D-Fender | FinboySlick: And originate against it |
18:14.58 | [TK]D-Fender | FinboySlick: Thats as close as you'll get |
18:15.09 | *** join/#asterisk benngard (n=benngard@213.88.138.230) |
18:17.43 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
18:17.52 | carrar | TK got SERVED! |
18:17.58 | carrar | :) |
18:17.59 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
18:19.02 | carrar | scrambles away from the cluebat |
18:19.54 | benngard | #"&¤(/"&¤/&"(/ |
18:19.58 | benngard | sorry |
18:20.08 | bmoraca_work | om nom nom...i ate your random characters |
18:20.58 | benngard | i hve been doing some tests with latests dahdi, spandsp and asterisk trunk |
18:21.41 | benngard | i can get fax to work, but not with the same settings i need for outgoing calls |
18:22.01 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
18:22.02 | benngard | so i have to choice, either fax or voice :( |
18:24.40 | stones_ | Anyone here needing a dedicated server? |
18:25.14 | Qwell | stones_: Please do not advertise here |
18:25.37 | FinboySlick | [TK]D-Fender: Given the lack of 'obvious' way to do what I'm trying to do, I'm tempted to consider that I'm not taking the right approach to my problem. I'm trying to setup something that re-sets all the right agents into all the right queues every morning at 8am. |
18:25.48 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
18:26.08 | FinboySlick | [TK]D-Fender: What would you suggest I look into for something like that? |
18:27.09 | FinboySlick | I was figuring cronjob calling AddQueueMember() apps |
18:27.26 | seanbright | asterisk -rx "queue remove member SIP/1157 from queue" |
18:27.30 | seanbright | heh |
18:27.40 | *** join/#asterisk korcan (n=kshamoun@ip65-44-169-89.z169-44-65.customer.algx.net) |
18:27.58 | *** join/#asterisk areay (n=areay@188-220-19-194.zone11.bethere.co.uk) |
18:28.42 | titter` | Say you have _70XX in your dial plan doing one thing, but you need say _700[0-5] do something else ... what would be the best way to write that? |
18:28.52 | Qwell | titter`: just like that |
18:28.56 | FinboySlick | seanbright: I'm starting to like you. |
18:29.04 | *** join/#asterisk ruied (n=ruied@92.250.95.7) |
18:29.13 | *** join/#asterisk toddejohnson (n=toddejoh@ppp-70-226-205-249.dsl.spfdil.ameritech.net) |
18:29.15 | titter` | When I dialout with that, it dials both since they both are in the same context |
18:29.18 | [TK]D-Fender | FinboySlick: Generate the dialplan to do all actions. Issues a dialplan reload, then a single originate, or call-file to trigger the batch |
18:29.27 | ruben23 | hi how do i setup h323 on asterisk, im currently using SIP. |
18:29.37 | [TK]D-Fender | ruben23: Go read the sample configs |
18:29.53 | carrar | titter: _70X[6-9] |
18:30.20 | carrar | err |
18:30.21 | carrar | no |
18:30.22 | carrar | haha |
18:30.24 | titter` | lol |
18:30.27 | titter` | bout to say |
18:30.39 | carrar | must find coffee |
18:31.34 | carrar | just create 5 extension 7000 - 7005 |
18:31.39 | carrar | they will match first |
18:31.44 | carrar | err 6 extensions |
18:31.54 | Qwell | so would _700[0-5] |
18:32.06 | *** join/#asterisk theron (n=theron@ip244.scolloc.lh.net) |
18:32.13 | titter` | yup |
18:32.27 | titter` | it shows first, but because they are in the same context, it dials both |
18:32.35 | Qwell | it doesn't dial both... |
18:32.38 | [TK]D-Fender | titter`: DIALS BOTH? PARDON? |
18:32.51 | [TK]D-Fender | titter`: PAStebiN the failed call and your dialplan |
18:35.53 | *** join/#asterisk luckyaba (n=lucky@ip72-194-218-169.sb.sd.cox.net) |
18:36.18 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
18:36.34 | titter` | dp - http://pastebin.com/d67447949 cli - http://pastebin.com/d2d99efb7 - dp show in cli - http://pastebin.com/m18de54fb |
18:36.42 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
18:37.28 | titter` | nvm, im an idiot ... i just realized what it did |
18:37.57 | [TK]D-Fender | titter`: Yeah... COULD match... not WOULD. |
18:38.03 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
18:38.07 | Katty | herroes. |
18:38.10 | [TK]D-Fender | titter`: And clearly isn't executing anything extra |
18:38.13 | [TK]D-Fender | kettMew. |
18:38.26 | [TK]D-Fender | Katty: Even... |
18:38.27 | Katty | tab tail. |
18:38.29 | Katty | ... |
18:38.34 | Katty | i fail, too. |
18:38.42 | *** join/#asterisk Alagar (n=Administ@122.164.18.195) |
18:38.43 | titter` | the firt two times i saw it in the cli someone else made a call in the 700x range, so i was like wtf is it doing ... it's been a long night. |
18:38.56 | titter` | see it's and its ... |
18:39.49 | p3nguin | looks like you have correctly used "it's" in your sentence. |
18:41.19 | titter` | ROFL |
18:41.33 | p3nguin | doesn't get it |
18:41.34 | titter` | I give up. |
18:42.30 | titter` | needs a nap |
18:42.55 | Katty | applies pillow and blanket to titter` |
18:43.15 | benngard | guys/girls i ran into a strange problem today: was playing around with trukn and and faxing: i hade i dialplan like below |
18:43.27 | benngard | exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) |
18:43.27 | benngard | exten => 101,5,System(mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf magnus.b@inputinterior.se < /dev/null) |
18:43.48 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
18:44.09 | benngard | 101,4 was executed without error bout 101,5 didnt got executed |
18:44.18 | benngard | changed to: |
18:44.31 | benngard | exten => 101,4,System(fax.sh) |
18:44.55 | benngard | tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf |
18:44.55 | benngard | mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf boras@inputinterior.se < /dev/null |
18:45.23 | benngard | in fax.sh and all starts to work... |
18:45.24 | *** join/#asterisk Alagar (n=Administ@122.164.18.195) |
18:45.56 | p3nguin | < /dev/null is going to take input FROM /dev/null. Is that really what you meant? |
18:46.07 | benngard | yes |
18:46.13 | p3nguin | okay, just making sure. |
18:46.35 | Katty | yawns |
18:46.43 | p3nguin | throws popcorn in |
18:47.00 | Katty | so so so tired |
18:47.09 | benngard | on me? |
18:47.22 | Katty | it's not always about you. |
18:47.22 | p3nguin | It's one of those days, apparently. |
18:47.28 | benngard | :) |
18:47.52 | Erestar | Does anyone else experience an issue when dialing out to cell phones where Answer() occurs way to early? If so, can someone point me to a place to resolve the issue? |
18:47.52 | benngard | had ones of those days today :( |
18:48.40 | Katty | i would think you'd be happy the cellphones answered promptly. |
18:48.51 | p3nguin | erestar: If you are dialing out, why would you have Answer() in your dialplan? |
18:49.06 | Katty | p3nguin: call file maybe. |
18:49.09 | Erestar | They're originated calls |
18:49.21 | titter` | benngard: I have had the most success with http://caspian.dotconf.net/menu/Software/SendEmail/ for sending e-mails via System() on my fax2email script |
18:49.35 | Katty | oh, i guess i could link my fax page. |
18:49.39 | Erestar | Well, the issue is that I'm playing a sound file, and it starts playing before the call is actually answered |
18:49.40 | Katty | but it's not on sending faxes. |
18:49.57 | titter` | Katty: link anyways, ill probably do that next |
18:49.58 | dustybin | is AGI the only way to execute shell commands from a dialplan in asterisk 1.6 ? |
18:50.00 | p3nguin | erestar: Originate it the other way around. |
18:50.14 | benngard | titter`: yes, but why isnt 101,5 not even got executed |
18:50.22 | p3nguin | erestar: Make the cell phone answer before going to the dialplan. |
18:50.52 | benngard | when i moved both "system" commands to a shell script it worked |
18:50.55 | Erestar | p3nguin, Hmm. I'm afraid I'm not sure how to do that |
18:50.57 | titter` | benngard: pastebin the CLI of the failed attempt and somone could help |
18:51.33 | Katty | titter`: http://42ndgeekstreet.blogspot.com/2009/11/asterisk-faxing.html |
18:51.44 | p3nguin | erestar: originate SIP/1234567@itsp.com extension 123@yourcontext |
18:51.47 | ruben23 | hi |
18:51.49 | p3nguin | erestar: Something like this. |
18:52.02 | benngard | i cant send a fax now, i am at home, can just receive |
18:52.31 | titter` | Katty: thanks, I have had that bookmarked form my research in the past |
18:52.38 | benngard | but i can do it tomorrow when i am back at office |
18:52.47 | *** join/#asterisk lupine_85 (n=lupine_8@unaffiliated/lupine-85/x-7392152) |
18:52.48 | Katty | titter`: orly, i wonder how you got your hands on it. |
18:53.01 | titter` | Katty: the magical power of google |
18:53.13 | p3nguin | erestar: In my experiences, the tech/data leg of the call must answer and be 'up' before extension will execute. |
18:53.23 | Katty | ohiseethenk |
18:53.35 | Erestar | p3nguin, I believe that's what I'm doing... Originating on Channel DAHDI/1/5555555 with the extension 500 (which is where the sound file gets played) |
18:53.45 | benngard | i can do alot of things remotly, but i cant put a paper in a fax and press the buttons :( |
18:53.54 | Erestar | And where 5555555 is the cell phone number I'm calling |
18:53.58 | titter` | :> |
18:54.01 | p3nguin | erestar: Oh, dahdi... that's why it is behaving that way. |
18:54.11 | Erestar | p3nguin, ?? |
18:54.28 | p3nguin | erestar: DAHDI channels go up as soon as the number is finished -- no waiting for an answer. |
18:54.39 | Katty | watches the squirrels |
18:54.47 | Erestar | Oh |
18:54.50 | Erestar | Yuck |
18:55.01 | p3nguin | erestar: So consider some magic with WaitForSilence() or AMD. |
18:55.03 | Erestar | So should I be waiting for silence or something? |
18:55.19 | Katty | waiting for the awkward silence. |
18:55.27 | Katty | it happens, you know |
18:55.32 | [TK]D-Fender | [13:54]<p3nguin>erestar: DAHDI channels go up as soon as the number is finished -- no waiting for an answer. <- many assumptions here |
18:55.35 | Katty | when the little old lady asks you if you've seen the tapicoa. |
18:55.49 | p3nguin | erestar: [tk]d-fender is willing to help you solve your issues. |
18:55.54 | [TK]D-Fender | :p |
18:55.55 | Erestar | :-p |
18:56.13 | Katty | you did it now fenrus |
18:56.22 | Katty | s/fenrus/ [TK]D-Fender |
18:56.26 | titter` | Katty: interesting blog ... asterisk advice and diet tips |
18:56.27 | Erestar | The reason I thought it was only cell phones is because the last time I messed with asterisk back in the zapata days this only occured when calling cell phones |
18:56.28 | Erestar | I think |
18:56.33 | [TK]D-Fender | Katty: "Fenris" |
18:56.34 | Katty | titter`: and product reviews. |
18:56.41 | Katty | [TK]D-Fender: i tried to tab complete fender |
18:56.45 | Katty | [TK]D-Fender: fenrus |
18:56.46 | titter` | Katty: yup, can't forget those |
18:56.55 | Katty | titter`: what can i say. |
18:56.59 | Katty | titter`: i have a wide range of interest. |
18:57.17 | Katty | titter`: there's also 42ndrecipestreet |
18:57.19 | [TK]D-Fender | Erestar: What card are you using? |
18:57.46 | Erestar | [TK]D-Fender, e159:0001 Wildcard TDM400P REV I |
18:57.52 | titter` | Katty: same, but my diet has consisted of lemon pepper chicken i make, and brown rice portioned into a 1:1 carb:protien diet |
18:58.06 | titter` | Katty: sonow your blog has made me hungry |
18:59.02 | Katty | titter`: i was doing a 350-400 per meal and 100-150 on the snacks. |
18:59.14 | Katty | titter`: but i've since just been doing 1500 a day |
18:59.43 | [TK]D-Fender | Erestar: If you use "callprogress = yes" for your channels 8 will not consider them "answered" the moment theya re placed. this runs the risk however of your calls being randomly disconnected |
19:00.28 | titter` | Katty: I try not to count calories as much as concetrating on the 1:1 ratio per pound of weight ... tends to keep the calorie intake more random, but also makes you focus on clean food. can't accomplish that goal without eating clean. |
19:00.31 | *** part/#asterisk toddejohnson (n=toddejoh@ppp-70-226-205-249.dsl.spfdil.ameritech.net) |
19:00.37 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
19:00.50 | Katty | titter`: well i don't just calorie count, i also nutrient count. |
19:00.59 | Erestar | [TK]D-Fender, Thanks... I'm taking a look at that now |
19:01.04 | Katty | titter`: g and mg of vitamins, minerals, etc. |
19:02.31 | titter` | Katty: that's good, good job on the goals |
19:03.10 | Katty | titter`: gotta keep that girlish figure. |
19:03.11 | Katty | i mean. |
19:03.16 | Katty | that 50 year old man in the basement figure. |
19:03.57 | titter` | heh, i just try to avoid the typical "i.t. guy" figure |
19:04.33 | titter` | living in florida also motivates me ... can't be the fatass of the bunch when going to the beach |
19:05.00 | Katty | what's wrong with Roy and Moss? |
19:05.35 | Katty | especially moss. |
19:06.26 | Erestar | [TK]D-Fender, Thanks, that worked perfectly for the first test. Do you know what circumstances the random disconnects start happening? Is it when there are lots of concurrent calls or something? |
19:06.53 | [TK]D-Fender | Erestar: No... * detects silence, etc.... lack of ringing, et |
19:07.16 | titter` | Katty: havent watched the show enough to say, watched one episode |
19:08.30 | Katty | titter`: it's lovely. |
19:09.09 | titter` | Katty: ill have to watch more ... |
19:09.19 | dustybin | how can one pass "${CALLERID}" into a shell script ? |
19:09.40 | Katty | echo? |
19:09.50 | Katty | and system() |
19:10.02 | dustybin | asterisk 1.6 doesnt have system |
19:10.05 | dustybin | i only have AGI |
19:10.14 | Katty | 1.6 doesn't have System()? |
19:10.20 | dustybin | nope :( |
19:10.21 | Katty | i find that very hard to believe. |
19:10.34 | dustybin | ill have another look :D |
19:10.45 | Katty | [TK]D-Fender: was System() taken out of 1.6? |
19:11.25 | dustybin | oh it worked! |
19:11.28 | dustybin | you are correct! |
19:11.29 | *** join/#asterisk daidoji (n=daidoji7@adsl-99-27-128-66.dsl.pltn13.sbcglobal.net) |
19:11.31 | titter` | I am using it now :< so if it was, news to me |
19:11.34 | Katty | [TK]D-Fender: nevermind. |
19:11.41 | dustybin | hugs Katty |
19:12.22 | [TK]D-Fender | :) |
19:12.24 | dustybin | now its time to test this |
19:12.25 | dustybin | exten => s,1,System(/usr/local/bin/mythtvosd --template=cid --caller_number="${CALLERID(num)}" --caller_date="${STRFTIME(${EPOCH},%F)}" --caller_time="${STRFTIME(${EPOCH},%r}" --caller_name="${CALLERID(name)}") |
19:12.35 | dustybin | if this works, this will be the best thing ever |
19:12.37 | Katty | Subject: Fire... Dear Sir/Madam, I am writing to inform you of a fire that has broken out at the premises.... wait, that's too formal. |
19:13.49 | dustybin | IT WORKS!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
19:13.59 | Erestar | dustybin, =) |
19:14.00 | dustybin | when somebody rings me, a OSD appears on mythtv |
19:14.05 | fenrus | sexy |
19:14.07 | dustybin | falls off chair |
19:14.08 | Erestar | Oh nice |
19:14.13 | fenrus | does it pause the movie aswell ? |
19:14.16 | fenrus | or mute the tv ? |
19:14.18 | dustybin | nope |
19:14.18 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
19:14.25 | fenrus | turn on the lights ? |
19:14.27 | fenrus | =)) |
19:14.30 | dustybin | heh |
19:14.31 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
19:14.34 | benngard | make coffee |
19:14.38 | dustybin | im in shock |
19:14.49 | fenrus | then you have work to do |
19:14.50 | Katty | benngard: throw coffee. |
19:14.51 | dustybin | rings himself |
19:14.52 | fenrus | =) |
19:15.00 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.172.179.dsl.dyn.forthnet.gr) |
19:15.39 | benngard | Katty: catch! |
19:15.43 | Katty | i need to velcro a pager to my remote |
19:15.51 | Katty | so i can find it when the dog runs off with it |
19:15.55 | fenrus | haha |
19:15.56 | *** join/#asterisk minotaur01 (n=minotaur@24.215.3.50) |
19:16.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
19:17.29 | beek | Katty: don't your ferrets ever take it from you? |
19:18.07 | Katty | no |
19:18.41 | Katty | guess it's not interesting enough |
19:18.42 | *** join/#asterisk minotaur01 (n=minotaur@24.215.3.50) |
19:19.10 | titter` | If my dog ran off with my remote, I would not be happy ... |
19:19.28 | Erestar | So if you guys had to call a bunch of people and play a message, but you only had 4 analog lines to do it, how would you make it work? |
19:19.36 | [TK]D-Fender | [14:14]<benngard>make coffee <- Mine does |
19:19.52 | benngard | lucky u! |
19:19.55 | fenrus | =)) |
19:20.04 | Katty | Erestar: i guess the first question is why are we calling a bunch of people, and what is the nature of the message. |
19:20.12 | titter` | lol |
19:20.23 | fenrus | Katty, you don't like voice-spammers? :) |
19:20.25 | Erestar | Katty, Its for an opt-in alert system |
19:20.39 | Katty | Erestar: now put that in english. |
19:20.42 | dustybin | i spend my friends 5 days solid with .call_me |
19:20.46 | dustybin | *spammed |
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19:20.49 | *** join/#asterisk stan (n=stan@87-194-216-82.bethere.co.uk) |
19:21.03 | Erestar | Katty, People can sign up to receive phone messages when their kid's school has a delay or is cancelled |
19:21.17 | Katty | Erestar: okay. |
19:21.25 | Katty | Erestar: how many people are we talking about here, roughly |
19:21.31 | Erestar | http://communitycrieronline.com/ |
19:21.40 | Erestar | In some cases, 15k ;) |
19:21.54 | Katty | Erestar: and how often do you really call them |
19:22.04 | Erestar | We don't know yet ;) |
19:22.19 | Katty | Erestar: my personal suggestion would be txt messages, not phone calls. |
19:22.20 | *** join/#asterisk minotaur01 (n=minotaur@24.215.3.50) |
19:22.28 | Erestar | We do texts too |
19:22.41 | Erestar | I've already written a system using the manager API that monitors channels and what not to find available lines |
19:22.47 | Erestar | And places calls based on what's available |
19:22.47 | Katty | Erestar: if you absolutely need phone calls, with audio, you need more than 4 analog lines |
19:23.07 | Erestar | I just had to ask to see if there was some built in way of doing it |
19:23.17 | benngard | thought about erlang? |
19:23.26 | Erestar | Oh, in some cases we'll have 24 analog lines, and sometimes we'll be using a VoIP provider |
19:23.32 | Katty | You mean a magically way to place a phone call without using a phone line? |
19:23.48 | Erestar | No, no |
19:23.59 | fenrus | like a built in system queueing up outgoing calls ? :) |
19:24.07 | Erestar | fenrus, Yeah |
19:24.07 | fenrus | waiting for available lines |
19:24.15 | Erestar | fenrus, Exactly that |
19:24.17 | Katty | i'm sure you can write one |
19:24.21 | Erestar | Ok |
19:24.22 | Erestar | I did |
19:24.26 | Erestar | And it works |
19:24.31 | fenrus | i've not seen anything - but google a bit :) |
19:24.32 | Katty | horay! |
19:24.38 | titter` | Anyone know off the top of their head how to redirect a call different when a Polycom is forwarded? I see it gives a diversion ;reason=deflection, then places a new call to the number ... but I would like to set the callerid of that outbound call |
19:24.56 | bmoraca_work | Erestar, unless you're prepared to pony up for the capacity to call 15K people, you might be best served outsourcing this (15,000 people at 24 people at a time, even if it's only 1 minute per call will take 10 hours to call) |
19:25.06 | Erestar | We're testing it this weekend, but I'm completely new to Asterisk and phone systems, so I _had_ to check before we move to production just to make sure I didn't just waste a week ;) |
19:25.08 | Katty | titter`: status |
19:25.21 | titter` | tahnky |
19:25.22 | daemon | hey guys, im using the asterisk 'ajax' gui, all is working excellently but non of my logs seem to be appearing |
19:25.26 | titter` | thanky* |
19:25.36 | daemon | they are in the database, but nothing ever gets displayed in my browser |
19:25.47 | Erestar | bmoraca_work, I know it. I've got a spreadsheet that figures that stuff out. I was most asking about the out going queue system that fenrus was talking about |
19:25.51 | Katty | daemon: we do not support gui's here. please visit whatever irc chanenl supports your box. |
19:25.54 | Erestar | Er, mostly* |
19:26.29 | *** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
19:26.47 | *** join/#asterisk puzzled (n=patrick@535335AA.cable.casema.nl) |
19:27.04 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
19:27.21 | *** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net) |
19:27.26 | daemon | Katty, asterisknow, is on asterisk.org... so the channel would be here I think |
19:27.41 | Katty | daemon: we DO NOT SUPPORT GUIS HERE. |
19:27.50 | bmoraca_work | daemon, #asterisknow |
19:27.53 | Katty | daemon: We are NOT asterisknow. |
19:27.56 | bmoraca_work | daemon, it's right in the topic |
19:28.06 | *** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net) |
19:28.07 | Katty | daemon: Please go to the appropriate IRC channel. |
19:28.35 | *** join/#asterisk minotaur01 (n=minotaur@24.215.3.50) |
19:28.46 | seanbright | jeez |
19:28.59 | devmod | Hello, If i wanted to setup a very simple asterisk setup at home, what voip provider would u guys recommend ? |
19:29.03 | [TK]D-Fender | gathers the pitchforks, torches, and villagers |
19:29.15 | [TK]D-Fender | ~itsplist-us |
19:29.16 | infobot | methinks itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
19:29.16 | titter` | I have this GUI.... *channel bursts out in rage* |
19:29.19 | [TK]D-Fender | devmod: ^^^^^^ |
19:29.20 | seanbright | i can't believe you would ask about a GUI in here you SON OF A BITCH |
19:29.30 | seanbright | heh |
19:29.40 | Erestar | Man, I miss IRC ;) |
19:29.45 | bmoraca_work | seanbright, it's not that. the problem is (especially with asterisknow) that there's almost no way for anyone in here to support it |
19:29.51 | Katty | seanbright: exactly. |
19:29.51 | seanbright | i know i know |
19:29.55 | Katty | seanbright: YOU KNOW |
19:30.04 | seanbright | the goal of this channel is clear |
19:30.06 | devmod | [TK]D-Fender, thanks :) |
19:30.34 | seanbright | to pawn people off on other channels so there is more "hug time" |
19:30.36 | seanbright | heh |
19:30.58 | seanbright | problems with extensions.conf? try in #asterisk-extensions-dot-conf |
19:31.24 | seanbright | i may be trolling |
19:31.30 | Katty | do you need a hug, mister cranky pants. |
19:31.39 | thehar | needs a hug |
19:31.42 | thehar | very cranky |
19:33.19 | seanbright | i require no hugs |
19:33.51 | Katty | you're gonna be a bitter old cranky man. |
19:33.57 | seanbright | god willing |
19:34.10 | Katty | kay |
19:34.18 | titter` | needs food |
19:34.29 | Katty | titter`: spaghetti! |
19:34.34 | benngard | needs sleep |
19:34.59 | Katty | nap would be good |
19:35.12 | Naikrovek | naps are always welcome |
19:35.14 | Naikrovek | even while sleeping |
19:35.16 | Katty | josh groban looks like Roy from the IT crowd |
19:35.19 | Naikrovek | i dream about sleeping |
19:35.25 | Naikrovek | while sleeping |
19:35.28 | Katty | i dream about having to pee |
19:35.31 | Naikrovek | hehe |
19:35.35 | titter` | Katty: hmm, good idea ... but nothing at my office to make it, so im thnking those new spicey nuggest from wendys will hold me over. |
19:35.45 | Katty | fast food :< |
19:35.47 | Katty | DON"T DO EET |
19:36.05 | titter` | lol |
19:36.30 | Katty | what is nearby, besides wendy's |
19:36.35 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
19:36.55 | titter` | everything basically ... |
19:36.55 | Naikrovek | HeartClogger's, Fatty McGee's, McDonalds, uh, and Best Buy |
19:37.31 | titter` | actually havent had fast food in amost 2 months ... but the spicey nuggets look good |
19:37.36 | Erestar | titter`, Today feels like a Chinese food day to me |
19:38.18 | titter` | 81 and sunny, with a bit of Chinese ... but ive been eating so much brown rice, don't know if I want more rice |
19:38.20 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
19:38.45 | Katty | black beans, kidney beans, seasonins, and cornbread. |
19:38.46 | p3nguin | You must be in Miami. |
19:39.14 | titter` | Other side of the state |
19:39.49 | Katty | titter`: http://www.tonychachere.com/recipesapp/ViewRecipe.asp?RecipeId=68 |
19:39.52 | p3nguin | Naples? |
19:40.00 | Katty | has relatives in Naples |
19:40.06 | titter` | very close to Naples |
19:40.09 | titter` | Fort Myers |
19:40.12 | Katty | titter`: i'm making that this weekend, no rice tho. substituting cornbread. |
19:40.13 | p3nguin | St. Pete? |
19:40.16 | p3nguin | oh |
19:40.27 | titter` | St Pete and Miami are two places I wouldn't live |
19:40.36 | titter` | St Pete Beach is nice |
19:40.40 | Katty | what's at Fort Myers |
19:40.47 | Katty | i'm guessing military personell? |
19:40.50 | Katty | personnel |
19:40.52 | titter` | Nope |
19:40.56 | titter` | it's a city |
19:41.00 | Katty | ah. |
19:41.07 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
19:41.11 | titter` | north of naples by about 20 minutes |
19:41.28 | Katty | i don't remember much being in Naples |
19:41.29 | p3nguin | so like 3 minutes from Cape Coral. |
19:41.31 | titter` | couple colleges, and the airport |
19:41.43 | titter` | yup 3 minutes from Cape Coral |
19:41.49 | titter` | depending how close you are to the bridges |
19:42.02 | titter` | and how many northerns are getting lost on the roads |
19:42.10 | titter` | northeners* |
19:42.43 | titter` | It was 28 last Sunday night ... now its 81 |
19:42.47 | titter` | Love the climate |
19:42.52 | Erestar | You guys made me hungry. I'm off to lunch. Thanks for all of your help and advice :) |
19:44.04 | titter` | http://www.wunderground.com/history/airport/KRSW/2010/1/11/DailyHistory.html?req_city=NA&req_state=NA&req_statename=NA |
19:44.09 | Katty | i could so nap at my keyboard |
19:44.21 | p3nguin | I might vacation in New Smyrna Beach this summer. |
19:44.28 | titter` | I have done that before |
19:44.32 | titter` | jacket on my desk |
19:44.34 | titter` | and nap away |
19:44.43 | titter` | to be woken by the sound of the polycom |
19:45.35 | *** join/#asterisk tzafrir (n=tzafrir@212.179.75.202) |
19:46.16 | Katty | p3nguin: what's at New Smyrna Beach? |
19:46.43 | Katty | titter`: i usually just lean back in my chair, and play some relaxing ambient music free rain/wind video. |
19:47.02 | Katty | titter`: after the 30 minutes, the slight change from ambience to quiet is enough to wake me up. |
19:47.09 | titter` | hmm |
19:47.13 | p3nguin | katty: Condos... and a beach. With less people than Daytona Beach. |
19:47.29 | Katty | not much of a beach person, personally |
19:47.31 | titter` | Daytona can get crazy |
19:47.53 | p3nguin | I used to spend a lot of time there. |
19:48.14 | titter` | Summer isnt as bad as Spring Break |
19:48.50 | titter` | off to find some food, ill be back ... thinking a turkey wrap |
19:49.16 | bmoraca_work | oh food...how great thou art |
19:49.35 | dustybin | i really got to stop ringing myself, this is getting silly |
19:49.57 | Katty | lol |
19:50.26 | titter` | how nice of dell to send me a whole new lid, bezel, and switchpanel for free for my xpls ... <3 warranties |
19:50.27 | benngard | spring break at pompano beach could be .... iÃÂntresting ;) |
19:50.40 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
19:50.47 | Katty | titter`: shoo |
19:50.49 | Katty | titter`: go eat. |
19:50.54 | titter` | :> |
19:51.34 | *** join/#asterisk ChrisWi (n=admin@mx2.wwserver.net) |
19:52.01 | dustybin | if somebody rings me, the caller name is the number, is there a way to replace numbers with peoples name? |
19:52.53 | *** join/#asterisk cadmium (i=mike@217.194.139.22) |
19:52.57 | p3nguin | dustybin: Of course. |
19:53.17 | cadmium | hi. what channel variable can I use to obtain the peername |
19:53.18 | dustybin | p3nguin: i mean from the outside |
19:53.39 | dustybin | p3nguin: this would require me setting up some kind of database? |
19:54.02 | p3nguin | dustybin: You could, but you can also run a script to do free lookups via internet. |
19:54.16 | p3nguin | That's what I do for my home number. |
19:54.20 | dustybin | oh nice |
19:54.33 | cadmium | p3nguin how can i reference the peername as a channel variable in the dialplan ? |
19:54.37 | p3nguin | For work, I let the ITSP do queries and they send me the name. |
19:54.43 | dustybin | can the internet find the name for any number |
19:54.52 | p3nguin | dustybin: no |
19:55.57 | dustybin | p3nguin: i only have 2 friends, so all i need to do is create some kind of table with name and number, if there is a match then use the name |
19:57.22 | bmoraca_work | dustybin, astdb should be more than easy enough. or even just global variables. just set CALLERID(name) to whatever you want. |
19:57.48 | bmoraca_work | or hell, just use callerid matching on your extension priority |
19:57.50 | dustybin | bmoraca_work: if my asterisk server rings my cellphone, my cellphone says 'private' |
19:57.56 | dustybin | displays no name or number |
19:57.59 | bmoraca_work | dustybin, no way to get around that. |
19:58.07 | dustybin | aye ok |
19:58.20 | dustybin | bmoraca_work: is that down to my VOIP provider? |
19:58.21 | bmoraca_work | dustybin, best you could do is have it display a number. you cannot send callerid name. CNAM is ALWAYS performed on a lookup basis by the CALLED party |
19:58.25 | p3nguin | cadmium: ${SIPCHANINFO(peername)} |
19:58.53 | dustybin | right lets check the outgoing first |
19:58.56 | bmoraca_work | dustybin, if your VOIP provider allows you to specify any callerid number, yes you could send number. but you will never be able to send name |
19:59.13 | cadmium | p3nguin thanks |
19:59.32 | bmoraca_work | many (probably most) providers do not allow you to specify CID to anything other than numbers you own. |
19:59.42 | bmoraca_work | there's been a big crackdown on that lately...callerid spoofing is bad |
19:59.43 | *** join/#asterisk guax (n=guax@unaffiliated/guaxinim) |
20:00.08 | [TK]D-Fender | [14:58]<bmoraca_work>dustybin, if your VOIP provider allows you to specify any callerid number, yes you could send number. but you will never be able to send name <- I've seen PRI's I can do this on... not a stretch to imagine its possible for them to apss the same on via SIP |
20:00.34 | guax | Got a attended transfer that when hung up turns into a blind transfer, when that happens the transfered channel stops playing music on hold and gets muted even if the other channel should be ringing. |
20:00.56 | dustybin | aye ok thanks, i will try and set the number |
20:01.05 | guax | even muted the other leg keeps ringing and if its answered the call place normaly |
20:01.44 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
20:02.19 | bmoraca_work | aaaaaand...lunchtime |
20:02.57 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
20:04.02 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
20:04.04 | dustybin | before the dialplan rings my VOIP provider, all i need before is: |
20:04.14 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
20:04.17 | dustybin | SetCallerID(number) |
20:05.10 | dustybin | whoops |
20:05.11 | dustybin | Set(CALLERID(num)=value) |
20:05.23 | dustybin | tests |
20:12.41 | dustybin | oh it was working anyway, i think i had a problem with .call_me showing a private number |
20:13.00 | dustybin | tests |
20:16.00 | titter` | adventures of dealing with tourists on the road |
20:19.38 | minotaur01 | im setting up a new building in a month or so with a fresh install on asterisk... they will require external paging... dose anyone have any recommandations |
20:19.47 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
20:20.19 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
20:20.29 | Katty | i recommend... |
20:20.39 | Katty | having a healthy breakfast. |
20:20.45 | minotaur01 | lol |
20:22.53 | Katty | we don't use external paging. |
20:22.54 | Katty | j |
20:23.01 | Katty | just the add sip header stuffs. |
20:24.04 | minotaur01 | i use a CyberData Voip Amplifier in one building now but im wondering if there is anything better out there |
20:24.12 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
20:24.50 | Katty | hi raden |
20:25.43 | titter` | really needs a nap now |
20:25.57 | *** part/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
20:26.04 | Katty | 2.5hrs before i go home still. |
20:26.09 | Katty | but a nap does sound pretty awesome. |
20:26.17 | titter` | I could go now |
20:26.39 | titter` | I just like the idea of working, and getting things finished lol |
20:27.37 | Katty | well currently nothing here is broken. |
20:27.48 | Katty | just waiting for someone to call, email, or otherwise run screaming to my office |
20:28.28 | titter` | lol, I know the feeling ... I restarted 11 switches and almost 400 shoretel phones last night after making a system wide change ... and today has been quiet ... maybe because they can't call me |
20:28.44 | titter` | ponders... |
20:28.51 | titter` | oh well. |
20:29.33 | Katty | sounds good to me |
20:30.02 | minotaur01 | titter: thats the best phone call |
20:30.05 | *** join/#asterisk ruied (n=ruied@bl7-217-61.dsl.telepac.pt) |
20:30.39 | titter` | yes ive gotten calls from people saying their phone isn't working |
20:31.11 | minotaur01 | titter: how are those shortel phones to work with? |
20:31.56 | titter` | since we run ipsec tunnels from some office to access internal websites, ive gotten emails stating my internet is down, as they e-mail me. |
20:32.24 | minotaur01 | lol |
20:32.25 | titter` | minotaur01: it's a pig. it's a nice system, but the costs of it isn't worth it |
20:33.09 | Katty | you know. |
20:33.14 | Katty | we're actually a shoretel partner |
20:33.21 | Katty | but we've never sold a single shortel system |
20:33.24 | titter` | it was from acquisition, and making asterisk work with shoretel ... bleh. |
20:33.41 | Katty | seems like everyone around here wants or already has, toshiba. |
20:34.06 | minotaur01 | i use Aastra's 57i my self but always looking for the next thing |
20:35.23 | titter` | shortel phones are proprietary to their hardware ... sort of |
20:35.27 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
20:35.42 | minotaur01 | ah no good for me then |
20:35.49 | titter` | shoretel only uses one rtp stream on port 5004 |
20:35.57 | titter` | and only supports MGCP |
20:36.29 | minotaur01 | i had a proprietary 3Com Nbx at one time... what a pice of crap... |
20:36.34 | simplydrew___ | does anyone know if there's a script out there or addon for asterisk that will let you dial into asterisk and define a time to call a number? |
20:37.08 | minotaur01 | like a hotel wake up service? |
20:37.11 | minotaur01 | then yes |
20:37.18 | simplydrew___ | pretty much, but to call an external number |
20:37.27 | simplydrew___ | I have a wake up call script integrated, but it won't dial out |
20:38.02 | bmoraca_work | it should be fairly simple to modify the script you have |
20:38.07 | minotaur01 | you have to change the channel its calling on |
20:38.09 | bmoraca_work | to do what you want |
20:38.24 | *** join/#asterisk hfb (n=hfb@96.247.53.139) |
20:38.26 | p3nguin | Wake up calls from outside the system? |
20:38.36 | minotaur01 | find the spot that says SIP/[ext] |
20:38.51 | simplydrew___ | essentially yes, I just want to be able to tell it to call my cell at a certain time or something |
20:38.54 | minotaur01 | and change it to TRUNK/[number] |
20:39.13 | p3nguin | minotaur01: You sound like a GUI. |
20:39.23 | minotaur01 | lol |
20:39.29 | minotaur01 | i use one |
20:39.33 | Katty | how does one sound like a GUI |
20:39.33 | p3nguin | It shows. |
20:39.41 | Katty | does it make specific, gurgly noises? |
20:39.46 | titter` | lol |
20:39.48 | Katty | or maybe whirly sounds? |
20:39.52 | Katty | mwrrrrrrrhhhhfff |
20:40.06 | minotaur01 | lol |
20:40.08 | titter` | pew pew |
20:40.57 | bmoraca_work | the sound of my belly trying to digest the McDonalds double cheeseburger i just ate |
20:41.21 | titter` | bleh |
20:41.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
20:42.14 | bmoraca_work | meh...it was cheap and quick and it's been pouring rain here for 4 days straight...didn't feel like going anywhere else |
20:42.15 | titter` | why is it old people have a hard time seeing a bright red car? that they much pull out of whereever they are, and cut me off, with no notice of their action |
20:43.21 | Katty | that's not strictly old people |
20:43.30 | bmoraca_work | most people drive like morons |
20:43.49 | Chainsaw | I think you'll have better luck with a bright yellow car. |
20:43.50 | titter` | true, but where I am it is mostly old people, or out of state plates |
20:44.07 | p3nguin | The problem with this country is that there is no death penalty for incompetence. |
20:44.07 | titter` | has road rage |
20:44.30 | bmoraca_work | right now, we're getting the morons who slow down to 1 MPH to go through a puddle that's not even 6" deep |
20:44.34 | *** join/#asterisk dpisites (n=cheng@dsl-67-204-18-213.acanac.net) |
20:45.01 | Katty | titter`: eww. |
20:45.08 | Katty | titter`: you need to work on that |
20:45.15 | Katty | titter`: very very bad for your health |
20:45.48 | titter` | lol |
20:45.52 | titter` | I have a lead foot too |
20:46.05 | p3nguin | I'm not too good to run someone off the road for driving like a twit. |
20:46.10 | Katty | i assure your, your foot is not lead. you just like to drive fast. |
20:46.12 | *** part/#asterisk dpisites (n=cheng@dsl-67-204-18-213.acanac.net) |
20:46.15 | dlynes | I'm just wondering what I'm missing? The following two lines tell me that I'm missing the trailing parenthesis? |
20:46.16 | dlynes | exten => s,2,Read(CHOICE,/var/spool/asterisk/autoattendants/default/${ARG2},1,,,20) |
20:46.17 | dlynes | exten => s,n,GotoIf($["${READSTATUS}"="OK"]?${CHOICE},1:1) ; if the user picked something, branch to the choice |
20:46.25 | *** join/#asterisk dpisites (n=cheng@dsl-67-204-18-213.acanac.net) |
20:46.33 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
20:48.22 | dlynes | It's the second of those two lines where I'm getting the error |
20:48.30 | voipmonk | hrmm |
20:48.34 | dlynes | ${CHOICE} has a value of '1', if it matters |
20:48.37 | minotaur01 | was anyone come across a problem with the queues conf not loading propery in version 1.6.0.20 on start/restart but loading fine on reload?? |
20:49.28 | dlynes | minotaur01, could be the order in which you're loading your modules |
20:49.46 | bmoraca_work | dlynes, your destination syntax for gotoif is wrong. it's looking for a label, as such ${CHOICE},1:1 is wrong. |
20:49.59 | Katty | dlynes: just stick another parenthesis at the end |
20:50.00 | titter` | Katty: I guess |
20:50.08 | *** join/#asterisk mlr78 (n=mratliff@74.5.197.138) |
20:50.20 | titter | Katty: I can't go 2 days without see 1xx in my car |
20:50.22 | dlynes | ${CHOICE},1 is one label (exten, priority); 1 is another (priority) |
20:50.33 | minotaur01 | hey dlynes, how do i change the order? |
20:50.51 | Katty | titter: i very rarely see my car up in the 3 digits. |
20:51.15 | bmoraca_work | dlynes, it's not looking for an exten,priority, it's looking for a LABEL |
20:51.18 | dlynes | minotaur01, autoload => no in your modules.conf file, and then do a load => module_name.so specifically for each module you want to load, in the order you want |
20:51.23 | p3nguin | I usually only go that fast when everyone else is also going that fast. |
20:51.38 | dlynes | bmoraca_work, so i can't use exten, priority or context, exten, priority there? |
20:51.46 | dlynes | bmoraca_work, only priority and n(label)? |
20:51.57 | bmoraca_work | dlynes, i don't believe so, no. your second question would be an affirmative |
20:52.05 | minotaur01 | dlynes: thank you |
20:52.09 | dlynes | that sucks, immensely |
20:52.13 | bmoraca_work | dlynes, if you want to do it the way you have been, Background() would be more appropriate |
20:52.14 | p3nguin | dlynes: I use exten,priority in 1.4 and it works fine. |
20:52.21 | titter | p3nguin: http://www.radarbusters.com/radar-detectors/escort-radar-detectors/escortpassport9500ix.cfm |
20:52.21 | dlynes | p3nguin, i'm using 1.6.1 |
20:52.40 | p3nguin | I can't imagine they would take out that functionality. |
20:52.52 | dlynes | bmoraca_work, how is background any different from read(), goto()? |
20:53.08 | Katty | oooh. it appears the sun is out. |
20:53.17 | Katty | the crittercam is BRIGHT for the first day in....days! |
20:53.18 | bmoraca_work | dlynes, background() goes directly to the extension entered in...it does not store the captured digits in a variable |
20:53.21 | p3nguin | titter: Is that what you have? |
20:53.27 | titter | YUP |
20:53.32 | titter | yup* |
20:53.40 | p3nguin | titter: Have you ever used a Valentine One? |
20:53.46 | titter | I have |
20:53.55 | p3nguin | How would you compare the two? |
20:54.13 | p3nguin | I find the Valentine to be bulky and ugly. |
20:54.16 | dlynes | bmoraca_work, yeah, but read gives me a bit more flexibility |
20:55.16 | p3nguin | dlynes: n,GotoIf($[${BLACKLIST()}]?misc,blocked,1) for example... misc context, blocked exten, priority 1. No problems. |
20:55.20 | titter | Honestly they are pretty close |
20:56.42 | p3nguin | titter: If you had an extra $500 sitting around and wanted to buy another detector for another car, would you buy either of the two or something different? |
20:57.03 | titter | 1 sec smoeone made a 911 call -.- |
20:57.11 | Katty | i can think of better things to buy with 500 bucks |
20:58.13 | dlynes | p3nguin, I just said forget it, and called it ch1, ch2, and ch3 instead |
20:58.20 | p3nguin | Some people want to have a good radar detector, though. |
20:58.31 | dlynes | p3nguin, so it should go to 1,n(ch1) and 1,n(ch2) and 1,n(ch3) instead |
20:58.34 | titter | I would get the Passport again |
20:58.38 | titter | more features |
20:58.40 | titter | same range |
20:58.45 | titter | V1 is better on X band |
20:58.48 | titter | bit overkill |
20:58.48 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
20:58.50 | titter | and has arrows |
20:59.04 | p3nguin | dlynes: So your labels are working, but con,ext,pri would not? |
20:59.08 | titter | I dont use arrows much, since I know where cops are probably sitting |
21:00.22 | p3nguin | The arrows seem like a novelty to me. If you know you are speeding, it shouldn't matter where the signal came from... SLOW DOWN anyway. |
21:00.47 | titter | A good detector goes a long way, knowing when tos peed, and not to speed plays a big roles |
21:01.43 | Katty | or you could just....not...speed. |
21:01.48 | Katty | or go 75 in a 70 zone. |
21:01.58 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
21:03.00 | [TK]D-Fender | Here the cops don't car if you go like 60-65 in a 50 zone... its when you pass by 20+ |
21:05.07 | titter | I avg 90 in a 70 |
21:05.10 | titter | on 75 |
21:05.22 | p3nguin | I normally don't go over 10 over on the freeway. |
21:05.32 | p3nguin | unless other traffic is also going over. |
21:05.40 | Skeeter- | p3nguin, its a called a freeway from some reason |
21:05.43 | bmoraca_work | I go 80 on freeways |
21:05.48 | Skeeter- | 100 neeeds to be around 140 |
21:05.54 | Skeeter- | kmh here |
21:06.00 | p3nguin | I prefer the freeways where there is no daytime speed limit. |
21:06.04 | bmoraca_work | speed limit on most of them is 65, but we're starting to get some 70 and 75s around |
21:06.31 | titter | it's easy with my car ... but most of the time I cruise around 90, and if get a blip ill slow down to 85ish until I know if its real or not |
21:06.39 | titter | the 9500ix hasnt failed me yet |
21:06.48 | p3nguin | On my way home from Phoenix, I didn't drive under 110 unless impeded by other traffic. That's 1600 miles. |
21:08.44 | p3nguin | Lots of speed corridors, too, but never say any highway patrol in any of them. |
21:08.50 | titter | ive picked up FHP almost 3 miles in front of me |
21:09.28 | p3nguin | How well does your detector do when the radar power is turned way down? |
21:10.00 | p3nguin | I've approached cops and their signal is so weak that even right on top of them, my detector isn't showing full bars. |
21:10.36 | titter | they are proabably using pop radar or instant on |
21:10.41 | p3nguin | no |
21:10.57 | p3nguin | Those are totally different than what I described. |
21:11.57 | p3nguin | POP isn't detected by a lot of units, and instant-on goes from off to on in a fraction of a second and stays on. They just often turn down the power on the normal mode. |
21:12.06 | titter | http://guysoflidar.com/august-2007/radar-detector-test.html |
21:12.09 | bmoraca_work | they've started using LIDAR around here |
21:12.26 | titter | laser is used a lot of places |
21:12.31 | bmoraca_work | radar detectors don't work on them |
21:12.44 | titter | they do... just have to have more than the detector |
21:12.55 | p3nguin | Most radar detectors don't come without laser built-in. |
21:13.08 | titter | http://www.radarbusters.com/radar-detector-accessories/laser-veil/laserveil.cfm |
21:13.20 | titter | I have his on my headlights |
21:13.39 | titter | and ive been through a few laser traps without a ticket |
21:13.53 | p3nguin | I can't imagine how that could even work. |
21:14.16 | p3nguin | It's not like lidar only reflects off headlights or something. |
21:14.56 | titter | thats the first place they aim |
21:15.02 | titter | and the only place on my car they can aim |
21:15.08 | titter | if they aim higher |
21:15.08 | p3nguin | No, that's not how it works. |
21:15.11 | titter | it is |
21:15.15 | p3nguin | It's not. |
21:16.21 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
21:17.00 | *** join/#asterisk glasshut (n=chatzill@67.200.59.158) |
21:17.32 | titter | http://www.youtube.com/watch?v=jdeIj1KKL0w&feature=fvw |
21:18.19 | titter | leo is trained to aim at front plates and headlights, they can get a reading back from the car itself, but the better response is from highly reflective surfaces |
21:19.01 | p3nguin | better response... does not mean LIDAR relies on headlights nor license plates. |
21:19.17 | p3nguin | It just means the light is more likely to reflect off those surfaces. |
21:19.21 | *** join/#asterisk ZX81 (n=ZX81@121-74-230-221.telstraclear.net) |
21:19.26 | titter | it doesnt rely on it, but with "MY" car it helps a lot |
21:19.35 | titter | because where my detector is |
21:19.39 | p3nguin | If you take off your plate and remove your headlights, you are not invisable to LIDAR. |
21:19.41 | titter | it will get the lidar hit |
21:19.49 | *** join/#asterisk settntrenz (n=joe@162.226.189.72.cfl.res.rr.com) |
21:20.03 | titter | not invisible, but the range is a lot less for them |
21:20.04 | Katty | yawn. |
21:20.05 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
21:20.09 | Katty | 1.5hrs |
21:20.23 | p3nguin | Why are you still awake? |
21:20.24 | titter | the idea with lidar is to slow down before they get you |
21:20.42 | titter | veil and my detector make that happen |
21:21.09 | titter | http://i109.photobucket.com/albums/n58/dotdotkeek/01ss/IMG_0360-1.jpg?t=1264108849 |
21:21.12 | Katty | well it looks like i'm gonna be driving to The Country tomorrow |
21:21.18 | titter | front of my car |
21:21.22 | titter | my headlights are huge |
21:21.23 | p3nguin | If I were to build a LIDAR unit, I would change the wavelength of light that is used to circumvent the detectors. |
21:21.38 | *** join/#asterisk jelly-bean (n=mikesmul@75-148-103-190-Utah.hfc.comcastbusiness.net) |
21:22.01 | jelly-bean | how could i get asterisk to notify me when a call has been waiting in queue for more than 1 minute? |
21:22.13 | p3nguin | I thing my headlights have more area, but are probably less reflective than yours. |
21:22.47 | [TK]D-Fender | jelly-bean: Write a script that monitors the queue and checks member wait time |
21:23.02 | [TK]D-Fender | jelly-bean: AMI or CLI via "-rx" |
21:23.13 | jelly-bean | what kind of script--any kind? AMI? |
21:23.43 | *** join/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
21:24.16 | settntrenz | Say I have a staff that logs into a queue to handle inbound support calls. The staff (supporting athletic events that don't have a set ending time) leaves for the night when the last event is over. We also have a nighttime answering service. Can I configure asterisk so that the last employee out can use their handset, dial an ext, input a password which would automatically forward calls destined for the support queue to the night extension |
21:25.14 | *** join/#asterisk luminblade (n=luminfor@cpe-24-28-78-39.austin.res.rr.com) |
21:25.16 | bmoraca_work | settntrenz, of course. use astdb to store which mode (day/night) the inbound calls should be treated as |
21:25.29 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
21:25.30 | mlr78 | anyone have a good example of a config file for a linksys spa942? |
21:25.44 | dlynes | ah...found out what the real issue was |
21:25.54 | Carlos_PHX | mlr78: What do you want in your config? |
21:25.59 | dlynes | It has an issue with a trailing parenthesis on the line: exten => 1,1,Dial(${ARG1}/${DB(${ARG2}/option1},${ARG3},ktx) ; enable call parking, call transfer, and call recording |
21:26.07 | p3nguin | dlynes: missing a "close paren"? |
21:26.07 | Carlos_PHX | mlr78: You know the phone can spit out its current one? |
21:26.13 | settntrenz | bmoraca_work: Thanks, I'll do some googling. I didn't know what this type of feature was actually called making it difficult to google. |
21:26.34 | dlynes | I'm guessing it has something to do with the DB function |
21:27.10 | bmoraca_work | settntrenz, it's just an alternate type of routing. you probably won't find anything prebuilt, but it owuld only take 6-7 lines to actually put something together. |
21:27.16 | luminblade | question: running 1.6.2.1. i have multiple peers (carriers) connected running g729 pass-through. if i enable T.38 (t38_udptl=yes) asterisk segfaults from time to time... is there anything else/special i need to do to set up Fax/T.38 pass-thru correctly? i am not trying to connect directly to any fax endpoints/devices. |
21:28.00 | bmoraca_work | dlynes, you are missing the closing parenthesis for your DB() call...but it's impossible for me to tell you where it should be because i don't have any other context for what you're trying to do |
21:28.38 | [TK]D-Fender | checkout time, later all |
21:29.07 | bmoraca_work | dlynes, but i suspect it should be after "option1" |
21:29.53 | settntrenz | bmoraca_work: thanks for pointing me in the direction, now I just gotta see how to do this using trixbox. Before this gig I've only worked on standard asterisk. I'm trying to figure out if I'll break things or have them overwritten by editing extensions.conf directly |
21:29.55 | p3nguin | dlynes: vim is your friend in a case like that. It highlights the matching open or close paren, brace, and bracket. |
21:30.43 | bmoraca_work | settntrenz, uhg...trixbox...let me just tell you that FreePBX (what generally handles trixbox's dialplan) has a module that will do day/night routing for you. |
21:30.59 | settntrenz | yup, just found it. http://www.freepbx.org/support/documentation/module-documentation/day-night-mode-control |
21:31.26 | settntrenz | it looks like it handles doing it from the handsets too |
21:31.38 | bmoraca_work | yes it does. |
21:31.42 | settntrenz | in the past I'd just get a call saying we're closing early and edit the gotoiftime's |
21:32.26 | dlynes | bmoraca_work, oh shit....just realized I'm missing the ')' on DB |
21:32.26 | settntrenz | ok, i'm going to read some docs. Thanks again |
21:32.30 | *** part/#asterisk settntrenz (n=joe@162.226.189.72.cfl.res.rr.com) |
21:32.43 | glasshut | Our company has an editor who listens to long winded stories from people on the phone then needs to transcribe them into text. To do this we set up recordings. The only problem is that on an outbound call if she hits mute, the recording pauses even though she can hear the other person. When you listen to it the times when she pushes mute are chopped right out of the recording. Asterisk... |
21:32.44 | glasshut | ...1.4.22. Any ideas? |
21:33.28 | bmoraca_work | glasshut, make sure that there's no VAD or comfort noise of any kind enabled on the endpoint |
21:34.05 | *** join/#asterisk [Outcast] (n=anonymou@64.202.62.5) |
21:34.15 | *** part/#asterisk stan (n=stan@87-194-216-82.bethere.co.uk) |
21:35.03 | glasshut | I believe I did. I tried to make sure it was turned off in /etc/asterisk/phone.conf silence suppression as well as /tftpboot/SIP* files. I may have missed something. |
21:35.08 | [Outcast] | so what is the deal with g.729 and 1.6.2?" |
21:35.48 | bmoraca_work | [Outcast], it works fine? |
21:36.17 | [Outcast] | no I have been told g.729 does not work with 1.6.2 |
21:36.33 | [Outcast] | There is no module fo rit |
21:36.42 | p3nguin | That would be a bother. |
21:36.57 | bmoraca_work | i'm using 1.6.2.0 right now with g729... |
21:37.02 | bmoraca_work | must be voodoo |
21:37.24 | glasshut | bmoraca_work: I have cisco 7960 phones, enable_vad: "0" in SIPDefault.cnf. Shouldn't that do the trick? |
21:37.28 | [Outcast] | yea it is pass through, but it will no terminate |
21:37.51 | bmoraca_work | once again, i'm using it just fine to terminate as well |
21:38.10 | luminblade | bmoraca_work: do you have fax / T.38 working as well? |
21:38.30 | bmoraca_work | luminblade, fax yes, but I don't use T.38 |
21:39.34 | [Outcast] | so you can have a g.729 call be answered by application in the dialplan |
21:42.17 | glasshut | I am trying to make sure VAD is not on to fix my recordings issue. If I make a call and select mute for both the Cisco 7960 using SIP as well as a cell phone. With the volume the whole way up on the desk phone, should vad make noise if it is turned on? |
21:42.36 | glasshut | It seems dead silent. |
21:42.56 | p3nguin | If you have set vad to be off, and reloaded the phone, it is off. |
21:44.06 | glasshut | Ok, so maybe that isn't the issue as to why pushing mute on the phone cuts that part out of the recording :-( any ideas? |
21:44.21 | p3nguin | What are you using to record? |
21:44.32 | glasshut | MixMonitor I think |
21:44.35 | mlr78 | for the linksys spa phones...is the correct; spaMAC.cfg or spa$MAC.cfg to do auto provisioning via option 66? |
21:45.16 | glasshut | exten => s,1+998(record),MixMonitor(${MIXMON_DIR}${CALLFILENAME}.${MIXMON_FORMAT},,${MIXMON_POST}) |
21:46.06 | Katty | AHHHHHHHHHIEEEEEEEEE. |
21:46.10 | Katty | 1hr. |
21:47.41 | glasshut | If I look at the call length and the recording length in audacity, the recording is significantly shorter if I use the pause button. |
21:48.02 | glasshut | *Mute button |
21:51.45 | Katty | so many pretty birds in the front yard. |
21:53.23 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
21:53.49 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
21:53.53 | LemensTS | Is there a way to get the username of a sip client on an outgoing call? I dont see it in predefined channel variables |
21:54.35 | p3nguin | lemensts: Yes. SIPCHANINFO(peername) |
21:54.37 | [TK]D-Fender | LemensTS: where would you grab this anyway? |
21:55.32 | *** join/#asterisk ctooley (n=ctooley@c-98-193-232-154.hsd1.tn.comcast.net) |
21:55.40 | LemensTS | TKD: Id pass it into an agi script from the dialplan. |
21:55.54 | ctooley | How do I use tshark to output just the SIP messaging from a pcap file? |
21:56.10 | [TK]D-Fender | LemensTS: parse out sip.conf |
21:56.16 | LemensTS | p3nguin: cool never knew that cmd before |
21:56.35 | p3nguin | lemensts: core show functions |
21:57.29 | [TK]D-Fender | LemensTS: Also isn't what you asked for..... |
22:02.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
22:06.23 | titter | [TK]D-Fender: If Asterisk gets a 302 Moved Temp request from a Polycom, is there anyway to change the outgoing Contact it recieved from the Polycom to append say a * to then dialout via a specific pattern matach. I want to forward on incoming callerid, to my forwarded phones. |
22:06.48 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
22:07.05 | *** join/#asterisk sflemming (n=stefan@85.183.40.40) |
22:11.43 | voipmonk | hey titter |
22:11.52 | titter | hey |
22:14.26 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
22:14.48 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
22:17.03 | *** part/#asterisk FinboySlick (n=shark@207.134.8.4) |
22:17.53 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
22:18.08 | Katty | pokes voipmonk |
22:18.50 | *** join/#asterisk Tech_Travis (n=tech_tra@208.179.137.131) |
22:20.21 | titter | bbl, time to diassemble my laptop |
22:23.45 | *** join/#asterisk rossand (n=aross@dhcp-233-179.tb-classrooms.carleton.ca) |
22:25.41 | *** join/#asterisk ruied (n=ruied@bl7-217-61.dsl.telepac.pt) |
22:27.12 | dustybin | its time for a cup of tea |
22:29.22 | dustybin | my lord |
22:29.23 | dustybin | database put cidname 8005555555 "Asterisk Caller #1" |
22:30.00 | p3nguin | hmm? |
22:30.12 | dustybin | p3nguin: this is exactly what im looking for |
22:30.15 | p3nguin | oh |
22:30.25 | dustybin | test |
22:30.26 | p3nguin | I thought someone suggested astdb earlier. |
22:33.26 | Katty | 20 minutes!! |
22:34.34 | dustybin | adding a entry in the asterisk database didnt change this |
22:34.36 | dustybin | --caller_name="${CALLERID(name)}" |
22:34.50 | dustybin | that still shows the incoming number, not name |
22:37.17 | dustybin | oh i need to change my dialplan |
22:37.18 | dustybin | <PROTECTED> |
22:37.18 | dustybin | <PROTECTED> |
22:37.18 | dustybin | <PROTECTED> |
22:37.21 | dustybin | <PROTECTED> |
22:37.50 | *** join/#asterisk KaneHau (n=KaneHau@133.40.166.155) |
22:38.37 | dustybin | exten => s,n,Set(CALLERID(name)=${IF($["${DB(cidname/${CALLERID(num)})}" = ""]?${CALLERID(name)}:${DB(cidname/${CALLERID(num)})})}) |
22:38.45 | KaneHau | question... I am using BackGround to play a message while waiting for a user to hit any DTMF tone. I don't care what key they hit... is there a way to label an extension so that any key (0-9) goes to the same extension (without using extension 'i')? |
22:38.47 | dustybin | man that is ugly |
22:39.03 | dustybin | tests |
22:39.26 | Katty | KaneHau: yes. |
22:39.34 | KaneHau | wonderful |
22:39.37 | seanbright | _X |
22:39.41 | Katty | ^- |
22:40.00 | seanbright | exten => _X,1,NoOp(they hit ${EXTEN}) |
22:40.07 | dustybin | IT WORKS!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
22:40.32 | dustybin | MY NAME SHOWED ON MYTHTV OSD WHEN I CALLED FROM MY CELLPHONE :D |
22:40.39 | fenrus | :D :D :D |
22:40.42 | fenrus | congratz |
22:40.57 | KaneHau | sean: thank you very much |
22:41.11 | seanbright | yup |
22:41.21 | Katty | gosh. |
22:41.25 | Katty | simmer down now. |
22:41.45 | KaneHau | sean: didn't work... I got this: Channel 'DAHDI/1-1' sent into invalid extension '8' in context 'outgoing', but no invalid handler |
22:42.00 | KaneHau | I used the exact syntax you showed |
22:42.05 | Katty | KaneHau: it did work. |
22:42.17 | Katty | KaneHau: else how would you get 8? |
22:42.25 | KaneHau | I didn't get the NoOp line |
22:42.35 | KaneHau | exten => _X,1,NoOp(they hit ${EXTEN}) |
22:42.41 | KaneHau | never said "they hit 8" |
22:43.03 | KaneHau | oh, I bet my '1' priority should be 'n' |
22:43.09 | Katty | ^_- |
22:43.26 | Katty | i am not following your thought process. |
22:43.34 | Katty | feel free to share some pastebin. |
22:44.19 | KaneHau | no, it wasn't the priority either... guess I'll just use 'i' as the extension |
22:44.27 | *** join/#asterisk JraNil (n=JraNil@85.133.218.30) |
22:44.37 | seanbright | or you could pastebin your extensions.conf |
22:44.41 | seanbright | and we could tell you what is wrong |
22:44.53 | KaneHau | sure, all 4 lines of it... hold on |
22:44.54 | Katty | personally, i'm not a mind reader. |
22:44.57 | Katty | seanbright here might be tho |
22:45.07 | Katty | seanbright: WHAT AM I THINKING |
22:45.21 | seanbright | i have no idea |
22:45.28 | Katty | :< |
22:45.31 | Katty | tacos. |
22:45.35 | Katty | the answer is always tacos. |
22:45.40 | seanbright | noted |
22:45.50 | Qwell | Who was the 32nd President of the United States? |
22:45.52 | Qwell | tacos. |
22:46.10 | KaneHau | sean: http://pastebin.com/m46625b30 |
22:46.23 | Katty | I'm pretty sure mister Roosevelt liked tacos. |
22:46.32 | seanbright | the priority should be 1 |
22:46.35 | seanbright | not n |
22:46.45 | Qwell | and Background doesn't wait |
22:46.45 | KaneHau | I tried '1' - also failed |
22:46.47 | p3nguin | kanehau: Well there's yer problem. |
22:46.47 | seanbright | and make sure you do a 'dialplan reload' after making changes |
22:46.52 | Qwell | It will fall through immediately |
22:47.05 | Katty | WaitExten() <- |
22:47.06 | KaneHau | ahhh, forgot the dialplan reload... thanks |
22:47.09 | seanbright | and there is that as well |
22:47.13 | p3nguin | kanehau: _X doesn't have a 1 priority. |
22:47.22 | Katty | WaitExten(Tacos) |
22:47.23 | seanbright | 17:46 < seanbright> the priority should be 1 |
22:47.25 | seanbright | 17:46 < seanbright> not n |
22:47.59 | sflemming | Hi all, is there someone who uses dahdi and has a problem with the dahdi_dummy timing source? It simply does not tick with my setup. Someone there who can help me? |
22:48.00 | p3nguin | kanehau: And the dialplan CANNOT flow from 's' to _X without a goto or something else to get it there. |
22:48.08 | Katty | exten => _X,1,Goto(Tacos,s,1) |
22:48.27 | KaneHau | ok... it was because I forgot the dialplan reload, thanks! |
22:48.30 | Katty | sflemming: it does not tick? |
22:48.37 | Katty | sflemming: have you replaced the batteries? |
22:48.48 | Qwell | sflemming: what version of dahdi? |
22:48.53 | seanbright | Katty: excellent |
22:48.54 | sflemming | no, when I use dahdi_test it hangs |
22:49.08 | sflemming | dahdi 2.2.1, the same with dahdi_tools |
22:49.21 | KaneHau | p3: I know that... I didn't want it to flow - I just wanted to get it to detect the DTMF tone and go to the _X extension... I need to drop a "hangup"between those |
22:49.23 | Katty | i hear that replacing batteries fixes a lack of ticking. |
22:49.35 | Qwell | sflemming: unload all the modules, load only the dahdi module |
22:49.38 | p3nguin | kanehau: Perfect. |
22:49.40 | Qwell | (not dahdi_dummy) |
22:49.50 | Katty | 10 minutes! |
22:49.55 | Katty | i will survive another day of work! |
22:50.08 | Katty | I WILL SURVIVE |
22:50.12 | Kobaz | katty: http://www.whowantsabalti.com/funpics/mild/squirrel-surprise.jpg |
22:50.14 | sflemming | Which batteries? I had a look into the sources of dahdi_dummy and it seems to use the HPET timer of the kernel, I tried to rise the frequency from 250hz to 1000 but same problem |
22:51.03 | Katty | Kobaz: mmmm, schmirnoff |
22:51.18 | Katty | sflemming: what do you mean which batteries?! |
22:51.34 | sflemming | I tried nearly all loading and unloading scenarios. Have a discussion about the problem since 3 month, there are three people that have the same problem. (All on 64 bit machines) On 32 bit there is no problem with it |
22:51.36 | Katty | Kobaz: what is that other little baggy thing? |
22:51.47 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
22:51.48 | Katty | Kobaz: sugar/salt? |
22:52.02 | citywok | i'm having issues with callerid names, and asked bandwidth.com to help me out, and they asked me this "Can you try and enable the use of Remote-Party-ID or P-Asserted-Identity and give me the call examples of either of those fails?" |
22:52.12 | Qwell | Katty: looks like a chocolate |
22:52.25 | Qwell | chocolate covered cherry perhaps |
22:52.33 | Katty | ooh |
22:52.35 | Katty | aww. |
22:52.36 | sflemming | katty asked if i replaced the batteries, but there are no batteries, okay there is a battery for the rtc but this seems to work correctly |
22:52.49 | seanbright | sflemming: she was making a joke |
22:52.49 | Katty | sflemming: just...nevermind...you didn't get it. |
22:52.57 | dustybin | you guys are clever sods :D |
22:53.01 | dustybin | *gals |
22:53.49 | Katty | citywok: I believe they're talking about something Native to toshiba systems. |
22:53.59 | sflemming | katty: ;-) sometimes this is not easy to recognize when english is not your motherlanguage. nevermind, any ideas? |
22:54.05 | Katty | citywok: what they're asking you for is information off the asterisk CLI that might give them a clue as to what's going on. |
22:54.30 | citywok | so i googled and there is a voip-info document about p-asserted http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header |
22:54.34 | seanbright | no they aren't |
22:55.44 | Qwell | sflemming: scroll up to what I said |
22:56.01 | seanbright | <@Qwell> sflemming: unload all the modules, load only the dahdi module |
22:56.06 | seanbright | <@Qwell> (not dahdi_dummy) |
22:56.12 | seanbright | i've reproduced it here |
22:56.16 | seanbright | heh |
22:56.34 | Qwell | with dahdi_dummy 2.2.1 loaded? |
22:57.21 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
22:57.54 | sflemming | without dahdi dummy i won't have a timing source since my card does not provide it |
22:58.04 | Qwell | sflemming: please do as I asked. :) |
22:58.39 | sflemming | i already did |
22:58.44 | KaneHau | Qwell: You said that "BackGround" doesn't wait... but the Asterisk manual states: Plays the specified audio file(s) while waiting for the user to begin entering DTMF digits. Once the user begins to enter DTMF digits, the playback is terminated. |
22:58.45 | Qwell | now run dahdi_test |
22:59.14 | seanbright | KaneHau: it waits for the sound to finish, but it doesn't wait beyond that |
22:59.25 | Katty | he said, she said. |
22:59.43 | seanbright | KaneHau: if you want to continue waiting after the sound plays, you need to put a WaitExten() after the Background() |
22:59.50 | Katty | i'm going to go back to lurking and trolling about batteries. |
22:59.57 | KaneHau | yes, that's fine... if no DTMF is detected by the time the audio plays, I want to hangup |
23:00.04 | seanbright | then you're done |
23:00.07 | sflemming | Unable to open dahdi interface: No such device or address |
23:00.12 | KaneHau | er, by the end of the audio I meant |
23:00.33 | KaneHau | ok, got it... so it should work as I want |
23:00.39 | seanbright | correct |
23:00.59 | Katty | ohhhhhhheyyy |
23:01.00 | Katty | it's 5 :> |
23:01.03 | Katty | BAI! |
23:01.04 | Katty | poofs |
23:01.05 | KaneHau | (this is to get around the fact that I can't easily detect a PSTN line that has picked up) |
23:01.52 | sflemming | Qwell: what do you suggest now? |
23:02.07 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
23:02.57 | seanbright | sflemming: can you pastevin the output from lsmod? |
23:03.00 | seanbright | pastebin* |
23:03.32 | Qwell | I thought all the timing stuff got pushed into dahdi-base.. maybe that's scheduled for2.3 |
23:04.32 | seanbright | no, it's in 2.2.1 |
23:04.51 | sflemming | here it is http://pastebin.com/d2acb64ba |
23:04.52 | seanbright | oh, but it's off by default |
23:04.57 | p3nguin | I had to install dahdi to get timing so I could use MeetMe(). |
23:05.06 | seanbright | CONFIG_DAHDI_CORE_TIMER |
23:05.11 | Qwell | seanbright: where are you seeing the disabling? |
23:05.24 | seanbright | in the commit message |
23:05.49 | KaneHau | The extension __X doesn't seem to do what I intend... it thinks the pickup of the line is the same as DTMF '1' |
23:05.53 | Qwell | right |
23:05.54 | KaneHau | er _X |
23:06.07 | KaneHau | put using extension 'i' works |
23:06.10 | KaneHau | but |
23:06.12 | KaneHau | damn |
23:06.28 | p3nguin | kanehau: You know that 'i' is the invalid extension, right? |
23:06.30 | seanbright | Qwell: http://svnview.digium.com/svn/dahdi?view=revision&revision=6863 |
23:06.38 | KaneHau | yes |
23:06.51 | KaneHau | I just want to detect ANY tone, I don't care what one |
23:07.02 | KaneHau | it tells me to "go ahead" with the rest of the announcement |
23:07.09 | Qwell | sflemming: How did you install dahdi? |
23:07.12 | seanbright | sflemming: that is lspci. lsmod. |
23:07.22 | sflemming | oh, I'm sorry |
23:08.04 | seanbright | Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] |
23:08.13 | seanbright | why does that sound familiar to me? |
23:08.27 | Qwell | mISDN. it wouldn't be providing timing |
23:08.30 | *** join/#asterisk bmoraca_work (n=bmoraca@66-242-174-254.ceres.bvn.net) |
23:08.34 | sflemming | it is only echo, dahdi, r8169 and forcedeth loaded |
23:08.44 | seanbright | ah. |
23:08.46 | KaneHau | Thanks once again for all your help! Mahalo nui loa! |
23:08.54 | Qwell | sflemming: how was dahdi installed? |
23:09.36 | sflemming | I know that hfc-s does not provide a timer, therefore i would like to use dahdi_dummy. I built an ebuild for gentoo, so it is compiled from source |
23:09.49 | Qwell | walks away |
23:09.57 | *** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil) |
23:10.00 | seanbright | heh |
23:10.37 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
23:10.48 | sflemming | dahdi_scan and dahdi_speed works when all the drivers are loaded, but dahdi_test shows there is no timing |
23:12.36 | p3nguin | I guess KaneHau doesn't care that it works here and I was going to pastebin the dialplan. |
23:14.27 | sflemming | okay, i reloaded everything and now i have a response the very first time. but it looks like it is quite unreliable. sometimes i get a response, sometimes it simply hangs, sometimes i get a segmentation fault |
23:16.15 | *** join/#asterisk krytzz (n=quassel@quassel/user/krytzz) |
23:17.24 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
23:20.08 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
23:20.19 | chilicuil | Hi there, I'm trying to setup a local sip account and it doesnt matter what softphone I use the sound is very low, I've configured the sound in the local machine (with alsamixer) as high as possible but it doesn't change anything I've tried with diferent audio devices and looked at the asterisk log but I see nothing, so right now I'm a bit lost, any ideas? |
23:22.47 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
23:23.14 | krytzz | hi, i am interested in using the fxs ports on the t-com speedport w701v on openwrt, i found no documentation if that is even possible, looking on the board a "cpc5621" chip is there and connected to the fxs ports, and i found the source for a driver, would it possible to use them? |
23:24.46 | Gido-E | the source is a kernel driver? |
23:25.12 | krytzz | yes, but only 300 lines of code |
23:25.25 | Gido-E | ok, normally there is a kernel part, and a asterisk part. |
23:25.31 | krytzz | but i dont know if that is enough |
23:25.38 | bmoraca_work | you'd be way better off in an openwrt channel |
23:25.44 | krytzz | ah ok, sounds reasonable |
23:26.12 | krytzz | ok, thank you, but can you tell me for the asterisk part, i would probably have to write it by myself? |
23:26.29 | Gido-E | check openwrt |
23:26.31 | krytzz | also i am not sure if there are other asics involed with the ports |
23:26.35 | krytzz | ok, thanks :) |
23:27.10 | bmoraca_work | krytzz, that depends how you want to use them...are you trying to run asterisk on the device itself or do you want to use the device as a media gateway? two very different scenarious that will require you to modify two different codebases |
23:27.31 | krytzz | bmoraca_work: i would like to run asterisk on the device |
23:27.41 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:27.50 | bmoraca_work | you'll need to build a dahdi driver for them |
23:28.13 | krytzz | my setup would be: analog phone - speedport - iax to another asterisk |
23:28.23 | krytzz | ok, thanks ill look into the source |
23:33.02 | *** part/#asterisk ruied (n=ruied@bl7-217-61.dsl.telepac.pt) |
23:35.52 | sflemming | does not work but will continue tomorrow, thank you for your help, bye |
23:36.27 | *** join/#asterisk dkirker-openmobl (n=dkirker@openmobl/ceo/dkirker) |
23:39.54 | *** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie) |
23:49.26 | titter | it's alive |
23:49.49 | *** join/#asterisk aidinb (n=Aidin@166.191.98.36) |
23:59.04 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.13.dyn.user.ono.com) |