| 00:01.59 | Waverly360 | Well, since we're talking about fax so much, I have an asterisk question regarding fax detection. |
| 00:02.29 | *** join/#asterisk tzafrir_laptop (i=tzafrir@69-94-204-127.biltmorecomm.com) |
| 00:02.33 | Waverly360 | What would keep asterisk from properly detecting an incoming fax, if 9/10 times it seems to work fine? |
| 00:03.29 | Waverly360 | Also, I have DIDs setup to auto detect faxes for about 3 seconds before moving on to the callee's phone. Most time, asterisk picks up the fax immediately, while other times, it takes longer than that 3 second wait. |
| 00:03.53 | samy_b1 | anyone |
| 00:04.06 | Waverly360 | That will typically cause the user's phone to ring once or twice, before asterisk decides to detect it and pass it onto the fax extension. |
| 00:04.19 | JT | maybe the sending fax machine is not sending the tone out very well? |
| 00:04.36 | Waverly360 | Possibly, but it happens pretty frequently. |
| 00:04.46 | Waverly360 | and even with the same fax machine over and over, the results vary. |
| 00:05.30 | *** join/#asterisk ChkDigit (n=mrw@static24-72-71-175.regina.accesscomm.ca) |
| 00:09.53 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:10.18 | Waverly360 | Anyways, it's late, and I need to head home. Maybe not think about this crap tonight. |
| 00:10.23 | Waverly360 | I'll talk to you all soon. |
| 00:11.06 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 00:11.11 | Mahmoud | hi folks |
| 00:11.37 | Mahmoud | is there any SIP-to-IAX tunneling online service? |
| 00:16.26 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
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| 00:27.29 | *** join/#asterisk vAd0r (n=IceChat7@65.67.210.121) |
| 00:27.33 | vAd0r | can someone help me with a one way sound issue. I have pfsense |
| 00:31.18 | vAd0r | anyone here |
| 00:31.37 | samy_b1 | sure man |
| 00:31.41 | vAd0r | can someone help me with a one way sound issue. I have pfsense |
| 00:31.48 | samy_b1 | vAd0r |
| 00:31.55 | vAd0r | yeah |
| 00:32.10 | samy_b1 | taht can happend due to unsupproted codecs |
| 00:32.27 | samy_b1 | or |
| 00:32.32 | vAd0r | i have a sip connection for my friend to connect to my house and im at my house |
| 00:32.35 | samy_b1 | rtp port fitrating |
| 00:32.38 | vAd0r | he can hear me but i cacn't hear him |
| 00:32.51 | samy_b1 | do you have a firewall |
| 00:32.54 | vAd0r | he couldn't even connect until i added this to my sip.conf |
| 00:33.17 | vAd0r | domain=mypubip,from-internal |
| 00:33.18 | samy_b1 | what |
| 00:33.29 | vAd0r | once i added that he was able to register |
| 00:34.06 | samy_b1 | humm |
| 00:34.12 | samy_b1 | i don't know |
| 00:36.25 | *** join/#asterisk xpotx (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 00:37.36 | mightnare | vAd0r: did you allow udp ports 10000-20000 on pfsense? |
| 00:37.45 | vAd0r | yeah |
| 00:37.58 | vAd0r | i think i might have just got it fixed |
| 00:38.04 | vAd0r | i just want to test 2 people from outside my house calling each other |
| 00:38.10 | vAd0r | any volunteers? |
| 00:38.15 | vAd0r | : ) |
| 00:40.34 | aptura | vAd0r just set up some sip clients to register back though your public ip adress. |
| 00:41.19 | JT | that's not a good test if they're on the internal lan |
| 00:42.31 | aptura | Jt mabey I was asuming he was talking about external clients loging in. |
| 00:42.52 | JT | i think that's what he's trying to do |
| 00:42.55 | vAd0r | I am internal and can talk to one external person fine now. i got that fixed |
| 00:43.06 | vAd0r | i want to have 2 external people talk to each other |
| 00:43.27 | vAd0r | not a good to test for me to do it to someone |
| 00:43.42 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 00:43.51 | sharp | join #gnuradio |
| 00:43.55 | sharp | crap |
| 00:44.34 | aptura | frack thaty buffer sharp! |
| 00:44.46 | aptura | :) |
| 01:00.25 | [TK]D-Fender | ~sipnat |
| 01:00.41 | jbot | i heard sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
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| 01:18.51 | *** join/#asterisk ronnocol (n=lance@bonobo.monkeymental.com) |
| 01:20.06 | ronnocol | I have what I hope is a quick question. I want to setup something to allow people to call in on a POTS line, leave a voicemail, and then have the box send email/call a specified # to alert me that a message has been left |
| 01:20.35 | *** join/#asterisk cody989887 (n=cody@monitor.itfreedom.com) |
| 01:20.46 | ronnocol | Can Asterisk do this? If so, what is the minimum h/w I need. I don't need any VoIP phones... just essentially an intelligent answering machine |
| 01:21.44 | Nugget | asterisk can certainly do that, but the POTS interface will either be expensive or crappy. |
| 01:21.47 | cody989887 | I just signed on with telasip, and after some time, I emailed their support because I couldn't dial local numbers. They say they don't support local numbers, but I can configure asterisk to do this |
| 01:22.09 | cody989887 | exten => _X.,1,Dial(SIP/telasip-gw/${EXTEN}), this is my outbound line |
| 01:22.38 | ronnocol | Nugget: Crappy in that incoming voice quality will be bad, or outbound? |
| 01:23.05 | ronnocol | Basically I'm looking for a way for customers to call and say "My site is down, trouble ticket xxxx" |
| 01:23.19 | Nugget | you'll spend your days chasing weird echo problems which are impossible to isolate |
| 01:24.37 | ronnocol | Nugget: Actually, I'd proably do ok, if the system recognizes there is a high priority ticket in the queue, and just called to alert me to that. Then the customers wouldn't even have to leave a v/m |
| 01:24.49 | Nugget | something like a sipura spa-3000 would be a pretty decent solution |
| 01:25.16 | Nugget | don't be tempted by the $9 "clone x100p" cards you see on ebay |
| 01:25.44 | rudholm | I've got a 3000 on ebay no one has bid on yet :) |
| 01:26.01 | ronnocol | how much am I looking at for that? |
| 01:26.04 | Nugget | why not just have your ticket system send you an SMS? that requires no hardware and no asterisk at all. |
| 01:26.23 | rudholm | starting price is 45$, BuyItNow is 65$. |
| 01:26.29 | ronnocol | to easy to sleep through SMS |
| 01:28.02 | [TK]D-Fender | ronnocol, $75 for a Linksys SPA-3102, or $150 for a Digium TDM01B (TDM400P + 1 FXO module) |
| 01:29.52 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:29.54 | tengulre | hi,all |
| 01:30.15 | tengulre | how to using rxgain and txgain cancel the echo in Zap channel? |
| 01:30.27 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 01:31.10 | Nugget | The TDM400P is a pain in the ass. |
| 01:31.11 | [TK]D-Fender | tengulre, go lookup "fxotune" on the WIKI |
| 01:35.30 | tengulre | [TK]D-Fender: tks! but when I running this command, it's display /dev/zap/1 absent: Device or resource busy, is it right? |
| 01:35.38 | tengulre | fxotune -i 4 |
| 01:35.59 | [TK]D-Fender | tengulre, maybe you shouldn't be running * while doing this... |
| 01:36.04 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 01:37.18 | *** join/#asterisk Cresl1n (n=matt@65-182-39-144.cre.bil.biltmorecommunications.net) |
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| 01:37.42 | tzafrir_laptop | tengulre, is that an fxo? |
| 01:37.52 | tengulre | tzafrir_laptop: yes |
| 01:38.02 | tzafrir_laptop | are you in the US? |
| 01:38.25 | tengulre | tzafrir_laptop: no, I m in CHINA! |
| 01:38.41 | tzafrir_laptop | look at the opermode parameter of wctdm first |
| 01:39.00 | tengulre | tzafrir_laptop: how to look it? |
| 01:39.26 | tengulre | in header file? |
| 01:40.05 | tzafrir_laptop | options wctdm opermode=CHINA |
| 01:40.11 | tzafrir_laptop | or something of that sort |
| 01:41.10 | tengulre | tzafrir_laptop: your mean is running modprobe wctdm opermod=CHINA?? |
| 01:41.12 | tzafrir_laptop | what distro do you use? |
| 01:41.58 | tengulre | asterisk-1.4.4 and zaptel-1.4.4 |
| 01:42.37 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:43.14 | tengulre | grep China * ,,, I got zonedata.c: { 33, "cn", "China", { 1000, 4000 }, ?? |
| 01:44.07 | tengulre | is it modprobe wctdm opermode=cn? |
| 01:45.10 | tengulre | or setting loadzone=cn in /etc/zaptel.conf? |
| 01:45.15 | tzafrir_laptop | no, that relates to the dialnone. Look in wctdm.c itself... |
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| 02:43.32 | *** join/#asterisk mtc (n=mtc@f049171151.adsl.alicedsl.de) |
| 02:43.35 | mtc | hallo all |
| 02:44.55 | mtc | I'm using asterik 1.4.4 <-- iaxmodem 0.21 ->> hylafax 4.3.1 |
| 02:46.09 | mtc | when I try to send a fax I get "Adjusting skew to -XXXXX" |
| 02:46.55 | mtc | I get it when I try to send to dus.net or even local to ananother asterisk server |
| 02:47.31 | mtc | We are trying to write an Manual, that is going to be printed soon |
| 02:47.59 | ez` | any one got a real good usb phone |
| 02:48.13 | ez` | all usb phone seem be crap |
| 02:48.49 | [TK]D-Fender | ez`, the first step is admitting you have a problem :) |
| 02:49.07 | [TK]D-Fender | ez`, you are on the path to buying quality equipment! |
| 02:49.16 | ez` | hehe ; cool |
| 02:50.58 | [TK]D-Fender | (translation : USB phones SUCK. Soft-phones SUCK. Together they are crapTASTIC!) |
| 02:50.59 | ez` | i am so impress about polycom ; i want something similar ; usb ;) |
| 02:50.59 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 02:50.59 | ez` | we use softphone a work ; on a 911 central ; and to tell you the true ; soft phone is not alway good ;) |
| 02:51.32 | *** join/#asterisk monstertruck (n=monstert@c-75-74-251-82.hsd1.fl.comcast.net) |
| 02:51.36 | nick125_lappy | does anyone here know of a guide (or something) that would show some estimate prices for a voice t1? (just ballpark ranges, really) |
| 02:51.41 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:51.50 | mtc | what does "Adjusting skew to -XXXXX" mean? |
| 02:52.05 | coppice | a softphone is inherently no better or worse than any other phone. in fact, all IP phones are soft phones. its only the package which varies. The key thing is most VoIP products are crap |
| 02:52.16 | mtc | I get it hundres of time and the fax does not reach |
| 02:52.41 | monstertruck | does anyone know what the technical term for "terminate voip traffic into an international call via pstn" is? |
| 02:53.24 | JT | nick125_lappy: it varies a lot in the US depending on where you are |
| 02:53.33 | JT | nick125_lappy: what you really want is a PRI ISDN T1 |
| 02:54.10 | JT | coppice: i disagree, good sip hardphones are way better than most softphones currently available |
| 02:54.38 | coppice | I can't actually think of a "good" hardphone |
| 02:54.42 | nick125_lappy | JT: What I'm looking for is just some basic prices on a 24 channel T1 line without having to contact a provider and get a bunch of sales bullcrap |
| 02:54.44 | ez` | coppice, try polycom and cisco hardphone; |
| 02:54.58 | mtc | is the a german asterisk channel? |
| 02:55.11 | coppice | polycoms are crap. ciscos are not much better. people have such very low standards |
| 02:55.20 | JT | nick125_lappy: 23 channels for PRI |
| 02:55.43 | nick125_lappy | JT: Yeah, I forgot about that. |
| 02:56.01 | JT | nick125_lappy: well i've heard it varies from a couple of hundred to thousands a month in the US, depending on location and exactly what type of service it is |
| 02:56.43 | ez` | coppice, i think you got a other problem ;) |
| 02:57.08 | coppice | the biggest problem with softphones is people use 50 cent headsets and complain about the softphone |
| 02:57.33 | [TK]D-Fender | coppice, well then... what DOES qualify as "good" for you? |
| 02:57.57 | ez` | coppice, try a rope + 2x steel can ; is better |
| 02:58.00 | JT | even so, they lack noise reduction and echo cancellation, and you need to set all the softphone levels just right |
| 02:58.25 | coppice | if you switch from G.711 to G.722 with a good phone the difference is *huge*. try it with most IP phones, and the difference is small |
| 02:59.20 | ez` | the most impressive thing i try lastly : softecho cancel from octasic with a cheap x100p ; very very spectacular |
| 02:59.25 | nick125_lappy | JT: Do you know of any providers that I can get a quote from (maybe just based on CO, I don't want to give out a bunch of information and have them calling day and night) and get a ballpark range from? |
| 02:59.45 | JT | ez`: HPEC or something bought direct from octastic? |
| 03:00.04 | mtc | can anyout help me out with hylafax? |
| 03:00.05 | JT | nick125_lappy: dunno, i'm not an american |
| 03:00.21 | coppice | ez: exactly. get the soft setup right, and it can work really well. there is nothing inherently wrong with a softphone. most hardphones have crappy handsets, though, with a lot of problems |
| 03:00.26 | [TK]D-Fender | coppice, so again, highlight a few "good" products for us. Any good service to reach the "real world" with them either? |
| 03:00.27 | ez` | they got a softecho cancel ; it really really good ; |
| 03:00.55 | JT | the point is, the soft setup is almost never "right" :) |
| 03:00.59 | JT | for softphones |
| 03:01.05 | ez` | coppice, are you satisfy from yourselft sometime ^ |
| 03:02.21 | coppice | there are no good IP hardphones. customers don't provide the incentive for makers to care about voice quality. nokia didn't get so big by good voice quality. they used to have the worst voice quality of any cellphone, though they have improved recently. voice quality doesn't seem to sell |
| 03:02.34 | ez` | octasic really worth to try it ; i highly suggest it |
| 03:02.34 | mtc | hmmm, what do you use instead of hylafax? |
| 03:02.52 | mtc | it seems like no one of you is using it!? |
| 03:02.59 | coppice | where did you buy the octasic software? they don't seem to be promoting it well |
| 03:03.12 | ez` | any cdma or gsm seem to be poor quality sound coppice |
| 03:03.47 | coppice | most GSM phones sound *far* worse than the system is capable of |
| 03:03.58 | [TK]D-Fender | coppice, Ok, so far EVERYTHING is bad in your books, so we're down to degrees of "bad" with nothing standing out in your view? |
| 03:04.23 | coppice | pretty much. like i said quality doesn't see, so people don't provide it |
| 03:04.39 | nick125_lappy | Anyone here use asterisk realtime? I'm thinking of maybe sharing my current VoIP account (it's a pay-per-minute account, so, no worries) with a few friends, but, I don't want my configs to get ugly when I add even more users. |
| 03:04.40 | ez` | coppice, right i think same about this ; they dont really promot it ; |
| 03:05.00 | coppice | I was amazed how good we got GSM half rate coding when I worked on that. I've never heard a phone sound as good as our test setups |
| 03:05.13 | JT | ez`: so where did you buy it and how much? |
| 03:05.48 | ez` | octasic mainly built echo cancel hardwafre to big telco; 1000 call concurent ; but recenlty they starteda new brnach |
| 03:06.06 | ManxPower | coppice: So what happened when it went to production? |
| 03:06.10 | coppice | mtc: what do you want to know about hylafax? |
| 03:06.22 | JT | ez`: yes, that's obvious, the digium and sanngoma cards with HWEC both have octastic chips i believe |
| 03:06.29 | JT | ez`: so how about answering the question? |
| 03:06.39 | ManxPower | ez`: once I got over The Bug From Hell, the HPEC worked pretty good. |
| 03:06.44 | ez` | right jt; i am looking for the url hehe |
| 03:06.46 | coppice | i keep telling you. nobody cares. connect a good codec to disgusting mic and earpiece amps and it sounds bad |
| 03:06.51 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:07.10 | ManxPower | coppice: Ah. So nothing technical, 9-) |
| 03:07.17 | mtc | coppice, I have configured it but all the time I want to send a fax I get "Asjusting skew to XXXX" |
| 03:07.28 | ez` | www.octasic.com/en/index.php |
| 03:07.30 | mtc | coppice, the fax never get through |
| 03:08.05 | mtc | coppice, I've tried it with dus.net and a local asterisk installation -> the same |
| 03:08.14 | coppice | ez: that doesn't tell you where to buy, or the price |
| 03:08.19 | ez` | JT, i really recommend their product ; its auto tunning ; i worht to try with a crappy x100p or any clone card fxo |
| 03:08.25 | mtc | I just don't know what I'm doing wrong |
| 03:08.34 | ez` | 10 candian dollars per channel |
| 03:08.37 | JT | ez`: ok, i don't use X100ps though |
| 03:08.49 | ez` | tdm400 work too |
| 03:09.08 | JT | i avoid analogue if i can |
| 03:09.09 | ez` | i dont know every supported hardware |
| 03:09.21 | ez` | JT, ofcoure it better to avoid |
| 03:09.53 | JT | avoiding analogue doesn't avoid echo though :/ |
| 03:10.03 | ez` | but for home use ; and if you got 16 or less pots ; its crystal clear voice |
| 03:10.19 | ez` | yeah ; i know hehe |
| 03:10.50 | ez` | octasic can be use with softphone; but i did not test it |
| 03:11.11 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 03:11.11 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:12.11 | Strom_C | it's the Qwell |
| 03:12.19 | ez` | i beleive xlite is one of good sip softphone; do you agree ? |
| 03:12.31 | mtc | coppice, you are still there? |
| 03:12.41 | nick125_lappy | I personally hate xlite, but, it seems a tiny bit better than the rest. |
| 03:12.42 | coppice | no |
| 03:12.46 | nick125_lappy | (least on linux) |
| 03:13.03 | ez` | maybe he put he laptop in garbage because is all crap ;) |
| 03:13.20 | ez` | nick125_lappy, yes; better ... |
| 03:13.35 | ez` | also multi platform ; |
| 03:13.47 | coppice | xlite illustrates what sells in the phone industry prefectly. how much effort went into the pretty skin, and how much into the code? |
| 03:13.57 | ez` | idefisk seem creata lot of echo ... |
| 03:14.11 | nick125_lappy | 90% skin, 9% code, 1% hot-air. |
| 03:14.19 | ez` | heh |
| 03:14.30 | coppice | i think you are a bit light on the hot air there |
| 03:14.44 | ez` | coppice, do you have a phone in your home ? which one |
| 03:15.02 | coppice | a $2 phone on a POTS line :-) |
| 03:15.07 | ManxPower | coppice: I think X-Lite is an example of everything a softphone should NOT be. |
| 03:15.08 | nick125_lappy | I've been pretty happy with my PAP2, I don't have experience with any other devices though |
| 03:15.17 | ez` | dollarama ; 1$ store |
| 03:15.26 | coppice | actually, no. its an $8 SMS phone the phone company gave us for free |
| 03:15.29 | ez` | ManxPower, why ^ |
| 03:15.48 | JunK-Y | coppice: 8$!!! u disapoint me now! :P |
| 03:15.57 | ManxPower | ez`: For one thing it's config system will drive you batty. |
| 03:15.58 | ez` | my sipura 3000 drive me crazy ; very unstable ... echo |
| 03:16.13 | JunK-Y | ez`: go get a pap2. |
| 03:16.15 | coppice | and it sounds better than a high end cisco. sad, isn't it |
| 03:17.04 | ez` | ManxPower, i agree |
| 03:17.31 | JunK-Y | sip is much more then only phone. |
| 03:18.11 | ez` | i mostly use iax on trunk and sip to hardphone ,, |
| 03:18.18 | *** join/#asterisk booray (n=ray@adsl-75-7-233-157.dsl.irvnca.sbcglobal.net) |
| 03:18.55 | JunK-Y | ez`: are ya comin to meeting, friday? |
| 03:19.15 | ez` | hope so ; but alone ; |
| 03:19.23 | JunK-Y | no gf this time? |
| 03:19.39 | JunK-Y | too bad, no gang-bang |
| 03:20.03 | ez` | maybe not ; or my gf and my 2 princess will be in a hotel with a pool ; this mean 300$ a night ;) |
| 03:20.09 | ez` | yes |
| 03:20.33 | JunK-Y | are they learning * yet =) |
| 03:21.29 | ez` | hehe not yet ; but they got sip phone in their room ; monitored ;) and with filter .,.. +/- 3 number on outbound ... |
| 03:21.52 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-104-76-82.pskn.east.verizon.net) |
| 03:21.58 | ez` | i try to tech her install linux ;) |
| 03:22.04 | ez` | teach |
| 03:22.39 | ez` | they use linux every day ;) |
| 03:22.40 | JunK-Y | 976 number! |
| 03:22.57 | ez` | i will add it l8tr |
| 03:23.21 | JunK-Y | they will can earn some money :P |
| 03:23.22 | *** join/#asterisk bbryant (i=Brett@65-182-39-131.cre.bil.biltmorecommunications.net) |
| 03:23.43 | ez` | i would like be able to draw power from usb or pcmcia to inject power to my polycom ;;; i am testing it ,,, |
| 03:23.53 | ez` | hah |
| 03:24.26 | ez` | usb seem be 5 volt ans my polycom ip 300 need 12v 400ma |
| 03:24.52 | ez` | i could plug x2 usb to my polycom ; |
| 03:26.42 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
| 03:26.43 | ez` | JunK-Y, you gf ask if she can sleep one more day here ; do you mind to ... ? |
| 03:26.53 | ez` | you = your |
| 03:27.06 | *** join/#asterisk Cresl1n (n=matt@65-182-39-144.cre.bil.biltmorecommunications.net) |
| 03:27.06 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 03:27.19 | JunK-Y | no, keep her when ur is under my desk... |
| 03:28.34 | ez` | k |
| 03:29.36 | ez` | are you w8ing a lot of people frindya %^? |
| 03:29.38 | JT | ez`: xlite, in terms of softphones, is ok for testing, but the audio is terrible compared to a polycom |
| 03:29.41 | JT | same for idefisk |
| 03:30.34 | JunK-Y | ez`: more then the next one i think. |
| 03:30.55 | ez` | JT, i agree; nothing beat polycom hardphone ... to me experience |
| 03:31.15 | ez` | idefisk is really lower to xlite ;( |
| 03:31.21 | nick125_lappy | Testing is fine, but, if you are going to do anything more than that, get a TA or even a cheap IP phone (just not a grandstream) |
| 03:32.20 | ez` | i tested aastra; well some one bougth alot of them ans i been desapoint using sip with them ,,,, |
| 03:32.37 | JT | english... |
| 03:32.50 | ez` | xd |
| 03:32.58 | JT | .,,..,..@@%@@ |
| 03:33.02 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:33.06 | ez` | heh |
| 03:33.06 | coppice | mtc: that skew message seems to indicate one of you machines is running at a significantly different speed from the other |
| 03:33.39 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 03:33.43 | `Sean | ez` i like my Cisco 7970 :D |
| 03:33.46 | ez` | JT, have you try any good usb phone ? |
| 03:33.50 | `Sean | i fell in love with that phone first time i saw it |
| 03:34.18 | coppice | Sean: you get lots of exercise lifting the handset on that one :-) |
| 03:34.20 | Qwell | ez`: usb "phones" are just sound cards |
| 03:34.36 | JT | ez`: usb phones are gimmicks |
| 03:34.43 | `Sean | coppice it doesn't seem heavy to me i liked how its curved tho |
| 03:34.49 | ez` | we use cisco at work ; but i did not test with asterisk ; i need to get somewhere sip stuff to convert it using tftp... i would borrow one form work ; we got 7940 |
| 03:34.53 | `Sean | its like perfectly made for people |
| 03:34.56 | JT | ez`: they are, as Qwell said, a phone, speaker and mic, and keypad |
| 03:34.56 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:35.05 | `Sean | like it curves around where youre chin is at etc so then you can talk more clearly |
| 03:35.11 | `Sean | unlike traditional phones |
| 03:35.24 | coppice | Sean: i rather hate that handset. on a long call it really pisses me off |
| 03:35.42 | `Sean | hrmp possible |
| 03:35.45 | Strom_M | `Sean: define "traditional" |
| 03:35.48 | `Sean | coppice what phone do you like |
| 03:35.51 | ez` | jt yeah i know ; every i try they was a gay sound card ; this not a autonom ip phone; i juste wondering if there is a real usb ip phone .... |
| 03:36.08 | coppice | I never met an IP phone I liked |
| 03:36.16 | `Sean | Strom_M like those umm vista 300 phones telcos used to sell and well any traiditonal anologue phone |
| 03:36.25 | `Sean | like the astra business ones perhaps |
| 03:36.26 | Strom_M | what the hell is a "vista 300"? |
| 03:36.30 | Strom_M | no no and no |
| 03:36.35 | ez` | coppice, : polycom , cisco give real great result |
| 03:36.40 | Strom_M | 1980s != traditional |
| 03:36.46 | coppice | I think people complain too much about grandstream, as nothing else is a whole lot better. there are *far* more usable traditional phones |
| 03:36.54 | `Sean | coppice |
| 03:36.58 | `Sean | but i like polycom too |
| 03:37.02 | JT | ez`: why would there be? that makes no sense, USB is not a networking standard |
| 03:37.02 | `Sean | like the soundstations :D |
| 03:37.39 | coppice | the soundstations are very poorly designed. there are far better products which never took off |
| 03:37.43 | ez` | JT, i got usb ethercard ; so why not real usb ip phone ? |
| 03:37.51 | Strom_M | `Sean: http://www.msu.edu/~atchiso5/302_files/image002.jpg http://www.oldtimephones.com/images/250_No._500.jpg |
| 03:38.04 | Strom_M | that's traditional :) |
| 03:38.10 | `Sean | no thats OLDTIME |
| 03:38.11 | `Sean | :P |
| 03:38.16 | Strom_M | doofus |
| 03:38.25 | JT | ez`: you are joking, right? |
| 03:38.30 | ez` | JT, nah |
| 03:38.32 | nick125_lappy | Anyone here know of a decent python AGI library? |
| 03:38.33 | coppice | oldtime is a candle stick phone |
| 03:38.36 | `Sean | hrmp i love cisco phones tho |
| 03:38.36 | `Sean | :P |
| 03:38.37 | JT | ez`: the ETHERNET side of the usb network adapter is networking |
| 03:38.47 | JT | usb is just a toy bus computers use |
| 03:38.48 | coppice | and for really old time the candle has a wick |
| 03:38.51 | JT | networks don't use usb |
| 03:39.29 | ez` | JT, i mean ; i got a small usb adapter ; i plug it and it give me a ethernet port |
| 03:39.40 | coppice | USB is not a toy. it solves a different problem from ethernet |
| 03:40.25 | JT | ez`: i know that, where on earth did i dispute that? |
| 03:40.30 | coppice | people think of USB as a toy, because of a long history of screwed up implementations |
| 03:40.50 | JT | ez`: i will say it again, USB is NOT a networking standard, it's for peripheral interconnect |
| 03:40.57 | JT | IP phones are network devices |
| 03:41.41 | coppice | a few USB phones connected to an NSLU2 could make a nice little small office PBX |
| 03:41.43 | ez` | eheh k ; i thought you was know knowing this crappy gadjet |
| 03:42.21 | ez` | JT, also this adapter dont give good result ; so .... |
| 03:42.24 | *** join/#asterisk juliusspencer (n=julius@60-234-132-23.bitstream.orcon.net.nz) |
| 03:42.35 | JT | ez`: "know knowing" |
| 03:42.36 | ez` | i know phone are network device ; |
| 03:43.26 | JT | coppice: sure, as long as they're not more than a few metres away from said NSLU2 :) |
| 03:44.06 | coppice | I did say small office. say, a little branch place with everyone in one room |
| 03:44.33 | coppice | a *lot* of offices are like that |
| 03:44.51 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 03:45.30 | coppice | have they stopped making the NSLU2? |
| 03:46.09 | ez` | i got to install about 110-130 polycom on 1 pbx next week; ; all g711 all sip .. any suggest power you recommend ? i dont know whats the maximum concurent call this busness do ... |
| 03:46.33 | ez` | cpu ; ram ... |
| 03:46.57 | coppice | power? allow 10W a phone, so that's about a kilowatt |
| 03:47.03 | ez` | i use to installl smaller 50-60 ... people |
| 03:47.55 | ez` | coppice, horsepower exacly |
| 03:48.07 | JT | 10W on startup perhaps, they're only meant to take up to 5W when running |
| 03:48.37 | coppice | you are ignoring power supply losses |
| 03:48.50 | JT | which power supply? |
| 03:48.58 | ez` | so 8w is good ? |
| 03:49.17 | ez` | heh |
| 03:50.07 | *** join/#asterisk kieranmullen2 (i=user@71.245.97.59) |
| 03:50.32 | coppice | it puzzles me why these phones take so much. they should be <1W |
| 03:50.34 | kieranmullen2 | Is there a hidden option other than * to mute all other users in a conference? |
| 03:50.51 | juliusspencer | hi, does anyone here know if any software other than OS needs to beinstalled for the asterisk card to show up with lspci? |
| 03:51.01 | JT | coppice: polycoms are rated at 5W max |
| 03:51.09 | JT | but greater efficiency would be nice |
| 03:52.09 | coppice | I spend much of my time working on things consumming 5mW, so I think 5W is *huge* |
| 03:53.18 | JT | imho speakerphone and ringing should draw the most |
| 03:54.19 | coppice | ringing might go up to a watt. a speakerphone doesn't take much |
| 04:09.24 | *** join/#asterisk bbryant (i=Brett@65-182-39-131.cre.bil.biltmorecommunications.net) |
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| 04:24.41 | rudholm | JT: we can power the phones with water! |
| 04:25.38 | JT | argh |
| 04:26.14 | sky_net | l |
| 04:26.20 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 04:26.43 | coppice | rudholm: in brazil you probably would |
| 04:27.53 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 04:28.01 | plasmid | how do I transfer a ringing extension to mine? in other words, if I hear an extension ringing but no one picks it up... can I pick up that extension from MY extension phone? |
| 04:28.04 | plasmid | sales and whatnot. |
| 04:28.46 | coppice | is the whatnot department anywhere near the overheads department? |
| 04:28.50 | rudholm | coppice: I'm just reminding JT of a conversation we had here yesterday with some moron who believed in a perpetual motion machine. |
| 04:29.11 | plasmid | well.. i could always run and pick up that extension phone. |
| 04:29.44 | coppice | rudholm: next thing you'll have people believing in hydrogen as a source of non-nuclear energy :-) |
| 04:30.25 | rudholm | coppice: you're pretty close to what he was saying. he seemed to think you could electrolyze water and yield a net positive |
| 04:30.49 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 04:31.00 | coppice | man, that guy is ahead of the game. invest in him |
| 04:31.06 | rudholm | hahaha |
| 04:31.07 | rudholm | yeah |
| 04:32.14 | coppice | I think powering brazilian IP phones by the direct use of water might be a little more effective |
| 04:32.25 | kaldemar | plasmid: take a look at pickup groups. |
| 04:32.56 | JT | rudholm: and you know, liquid hydrogen takes hardly any power to make ;) |
| 04:34.20 | nick125_lappy | Is there any kind of way to get debugging information from AGI? (I need to see what variables my script is getting, what it is returning, etc)? |
| 04:34.38 | plasmid | kaldemar, err... pickup groups? |
| 04:34.47 | JT | plasmid: google :) |
| 04:34.53 | JT | ~thewiki |
| 04:35.14 | jbot | well, thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 04:35.15 | coppice | JT: as long as you don't need it very pure, the sea is full of it |
| 04:35.15 | JT | coppice: heh |
| 04:43.01 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 04:47.46 | *** part/#asterisk kieranmullen2 (i=user@71.245.97.59) |
| 04:57.29 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 05:02.48 | rue_mohr | is there a new signalling definition is * for newbridge? |
| 05:02.48 | rue_mohr | based on koolstart? |
| 05:02.48 | rue_mohr | er kewlstart |
| 05:04.18 | plasmid | ok.. pickupgroup i put a 1 for extesions 100 and 200. But when i call extension 100 (on my cell) and try to pickup with extension 200 (by pressing *8) nothing happens... what did I do wrong? |
| 05:06.29 | kaldemar | plasmid: have you defined a callgroup also? |
| 05:06.41 | plasmid | ahh shit. |
| 05:07.15 | *** join/#asterisk simonr (n=simonr@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com) |
| 05:08.56 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 05:12.29 | plasmid | callgroup is now set to "2" (without the quotes) but I still get zilch... well.. a busy signal. Hmm. |
| 05:13.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:16.37 | kaldemar | plasmid: if you have 2 as the callgroup for the extensions, then extensions in pickup group 2 can pick up calls to them. IIRC. |
| 05:18.53 | plasmid | yes.. it's odd though I should be able to pick up whoever is calling extension 100 by pressing *8 on my 200 extension but i get silence. |
| 05:24.15 | stoffell | plasmid, if pickup and callgroup are both 2, what does the CLI indicate? |
| 05:24.40 | plasmid | checking cli... |
| 05:25.39 | *** join/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 05:26.37 | kraptv | I'm trying to figure out why I can get my Cisco 7940 to register just fine with trixbox but it doesn't want to register with normal plain asterisk. I keep getting a 401 Unauthorized. |
| 05:26.58 | kraptv | But the secret information should be fine. |
| 05:27.16 | stoffell | kraptv, try sip debug ip <phone's ip> |
| 05:28.20 | *** join/#asterisk hypa7ia (i=hypatia@judecca.aculei.net) |
| 05:28.28 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:29.33 | kraptv | I'm getting a bunch of things that look like this: http://pastebin.ca/506772 |
| 05:30.42 | kraptv | the system running asterisk is 192.168.0.99. |
| 05:31.42 | kaldemar | does the phone answer the 401 with the challenge response? |
| 05:32.24 | drrt | kraptv, paste section of your sip.conf for actual phone |
| 05:33.09 | kraptv | mind you, I've been hacking it over and over quite a bit... ;-) |
| 05:33.48 | kraptv | http://pastebin.ca/506779 |
| 05:33.51 | kraptv | updated. |
| 05:34.02 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 05:34.28 | kraptv | I guess I could turn off nat=yes effectively as it is on the same net... |
| 05:35.28 | adorah | Hi has anyone experienced with connecting asterisk to a legacy PBX? |
| 05:37.22 | JT | adorah: yes |
| 05:37.48 | drrt | kraptv, drop username because it`s useful for peer not for user. which distro do u use? |
| 05:38.18 | adorah | JT: if u connect asterisk to pbx extension ports, I understand one has to use fxo on * side right? |
| 05:38.39 | JT | adorah: depends what pbx ports you connect it to |
| 05:38.45 | kraptv | drrt: I'm using Asterisk 1.4.4 on CentOS 5 i386. |
| 05:39.34 | drrt | kraptv, try to use prebuild rpm package for 1st time and rebuild your conf |
| 05:39.37 | adorah | JT well it is an old pbx but has a few free regular extension ports |
| 05:40.15 | kraptv | drrt: I built from source this time, unfortunately, it seems as though there's no asterisk 1.4.4 rpm's?? maybe I was a bad searcher. |
| 05:40.36 | adorah | maker is Telrad keyBX I know it was sold in the usa but not sure how it was branded there |
| 05:41.31 | drrt | kraptv, did you try fedora`s project repository ? |
| 05:41.36 | adorah | JT My sense tells me that if I have fxs ports on pbx's side I should use fxo on * side.. |
| 05:41.51 | adorah | and VV |
| 05:42.38 | drrt | adorah, sure. u cant connect two fxs ports with |
| 05:43.22 | *** join/#asterisk tenzind (n=tenzind@202.144.144.77) |
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| 05:47.24 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) [NETSPLIT VICTIM] |
| 05:47.32 | kraptv | drrt, no go. looking all over, I get no asterisk 1.4, and the make rpm script within source is not so good. |
| 05:55.20 | *** join/#asterisk codefreeze (n=steve_mu@65-182-39-254.cre.bil.biltmorecommunications.net) [NETSPLIT VICTIM] |
| 05:57.02 | *** join/#asterisk mosty (n=mostyn@203-206-139-58.perm.iinet.net.au) |
| 05:57.29 | mosty | i'm trying to setup a telstra onramp30 connection with a te412p, the port is lit green |
| 05:57.38 | JT | ok |
| 05:57.43 | mosty | "Primary D-Channel on span 1 down" is what it says when i dial |
| 05:57.50 | JT | telstra, getting ripped off ;) |
| 05:57.54 | drrt | kraptv, remake your conf file with minimal options you need. will be much easy to find a source |
| 05:57.55 | JT | heh |
| 05:58.00 | mosty | jt, any tips? |
| 05:58.09 | JT | pastebin.ca zapata.conf and zaptel.conf |
| 05:58.19 | kraptv | drrt: alrighty... I suppose I'll empty it out. |
| 05:58.24 | *** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
| 05:59.21 | mosty | http://pastebin.ca/506799 |
| 05:59.34 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:00.06 | JT | mosty: are the other spans in use? |
| 06:00.17 | mosty | jt: i have them commented out |
| 06:00.27 | mosty | http://pastebin.ca/506800 this has the console output |
| 06:00.33 | nick125_lappy | Does anyone here have any experience with realtime? Good, bad, any different |
| 06:01.15 | JT | span=1,2,0 should be span=1,1,0 then |
| 06:01.17 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:01.26 | webman | I'm trying to setup a stun server, but it seems to want two IP addresses, does anyone know what these addresses are used for?? (there is no documentation in my package, and google doesn't seem to have much either) |
| 06:01.39 | JT | you must set the switchtype too |
| 06:01.55 | nick125_lappy | I'm thinking of setting up a small asterisk server, mainly to play with realtime, and roll some custom billing/administration software probably |
| 06:02.02 | webman | nick125_lappy: yeah, been using it for 2 years, but it is a very old version..... |
| 06:02.05 | mosty | jt: i have switchtype=euroisdn |
| 06:02.20 | JT | mosty: i can't see that parameter at all |
| 06:02.36 | mosty | i didn't paste the whole file |
| 06:02.44 | JT | please paste the whole file |
| 06:02.46 | nick125_lappy | webman: How is it working for you? |
| 06:02.47 | JT | it's sort of important |
| 06:02.55 | JT | especially when you leave out switchtype |
| 06:03.24 | mosty | jt: http://pastebin.ca/506804 |
| 06:03.32 | webman | nick125_lappy: very well ..... was there something specific you wanted to know? |
| 06:04.04 | nick125_lappy | webman: I'm wondering if it is stable, and how you are using it |
| 06:05.12 | nick125_lappy | One thing I want to do with my little setup is be able to add "feature codes" to specific "accounts", that will somehow translate into some extensions entries (I still have no idea exactly how...) |
| 06:05.14 | webman | nick125_lappy: using it as a SIP/IAX2 server, loading from a mysql DB, storing SIP users, and CDR's and voicemail in the DB |
| 06:05.39 | webman | oh yeah, extensions are in the DB as well |
| 06:06.42 | JT | mosty: i think you stuffed that paste up |
| 06:06.47 | JT | mosty: repeated info |
| 06:06.50 | JT | no span definitions |
| 06:08.29 | JT | also, to save time |
| 06:08.29 | JT | also include a call with pri intense debug turned on, and the full output of that |
| 06:08.29 | *** join/#asterisk juliusspencer (n=julius@60-234-130-224.bitstream.orcon.net.nz) |
| 06:08.56 | juliusspencer | hi... anyone know how commands to find out if the motherboard has version 2.2 compliant PCI slots? |
| 06:09.06 | JT | what chipset is it |
| 06:09.06 | mosty | jt: i've had to stop testing now, the client is desperate to use the line |
| 06:09.33 | juliusspencer | <PROTECTED> |
| 06:09.34 | mosty | JT, ie i'm going to have to research the error and try again when i'm onsite next, which is sunday |
| 06:10.03 | JT | mosty: delete [trunkgroups] |
| 06:10.11 | JT | check zap timing |
| 06:10.15 | JT | and do pri intense debugs |
| 06:11.07 | juliusspencer | I've got me pcm400p card in but it's a no show with lspci....\ |
| 06:11.27 | JT | tdm? |
| 06:11.29 | nick125_lappy | webman: Pretty stable and such? |
| 06:11.33 | juliusspencer | oh yeah tdm :) |
| 06:11.44 | juliusspencer | too uch telcoing... :) |
| 06:11.49 | mosty | jt: it's a te412p, i've deleted [trunkgroups] and will try with pri debug |
| 06:12.02 | JT | mosty: why telstra... was it pre-existing or the only available service? |
| 06:12.05 | mosty | i may get another 10 minutes at 5pm to try again |
| 06:12.07 | webman | nick125_lappy: yep, working fine.... very nice, hardly ever a reboot needed (if at all) |
| 06:12.08 | JT | pri intense debug, make sure :) |
| 06:12.24 | mosty | jt: the system i'm replacing was using telstra |
| 06:12.28 | JT | ah right |
| 06:12.36 | mosty | it's an external client |
| 06:12.47 | JT | for your info, optus will happily take it over for no install charge :) |
| 06:12.49 | juliusspencer | I'm (almost) certain that the mobo has pci 2.2 slots |
| 06:12.57 | JT | if they are in an optus exchange area |
| 06:13.23 | webman | JT: how do you get optus to do that ?? I've always been told optus won't do ISDN |
| 06:13.31 | mosty | JT, i heard reports that optus requires wierd cabling |
| 06:13.49 | JT | webman: they don't do BRI, they certainly do PRI :) |
| 06:14.09 | JT | webman: since december, it's been $20/channel |
| 06:14.13 | JT | inc gst |
| 06:14.20 | JT | $0 install fee |
| 06:14.30 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:15.09 | JT | and if you have existing phone bills with a bit of spend ($1k+), they can get you better rates than their already cheaper than telstra rates, and will even credit your bill (for the cost of buying a pri card, or whatever :) ) |
| 06:15.12 | webman | JT: oh, only a resent thing :) |
| 06:15.26 | JT | webman: they've been around before that, just weren't pushing so hard |
| 06:15.55 | webman | JT: yeah, I know, a place I worked got a 10 channel PRI in 1996 .... and a 100+ analog lines... |
| 06:16.07 | JT | haha |
| 06:16.29 | webman | mind you, we were in one of the three buildings they put their fibre nodes into in sydney CBD |
| 06:16.34 | JT | mosty: haven't seen any evidence of "weird cabling" |
| 06:16.59 | JT | optus pri config is the same as telstra as far as i recall too |
| 06:17.28 | mosty | jt: this client is moving buildings in a few weeks, i think it's non-telstra pri at the new place |
| 06:17.40 | JT | but yeah, if you talk to the sales guy right and you have a decent bill, you can get up to a $1000 CREDIT :) |
| 06:17.58 | JT | mosty: optus will also move the PRI for free if you move offices |
| 06:18.47 | JT | also, they will do increments of 1 channel, starting at 10, so they will do any number of channels from 10 up |
| 06:20.37 | mosty | jt: well i will recommend optus to the next client that comes along, these guys are moving to powertel |
| 06:21.10 | JT | i've priced powertel, they weren't very competitive on the line rental side of things, at least from what i saw |
| 06:21.10 | mosty | how do i turn on pri intense debugging? pri intense debug does not work |
| 06:21.25 | JT | help pri debug |
| 06:21.28 | JT | pri intense debug span 1 |
| 06:21.47 | mosty | thanks. wow it sure is intense |
| 06:22.02 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:22.23 | mosty | it's currently full of ff's since the cable is unplugged |
| 06:22.25 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:28.05 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:34.43 | juliusspencer | hi JT any more ideas...? |
| 06:37.04 | JT | juliusspencer: i asked for the motherboard chipset, not the pci bridge, i don't memorise pci bridge model numbers in my head |
| 06:37.07 | *** join/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net) |
| 06:38.32 | juliusspencer | ah sorry... I'll try get that info |
| 06:39.27 | JT | it would have to be pretty old to not be PCI2.2 compliant |
| 06:39.54 | juliusspencer | that's what I figured too |
| 06:41.40 | juliusspencer | Intel 875P Motherboard |
| 06:42.43 | juliusspencer | does that sound like the info which would help? |
| 06:45.04 | webman | does anyone know much about stun here? I'm trying to work out why I would be asked for a primary + secondary IP address. |
| 06:46.21 | juliusspencer | webman, this article might help... http://www.linuxjournal.com/node/9540/print |
| 06:47.46 | JT | webman: stun always needs 2 IPs |
| 06:47.50 | JT | stun is a waste of time |
| 06:48.03 | JT | unless you use a rubbish sip server that's completely NAT dumb |
| 06:48.16 | JT | hopefully not many of those are left |
| 06:48.42 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 06:49.03 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
| 06:49.14 | JT | juliusspencer: 875P is a P4 chipset, well and truly years since everything has been PCI2.2 compliant :) |
| 06:49.22 | JT | it's really only a worry on PII and lower machines |
| 06:49.32 | juliusspencer | ah cool... hmmm |
| 06:49.42 | juliusspencer | any ideas baout why it might not show up with lspci? |
| 06:50.20 | juliusspencer | I've tried it on another PC (compaq evo) don't know the mobo chjipset.. but can find out... I'll just ssh in |
| 06:50.43 | webman | JT: do you know what the 2 IP's are?? ie, does the stun server need 2 IP's of it's own ?? |
| 06:50.47 | juliusspencer | it would be perfect in my mythtv box as it's always on |
| 06:53.14 | webman | juliusspencer: what card doesn't show up in lspci ?? |
| 06:53.14 | juliusspencer | JT the other one has an Intel 845 chipset. I was using a PCI riser board so maybe that was the problem there.... |
| 06:53.14 | juliusspencer | oh |
| 06:53.14 | juliusspencer | JT, it's a TDM400P (one FXO and one FXS) |
| 06:54.02 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 06:54.46 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 06:54.46 | juliusspencer | JT, kernel versions are, (evo, 845) 2.6.20-1.2316.fc5 and (dimension, 875P) 2.6.20-1.2948.fc6 |
| 06:55.52 | juliusspencer | The manual says 2.4 kernel or later so should be pretty sweet... |
| 06:58.17 | *** part/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
| 06:58.41 | JT | web |
| 06:58.42 | JT | bah |
| 06:58.44 | JT | gone already |
| 06:58.47 | JT | useless |
| 06:58.57 | JT | dumb questions for a dumb protocol |
| 06:59.02 | JT | stun is an utter waste of time |
| 06:59.30 | coppice | stun is an extremely useful thing |
| 06:59.53 | JT | if your sip server is rubbish |
| 07:00.09 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 07:00.21 | coppice | if you think that, it seems you don't understand stun |
| 07:00.21 | Know1 | well I'm using asterisk and I can't see how to do this currently without STUN |
| 07:00.46 | Know1 | sip client behind work firewall registering to home asterisk box |
| 07:00.59 | Know1 | currently doesnt wokr even with stun |
| 07:01.01 | JT | coppice: what exactly is it so useful for? |
| 07:01.16 | Know1 | has buckleys chance without stun |
| 07:01.19 | JT | Know1: you should be able to set that up without stun if it can be setup at all |
| 07:01.28 | Know1 | I'm thinking the later |
| 07:01.31 | coppice | two rtp end points behind NATs can eaily find each other with STUN |
| 07:01.36 | JT | Know1: likely the firewall is the showstopper anyway |
| 07:01.48 | Know1 | yeah that's the prob work firewall means no can touch it |
| 07:02.01 | Know1 | i.e no passwords to change it |
| 07:02.08 | juliusspencer | hey JT, any ideas for this TCM400P? |
| 07:02.10 | Know1 | so STUN looks like the only route |
| 07:02.17 | JT | juliusspencer: tcm? |
| 07:02.24 | juliusspencer | TDM |
| 07:02.25 | JT | Know1: or iax |
| 07:02.30 | juliusspencer | TDM400P |
| 07:02.43 | Know1 | lol yeah that is fast becoming an option if I can find a decent IAX softphone |
| 07:02.54 | JT | no softphones are decent. |
| 07:03.01 | JT | idefisk works |
| 07:05.04 | JT | coppice: i'd rather pass the media through a proxy os B2BUA that run directly between 2 NATed clients |
| 07:06.02 | coppice | JT: you must like paying for bandwidth and having extended latency, then. most folk don't |
| 07:06.16 | juliusspencer | heh I'd just like to be able to see my card in my OS :) |
| 07:07.01 | JT | coppice: actually media proxying can reduce latency in many cases, and ideally the server would be running on decent bandwidth anyway |
| 07:07.13 | JT | in terms of costs.... very neglibible data use with voip |
| 07:07.21 | JT | negligible |
| 07:07.44 | coppice | JT: in your setup, maybe. in the general case, its *bad* |
| 07:08.15 | juliusspencer | hey coppice, do you know how I might go about getting a TDM400P card to show up in Linux? |
| 07:08.19 | JT | most (read all) ITSPs proxy media, even if the two clients are on their network |
| 07:09.09 | coppice | juliusspencer: if you have zaptel installed it should show up. if not, see if misdn is installed and grabbing the card as being an ISDN card |
| 07:09.20 | JT | coppice: so that leaves 2 NATed clients talking directly to each other as a small niche case |
| 07:09.28 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 07:09.28 | coppice | JT: the world is more that stinky little ITSPs |
| 07:09.30 | JT | mainly home users without asterisk |
| 07:09.35 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 07:09.46 | JT | coppice: why are you so angry? |
| 07:09.49 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 07:09.58 | juliusspencer | coppice, so I need the driver installed before seeing the card with lspci? |
| 07:10.00 | coppice | I'm not. you are just myopic |
| 07:10.23 | coppice | oh, lspci should see it as a tigerjet card without any driver |
| 07:10.27 | juliusspencer | I actually am myopic |
| 07:10.29 | juliusspencer | heh |
| 07:10.46 | JT | coppice: i find that offensive, to me and people who actually are myopic |
| 07:10.52 | juliusspencer | hahah I don't |
| 07:10.58 | coppice | 03:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 07:11.05 | juliusspencer | yeah that's what I'm looking for |
| 07:11.12 | juliusspencer | just... isn't there |
| 07:11.16 | juliusspencer | I'll try installing all the sw |
| 07:11.23 | juliusspencer | just that... it didn't help on another machine |
| 07:11.26 | JT | coppice: perhaps you should explain all the wonderfully big market for sip clients behind NAT talking directly |
| 07:12.17 | coppice | I use SIP regularly, talking to other SIP users, mostly behind NAT, and I've never touched an ITSP in my life |
| 07:12.56 | JT | do you use asterisk or similar? |
| 07:13.37 | coppice | something similar |
| 07:13.53 | JT | right |
| 07:13.57 | JT | and that should do the job |
| 07:15.08 | juliusspencer | so coppice, once I have zaptel installed I'll come back... |
| 07:16.02 | JT | usually i'm seeing stun prescribed for situations where it's completely unnecessary and just adds complication and reliance on outside services |
| 07:17.24 | Know1 | danged why can't SIP be better |
| 07:17.35 | Know1 | created an IAX extension and it worked first time |
| 07:17.49 | Know1 | not nat trickery or anything |
| 07:17.51 | Know1 | no* |
| 07:18.24 | coppice | JT: so how to my two boxes behind NAT allow their RTP streams to find each other? |
| 07:18.45 | JT | coppice: behind the same NAT device or different? |
| 07:19.15 | coppice | on different sides of the planet |
| 07:19.24 | JT | Know1: sip works great for a lot of situations, highly restrictive networks is not one of them |
| 07:19.38 | JT | coppice: just put the B2BUA on the DMZ or port forward :) |
| 07:20.14 | Know1 | yeah I think SIP will be all the devices at home as sip phones are easy to come by |
| 07:20.26 | Know1 | but for the lappie when away from home I'll have to use the iax softphone |
| 07:20.35 | coppice | you can have a setup like that, which only an expert gets right, or you go plug and play with STUN |
| 07:21.06 | JT | coppice: and can you guarantee that STUN will always work, 100% of the time? :) |
| 07:21.30 | coppice | there are masses of freely usable STUN servers out there. they just work |
| 07:21.42 | JT | not talking about the servers |
| 07:21.56 | JT | talking about the weird networks people have |
| 07:22.05 | JT | some like to make it difficult for data to pass |
| 07:22.06 | *** join/#asterisk juliusspencer (n=julius@60-234-130-224.bitstream.orcon.net.nz) |
| 07:22.18 | rudholm | I think STUN isn't so reliable when dealing with larger enterprise NAT devices. |
| 07:22.25 | coppice | unless you have one of those perverse firewall arrangements, yes. |
| 07:22.38 | juliusspencer | coppice, still no go.... |
| 07:22.53 | JT | a lot of places have firewalls |
| 07:22.57 | JT | they're everywhere |
| 07:23.03 | coppice | if people are trying to block you with perverse arrangements they won't set up your arrangement anyway |
| 07:23.14 | rudholm | yeah, like pretty much every large corporate network. |
| 07:23.25 | juliusspencer | coppice, I even rebooted :) it doesn't even appear to recognize a card is installed |
| 07:23.25 | JT | just like NAT is everywhere |
| 07:23.48 | coppice | but STUN gets through 99% of NAT with no setup effort. it just works |
| 07:23.50 | *** join/#asterisk friedrich| (n=friedric@e177244192.adsl.alicedsl.de) |
| 07:23.57 | *** join/#asterisk federicoco (n=federico@host130-56-static.206-80-b.business.telecomitalia.it) |
| 07:24.00 | juliusspencer | unlike my card |
| 07:24.14 | coppice | juliusspencer: sounds like a dead card |
| 07:24.19 | JT | call support or something already |
| 07:24.36 | juliusspencer | bugger... brand new too! |
| 07:24.42 | JT | how is mentioning your card every 2 seconds in here going to help? :) |
| 07:25.06 | coppice | JT: you are saying STUN won't work 100% of the time, but offering a solution which will never work without serious cooperation |
| 07:25.19 | tsurko | hello, I've a problem with ChannelRedirect on Asterisk 1.4.4. I'm trying to make n-way call as shown in voip-info.org, but I get this in CLI: |
| 07:25.34 | tsurko | WARNING[2981]: app_channelredirect.c:112 asyncgoto_exec: ChannelRedirect failed for SIP/alpha-081f4608 |
| 07:25.49 | tsurko | have you got any idea why it fails? |
| 07:26.22 | juliusspencer | well figured being a PITA might get some people giving some ideas for troubleshooting... rather than arguing security semantics. |
| 07:26.25 | JT | coppice: yes, but once setup right, an asterisk box can easily connect to a bunch of clients, without the use of stun at all |
| 07:27.10 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 07:27.22 | coppice | true, but it needs setting up on the firewall. stun is just "Enter stun server URL:". In works in hotels. It works in lots of places where setup is difficult or impossible |
| 07:28.30 | JT_ | i'd argue that iax2 has a higher chance of working in those sort of environments :) |
| 07:29.04 | JT_ | if they let through 5060, and 2 ports between 10000 and 20000, chances are good they'll let 4569 through |
| 07:29.09 | coppice | IAX2 has no better chance of working, unless you are happy pumping media through a central point |
| 07:30.08 | JT_ | a single port versus 3 ports, 2 of which aren't easily associated with the other from the nat or firewall's point of view |
| 07:31.51 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 07:32.23 | coppice | JT: face it. STUN just works. it works with masses of products. its service far more people quite happily right now than use IAX2 or asterisk. now RTP is almost always symmetric, its the normal way of doing things these days. The next step will be merging RTP and RTCP on a single port, making things simpler still |
| 07:33.01 | JT | what about sip? |
| 07:33.18 | coppice | what about SIP? |
| 07:34.44 | JT | the sip port |
| 07:34.52 | JT | merging with rtp anytime soon? |
| 07:36.05 | coppice | that wouldn't make any sense,. SIP is a TCP protocol now, with UDP being deprecated |
| 07:36.34 | *** join/#asterisk rrrobert (n=rrrobert@58-65-160-140.nayatel.pk) |
| 07:36.48 | JT_ | deprecated where? |
| 07:36.55 | coppice | it wouldn't be easy to do, anyway. if you look at RTP and RTCP you'll see they were carefully designed to allow merging on a single port |
| 07:37.13 | coppice | the current SIP spec says TCP is mandatory and UDP should be phased out |
| 07:37.28 | JT | good to see anyone is follow it ;) |
| 07:37.32 | JT | following |
| 07:38.12 | coppice | they are |
| 07:38.33 | coppice | most good SIP implementations support TCP and TLS over TCP. many do SCTP too |
| 07:39.11 | JT | are there any "good" open source sip implementations? |
| 07:41.16 | coppice | a number. sofia is the best |
| 07:41.51 | coppice | sofia is definitely the one to use |
| 07:42.42 | coppice | I've seen some support it. the makers can be really slow to catch up, but they are all implementing it. |
| 07:42.51 | coppice | SIP over UDP was brain dead |
| 07:43.18 | JT | maybe so, but sip over tcp won't take off until the great majority of softphones and hardphones support sip over tcp |
| 07:43.56 | coppice | people want security. security == TLS/TCP/SIP the phone makers are under pressure to get that working |
| 07:44.07 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 07:44.09 | JT | heh |
| 07:44.26 | JT | hopefully they'll update even the low end models |
| 07:44.32 | coppice | on the other hand most T.38 implementations still only support the 1999 version :-) |
| 07:44.59 | JT | t.38, so endpoints can still only be done with cw for free, right? |
| 07:45.00 | drrt | what manufacter are u talking about ? |
| 07:45.08 | JT | none |
| 07:46.04 | rudholm | Yahoo! Instant Messenger Voice is using SIP over TCP fwiw. |
| 07:46.57 | coppice | it appears Digium are determined for * to not support T.38, so yeah, CW is the only option right now |
| 07:47.25 | JT | coppice: there's more than * and cw ;) |
| 07:47.37 | JT | yeah dig don't seem to be pioneering much at the moment |
| 07:48.12 | coppice | freeswitch wants my stuff, and will probably implement it in a while through pipes, to avoid licence conflicts. I don't know about yate |
| 07:48.49 | JT | hate i heard "one day" and they reckon spandsp can work with it but don't recommend it |
| 07:49.17 | coppice | yate is a PITA its best to avoid |
| 07:49.23 | JT | why so? |
| 07:49.45 | coppice | you haven't encountered the lovely diana? :-) |
| 07:50.03 | JT | i've seen far more vocal people in other circles |
| 07:50.05 | mbranca | they think that everystuff outside yate is crap and want to implement from scratch |
| 07:50.18 | JT | i don't judge software packages based on one person |
| 07:50.28 | JT | mbranca: so? |
| 07:50.40 | coppice | she *is* yate, though she doesn't develop it. |
| 07:50.50 | JT | heh |
| 07:50.59 | mbranca | so they cannot be able to do everything, they cannot have the knowledge of everything |
| 07:51.10 | coppice | or the manpower |
| 07:51.14 | mbranca | yeah |
| 07:51.19 | JT | mbranca: sorry, i don't really go for rheotoric like that |
| 07:51.19 | JT | sure |
| 07:51.22 | JT | no project does |
| 07:51.31 | JT | they all have their advantages |
| 07:51.31 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 07:51.59 | coppice | any encounter with yate is just a miserable experience. its just better to avoid it |
| 07:52.04 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 07:52.16 | JT | uhuh |
| 07:52.19 | JT | someone's bitter |
| 07:52.38 | coppice | no. i just hate dealing with obnoxious people, so I don't |
| 07:53.00 | JT | there's no imperative to deal with a single person |
| 07:53.18 | coppice | there is with yate. there is only one person to deal with |
| 07:53.42 | JT | there are others on the mailing list who help |
| 07:54.29 | coppice | dunno. i have successfully avoided contact with it for a while. there used to be no alternative |
| 07:54.47 | JT | i get it, you two hate each other, get a room already :) |
| 07:55.15 | coppice | i don't hate her. I just avoid her. most people in telephony do |
| 07:56.03 | coppice | what puzzles me is what hold she has over the smart guy who does the real work. i can't remember his name |
| 07:56.19 | *** join/#asterisk ishi (n=ishi@81.210.48.62) |
| 07:56.21 | JT | clearly you dislike her |
| 07:56.25 | ishi | hello :) |
| 07:56.38 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 07:56.59 | coppice | oh, i think everyone dislikes her. its hard to imagine anything different |
| 07:57.21 | JT | it shows real signs of bitterness when you speak for everyone like that :) |
| 07:57.57 | coppice | huh? |
| 07:58.17 | JT | "i dislike her, so clearly everyone must dislike her" |
| 07:58.32 | coppice | I didn't say that |
| 07:58.38 | JT | i have a healthy skepticism towards everyone |
| 07:59.06 | JT | but i also try not to pass judgements, especially based off limited online contac |
| 07:59.09 | coppice | since many people comment abot her, and its almost universally negative, I conclude what I said |
| 07:59.24 | JT | coppice: yes but perhaps in certain circles more than others? |
| 07:59.33 | JT | i know certain projects have a level of rivalry |
| 08:00.39 | coppice | healthy rivalry is friendly rivalry. |
| 08:00.39 | JT | i know there is unfriendly rivalry too |
| 08:00.43 | JT | such is humanity |
| 08:00.49 | coppice | I have a good sparing relationship with freeswitch, for example |
| 08:00.55 | JT | i realise |
| 08:01.05 | ishi | hoom... sorry to barge in... I don't mean to change the subject like that, but since I don't know "her", I cannot help the feeling, i'm overhearing somthing I should not... |
| 08:01.22 | JT | ishi: heh, diana, of yate |
| 08:01.42 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 08:02.21 | JT | i realise not everyone are friends, i just try to stay the hell out of it :) |
| 08:02.44 | ishi | anyways... I have a pragmatic question so indulge me... I'd like to write a filter "as in signal processing, that is", and filter an asterisk controlled conversation through it... |
| 08:02.57 | JT | what sort of filter? |
| 08:03.00 | coppice | that's what i said before. I simply try to steer clear of people who are determined to be a PITA |
| 08:03.25 | JT | coppice: i'm not sparring myself, so no need to do so there :) |
| 08:03.36 | ishi | JT: well.. any sort ;) an adaptive most probably... but first of all I'd like to know if something like this exist alread... |
| 08:03.50 | coppice | ishi: * doesn't provide a good framework for that kind of thing, though you can often achieve the effect through a custom app |
| 08:03.51 | ishi | I couldn't find anything in your webpage... :( |
| 08:03.52 | JT | ishi: what would the filter do, functionally? |
| 08:04.04 | JT | my webpage? |
| 08:04.21 | ishi | "your" as in the projects |
| 08:04.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:04.40 | coppice | ishi what are you trying to achieve? |
| 08:04.43 | JT | most of us aren't directly part of the project apart from using it |
| 08:04.51 | ishi | :) |
| 08:05.10 | JT | ishi: so what would the filter do already, or is that a national security secret? |
| 08:07.02 | ishi | ok, the imediate goal is to produce an IVR service, where a user might choose a "voice changer" via some kind of menu, and call a friend... I did something like this for the 1st of april but hmm... "off line" -- sort of voice post cards... |
| 08:07.44 | coppice | someone has implemented a voice changer for * |
| 08:07.48 | ishi | now they want me to do it "live"... but the "filtering API" might be used for other things... |
| 08:07.51 | ishi | coppice: doesn't work |
| 08:08.22 | coppice | it does work. maybe it doesn't do the kind of changing you want |
| 08:08.35 | coppice | its rather slow. a fast implementation is possible |
| 08:08.37 | ishi | it was done for version 1.2 (or somthing like this)... when I "plug it in" all I get is cracks ;) |
| 08:08.58 | coppice | use 1.2. that's what the smart people do :-) |
| 08:09.35 | coppice | seriously, if it doesn't work with 1.4 the required changes should be small ones |
| 08:09.38 | ishi | heh... ok, will try... but as I said, I'd like to do something more general... that's why I came here asking for some docs... |
| 08:10.05 | ishi | but perhaps you are right... I should just read the code of the previous implementation... |
| 08:10.08 | ishi | oh well.. |
| 08:10.11 | ishi | :) |
| 08:10.43 | JT | ishi: is your . button stuck? |
| 08:10.57 | ishi | JT: you *have* noticed? ;) |
| 08:11.05 | coppice | I have something to do the opposite - vary the playback speed, without changing the pitch :-) |
| 08:11.50 | JT | to listen to recorded calls very fast? ;) |
| 08:11.53 | ishi | coppice: well... it seems a hard thing to do ;) --- especialy if you slow things down ;) |
| 08:13.00 | coppice | not really. there are several ways to vary playback speed which are not too complex and work well for voice. varying speed for high quality music is tough, though |
| 08:13.13 | ishi | JT: btw, I'm sorry about the '...' --- it's a habit of mine, and it's a bit hard to break after 10 years of ircing, but I'll do my best ;) |
| 08:14.30 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 08:15.52 | ishi | s/\.\.\././g |
| 08:16.17 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
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| 08:29.43 | *** join/#asterisk MerlinZShell (n=merlin@212.38.147.249) |
| 08:29.52 | MerlinZShell | hello giuys |
| 08:30.30 | MerlinZShell | im new to asterisk |
| 08:30.46 | MerlinZShell | is there any quick start guide for asterisk you recommend |
| 08:31.39 | *** join/#asterisk mendol (i=mendol@bobas.nowytarg.pl) |
| 08:33.27 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
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| 08:54.22 | MerlinZShell | any body alive here ? |
| 08:59.19 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:01.35 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
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| 09:06.27 | *** join/#asterisk pejo_ (n=peter@136.240.13.217.in-addr.dgcsystems.net) |
| 09:08.19 | *** join/#asterisk EEE53EEE (n=EEE53EEE@p57B870D8.dip.t-dialin.net) |
| 09:08.33 | EEE53EEE | hi |
| 09:15.21 | *** join/#asterisk rulereric (n=snellman@218.188.202.68.cfl.res.rr.com) |
| 09:16.00 | EEE53EEE | maybe somebody of you can help me. I have got a "busy" problem. When making an internal call to a busy phone, asterisk signing "busy" immediately. When calling from outside it first rings twice and then switchs to busy, in spite of the fact that the dialplan is the same. somebody knows this problem? |
| 09:17.09 | MerlinZShell | hi EEE53EEE well i guess you can help me with a very basic issue |
| 09:17.15 | MerlinZShell | are you willing to ? |
| 09:17.25 | EEE53EEE | i will try :) |
| 09:18.24 | MerlinZShell | I heard asterisk can help me make PC to phone calls |
| 09:18.31 | MerlinZShell | yes im that new to asterisk |
| 09:18.40 | MerlinZShell | but im good with linux in general |
| 09:20.12 | MerlinZShell | ok ste one |
| 09:20.37 | MerlinZShell | should i buy some kind of account what kind of account and from who |
| 09:20.38 | EEE53EEE | yes, asterisk can. but you will need either a ISDN-Card or a sip-provider to make calls |
| 09:20.48 | MerlinZShell | and where should i configure that |
| 09:21.00 | MerlinZShell | perfect |
| 09:21.05 | MerlinZShell | sip provider |
| 09:21.25 | MerlinZShell | can you recommend one that you know of with good quality and low rates |
| 09:22.18 | EEE53EEE | no, sorry. i do not use a sip-provider |
| 09:22.48 | MerlinZShell | ok thank you EEE53EEE |
| 09:22.54 | MerlinZShell | one more question |
| 09:22.54 | EEE53EEE | np |
| 09:23.04 | EEE53EEE | jo |
| 09:23.06 | MerlinZShell | can asterisk act as a server |
| 09:23.38 | MerlinZShell | like h323 client connects to my asterisk server |
| 09:24.49 | EEE53EEE | yes... you can connect many devices to the asterisk-server through the lan and make outgoing calls through the internet (sip-provider) |
| 09:25.22 | EEE53EEE | the other option is to make outgoing calls through isdn-cards |
| 09:28.28 | MerlinZShell | can i ask an ignorant question |
| 09:28.52 | MerlinZShell | ISDN card is a card that i plug into my local land line to be the gateway for my calls |
| 09:29.15 | MerlinZShell | while a sip provider is someone else with a asterisk server that i can relay my calls to right ? |
| 09:32.15 | EEE53EEE | the isdn-card you should plug into the computer on which asterisk is running so asterisk can make calls through the telephone line |
| 09:32.46 | JT | argh |
| 09:32.46 | JT | no |
| 09:32.49 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 09:32.55 | JT | isdn line connects to a digital ISDN line |
| 09:33.07 | skirmisha | guys how can i add prefix to dialed number before routing decision |
| 09:33.08 | JT | it's a different card for an analogue POTS line |
| 09:33.44 | JT | h.323, you can try, but support is poor in asterisk and you'll have better luck with SIP |
| 09:34.30 | JT | i meant this before: isdn card connects to a digital ISDN line |
| 09:34.39 | JT | isdn is not the same as analogue |
| 09:34.42 | JT | utterly different |
| 09:35.39 | JT | MerlinZShell: yes asterisk acts as a server |
| 09:36.35 | MerlinZShell | ok now i did install asterisk |
| 09:37.08 | defswork | anyone know of a simple app that shows extension status ? our phone don't have enough busy lamps |
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| 09:41.10 | MerlinZShell | does asterisk have an interface |
| 09:41.29 | MerlinZShell | or if i have to configure it using a config file what is the config file in question |
| 09:41.42 | JT | configure it with the config files |
| 09:41.50 | JT | it has a console, but that's more for monitoring |
| 09:42.30 | defswork | MerlinZShell: install freepbx if you want |
| 09:42.39 | JT | or not |
| 09:43.33 | MerlinZShell | i went to /etc/asterisk there is no config files |
| 09:43.42 | MerlinZShell | folder is empty |
| 09:43.44 | JT | did you compile from source? |
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| 09:43.58 | MerlinZShell | do i have other options im on ubuntu |
| 09:44.05 | JT | it's a directory btw, this isn't windows :) |
| 09:44.10 | JT | maybe, but the question stands |
| 09:45.43 | MerlinZShell | :) im more of a linux guy :) |
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| 09:45.55 | JT | right |
| 09:45.56 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 09:46.09 | MerlinZShell | I went to cd /etc/asterisk/ | ls -al nothing |
| 09:46.12 | JT | so did you compile from source or not? |
| 09:46.20 | JT | yes you've said that 3 times now |
| 09:47.56 | MerlinZShell | i compiled from source |
| 09:48.10 | JT | so make samples |
| 09:50.10 | MerlinZShell | thank you guys |
| 09:52.40 | JT | for future reference, we can help you faster if you pay more attention to our questions, they're so we can work out how to help you :) |
| 09:58.46 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 09:58.54 | eeos | hi there! |
| 09:59.17 | eeos | we have VOIP lines, but we would like to have our local configurable Asterisk. |
| 09:59.38 | eeos | Can we keep the extisting lines and have our local Asterisk working on them? |
| 10:01.18 | JT | i take it you're asking if you can connect a landline to asterisk? if so, the answer is yes |
| 10:09.22 | eeos | what I ask is: we have broadband, on this broadband we have voip lines offered by a provider, we want to keep the existing number and feed them into our local Asterisk server |
| 10:12.55 | eeos | so that we can mange all aspects of the phone lines, like redirection and so on, locally |
| 10:14.30 | Winkie | hey guys, having an issue trying to set up a bidirectional IAX trunk |
| 10:14.43 | Winkie | i was recommended not to use a single 'friend' section but i can't even get peer/user to work properly |
| 10:14.47 | Winkie | any advice? |
| 10:15.41 | *** join/#asterisk saftsack (n=saftsack@pd9e064e9.dip.t-dialin.net) |
| 10:18.00 | eeos | JT: what do you think? |
| 10:24.37 | JT | eeos: you're not being specific.... where are these numbers coming from, landline connection or purely VoIP over Internet? |
| 10:29.03 | eeos | I said in my original post they were VOIP lies :) |
| 10:29.17 | eeos | s/lies/lines |
| 10:29.29 | eeos | purely VOIP |
| 10:30.06 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 10:30.13 | JT | the use of the word "lines" is confusing |
| 10:30.18 | JT | voip isn't really lines |
| 10:30.49 | JT | so yes, if they use a standard VoIP protocol like SIP and you know the SIP password, you should be able to connect it to asterisk |
| 10:33.53 | eeos | JT: yes, they use SIP protocol and I know the password. What do you think will happen with the voice quality? Have you ever tried? |
| 10:34.15 | defswork | ever heard a dalek talk ? ;) |
| 10:35.05 | eeos | defswork: a what???? |
| 10:35.18 | JT | eeos: shouldn't change at all |
| 10:35.23 | JT | provided you use the same codecs |
| 10:37.45 | eeos | JT: thanks a lot you have been so very helpful!!!! |
| 10:38.05 | JT | you may want to check out the book |
| 10:38.08 | JT | ~thebook |
| 10:38.46 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:49.00 | eeos | JT: the book? |
| 10:54.05 | JT | eeos: ref the url above |
| 10:54.08 | eeos | JT: done and downloaded. thanks. defswork! |
| 10:54.28 | defswork | me ? all I did was mention daleks |
| 10:54.47 | eeos | sorry, thanks jbots |
| 10:54.56 | JT | thank the bot now |
| 10:55.01 | JT | it just works under command |
| 10:55.31 | *** join/#asterisk alexzz (n=chatzill@122.166.0.71) |
| 10:55.39 | eeos | JT: how did you call it? is it not the usual ! ? |
| 10:55.44 | *** join/#asterisk tdonahue-laptop (n=tdonahue@c-71-235-114-164.hsd1.ct.comcast.net) |
| 10:55.47 | tdonahue-laptop | morning all |
| 10:55.54 | JT | eeos: just scroll up :) |
| 10:56.06 | tdonahue-laptop | having some G.729 trouble |
| 10:56.38 | eeos | JT: :D:D:D:D I see! I just thought you had mistyped! |
| 10:57.15 | tdonahue-laptop | I registered a G.729 licence with the register utility yesterday and this morning when i restarted asterisk to bring the 729 codec up I am getting an error "codec_g729.c: Failed to initialize G.729 copy protection!" |
| 10:57.29 | tdonahue-laptop | anyone know where to look to fix this? |
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| 11:02.04 | tdonahue-laptop | no one here has seen this error before? |
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| 11:47.05 | skirmisha | guys can asterisk do echo cancelation on voip calls? |
| 11:51.57 | Tebi_ | sure |
| 11:52.43 | skirmisha | ? |
| 11:52.55 | Tebi_ | yes it can |
| 11:52.58 | skirmisha | i mean there is on isdn card |
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| 12:01.00 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 12:06.26 | sebastian|foo | hi |
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| 12:31.08 | deegan | I got a question, i'm very new to asterisk. We swapped machines and moved the ISDN card over to the new machine and now it continues to spit out that it can't find a D-channel 16. Been going on since yesterday. |
| 12:31.08 | deegan | The config is the same as the previous machine and it didnt have theese problems. Is this a ISDN problem or a zap problem? |
| 12:31.08 | Dovid | anyone ere using KVM over IP ? |
| 12:31.08 | deegan | pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 12:31.08 | Dovid | i know this is OT |
| 12:31.08 | *** join/#asterisk samarora (i=minesh@203.88.149.166) |
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| 12:31.09 | deegan | zttool does not repot any errors. |
| 12:31.09 | deegan | s/repot/report. |
| 12:31.09 | JT | deegan: more info? |
| 12:31.09 | deegan | JT, what kind of info would you need? I just got shoved into the asterisk managing and learning it now so help me help you if you dont mind. :) |
| 12:31.09 | JT | zapata.conf zaptel.conf in pastebin.ca, and what card you are using, if it is a t1 or e1, pri or not, what country |
| 12:31.09 | deegan | roger, hold on. |
| 12:31.46 | samarora | i found one error "ZT_CHANCONFIG failed on channel 1 : No such device or address (6)" after "ztcfg" command..pl suggest |
| 12:31.49 | *** part/#asterisk alexzz (n=chatzill@122.166.0.71) |
| 12:33.03 | samarora | can anybody suggest me on same..? |
| 12:33.20 | deegan | JT, http://www.pastebin.ca/507175 |
| 12:33.39 | deegan | there's only one card in that macine. |
| 12:33.44 | deegan | s/macine/machine. |
| 12:34.03 | JT | that's not the card modelnumber |
| 12:34.10 | JT | that's the generic name that comes up |
| 12:34.53 | JT | deegan: you have WAAAY too many modules loaded |
| 12:35.01 | JT | you only need the relevant ones |
| 12:35.08 | JT | not everyone under the planet |
| 12:35.45 | deegan | JT, it's a druid-os install so that's the default really but yea i was a bit confused by the amount of modules loaded. |
| 12:36.04 | JT | druidos? |
| 12:36.40 | deegan | yea it's a linux distro that installs asterisk and the druid webb-interface. |
| 12:36.45 | JT | i see |
| 12:36.57 | deegan | voiceroute.net i think is the creator of it. |
| 12:37.05 | deegan | Not that it matters but still. :) |
| 12:38.37 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 12:39.04 | deegan | Anyway, i could get back to this when i'm not working from home. just figured i'd give it a shot here, we've been googling and talking about it since yesterday when we installed the test enviroment for this druidos. |
| 12:40.07 | deegan | basicly, no solid answers from the www. it's not really _that_ big of a deal except that it spews out a shitload of debuging info and i guess it could have some effect on the calls people will be making on the system. |
| 12:42.42 | *** join/#asterisk jtexter3 (n=jtexter3@69-94-197-145.biltmorecomm.com) |
| 12:43.41 | JT | deegan: what are the results of zttest like? |
| 12:44.00 | deegan | JT, found the real name of the card. zttool, Digium Wildcard TE110P T1/E1 Card 0 |
| 12:44.28 | JT | ah ok |
| 12:44.34 | JT | makes sense |
| 12:45.06 | deegan | zttest says, Best: 99.987793 -- Worst: 99.987793 -- Average: 99.987793 |
| 12:45.16 | JT | how many did you run? |
| 12:45.31 | deegan | that was just 7 but i can have it going for as long as is necesary. |
| 12:46.26 | JT | may not be the problem anyway |
| 12:46.28 | JT | hrm |
| 12:46.29 | JT | well |
| 12:46.33 | JT | is it plugged in? :) |
| 12:46.55 | deegan | As a mather of fact yes it is haha. :) |
| 12:47.28 | deegan | It's just so wierd, i mean we didnt really notice any losses because of it but still it's not suppose to be spewing out warnings a few times a minute you know. |
| 12:47.35 | deegan | --- Results after 55 passes --- |
| 12:47.35 | deegan | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.988015 |
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| 12:55.05 | samarora | <PROTECTED> |
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| 13:05.58 | JT | deegan: with the same cable as before? |
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| 13:06.53 | Katty | morning sleepies. |
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| 13:10.01 | [TK]D-Fender | Katty: *yawn* Mew. |
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| 13:11.20 | deegan | JT, yea just moved the card from one box to another. |
| 13:11.45 | Qwell | yay wifi? |
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| 13:15.08 | JT | deegan: pastebin console output with errors, with pri intense debug turned on |
| 13:16.44 | intralanman | anyone know if a spa3k can pass callerid when being used as a PSTN->SIP gateway? |
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| 13:19.36 | Dr-Linux | need some idea, what's the best opensourse incoming call notification program out there? |
| 13:19.52 | Katty | could you write a batch script, that would resolve your public ip address, then dump that into a file, and ftp it somewhere? |
| 13:19.56 | Katty | or bash. |
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| 13:22.34 | [TK]D-Fender | intralanman: From PSTN to SIP, yes |
| 13:23.08 | intralanman | [TK]D-Fender: could you point me to a doc that would tell me how? |
| 13:23.25 | [TK]D-Fender | intralanman: www.voxilla.com <- go read the forums on this |
| 13:23.30 | intralanman | thnx |
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| 13:55.41 | syzygyBSD | gotoif(timeevaluation,afterhours,daytime) |
| 13:55.46 | syzygyBSD | or something... |
| 13:55.54 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:55.57 | syzygyBSD | the gotoif wiki probably has good examples |
| 13:56.01 | *** part/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:00.22 | [TK]D-Fender | Katty: "core show application gotoif" |
| 14:00.30 | [TK]D-Fender | Katty: "core show application gotoiftime" |
| 14:00.50 | Katty | fascinating. |
| 14:01.24 | Katty | meep |
| 14:02.05 | redax | hi n3glv, |
| 14:03.37 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 14:04.31 | *** join/#asterisk berktr (n=canberk@teknopet.com) |
| 14:07.04 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-96-8.hvc.res.rr.com) |
| 14:08.03 | cpm | Katty, you a fonality reseller yet? |
| 14:08.07 | Katty | cpm: no |
| 14:08.10 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 14:08.11 | zeeesh | hi |
| 14:08.11 | cpm | :) |
| 14:08.28 | Katty | cpm: but the tech support offer is appealing to someone who doesn't have a lot of linux knowledge. |
| 14:08.39 | cpm | it was funny, right after you mused about that, fonality got a lot of really good press |
| 14:08.43 | Katty | cpm: 12 grand to build the same thing that i can build for 2 or 3... |
| 14:08.52 | Katty | meh |
| 14:09.12 | [TK]D-Fender | Katty: Just resell Trixbox installs instead. Pure profit :) |
| 14:09.16 | zeeesh | making sip call through xlite to cell phone ... call has been established ... but there is no voice??? |
| 14:09.21 | Katty | [TK]D-Fender: but i need tech support |
| 14:09.25 | Katty | [TK]D-Fender: I LIVE FOR TECH SUPPORT |
| 14:09.26 | cpm | Well, what about the tech support offer for someone who only wants to implement then walk the heck away from it all |
| 14:09.49 | Katty | mew? |
| 14:10.31 | Katty | ah |
| 14:10.39 | Katty | yes. |
| 14:10.45 | Katty | they take over everything |
| 14:10.53 | Katty | right down to hiring it out for you if they can't walk you through it |
| 14:10.55 | Katty | tis for a price tho |
| 14:11.31 | Katty | [TK]D-Fender: oooh, so if it's in a certain time, it just changes the context. |
| 14:11.32 | cpm | I thought that was the point of Katty being a reseller, was to get summa dat price |
| 14:11.41 | Katty | no |
| 14:11.52 | Katty | the point of katty being a reseller is for pretty software for a receptionist, and tech support |
| 14:11.57 | Katty | katty doesn't make any money |
| 14:12.00 | Katty | katty is salary |
| 14:12.22 | *** join/#asterisk Fieldy (i=Yqub7lWa@gentoo/contributor/Fieldy) |
| 14:12.25 | Katty | katty just works in southern missouri for ten bucks an hour ;) |
| 14:12.53 | Katty | could probably go work for walmart, for that much... but at least this is fun |
| 14:13.18 | Katty | [TK]D-Fender: you know of a wiki example page where i can test this out? |
| 14:15.00 | [TK]D-Fender | Katty: jsut make an exten and test it. |
| 14:15.18 | [TK]D-Fender | Katty: The instruction sheet is clear. |
| 14:15.33 | Katty | yes, i suppose it is. |
| 14:15.36 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
| 14:15.46 | [TK]D-Fender | Katty talks about Katty's self in the 3rd person.... |
| 14:15.59 | Katty | naturally. |
| 14:15.59 | saftsack | hi does someone knows an octasic ec distributor where i can pay with a credit card? |
| 14:17.17 | cpm | pretty software for receptionists is key |
| 14:18.10 | Katty | and apparently, Fop Isn't Good Enough(tm) |
| 14:18.23 | cpm | So, who here does really clever turn key implementations? |
| 14:18.48 | Katty | so, clearly, we need some sort of database retrival software that pulls our clients records based on incoming callerid info |
| 14:21.21 | Katty | Hades and Hera on the same network. |
| 14:22.37 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 14:22.37 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:23.47 | zeeesh | <[TK]D-Fender> hi |
| 14:23.58 | [TK]D-Fender | zeeesh: y0 |
| 14:24.25 | saftsack | hasnt someone an octasic software ec? |
| 14:24.37 | [TK]D-Fender | Katty: Just need to name your domain "olympia" and you're set! |
| 14:24.44 | Katty | ha |
| 14:24.45 | zeeesh | <[TK]D-Fender> : making SIP call through xlite ... to cell phone ... connection has been made but there was no voice from both side ?? will u pls guide |
| 14:25.01 | [TK]D-Fender | zeeesh: ... |
| 14:25.02 | [TK]D-Fender | ~sipnat |
| 14:25.06 | jbot | hmm... sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:25.32 | zeeesh | <PROTECTED> |
| 14:25.36 | [TK]D-Fender | zeeesh: And go over that conference's recordings we did together:) |
| 14:26.37 | *** join/#asterisk punani (n=m@87.127.7.210) |
| 14:26.43 | Katty | so, i guess.... something like exten => 3000,1,GotoIfTime(8:00-17:00|mon-fri|*|*?newcontext and i dunno what to do after that |
| 14:27.19 | [TK]D-Fender | Katty: not just contexxt. you need to specify a priority (exten & context are optional) |
| 14:27.45 | Katty | so what would newcontext,s,1 do? |
| 14:28.03 | zeeesh | <[TK]D-Fender>: one more thing .. is there any issue .. if i m using non standard port except 5060??? |
| 14:28.06 | [TK]D-Fender | Katty: Assuming you actually want to CHANGE contexts |
| 14:28.07 | Katty | is that were it goes if it doesn't match? |
| 14:28.25 | [TK]D-Fender | zeeesh: No, and that has nothing to do with audio. Thats only the SIP signalling port |
| 14:28.27 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:28.32 | *** join/#asterisk edguy3 (n=edguy3@87-194-69-156.bethere.co.uk) |
| 14:28.35 | [TK]D-Fender | Katty: Next priority naturally :) |
| 14:28.47 | Katty | [TK]D-Fender: the word priority confuses me |
| 14:28.59 | Katty | is that the 3000,1 number? |
| 14:29.02 | Katty | prority 1? |
| 14:29.14 | Katty | oh. |
| 14:29.19 | [TK]D-Fender | Katty: extten => s,5,NoOp(I'm on priority 5, yippkiayay!) |
| 14:29.30 | Katty | so i could just skip context change and tell it to just keep going |
| 14:29.56 | Katty | well that wouldn't work either. |
| 14:30.04 | [TK]D-Fender | Katty: No. if it DOES match the time it will JUMP. if it DOESN'T then it will continue onward with the next |
| 14:30.10 | Katty | gotcha |
| 14:30.21 | Katty | context staying the same good |
| 14:30.28 | Katty | putting a lil jump somewhere in the loop isn't bad. |
| 14:30.34 | Katty | i can live with that |
| 14:30.42 | Katty | so, how do i skip the context change? another *? |
| 14:30.52 | *** join/#asterisk newvn (n=newvn@58.187.113.254) |
| 14:31.44 | Katty | (8:00-17:00mon-fri|*|*,4000,7) |
| 14:31.48 | Katty | like that? |
| 14:32.14 | [TK]D-Fender | Katty: http://www.pastebin.ca/507353 |
| 14:32.18 | Katty | thanks |
| 14:32.52 | Katty | ahh |
| 14:32.58 | Katty | but what if i want to move it... |
| 14:33.07 | Katty | to a new [area] and such |
| 14:33.22 | *** join/#asterisk oej (n=olle@69-94-197-20.biltmorecomm.com) |
| 14:33.22 | [TK]D-Fender | Katty: then you can specify wherever you want to go right away. |
| 14:33.37 | [TK]D-Fender | Katty: I just showed you the MINIMIM (changing priority) |
| 14:34.06 | [TK]D-Fender | Katty: you can jump to any specific spot you want just as the instructions say you can. |
| 14:34.15 | Katty | ya |
| 14:34.26 | *** join/#asterisk Acidcrawl (n=Miranda@12.168.96.254) |
| 14:34.28 | Katty | this is good. |
| 14:34.52 | Katty | i dunno why this is good |
| 14:34.53 | Acidcrawl | Is there any way to make a call that goes into a queue go to voicemail if nobody is logged into the queue? |
| 14:35.06 | Katty | maybe if someone wanted it to go to their cellphone after hours, rather than their voicemail after hours |
| 14:35.52 | Katty | or to a directory of cellphoens |
| 14:36.02 | Katty | cause i don't wanna rewrite my _XXX thingy |
| 14:36.04 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 14:36.15 | [TK]D-Fender | Acidcrawl: Yes, go read the sample config for the options to abort on. |
| 14:36.36 | Katty | you know, it really would help if they gave me some idea of what they wanted their dialplan to look like |
| 14:37.14 | [TK]D-Fender | Katty: Better approach : make sure of what they want to DO. Its up to YOU to make the dialplan fulfill their needs :) |
| 14:37.49 | Katty | well yeah |
| 14:37.56 | Katty | that's waht i meant, you goofball |
| 14:38.07 | Katty | now start reading my mind! |
| 14:38.32 | Katty | [TK]D-Fender: also, thanks for pastebin thingy. makes more sense now. |
| 14:39.40 | *** join/#asterisk iulius_ (n=iulius@mail1.technologieshq.com) |
| 14:40.12 | [TK]D-Fender | Katty: I *knew* you were going to do that! ;) |
| 14:40.26 | Katty | clearly. |
| 14:42.22 | mishehu | [TK]D-Fender: what you didn't know is that cookie is stale! |
| 14:42.37 | mishehu | it's what happens to cookies in Misery |
| 14:42.59 | *** join/#asterisk fujin (i=aj@unaffiliated/fujin) |
| 14:43.13 | [TK]D-Fender | mishehu: if croutons are just stale bread.... why are boxes of them sealed air-tight? :) |
| 14:43.24 | Acidcrawl | : [TK]D-Fender Well, I can see a couple of options, that prevent the queue from being joined if there are no agents, but it doesn't explain how to control what happends after, right now it just hangs up on the caller. |
| 14:43.40 | mishehu | [TK]D-Fender: I've always wonder that too. btw, why do we park on a driveway and drive on a parkway? |
| 14:43.59 | mishehu | [TK]D-Fender: why do our feet smell and our noses run? are we built backwards? |
| 14:45.50 | mishehu | hmm time to swap out this monitor... but need to clean up these bird feathers first... |
| 14:45.53 | Katty | mishehu: meeeeeesahhhhhh whoooo |
| 14:46.00 | mishehu | kitty katty |
| 14:46.08 | Katty | mishehu: how's chicago? |
| 14:46.13 | Katty | mishehu: i wanna come visit :< |
| 14:46.39 | mishehu | Katty: cluecon is coming up... tell those sales people to can it, and tell your boss he better send you |
| 14:46.58 | Katty | oh? when's cluecon this year? |
| 14:46.59 | Katty | august? |
| 14:47.07 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:47.23 | Katty | ooh, presents in warehouse *disappears* |
| 14:47.58 | mishehu | Katty: june |
| 14:50.42 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 14:51.06 | jmls | anyone help some poor sod with a headache regarding originate and dial status ? |
| 14:51.37 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 14:54.33 | LeddyHM | why is it called a "garage sale" when you aren't selling your garage? |
| 14:54.39 | Dovid | jmls: i am no expert but I can try to help. whats up ? |
| 14:54.44 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 14:54.52 | jmls | Dovid: thanks |
| 14:56.51 | jmls | I _want_ to be able to use originate to place calls. however I want to call the external number first, and then connect to a local channel that joins a meetme |
| 14:56.51 | jmls | I can do that |
| 14:56.51 | jmls | :) |
| 14:56.51 | jmls | however, I *also* want to be able to capture the dial status of the external call if the call does not go through |
| 14:56.52 | jmls | (busy, outoforder etc) |
| 14:56.52 | jmls | I can't seem to do that :( |
| 14:57.27 | [TK]D-Fender | Acidcrawl: it hangs up on the caller because you don't have a priority after calling Queue |
| 14:57.30 | jmls | best I can seem to do is to connect to a local channel first, dial the external numnber, run a maco on connection that does a meetme |
| 14:57.37 | jmls | is that the only solution |
| 14:57.38 | jmls | ? |
| 14:58.03 | Dovid | jmls: there is a variable that u can check. |
| 14:58.11 | Dovid | I think it is ${DIALSTATUS} |
| 14:58.25 | SwK | anyone know anything about voxbone? are they worth a damned? |
| 14:58.26 | [TK]D-Fender | jmls: Sure you can, but first you have to realize that your channel should not be SIP/IAX/ZAP :) |
| 14:58.45 | jmls | [TK]D-Fender: ? |
| 14:58.56 | *** join/#asterisk GuruJee (n=tilde@12.186.161.117) |
| 14:59.00 | GuruJee | hi everyone |
| 14:59.16 | GuruJee | i need the syntax for a macro, that dial calls on the Zap/g1 trunk |
| 14:59.21 | [TK]D-Fender | jmls: think about the channel you are using for your originate.... the answer is in there. Ask yourself HOW you could dial what you need to and be able to PROCESS the ATTEMPT :) |
| 14:59.22 | GuruJee | can anyone help me with that please? |
| 14:59.39 | jmls | [TK]D-Fender: I am using local channels |
| 14:59.48 | Katty | i smell blueberry muffins. |
| 14:59.49 | jmls | (see the "solution" above) |
| 15:00.02 | [TK]D-Fender | GuruJee: Macro's have nothing to do with dialing or ANYTHING specific really. Go lookup "stdexten" on the WIKI for some inspiration. |
| 15:00.21 | GuruJee | ok |
| 15:00.23 | [TK]D-Fender | jmls: pastebin what you've got NOW. |
| 15:00.29 | GuruJee | u are right, perhpas [TK] |
| 15:00.54 | GuruJee | what I am trying to do is to basically, recieve calls on IAX2 trunk and then dial them out on a pstn-trunk |
| 15:01.36 | *** join/#asterisk brea (n=brea@c-71-195-248-169.hsd1.ma.comcast.net) |
| 15:02.03 | [TK]D-Fender | GuruJee: Answer the call, then dial right back out. This isn't rocket science, its *1* line of dialplan. |
| 15:02.36 | [TK]D-Fender | GuruJee: No implicit need for Macro's or anything. - Answer, Dial, Hangup. |
| 15:02.48 | GuruJee | ok |
| 15:02.53 | GuruJee | thanks tk |
| 15:03.09 | GuruJee | can you help me with the syntax please. I have been working on thsi for the past 3 days and i am stuck |
| 15:03.25 | Katty | a former employee just called me in an attempt to sell me insurance. |
| 15:04.42 | [TK]D-Fender | GuruJee: there is nothing to say. "show application dial". |
| 15:04.49 | *** join/#asterisk renier (n=renier@69.79.111.24) |
| 15:04.54 | GuruJee | ok thanks |
| 15:04.57 | Dovid | Katty: i dont know which business is worse. We just had hardware failure. Loads of fun ;) |
| 15:05.06 | Katty | joy. |
| 15:05.23 | Katty | i would hate my life if i was a sales person. |
| 15:05.34 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:06.09 | Dovid | hehe |
| 15:06.25 | Dovid | i smoke on averge less than a pack. i am on pack2 today |
| 15:06.32 | Dovid | and to top it all off its my birthday :( |
| 15:06.33 | Katty | then again, our sales reps say they would kill themselves if they had to work with computers all day; |
| 15:06.37 | Dovid | hehe |
| 15:07.07 | Katty | you can threaten computers. |
| 15:07.11 | Katty | but not potential clients. |
| 15:07.15 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 15:07.17 | Katty | i win ^_^ |
| 15:07.29 | Katty | Dovid: sorry you're having such a crappy day )= |
| 15:07.39 | Katty | Dovid: just remember to breathe. take 5 minutes at a time. |
| 15:07.40 | Dovid | katty: all in a days work. |
| 15:07.51 | Dovid | Katty: i goto dig up my bottle of Jack.... |
| 15:07.58 | Katty | hehe |
| 15:08.05 | Katty | i'm more of a vodka and sprite person. |
| 15:08.09 | Dovid | ;) |
| 15:08.10 | Katty | or bailey's and milk |
| 15:08.48 | renier | hello. would there be a way to mute the mic of a sipphone at the asterisk level? maybe with some feature code? |
| 15:09.34 | Katty | oooh, i'd get in trouble with that. |
| 15:09.34 | Dovid | renier: i personally don't know of any. You can try putting the call in a meetme room |
| 15:09.35 | [TK]D-Fender | renier: not without recoding a ton of stuff to invent it. * is not a "mute" button. |
| 15:09.49 | PioneerVM2 | is h323 another peer protocol like sip? my provider supports h323 |
| 15:10.03 | [TK]D-Fender | renier: Waht kind of crap phone doesn't have a mute button? |
| 15:10.11 | PioneerVM2 | lol |
| 15:10.19 | Katty | [TK]D-Fender: one from 1960 |
| 15:10.26 | [TK]D-Fender | PioneerVM2: Yes. Go read THE BOOK for info on the various protocols * supports |
| 15:10.28 | [TK]D-Fender | ~book |
| 15:10.30 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:10.30 | Katty | [TK]D-Fender: that's what we had. we went from 1960 to polycom ;) |
| 15:10.30 | PioneerVM2 | why clutter your phone with buttons when you can just cover the moutpiece |
| 15:10.35 | GuruJee | hey guys |
| 15:10.37 | renier | [TK]D-Fender: gigaset s450 IP |
| 15:10.40 | Dovid | PioneerVM2: H323 went under a face change recently. I am using it and it works fine. |
| 15:10.41 | GuruJee | I need some help with DUNDi |
| 15:10.47 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 15:10.52 | GuruJee | [TK]. have u any idea about dudi? |
| 15:10.54 | Dovid | <--- Never used Dundi ;) |
| 15:10.55 | GuruJee | dundi* |
| 15:10.57 | PioneerVM2 | df: do you know how to view peers in asterisk for h323, similar to sip show peers |
| 15:11.08 | Dovid | Kattyl I would love to help you but I think I am past that point |
| 15:11.09 | PioneerVM2 | there is no "h323 show peers" |
| 15:11.12 | Katty | i think i remember meeting the guy who did dundi support for asterisk |
| 15:11.21 | Dovid | PioneerVM2: did u build H323 ? |
| 15:11.31 | PioneerVM2 | hmm maybe not |
| 15:11.41 | GuruJee | DUNDi anyone? |
| 15:11.42 | PioneerVM2 | that would explain it |
| 15:11.49 | Katty | file: did the dundi developer attend the cluecon i went to? |
| 15:11.51 | Dovid | PioneerVM2: do u have asterisk-addons instaled ? |
| 15:11.54 | Dovid | installed* |
| 15:11.56 | GuruJee | I want my system to bridge calls to pstn from a peer |
| 15:12.01 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 15:12.10 | GuruJee | my dundi for itnernal calls work |
| 15:12.15 | GuruJee | I cannot dial out to pstn, however |
| 15:12.16 | PioneerVM2 | i checked out the full 1.4.x latest version with svs |
| 15:12.30 | nick125_lappy | Anyone here play with asterisk 1.4.4? I'm wondering if I should move my personal voip system to 1.4.4 or leave it at 1.2.17 |
| 15:12.32 | Dovid | GuruJee: anything in the CLI or logs, any errors ? |
| 15:12.43 | GuruJee | dovid, nop |
| 15:12.48 | GuruJee | i have the iax2 debugging on |
| 15:13.09 | GuruJee | on the dialing phone, when i dial for example 713.715.XXXX I get a all lines are busy error |
| 15:13.19 | Dovid | nick125_lappy: i used 1.4.X a bit but I still dont feel comfortable with it. If its a personal system at home then it dosent hurt to try |
| 15:13.25 | PioneerVM2 | dovid: i see "--with-h323" in configure |
| 15:13.31 | GuruJee | on the IAX2 debug i see nothing. Which means that the there are no outbound routes |
| 15:13.33 | [TK]D-Fender | GuruJee: No |
| 15:13.43 | PioneerVM2 | i assume i need to install openh323 source and link to that? |
| 15:13.46 | nick125_lappy | Dovid: Was it stable when you played with it? |
| 15:13.46 | Dovid | GuruJee: i never tried but u can debug ur zaptel |
| 15:13.52 | Dovid | yes it was |
| 15:13.54 | GuruJee | but if i dial a 4 digit extension, it goes through |
| 15:14.06 | [TK]D-Fender | GuruJee: Don't expect any help here for your FreePBX problems.... |
| 15:14.09 | GuruJee | well, I think the call is not even passed on the IAX2 trunk |
| 15:14.15 | GuruJee | zaptel doesnt come into play |
| 15:14.18 | Dovid | TK: hehe :) |
| 15:14.25 | GuruJee | its not a freepbx dude |
| 15:14.26 | nick125_lappy | Dovid: What version? 1.4.4? |
| 15:14.31 | GuruJee | its Asterisk |
| 15:14.33 | [TK]D-Fender | GuruJee: And don't think we've forgotten either :) |
| 15:14.34 | Dovid | yes |
| 15:14.37 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 15:14.47 | GuruJee | honestly man, its not freepbx |
| 15:14.52 | GuruJee | its all asterisk |
| 15:15.01 | Dovid | PioneerVM2: what version of aasterisk r u using ? |
| 15:15.04 | GuruJee | all the inetrnal contexts are generic |
| 15:15.13 | PioneerVM2 | latest checked out of svs |
| 15:15.18 | PioneerVM2 | 1.4.4 i assume |
| 15:15.20 | Dovid | 1.4.xor 1.2.x ? |
| 15:15.31 | PioneerVM2 | i see /channels/h323 driver |
| 15:15.44 | Dovid | there are several h323 drivers |
| 15:15.51 | Dovid | i used this to get h323 working |
| 15:16.13 | PioneerVM2 | well before i go crazy, is it suggested to use h323 instead of sip if it is an option? |
| 15:16.37 | Dovid | nope. |
| 15:16.40 | Dovid | use sip if u can |
| 15:16.43 | PioneerVM2 | lol ok |
| 15:16.43 | Dovid | if u must then use h323 |
| 15:16.44 | PioneerVM2 | :) |
| 15:16.51 | PioneerVM2 | ok that solves that problem |
| 15:17.15 | PioneerVM2 | you use 1.4.x with sip by any chance? |
| 15:17.22 | Dovid | i am still on 1.2 |
| 15:17.33 | Dovid | I use SIP, H323 and IAX |
| 15:17.33 | PioneerVM2 | ok, im having probs with 1.4.4 and caller ID in sip |
| 15:17.48 | Dovid | what caller ID issues are you having ? |
| 15:18.17 | Katty | i'm having callerid issues too |
| 15:18.18 | PioneerVM2 | if I set CALLERID(num) to a number, and CALLERID(name) to anything that is not a number like "Joe", caller ID comes up empty |
| 15:18.19 | Katty | it's called dumb telco |
| 15:18.37 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 15:18.44 | PioneerVM2 | if I set CALLERID(num) to one number, and CALLERID(name) to a different one -- then the end caller sees the one I assigned to (name) |
| 15:18.53 | coppice | Katty: isn't the dumb there superfluous? |
| 15:18.54 | nick125_lappy | Katty: Which provider? :p |
| 15:19.08 | Dovid | PioneerVM2: Most providers wont allow u to pass CallerIDName |
| 15:19.20 | Dovid | most providers do thier own CNAM dips |
| 15:19.27 | Dovid | sorry carriers* |
| 15:19.49 | PioneerVM2 | the only way i can get it to work is to set (num) to a number and (name) to empty |
| 15:19.50 | Dovid | also some providers wont let you set CID at all. |
| 15:20.01 | Dovid | PioneerVM2: who is ur carrier ? |
| 15:20.41 | PioneerVM2 | dovid: yea i know that -- but i would figure they would ignore it |
| 15:20.47 | PioneerVM2 | dovid: well one problem is with VoicePulse |
| 15:21.05 | PioneerVM2 | dovid: voicepulse traced packets, and it seems they are seeing the data i set in (name) rather than (num) |
| 15:21.06 | Dovid | have u tried teliax or voipjet ? |
| 15:21.12 | PioneerVM2 | im trying agn voip now |
| 15:21.24 | Dovid | i had success with Teliax and Voipjet |
| 15:21.24 | PioneerVM2 | but i cant seem to set the # at all, im waiting to hear back from tech support |
| 15:21.37 | Dovid | do u have a debug output ? |
| 15:21.46 | PioneerVM2 | if what voicepulse is telling me is correct, then the "name" field is overriding the number field |
| 15:21.56 | PioneerVM2 | meaning sip protocol is sending the name as the number |
| 15:22.13 | PioneerVM2 | well i put up a whole post on it on bug boards finally i have to figure out how to get a simple debug of it |
| 15:22.33 | Dovid | PioneerVM2: i see no point in setting caller id name since almost *all* terminators will do thier own CNAM dips |
| 15:22.39 | PioneerVM2 | yes i know that |
| 15:22.46 | Dovid | in the CLI do: |
| 15:22.48 | PioneerVM2 | however (name) should not override (num) regardless |
| 15:22.50 | Dovid | SIP DEBUG |
| 15:22.54 | Dovid | Set Verbose 6 |
| 15:22.59 | Dovid | correct |
| 15:23.17 | PioneerVM2 | and what was happening, it turns out |
| 15:23.25 | PioneerVM2 | is that my original caller in data was getting set, s |
| 15:23.36 | PioneerVM2 | so if someone called in with caller ID "joe" <123> |
| 15:23.36 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 15:23.41 | PioneerVM2 | and i changed the number to "456" |
| 15:23.50 | PioneerVM2 | voicepulse was getting send "joe" for the number |
| 15:23.54 | PioneerVM2 | so it is coming up empty |
| 15:24.07 | PioneerVM2 | at least that is what they are saying, they traced packets with me on phone |
| 15:24.38 | PioneerVM2 | i didnt realize that name was getting set automatically |
| 15:24.45 | PioneerVM2 | until i did (name)="" |
| 15:24.49 | Dovid | PioneerVM2: if what u r saying is thrue then this is a BUG |
| 15:25.03 | PioneerVM2 | yea thats what im guesing, i checked everything like 20 times |
| 15:25.09 | PioneerVM2 | and before i made that change if i use IAX2 it is fine |
| 15:25.25 | PioneerVM2 | i just changed SIP to IAX2 and caller ID worked fine |
| 15:25.32 | Dovid | Katty: what kind of phone and are you behind NAT ? I had a client that has polycoms behind an expensiove router, It was droping the NOTIFY packers |
| 15:25.47 | PioneerVM2 | and now im having problems with AGN not taking my caller ID but im going to double check with them first to make sure its not something else in my account setting |
| 15:26.07 | Dovid | PioneerVM2: try SIP with Teliax. |
| 15:26.25 | PioneerVM2 | dovid: gotta go get the door brb |
| 15:26.42 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:26.45 | Dovid | kk |
| 15:26.47 | *** join/#asterisk ploieel (n=manni@Fb2c8.f.ppp-pool.de) |
| 15:26.55 | Katty | Dovid: 501s, and internal lan only |
| 15:26.57 | Dovid | hello Mercestes. good to see some pro's here :) |
| 15:27.03 | Katty | Dovid: rebooting phone fixed it. |
| 15:27.11 | Mercestes | :) |
| 15:27.14 | Katty | Mercestes: (= |
| 15:27.18 | Mercestes | Morning to you too |
| 15:27.23 | Katty | Mercestes: guess what guess what!!! |
| 15:27.26 | Mercestes | ?? |
| 15:27.29 | Mercestes | I'm hired? |
| 15:27.33 | Dovid | Katty: do a TCP DUMP on the server and then a trace on the phone. see if the phone is getting the packets |
| 15:27.35 | Katty | [TK]D-Fender helped me with gotoiftime stuff (= |
| 15:27.42 | Mercestes | oh....yay! |
| 15:27.44 | Katty | Dovid: uhh, no |
| 15:27.45 | Katty | Dovid: it's fine now |
| 15:27.51 | Katty | Dovid: i've better things to do than trace packets. |
| 15:27.53 | Dovid | okies. |
| 15:27.54 | Dovid | hehe |
| 15:28.05 | Katty | like....trace on the walls! |
| 15:28.10 | Katty | that's much more fun. |
| 15:28.18 | Mercestes | and happy trails |
| 15:28.21 | Mercestes | >.> |
| 15:28.39 | Dovid | Katty: I would like to know the cause of it, this client is pissing me off with the daily phone calls. This is especially pissing me off because we dont officially support Polycoms |
| 15:28.53 | Katty | ^_- |
| 15:28.56 | irule | anybody have the link to download the free classical music? thanks |
| 15:28.57 | Katty | right. |
| 15:29.15 | Mercestes | irule, um....emerge asterisk-addons ?? |
| 15:29.24 | Mercestes | s/asterisk-addons/asterisk-sounds/ |
| 15:29.27 | [TK]D-Fender | irule: "the" free classical music? |
| 15:29.27 | Dovid | irule: wana calm down pissed off clients ? ;) |
| 15:29.30 | Mercestes | ~botsnack |
| 15:29.30 | jbot | Mercestes: aw, gee |
| 15:29.42 | irule | yes hehe |
| 15:30.19 | Mercestes | apt-get pantera |
| 15:30.26 | [TK]D-Fender | irule: "the" implies ONE source/kind of item. |
| 15:30.41 | [TK]D-Fender | Mercestes: Don't forget to add Stryper too! :) |
| 15:30.50 | Mercestes | Brar! :D |
| 15:30.52 | [TK]D-Fender | Mercestes: And WASP for good measure! |
| 15:31.00 | [TK]D-Fender | Mercestes: Or GWAR ;) |
| 15:31.21 | [TK]D-Fender | irule: Heres a thought : http://freeplaymusic.com/ |
| 15:31.26 | Mercestes | or anthrax |
| 15:31.41 | Mercestes | Or Johnny Cash! ....oh wait..no I screwed it up |
| 15:31.43 | Mercestes | damnit |
| 15:31.45 | Mercestes | stupid play list |
| 15:31.48 | PioneerVM2 | dovid: with teliax, what option do i choose |
| 15:32.01 | Dovid | u want pay as u go |
| 15:32.02 | PioneerVM2 | just normal pay as you go plan using "My own sip devices"? |
| 15:32.07 | irule | [TK]D-Fender thanks |
| 15:33.31 | Dovid | pioneerVM2: I believe so |
| 15:33.37 | Dovid | let me hop on thier sit |
| 15:33.39 | Dovid | site* |
| 15:33.50 | PioneerVM2 | yea i just went to products and services, pay as u go |
| 15:33.53 | PioneerVM2 | then under "choose hardware" |
| 15:34.01 | PioneerVM2 | you can jump right to that on top |
| 15:34.19 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 15:34.23 | Dovid | pioneerVM2: Pay as you go and then use my own SIP device |
| 15:34.34 | PioneerVM2 | these guys any good or only for simple stuff |
| 15:34.40 | Dovid | i personally like IAX better but choose SIP for now to trouble shoot. i can allways switch it with them later |
| 15:35.18 | PioneerVM2 | yea i had IAX be a little choppy with voicepulse so went to sip |
| 15:35.24 | Dovid | PioneerVM2: i used to use them as my primary terminator till I started working for a terminator (where I get termination for 0 to nothing.... ) |
| 15:35.34 | *** join/#asterisk doolph (n=doolph@200.105.35.219) |
| 15:35.50 | doolph | anyone has experience with nortel? |
| 15:35.53 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 15:35.54 | Dovid | pioneerVM2: Then go for SIP and see what happens |
| 15:36.25 | PioneerVM2 | well i want to trouble shoot this too so ill do sip |
| 15:36.31 | Dovid | okies. good luck |
| 15:37.09 | [TK]D-Fender | doolph: Yes, I owned stock.... |
| 15:37.40 | Dovid | anyone have an AGI to connect to XM radio ? |
| 15:38.02 | Mercestes | PioneerVM2, how can anyone that advertises in IRC via a username have any problems with their audio??? |
| 15:38.10 | doolph | ok I have an office, with an nortel extension ip phone |
| 15:38.22 | Mercestes | doolph: I will pray for you tonight |
| 15:38.23 | doolph | but I want more phones, so i can transfer calls, etc |
| 15:38.29 | doolph | lol |
| 15:38.53 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 15:39.08 | [TK]D-Fender | doolph: And this has WHAT to do with *? |
| 15:39.27 | *** join/#asterisk shodan (n=shodan@ip120.96-113-216.pppoe1.joliette.intermonde.net) |
| 15:39.28 | doolph | yes |
| 15:40.02 | doolph | I want to connect an * instead of that ip phone |
| 15:40.13 | Mercestes | do you have a printer and a soldering iron? |
| 15:40.38 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:41.21 | Mercestes | neverblue, good timing. I was just thinking about you. |
| 15:41.27 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 15:41.52 | [TK]D-Fender | doolph: Time to pay them for a SIP license... |
| 15:43.20 | doolph | I dont have access to the nortel server |
| 15:43.31 | Mercestes | doolph: no kidding? |
| 15:43.36 | Mercestes | doolph: Did you ask them for it? |
| 15:43.41 | doolph | I only have an ip phone from them |
| 15:43.51 | doolph | I don't think they want me touch that |
| 15:44.12 | neverblue | Mercestes, ? |
| 15:44.13 | [TK]D-Fender | doolph: Sorry, there is no "miracle" tool to make a Nortel system you don't have access to bend to your will... |
| 15:44.14 | Mercestes | what? no way. They gave me full access just for the asking. Give your rep a call. bet he'll email you the admin guide. |
| 15:44.35 | Mercestes | it's not really hard at all. What kind of connectors do you have on that Nortel? |
| 15:44.39 | doolph | really that's bad |
| 15:44.59 | neverblue | need a hand deciding between an Intel and an AMD system |
| 15:45.06 | neverblue | anyone "knowledgeable" ? |
| 15:45.07 | Mercestes | not really. Nortel's are really easy to program. nice guys |
| 15:45.27 | Mercestes | neverblue, no. But doolph is trying to plug an * into a Nortel IP phone. Can you walk him through it? |
| 15:45.58 | neverblue | Mercestes, I doubt it |
| 15:45.58 | doolph | is that hard? |
| 15:45.59 | cpm | Mercestes, see the big hoopla betwixt Nortel and Fonality last week? |
| 15:45.59 | Mercestes | I'd do it but I'm a little busy righ tnow. |
| 15:46.01 | Mercestes | cpm: Yes, yes I did. :) |
| 15:46.13 | doolph | Mercestes can you help me later? |
| 15:46.14 | cpm | pretty fun, that. |
| 15:46.17 | Mercestes | doolph: Not at all, Neverblue is the resident expert on that. |
| 15:46.20 | doolph | or at least give me some tips |
| 15:47.32 | [TK]D-Fender | doolph: You cannot have * act as a phone to the BCM without paying for a SIP channel license. |
| 15:47.36 | [TK]D-Fender | doolph: "It's not possible to connect a Nortel Succession/Meridian/BCM to Asterisk via chan_unistim. Use either E1/T1 trunks, or buy UTPS (UNISTIM Terminal Proxy Server) from Nortel. " |
| 15:47.55 | neverblue | doolph, I wouldn't really listen to what Mercestes is saying, he is just yanking your chain |
| 15:48.19 | [TK]D-Fender | doolph: indeed.... Mercestes likes to "yank" ;) |
| 15:48.27 | doolph | :/ |
| 15:48.33 | doolph | so I cannot do nothing |
| 15:48.46 | Katty | i leave for 10 minutes...and Mercestes goes nuts?! |
| 15:48.53 | Katty | hopeless. utterly hopeless. |
| 15:49.08 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 15:49.12 | Katty | at least the dell music on hold in this queue is nice. |
| 15:49.13 | Mercestes | doolph: I wouldn't say "nothing." But it's far more complex/difficult then you think. |
| 15:49.35 | doolph | I though I can connect that rj45 port to an E1 port |
| 15:49.47 | Mercestes | doolph: You could just run a T1 trunk to the Nortel but without knowing how the Nortel is programmed that could be an endeavor into futility. |
| 15:49.47 | doolph | and do something with asterisk |
| 15:49.54 | [TK]D-Fender | doolph: Ethernet != T1. |
| 15:50.07 | Mercestes | doolph: What you *really* need to do is scrap that nortel POS and go * from start to finish and do it right the first time instead of trying to hack it |
| 15:50.12 | [TK]D-Fender | doolph: I've told you what you have to do. No go do it or stop whining! |
| 15:50.52 | Mercestes | doolph: And .....how do you connect a rj45 port to a e1 port?? |
| 15:50.59 | Mercestes | doolph: your into lesbians aren't you? |
| 15:51.21 | doolph | something like that |
| 15:51.23 | doolph | i dont know |
| 15:51.36 | neverblue | mmm lesbians |
| 15:51.43 | Mercestes | It's what's for dinner! |
| 15:52.10 | PioneerVM2 | Dovid: u there? |
| 15:52.27 | PioneerVM2 | dovid: I got teliax working right away, and SIP caller ID worked no problem -- just set the num and it worked |
| 15:52.39 | Mercestes | PioneerVM2, feels good, doesn't it? |
| 15:53.08 | neverblue | intel or AMD? |
| 15:53.14 | Dovid | hehe |
| 15:53.21 | Mercestes | neverblue, desktop or laptop? |
| 15:53.25 | neverblue | desktop |
| 15:53.32 | Mercestes | amd |
| 15:53.35 | Dovid | pioneerVM2: So it may be an issue with ur orig. carrier |
| 15:53.49 | neverblue | same price? |
| 15:53.49 | Mercestes | unless it's a Dell PE server. then Intel |
| 15:53.57 | PioneerVM2 | yea im wondering |
| 15:54.19 | Mercestes | neverblue, last I checked AMD was cheaper, but.....very similiar in price |
| 15:54.30 | neverblue | same price, AMD or Intel |
| 15:54.58 | neverblue | im almost tempted to do Intel |
| 15:55.07 | Mercestes | don't do it |
| 15:55.11 | neverblue | ha |
| 15:55.21 | neverblue | i never take anything you say seriously |
| 15:55.30 | neverblue | so ill just get Intel to spite you |
| 15:55.48 | neverblue | intel 2 AMD 0 |
| 15:55.50 | Mercestes | and then you cry that no one likes you. |
| 15:55.57 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 15:56.43 | anonymouz666 | anyone in here already tried to transfer a call from SIP to Google talk ? |
| 15:57.44 | anonymouz666 | it does not work. |
| 15:58.08 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 15:59.13 | *** join/#asterisk ghenry (n=ghenry@fw2-inet.semantico.net) |
| 15:59.28 | PioneerVM2 | dovid: im on with voicepulse now going to try and trouble shoot agian, they are not very friendly though on the phone |
| 15:59.32 | anonymouz666 | only mog can answer that |
| 15:59.58 | PioneerVM2 | no matter what you call with the guy acts like he has no desire to deal with people |
| 16:00.27 | PioneerVM2 | agn networks is worse, they wont even talk on the phone -- they force me thru email even after they wont answer my questions without canned responses. |
| 16:00.34 | Mercestes | PioneerVM2, Again, I can't imagine that a company that advertises via an IRC username could have any problems, especially with Customer Service. |
| 16:00.58 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 16:01.52 | *** join/#asterisk berktr (n=canberk@212.156.213.131) |
| 16:04.09 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:05.40 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:05.43 | ManxPower | All ISTPS suck when it comes to customer service. |
| 16:05.51 | Qwell | ManxPower: not all |
| 16:05.57 | ManxPower | Teliax at least when I was using them was pretty resonsive. |
| 16:05.58 | Qwell | but the ones that don't, suck in other ways |
| 16:06.23 | ManxPower | Qwell: They all suck in some why. If their customer service does not suck then they are too small to consider. |
| 16:06.31 | Mercestes | Katty, eat chinese |
| 16:07.02 | Mercestes | ManxPower, I never heard you complain about anyone sucking before. |
| 16:07.03 | Katty | Mercestes: i had that yesterday. twice |
| 16:07.09 | PioneerVM2 | merc: not sure i get what u mean |
| 16:07.37 | PioneerVM2 | merc: i assume you wer emaking a joke :) |
| 16:07.40 | Mercestes | PioneerVM2, Oh, to put it another way "surprise, surprise." |
| 16:07.48 | PioneerVM2 | lol |
| 16:07.50 | Mercestes | PioneerVM2, it's only a joke ifyou get it. ;) |
| 16:07.57 | GuruJee | <PROTECTED> |
| 16:07.57 | Mercestes | Katty, hmm..what goes good after chinese? |
| 16:08.03 | PioneerVM2 | yea i couldnt tell if you were picking on me or making a joke :) |
| 16:08.04 | Mercestes | Katty, Italian |
| 16:08.08 | Katty | Mercestes: hrmm. italian |
| 16:08.09 | Mercestes | PioneerVM2, yes. |
| 16:08.20 | PioneerVM2 | the guys at voicepulse sent me a bunch of examples of why i was wrong, and it turns out they show i was right |
| 16:08.29 | PioneerVM2 | im on with them now and they are al lfrustrated with me lol |
| 16:08.40 | PioneerVM2 | they are like "whats your point", my point is that your systems are handling caller ID info wrong |
| 16:08.40 | ManxPower | PioneerVM2: So, was *I* right? |
| 16:08.49 | PioneerVM2 | manx: possibly im waiting to see what they say |
| 16:09.08 | PioneerVM2 | manx: they were tracing packets and saying they saw the data a certain way but i think they were not telling me the right thing |
| 16:09.27 | PioneerVM2 | if what they told me originally about the traced packets was true, then they are right, if not then they are wrong -- i tested teliax and it works fine |
| 16:09.59 | PioneerVM2 | it seems with voicepulse, that if you set (num) and (name) is set to ANYTHING (a number or a name) then they take what is in the (name) field and use that |
| 16:10.12 | PioneerVM2 | so if you set (num) you have to set (name)="" |
| 16:10.24 | PioneerVM2 | they should be ignoring (name) |
| 16:10.48 | PioneerVM2 | and im not really "setting" name anyway, its being passed thru from the original incoming call before i forward it -- |
| 16:10.53 | Mercestes | that's completely wrong |
| 16:10.54 | PioneerVM2 | so i have to specifically empty it with them |
| 16:10.56 | PioneerVM2 | yes of course :) |
| 16:11.13 | PioneerVM2 | these guys are obnoxious, at least they actually talk with me though |
| 16:11.29 | Mercestes | google BOFH |
| 16:11.38 | PioneerVM2 | i cant even talk to agn and it took them 5 days to setup my account when they advertise "provisioned in 30 minutes" |
| 16:11.49 | tzanger | morning everyone |
| 16:11.55 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 16:11.55 | Mercestes | moring tzanger |
| 16:12.10 | *** join/#asterisk bbryant (n=brett@69-94-196-94.biltmorecomm.com) |
| 16:12.20 | PioneerVM2 | now i just have to figure out why i cant set caller ID on AGN, just doesnt do anything, teliax worked fine |
| 16:12.36 | Dr-Linux | need some idea, what's the best opensourse incoming call notification program out there? |
| 16:13.44 | ManxPower | PioneerVM2: I'll bet AGN is simply growing fast. That is usually when the customer service goes to hell. |
| 16:14.04 | PioneerVM2 | yea like vonage |
| 16:14.09 | PioneerVM2 | vonage is great unless you need to talk with someone |
| 16:14.21 | PioneerVM2 | yea well sales answered my calls all the time |
| 16:14.24 | Mercestes | Dr-Linux: A ringer. :D |
| 16:14.48 | Dr-Linux | Mercestes: link? |
| 16:15.10 | Dr-Linux | http://www.voip-info.org/wiki/view/Asterisk+call+notification |
| 16:15.18 | Dr-Linux | here are alot, but not sure which is best |
| 16:15.36 | Mercestes | Dr-Linux, http://en.wikipedia.org/wiki/Ring_%28telephone%29 |
| 16:15.38 | nick125_lappy | Can you mix AEL and regular dial plan? I think AEL would be really cool for writing some features, but, overkill for anything else.. |
| 16:16.07 | Dr-Linux | Mercestes: okey i've a question about dialplan |
| 16:16.43 | Mercestes | By all means |
| 16:17.09 | PioneerVM2 | idiots at voicepulse |
| 16:17.18 | PioneerVM2 | they are like "oh yea we talked to our engineers and that is currently how it works" |
| 16:17.29 | PioneerVM2 | "we may fix it in future versions" |
| 16:17.47 | PioneerVM2 | i was like, it would be nice if you told customers that so they dont waste 3 days on this crap, and if you dont tell them they are wrong |
| 16:17.49 | Dr-Linux | Mercestes: |
| 16:17.50 | Dr-Linux | exten => 1234,1,DBget(USERx_IP=SIP/Registry/userx) |
| 16:17.50 | Dr-Linux | <PROTECTED> |
| 16:18.11 | Dr-Linux | how can i add users and their ips in DB? :S |
| 16:18.29 | Mercestes | s/we talked to our engineers/we googled it/ |
| 16:18.33 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 16:19.12 | Mercestes | Dr-Linux, you mean like in RTA? |
| 16:19.27 | PioneerVM2 | if you "don't support caller ID names", why the heck would you even look at the name field, let alone use the data in the name field as a number |
| 16:19.28 | Dr-Linux | i'm sorry i don't know what's RTA |
| 16:19.43 | [TK]D-Fender | Dr-Linux: Yes. |
| 16:20.17 | [TK]D-Fender | nick125_lappy: Yes. |
| 16:20.27 | *** join/#asterisk djPepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
| 16:20.32 | Dr-Linux | but, i need to store IP addresses for each user, and system will send call through the IP address against the IP address |
| 16:20.35 | djPepse | Morning everyone |
| 16:20.42 | ManxPower | PioneerVM2: no company support sending the CALLERID name, BTW. |
| 16:21.09 | ManxPower | some allow you to set it and it will work on their own network. |
| 16:21.42 | PioneerVM2 | yea i know that |
| 16:21.44 | ManxPower | When a call comes in from outside a carrier's network they normally do a database dip (CNAME DIP?) to look up the name associated with the received number. |
| 16:21.45 | *** join/#asterisk nays85 (i=nays85@shell.thehostbusters.com) |
| 16:21.48 | djPepse | I'm having a strange problem but I know most of you will cringe when I describe it. I'm using a Skype gateway (Uplink Skype2Sip) and I can get audio in both directions when I call out to a Skype user. When a Skype user calls me, however, the phone rings and audio only goes one way. |
| 16:21.51 | blitzrage | we're going to be able to support CIDNAME once I subscribe to some 411 service with Verizon, but we haven't peered with it yet |
| 16:21.59 | PioneerVM2 | the problem was that th eincoming call was being forwarded, and by default asterisk was keeping the "name" data that came in |
| 16:21.59 | djPepse | The skype gateway and the pbx are on the same local network, so I'm not sure what the deal is |
| 16:22.11 | blitzrage | "I'm using a Skype..." |
| 16:22.21 | PioneerVM2 | and when i called in, both name and number were, for example set to "123" <123> |
| 16:22.23 | ManxPower | blitzrage: me too. |
| 16:22.29 | PioneerVM2 | and when i called out and tried to set (num)="456" |
| 16:22.32 | djPepse | That's why I prefaced it with "I know most of you will cringe" |
| 16:22.33 | nays85 | are you still talking about your CID problem? |
| 16:22.36 | PioneerVM2 | they would take the "123" from the name field and use that |
| 16:22.44 | PioneerVM2 | or whatever else was in there |
| 16:22.48 | ManxPower | PioneerVM2: sounds evil to me. |
| 16:23.16 | GuruJee | hey guys |
| 16:23.22 | GuruJee | how can I connect two asterisk servers together |
| 16:23.34 | GuruJee | so that 1 server can have a PRI card, and other doesnt. |
| 16:23.52 | GuruJee | The server without the pri card may still be able to make pstn calls via other server |
| 16:23.59 | [TK]D-Fender | GuruJee: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers&view_comment_id=13799 |
| 16:24.10 | djPepse | Please, is there anything simple that I'm forgetting to do? Why would I only get audio one way on the same local network? |
| 16:24.15 | GuruJee | thanks tk |
| 16:24.38 | doolph | djPepse ports? |
| 16:24.39 | nays85 | PioneerVM2 : sounds like a 1 line fix to me, what took you 3 days |
| 16:24.56 | PioneerVM2 | nays: yea annoying |
| 16:24.57 | Dr-Linux | [TK]D-Fender: so can you give me hints, how can i add ips to the DB? |
| 16:25.10 | PioneerVM2 | nays: but it should not have needed to be anyway, so as far as im aconcerned its a "workaround" |
| 16:25.15 | doolph | any good asterisknow ? |
| 16:25.15 | djPepse | doolph: hm, i'm pretty sure i've disabled windows firewall but lemme doublecheck |
| 16:25.21 | Dr-Linux | [TK]D-Fender: here is dialplan: |
| 16:25.21 | Dr-Linux | exten => 1234,1,DBget(USERx_IP=SIP/Registry/userx) |
| 16:25.21 | Dr-Linux | <PROTECTED> |
| 16:25.24 | PioneerVM2 | nays: and the engineer told me the wrong debug info and made me believe it was asterisk |
| 16:25.31 | nays85 | PioneerVM2 : different providers work different ways, it takes 2 seconds to accomodate differences in extensions.conf |
| 16:25.32 | [TK]D-Fender | Dr-Linux: Go play around with it a bit. |
| 16:25.45 | PioneerVM2 | nays: yes but they told me the wrong info, so i didnt realize it |
| 16:25.48 | Dr-Linux | hhm.. ok |
| 16:25.49 | [TK]D-Fender | doolph: Good for WHAT is the question.... |
| 16:25.55 | djPepse | Yeah, firewall is turned off :/ |
| 16:26.04 | PioneerVM2 | nays: they didnt even know how their own caller ID worked |
| 16:26.33 | nays85 | PioneerVM2 : if you know the name is going out as the number, why don't you just set it to whatever you want it to be? |
| 16:26.45 | PioneerVM2 | because i use multiple providers |
| 16:26.45 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:27.06 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 16:27.18 | PioneerVM2 | and i dont want to do tons of hacks to make it work -- i wanted to find out what the problem was anyway.. when things dont work the right way you usually want to get them working correctly |
| 16:27.24 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 16:27.27 | PioneerVM2 | and also so it doesnt break when its fixed in the future |
| 16:27.28 | BSD_Tech | mornig |
| 16:27.28 | djPepse | Does it make sense that 2way audio is fine with outgoing calls (when Asterisk calls Skype) but only 1way audio on incoming calls (when Skype calls Asterisk)? |
| 16:27.45 | nays85 | So you want a provider to make a system-wide change that will affect thousands of users when there's a 1 line workaround on your end... sounds logical |
| 16:27.55 | NoCarrier | djPepse: are any of them behind NAT? |
| 16:28.02 | PioneerVM2 | nays85 -- they dont even realize that it works that way |
| 16:28.03 | djPepse | NoCarrier: no |
| 16:28.03 | *** join/#asterisk dr_decimal (n=stefan@c-68-45-144-101.hsd1.pa.comcast.net) |
| 16:28.04 | NoCarrier | djPepse: I mean "either of them" |
| 16:28.11 | djPepse | NoCarrier: well, both of them together are, on the same local net |
| 16:28.12 | PioneerVM2 | nays: its not that they are saying "do it this way and it will work" |
| 16:28.21 | NoCarrier | djPepse: and then what's the other endpoint? |
| 16:28.24 | PioneerVM2 | nays: and yes, its a bug they should fix it properly |
| 16:28.36 | NoCarrier | djPepse: skype <--> Asterisk <--> ??? |
| 16:28.42 | PioneerVM2 | nays: or at least put a FAQ that tells you exactly how to do it, not say "its your fault" |
| 16:28.42 | *** part/#asterisk dr_decimal (n=stefan@c-68-45-144-101.hsd1.pa.comcast.net) |
| 16:28.43 | NoCarrier | djPepse: what's "???" |
| 16:28.54 | djPepse | NoCarrier: ATA on the local network |
| 16:29.06 | BSD_Tech | http://pastebin.ca/507487 a gift for all |
| 16:29.23 | NoCarrier | when Asterisk replies to skype, it's probably giving the ATA's private IP in the SDP endpoint info |
| 16:29.37 | NoCarrier | or it's own private IP in the SDP endpoint info |
| 16:29.45 | NoCarrier | so the RTP stream from skyp back to you never reaches you |
| 16:29.51 | PioneerVM2 | nays: its like a Shell Gas stations having mislabled gas -- 87 = 91 and vice versa. But not telling you and not knowing that its like that -- and thinking its right for them to just leave it that way |
| 16:29.55 | nays85 | they tell you how to set it in the example config... even if it is a bug, they can't just blindly change how it's been working for months and screw everyone else because you ask them to |
| 16:30.23 | PioneerVM2 | nays: they set the caller ID using a depricated option |
| 16:30.35 | Strom_C | PioneerVM2: so stop bitching already and find another carrier |
| 16:30.41 | nays85 | PioneerVM2 : they have to support every version of asterisk, not just your version |
| 16:30.52 | PioneerVM2 | nays: its not asterisk. |
| 16:31.08 | PioneerVM2 | ugh forget it, you are just out to argue everything |
| 16:31.10 | djPepse | NoCarrier: Hm. but since it's all private IPs, shouldn't Skype be able to send rtp stream to the *? |
| 16:31.19 | nays85 | PioneerVM2 : what is it then? |
| 16:31.22 | PioneerVM2 | strom: sorry nays was just being difficult |
| 16:31.35 | jmls | [TK]D-Fender: ping |
| 16:31.35 | PioneerVM2 | nays: they are not following the proper protocol for setting caller ID. period. |
| 16:31.42 | [TK]D-Fender | jmls: pong |
| 16:31.45 | jmls | http://www.pastebin.ca/507498 |
| 16:31.56 | djPepse | I can see * doing "Packet2Packet bridging SIP/XXX and SIP/XXXX" |
| 16:32.41 | redax | good afternoon, |
| 16:32.42 | nays85 | PioneerVM2 : what is it then? |
| 16:32.48 | jmls | [TK]D-Fender: sorry, channel should read "Channel: Local/dialout@dialout/n" |
| 16:32.53 | PioneerVM2 | nays: what is what |
| 16:33.06 | nays85 | PioneerVM2 : what are you using if it's not asterisk |
| 16:33.13 | PioneerVM2 | i am using asterisk |
| 16:33.28 | djPepse | NoCarrier: Could it be that it's trying different RTP ports than what * is expecting? |
| 16:33.39 | NoCarrier | djPepse: one sec, otp |
| 16:34.30 | [TK]D-Fender | jmls: ... and? Looks like it went to do "its thingA" just fine. Whats the issue? |
| 16:34.32 | ManxPower | watching the sip debug will tell you if it is sending out it's wrong IP address |
| 16:35.17 | NoCarrier | djPepse: no, skype would try whatever * sends it |
| 16:35.36 | djPepse | debug on what? |
| 16:35.37 | *** part/#asterisk jtexter3 (n=jtexter3@69-94-196-106.biltmorecomm.com) |
| 16:35.39 | NoCarrier | djPepse: can you just use ethereal to capture the traffic between * and skype and then look at it? |
| 16:36.00 | djPepse | NoCarrier: I may be able to |
| 16:36.58 | NoCarrier | djPepse: I think that's your best bet |
| 16:37.09 | djPepse | i'll try using smsniff on the machine running skype, so i don't have to go hunting for a hub |
| 16:38.14 | *** join/#asterisk ccesario (n=ccesario@201-0-124-6.dsl.telesp.net.br) |
| 16:38.36 | djPepse | What should I look for specifically? |
| 16:38.42 | ManxPower | djPepse: sip debug would be easier to parse. |
| 16:38.59 | ManxPower | and still provides quite a bit of info. |
| 16:39.15 | djPepse | ManxPower: do I just set the vorbicity higher? |
| 16:41.08 | djPepse | verbosity. i did just get a capture tho. |
| 16:41.31 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-5579be0fc4a225a8) |
| 16:41.48 | jmls | [TK]D-Fender: I'm back !] |
| 16:42.05 | ManxPower | djPepse: you don't. "sip debug" will print every sip message on the console |
| 16:42.35 | Dr-Linux | hhm.. no luck |
| 16:42.56 | Dr-Linux | maybe i should grab ManxPower brain :P |
| 16:42.57 | djPepse | not sure what I'm looking for. I see it inviting and identifying itself with the internal ip |
| 16:43.27 | ManxPower | djPepse: then that might be your problem. |
| 16:43.50 | NoCarrier | djPepse: what about the SDP endpoint info |
| 16:44.04 | *** join/#asterisk Dr-Linux|work (n=Nothing@202.125.139.198) |
| 16:44.10 | ManxPower | djPepse: From what little I understand about that FrankenPBX/Skype abomination, perhaps you need to set up Asterisk as being behind NAT. |
| 16:44.11 | Dr-Linux|work | back |
| 16:44.13 | NoCarrier | djPepse: you don't have a signaling problem - just a media problem |
| 16:44.30 | djPepse | ManxPower: everything is on the local network, skype, *, and ATA |
| 16:44.38 | djPepse | NoCarrier: looking for SDP |
| 16:44.45 | ManxPower | djPepse: Yes, but the calls go out to the internet at some point. |
| 16:44.57 | NoCarrier | djPepse: also, do you have "canreinvite=no" for the skype peer? |
| 16:45.09 | ManxPower | djPepse: I assume you do NOT have the same problem if you remove asterisk from the setup? |
| 16:45.09 | jmls | [TK]D-Fender: what I need to be able to do is to pass the zap channel the call was connected to along to the second channel where the meetme is about to take place |
| 16:45.15 | djPepse | I also see something that says m=audio 8180 RTP |
| 16:45.20 | djPepse | shouldn't it be expecting 10000+ |
| 16:45.21 | djPepse | ? |
| 16:45.27 | [TK]D-Fender | jmls: looks like its doing that already. Whats the problem? |
| 16:45.31 | jmls | [TK]D-Fender: you see that the TESTVARIABLE although set with inheritance does not make it through |
| 16:45.45 | djPepse | ManxPower: I have 2 way audio if I call skype from the outside |
| 16:46.14 | ManxPower | djPepse: when do you get one-way audio. |
| 16:46.15 | djPepse | Skype should be handling audio communication with the public networks on its own, nothing to do with the gateway or the * (i would think) |
| 16:46.23 | ManxPower | one-way audio is almost always a NAT issue. |
| 16:46.31 | ManxPower | djPepse: don't count on that. |
| 16:46.36 | djPepse | ManxPower: My mistake, I said it backwards. One-way audio when I call in from outside, 2way when I call out from inside |
| 16:46.49 | Dr-Linux|work | ManxPower: please observe my question :) |
| 16:46.49 | ManxPower | H.323, SCCP/Skinny, SIP, and MGCP all use the SAME protocol for the audio (RTP). |
| 16:46.51 | djPepse | NoCarrier: I have canreinvite=yes |
| 16:46.52 | Dr-Linux|work | ManxPower: http://phpfi.com/236646 |
| 16:47.07 | djPepse | NoCarrier: and caninvite=yes |
| 16:47.20 | ManxPower | djPepse: set canreinvite=no |
| 16:47.42 | ManxPower | Dr-Linux|work: your question is not interesting enough for me. |
| 16:47.45 | [TK]D-Fender | jmls: I'd be betting thats because the first channel does not DIAL the second. |
| 16:47.52 | Dr-Linux|work | ManxPower: i want call notification on the windows client, all done, tested working, but little things remains i.e. how can i add ips against users in asterisk DB |
| 16:47.55 | Dr-Linux|work | opss :( |
| 16:47.56 | djPepse | ManxPower: Will try, but uplink docs say to use =yes :) |
| 16:48.05 | *** join/#asterisk ToyMan (n=Stuart@74-32-22-252.dsl1.mdl.ny.frontiernet.net) |
| 16:48.27 | Dr-Linux|work | ManxPower: you didn't understand my english or something else :S |
| 16:48.32 | ManxPower | djPepse: As I said, your symptom of one-way audio on incoming calls is a class NAT sympton |
| 16:48.43 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 16:48.47 | vAd0r | Why is it that i can register w/ my ip but not the a record that points to it even though it resolves to the ip |
| 16:48.52 | djPepse | changed canreinvite, no difference tho |
| 16:48.55 | ManxPower | Dr-Linux|work: I said your question is not interesting enough for me to do the significant amount of work to help you. |
| 16:48.56 | [TK]D-Fender | jmls: have you tried accessing the variable with "_" , and NO prefuix as well? |
| 16:49.23 | ManxPower | vAd0r: I have no idea what you just said. |
| 16:49.26 | djPepse | ManxPower: weird. shouldn't i have the same problem if I were to remove the * gateway? |
| 16:49.27 | Dr-Linux|work | ok, btw i'd need some hints .. anyway thanks |
| 16:49.43 | ManxPower | djPepse: I did ask you if that was the case. |
| 16:49.43 | [TK]D-Fender | ManxPower: He's talking about DNS lookups |
| 16:50.07 | djPepse | I'm going to triple-verify that.. |
| 16:50.13 | [TK]D-Fender | vAd0r: please PASTEBIn some backup for any of these kinds of questions. we will always want to see what you're doing. |
| 16:50.39 | jmls | [TK]D-Fender: gotta run, just tried with _, __, and nothing, same result. |
| 16:50.52 | djPepse | ManxPower: yeah, audio works fine when * gateway is removed |
| 16:51.00 | vAd0r | I am adding my hostname that points to my server in x-lite. it will not work that way. when i change it to the ip it will |
| 16:51.05 | vAd0r | what am i suppose to paste? |
| 16:51.11 | [TK]D-Fender | jmls: You need to come up with an alternative plan. |
| 16:51.15 | ManxPower | vAd0r: what is the hostname? |
| 16:51.18 | djPepse | NoCarrier: sorry, I wasn't able to find anything that specified SDP.. is there some syntax that it would be in? |
| 16:51.40 | [TK]D-Fender | vAd0r: the bits that use the domain/hostname you are referring to. Everything relevent to the problem |
| 16:51.42 | NoCarrier | djPepse: first set canreinvite=no on both peers and test that |
| 16:51.56 | djPepse | i did, no change |
| 16:51.57 | vAd0r | trix.kanetworks.com |
| 16:52.12 | [TK]D-Fender | vAd0r: PASTEBIN the EXACT way you are using these. |
| 16:52.19 | vAd0r | that is |
| 16:52.32 | NoCarrier | just so I'm clear - ATA makes outgoing call, and that's when you hear no audio on the ATA? |
| 16:52.38 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 16:52.45 | ManxPower | vAd0r: You are putting the hostname in X-Lite, right? |
| 16:52.50 | [TK]D-Fender | vAd0r: No, I want to see it exactly how its being used in your setup. |
| 16:52.51 | vAd0r | yes |
| 16:52.52 | vAd0r | under domain |
| 16:53.03 | vAd0r | there is no setup TK |
| 16:53.05 | djPepse | NoCarrier: no, if the ATA makes an outgoing call via skype, 2way audio is fine.. When skype calls ATA, ATA gets no audio in but sends audio out |
| 16:53.08 | [TK]D-Fender | vAd0r: pastebin so config files. |
| 16:53.13 | ManxPower | vAd0r: domain is the SIP domain |
| 16:53.14 | [TK]D-Fender | s/so/some |
| 16:53.21 | vAd0r | i goto my dns on the internet and create a A record |
| 16:53.23 | vAd0r | ok |
| 16:53.24 | nick125_lappy | whee, reconfiguration time! |
| 16:53.25 | ManxPower | vAd0r: not the DNS domain |
| 16:53.28 | vAd0r | so add that instead of just the ip? |
| 16:53.39 | ManxPower | [TK]D-Fender: I don't think he is doing anything on asterisk |
| 16:53.46 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 16:53.48 | vAd0r | in the sip.conf right |
| 16:53.56 | [TK]D-Fender | @#@*#@*#@^&@#%^&$(&*(#$^)*#&*$^)^$&*()^)*&)&$^)#$ |
| 16:54.03 | NoCarrier | djPepse: ok, and ata, Asterisk, and skype box are all on the same subnet? |
| 16:54.06 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 16:54.10 | djPepse | NoCarrier: correct |
| 16:54.15 | ManxPower | vAd0r: you said you are doing these changes in X-Lite. Which is it, X-Lite or sip.conf. |
| 16:54.39 | vAd0r | xlite |
| 16:54.50 | vAd0r | i was thinking after what you said i need to add it to my sip.conf |
| 16:54.57 | ManxPower | vAd0r: then you have to look at why X-Lite won't register with your setup. |
| 16:55.13 | [TK]D-Fender | vAd0r: No, in X-Lite that is where you put the host/IP of your * server. |
| 16:55.22 | vAd0r | domain=trix.kanetworks.com,from-internal like something like that |
| 16:55.30 | [TK]D-Fender | vAd0r: this does NOT require you to do anything speical in sip.conf |
| 16:55.55 | [TK]D-Fender | vAd0r: "domain=trix.kanetworks.com" is legit. that suffix at the end is NOT. |
| 16:56.06 | NoCarrier | djPepse: during an inbound call where you have the audio issue, can you use ethereal to see if you're at least receiving a crapload of RTP packets from skype on your skype box? |
| 16:56.09 | vAd0r | i know but it is easier for people to remember hostname instead of ip. no one could register w/ me externally untill i added that statement above w/ the public ip instead of the hostname |
| 16:56.26 | ManxPower | vAd0r: domain should be able to be left BLANK |
| 16:56.34 | djPepse | NoCarrier: in one place in my capture, it says m=audio 8180 RTP/AVP 3 0 8 101... the other lines with m=audio yadda yadda all have 10000 |
| 16:56.36 | vAd0r | in sip.conf you mean |
| 16:56.40 | ManxPower | vAd0r: no. |
| 16:56.48 | [TK]D-Fender | ManxPower :X-lite uses the "domain" to imply the IP of the server. |
| 16:56.48 | ManxPower | vAd0r: in X-Lite |
| 16:56.56 | ManxPower | vAd0r: your problem is NOT in sip.conf |
| 16:56.58 | vAd0r | how does it know to goto my house then |
| 16:57.08 | vAd0r | where do i put the ip? |
| 16:57.18 | ManxPower | [TK]D-Fender: another example of why X-lite is an example of how NOT to do a softphone. |
| 16:57.31 | djPepse | NoCarrier: if I've set rtpstart and rtpend at 10000 and 10016 (i was trying to test being behind nat), would that make a difference? |
| 16:57.35 | vAd0r | I have cisco phones im gonna use but this is for testing before i flash everything |
| 16:57.43 | NoCarrier | djPepse: it should also have a "Connection Information (c)" field with an IP address |
| 16:57.50 | NoCarrier | djPepse: what's that ip address? |
| 16:58.02 | ManxPower | vAd0r: in a DOS/CMD prompt on the machine x-lite is running on, can you "ping trix.kanetworks.com" ? |
| 16:58.07 | vAd0r | where do i put the address of my server at in xlite if not in the Domain field |
| 16:58.10 | *** join/#asterisk CyberPony (n=CyberPon@66-194-25-11.static.twtelecom.net) |
| 16:58.11 | vAd0r | icmp is blocked so no |
| 16:58.15 | vAd0r | but it does resolve |
| 16:58.33 | ManxPower | vAd0r: blocking ICMP will break your network in subtle ways. |
| 16:58.42 | backblue | anyone can give me a hand, with zaptel and echo? |
| 16:58.45 | ManxPower | Some sites won't load, some will load REALLY REALLY slow. |
| 16:58.56 | Dovid | backblue: good luck |
| 16:58.57 | CyberPony | is there a way to reset/clear all channels associated with SIP Peer without reloading chan_sip? |
| 16:59.02 | dasuberdavid | backblue: what version of vaptel are you using? |
| 16:59.02 | Dovid | backblue: what kind of card ? |
| 16:59.03 | djPepse | NoCarrier: hm. i think the ATA |
| 16:59.10 | backblue | Dovid: x100p |
| 16:59.12 | vAd0r | You said that i dont put ip in Domain. Where am i suppose to put it? |
| 16:59.21 | NoCarrier | djPepse: and is this a packet from the ATA to Asterisk? |
| 16:59.30 | backblue | dasuberdavid: last in the svn stable. |
| 16:59.33 | backblue | 1.2 release. |
| 16:59.35 | ManxPower | vAd0r: [TK]D-Fender said that X-Lite uses domain to mean hostname. |
| 16:59.49 | vAd0r | That is where i put it |
| 16:59.49 | ManxPower | seems very stupid to me, but I don't use softphones. All Softphones Suck(tm) |
| 16:59.55 | vAd0r | i agree |
| 17:00.35 | dasuberdavid | backblue: edit zconfig.h in your zaptel sources directory and change your echo can algorithm to mg2. |
| 17:00.35 | vAd0r | just the A record i created doesn't resolve to it |
| 17:00.35 | vAd0r | ill try it in the sip.conf and let you know if it works |
| 17:00.35 | djPepse | nocarrier: that part's a little confusing because it seems smsniff has combined them. It says 11 packets, and it says local add: (skype ip) remote add: (asterisk ip) |
| 17:00.35 | backblue | dasuberdavid: what will that do, diferent from the default? |
| 17:00.52 | dasuberdavid | backblue: yeah i think the default is kb1 , and mg2 kills echo better |
| 17:01.04 | *** part/#asterisk newvn (n=newvn@58.187.113.254) |
| 17:01.05 | vAd0r | that fixed it |
| 17:01.08 | irule | <PROTECTED> |
| 17:01.14 | backblue | dasuberdavid: if you say so. |
| 17:01.26 | vAd0r | domain=trix.kanetworks.com,from-internal |
| 17:01.29 | backblue | i will try, i dont have almost any experience with zaptel problems |
| 17:01.33 | irule | how can I do a conference call, where I am in a normal call, just put the other en on hold, dial someone else, and put them together? is it possible without meetme? I am looking for something more standardized |
| 17:01.36 | backblue | since we dont use it |
| 17:01.40 | [TK]D-Fender | irule: 3-way call done by your phone. |
| 17:01.45 | djPepse | NoCarrier: i see a bunch of stuff between the skype ip and the ATA ip on 10000 |
| 17:01.48 | [TK]D-Fender | irule: app_conference |
| 17:01.49 | vAd0r | i had to do it w/ that and the ip for anyone externally to work. Maybe this is new w/ 2.2? |
| 17:01.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:01.58 | irule | [TK]D-Fender thanks a lot |
| 17:02.00 | [TK]D-Fender | irule: app_meetme |
| 17:02.03 | backblue | but in ztmonitor, i see all the audio that it's in the tx, it's in the rx too, when i speak |
| 17:02.04 | [TK]D-Fender | irule: take your mick |
| 17:02.04 | backblue | very weird. |
| 17:02.09 | [TK]D-Fender | pick* |
| 17:02.19 | NoCarrier | djPepse: nothing should be going from the skype IP directly to the ATA IP - it should all be going through your Asterisk box |
| 17:02.38 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 17:03.14 | djPepse | NoCarrier: that's weird. it says local add: skypeip.. remote add: ataip.. local port: 10000, remote port: 8180 (the one i was mentioning earlier) |
| 17:03.27 | ManxPower | backblue: echo is one of the hardest things to fix in VoIP. |
| 17:03.32 | djPepse | NoCarrier: actually, tha tmakes sense |
| 17:03.36 | carrar | irule: http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro |
| 17:03.41 | djPepse | NoCarrier: the skype gateway and the skype app are on the same computer |
| 17:03.47 | carrar | already written for you |
| 17:03.49 | djPepse | NoCarrier: it connects to * with SIP |
| 17:03.50 | NoCarrier | djPepse: ohhh |
| 17:04.35 | *** part/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 17:05.09 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:05.25 | [TK]D-Fender | carrar: Psycho macro that jsut uses Meetme. He asked if there were OTHER MEANS of conferencing than Meetme. |
| 17:06.21 | carrar | yeah but the tradditional use of meetme and this app might be different for the new user |
| 17:06.54 | ManxPower | app_conference |
| 17:07.04 | backblue | ManxPower: i know, i never have very trouble, because we mostly use ISDN. |
| 17:07.04 | ManxPower | maybe in asterisk-addons, but I'm not sure. |
| 17:07.13 | backblue | in BRI and PRI |
| 17:07.17 | ManxPower | backblue: that is not the reason |
| 17:07.24 | ManxPower | echo comes from the FAR end analog loop. |
| 17:08.06 | ManxPower | so if you are calling another ISDN destination or a cell destination or any kind of destination that is not on analog, then you won't get echo since the far end is not analog. |
| 17:08.20 | backblue | i only have this echo problem, with this FXO cards |
| 17:08.31 | ManxPower | backblue: Let me explain echo to you. |
| 17:08.44 | backblue | ok, go head |
| 17:09.28 | ManxPower | backblue: Imagine you are in the middle of a large empty field and you say "HELLO!". Your voice sounds like it is outside because there is none of what is called "side tone". Similar to when you talk into a phone that is not powered on and not plugged into anything. |
| 17:10.01 | plasmid | http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups <-- tells me how to setup a pickupgroup and a callgroup. I set extensions 100 & 200 to "1" (without the quotes) in pickupgroup and call group. I called extension 100 on my cell and tried to pickup extension 200 phone by pressing *8. I get silence. What did I do wrong? Half right? |
| 17:10.12 | ManxPower | Now go into an average sized room and say "HELLO!". Your voice sounds like you are inside. This is called "sidetone". Similar to when you are talking into a phone that is powered up and connected to a line. |
| 17:10.31 | ManxPower | Now go into a really big room and say "HELLO!". You hear an echo. |
| 17:11.28 | ManxPower | When you were in the small room, there was echo, but it came back so fast that you do not hear it as echo, you hear it as a sidetone. This is similar to calling an analog phone. THE ECHO IS THERE, but you don't hear it. |
| 17:11.53 | ManxPower | On VoIP you are in a large room, the echo is there and the delay of the echo is so much that you hear the echo. |
| 17:12.53 | ManxPower | Echo is a fact on all 2-wire analog lines. You just don't hear it as anything but a sidetone until you start having a delay because of network latency, buffering, and codec conversion delays. |
| 17:13.51 | ManxPower | backblue: If you are on analog digium cards that are in-warrenty then you can get a free HPEC (commercial grade echo cancel software module). If you are out of warrenty or do not have a digium analog card then you can buy the HPEC for like $10/channel |
| 17:14.32 | djPepse | I don't think he's coming back :) |
| 17:14.45 | vAd0r | anyone here i can make a test call to |
| 17:15.44 | djPepse | ManxPower: Oo where do you get that? |
| 17:17.00 | djPepse | vador: i don't think you can make a sip call to me, but i'm on FWD and Stanaphone |
| 17:17.01 | ManxPower | djPepse: From Digium |
| 17:17.16 | vAd0r | i wanna try a direct sip call and one that has a custom extension that routes through |
| 17:17.19 | djPepse | I wonder how I can tell if my digium card is under warranty :) probably isn't tho |
| 17:17.36 | vAd0r | everything works fine. just when im at work my asa blocks crap |
| 17:17.36 | ManxPower | djPepse: your original invoice. Digium can't tell you based on the serial number |
| 17:17.47 | djPepse | ManxPower: bummer, no clue where that is |
| 17:17.59 | ManxPower | djPepse: $10/channel is cheap. |
| 17:18.05 | backblue | ManxPower: i'm downloading the octasis echo cancel now. |
| 17:18.13 | djPepse | indeed |
| 17:18.21 | djPepse | I could use a couple of those |
| 17:18.29 | ManxPower | I COULD have found a serial number and invoice that would make digium think I had an in-warrenty analog card. I found it easier to spend $40 on 4xHEPC licenses. |
| 17:18.32 | djPepse | backblue: Please tell me how that works out |
| 17:18.42 | ManxPower | backblue: it won't work woutout a license. |
| 17:18.53 | ManxPower | The HPEC works fine for me. |
| 17:19.24 | backblue | ManxPower: i have a licence :) |
| 17:19.38 | *** part/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 17:20.12 | djPepse | I see a bunch of stuff about HPEC making the volume low |
| 17:20.47 | Qwell | djPepse: and where are you seeing this? |
| 17:20.58 | djPepse | http://forums.digium.com/viewtopic.php?p=51230&sid=54a05f7536ea40112cd23302c97a1d37 |
| 17:21.18 | djPepse | but it said there's a new version of HPEC out on may 17th |
| 17:22.09 | redax | is this loop sytactical OK: While( ${VAR} ); Set(U=${Cut(VAR,,1)}); ...do something with ${U} ... ; Set(VAR=${Cut(VAR,,2-)}); EndWhile |
| 17:24.04 | djPepse | ManxPower: any chance I could get some more help on the 1way audio issue? :).. It seems like it should just be sip between the skype gateway box and the ATA |
| 17:24.10 | djPepse | doh |
| 17:28.42 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 17:28.44 | PioneerVM2 | is there any need to "register" with a sip peer? With one provider I cant seem to register, yet everything seems to work fine. |
| 17:29.37 | *** join/#asterisk Fieldy (i=q2RueeaF@gentoo/contributor/Fieldy) |
| 17:29.50 | ghenry | PioneerVM2: depends on provider |
| 17:29.59 | ghenry | some auth by ip and cli |
| 17:30.06 | [TK]D-Fender | PioneerVM2: Registration is not technically required to place/get calls |
| 17:30.18 | [TK]D-Fender | PioneerVM2: AGN specifically does not support/need it. |
| 17:30.22 | PioneerVM2 | so if making/getting calls is working then I should have to worry about it? |
| 17:30.40 | PioneerVM2 | ok, they mentioned i "can" do it in my setup docs but their documentation is sparse and i could not get it to work |
| 17:30.56 | PioneerVM2 | but phone seems to work, at least dial out so far |
| 17:31.08 | [TK]D-Fender | PioneerVM2: "Register" is jsut a way of updating the IP they should be sending calls to. If yours doesn't change, then there is no need to register. |
| 17:31.16 | PioneerVM2 | ahh ok |
| 17:31.18 | PioneerVM2 | thx |
| 17:31.36 | PioneerVM2 | so if you had dynamic IP like at home you may want to register |
| 17:32.12 | djPepse | Can anyone else by chance offer any help with my one-way audio on the same local subnet problem? |
| 17:32.20 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:33.40 | [TK]D-Fender | PioneerVM2: Typically yes. Or provide them a host/domain and keep it updated via DynDNS or the like. |
| 17:33.58 | PioneerVM2 | now what about "status monitored/unmonitored" -- i just setup teliax and teliax has no status info , says unmonitored |
| 17:34.03 | PioneerVM2 | is that a cfg issue or just how that provider works |
| 17:34.05 | [TK]D-Fender | djPepse: pastebin your sip.conf minus only passwords |
| 17:34.49 | [TK]D-Fender | PioneerVM2: "unmonitored" means you aren't using "qualify" to validate that the host is still reachable |
| 17:35.35 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 17:35.36 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:35.41 | PioneerVM2 | df: ahh yep that fixed it -- their default cfg they sent me didnt show that |
| 17:35.58 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 17:36.09 | _VoiceMeUp_COM | is there a specific version for radius ? |
| 17:37.26 | djPepse | [TK]D-Fender: http://pastebin.ca/507640 |
| 17:37.49 | CyberPony | can anyone explain why I might see sip channels with "INVITE" status when the SIP peer is no longer powered on ? |
| 17:38.01 | djPepse | I tried canreinvite=no on both, no difference, so I set it back |
| 17:38.04 | [TK]D-Fender | PioneerVM2: I wouldn't say "fixed", as this state does not actually pose a PROBLEM, but it may be good for your peace of mind. |
| 17:38.21 | `Sean | Guys i need youre help i need to find a Soloution to get my Fax machine running Over IP maybe a good working ATA by cisco or someone else, or even a seperate application with asterisk/callweaver or something that runs on windows even |
| 17:38.37 | PioneerVM2 | dk: gotya thx |
| 17:39.12 | *** join/#asterisk saftsack (n=saftsack@pD9E064E9.dip.t-dialin.net) |
| 17:39.24 | [TK]D-Fender | djPepse: "canreinvite=no", and I do not see you define "localnet=" in there |
| 17:39.52 | djPepse | where does that get defined? |
| 17:40.07 | djPepse | (and what should it look like).. i'd never seen the localnet def before |
| 17:40.19 | [TK]D-Fender | djPepse: under [general] along with the "nat=yes" you are missing. |
| 17:40.36 | irule | how can I tell * what to do when the caller pesses on flash in a running call? |
| 17:41.08 | djPepse | localnet=192.168.0.0/24? |
| 17:41.12 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 17:42.59 | djPepse | localnet=192.168.0.0/255.255.0.0 |
| 17:43.25 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 17:43.48 | plasmid | [TK]D-Fender, how can I tell if my callgroup, pickupgroup is enabled? I already set callgroup and pickupgroup to "1" (without the quotes) and I have extensions 100 and 200 on them. However, if I call extension 100 on my cell and try to pickup the call on extension 200 by pressing *8 I get silence. I must have done something wrong or half right. |
| 17:47.01 | djPepse | [TK]D-Fender: Still no change :/ |
| 17:49.55 | GuruJee | helo people |
| 17:50.06 | GuruJee | i need some advice on connection two Asterisk Boxes |
| 17:50.19 | GuruJee | [TK] man u gota help me!! |
| 17:50.37 | bkruse | [TK]D-Fender: run! |
| 17:50.55 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 17:50.57 | GuruJee | connecting* |
| 17:51.03 | GuruJee | listen up guys |
| 17:51.04 | backblue | octasis echo cancel, solved all the problems |
| 17:51.10 | backblue | no echo, perfect! :) very good |
| 17:51.12 | djPepse | how much is that one |
| 17:51.18 | backblue | $10 i think |
| 17:51.22 | backblue | dont know |
| 17:51.23 | GuruJee | I have a iax2 trunk up between the two systems and i can call the extensions |
| 17:51.37 | GuruJee | but what I cant do is to dial local pstn calls over that trunk |
| 17:51.44 | GuruJee | one system has a pri card |
| 17:51.52 | GuruJee | I want that system to be able to dial out on the pstn |
| 17:52.11 | GuruJee | any help is greatly appreciated |
| 17:53.19 | *** join/#asterisk lee_is_me (n=chatzill@66.16.60.61) |
| 17:53.33 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 17:53.38 | GuruJee | I need help with linking two asterisk boxes |
| 17:53.44 | GuruJee | one has a pri and the other one dont |
| 17:53.47 | `Sean | i need help with FAX! |
| 17:53.59 | *** join/#asterisk ssokol (n=ssokol@69-94-196-106.biltmorecomm.com) |
| 17:54.01 | GuruJee | sean, faxing doesnt work good with asterisk |
| 17:54.25 | lee_is_me | Hey, all. Can anyone tell me what "Web Recovery atA: xxx.xxx.xxx.xxx" on an Aastra phone indicates? |
| 17:54.38 | `Sean | Gurujee i know butpeople have gotten good results with some ata's just wanna find out those ata's |
| 17:55.00 | lee_is_me | It goes to that message as soon as the phone is booted and I cannot use any of hte buttons. |
| 17:55.23 | GuruJee | I need help with linking two asterisk boxes |
| 17:55.24 | GuruJee | I need help with linking two asterisk boxes |
| 17:55.35 | GuruJee | I need help with linking two asterisk boxes |
| 17:55.39 | *** kick/#asterisk [GuruJee!n=north@pdpc/sponsor/digium/Qwell] by Qwell (we heard you the first 8 times) |
| 17:55.42 | Qwell | jesus |
| 17:55.46 | *** join/#asterisk GuruJee (n=tilde@12.186.161.117) |
| 17:55.51 | GuruJee | aah |
| 17:55.53 | Qwell | This isn't AOL. |
| 17:55.59 | GuruJee | sorry dude |
| 17:56.02 | GuruJee | i apologize |
| 17:56.09 | GuruJee | i started playing with the colors |
| 17:56.14 | Qwell | colors? |
| 17:56.16 | Qwell | again... |
| 17:56.16 | GuruJee | yeah |
| 17:56.17 | Qwell | This isn't AOL. |
| 17:56.21 | GuruJee | yeah i know |
| 17:56.22 | GuruJee | sorry |
| 17:56.29 | GuruJee | so qwell can u help me? |
| 17:56.30 | LeddyHM | no you're not |
| 17:56.32 | [hC] | Qwell; ASL? |
| 17:56.35 | [hC] | :P |
| 17:56.38 | Qwell | [hC]: zomg! |
| 17:56.42 | GuruJee | with linking two asterisk ? |
| 17:56.46 | GuruJee | boxes |
| 17:56.50 | [TK]D-Fender | GuruJee: show us how hard you've TRIED with the link I gave you earlier today... |
| 17:56.52 | [hC] | GuruJee: www.voip-info.org |
| 17:56.57 | GuruJee | tk |
| 17:57.00 | GuruJee | I have tried all that |
| 17:57.10 | GuruJee | here is the thing |
| 17:57.14 | Qwell | The enter key is not punctuation. |
| 17:57.15 | GuruJee | i have tried DUNDi (local calls work) |
| 17:57.16 | Qwell | Use commas. |
| 17:57.26 | [TK]D-Fender | GuruJee: NOT WHAT I ASKED |
| 17:57.28 | GuruJee | i have tried iax2 (local calls work) |
| 17:57.34 | GuruJee | I want to be able to dial onto the pstn |
| 17:57.41 | GuruJee | any hint? |
| 17:57.51 | GuruJee | <[TK]D-Fender> GuruJee: NOT WHAT I ASKED //what do u wanna see? |
| 17:57.56 | [hC] | if you can dial from one * to another *, on local extensions, but not from one * to another *'s pstn, |
| 17:58.03 | [hC] | your dial plan is wrong. period. the link is fine. |
| 17:58.07 | [TK]D-Fender | GuruJee: Show me how you've set up your 2 boxes with the methods explained in the page I linked you earlier |
| 17:58.18 | GuruJee | hc: I can call the local lines. One of my pbx has a pri card; the otehr one doesnt |
| 17:58.22 | [hC] | I would check what context you are dropping calls into when they come from the other box. |
| 17:58.30 | GuruJee | I want the one without to use the other pbx's pri card to dial out |
| 17:58.35 | [hC] | You're probably sending into a context that is only able to dial local extensions. |
| 17:58.46 | GuruJee | well, i tried from-internal |
| 17:58.49 | GuruJee | and also from-pstn |
| 17:58.51 | _VoiceMeUp_COM | so can asterisk be used iwth radius ? |
| 17:58.55 | mcab | //llp |
| 17:58.57 | GuruJee | yes it can voice |
| 17:59.00 | [TK]D-Fender | GuruJee: stop talking about your configs and SHOW THEM. |
| 17:59.01 | bkw_ | GuruJee, have you tried an ethernet cable to link them? |
| 17:59.03 | djPepse | D-Fender: was localnet=192.168.0.0/255.255.255.0 correct syntax? |
| 17:59.05 | GuruJee | do u want to see my config files? |
| 17:59.09 | GuruJee | ok wait |
| 17:59.09 | mcab | errm, oops, sry |
| 17:59.11 | [hC] | GuruJee: are you using trixbox/freepbx? |
| 17:59.17 | bkw_ | djPepse, 192.168.0.0/24 is also correct |
| 17:59.17 | [TK]D-Fender | djPepse: Yes, pastebin your new sip.conf |
| 17:59.28 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 17:59.28 | GuruJee | I am using Asterisk by Digium, and the other box is Trixbox |
| 17:59.32 | Assid | heya |
| 17:59.37 | GuruJee | hwoever, the configs and contexts on both of them are same |
| 17:59.46 | [TK]D-Fender | GuruJee: do NOT expect help with the Trixbox system at all. |
| 18:00.05 | GuruJee | TK |
| 18:00.05 | djPepse | IE is not having a good day. |
| 18:00.27 | GuruJee | just show me the way; dont worry about the distribution. As I mentioned earlier the contexts are not of trixbox or freepbx |
| 18:00.37 | GuruJee | hold on let me paste the configs |
| 18:00.38 | [hC] | [TK]D-Fender: So, I built that polycom microbrowser company directory app, which builds the directory based on the lines configured in all of our polycom config files (reg.1.address/displayName, reg.2.address/displayName), so as soon as a phone is provisioned, it shows up. Kinda neat. |
| 18:00.42 | Katty | [TK]D-Fender: in my intercom example, they setup exten *96 which was a goto [intercom] which executed answer, play beep, time out 5 seconds, then WaitExten(10), then it jumped down to *,1 and *,2. 1 added teh ring answer header, and 2 had dial SIP$EXTEN |
| 18:01.04 | Katty | [TK]D-Fender: but that's not really want i'm trying to do. |
| 18:01.24 | LeddyHM | hc: hook us up! |
| 18:01.25 | Katty | [TK]D-Fender: i want an answer, playback, set timeout, add header, then page(SIP/whoever&SIP/thisonetoo |
| 18:01.48 | [hC] | GuruJee: well, the distribution matters, as the contexts will be named accordingly. trixbox does a lot and implies alot. |
| 18:01.48 | Katty | [TK]D-Fender: i think. |
| 18:01.51 | [hC] | LeddyHM: I will release it soon, i just have to make it a touch less centric to the way i do things |
| 18:01.56 | djPepse | [TK]D-Fender: http://pastebin.ca/507671 |
| 18:01.57 | [hC] | I am also going to add support for it to pull data from mysql. |
| 18:02.03 | Katty | [TK]D-Fender: what's the difference in the example with WaitExten(10) with dial(SIP/EXTEN) and Page(SIP/multiple) |
| 18:02.10 | [TK]D-Fender | Katty: You had the little menu to choose between page all / someone specific IIRC |
| 18:02.16 | LeddyHM | hc: I created one that pulls info from sharepoint |
| 18:02.18 | LeddyHM | kinda nifty |
| 18:02.31 | Katty | [TK]D-Fender: but how does Page/lots compare to Dial/lots |
| 18:02.45 | Katty | [TK]D-Fender: is dial just dial and make them pick up, where page is a meetme? |
| 18:02.45 | [hC] | LeddyHM: ah yeah another feature i will be adding is pulling from ldap (including MS Exchange) |
| 18:02.48 | [TK]D-Fender | Katty: "lots"? |
| 18:02.57 | Katty | SIP/lots&SIP/more |
| 18:03.00 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 18:03.06 | djPepse | [TK]D-Fender: the two machines are on the same local net, so why nat=yes? |
| 18:03.11 | [TK]D-Fender | Katty: Page = 1-way multi-dial meetme |
| 18:03.15 | djPepse | err the three machines |
| 18:03.30 | [TK]D-Fender | djPepse: You have implied that your * is behind NAT. *just because*. |
| 18:03.32 | Katty | [TK]D-Fender: and dial is a they can talk back thing too? |
| 18:03.37 | djPepse | ok |
| 18:03.47 | [TK]D-Fender | djPepse: Now pastebin the output of a failed call at verbose 10, and sip debug enabled. |
| 18:03.50 | GuruJee | ok gentlement: http://paste.debian.net/28759 this is the link to extensions_custom.conf file |
| 18:03.59 | [TK]D-Fender | Katty: that does not parse |
| 18:04.13 | *** join/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 18:04.18 | GuruJee | [hc]: have a look at the logic of the dial plan please |
| 18:04.23 | Katty | [TK]D-Fender: Page(SIP/multiple) means you call multiple people on ring answer, and they can't talk back. |
| 18:04.31 | GuruJee | http://paste.debian.net/28759 this is the link to extensions_custom.conf file |
| 18:04.44 | Katty | [TK]D-Fender: and dial(SIP/multiple) means after ring answer, they /can/ talk back? |
| 18:04.48 | [TK]D-Fender | GuruJee: that is not at all like the methods described byt he page I gave you |
| 18:04.50 | *** join/#asterisk Un1x_laptop (i=Sean@72.53.146.162) |
| 18:04.50 | Katty | [TK]D-Fender: or is dial for only one? |
| 18:05.10 | [TK]D-Fender | Katty: AH, yes, dial is for jsut 1 |
| 18:05.12 | GuruJee | TK: I tried that and I lost all the connectivity to it |
| 18:05.13 | Katty | ok |
| 18:05.21 | GuruJee | TK: nothing worked with that |
| 18:05.25 | djPepse | guh what was the verbosity command again |
| 18:05.35 | [TK]D-Fender | GuruJee: translation : you have nothing sueful to show me. Thanks for stopping by. |
| 18:05.38 | GuruJee | however, I created a trunk from gui using IAX2 |
| 18:05.55 | GuruJee | TK are you on a crow diet? |
| 18:06.21 | djPepse | core verbose |
| 18:06.27 | [TK]D-Fender | GuruJee: You also have not shown IAX2 debug for the connection attempt, nor your IAX configs. How hard do you think you appear to be trying? |
| 18:06.43 | GuruJee | sorry, all u had to do was ask |
| 18:06.49 | [TK]D-Fender | GuruJee: You've done little, and shown even less. |
| 18:06.51 | GuruJee | I am a noob, but I am ready to learn. So just ask |
| 18:07.00 | [TK]D-Fender | GuruJee: You want help? Don't make us ASK too much. |
| 18:07.15 | GuruJee | sorry dude |
| 18:07.22 | GuruJee | would u wanna vnc into my system? |
| 18:07.30 | GuruJee | never mind |
| 18:07.37 | GuruJee | let me create a text file with all the things |
| 18:07.43 | Assid | hey [TK]D-Fender hows you |
| 18:07.54 | [TK]D-Fender | GuruJee: No, I want you to show you have a brain and provide simple pastebin'd configs and output that clearly show your problem. |
| 18:08.08 | [TK]D-Fender | GuruJee: And you have only shown us the TRIXBOX side. |
| 18:08.19 | [TK]D-Fender | Assid: can't you tell by my tone? :) |
| 18:08.42 | Assid | oh that was you? I thought the bot was writing for ya.. never seen you this red |
| 18:10.36 | [TK]D-Fender | GuruJee: and I don't do Dundi support. Go try to set up a proper basic IAX2 link as you were provided info on and then we can work out the issues |
| 18:11.30 | Assid | breathe brother breathe!!! |
| 18:14.40 | djPepse | man, i have to start all over :) this is taking a bit |
| 18:15.00 | GuruJee | [tK]: http://paste.debian.net/28762 please check it now. If you need anything else just let me know. I have the outputs of everything you asked for |
| 18:15.17 | GuruJee | you too [HC] |
| 18:16.25 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 18:16.28 | djPepse | [TK]D-Fender: Can I send this in a msg? (the pastebin url) |
| 18:16.55 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:17.16 | GuruJee | by the way, what is the default context for outgoign calls? |
| 18:18.02 | [TK]D-Fender | GuruJee: Sorry, as I said I do not support Dundi. When you've set it up according to the guide I provided I may be able to assist |
| 18:18.10 | GuruJee | thats fine |
| 18:18.14 | GuruJee | thank you |
| 18:18.22 | GuruJee | one last question |
| 18:18.29 | GuruJee | what is the default context for the outgoing calls on pstn? |
| 18:18.32 | GuruJee | from-pstn? |
| 18:19.27 | [TK]D-Fender | GuruJee: There is no such concept as "default" |
| 18:19.43 | Assid | he is asterisknow based on pgsql/mysql backend ? |
| 18:19.51 | djPepse | [TK]D-Fender: May I send this pastebin url in a msg? |
| 18:20.38 | GuruJee | well if I am using Digium TE410p card and zaptel. what context would that be by defautl? |
| 18:20.48 | [TK]D-Fender | djPepse: just do it HERE |
| 18:21.11 | [TK]D-Fender | GuruJee: the is NO SUCH CONCEPT |
| 18:21.25 | djPepse | http://pastebin.ca/507696 |
| 18:22.11 | *** join/#asterisk smooth_penguin (n=smooth_p@59.95.11.70) |
| 18:22.13 | djPepse | i don't know if there's passwords in that or what :P |
| 18:22.29 | [TK]D-Fender | djPepse: You have EYES, use them :) |
| 18:22.39 | smooth_penguin | hey im not very good with asterisk .. but can you use a internal modem as an FX0 device? |
| 18:23.06 | djPepse | Yes, but which is easier: msging you a url or reading 412 individual lines looking for passwords |
| 18:23.12 | djPepse | :) |
| 18:23.23 | djPepse | smooth_penguin: no |
| 18:24.24 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:24.32 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 18:25.38 | brodiem | can anyone think of any decent IP phones that have functional BLF support? I'm already aware of its support on Aastra, Snom and GS |
| 18:26.25 | smooth_penguin | djPepse, I was looking at this post and hence asked .. http://forums.whirlpool.net.au/forum-replies.cfm?t=462829#r1 |
| 18:26.55 | [TK]D-Fender | brodiem: Polycom > All/ |
| 18:27.33 | djPepse | smooth_penguin: I guess I was wrong then :) |
| 18:27.35 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 18:28.03 | brodiem | [TK]D-Fender thanks, you figure that about sums them up? I know thompson does too but no real interest in those |
| 18:28.38 | Katty | anyone got a good .wav for an intercom notification? |
| 18:28.51 | brodiem | wonder if the ciscos do.. |
| 18:29.24 | [TK]D-Fender | brodiem: Cisco = No BLF |
| 18:30.01 | d3wayne | doh.wav ? |
| 18:30.05 | djPepse | [TK]D-Fender: no ideas from looking at that debug output? |
| 18:31.46 | fall0ut | hrm, there a way to test if a meetme conf exists and if it doesn't prompt to authenticate, if it does drop into the meetme and check for pin etc? |
| 18:31.54 | [TK]D-Fender | djPepse: LOOKS fine... |
| 18:31.56 | vAd0r | when i am behind my ASA i have problems w/ talk path when i connect to my server at my house. anyone know how to fix this |
| 18:32.08 | [TK]D-Fender | brodiem: Cisco = expensive for no real point. |
| 18:32.29 | [TK]D-Fender | vAd0r: ... |
| 18:32.30 | [TK]D-Fender | ~sipnat |
| 18:32.32 | jbot | it has been said that sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:32.32 | fall0ut | vAd0r: turn off sip "fixup" in the ASA and just rely on asterisk to do NAT fixing |
| 18:32.33 | djPepse | [TK]D-Fender: I'm confused as to why it uses a port in the 8000s for rtp.. is that just on the ATA (.190)? |
| 18:32.39 | *** join/#asterisk tzafrir_laptop (i=tzafrir@69-94-197-125.biltmorecomm.com) [NETSPLIT VICTIM] |
| 18:32.39 | *** join/#asterisk kmshanah (n=kmshanah@ppp235-99.static.internode.on.net) |
| 18:32.39 | *** join/#asterisk Know1 (i=know1@creep.bur.st) [NETSPLIT VICTIM] |
| 18:32.39 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@210.18.216.66) [NETSPLIT VICTIM] |
| 18:32.39 | *** join/#asterisk masked (i=masked@shell.iinet.net.au) [NETSPLIT VICTIM] |
| 18:33.02 | [TK]D-Fender | djPepse: each side just picks its own port.... shouldn't be an issue |
| 18:33.18 | vAd0r | it works fine except when im at my work and im behind an asa there |
| 18:33.31 | djPepse | So when ATA initiates call, it's fine.. When gateway initiates call, it isn't :/ |
| 18:33.32 | vAd0r | no fixup there |
| 18:33.50 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:34.41 | Assid | nice.. asterisk can link directly with pgsql |
| 18:34.59 | Assid | hey [TK]D-Fender you ever wrote somethiing that can handle LCR ? |
| 18:35.23 | [TK]D-Fender | Assid: Nope.... I just pass all of my home calls through the office... NCR :D |
| 18:35.31 | [TK]D-Fender | (to ME anyways!) |
| 18:35.32 | Assid | hehe |
| 18:36.05 | [TK]D-Fender | Assid: Most I'd do if it comes to it is NANPA area-code seperation.... |
| 18:36.39 | Assid | im trying to mimick the vendors.. have a credit per user.. pre-paid.. and cut off the call if they finish that much credit |
| 18:37.15 | fall0ut | that's fraud control, not so much LCR |
| 18:37.34 | Assid | well.. that too i guess.. basically i wanna know how much they are using |
| 18:37.45 | Katty | where's the sounds that you Playback located? |
| 18:37.46 | fall0ut | but LCR is done best by ratecenter and/or npa-nxx or tier depending on rates from the carriers |
| 18:38.11 | Assid | yeah.. i need the rates per min to be calculated |
| 18:38.23 | Assid | cause they call all over.. europe/us/canada etc. |
| 18:38.33 | Assid | but generally 4-5 countries |
| 18:38.48 | fall0ut | typically carriers charge either a flat-rate per ratecenter or a flat rate per OCN in a rate center |
| 18:38.51 | Assid | and the rates are generally different in europe .. |
| 18:38.58 | fall0ut | yea |
| 18:39.04 | [TK]D-Fender | Katty: typically /var/lib/asterisk/sounds |
| 18:39.12 | fall0ut | usually for intl rates, you get 3-4 rates per countyr |
| 18:39.12 | Assid | so i want it billable to the user so they dont abuse the system |
| 18:39.25 | fall0ut | depending on cell/landline/etc |
| 18:39.27 | Katty | [TK]D-Fender: i'm looking but i can't seem to find the one after she says "youhave" and then the numbers... |
| 18:39.41 | Assid | right.. so i m trying to figure out how to differentiate between +44 +4414 +4415 +44156 +44146 |
| 18:39.56 | Assid | since you actually have to put it in a loop and keep counting backwards |
| 18:39.59 | smooth_penguin | anyone here used a dlink winmodem as a FXO device? |
| 18:40.02 | Katty | [TK]D-Fender: i want the "3" file..but can't seem to locate it |
| 18:40.11 | Dovid | +44 is the UK |
| 18:40.12 | [TK]D-Fender | Katty: under "/digits" |
| 18:40.18 | [TK]D-Fender | Assid: AGI time <- |
| 18:40.44 | Katty | [TK]D-Fender: oooh. but it's still Playback(3) |
| 18:40.46 | [TK]D-Fender | smooth_penguin: Extremely unlikely. Yuo can't take jsut any craptastic modem and use it as an FXO. |
| 18:40.53 | Katty | [TK]D-Fender: don't need to tell it to switch directories, right? |
| 18:41.06 | [TK]D-Fender | Katty: Playback(digits/3) |
| 18:41.14 | Katty | sweet |
| 18:41.23 | smooth_penguin | [TK]D-Fender, I read some of the winmodems could be configured as an FXO device |
| 18:41.30 | Mercestes | Festival("I love mercestes") |
| 18:41.33 | Mercestes | that is all. thank you |
| 18:41.40 | [TK]D-Fender | smooth_penguin: Where some = <.5% of whats out there. |
| 18:41.47 | smooth_penguin | hrm |
| 18:41.55 | [TK]D-Fender | smooth_penguin: ... |
| 18:41.56 | Assid | [TK]D-Fender: yeah but how :| my agi knowledge.. well.. sucks! |
| 18:41.58 | [TK]D-Fender | ~wglwat |
| 18:42.01 | jbot | i heard wglwat is well, good luck with all that |
| 18:42.02 | smooth_penguin | can you link me to any links on it? |
| 18:42.04 | Assid | time to learn |
| 18:42.29 | [TK]D-Fender | smooth_penguin: FORGET your modem and realize you're going to have to buy SUPPORTED equipment. |
| 18:42.33 | Assid | just curious. i should be able to take the first 6 digits.. right ? i dont need to match all and count backwards do i ? |
| 18:42.44 | irule | hi there, I have a couple x100p and would like to asign them different contexts when I have incomming calls, I currently just have 'context 1-2', will it be context 1 and in another line context 2 or something? thanks\ |
| 18:42.45 | smooth_penguin | [TK]D-Fender, http://www.velocityreviews.com/forums/t236754-why-not-use-a-voice-modem-as-fxo-card.html |
| 18:42.47 | Mercestes | smooth_penguin, I read somewhere that Elvis is still alive, UFOS are real, the president is an alien, and the Apollo missions were all hoaxes. oh, and there is a russian moon base on the dark side of the moon |
| 18:42.53 | Assid | most centers are just a maximum of 4 digits? |
| 18:43.03 | smooth_penguin | Mercestes, haha ok man I get you |
| 18:43.09 | Mercestes | :) |
| 18:43.32 | [TK]D-Fender | irule: context=context1 , channel =>1 ,context=context2 , channel =>2 |
| 18:43.34 | smooth_penguin | was just asking if it was possible because of a few posts I read |
| 18:43.58 | Mercestes | not with any reliable/impressive/good results. |
| 18:44.08 | djPepse | i'm interested in that cell driver for asteriskwin32 |
| 18:44.18 | [TK]D-Fender | smooth_penguin: check your chipset. See if Zaptel acknowledges it. if not : TFB |
| 18:44.18 | djPepse | that would be pimp to connect * to my cellphone |
| 18:44.28 | Mercestes | and even if you did get it to work, you will get random faxes out of your ass while your on the phone. |
| 18:44.46 | irule | [TK]D-Fender thanks a lot man |
| 18:45.01 | Mercestes | It's kinda like, "Yay! It works....omg..this sounds like CRAP!" |
| 18:45.14 | Mercestes | and that's .5% of the time. The other 99.5% of the time it doesn't work at all |
| 18:45.42 | djPepse | Can anyone else suggest something to try with a 1way audio issue on the same local network? |
| 18:45.43 | Katty | [TK]D-Fender: if i play the 3,2,1 files right after the user dials an extension, it cuts off the first bit of the 3 and sounds awkward. is there anyway to tell it to pause for a second? |
| 18:46.02 | [TK]D-Fender | Katty: Wait(1) |
| 18:46.05 | Katty | thanks |
| 18:46.37 | irule | so there are no changer in /etc/zaptel.conf? I just have fxsks = 1-2 in there |
| 18:46.44 | irule | changes |
| 18:46.51 | smooth_penguin | [TK]D-Fender, http://www.voip-info.org/wiki/index.php?page=X100P+clone |
| 18:46.58 | smooth_penguin | ive got an sm56 |
| 18:47.13 | [TK]D-Fender | smooth_penguin: Perhaps you weren't paying attention : |
| 18:47.22 | [TK]D-Fender | [TK]D-Fender>smooth_penguin: check your chipset. See if Zaptel acknowledges it. if not : TFB |
| 18:47.42 | djPepse | I'm going on vacation soon and I would really like to be able to call home to my girlfriend from places :/. As it is, she can call me, .. but she's a girl. |
| 18:47.48 | [TK]D-Fender | smooth_penguin: that means get off your ass, TRY and and THEN give up all hope if/when it fails :) |
| 18:47.58 | irule | btw guys I am using a digium x100p and a clone, both of them work fine and dandy for me |
| 18:48.21 | *** join/#asterisk Divious1 (i=chatzill@65.112.134.160) |
| 18:48.25 | Divious1 | hello |
| 18:48.34 | smooth_penguin | [TK]D-Fender, I like the trial and error method :) |
| 18:48.39 | Divious1 | I have a question, is there a migration tool for asterisk? |
| 18:48.40 | smooth_penguin | teaches a lot |
| 18:48.47 | irule | Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 18:49.07 | plasmid | [TK]D-Fender, how can I tell if my callgroup, pickupgroup is enabled? I already set callgroup and pickupgroup to "1" (without the quotes) and I have extensions 100 and 200 on them. However, if I call extension 100 on my cell and try to pickup the call on extension 200 by pressing *8 I get silence. I must have done something wrong or half right. |
| 18:49.30 | Mercestes | plasmid: Is *8 enabled in features.conf?? |
| 18:49.51 | plasmid | Mercestes, checking... |
| 18:49.56 | Mercestes | plasmid: And does it *answer* the call and give you silence or does it just sit there? |
| 18:50.14 | plasmid | Mercestes, it just sits there for a while then i get a busy signal |
| 18:50.26 | fall0ut | how do you check to see if a meetme conference exists? |
| 18:50.38 | Mercestes | [TK]D-Fender, see how he automatically asked you instead of the channel??? If you were like me, and hosed a few asterisk boxes, maybe they wouldn't bother you as much. :P |
| 18:50.52 | Mercestes | plasmid: What kind of phone? Does your phone's digitmap support *8? |
| 18:51.09 | djPepse | how does *8 work, anyway? |
| 18:51.09 | plasmid | Mercestes, it's a regular PSTN phone attached to a sipura3102 |
| 18:51.22 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
| 18:51.23 | Mercestes | Does the ata's digitmap support *8? |
| 18:52.06 | plasmid | Mercestes, ack! i don't see *8 in my features.conf file... what is the correct syntax? |
| 18:52.40 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 18:52.42 | Mercestes | pickupexten = *8 |
| 18:53.07 | djPepse | is that for picking up a call that's on hold? |
| 18:53.15 | [TK]D-Fender | Divious1: Migration do/from WHAT? |
| 18:53.19 | Mercestes | djPepse, It answers a ringing phone that's not at your desk. |
| 18:53.26 | djPepse | oh i see. |
| 18:55.26 | plasmid | Mercestes, coolbeans. Thanks.. testing in a bit. |
| 18:55.29 | Mercestes | coolbeans! i love it. :D That takes me back |
| 18:55.29 | djPepse | I've been trying to figure out transfering :) |
| 18:55.29 | Mercestes | djPepse, try the transfer button |
| 18:55.29 | Divious1 | from an old distribution of linux into a newer one? |
| 18:55.30 | djPepse | well, i'm using a cellphone (E61 with built-in SIP) and an ATA |
| 18:55.30 | djPepse | they don't have those |
| 18:55.30 | Mercestes | Divious1, Can you give us the output of dd if=/dev/urandom of=...haha, just kidding. |
| 18:55.31 | Divious1 | lol ok |
| 18:55.31 | plasmid | Mercestes, is it: pickupexten = *8 OR pickupexten => *8 because the rest of the entries have that arrow |
| 18:55.32 | Mercestes | Divious1, and...are you trying to upgrade your linux? |
| 18:55.32 | Mercestes | Divious1, ...why not...upgrade the existing box? |
| 18:55.32 | Divious1 | yep my boxes but i want to keep my old databases and conf intact |
| 18:55.41 | Divious1 | I should have mentioned I'm kinda of a noob at this |
| 18:55.41 | Mercestes | plasmid, My example is pickupexten = *8 , but some arses in Digium want to rearrange the syntax every other revision just to feel 133t so there is no telling now. |
| 18:55.48 | Mercestes | Divious1, I would just .....reinstall on the new box. |
| 18:55.48 | Katty | it's a little different than i expected, [TK]D-Fender. |
| 18:55.48 | plasmid | Mercestes, lol.. okie dokie. |
| 18:55.57 | [TK]D-Fender | Divious1: * doe not have databases (other than ASTDB). that file is clearly copyable. the rest is just backing up your configs, VM folder,etc |
| 18:56.03 | Katty | [TK]D-Fender: answer, add auto answer header, wait 1 second, play back 3,2,1 dial sip extens, then playbackbeep |
| 18:56.07 | djPepse | Mercestes: Do you have any suggestions in my problem? Skype gateway, *, ATA.. all sip, all on local subnet.. when skype calls ATA, only audio from ata to skype is heard. When ATA calls skype, 2way audio is fine. |
| 18:56.08 | Katty | [TK]D-Fender: i think our 50 year old can handle that |
| 18:56.17 | djPepse | in/on |
| 18:56.26 | [TK]D-Fender | Katty: Woah there! Lets not rattle the cage too much! |
| 18:56.29 | Mercestes | Divious1, the only things you'll need is /etc/asterisk /var/log/asterisk /var/lib/asterisk/ /var/spool/asterisk and that's pretty much everything |
| 18:56.32 | Katty | [TK]D-Fender: :P |
| 18:56.40 | Katty | [TK]D-Fender: if she can't count down till beep, then she's got a major issue. |
| 18:57.05 | Mercestes | djPepse, firewall? |
| 18:57.09 | [TK]D-Fender | Katty: make a recording saying "wait until the damn beep before talking!" |
| 18:57.16 | djPepse | mercestes: no firewall |
| 18:57.21 | Mercestes | djPepse, no clue. |
| 18:57.32 | Mercestes | djPepse, surely there is NAT to Skype though. |
| 18:57.38 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 18:57.43 | djPepse | well audio between skype and the blagoblag is fine |
| 18:57.53 | djPepse | it's just localy between the skype gateway and the ATA |
| 18:57.58 | [TK]D-Fender | Divious1: /var/lib/asterisk/astdb <- |
| 18:58.07 | Katty | [TK]D-Fender: LOL |
| 18:58.13 | Mercestes | I think /var/lib/asterisk will get that. :P |
| 18:58.17 | djPepse | skype and skype gateway software are on same machine |
| 18:58.29 | [TK]D-Fender | Mercestes: He asked specific, I answered specific. |
| 18:58.33 | sandorp | I need to re-install the OS on my asterisk machine, I've backed up /etc/zaptel.conf and /etc/asterisk ... do I need anything else? |
| 18:58.35 | Mercestes | djPepse, well, there is still nat from the ATAs perspective. I would treat it as NAT. |
| 18:58.47 | djPepse | hm, why? |
| 18:58.51 | djPepse | sip ends at skype gateway |
| 18:59.03 | djPepse | cause then it's an audio device to skype |
| 18:59.09 | [TK]D-Fender | sandorp: Maybe your voicemail setup, etc... THINK about the other supporting folders... |
| 18:59.12 | djPepse | that's all within the OS |
| 18:59.12 | Mercestes | sandorp, If you have to ask...probaly. See my above posts for Divious1 |
| 18:59.21 | [TK]D-Fender | Mercestes: he jsut joined. |
| 18:59.29 | Mercestes | oh |
| 18:59.36 | Mercestes | <PROTECTED> |
| 18:59.39 | Mercestes | there |
| 19:01.05 | sandorp | Mercestes: thanks |
| 19:01.05 | Divious1 | really, thanks what about the mysql databases? r they in var/lib/asterisk? |
| 19:01.05 | Mercestes | sandorp, No problem. Good luck. |
| 19:01.05 | [TK]D-Fender | Divious1: Mysql has NOTHING to do with * |
| 19:01.05 | Mercestes | Divious1, ... of course not, silly |
| 19:01.06 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 19:01.06 | Mercestes | Divious1, mysqldump -u -p -Dasterisk |
| 19:01.06 | Mercestes | I think |
| 19:01.06 | [TK]D-Fender | Divious1: that is its own little universe |
| 19:01.06 | Divious1 | gotcha |
| 19:01.06 | [TK]D-Fender | Divious1: with its own support channel. |
| 19:01.06 | Mercestes | Assuming you named your asterisk database "asterisk" and not ..."spot" |
| 19:01.06 | Divious1 | what about if I wanted to backup mysql dbs also |
| 19:01.08 | Mercestes | ... |
| 19:01.08 | Mercestes | man mysqldump |
| 19:01.22 | [TK]D-Fender | Divious1: again, that is a MYSQL quesiton that has nothing to do with *. |
| 19:01.55 | Divious1 | thanks |
| 19:04.47 | Divious1 | thank you very much for your help |
| 19:04.51 | [TK]D-Fender | Mercestes: I suppose its hardly even a GUESS as to the reason behind that line of questioning, isn't it? :) |
| 19:05.47 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 19:11.33 | Katty | [TK]D-Fender: paging is Boss Approved(tm) ^_^ |
| 19:11.45 | [TK]D-Fender | Katty: whee! |
| 19:11.51 | Katty | also. |
| 19:11.53 | Katty | right now |
| 19:12.03 | Katty | it amuses me to transfer a client to this extensions |
| 19:12.12 | Katty | and have the client blaring across all the speakers. |
| 19:12.16 | Katty | ^_^ |
| 19:12.41 | Katty | i'll fix that later. |
| 19:12.55 | Katty | somehow |
| 19:14.24 | [TK]D-Fender | Katty: You should insurreptiously sneak in a "silent" targeting page for "DHS" usage ;) |
| 19:15.24 | *** join/#asterisk ToyMan (n=Stuart@74-32-22-252.dsl1.mdl.ny.frontiernet.net) |
| 19:15.29 | [TK]D-Fender | Katty: Means use an alert-info that auto-answers WITHOUT A SOUND so you can spy on people! |
| 19:16.15 | *** join/#asterisk mxmasster (n=Max@129.47.12.101) |
| 19:16.15 | mocker | Weeeird. I get a LEC announcment *after* a dialed extension is rining. |
| 19:16.18 | mxmasster | hi all |
| 19:16.29 | mocker | Like it rings for 10 seconds and then gives me a LEC all circuits busy |
| 19:16.29 | mxmasster | i have a sip endpoint |
| 19:16.38 | mxmasster | i want to configure a dialplan to send calls to this endpoint |
| 19:16.49 | mxmasster | but i want to limit the number of calls to only x concurrent |
| 19:16.56 | mxmasster | how do I configure this? |
| 19:17.14 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 19:17.17 | [TK]D-Fender | mxmasster: lookup "GROUPCOUNT" on the WIKI |
| 19:17.19 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:17.25 | mxmasster | thanks |
| 19:18.24 | [TK]D-Fender | mxmasster: and "checkgroup" |
| 19:21.45 | mocker | Weird, having the 'm' option in my Dial command would give a LEC announcement when going to voicemial. |
| 19:22.06 | mocker | Didn't need the m option anyway, but still strange. |
| 19:25.21 | *** join/#asterisk MarkWD (n=Mark@rrcs-67-78-88-186.sw.biz.rr.com) |
| 19:25.27 | *** join/#asterisk Exhar (n=Roy@cl-974.ams-05.nl.sixxs.net) |
| 19:28.36 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:28.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:32.08 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de) |
| 19:32.42 | *** join/#asterisk jkimball4 (n=user@ip24-252-32-248.om.om.cox.net) |
| 19:33.28 | jkimball4 | I'm working with realtime in Asterisk and am curious about how Asterisk knows when an update occurs in the database. |
| 19:33.41 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 19:33.44 | jkimball4 | It seems that Asterisk's connection is always "sleeping" |
| 19:34.04 | _VoiceMeUp_COM | kik\ |
| 19:34.09 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:34.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:36.16 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:36.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:38.43 | fall0ut | hrm, so no way to check to see if a meetme conf already exists? |
| 19:39.00 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:39.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:41.22 | *** join/#asterisk mrdigital (n=mrdigita@207-172-228-21.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 19:41.32 | mrdigital | when i record a voicemail greeting the greeting never plays on the phone |
| 19:41.46 | mrdigital | but if yuo dial the VM extenstion it plays the greeting |
| 19:43.16 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 19:43.36 | yannj_fr | Hello everybody |
| 19:44.00 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:44.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:44.05 | yannj_fr | does anybody have test the compatibility with QSIG? |
| 19:44.21 | [TK]D-Fender | mrdigital: "show application voicemail" |
| 19:44.32 | mrdigital | huh [TK]D-Fender? |
| 19:45.01 | [TK]D-Fender | mrdigital: Go type that into * CLI and read the instructions on how you should be calling voicemail to PLAY the appropriate message mack |
| 19:45.04 | [TK]D-Fender | back* |
| 19:45.21 | mrdigital | what instructions though? |
| 19:45.38 | mrdigital | it looks like the recording isnt in the right format |
| 19:45.43 | mrdigital | but i used asterisk to record the greeting |
| 19:46.01 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:46.01 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:46.10 | [TK]D-Fender | mrdigital: type the line I gave you into * CLI and READ. |
| 19:46.34 | mrdigital | ok |
| 19:47.28 | mrdigital | the wav isnt in the right format |
| 19:47.31 | mrdigital | hmm |
| 19:47.56 | [TK]D-Fender | mrdigital: how did you record it? |
| 19:48.03 | mrdigital | though the voicemail prompts |
| 19:48.17 | [TK]D-Fender | mrdigital: then that shouldn't be an issue. |
| 19:48.18 | mrdigital | the wav will play if i call it from the softphones if i call from a pstn line |
| 19:48.19 | mrdigital | it wont play |
| 19:48.33 | mrdigital | it says its playing but skips over it |
| 19:48.35 | [TK]D-Fender | mrdigital: pastebint he CLI output of both attempts |
| 19:48.37 | mrdigital | as if its not in the right format |
| 19:48.52 | [TK]D-Fender | at verbose 10 |
| 19:50.23 | mrdigital | http://pastebin.ca/507873 |
| 19:50.35 | yannj_fr | does anybody have test the compatibility with QSIG? |
| 19:50.47 | [TK]D-Fender | mrdigital: Is this the one that WORKS? |
| 19:51.17 | mrdigital | ohhhhh |
| 19:51.18 | mrdigital | i see |
| 19:51.22 | mrdigital | its not even playing the greeting at all |
| 19:51.36 | [TK]D-Fender | mrdigital: GOOD.. you are beginning to PAY ATTENTION. |
| 19:51.36 | mrdigital | if i call the vm exten directly it shows it playing the wav |
| 19:52.05 | mrdigital | any idea? |
| 19:52.07 | [TK]D-Fender | -- Executing VoiceMail("IAX2/voicemeup-2", "1020@default|s") in new stack <- this is told not to play anything |
| 19:52.29 | [TK]D-Fender | mrdigital: this is FreePBX BS. Don't even DREAM of asking us to suggest a fix for it :) |
| 19:52.30 | mrdigital | could it be because of a ringgroup? |
| 19:53.32 | [TK]D-Fender | mrdigital: Look at the options the interface provides you. If you don't like it, or it doesn't work, we can't help you. |
| 20:02.03 | *** join/#asterisk Skaag (n=skaag@80.178.5.165.adsl.012.net.il) |
| 20:05.00 | sandorp | I've been testing asterisk (1.2) and am ready to build a system for production use ... any reason I should stick to 1.2 instead of using 1.4? |
| 20:05.21 | blitzrage | not really |
| 20:05.27 | blitzrage | I use 1.4 in production -- very stable for me |
| 20:05.29 | Mercestes | \1.2 works and there is no 1.4 bs |
| 20:05.54 | russellb | except 1.2 is not going to be maintained for forever ... |
| 20:06.04 | russellb | if you're doing a new install, I *******STRONGLY********* recommend using 1.4. |
| 20:06.05 | sandorp | Mercestes: can you elaborate on "1.4 bs"? |
| 20:06.54 | Mercestes | not really. I just didn't like it. This was like...1.4.0 tho, I hear it's pretty ok now |
| 20:07.21 | Mercestes | They randomly changed all the syntax for no good reason tho |
| 20:07.33 | [TK]D-Fender | Mercestes: Thats not a |
| 20:07.45 | [TK]D-Fender | "problem" really.. jsut a speed-bump |
| 20:07.46 | Mercestes | your right, it's not |
| 20:07.50 | *** join/#asterisk Greek-Boy (n=g@196.45.144.42) |
| 20:07.51 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:07.55 | sandorp | so I won't be able to use the config files I've built during testing? |
| 20:07.56 | Mercestes | preemptive enter complex? |
| 20:08.04 | [TK]D-Fender | Mercestes: Broken channel drivers, etc... now THAT would be :) |
| 20:08.21 | [TK]D-Fender | Mercestes: its right next the the " key :) |
| 20:08.29 | Qwell | russellb: heh, cheater |
| 20:10.02 | russellb | how did i cheat? |
| 20:10.37 | Qwell | What you just talked about :D |
| 20:10.56 | sandorp | should I use the stable, released 1.4 package or should I download the latest 1.4 src ? I know I had to recompile 1.2 because my new TDM808 card would not work with the 1.2 version that was installed |
| 20:11.09 | Qwell | sandorp: it should be supported in the latest 1.4 release |
| 20:12.03 | sandorp | will my 1.2 configs work with 1.4? |
| 20:12.11 | Mercestes | no |
| 20:12.26 | Mercestes | random new syntax |
| 20:12.47 | sandorp | ooh, that's a big bummer |
| 20:13.20 | Qwell | sandorp: read UPGRADE.txt in the 1.4 source |
| 20:13.46 | sandorp | ok |
| 20:15.38 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 20:16.39 | boch | is it possible to force * not using rtp? just all reinvites? |
| 20:20.44 | *** join/#asterisk Skaag (n=skaag@80.178.86.64.adsl.012.net.il) |
| 20:23.10 | *** join/#asterisk iEatBabies (n=DudeMan@asa.redglaze.com) |
| 20:23.50 | iEatBabies | i have a question about asterisk compatibility. The 3com 3102 business phone support sip, does this mean that they will work in an asterisk system? |
| 20:24.09 | dasuberdavid | iEatBabies: as long as it uses the standard sip protocol it shoud work |
| 20:24.37 | *** join/#asterisk drrt (n=junior@ppp-static2-140.tis-dialog.ru) |
| 20:25.21 | iEatBabies | thanks dasuberdavid. i was hoping to get a uber fast responce |
| 20:25.28 | dasuberdavid | iEatBabies: no problem |
| 20:27.17 | *** join/#asterisk oej (n=olle@69-94-197-20.biltmorecomm.com) |
| 20:27.59 | *** join/#asterisk adr3nalin3 (n=adr3nali@65-123-189-218.dia.static.qwest.net) |
| 20:30.13 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:31.15 | *** join/#asterisk [hC] (n=hardcore@d154-5-29-79.bchsia.telus.net) |
| 20:37.22 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 20:39.27 | *** join/#asterisk ber123 (i=brad@neu.cow.org) |
| 20:39.33 | ber123 | Hi, |
| 20:39.38 | ber123 | I want to use the A option with the Dial command |
| 20:39.47 | ber123 | whats the syntax to do that? Dial |
| 20:39.58 | ber123 | i didnt see an example explicitly on voip-info |
| 20:41.13 | *** join/#asterisk boster (n=boster@38.98.147.68) |
| 20:41.24 | ber123 | Dial(type1/identifier1 [& type2/identifier2 [& type3/identifier3 ... ] ], timeout, options, URL) |
| 20:41.33 | *** join/#asterisk VJFROMGT (n=vijay_0@190.80.51.16) |
| 20:41.56 | ber123 | so it would just be Dial(SIP/XXX@VOIP.PROVIDER,60,A(play_announcement.gsm)) ? |
| 20:42.37 | Mercestes | Ber123: Dial(SIP/69@sip.com,60,a(you_sound_cute)) |
| 20:42.40 | VJFROMGT | i have a client who needs to connect to me but his sip gateway is designed to authenticate by IP is there anyway i can set an extension to authenticate by IP only |
| 20:46.03 | *** join/#asterisk fiber0pti (n=Sean@216.31.101.41) |
| 20:47.30 | Mercestes | fiber0pti, Is that what you call several orange cables in a bundle??? |
| 20:49.40 | *** join/#asterisk FastFeet (n=FastFeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com) |
| 20:49.45 | fiber0pti | heh |
| 20:49.47 | fiber0pti | sure |
| 20:49.48 | fiber0pti | :P |
| 20:52.30 | *** part/#asterisk fiber0pti (n=Sean@216.31.101.41) |
| 20:53.23 | anonymouz666 | haha |
| 20:53.47 | anonymouz666 | I just removed 10GB |
| 20:53.55 | anonymouz666 | these days |
| 20:54.54 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 20:55.30 | *** part/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 21:00.06 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 21:01.23 | ber123 | thanks mercestes |
| 21:01.38 | Mercestes | sorry. Hope you have good backups. |
| 21:01.43 | Mercestes | ...oh wait, no, I really helped you. |
| 21:01.44 | Mercestes | :) |
| 21:01.47 | Mercestes | Your welcome. |
| 21:05.55 | Qwell | my welcome what? |
| 21:06.09 | d3wayne | your welcome mat |
| 21:06.45 | boster | is it possible to push calledID information on a LD T1? |
| 21:06.54 | boster | eer, callerid |
| 21:06.59 | Mercestes | woolbeo, 3 pris all from the same telco or different telcos?? |
| 21:07.16 | woolbeo | Mercestes, 2 from one telco one from another |
| 21:07.25 | Mercestes | There's your problem. |
| 21:07.45 | Hmmhesays | it would be cool if asterisk dialplan had an eval type function |
| 21:07.50 | Mercestes | you have two timing sources and timing source A is jacking up timing source B. |
| 21:08.15 | *** part/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 21:08.21 | Mercestes | there is special firmware to make the sangoma time per port instead of the entire card (I'm assuming you have it since it works at all) but I've found out from experience that that is nowhere near perfect. |
| 21:08.29 | Mercestes | your best bet is to seperate your telcos into seperate boxes |
| 21:08.39 | Mercestes | then IAX2 or SIP from box to box |
| 21:08.47 | woolbeo | Mercestes, seriously? that sux... |
| 21:08.53 | Mercestes | yes, and....yes. |
| 21:08.59 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:09.48 | woolbeo | Mercestes, I just convinced my boss to buy a new a104d, to replace a t410p with a bad port... He's not going to like the idea of buying another another t1 card... :( |
| 21:10.02 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 21:10.15 | Mercestes | and a new server too. :) |
| 21:10.38 | woolbeo | We have plenty of servers sitting around doing nothing... |
| 21:10.57 | woolbeo | We have a couple t100ps but I really really hate them. |
| 21:11.30 | FastFeet | I need some help with an IAX2 problem: http://pastebin.ca/508052 |
| 21:11.35 | FastFeet | I keep getting: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 21:12.04 | FastFeet | Any insight would be appreciated. |
| 21:12.17 | FastFeet | http://pastebin.ca/508052 |
| 21:13.48 | FastFeet | I have no problems with my box, but my production box is giving me this error: WARNING[7850]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 21:16.14 | woolbeo | Mercestes, so with two different pris from two telcos, how do I setup the spans in zaptel.conf? right now I have the timing in zaptel.conf set to telco 1 pri 1 primary, Telco 1 pri 2 secondary, and Telco 2 pri 1 tertiary(sp?) and in wanpipeX.conf I have TE_REF_CLOCK = 0 and TE_CLOCK = NORMAL. Is this how I should have it set to do per port timing? |
| 21:19.09 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:20.47 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
| 21:21.25 | n00dle | Anyone working with SLA? I have tried a simple config, but: [May 24 15:13:47] WARNING[15644]: pbx.c:1797 pbx_extension_helper: No application 'SLAStation' for extension (sla-stations, station224_line1, 1) |
| 21:21.26 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:21.34 | Mercestes | woolbeo, It hasn't worked for me at all |
| 21:21.51 | VJFROMGT | <PROTECTED> |
| 21:22.58 | Mercestes | VJFROMGT, with my limited amount of knowledge I believe that is an all or nothing thing. |
| 21:23.07 | Mercestes | But if his SIP gateway authenticates by IP only then his SIP gateway sucks |
| 21:23.46 | *** join/#asterisk marta (n=marta@host35-199-dynamic.17-87-r.retail.telecomitalia.it) |
| 21:24.05 | marta | hi |
| 21:24.28 | woolbeo | Mercestes, What about digiums cards with two telcos? |
| 21:25.25 | Mercestes | woolbeo, no. |
| 21:25.58 | woolbeo | This explains alot about the problems we have been having... |
| 21:27.04 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 21:27.47 | skirmisha | guys |
| 21:27.51 | skirmisha | any ideas about this |
| 21:27.53 | skirmisha | channel.c:785 channel_find_locked: Avoided deadlock for '0x8312ce0', 10 retries! |
| 21:28.34 | n00dle | Anyone here got SLA to work at all? |
| 21:28.38 | woolbeo | Mercestes, Let me make sure this is the problem. We have two long distance pris from UCN, and a local pri from Qwest. Qwest provides the loop for all three, so does this count as two seperate telcos? |
| 21:28.52 | marta | does anyone have ideas of what I can do to have a core file from asterisk? (I tryed ulimit in bash and unlimit coredumpsize in tcsh with asterisk -gcvvv) |
| 21:29.41 | *** join/#asterisk marc\cba (n=marc@cpc2-whit2-0-0-cust200.cdif.cable.ntl.com) |
| 21:29.54 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:31.14 | skirmisha | marta what is the problem? |
| 21:32.01 | woolbeo | n00dle, sorry what is SLA? When I see SLA I think Service Level Agreement. |
| 21:32.05 | marta | I want a dump (corefile) when a segmentation fault happens |
| 21:32.26 | marta | I don't know what I am missing |
| 21:32.41 | skirmisha | should be in /tmp dir |
| 21:33.02 | marta | no, it isn't |
| 21:33.17 | n00dle | SLA = Shared Line Appearance (key- or hybrid-system emulation) |
| 21:33.25 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 21:34.02 | n00dle | src/asterisk-1.4.4/doc/sla.pdf |
| 21:34.33 | woolbeo | n00dle, ah share line appearance... stupid me... didn't even have to look |
| 21:34.50 | skirmisha | marta did u search for it |
| 21:34.53 | woolbeo | n00dle, if you would have said SIP SLA, I would have understood |
| 21:37.02 | n00dle | woolbeo: I have followed config examples, yet it doesn't work. |
| 21:37.57 | n00dle | I read the dox (actually doc, singular) and did forum searches, but haven't come up with much in the way of a working example. |
| 21:38.30 | woolbeo | n00dle, have you read the blog on asterisk.org about it? |
| 21:38.45 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 21:38.58 | n00dle | I drifted through it, but it seemed to be only about pre-1.4 stuff before the major overhaul. |
| 21:39.04 | coolbeans | Is there any magic to getting music on hold when a sip phone is placed on hold? (Polycom) |
| 21:39.51 | marta | skirmisha: I searched in /tmp and in $HOME. I see only "Segmentation fault" and not "Segmentation fault core dumped". and I forgo to say that I'm using trunk. ideas? |
| 21:40.28 | skirmisha | how do you start your asterisk server |
| 21:40.39 | skirmisha | do you use safe_asterisk script |
| 21:40.59 | n00dle | woolbeo: I just re*read it... it has much less than the sla.pdf it refers to in the footer. |
| 21:41.47 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 21:42.30 | marta | skirmisha: I use asterisk -gcvvv |
| 21:43.54 | n00dle | Anyhow, the * complains that SLAStation is an unknown app... I don't see a reference to having to load a special module (no module has any name like sla) or enabling a specific switch on compile... |
| 21:44.09 | Qwell | n00dle: meetme |
| 21:44.51 | coolbeans | For some reason, we have no music on hold when a call is placed on hold from a sip phone (polycom) but our queues and meetme MOH works fine. Any suggestions? Asterisk doesn't report any errors. |
| 21:44.52 | skirmisha | marta try find / -name core* |
| 21:45.26 | coolbeans | And asterisk also doesn't show " -- Started music on hold, etc" |
| 21:45.48 | FastFeet | I really need some help with this IAX2 problem: http://pastebin.ca/508052 |
| 21:45.56 | n00dle | Qwell: I have heard of that... thanks for the pointer, I will go read more. |
| 21:46.06 | FastFeet | Google and voip-info.org is of no help. |
| 21:47.19 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:47.25 | coolbeans | For some reason, we have no music on hold when a call is placed on hold from a sip phone (polycom) but our queues and meetme MOH works fine. Any suggestions? Asterisk doesn't report any errors. |
| 21:49.24 | coolbeans | Do you have to explicitly set MOH in the sip peer definition or sip.conf? Dialplan? |
| 21:49.25 | FastFeet | Anyone have any clues as to what is causing this: |
| 21:49.26 | FastFeet | dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 21:49.26 | FastFeet | <PROTECTED> |
| 21:50.47 | [TK]D-Fender | FastFeet, we'd have to see the line that CAUSES the error, and the config is any entries it uses |
| 21:51.35 | FastFeet | Thanks again Fender; Will this help : http://pastebin.ca/508052 |
| 21:51.36 | coolbeans | Let me rephrase, if I hit a hold button on a Polycom 501, should asterisk play MoH to the caller and (when verbose is set to 9) display the " -- Starting Music On Hold .." message? |
| 21:52.08 | coolbeans | 1.2.18 |
| 21:54.05 | FastFeet | My Home Asterisk Box (which works) doesn't get this message, but my production box has an identical config, yet gives me this problem |
| 21:55.09 | [TK]D-Fender | coolbeans, yes,if a MoH class is defined for that channel |
| 21:55.24 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:55.57 | coolbeans | [TK]D-Fender: Thanks. If I don't have one defined, should it default to 'default'? And where would I explicitely define the class? (Thanks for the insight) |
| 21:56.10 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:56.11 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 21:56.37 | [TK]D-Fender | coolbeans, in the phone [something] |
| 21:57.03 | [TK]D-Fender | FastFeet, Ok, looks largely legit. verify your host name and try to ping it and make sure it resolves |
| 21:57.26 | FastFeet | dude thanks, I shall do that. |
| 21:57.54 | coolbeans | [TK]D-Fender: .. in the phone [something] .. Sorry, you lost me on this?! lol |
| 21:58.03 | n00dle | Qwell: Ok, meetme seems simple enough, but no doc gives me anyting to do in meetme.conf. |
| 21:58.26 | [TK]D-Fender | coolbeans, in each phons definition like [20] , type=friend , secret=hhfdd and so on |
| 21:58.29 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 21:58.43 | *** join/#asterisk b00gz (n=b00gz@d233-124-245.col.wideopenwest.com) |
| 21:58.56 | yannj_fr | Is there people using Asterisk for Business? |
| 21:59.07 | coolbeans | Ahh.. I'm using realtime static, so in my peer definition, specify MUSICONHOLD and the the class, "default" ?? |
| 21:59.16 | [TK]D-Fender | coolbeans, "musicclass=[yourclassnameherewithoutbrackets]" |
| 21:59.30 | [TK]D-Fender | yannj_fr, No, its a complete marketing failure and only there to confuse you. |
| 21:59.47 | yannj_fr | what do you mean? |
| 21:59.56 | lesouvage | yannj_fr: most of us I guess |
| 21:59.59 | [TK]D-Fender | coolbeans, Make sure the class you wish you to exists and is ready to function to your expectations |
| 22:00.33 | [TK]D-Fender | yannj_fr, I mean DUH, of course people use it. Not most people here, but clearly enough to warrant it |
| 22:00.45 | [TK]D-Fender | yannj_fr, got a meaningful question about it to ask? |
| 22:01.24 | yannj_fr | no just want to know if people using it are satisfied by the support |
| 22:01.45 | Mercestes | yannj_fr, The only people who use a4b are the ones too stupid to tell Polycom "yes we have it" when they don't to get the Polycom support. |
| 22:01.58 | Mercestes | which is one of the reasons I cite to prove Polycom ppl are retarded |
| 22:02.17 | yannj_fr | ok |
| 22:02.26 | yannj_fr | I need to us asterisk in my company |
| 22:02.38 | Mercestes | the other reason is that they require RHEL. ...'tards |
| 22:02.43 | yannj_fr | but my boss say me "ok , if we can have a full support" |
| 22:02.50 | lesouvage | oej: are you there (edw) |
| 22:02.59 | Mercestes | yannj_fr, Buy a support contract from Digium. |
| 22:03.17 | Mercestes | Or buy a4b, leave the box sealed, put it in a nice glass case, then install the *source*. |
| 22:03.57 | coolbeans | Is there an explicit way to disable silence surpression on polycom 550's and 650's that anyone knows of? |
| 22:04.00 | yannj_fr | support contract is different from asterisk business edition |
| 22:05.05 | lesouvage | mvanbaak: we have 175 registrations for coming thursday. |
| 22:05.23 | [TK]D-Fender | yannj_fr, support is the same... |
| 22:06.03 | coolbeans | Is MoH automatically disabled when a call is using g729? |
| 22:06.19 | Mercestes | probably better, really. |
| 22:06.32 | [TK]D-Fender | coolbeans, nope, but may break is you don't have licenses to cover it or G.729 encoded MoH |
| 22:06.37 | Mercestes | You go, "I have asterisk for business" the phone tech will go "what? ......oh yea, that! we forgot about that." |
| 22:06.43 | coolbeans | hrm... |
| 22:06.53 | [TK]D-Fender | Ok, off for a few hours, back later-ish |
| 22:07.01 | yannj_fr | is the digium support good? of equivalent that asking a question on irc? |
| 22:07.04 | Mercestes | l8s, Fender |
| 22:07.19 | yannj_fr | or* |
| 22:07.30 | Mercestes | yannj_fr, I would say Digium support is far better than IRC support. |
| 22:07.56 | Mercestes | for one, D-Fender doesn't work for digium so you'll never have to deal with him and his random capitialization to emphasize random words that dont' really make sense to emphasize. |
| 22:08.01 | b00gz | I am having issues hearing anything I am over a Internet connection connecting to the Asterisk Box. I am in sip_nat.conf is there anyway to just add everyone as a trusted source for testing? |
| 22:08.19 | Mercestes | two, they'll never ask you to give you the output of dd if=/dev/urandom of=wherever/your/primary/hard drive/is |
| 22:08.34 | Mercestes | three, Digium can/may remote into your machine and just fix it.. |
| 22:09.01 | Mercestes | four, You can expect Digium to give you support, you have to ask us and we can deny you if we don't like you. |
| 22:09.33 | yannj_fr | ok... I still dont understand why they do several contract regarding to the extensions number... |
| 22:09.50 | Mercestes | yannj_fr, $$$. |
| 22:09.53 | yannj_fr | lol |
| 22:09.55 | Mercestes | b00gz: ~freepbx |
| 22:10.02 | yannj_fr | dummie I am! |
| 22:10.05 | Mercestes | ... |
| 22:10.07 | Mercestes | ~freepbx |
| 22:10.23 | jbot | i heard freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:10.23 | Mercestes | yannj_fr, for more money |
| 22:10.52 | yannj_fr | I understand... the contract is really expensive, regarding to the fact asterisk is free! |
| 22:11.30 | Mercestes | yannj_fr, When you go to a mechanic, with a busted car, and they replace a fuse, it takes them 30 seconds, and they charge you $90. |
| 22:11.44 | Mercestes | they are not charging you for the 30 seconds it took to do it, they are charging you for knowing to replace the fuse. |
| 22:12.00 | Mercestes | Asterisk is the same. Your not paying for asterisk, your paying for the knowledge and support. |
| 22:12.05 | yannj_fr | you're true |
| 22:12.11 | yannj_fr | right* |
| 22:12.24 | Nugget | I pay for the opportunity to pick on file. |
| 22:12.25 | yannj_fr | (fuc**** french) |
| 22:13.52 | yannj_fr | Do you think they can also fix some bugs, I mean by coding patches? |
| 22:14.53 | lesouvage | Does it make sence to use RAID1 on an Asterisk production machine that will be heavely used or is this adding serious extra latency and taken resources beyond proportions? |
| 22:15.06 | Mercestes | yannj_fr, I agree. fuc***** frency |
| 22:15.07 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 22:15.23 | Mercestes | lesouvage, use raid 5. |
| 22:15.35 | yannj_fr | ;-) mercestes, where do you come from? |
| 22:15.36 | Mercestes | lesouvage, and....yes, it makes sense. |
| 22:15.41 | Mercestes | yannj_fr, Texas |
| 22:15.52 | yannj_fr | ok, I understand why you say that |
| 22:15.55 | Mercestes | lol |
| 22:16.01 | Mercestes | yea, us rednecks hate everyone...'cept rednecks |
| 22:16.05 | Mercestes | ...'cept, I hate rednecks too |
| 22:16.59 | yannj_fr | But it still be quite difficult for french people to speak in a perfect english, because of several words meanings. |
| 22:17.11 | Mercestes | Dont feel bad.... |
| 22:17.20 | yannj_fr | thk |
| 22:17.22 | Mercestes | it will be quite difficult for texans to speak in a perfect english |
| 22:18.10 | yannj_fr | I know, last time I met a texan, I didnt understand any word ('cept f*ck) |
| 22:18.53 | Nugget | Never ask a man if he's from Texas. If he is he's going to work it into the conversation anyway, and if he's not you'll spare him the embarassment. |
| 22:19.34 | coolbeans | [TK]D-Fender (and others): No matter what I do I can't get MoH to play when I put one of our polycom phones on hold... Setting the musicclass in my sip.conf doesn't help. It's almost like I have it turned off. Is there a global MoH setting? Maybe it's in the phone config files? Any help would be greatly appreciated. |
| 22:20.07 | *** join/#asterisk fbffff (n=fbffff@24-148-35-123.grn-bsr1.chi-grn.il.cable.rcn.com) |
| 22:21.40 | *** join/#asterisk bluelinq2 (n=bluelinq@dsl-7-36.cofs.net) |
| 22:22.02 | bluelinq2 | hi there.. does anyone know if is possible to support login/logout in a phone to support |
| 22:22.02 | bluelinq2 | <PROTECTED> |
| 22:22.04 | boch | is it possible to force * not using rtp? just all reinvites? |
| 22:22.34 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:23.25 | FastFeet | <[TK]D-Fender> I was able to resolve my host name and verified ping tests |
| 22:23.43 | FastFeet | Still having issues with http://pastebin.ca/508052 |
| 22:24.07 | lesouvage | Mercestes: Do you advice raid5 because it is faster or is there any other reason (just did some reading on wikipedea) |
| 22:24.25 | FastFeet | 2 weeks searching for answers and getting now where. Ready to pull what little hair I have out. |
| 22:24.35 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:25.07 | bluelinq2 | raid 5 is slooooooooowww.. |
| 22:25.11 | coolbeans | FastFeet: Firewall? |
| 22:25.23 | bluelinq2 | it is actually the slowest of all raid configurations... |
| 22:25.33 | FastFeet | I do, but I am forwarding UDP 4569 |
| 22:26.04 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:26.10 | n00dle | Ok, I've read all I can find about SLA, requiring zaptel for meetme, looked through configurations.... I'm still getting a 603 on the phone and "[May 24 16:25:02] WARNING[20441]: pbx.c:1797 pbx_extension_helper: No application 'SLAStation' for extension (sla-stations, station224_line1, 1)" |
| 22:26.10 | lesouvage | coolbeans: if you do exten => 5000,s,MusicOnHold() and you call 5000 do you hear music |
| 22:26.27 | coolbeans | lesouvage: Yep. Also works fine in queues and meetme. |
| 22:27.02 | coolbeans | Just not when I put a phone on hold. Don't even get the "-- Starting music on hold" in the console... Like I have it disabled on sip peers or something. |
| 22:27.22 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:27.39 | n00dle | coolbeans: what do you see at the CLI with verbosity of about 10? |
| 22:27.39 | lesouvage | coolbeans: did you try a softphone, just for testing. |
| 22:28.55 | coolbeans | n00dle: Nothing at all as pertaining to music on hold, just the call being initiated. |
| 22:29.05 | coolbeans | lesouvage: Nope... |
| 22:29.46 | lesouvage | coolbeans: if moh works on a softphone I guess you pinpoint the problem to the phone. |
| 22:30.00 | n00dle | Hm. |
| 22:30.30 | coolbeans | How does asterisk know the phone has been placed on hold? A sip message? |
| 22:30.59 | yannj_fr | is there a way to integrate directly in asterisk a tftp server for configuration file used by phone? |
| 22:31.20 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:31.59 | n00dle | coolbeans: I started over with a fresh box after trixbox fsked itself on upgrade... installed the samples and voila... all my sip calls have MOH. |
| 22:32.27 | coolbeans | Right, and that used to be the case with us. Dunno what happened or when.. hrm.. |
| 22:32.55 | n00dle | coolbeans: Yah, when similar stuff happens to me, it makes me wish I'd done the configs in CVS! |
| 22:33.25 | coolbeans | I'm using realtime static... |
| 22:34.24 | Mercestes | lesouvage, it's faster *and* provides redunancy |
| 22:37.58 | coolbeans | Is it possible to convert mp3's to sln via sox? |
| 22:38.24 | n00dle | ..but if sox knows it, then yes. |
| 22:38.55 | bluelinq2 | does anyone know if * supports login in and out of a phone configuration, not necesarily of a queue? |
| 22:39.21 | coolbeans | Is there another name for sln? Is it just a .raw file? |
| 22:39.46 | Mercestes | bluelinq2, Yes. to log out you just unplug the cord. |
| 22:39.50 | lesouvage | Mercestes: I'm actually not sure that the raid controller supports raid5. Btw: suse 10.2 supports the raidcontroller on my asus board while Centos 4.? doesn't. |
| 22:39.59 | bluelinq2 | funny guy... |
| 22:40.05 | Mercestes | Thank you. :) |
| 22:40.14 | bluelinq2 | seriouly now? |
| 22:40.27 | Mercestes | Pretty much |
| 22:40.33 | n00dle | coolbeans: from the voip-info wiki: Converting to sln format |
| 22:40.34 | n00dle | Starting from Asterisk 1.2.0, the .sln (SLINEAR) format seems to be the preferred format. |
| 22:40.34 | n00dle | To convert wav file to sln, use the following command: |
| 22:40.34 | n00dle | <PROTECTED> |
| 22:40.43 | n00dle | Damnation... my timing strikes again. |
| 22:42.04 | bluelinq2 | there must be a way of doing it.. particularly when you have different shift of people using the same phone, but with diff extensions... |
| 22:42.26 | Mercestes | bluelinq2, nope |
| 22:42.37 | Mercestes | bluelinq2, agentcallbacklogin is the only way to do that |
| 22:42.45 | Mercestes | but the phone extension will stay the same. |
| 22:42.52 | Mercestes | you only change the agent who sits there |
| 22:42.59 | Mercestes | which is *precisely* what is happening in reality too |
| 22:43.12 | n00dle | bluelinq2: You could have a virtual extension with an AGI db dip to get it to the right hard extension. |
| 22:43.40 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:43.46 | Mercestes | heh, yea, and you could System(mv sip.1st.shift sip.conf && asterisk -rx 'sip reload') too |
| 22:43.52 | *** join/#asterisk anthm (n=anthm@adsl-76-230-214-27.dsl.milwwi.sbcglobal.net) |
| 22:43.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:44.07 | Mercestes | and system() change your phone configs and asterisk -rx a sip notify polycom_check_cfg <peer> |
| 22:44.07 | boch | do you know if there is a working sip/iax java applet? |
| 22:44.17 | n00dle | Necessity being the mother of invention, I'm sure that someone could come up with a mother of a workaround! |
| 22:44.18 | bluelinq2 | M and if the agents all come a different times? |
| 22:44.21 | Mercestes | but....man, that's alot of work for something agentcallbacklongin takes care of. |
| 22:44.50 | n00dle | I haven't messed with agentcallbacklogin, so... :) |
| 22:45.15 | bluelinq2 | maybe I don't know the agentcallbacklogin.. but does not sound like is what we are looking to do |
| 22:45.50 | bluelinq2 | if you use that, will my phone change extensions to be mine? If not, then your solutions does not work |
| 22:46.02 | Mercestes | ... |
| 22:46.17 | Mercestes | You need to read slower. |
| 22:46.27 | Mercestes | the EXTENSION will stay the same, but the AGENT sitting at the phone will change. |
| 22:46.41 | Mercestes | and AgentCallBackLogIn tells asterisk at which EXTENSION an AGENT can be called from a QUEUE |
| 22:47.21 | Mercestes | so if you create a VIRTUAL EXTENSION of what you want, and point it to a queue named joe_somebody and they log into the REAL extension, then the call to their VIRTUAL EXTENSION will be directed to the REAL EXTENSION which is where they are sitting |
| 22:47.53 | lesouvage | bluelinq2: it is a kind of standard follow me functionality. |
| 22:48.00 | Mercestes | gah, now you got me talking to [TK]D-Fender |
| 22:48.07 | Mercestes | s/to/like/ |
| 22:48.45 | lesouvage | bluelinq2: and actually just what you are looking for. |
| 22:49.30 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 22:50.58 | lesouvage | Mercestes: maybe your capitals make sense :-) |
| 22:51.10 | *** join/#asterisk ssokol (n=ssokol@65-182-39-203.cre.bil.biltmorecommunications.net) |
| 22:51.25 | Mercestes | lesouvage, ROFLMAO |
| 22:51.40 | n00dle | Yeah, what Mercestes said... point the virtual at the queue with the person=agent logged in to that queue. |
| 22:51.45 | lesouvage | ROFLMAO ? |
| 22:51.57 | Mercestes | Rolling On Floor Laughing My Ass off. |
| 22:52.03 | Mercestes | I pick on him alot but he's pretty cool. |
| 22:52.31 | Mercestes | This one time he jumped my ass like Cory at a roofies tasting convention but other than that he could almost pass for a friend of mine. |
| 22:53.17 | JT | who's this? |
| 22:53.28 | Mercestes | D-Fender |
| 22:53.30 | Mercestes | who else?\ |
| 22:53.43 | *** part/#asterisk boster (n=boster@38.98.147.68) |
| 22:53.43 | JT | heh |
| 22:56.23 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 22:59.37 | lesouvage | Mercestes: if I send info about an open source voip event to digium will they add it to the events agenda? |
| 22:59.40 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 22:59.59 | blitzrage | lesouvage: not sure how anyone can respond to that other than Digium |
| 23:04.02 | Mercestes | Dunno |
| 23:04.38 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 23:05.42 | yannj_fr | time to sleep in france (01 am...) , bye everybody |
| 23:05.57 | n00dle | Bonne nuit |
| 23:06.03 | yannj_fr | french? |
| 23:06.50 | n00dle | Non, seulement americain qui a etudie un peu. |
| 23:07.10 | yannj_fr | a a, un américain peut faire des études?? (joke) |
| 23:07.33 | n00dle | Je ne suis pas "normal" :O :) |
| 23:11.08 | marta | I did'n solve my problem, but thanks anyway. i will try tomorrow |
| 23:11.10 | marta | night |
| 23:14.35 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:15.28 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 23:16.38 | *** join/#asterisk zirman (n=zirman@ip194.207.107.216.seg.net) |
| 23:23.16 | *** join/#asterisk sky_net (n=mike@76.182.134.182) |
| 23:27.03 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 23:33.11 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 23:34.07 | JT | my goodness |
| 23:34.15 | JT | i just read a thread on the trixbox forums |
| 23:34.18 | JT | and feel dumber for it |
| 23:34.22 | JT | blind leading the blind |
| 23:35.38 | Mercestes | whcih one? lol |
| 23:35.42 | n00dle | Heh... |
| 23:36.02 | JT | http://www.trixbox.org/forums/vendor-moderated-forums/sangoma/bad-sangoma-a101d |
| 23:36.18 | n00dle | It kept me hemmed in too much, I kept having to add custom stuff to the conf's. |
| 23:36.39 | JT | so sangoma didn't ship drivers with the a101d that worked with trixbox, an oversight yes, but damn these people i would not want anywhere near a PRI :/ |
| 23:40.50 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 23:43.47 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 23:46.30 | n00dle | JT: It took me a bit of wrangling to get my rhino card working in the trixbox... when I reformatted and went with just asterisk, it was not trouble at all. |
| 23:46.55 | supjigatr | Anyone had trouble with Bellsouth PRI not dialing 911? |
| 23:47.16 | JT | n00dle: yeah and the problem is most people using trixbox don't have the competance to properly administer a linux system, such as installing new hardware |
| 23:47.29 | supjigatr | I have pridialplan set to national cause thats what I ordered but they are telling me to send unknown on x11 calls only. |
| 23:47.35 | supjigatr | Can asterisk even do that? |
| 23:47.52 | JT | can't you set it to unknown all the time |
| 23:48.19 | n00dle | Wish I could help with the PRI stuff, but it's not available here. :( |
| 23:48.30 | n00dle | ...therefore I have no experience with it. |
| 23:50.40 | n00dle | ...but I think I may finally have discovered why my SLAStation app wasn't there... |
| 23:51.06 | *** join/#asterisk Skaag (n=skaag@87.69.217.36) |
| 23:51.08 | n00dle | I compiled * before zaptel. DUH! |
