| 00:00.05 | Druken | since when does 3 slices of pizza fill me up... god i can't eat shit lately |
| 00:00.36 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 00:01.38 | LostFrog | apt-get install second-stomach |
| 00:02.24 | Shaun2222 | ok, guess i still need it.... how can i get the extension a agent is assigned to? |
| 00:02.28 | twisted[asteria] | Druken, i'm in the same boat |
| 00:02.49 | dlynes | Druken: cheaper that way, isn't it? |
| 00:02.54 | *** join/#asterisk austinnichols102 (n=austinni@h4608adc4.area4.spcsdns.net) |
| 00:03.45 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.75.171) |
| 00:03.49 | Druken | no.... |
| 00:04.08 | key2 | someone has succeded to use the Wipphone service with Asterisk ? |
| 00:04.11 | *** join/#asterisk demigod2k (n=joey@71-13-80-162.static.bycy.mi.charter.com) |
| 00:04.21 | demigod2k | yaaaaaa finally got rid of most of the echo! that took forever |
| 00:04.58 | Druken | wtf is wipphone ? |
| 00:05.37 | demigod2k | a typo of sipphone? |
| 00:05.53 | dlynes | demigod2k: probably some wireless internet protocol phone |
| 00:05.57 | *** join/#asterisk ACiDV (n=acidv@modemcable247.11-37-24.mc.videotron.ca) |
| 00:05.59 | Druken | twisted[asteria]: why you in the same vote? |
| 00:09.35 | demigod2k | ya my next task is to find a decent cordless IP phone. that is proving quite difficult |
| 00:09.40 | demigod2k | every one I've seen so far works like crap |
| 00:10.14 | Druken | doesn't linksys have one now? |
| 00:10.19 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-52-38.dsl.irvnca.pacbell.net) |
| 00:11.02 | *** part/#asterisk austinnichols102 (n=austinni@h4608adc4.area4.spcsdns.net) |
| 00:11.17 | demigod2k | I saw a linksys that wasnt available for sale yet (maybe that changed) but it looked like crap |
| 00:11.32 | demigod2k | tiny things like "does it have a charging cradle" |
| 00:11.45 | demigod2k | I've never seen one yet that works as well as a $39 cordless phone |
| 00:13.30 | dlynes | demigod2k: it's never gonna happen until wireless networks improve |
| 00:13.49 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 00:13.54 | dlynes | or until mesh is everywhere |
| 00:13.55 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-52-38.dsl.irvnca.pacbell.net) |
| 00:14.27 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 00:14.30 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.75.171) |
| 00:14.36 | demigod2k | or until voip displaces conventional systems |
| 00:14.44 | demigod2k | I havent seen the cisco one yet maybe that's a decent cordless |
| 00:15.07 | dlynes | well for whatever the price those bloodsuckers are selling it for, i should hope so |
| 00:15.12 | demigod2k | I think theres room for an ip cordless in the business environment without needing mesh, etc |
| 00:15.28 | demigod2k | I just want a cordless that goes 100ft around the office and works with my phone system and is less than 400 a phone |
| 00:15.30 | demigod2k | ya know |
| 00:15.39 | dlynes | ummm |
| 00:15.44 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:15.58 | dlynes | demigod2k: Have you ever gotten 802.11b/g to go 100ft, flawlessly? |
| 00:16.08 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.75.171) |
| 00:16.17 | Ariel_ | evening everyone |
| 00:16.22 | dlynes | i.e. where you're doing a download of a large file, walked around the office, and it remains a steady speed? |
| 00:17.35 | Druken | hey Ariel_ |
| 00:17.48 | Ariel_ | Druken, sup |
| 00:18.30 | Druken | nuttin, bored outta my mind with an inside headache, watching dvd's to pass the time... |
| 00:18.32 | Druken | hehe |
| 00:18.46 | Ariel_ | wow |
| 00:19.11 | Ariel_ | I am watching dvd 's as well. But kids movies with my 3 year old. |
| 00:19.32 | Druken | ahh, my son is watching the tv in the livingroom |
| 00:21.08 | justinu | <PROTECTED> |
| 00:21.59 | dlynes | justinu: not to mention getting reception deadspots, randomly |
| 00:22.30 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 00:23.24 | dlynes | tzafrir: so you guys are coming out with some 16-port astribanks and some 8 port fxo astribanks? |
| 00:26.37 | Druken | i'd like to get a generic version of the cordless vonage uses, but the asshole aren't selling one... |
| 00:27.27 | demigod2k | dlynes, back... I'm sure the latency could kill voice (echo). actually I dont even need it to be 802.11b. |
| 00:27.46 | demigod2k | mainly I just want something that actually has a hold/transfer button that works with the system, instead of memorizing tons of pound-key combinations |
| 00:28.04 | demigod2k | otherwise an ATA with a classic phone would be fine. it's just that the integrate very poorly |
| 00:28.23 | dlynes | demigod2k: yeah...i think the best solution, personally would be a cordless phone with a sipura 2000 or sipura 1000 |
| 00:28.40 | dlynes | demigod2k: i.e. a 4.6 or whatever ghz phone |
| 00:28.57 | demigod2k | maybe a cordless with a macro button would do the job though |
| 00:29.12 | dlynes | or a nortel cordless set with base station and roof transmitter |
| 00:29.28 | Druken | hold is easy on an ata, press link or flash and hang the phone up |
| 00:29.47 | dlynes | demigod2k: you can do hook flash to achieve call waiting, transfer |
| 00:30.14 | dlynes | demigod2k: and hold as druken stated |
| 00:30.16 | demigod2k | understood. a few will be easy, a few will be hard. |
| 00:30.53 | demigod2k | I've seen one implementation of it that looked too hard. everybody had cheetsheets on their desk with the commands. |
| 00:31.15 | demigod2k | what I like about the gxp2000 right now is that it has a million buttons that can be programmed to do the common features |
| 00:32.28 | demigod2k | you dont have to remember flash+# to page the intercom, you just can program a button for it. etc |
| 00:33.16 | demigod2k | oh well, I'm still happy that this system didnt cost $15k like an avaya |
| 00:36.35 | *** join/#asterisk DAMWorking (n=DAM@c-24-0-137-128.hsd1.tx.comcast.net) |
| 00:41.11 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:49.25 | *** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool4-a113.nwlnnh.tds.net) |
| 00:49.44 | *** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool4-a113.nwlnnh.tds.net) |
| 00:53.04 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 00:54.03 | [hC] | So, anyone have any idea why the asterisk manager seems to be sending me <unknown> for caller id name and number? I have a program that connects via manager to display caller id popups... doing a tcpdump, it initially sends proper caller id and name, then in a later request, i see it sending <unknown> instead. Am i doing something wrong, or is it possibly a bug, or my clients fault? |
| 00:54.21 | dlynes | demigod2k: The aastras have seven programmable buttons, and they're relatively inexpensive as well |
| 00:54.31 | dlynes | demigod2k: not to mention a good quality full duplex speaker phone |
| 00:56.32 | *** join/#asterisk Omfgicbts (i=ray@cpe-65-189-198-222.neo.res.rr.com) |
| 00:58.35 | Ariel_ | anyone here know what settings are needed for a tdm400p and BT lines... the wiki info is not working out. |
| 00:58.55 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 00:59.21 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 00:59.24 | Omfgicbts | Howdy folks. I'm looking for a way to strech a telephone extension from our existing PBX to a users house. Has anybody seen any docs discussing an asterisk configuration along this line ? |
| 01:00.13 | ManxPower | Omfgicbts, what is the distance? |
| 01:00.28 | *** join/#asterisk frawd (n=frawd@87.223.190.128) |
| 01:00.36 | Omfgicbts | ManxPower several hunders miles in one case. |
| 01:00.36 | wunderkin | is it too far for it to travel on a string? |
| 01:00.43 | wunderkin | i guess so then |
| 01:00.57 | ManxPower | Omfgicbts, Well, you won't do it with analog. |
| 01:01.09 | Omfgicbts | I personaly don't have a problem with the string. Tends to get in the way of other peoples plans though. |
| 01:01.50 | ManxPower | Omfgicbts, you would basically have to set up a mini PBX (asterisk) to tie in with your existing PBX, then install an IP phone at the remote location. However, expect to spend 20 - 30 hours (more if you don't understand linux or telecom) to get it working. |
| 01:03.10 | [hC] | nobody here's familiar with manager? :/ |
| 01:03.44 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 01:03.59 | Omfgicbts | That's about where I am now ManxPower. I've tried Asterisk@Home and thier idea of a dialplan is really getting in the way here. |
| 01:04.33 | Omfgicbts | The worst part is I can't string 8 uninterrupted minutes together at work to figure this out. |
| 01:07.34 | ManxPower | Omfgicbts, there IS another way, BTW. |
| 01:07.56 | Omfgicbts | Another way ? |
| 01:08.01 | ManxPower | ASSUMING your PBX provides Far End Disconnect Supervision then you can do it all without Asterisk |
| 01:08.14 | ManxPower | just two SIPura boxes and some celeverness. |
| 01:08.41 | Omfgicbts | My lack of cleverness is why I'm here. |
| 01:09.23 | ManxPower | A SIPura on the far end, set up so when the user picks up the phone, it automatically "dials" the SIPura at the office, which then goes off hook and then the user gets a PBX dialtone. |
| 01:09.37 | Omfgicbts | I had originally thought about that soft of configuration. But ran into issues with the cleverness deficiiency and getting sip/rtp through the firewall. |
| 01:09.39 | ManxPower | Have the SIPura at the office call the remote SIPura whenever it gets a call from the PBX. |
| 01:10.07 | ManxPower | Omfgicbts, Oh. Add another 10 hours to my estimate if you have firewall or nat involved. |
| 01:10.15 | demigod2k | 10? more |
| 01:10.38 | ManxPower | Omfgicbts, You basically have to learn enough to set up a full scale PBX with remote extensions in order to do what you want it to do. |
| 01:10.47 | demigod2k | 10 if you're using iax. 50 if you're using sip |
| 01:10.49 | ManxPower | demigod2k, I am assuming he already knows linux, telecom, networking |
| 01:10.56 | demigod2k | hah ya that would help |
| 01:11.40 | demigod2k | just the tiniest things. I've been beating my head against the wall for an hour trying to figure out why voicemails werent saving. it turns out there's a minimum message length parameters |
| 01:11.45 | Omfgicbts | My experience with IAX has been fantastic so far. That silly looking iaxy has been able to connect to the asterisk box from every place I've tried it. |
| 01:12.37 | demigod2k | this gxp2000 is sort of lame. what a pain to configure |
| 01:14.24 | Omfgicbts | Maybe if I ask specific questions.... |
| 01:15.34 | Omfgicbts | I set up the outgoing dialing rule something like 9|5XX and then the remote user is able to dial a 3 digit extension on our PBX if they dial a 9 first. |
| 01:16.17 | Omfgicbts | When I replced that rule with 5XX I got a fast busy. |
| 01:18.15 | SplasPood | 95XX ? |
| 01:19.28 | Omfgicbts | Yea, 9|5XX . Asterisk strips off the 9, connects through an X100P to our PBX, and passes the 3 digit extension. |
| 01:19.37 | *** join/#asterisk Abydos313 (i=abydos31@ppp-71-138-7-212.dsl.irvnca.pacbell.net) |
| 01:20.45 | Omfgicbts | Life would be a LOT easier if I could just dump the existing PBX and replace it with an Asterisk solution. That's not an option right now though. |
| 01:21.00 | demigod2k | how many users in your facility? |
| 01:21.46 | Omfgicbts | We have 38 bodies on site. 3 full time remote users. Plus an assortment of conference rooms and such. |
| 01:22.15 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:22.51 | Omfgicbts | Lots of users but relativly low phone useage. I just dropped down to 8 trunks and nobody has noticed. |
| 01:24.06 | paolob | Hi guys! I'm installing asterisk, and I discovered the way to transfer calls with the # key. The question I have is: how do you get that you receive the call again if the extension you transfered to didn't answer. Any hint? thank you! |
| 01:26.36 | *** join/#asterisk demolish50 (n=bla@uslec-63-243-127-240.cust.uslec.net) |
| 01:26.40 | demolish50 | anyone around? |
| 01:27.38 | *** join/#asterisk kemp (n=10330405@221.232.225.120) |
| 01:35.52 | wunderkin | paolob, use attended transfer |
| 01:36.13 | paolob | wunderkin, what should I change? |
| 01:36.20 | wunderkin | are you using 1.2 |
| 01:38.31 | demolish50 | wow someone is actually here :) |
| 01:39.11 | demolish50 | I need help when anyone gets a free minute....:) |
| 01:41.24 | wunderkin | you would get a better response if you just ask your question |
| 01:41.44 | demolish50 | wow n/m I just figured it out |
| 01:42.03 | demolish50 | Softphone had silent surpression turned on underneath the SIP settings....that was making moh choppy |
| 01:43.11 | Omfgicbts | You just need louder music. |
| 01:43.13 | paolob | wunderkin, 1.0.9 |
| 01:43.25 | wunderkin | paolob, sucks to be you |
| 01:43.48 | paolob | wunderkin, :-( |
| 01:44.43 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 01:47.58 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 01:47.59 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 01:56.06 | sevard | anyone want to stream some *(#&$ing tv to me |
| 01:56.11 | sevard | i'm bored as hell |
| 01:58.47 | *** join/#asterisk Eggplant (i=No@dsl-126.cascadeaccess.com) |
| 02:04.41 | *** join/#asterisk iq (n=iq@71-38-73-211.omah.qwest.net) |
| 02:07.12 | sleepy_one | ~wiki |
| 02:07.43 | sleepy_one | ~docs |
| 02:07.45 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:07.46 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 02:08.02 | sleepy_one | jbot, hi |
| 02:08.03 | jbot | hello, sleepy_one |
| 02:08.11 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 02:08.25 | sleepy_one | jbot, is everyone alive? |
| 02:08.45 | Sedorox | it is quiet for the #asterisk channel |
| 02:09.16 | file | we're all out with friends, leave a message after the beep |
| 02:09.41 | Omfgicbts | That sounds so much better coming from Alison Smith. |
| 02:10.39 | alephcom | I guess I could dream up something to have a flame ware over. |
| 02:10.40 | sleepy_one | jbot, are you alive? |
| 02:10.41 | jbot | The dead cannot live |
| 02:10.52 | sleepy_one | jbot, is anyone here? |
| 02:11.19 | yxa | anyone interested in a callback bounty? please pm me |
| 02:11.55 | sleepy_one | Omfgicbts, does your nick stand for oh my %*^(%*%^*%*%($#%*#$( god I can't believe this $%*#$(%*#$(%$*%($*% ? |
| 02:12.29 | Omfgicbts | It does Sleepy_one. |
| 02:12.58 | sleepy_one | lol |
| 02:13.07 | sleepy_one | very clever and original |
| 02:13.13 | sleepy_one | :-) |
| 02:13.32 | Omfgicbts | Hmm. FreePBX is just as slow on a 3 gig P4 box as it is on an 800 meg P3. |
| 02:13.53 | Omfgicbts | I don't know how original it is. It certainly is hard to type. |
| 02:15.05 | Omfgicbts | I take that back....freepbx seems snappy now. Webmin is still dog slow though :( |
| 02:15.50 | sleepy_one | webmin isn't known for speed |
| 02:16.20 | Sedorox | webmin is a last resort tool for me |
| 02:17.08 | Omfgicbts | I'm still lame. So webmin is a requirement. |
| 02:17.58 | Omfgicbts | I wouldn't have the slightest clue how to set up a static route without it. |
| 02:18.39 | Sedorox | to network or to host? :p |
| 02:19.03 | Omfgicbts | lol |
| 02:19.48 | Omfgicbts | At least I'm up to speed on rm -rf at / |
| 02:19.49 | Sedorox | route add -host 1.2.3.4/32 gw 4.3.2.1 dev eth0 |
| 02:19.49 | Sedorox | :p |
| 02:19.53 | Sedorox | 0_o |
| 02:20.00 | Sedorox | dd if=/dev/random of=/dev/hda |
| 02:20.42 | Omfgicbts | Ahh...pretty much like the route add in DOS. |
| 02:20.42 | sleepy_one | cat /dev/zero > /dev/sda |
| 02:22.17 | Sedorox | Omfgicbts: ummm....... no.. windows/dos is a bit different :p |
| 02:22.28 | Sedorox | actually.. I think route can't do cidr... |
| 02:22.39 | Sedorox | route add -host 1.2.3.4 netmask 255.255.255.255 gw 4.3.2.1 dev eth0 |
| 02:22.42 | Sedorox | lol |
| 02:23.30 | sleepy_one | lol |
| 02:23.50 | Omfgicbts | cidr ? |
| 02:24.53 | Omfgicbts | Well, being a redneck, I understand the classless part |
| 02:25.46 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 02:26.39 | Omfgicbts | I'm not sure I see the differrence betwen that and the DOS flavor |
| 02:27.05 | websae | how are you doing? |
| 02:27.35 | sleepy_one | #:: ::-| ::-| .-. :||-:: 0-| .-| ::||-| .:|-. :||open(Q,$0);while(<Q>){if(/^#(.*)$/){for(split('-',$1)){$q=0;for(split){s/\|/:.:/xg;s/:/../g;$Q=$_?length:$_;$q+=$q?$Q:$Q*20;}print chr($q);}}}print"\n";#.: ::||-| .||-| :|||-| ::||-| ||-:: :|||-| .:| |
| 02:27.42 | websae | anyone run asterisk on VMWARE? |
| 02:28.47 | sleepy_one | you can but it's not recommended |
| 02:30.55 | *** join/#asterisk tengulre (n=root@221.11.5.180) |
| 02:32.19 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 02:35.53 | sleepy_one | websae, asterisk is sensitive to latency and probably won't work very well on a machine which is having it's resources shared between the HOST OS and the GUEST OS |
| 02:36.13 | file | vmware and Asterisk are... interesting |
| 02:36.21 | websae | have you tried? |
| 02:36.33 | kemp | can I install Asterisk on Gentoo? does anyone know this? |
| 02:36.39 | file | we tried to use it for the dev conference :D didn't work |
| 02:36.40 | websae | yes you can kemp |
| 02:36.43 | sleepy_one | kemp, of course you can |
| 02:37.21 | sleepy_one | websae, I haven't tried it, I'm sure it can be made to work but I don't know how effective it would be, what are you trying to do? |
| 02:37.40 | websae | just see how it responds to such an environment |
| 02:37.45 | websae | setup a mini peer |
| 02:38.32 | *** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
| 02:41.08 | demigod2k | can anybody recommend a good (cheap) service for recording IVR prompts? |
| 02:42.03 | Hmmhesays | Hmmhesays wacky IVR |
| 02:42.56 | websae | anyone here played around with fax to email? |
| 02:43.02 | Omfgicbts | Doesn't Digium still do the $50 messages ? |
| 02:43.02 | websae | I am curious how it works out? |
| 02:43.31 | kemp | thank you all, now I have an NMS AG4000-E1 card and an Aculab Prosody 2 Port E1/J1 Card, Can I use one of them to connect telephone network with Asterisk? |
| 02:43.43 | demigod2k | no idea, I saw it on digiums site but I figured there may be other good services |
| 02:45.17 | Omfgicbts | What's the need for "pro" recordings ? |
| 02:45.39 | kemp | or I have to buy another card from digium,haven't I? |
| 02:47.08 | demigod2k | Omfgicbts, sounds better |
| 02:48.58 | Omfgicbts | Is it just me or is Alison Smith talking about being eaten by weasles relly hot ? |
| 02:52.34 | demigod2k | ok I setup a ring group for sales and support. whats the typical way to identify those versus direct-dial to the phone on my desk |
| 02:52.39 | demigod2k | distinctive ring? something on the display? |
| 02:52.41 | sleepy_one | kemp, http://www.asterisk.org/hardware |
| 02:54.03 | *** join/#asterisk Qwell[] (n=chatzill@unaffiliated/qwell) |
| 02:54.42 | kemp | thank you, sleepy_one, I'll read them |
| 02:56.26 | sleepy_one | yw |
| 02:57.19 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 03:02.16 | Hmmhesays | drivin;n me nuts |
| 03:02.43 | sleepy_one | Hmmhesays, ? |
| 03:09.44 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 03:12.00 | *** part/#asterisk Omfgicbts (i=ray@cpe-65-189-198-222.neo.res.rr.com) |
| 03:18.29 | *** join/#asterisk inv_Arp (i=junya@adsl-10-153-159.mia.bellsouth.net) |
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| 03:19.44 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 03:21.24 | *** join/#asterisk jsaunders (n=root@216.86.121.58) |
| 03:22.02 | jsaunders | w00t, got my tdm2432e up n' runnin' |
| 03:23.49 | jsaunders | I take it it's considered bad practive to connect / unconnect stations/trunks well card is live? Or is that ok? |
| 03:23.55 | jsaunders | practice even |
| 03:25.30 | jsaunders | d- for tardy response |
| 03:26.32 | *** part/#asterisk elliot (n=elliot@rdu-nat.rpath.com) |
| 03:28.08 | demigod2k | no idea |
| 03:29.56 | *** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
| 03:32.22 | *** part/#asterisk gursikh (n=guriskh1@adsl-68-93-83-152.dsl.hstntx.swbell.net) |
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| 03:44.59 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 03:45.01 | websae | it's dead in here |
| 03:45.22 | sleepy_one | aye |
| 03:45.37 | websae | howdy |
| 03:45.49 | sleepy_one | nay actually |
| 03:46.00 | websae | sleepy: have you done any fax to email with asterisk |
| 03:46.01 | websae | ? |
| 03:46.25 | sleepy_one | sort of |
| 03:46.33 | sleepy_one | have you checked the wiki? |
| 03:46.57 | websae | i am curious your experience with it |
| 03:47.06 | websae | yeah I saw an entry in the wiki regarding it |
| 03:47.43 | sleepy_one | well I played with it but didn't get it fully working in a production environment |
| 03:48.02 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:48.12 | websae | welcome shmaltz |
| 03:48.26 | shmaltz | what does this mean: |
| 03:48.28 | shmaltz | Supervised transfer requested, but unable to find callid '90c31293-2dd9c8d8@192.168.1.100'. Both legs must reside on Asterisk box to transfer at this time. |
| 03:48.30 | shmaltz | hi websae |
| 03:49.52 | sleepy_one | are both calls running thru your * box |
| 03:49.53 | sleepy_one | ? |
| 03:50.39 | *** join/#asterisk opus_ (n=opus@dahphish.org) |
| 03:50.41 | opus_ | yo |
| 03:50.55 | *** join/#asterisk copantl (n=galellop@207.13.77.62) |
| 03:50.56 | opus_ | why is it sometimes MoH doesn't work with SIP? |
| 03:51.13 | sleepy_one | it should always work |
| 03:51.23 | sleepy_one | what kind of SIP device do you have? |
| 03:52.41 | copantl | any body know what is ASR and ACT? |
| 03:52.58 | shmaltz | sleepy_one, I'm trying to use the conf function on a SPA941 |
| 03:53.23 | _asr | i'm asr |
| 03:53.35 | *** join/#asterisk bmg505 (n=leon@dsl-146-28-218.telkomadsl.co.za) |
| 03:54.02 | sleepy_one | shmaltz, you either have to conf on the SPA941 or you have to conf using asterisk |
| 03:54.13 | *** join/#asterisk Apach (n=Apach@Sudbury-cable-69-60-231-77.unitz.ca) |
| 03:54.19 | copantl | _asr: but do you know whats means? |
| 03:54.28 | shmaltz | sleepy_one, I'm trying to conf using the SPA and thats the error I'm getting |
| 03:54.29 | _asr | it's my initials |
| 03:54.34 | sleepy_one | shmaltz, I think you should do an asterisk conference |
| 03:55.05 | Apach | Hi, where would be a good place to find information on the requirments of starting a VOIP service? |
| 03:55.14 | *** part/#asterisk file (i=jcolp@216.237.114.82) |
| 03:55.18 | Apach | I mainly just don't know who deals with the phone number registry |
| 03:55.18 | *** join/#asterisk file (i=jcolp@216.237.114.82) |
| 03:55.25 | copantl | ACT is = average call time? right? |
| 03:55.33 | shmaltz | sleepy_one, thanks for you suggestion, but it's much too complicated to do such a thing, besides the conf function should work and shouldn't give me this error |
| 03:55.47 | copantl | and ASR? |
| 03:55.49 | shmaltz | ~act |
| 03:55.50 | jbot | from memory, act is the Archive Comparison Test at http://web.act.by.net/~act/ |
| 03:56.17 | sleepy_one | wrong ACT jbot |
| 04:01.05 | *** join/#asterisk copantl (n=galellop@207.13.77.62) |
| 04:01.19 | copantl | any idea about ASR an ACT? |
| 04:01.40 | file | average success rate? |
| 04:01.45 | copantl | yes |
| 04:01.48 | copantl | file |
| 04:02.05 | copantl | can you tell me how can i found it? |
| 04:02.17 | file | find it from what? |
| 04:02.23 | copantl | from asterisk |
| 04:02.43 | file | Asterisk doesn't do anything for that... it's a PBX, you'd have to keep billing records and use them to calculate it somehow |
| 04:03.13 | copantl | is any open source software for that? |
| 04:03.18 | sleepy_one | use the mysql CDR and then write a script to calculate it |
| 04:03.23 | copantl | ok |
| 04:03.23 | file | no clue, Google |
| 04:03.47 | copantl | but means, how many calls got right termination? |
| 04:03.51 | websae | writing a billing platform can be very complicated |
| 04:09.07 | *** join/#asterisk joelsolanki (n=joelsola@202.160.163.144) |
| 04:14.54 | sleepy_one | cya l8r all |
| 04:20.28 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 04:20.40 | FuriousGeorge | hey all |
| 04:20.42 | websae | hey |
| 04:21.51 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 04:29.06 | *** join/#asterisk Stp1800 (n=Stp1800@68-235-251-98.atlsfl.adelphia.net) |
| 04:31.46 | *** part/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 04:33.55 | LostFrog | Has anyone had their broadvoice "unlimited business" account suspended for "being used for business?" |
| 04:35.54 | wunderkin | lol, maybe they clicked the wrong reason, but 'unlimited' is not unlimited, you know |
| 04:39.25 | *** join/#asterisk angom_h (n=angom@red-corp-201.143.97.166.telnor.net) |
| 04:45.31 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:46.44 | Stp1800 | Anybody ever compiledAsterisk on FreeBSD before? |
| 04:47.20 | file | yes |
| 04:50.57 | *** join/#asterisk Apach (n=Apach@Sudbury-cable-69-60-231-77.unitz.ca) |
| 04:51.58 | Stp1800 | Tried to install it from the ports tree this evening and it wouldn't compile at all. |
| 04:52.16 | Stp1800 | There has a message that said the package should not be installed because of security problems. |
| 04:52.50 | file | don't install it from ports... build it from a clean checkout of trunk |
| 04:54.48 | Stp1800 | How should I do this. |
| 04:54.58 | Stp1800 | Also this is on the amd64 version. |
| 04:55.08 | file | the instructions for checking out trunk are available on http://www.asterisk.org/ |
| 04:55.31 | Stp1800 | Also when I tried to build libpri from ports I get a libintl.so file format not supported or incorrect file format. An error like that. |
| 04:55.31 | file | and with that, I'm going to sleep |
| 04:55.43 | Stp1800 | Not really sure what that's about but seems to be a show stopper. |
| 04:56.04 | file | ports is out of date... |
| 04:57.29 | *** join/#asterisk phrog123 (n=francois@ns.menards.ca) |
| 04:58.09 | phrog123 | why does a SIP phone behind NAT shows NAT=no in SIP show peers when in fact it should be NAT=yes? |
| 04:59.23 | Corydon76-home | Have you set it to yes in sip.conf? |
| 05:01.13 | Corydon76-home | I'm guessing that would be "no" |
| 05:01.55 | phrog123 | of course it is set to NAT=yes, even tried NAT=1 |
| 05:02.25 | Corydon76-home | Have you done a 'sip reload' since then? |
| 05:02.47 | phrog123 | yes |
| 05:03.59 | phrog123 | I have this comment in A@H sip.conf: ; Note: If your SIP devices are behind a NAT and your Asterisk |
| 05:03.59 | phrog123 | ; server isn't, try adding "nat=1" to each peer definition to |
| 05:04.05 | Corydon76-home | Oh, wait. How do you know it is nat=no in "sip show peers"? |
| 05:04.14 | phrog123 | wtf is nat=1? |
| 05:04.49 | phrog123 | sip show peers shows: 205/205 205.151.208.34 D N 48376 OK (17 ms) |
| 05:04.56 | Apach | hehe, so many library files required ;d fianlly got it to compile |
| 05:05.11 | Corydon76-home | So? |
| 05:05.15 | Apach | Arrogent me doesnt like reading the instructions and had to learn the hard way I was missing like 10 required libraries |
| 05:05.32 | phrog123 | the N in there is no, and my SIP phone is behind 205.151.208.34 which does NAT |
| 05:05.42 | phrog123 | it used to show Y |
| 05:05.44 | Corydon76-home | phrog123: No, the N stands for NAT |
| 05:05.54 | Corydon76-home | phrog123: It never has shown 'Y' |
| 05:06.13 | Corydon76-home | It's blank if nat=no |
| 05:06.16 | Corydon76-home | Try it |
| 05:06.42 | phrog123 | ok, I'm dumb |
| 05:07.03 | phrog123 | would you have any idea why DTMFs are not going through at all across the firewall then? |
| 05:07.17 | Corydon76-home | Entirely separate issue |
| 05:07.25 | phrog123 | I've exhausted variantes of DTMF=inband, RFC2833, etc. |
| 05:07.37 | Corydon76-home | Your Asterisk is behind a nat? |
| 05:07.49 | phrog123 | no * is with public IP |
| 05:08.05 | Corydon76-home | Then the firewall is likely dropping DTMF packets |
| 05:08.24 | phrog123 | same problem with gxp2000 and xlite |
| 05:08.49 | Corydon76-home | Try dtmfmode=info |
| 05:09.00 | Corydon76-home | and make sure the phone config matches |
| 05:09.03 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 05:09.38 | phrog123 | tried that too - will have to try it again - would dtmf=info be easy to debug with sip debug? |
| 05:09.50 | phrog123 | easier than rtp stuff? |
| 05:10.02 | Corydon76-home | If you're typing dtmf=info, then that's what is wrong. The keyword is dtmfmode |
| 05:12.07 | Corydon76-home | Bedtime for me |
| 05:12.23 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 05:22.38 | *** join/#asterisk gursikh (n=guriskh1@adsl-68-93-83-152.dsl.hstntx.swbell.net) |
| 05:30.13 | znoG | guys, if im using g729 and dtmfmode = rfc2833 is not working (ignores the dtmf tones), as I can't use inband (not a 64k codec), and SIP info doesn't work .. I don't have much choice, do I ...? |
| 05:36.28 | *** join/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
| 05:37.01 | DoktorGreg | anyone here do multi threaded programming at all? |
| 05:38.57 | DoktorGreg | I have a thread that is communicating asynchronously with server |
| 05:39.00 | DoktorGreg | problem is |
| 05:39.13 | DoktorGreg | I cant guarantee that the packets come back in perfect order |
| 05:39.46 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 05:39.52 | DoktorGreg | So my thread... is going to spawn another thread, which is spawned suspended |
| 05:40.16 | DoktorGreg | until an event que picks up the response packet |
| 05:40.30 | DoktorGreg | then it will let the thread finish by resuming it |
| 05:40.48 | Alystair | Anyone know where I can find VOIP solution providors for small/medium businesses? |
| 05:40.54 | Alystair | *provider |
| 05:41.02 | *** join/#asterisk John-Fblt (i=alleged@freenode/staff/john-fblt) |
| 05:41.10 | Alystair | .. in Canada, sorry I didn't make it clear. |
| 05:41.12 | DoktorGreg | vonage has business accounts |
| 05:41.16 | DoktorGreg | oh |
| 05:41.23 | DoktorGreg | no idea |
| 05:41.35 | DoktorGreg | vonage has a business plan |
| 05:41.37 | DoktorGreg | 10 trunks |
| 05:41.50 | DoktorGreg | like 10000 minutes |
| 05:41.59 | DoktorGreg | 400 american a month |
| 05:42.03 | *** join/#asterisk DarthClue (n=DarthClu@adsl-69-152-249-37.dsl.snantx.swbell.net) |
| 05:43.33 | DoktorGreg | Vonage business waves canadian rates |
| 05:43.50 | *** join/#asterisk weasel00 (n=weaesl00@c-71-198-203-98.hsd1.ca.comcast.net) |
| 05:43.51 | Alystair | you mean they are par? |
| 05:48.10 | DoktorGreg | http://www.thevoipconnection.com/store/catalog/product_16220_Vonage_Business_Plus.html |
| 05:48.14 | DoktorGreg | I dont know |
| 05:48.22 | DoktorGreg | it would be a good baseline though i suspect |
| 05:49.14 | gursikh | What VOIP providers do you guys suggest for use with * ? (Price is an issue) but reliability comes first. Location is USA-NY-NYC. Both inbound and outbound carrier suggestions welcome please. |
| 05:49.35 | DoktorGreg | gursikh, how many trunks do you need? |
| 05:50.11 | gursikh | just 2 or 3 |
| 05:50.52 | gursikh | I'm looking to do one landline, 2 inbound carriers and 2 outbound. |
| 05:52.44 | DoktorGreg | http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Business |
| 05:56.45 | znoG | strangely enough, apart from the big warning that dtmfmode=inband does not work with g729, it does actually work. |
| 05:57.20 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 05:58.54 | justinu | unreliably |
| 05:59.28 | DoktorGreg | That sounds like one of those things that works until it doesnt |
| 05:59.35 | gursikh | DoktorGreg: thanks for the link |
| 05:59.59 | DoktorGreg | the voip-info site can probably anser 90% of the noobish questions |
| 06:00.24 | DoktorGreg | the only place it didnt help me was in final groking of extensions.conf |
| 06:00.40 | DoktorGreg | ok im off for my late night bike ride |
| 06:00.45 | DoktorGreg | bbiab |
| 06:00.47 | znoG | justinu: so far, out of 3 attempts, it detected the dtmf all 3 times :) i tried using info (got a "method not allowed" from the SIP server.. i wonder why they dont allow dtmf by info) and rfc2833 didn't work either |
| 06:02.12 | justinu | yeah, but don't run production traffic like that |
| 06:02.39 | znoG | you think in a big enough number of calls, dtmf would fail fairly often? |
| 06:03.02 | justinu | yep |
| 06:03.07 | justinu | often enough to piss people off |
| 06:03.15 | justinu | and guess who takes the heat... you |
| 06:03.36 | znoG | heh, yeah, but its either this or using a/u law ... with dtmf inband |
| 06:03.47 | znoG | dtmf out of band doesn't seem to work |
| 06:03.50 | justinu | it does |
| 06:04.01 | justinu | your provider isn't set up right most likely |
| 06:04.28 | znoG | yea i know it works, just not with my provider. |
| 06:04.41 | justinu | if they don't include the rtpmap 101 thing in the sdp, they're not offering it |
| 06:04.44 | znoG | and to get them to change that .. |
| 06:05.01 | justinu | ask them to deliver you dtmf via rfc2833 |
| 06:05.20 | opus_ | hey justinu |
| 06:05.22 | opus_ | whats up |
| 06:05.25 | justinu | nada |
| 06:05.33 | justinu | processing my wedding pics |
| 06:05.33 | znoG | from what i understand dtmfmode=info means when you press a dtmf, it will send it as a SIP packet indicating which digit was pressed.. right |
| 06:05.36 | znoG | ? |
| 06:05.44 | justinu | yeah, but info isn't commonly used |
| 06:05.51 | znoG | ok, how does rfc2833 work? |
| 06:06.01 | justinu | rfc2833 encodes dtmf in the rtp, but with a different codec type |
| 06:06.10 | justinu | technically out of band |
| 06:06.16 | justinu | but on the media plane |
| 06:06.17 | opus_ | znoG its a specially crafted rtp packet with press down and up and duration |
| 06:06.53 | justinu | dtmf inside info packets is on the signalling plane |
| 06:06.58 | justinu | rfc2833 is on the media plane |
| 06:07.04 | opus_ | znoG asterisk is not fully compatible with all rfc2833 hardware, and sometimes doesn't even like its own freaking rfc2833!! |
| 06:07.16 | justinu | heh, lots of double digit detection issues |
| 06:07.37 | justinu | i'm not sure what causes it, possible bad dtmf clamping |
| 06:07.39 | znoG | ah |
| 06:07.46 | znoG | i see, thanks for clearing that up |
| 06:07.53 | justinu | maybe inband 711 is better |
| 06:07.55 | opus_ | justinu, its because there is no duration in the dtmf stack |
| 06:07.58 | znoG | ok well rfc2833 is being completely ignored |
| 06:08.08 | znoG | strangely enough it seems to work OK if I use alaw/ulaw |
| 06:08.12 | znoG | just not with g729 |
| 06:08.13 | justinu | it has to be provisioned on their side too |
| 06:08.18 | justinu | for both codecs probably |
| 06:08.31 | opus_ | znoG inband dtmf (not rfc2833/ not info) won't work in g729 that well |
| 06:08.38 | justinu | some of the cisco experts probably know the exact commands :) |
| 06:08.55 | znoG | yea their side has to look for specially crafted dtmf packets |
| 06:09.03 | justinu | and send them |
| 06:09.11 | znoG | yeah they use a Cisco MediaGateway or something |
| 06:09.24 | opus_ | znoG are you having problems with rfc2833 digits? |
| 06:09.26 | znoG | opus_: yeah unfortunately until my provider gets back to me, i'm forced to use inband |
| 06:09.32 | justinu | pstn gateway must decode the inband dtmf on the TDM trunk, then convert to rfc2833 and strip out the digits from the g711 stream |
| 06:09.52 | znoG | opus_: yep basically no digits are getting through to my provider.. the only one that works (somewhat reliably, haven't tried it much) is inband |
| 06:09.57 | znoG | but, as we know, inband is not reliably. |
| 06:10.14 | opus_ | znoG does it work some of the time with rfc2833 or not at all /never? |
| 06:10.19 | znoG | not at all |
| 06:10.40 | znoG | i got annoyed at one stage and pressed the 1 button 50 times and i eventually got one through |
| 06:10.54 | opus_ | does it work if you hit it really really fast? |
| 06:11.21 | Alystair | Our router has 2 WAN connections from 2 different providers and splits the usage between them, would that break anything relating to VOIP? |
| 06:11.24 | justinu | you could turn on rtp debug and see if you see the rfc2833 packets being received |
| 06:11.54 | znoG | opus_: yeah i think if i press it fast enough it works.. i'll try again now |
| 06:13.11 | justinu | off to bed |
| 06:14.07 | gursikh | where can I find a diagram of how the basic hardware ina general * setup is supposed to look like? |
| 06:15.35 | znoG | opus_: that time it worked first go, then the second call again nothing.. |
| 06:15.36 | opus_ | gursikh what type of setup? |
| 06:15.45 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
| 06:15.49 | opus_ | znoG yeah what provider are you using? |
| 06:15.52 | John-Fblt | people still use inband signaling? |
| 06:16.31 | gursikh | I'm trying to wrap my head around how the hardware side of a setup with, small office, 4 computers, 5 phones, 1 PSTN line, 2-3 VOIP lines would look? With DSL modem, router etc. |
| 06:16.31 | znoG | opus_: im in argentina so you probably haven't heard of them :) they call themselves Call IP |
| 06:17.09 | opus_ | znoG how is argentina? :) how much is a gallon of gas right now? |
| 06:17.16 | znoG | ... a lot :) |
| 06:17.39 | znoG | at least they charge us a lot for the gas! |
| 06:18.35 | *** join/#asterisk Corydon76-home (i=yellow@pdpc/supporter/sustaining/Corydon76-home) |
| 06:18.53 | gursikh | the diagrams page on voip-info is pretty useless: http://www.voip-info.org/wiki/view/Asterisk+Implementation+Diagrams ..grr. |
| 06:19.39 | gursikh | Just trying to figure out how I need to "wire" everything up before I can hire someone else to do the * installation/setup. |
| 06:21.51 | znoG | there's no one size fits all installation gursikh |
| 06:22.21 | znoG | 5 phones... well you need to decide if you're going to use FXS ports from Digium on a TDM400 card, or use ATAs, or softphones, or ... |
| 06:22.27 | DoktorGreg | How many computers? |
| 06:22.30 | znoG | 1 PSTN line = Digium X100P card |
| 06:22.39 | znoG | or it could be one of the TDM400 modules |
| 06:22.39 | gursikh | SIP phones |
| 06:22.39 | DoktorGreg | doe! |
| 06:22.45 | DoktorGreg | dont get the x100p |
| 06:22.55 | DoktorGreg | go with the tdm400 for your fxo |
| 06:23.10 | DoktorGreg | even tho it costs more |
| 06:23.15 | gursikh | man, half the people tell me get an x100p from ebay, half tell me not to. I'm losing my mind! |
| 06:23.42 | znoG | x100p from ebay is not the best solution.. you may have caller ID issues, volume issues, etc. |
| 06:24.11 | znoG | as greg said, the tdm400 is best but its more expensive, hence why i suggested the x100p |
| 06:24.14 | znoG | Digium |
| 06:25.16 | DoktorGreg | the 300 you spend the the tdm400 will be well worth it |
| 06:25.24 | DoktorGreg | and if you ever need to upgrade |
| 06:25.29 | gursikh | I want this: 1 landline, 2-3 voip lines, DSL Modem, router, 4 wired computers+wireless network. Dell box with some card running asterisk. 5 SIP phones. I think that's about it. |
| 06:25.38 | DoktorGreg | all you have to do is pop an extra port in |
| 06:25.54 | gursikh | I think I like the TDM400's anyway for the modularity |
| 06:26.00 | DoktorGreg | ok, you will need a switch |
| 06:26.09 | DoktorGreg | you need enough ports on your switch |
| 06:26.18 | DoktorGreg | for each computer + each phone |
| 06:26.28 | gursikh | ok |
| 06:26.41 | DoktorGreg | next |
| 06:27.00 | DoktorGreg | install that switch in a closet where the phones come in the building |
| 06:27.11 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:27.22 | DoktorGreg | next pull a cat5e cable to every spot where you will have a computer |
| 06:27.27 | DoktorGreg | or phone |
| 06:27.35 | DoktorGreg | also |
| 06:27.40 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 06:27.46 | DoktorGreg | just go ahead and get a switch that had POE nowdays |
| 06:28.15 | gursikh | Power over etherenet? |
| 06:28.20 | DoktorGreg | yah |
| 06:28.30 | gursikh | Ok got it |
| 06:28.31 | DoktorGreg | that way you dont need a powersupply for your phones |
| 06:28.38 | DoktorGreg | in event of power outage |
| 06:29.18 | DoktorGreg | oh make a drawing of this whole wire thing, even if it is a primitive drawing |
| 06:29.19 | DoktorGreg | it will save big time later |
| 06:30.11 | DoktorGreg | ok, you know how to terminate simple straight through cat5e? |
| 06:30.34 | gursikh | that's what i'm trying to do now, beforehand, so i know what to purchase, and how it's going to go down, as I have limited time to install this (volunteer for a non-profit based in NYC, i live in TX. Will visit for 9 days to get this implemented) |
| 06:30.36 | gursikh | Yes |
| 06:31.05 | DoktorGreg | ok, so you have your cables in place |
| 06:31.08 | DoktorGreg | terminated |
| 06:31.12 | DoktorGreg | in the closet |
| 06:31.21 | gursikh | right. |
| 06:31.26 | DoktorGreg | next plug the switch into the largest UPS you can afford |
| 06:31.33 | gursikh | ok |
| 06:31.35 | DoktorGreg | its POE right? |
| 06:31.47 | DoktorGreg | ok with your load |
| 06:31.50 | gursikh | Sounds good to me. |
| 06:32.07 | gursikh | (if not, I assume the UPS would be pointless here) |
| 06:32.11 | DoktorGreg | your computer should be purchased from salvation army |
| 06:32.20 | gursikh | well we have the computer already |
| 06:32.28 | gursikh | I will "repurpose" it for * |
| 06:32.46 | DoktorGreg | ok, will you ever in your wildest dreams upgrade to a pri line for phones? |
| 06:33.02 | Alystair | what's a pri line? |
| 06:33.17 | DoktorGreg | if not |
| 06:33.19 | gursikh | No, that is probably not even possible at the location AFAIK |
| 06:33.23 | DoktorGreg | ok |
| 06:33.30 | DoktorGreg | so you want the rapid asterisk distro |
| 06:33.57 | DoktorGreg | http://xorcom.com/rapid/ |
| 06:34.04 | DoktorGreg | go download that and burn the cd |
| 06:34.06 | znoG | gursikh: hah, that'd be pretty cool to be sent somewhere to implement what you're doing, specially if you're paying someone to do it! 9 days will be more than enough. |
| 06:34.42 | znoG | unless that includes shopping, delivery, install time |
| 06:34.59 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-141.claranet.co.uk) |
| 06:35.00 | DoktorGreg | Day one pull 3/4s wire |
| 06:35.06 | DoktorGreg | Day two finish pulling wire |
| 06:35.08 | gursikh | I'm not being sent, I'm volunteering my time to go up there and help them up, the 9 days will not just be * setup, I will redo the entire network, setup a million other things, spend time in the city :-) , setup * and go crazy |
| 06:35.29 | gursikh | I will have some help with the wiring, so thats fun. |
| 06:35.41 | DoktorGreg | help makes pulling wire a lot easier |
| 06:35.56 | gursikh | Actually, last time i was there I pulled a lot of wiring, so 1/2 of it is done already by me :-) |
| 06:36.13 | Alystair | cable tester + walkitalki = necessary |
| 06:36.24 | DoktorGreg | meh |
| 06:36.26 | gursikh | It's a small office. talking loudly suffices :-P |
| 06:36.27 | asterboy | I'm always pulling my wire. |
| 06:36.43 | Alystair | asterboy: take your hand out of your pants please ;) |
| 06:36.44 | gursikh | wait, where does the router, modem, and * card go? |
| 06:36.45 | DoktorGreg | I never use my cable tester |
| 06:36.47 | asterboy | lol |
| 06:36.55 | asterboy | Beats pulling somebody elses. |
| 06:37.01 | Alystair | hah |
| 06:37.15 | DoktorGreg | we will get to that |
| 06:37.21 | asterboy | Seriously though, I do have some wiring to do. |
| 06:37.28 | DoktorGreg | did you see the software I linked to? |
| 06:37.35 | gursikh | uhm.. no |
| 06:37.43 | DoktorGreg | http://xorcom.com/rapid/ |
| 06:37.48 | asterboy | Typical customer does not want to put money into wires...just duct tape them down the wall. |
| 06:37.54 | DoktorGreg | for no nonsense install, thats what you want |
| 06:37.57 | *** join/#asterisk Hali_303 (n=surfk@dsl51B6E6BC.pool.t-online.hu) |
| 06:38.10 | DoktorGreg | if you are never gonna use PRI |
| 06:38.27 | gursikh | well, actually I will want to Hire someone to do the installation for me. I don't want to even begin with messing with it :-P |
| 06:38.43 | asterboy | that xorcom looks great |
| 06:38.43 | gursikh | (from my own pocket) |
| 06:38.51 | DoktorGreg | all you are gonna do in put that cd in computer and follow prompts |
| 06:39.08 | DoktorGreg | takes about an hour to get asterisk up and running |
| 06:39.15 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 06:39.19 | asterboy | I could get it done fast with dd |
| 06:39.45 | gursikh | Ok, i'll look into it now. I'll DL it and put it on a computer here to mess around with. |
| 06:39.58 | *** join/#asterisk lorinc (n=ang@caracas-1084.adsl.interware.hu) |
| 06:40.09 | DoktorGreg | that cd makes whatever computer you put it in into an asterisk box |
| 06:40.16 | DoktorGreg | automagically |
| 06:40.20 | gursikh | ic, same like A@H |
| 06:40.24 | asterboy | ok hold on...asterisk v 1.0.9? |
| 06:40.29 | DoktorGreg | yah |
| 06:40.40 | DoktorGreg | no frills |
| 06:40.41 | asterboy | forget that |
| 06:40.54 | DoktorGreg | lol |
| 06:41.07 | asterboy | I want 1.3! |
| 06:41.29 | DoktorGreg | build it from cvs! |
| 06:41.39 | gursikh | but i've had A@H described to me by various people as being along the lines of "frontpage for asterisk* outputs crappiest code ever. |
| 06:41.54 | Alystair | wow, that bad? |
| 06:41.57 | DoktorGreg | xorcom doesnt even output any code |
| 06:41.57 | gursikh | and nobody likes crappy code. |
| 06:42.04 | DoktorGreg | it is already set up |
| 06:42.28 | DoktorGreg | all you are gonna do is point your sip phones on the lan at it |
| 06:42.32 | asterboy | nice web page though |
| 06:42.52 | gursikh | Ok |
| 06:42.57 | DoktorGreg | and it will all work |
| 06:43.10 | *** join/#asterisk weasel00 (n=weaesl00@c-71-198-203-98.hsd1.ca.comcast.net) |
| 06:43.24 | DoktorGreg | thats one option, so far that is the easiest i have seen |
| 06:43.24 | asterboy | wouldn't be too hard to upgrade. |
| 06:43.31 | asterboy | might end up playing with it. |
| 06:43.51 | DoktorGreg | I found the rapid distro to be an excellent learning volume |
| 06:44.03 | DoktorGreg | its nice to tinker with an already working thing |
| 06:44.07 | asterboy | and learning value |
| 06:44.22 | Alystair | DoktorGreg did you just discribe the whole setup of a standard Asterisk install? |
| 06:44.26 | gursikh | 60% dled already, should be burning it and popping it soon. |
| 06:44.29 | asterboy | ya, that would be nice to see what a supposed proper config is like. |
| 06:44.40 | gursikh | Alystair: I think he's getting there |
| 06:44.48 | gursikh | I'm just taking notes Right now. |
| 06:44.49 | DoktorGreg | gursikh, is erases whatever computer you put it on |
| 06:44.55 | gursikh | yah i c that |
| 06:45.07 | DoktorGreg | just checking |
| 06:45.46 | DoktorGreg | now this computer, goes in the closet, and gets plugged into the switch |
| 06:46.04 | DoktorGreg | the phoen line, whatever fxo card you choose |
| 06:46.19 | DoktorGreg | and the big UPS you have installed |
| 06:46.26 | gursikh | ok, what port? do you plug the computer into? WAN? |
| 06:46.32 | DoktorGreg | no |
| 06:46.43 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 06:47.01 | gursikh | just another port. |
| 06:47.28 | DoktorGreg | ethernet port goes into switch |
| 06:47.36 | gursikh | ok |
| 06:47.42 | DoktorGreg | fxo port goes into pots line |
| 06:47.53 | gursikh | ok |
| 06:48.08 | DoktorGreg | one more device to stuff in closet |
| 06:48.20 | X-Gen | yo freaks |
| 06:48.30 | DoktorGreg | basic linksys router |
| 06:48.52 | gursikh | right. More like corner where the luddites currently pile all wires and devices into but yeah "closet" :-P |
| 06:48.56 | DoktorGreg | dsl modem plugs into that, then one of the ports in router plug into switch |
| 06:49.02 | gursikh | ok |
| 06:49.24 | DoktorGreg | set every device on your lan to dchp config |
| 06:49.29 | gursikh | ok |
| 06:49.31 | DoktorGreg | asterisk will already be configured that way |
| 06:49.41 | gursikh | got it. |
| 06:49.57 | Alystair | DoktorGreg: POTS = standard phone line hooked up with phone company right? |
| 06:50.01 | DoktorGreg | yup |
| 06:50.28 | Alystair | FXO port is not used for anything but 911 or if the power goes out? |
| 06:50.38 | DoktorGreg | yup |
| 06:50.49 | DoktorGreg | no, we have everything on UPS |
| 06:50.57 | DoktorGreg | internet should still work if power goes out |
| 06:51.01 | gursikh | and in ourcase accepting collect calls. |
| 06:51.11 | Alystair | alright |
| 06:51.21 | LostFrog | assuming your ISP is up. |
| 06:51.36 | DoktorGreg | if your ISP doesnt survive power outage |
| 06:51.41 | DoktorGreg | get a better ISP |
| 06:51.52 | DoktorGreg | the only exception i can think of |
| 06:52.01 | Alystair | what happens if it doesn't, does the system revert to FXO automagically? |
| 06:52.05 | DoktorGreg | is that massive multi state power outage we had a few years back |
| 06:52.09 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 06:52.22 | DoktorGreg | no that is configured in the extensions.conf file |
| 06:52.38 | gursikh | that's what cell phones, walki-talkies , nextel DC are for |
| 06:52.44 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 06:52.48 | DoktorGreg | in gursikh's case |
| 06:53.01 | DoktorGreg | he will need to add about 5 lines of code to extensions.conf |
| 06:53.08 | DoktorGreg | and un comment some things in sip.conf |
| 06:53.27 | DoktorGreg | to make everything route correctly |
| 06:53.44 | gursikh | sounds good, I'm not one afraid of nano or vi . |
| 06:53.50 | Alystair | hey DoktorGreg, wanna do a project with me? |
| 06:54.03 | Alystair | You create a neat tutorial and I add graphics/charts/etc |
| 06:54.10 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 06:54.11 | DoktorGreg | kk |
| 06:54.24 | gursikh | Ok DoktorGreg: Is that about it in terms of hardware setup? |
| 06:54.29 | DoktorGreg | that is all of it |
| 06:54.35 | DoktorGreg | did you draw a picture? |
| 06:54.42 | gursikh | yes, i did. |
| 06:54.49 | DoktorGreg | so you grok it now? |
| 06:54.55 | gursikh | most definetly |
| 06:54.57 | Alystair | http://img354.imageshack.us/img354/7613/cagsimplenetwork0sz.png <-- our office network |
| 06:55.00 | DoktorGreg | kewl |
| 06:55.06 | gursikh | I thank you for your most gracious help |
| 06:55.11 | gursikh | and patience |
| 06:55.25 | DoktorGreg | if you run into any snags just hop on here and look for me |
| 06:55.30 | Alystair | DoktorGreg should be named VOIP_Buddha |
| 06:55.35 | gursikh | Modem-->router-->Switch |
| 06:55.35 | gursikh | Switch --->computers/phones |
| 06:55.35 | gursikh | Computer-->Switch |
| 06:55.35 | gursikh | POTS line--->FXO port |
| 06:55.41 | gursikh | look about right? |
| 06:55.51 | DoktorGreg | yup |
| 06:55.56 | DoktorGreg | and really |
| 06:56.05 | DoktorGreg | get the TDM400 even though its more expensive |
| 06:56.15 | gursikh | second COmputer= * server |
| 06:56.38 | DoktorGreg | I wouldnt put anything on the asterisk server but for * |
| 06:56.44 | DoktorGreg | but thats me |
| 06:57.07 | DoktorGreg | and all the telco equipment goes on the largest UPS you can find |
| 06:57.09 | gursikh | I am. Because we might have the potential of good growth and getting a few more emloyees in the office, so we'd need it eventually anyways |
| 06:57.23 | Alystair | DoktorGreg: what about as a VM machine? |
| 06:57.25 | DoktorGreg | 1100va or bigger |
| 06:57.35 | gursikh | I'm not planning on putting anything onthe machine but * |
| 06:57.46 | DoktorGreg | oh one final thing |
| 06:58.08 | DoktorGreg | you would do well to get a telco company grade lightning supressor |
| 06:58.26 | DoktorGreg | it will pay for its self on first big surge |
| 06:58.38 | gursikh | how much do they run? |
| 06:58.46 | DoktorGreg | i dont remember off hand |
| 06:58.52 | DoktorGreg | I didnt have one once |
| 06:58.59 | DoktorGreg | and lost EVERY COMPUTER IN OFFICE |
| 06:59.12 | DoktorGreg | from single computer connected to modem |
| 06:59.28 | Alystair | wow |
| 06:59.28 | DoktorGreg | that is rare though |
| 06:59.43 | DoktorGreg | it was a new system too |
| 06:59.46 | DoktorGreg | like two months old |
| 06:59.53 | Alystair | modem = 56k onboard modem or the main office one? |
| 07:00.04 | DoktorGreg | yah, this was 95 or so |
| 07:00.09 | gursikh | something like: http://www.overcost.com/cat.asp?c=3&n=Computer+Hardware,+Supplies+%26+Data+Storage&s=12&sn=Computer&ss=109&ssn=Surge+Protectors&g=11797&gn=Isobar%C2%AE+and+Isobar%C2%AE+Ultra+Premium+Surge+Suppressor&referer=ff_TRPISOBAR6&utm_source=froogle&utm_medium=SE&utm_campaign=froogle |
| 07:00.22 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 07:00.53 | gursikh | ? |
| 07:01.16 | gursikh | or something like: http://www.buy.com/retail/product.asp?sku=10243414&SearchEngine=Froogle&SearchTerm=10243414&Type=PE&Category=Comp&Gad=0&dcaid=17379 ? |
| 07:02.27 | Alystair | heh |
| 07:02.44 | gursikh | lol |
| 07:03.15 | gursikh | I'm so tired, it's 2am here, been working all day, while trying to grok all this phone business |
| 07:04.03 | LostFrog | buy.com won't show me that item. |
| 07:04.33 | gursikh | http://froogle.google.com/froogle_cluster?q=lightning+suppressor+telephone&pid=2411200859355976277&oid=16229992021911069401&btnG=Search+Froogle&scoring=mrd&hl=en |
| 07:04.41 | gursikh | maybe that link? |
| 07:04.55 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 07:05.06 | DoktorGreg | my computer upgraded its self |
| 07:05.11 | DoktorGreg | er patched |
| 07:05.14 | DoktorGreg | then rebooted |
| 07:05.20 | gursikh | exciting |
| 07:05.24 | DoktorGreg | what was i blathering about? |
| 07:05.28 | gursikh | UPS |
| 07:05.33 | DoktorGreg | oh yah |
| 07:05.38 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-141.claranet.co.uk) |
| 07:05.40 | DoktorGreg | yah |
| 07:05.47 | DoktorGreg | network |
| 07:05.50 | gursikh | I was saying: something like this: http://froogle.google.com/froogle_cluster?q=lightning+suppressor+telephone&pid=2411200859355976277&oid=16229992021911069401&btnG=Search+Froogle&scoring=mrd&hl=en or http://www.overcost.com/cat.asp?c=3&n=Computer+Hardware,+Supplies+%26+Data+Storage&s=12&sn=Computer&ss=109&ssn=Surge+Protectors&g=11797&gn=Isobar%C2%AE+and+Isobar%C2%AE+Ultra+Premium+Surge+Suppressor&referer=ff_TRPISOBAR6&utm_sourc |
| 07:06.18 | gursikh | I wouldn't mind some of those Alystair |
| 07:06.23 | DoktorGreg | make sure they are actually grounded |
| 07:06.32 | LostFrog | If my phone lines weren't brought in on cable, I would buy a couple of those. |
| 07:07.08 | gursikh | the first or second link? |
| 07:07.19 | Alystair | what's the difference between analog and digital interface cards? |
| 07:07.32 | DoktorGreg | digital interface cards are for t1 e1 |
| 07:07.32 | gursikh | pretty cheap, that first one. |
| 07:07.34 | LostFrog | I am so pissed off at broadvoice.. And I can't call them to vent. |
| 07:07.57 | LostFrog | APC ProtectNet PDigiTel |
| 07:08.16 | gursikh | yah 20 bucks is nice |
| 07:09.47 | gursikh | and i guess i would stick it between modem and router |
| 07:09.48 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 07:09.56 | DoktorGreg | http://www.blackbox.com/Catalog/Category.aspx?cid=45,75,1426 |
| 07:10.02 | DoktorGreg | buy some stuff from that page |
| 07:10.03 | Alystair | DoktorGreg: so unless I'm using T1/ISDN lines they are useless right? |
| 07:10.11 | DoktorGreg | yup |
| 07:10.22 | Alystair | awesome |
| 07:10.34 | DoktorGreg | unless you wanted to connect asterick to a key system |
| 07:10.50 | DoktorGreg | and the key system had a pri interface on it |
| 07:11.08 | DoktorGreg | but yah |
| 07:11.18 | DoktorGreg | t1 e1 is IMO legacy tech |
| 07:11.42 | DoktorGreg | a tiny little 1.5Mbs connection |
| 07:11.47 | Alystair | yep. |
| 07:12.17 | gursikh | sucks though, but I have MUCH MUCH faster connecion here at home that anywhere else. |
| 07:12.19 | DoktorGreg | nowdays with cable modems delivering 1.5Mbs up and 8Mbs down |
| 07:12.25 | gursikh | (payed by company) |
| 07:12.46 | DoktorGreg | and everybody had a gigabit network card in their computer... |
| 07:12.54 | DoktorGreg | t1 seems quaint |
| 07:14.36 | DoktorGreg | actually |
| 07:14.39 | DoktorGreg | I was working on it |
| 07:14.50 | gursikh | ? |
| 07:14.55 | DoktorGreg | and i am contemplating selling my nortel mics system on ebay |
| 07:15.04 | DoktorGreg | upgrading the whole phone network to voip |
| 07:15.20 | DoktorGreg | and pocketing the cash i have left over |
| 07:15.55 | DoktorGreg | for a business anecdote for what voip is doing to the phone business |
| 07:16.21 | gursikh | well fellas, I really appreciate the time, and help, but I must be off to bed as it is quite late and i'm quite tired. You will see me around though for the next few weeks. Good night. |
| 07:18.58 | Alystair | ok now I gotta recap everything DoktorGreg said so it makes sense to me |
| 07:19.39 | DoktorGreg | the details on how to build a 30k phone system for 2k are too good to pass up? |
| 07:22.13 | Alystair | heh |
| 07:22.32 | Alystair | well our company is just starting up and I don't want to get stuck in a bad solution off the ground :) |
| 07:22.58 | DoktorGreg | well if you are just starting |
| 07:23.04 | DoktorGreg | if you want really cheap |
| 07:23.12 | DoktorGreg | get xlite voice clients |
| 07:23.14 | Alystair | I want expandablility |
| 07:23.18 | DoktorGreg | and usb headsets |
| 07:23.25 | LostFrog | What is a good provider for outgoing calls other than teliax? I'm looking for a Long Island, NY exchange. |
| 07:23.46 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 07:23.54 | luke-jr_ | LostFrog: I wouldn't necessarily consider teliax one... but |
| 07:24.05 | luke-jr_ | LostFrog: Voipjet has worked great for me for a year or so |
| 07:24.19 | luke-jr_ | Not sure where their exchange is |
| 07:25.09 | DoktorGreg | Alystair, the beauty of putting the phone system on commodity hardware is |
| 07:25.18 | DoktorGreg | you no longer need expandability |
| 07:25.29 | DoktorGreg | when you want to expand in a couple of years |
| 07:25.41 | DoktorGreg | you head down to salvation army store |
| 07:25.55 | DoktorGreg | and get a nice 2 year old computer |
| 07:26.07 | DoktorGreg | for $100 |
| 07:26.19 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:26.19 | Alystair | here's the thing |
| 07:26.28 | Alystair | we have a dual xeon server at the office which is barely doing anything |
| 07:26.42 | Alystair | can't I just run it as a VM? |
| 07:26.57 | DoktorGreg | I dont know if its gonna see the hardware |
| 07:27.05 | Alystair | true |
| 07:27.15 | luke-jr_ | why a VM? |
| 07:27.25 | DoktorGreg | yah if you have a dual xeon |
| 07:27.34 | DoktorGreg | just install debian, asterisk + samba |
| 07:27.46 | luke-jr_ | pfft, who needs samba |
| 07:27.54 | Alystair | ok, here, I want to tell you what we have already |
| 07:28.12 | Alystair | or you could just look at my networking diagram which I pasted previously |
| 07:28.28 | Alystair | http://img354.imageshack.us/img354/7613/cagsimplenetwork0sz.png |
| 07:29.05 | DoktorGreg | mmmmk |
| 07:29.06 | Alystair | the router's a hotbrick LB2, so we have 2 incoming wan ports that it load balances + 4 lan segments which can be totally seperated |
| 07:29.23 | DoktorGreg | nice router |
| 07:29.44 | Alystair | killer price two, the canadian distributer happened to live a block away from my house! |
| 07:29.47 | Alystair | *too |
| 07:29.54 | DoktorGreg | so ditch your phone lines |
| 07:29.55 | Alystair | I got it for about $200 :) |
| 07:30.06 | Alystair | Anyway, everythings already hooked up to a nice UPS so that's not an issue |
| 07:30.09 | DoktorGreg | install asterisk on the dell |
| 07:30.35 | DoktorGreg | put in a $300 poe switch |
| 07:30.44 | DoktorGreg | and start installing phones |
| 07:30.46 | Alystair | alright |
| 07:31.29 | rkr245 | DoctotGreg:can you tell me how to connect asterisk with SER |
| 07:31.40 | *** part/#asterisk luckyduck (i=lucky@gentoo/developer/luckyduck) |
| 07:32.08 | DoktorGreg | i dont know what SER is |
| 07:32.24 | rkr245 | sip express router |
| 07:32.33 | luke-jr_ | competition =p |
| 07:32.35 | DoktorGreg | oh i have no idea |
| 07:32.44 | rkr245 | o..k thank you |
| 07:32.46 | luke-jr_ | rkr245: you need to use SIP, I think |
| 07:32.52 | rkr245 | yes |
| 07:33.07 | rkr245 | luke-jr_:can you help me? |
| 07:33.12 | DoktorGreg | well that and the internets are promising to not support QOS |
| 07:33.31 | luke-jr_ | rkr245: I can tell you it needs to use SIP :) |
| 07:33.58 | DoktorGreg | again... |
| 07:34.00 | luke-jr_ | DoktorGreg: dunno... Google's parellel internet might |
| 07:34.28 | DoktorGreg | I havent had ANY issue sharing my pri over commodity dsl and cable connections |
| 07:34.44 | rkr245 | luke-jr_: SER uses sip but i cannot connect this ser to my asterisk |
| 07:35.09 | rkr245 | iam using this ser as sip proxy |
| 07:35.29 | Alystair | DoktorGreg: so we want 7 extensions (phones), 1 POTS and 3-4 VOIP lines... |
| 07:35.52 | luke-jr_ | Alystair: VoIP "lines" are for residential users |
| 07:35.55 | Alystair | so POE Switch and some kind of server are accounted for, what else do I need |
| 07:36.01 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 07:36.14 | DoktorGreg | are you using terminal services on that server? |
| 07:36.35 | luke-jr_ | Alystair: why any POTS? why a limited number of lines? |
| 07:36.54 | DoktorGreg | POTS for 911 |
| 07:37.05 | luke-jr_ | 911 is overrated |
| 07:37.13 | DoktorGreg | well i agree |
| 07:37.14 | luke-jr_ | you can do that with your dialplan |
| 07:37.18 | *** join/#asterisk xphreak (n=zsolti@ns1.zrlocal.net) |
| 07:37.39 | Alystair | someone would have a cellphone I guess |
| 07:37.42 | xphreak | hi |
| 07:37.44 | luke-jr_ | and chances are your POTS won't have E911, IIRC, so no benefit there |
| 07:37.49 | Alystair | alright |
| 07:37.51 | xphreak | enyone installed zapted on FC5 ? |
| 07:37.57 | luke-jr_ | Alystair: just route '911' to its normal 10-digit # |
| 07:37.57 | Alystair | so no POTS |
| 07:38.10 | xphreak | I get and error all the time |
| 07:38.27 | xphreak | zaptel version 1.2.5 |
| 07:38.33 | Alystair | ok, so I still need some sort of wierd cards for my server right |
| 07:38.50 | Alystair | FXS/FXO? |
| 07:38.54 | luke-jr_ | Alystair: what you need is 7 extensions, termination service, and X (?) DIDs |
| 07:39.18 | luke-jr_ | Alystair: you only need a FXS (?) card if you plan to have all the phone lines go to the server as analog |
| 07:39.31 | Alystair | ok |
| 07:39.32 | DoktorGreg | yah, no need for fxs |
| 07:39.47 | Alystair | so I don't need a Digium card at all |
| 07:39.49 | luke-jr_ | Alystair: if you want to use Ethernet cables and network switches for phone data, you don't need any cards, just ATAs |
| 07:39.53 | DoktorGreg | and once you put in ata analog phones cost just as much as voip phones... |
| 07:40.05 | Alystair | ATA? |
| 07:40.15 | Alystair | DIDs? |
| 07:40.16 | DoktorGreg | Analog Terminal adaptor IIRC |
| 07:40.23 | luke-jr_ | ATA is a small box that does analog to IP protocol |
| 07:40.26 | Alystair | why is there no acronym bot :[ |
| 07:40.27 | luke-jr_ | err, to IP anyway |
| 07:40.29 | DoktorGreg | DID = phone number |
| 07:40.30 | xphreak | people anyone installed zaptel on fedora core 5 ???? successfully ???? |
| 07:40.32 | xphreak | :-D |
| 07:40.32 | luke-jr_ | DID is a phone # |
| 07:40.37 | Alystair | alright |
| 07:40.41 | luke-jr_ | xphreak: I would assume so |
| 07:40.52 | xphreak | need help |
| 07:40.58 | xphreak | I'm having troubles real time |
| 07:41.09 | xphreak | have read the instructions at asterisk documentation site |
| 07:41.15 | xphreak | but not helped a lot |
| 07:41.22 | luke-jr_ | xphreak: try voip-info.org |
| 07:41.26 | tzafrir | ~ata |
| 07:41.27 | jbot | rumour has it, ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
| 07:41.35 | xphreak | have already done so |
| 07:41.48 | xphreak | the same info as in asterisk doc |
| 07:42.00 | tzafrir | next time, /msg jbot ata |
| 07:42.03 | Greek-Boy | who knows some cheap and high-quality wholesale voip providers? I've been checking out five9snetwork.com. dlynes recommended it but i want to look at more |
| 07:42.06 | luke-jr_ | xphreak: so use Debian or Gentoo or something |
| 07:42.21 | Alystair | ATA is FXS then. |
| 07:42.23 | xphreak | that's not a solution at all |
| 07:42.29 | DoktorGreg | Dont stress about termination too much!!!! |
| 07:42.33 | Alystair | heh |
| 07:42.36 | luke-jr_ | Greek-Boy: SellVoIP seems to work decent, though poor customer service response time |
| 07:42.38 | xphreak | I have asterisk with zaptel on fc4 |
| 07:42.40 | DoktorGreg | analog phone system will be bankrupt by 2010 |
| 07:42.42 | luke-jr_ | Greek-Boy: VoipJet is good for termination |
| 07:42.48 | xphreak | I want to make it work on fc5 as well |
| 07:43.01 | xphreak | luke-jr_: have you installed it |
| 07:43.03 | xphreak | ? |
| 07:43.06 | luke-jr_ | Alystair: it's FXS plus an embedded system to take care of the IP |
| 07:43.13 | luke-jr_ | xphreak: I've installed it in Gentoo |
| 07:43.19 | xphreak | :-) |
| 07:43.21 | Alystair | alright, but that's only if I want normal phones hooked into the system rigth |
| 07:43.24 | Alystair | *right |
| 07:43.26 | luke-jr_ | right |
| 07:43.28 | DoktorGreg | right |
| 07:43.37 | DoktorGreg | but you want fancy office phones! |
| 07:43.41 | luke-jr_ | if everything's IP phone (hard or soft), you just plug them into the ethernet |
| 07:43.48 | Alystair | ok so |
| 07:43.53 | xphreak | will try it little later maybe someone will be online who have installed it on fc5 |
| 07:44.05 | xphreak | greets |
| 07:44.12 | DoktorGreg | sipura 942;s |
| 07:44.44 | DoktorGreg | the main feature is!!!! |
| 07:44.45 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:44.45 | Alystair | IP phone -> POE Switch -> Asterisk box -> *magic* -> WAN Router -> internet |
| 07:44.48 | DoktorGreg | custom ring tones!!! |
| 07:44.50 | *** part/#asterisk xphreak (n=zsolti@ns1.zrlocal.net) |
| 07:44.51 | DoktorGreg | muhahaha! |
| 07:44.56 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-99-210.telkomadsl.co.za) |
| 07:44.59 | luke-jr_ | wtf is a POE Swtich? |
| 07:45.06 | DoktorGreg | Power over ethernet |
| 07:45.06 | Alystair | Power of Ethernet |
| 07:45.13 | luke-jr_ | oh |
| 07:45.19 | Alystair | to aid my ethernet connected shaver :) |
| 07:45.20 | luke-jr_ | do many IP phones support PoE? |
| 07:45.24 | DoktorGreg | that way you put the switch on UPS |
| 07:45.26 | *** join/#asterisk inv_arp[work] (i=junya@adsl-10-153-159.mia.bellsouth.net) |
| 07:45.31 | DoktorGreg | and install poe phones |
| 07:45.41 | DoktorGreg | then you dont need batteries for the phones |
| 07:45.55 | DoktorGreg | it is easily the least expensive way to do it |
| 07:45.59 | Alystair | So what's wrong with my amazing diagram? |
| 07:46.04 | Alystair | IP phone -> POE Switch -> Asterisk box -> *magic* -> WAN Router -> internet |
| 07:46.07 | DoktorGreg | nothing wrong |
| 07:46.12 | luke-jr_ | it's missing the other end |
| 07:46.15 | Alystair | what happens at the *magic* location |
| 07:46.21 | luke-jr_ | nothing |
| 07:46.26 | Alystair | uh |
| 07:46.27 | DoktorGreg | take the magic part out |
| 07:46.38 | Alystair | how do we connect a local phone # then? |
| 07:46.44 | Alystair | I mean if we want inbound calling |
| 07:46.53 | luke-jr_ | that's the other end |
| 07:47.03 | Alystair | from you know, silly people with normal phones |
| 07:47.13 | DoktorGreg | just have your voip provider give you some dids |
| 07:47.17 | luke-jr_ | POTS -> your DID -> origination provider -> internet -> asterisk box |
| 07:47.32 | Alystair | ! |
| 07:47.38 | Alystair | so the provider does all that? |
| 07:47.41 | DoktorGreg | yah |
| 07:47.42 | luke-jr_ | ... |
| 07:47.45 | Alystair | WOW |
| 07:47.50 | luke-jr_ | origination provider does |
| 07:47.53 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
| 07:47.56 | luke-jr_ | termination provider does the opposite |
| 07:47.58 | DoktorGreg | he groks itm he groks it! |
| 07:48.10 | luke-jr_ | asterisk box -> termination provider -> POTS -> destination phone |
| 07:48.19 | Greek-Boy | thanks for the recommendations luke-jr_ |
| 07:48.19 | DoktorGreg | find a nice easy voip provider |
| 07:48.42 | luke-jr_ | DoktorGreg: "nice easy" usually means bad news, IMO |
| 07:48.44 | Alystair | why did this sound so damn complicated the first time I was in here asking for help |
| 07:48.59 | luke-jr_ | "nice easy" means no Asterisk support, and paired origination/termination generally |
| 07:49.10 | luke-jr_ | Alystair: maybe Digium employees were answering |
| 07:49.14 | Alystair | this guy took me under his wing and told me scary things involving standard hpones and what not |
| 07:49.16 | DoktorGreg | actually, even Vonage Business plus supports asterisk now... |
| 07:49.43 | luke-jr_ | DoktorGreg: I'd avoid Vonage at all costs |
| 07:49.51 | DoktorGreg | why? |
| 07:49.56 | luke-jr_ | they suck |
| 07:50.04 | DoktorGreg | ive had no problem... |
| 07:50.07 | Alystair | wow so in theory I CAN run Asterix on the VM machine as long as it has it's own independant LAN card |
| 07:50.15 | luke-jr_ | mandatory locked ATAs that only work with their service |
| 07:50.25 | luke-jr_ | Alystair: why VM?!?! |
| 07:50.36 | DoktorGreg | well it take about 10 minutes of googling to find the admin password... |
| 07:50.36 | Alystair | why not? |
| 07:50.37 | luke-jr_ | and it doesn't need an independant LAN |
| 07:50.51 | luke-jr_ | DoktorGreg: that won't let you connect * to their servers |
| 07:51.06 | DoktorGreg | luke-jr_, business plus is basic sip service |
| 07:51.18 | luke-jr_ | DoktorGreg: "business plus" shouldn't be needed |
| 07:52.03 | Alystair | luke-jr_: this server is our domain controller, our profile server, our database backup AND a dumb terminal server, and it's not even over 10% proccessor/ram usage |
| 07:52.21 | luke-jr_ | Alystair: still no reason to mess w/ VMs |
| 07:52.32 | Alystair | It's not messing with VMs if you know what you're doing. |
| 07:53.04 | DoktorGreg | I kinda think you are gonna have problems with vm |
| 07:53.17 | Alystair | worst case, I buy another piece of hardware and run a V2P |
| 07:53.17 | DoktorGreg | I think you would do better to scrounge a computer on garbage day |
| 07:53.31 | Alystair | we actually have a crappy P3 laying around there |
| 07:53.38 | luke-jr_ | VM can introduce latency, in theory |
| 07:53.46 | DoktorGreg | no theory about it |
| 07:53.54 | DoktorGreg | asterisk needs to run near real time |
| 07:53.58 | Alystair | alright |
| 07:54.06 | Alystair | fine, it has it's own hardware. |
| 07:54.22 | DoktorGreg | with 5 simultanious calls... |
| 07:54.35 | Alystair | oh christ, I just realized I have a dual capable P3 1U in the basement going unused |
| 07:54.37 | DoktorGreg | you should do alright with a piece of trash computer |
| 07:54.45 | DoktorGreg | there you go |
| 07:54.50 | DoktorGreg | wildly more than enough |
| 07:54.53 | Alystair | yeah. |
| 07:55.08 | DoktorGreg | open it up, blow it out with canned air |
| 07:55.19 | DoktorGreg | maybe if you are paranoid a couple of new hard drives |
| 07:55.24 | Alystair | nah it's clean, the only problem is that the CPU fan is pretty noisey |
| 07:55.35 | DoktorGreg | so $10 and you are good to go |
| 07:55.45 | luke-jr_ | avoid any monohydroxide bioxygen |
| 07:56.22 | luke-jr_ | or was it monohydrogen dioxide? |
| 07:56.25 | luke-jr_ | hehe |
| 07:56.27 | Alystair | no |
| 07:56.29 | Alystair | you got it wrong |
| 07:56.36 | DoktorGreg | too much di-hydrogen monoxide can kill you! |
| 07:56.36 | Alystair | reverse it heh |
| 07:56.43 | Alystair | there you go |
| 07:56.55 | luke-jr_ | :x |
| 07:57.06 | DoktorGreg | there is also the hydrogen hydroxide combination |
| 07:57.08 | tainted- | anyone good with regex? |
| 07:57.16 | luke-jr_ | tainted-: somewhat |
| 07:57.34 | luke-jr_ | not Perl-level, tho |
| 07:57.40 | tainted- | i just need to match whether a string has $ character |
| 07:57.44 | luke-jr_ | ... |
| 07:57.47 | tainted- | hmm actually i can use instr |
| 07:57.53 | luke-jr_ | eww, basic |
| 07:57.56 | DoktorGreg | omg ouch |
| 07:58.07 | luke-jr_ | Alystair: what shopping list? |
| 07:58.19 | DoktorGreg | Alystair, demo it to the controller all working with sip phones |
| 07:58.32 | DoktorGreg | soft phones |
| 07:58.38 | Alystair | I'm on the board, and the CTO so.... |
| 07:58.41 | DoktorGreg | then ask for budget |
| 07:59.03 | DoktorGreg | sheesh what is with nerds in management lately |
| 07:59.11 | DoktorGreg | Im VP where i work |
| 07:59.17 | Alystair | :D |
| 07:59.18 | DoktorGreg | and i am not at all qualifed |
| 07:59.22 | DoktorGreg | but for |
| 07:59.29 | DoktorGreg | im the only one who knows all the password |
| 07:59.33 | DoktorGreg | lol |
| 07:59.37 | Alystair | haha |
| 07:59.44 | Alystair | that's what scares me about documentation! |
| 08:00.15 | Alystair | $3k should be enough though right? |
| 08:00.20 | DoktorGreg | dude, i havent worn a tie since high school graduation |
| 08:00.21 | Alystair | also I need to factor in monthly costs etc |
| 08:00.31 | DoktorGreg | I think your monthly costs are gonna go down |
| 08:00.35 | luke-jr_ | Alystair: what are you using $3k for? |
| 08:00.46 | DoktorGreg | Break it up over three years |
| 08:01.05 | DoktorGreg | because thats the depreciation of the hardware |
| 08:01.17 | *** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn) |
| 08:01.20 | DoktorGreg | then look at phone bill for same time |
| 08:01.32 | DoktorGreg | which if it is like most companies is outragious |
| 08:01.54 | DoktorGreg | I knocked $2000 a month off our phone bill |
| 08:02.06 | DoktorGreg | it looks like we are a little bigger than you though |
| 08:02.23 | Alystair | a lot bigger, for now :) |
| 08:02.37 | Alystair | give us a year or two, heh. |
| 08:02.59 | DoktorGreg | Well i cant beleive how much work it is to get to 10 million annual sales |
| 08:03.07 | Alystair | all we are buying are the ip phones, POE Switch, the termination service and possibly an upgrade for our internet services |
| 08:03.09 | DoktorGreg | just an awsome amount of work |
| 08:03.24 | luke-jr_ | Alystair: termination service is pretty much always per minute |
| 08:03.50 | Greek-Boy | voipjet recommends iLBC codec? |
| 08:03.51 | DoktorGreg | if you buy it in blocks it costs less though |
| 08:03.56 | Alystair | We aren't in the millions yet :( |
| 08:04.04 | luke-jr_ | Greek-Boy: Voipjet recommends ulaw |
| 08:04.10 | DoktorGreg | when i started about 10 years ago at my main gig |
| 08:04.18 | DoktorGreg | we were less than 1 mil |
| 08:04.20 | DoktorGreg | just under |
| 08:04.33 | Greek-Boy | ulaw only if the conditions are good |
| 08:04.36 | DoktorGreg | you cant even recognize the company anymore |
| 08:04.47 | DoktorGreg | from what it was when i started |
| 08:04.51 | Alystair | what business is your business? :) |
| 08:04.58 | Alystair | We're in magazines and advertising. |
| 08:05.09 | luke-jr_ | crap, you're a telemarketer?!?! |
| 08:05.12 | Alystair | Hell no! |
| 08:05.15 | DoktorGreg | http://www.swancreekcandle.com/ |
| 08:05.15 | luke-jr_ | =p |
| 08:05.28 | Greek-Boy | lol |
| 08:05.30 | DoktorGreg | my web site (haw haw) does about a million a year |
| 08:05.39 | *** join/#asterisk op3r (n=op3r@202.71.189.66) |
| 08:05.40 | Alystair | You're kidding me. |
| 08:05.44 | DoktorGreg | nope |
| 08:05.47 | Alystair | 10 million off of candles. |
| 08:05.57 | Alystair | Do you have ones that smell like money? |
| 08:05.57 | DoktorGreg | fragrant candles! |
| 08:05.58 | luke-jr_ | 'a' != '10' |
| 08:06.07 | Alystair | heh |
| 08:06.30 | DoktorGreg | We have about a 10% market share |
| 08:06.40 | DoktorGreg | and no one even knows we exist... |
| 08:06.57 | DoktorGreg | because we have apple loyal customers |
| 08:06.57 | Alystair | That's a good thing usually |
| 08:07.27 | Alystair | Do you guys do bulk sales to places like walmart though? I'm assuming that's where the cash is made |
| 08:07.39 | DoktorGreg | no money to be made in department stores |
| 08:07.43 | DoktorGreg | not a penny |
| 08:07.45 | Alystair | alright |
| 08:07.50 | DoktorGreg | unless you manufacture in china |
| 08:08.02 | Alystair | Have you ever heard of something called Somethingawful.com? |
| 08:08.08 | DoktorGreg | yah |
| 08:08.13 | DoktorGreg | uhh ohh |
| 08:08.15 | luke-jr_ | DoktorGreg: got any bees wax candles? |
| 08:08.17 | Alystair | Have you seen their recent photoshop phriday? |
| 08:08.24 | DoktorGreg | nope bees wax makes horrible candles |
| 08:08.27 | Alystair | http://www.somethingawful.com/articles.php?a=3728 |
| 08:08.29 | DoktorGreg | smokes too much |
| 08:08.31 | Alystair | I think you would like it :) |
| 08:08.48 | Alystair | Yankee candle scents photoshops |
| 08:09.12 | luke-jr_ | bees wax candles are hard to find :( |
| 08:09.28 | DoktorGreg | lol thanks for that! |
| 08:09.36 | DoktorGreg | Im gonna put that link on the list |
| 08:09.40 | Alystair | the list? |
| 08:09.51 | DoktorGreg | email list of all employees who read email |
| 08:09.55 | Alystair | you should be careful, some of the entrys are rude ;) |
| 08:10.07 | DoktorGreg | naw, they all adults they can handle it |
| 08:10.10 | Alystair | alright :D |
| 08:10.12 | luke-jr_ | rude is a somewhat understatement |
| 08:10.13 | op3r | does nayone knows Asterisk@home here? |
| 08:10.24 | luke-jr_ | op3r: everyone hates Asterisk@home here |
| 08:10.30 | Alystair | op3r: #freepbx |
| 08:10.39 | Alystair | oh he's there. |
| 08:11.23 | Alystair | DoktorGreg: what's your take on this: http://www.dlink.com/products/?sec=2&pid=485 |
| 08:11.55 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:12.25 | DoktorGreg | wait a year, ask again |
| 08:12.34 | DoktorGreg | the wifi voip stuff is just WAY TOO NEW |
| 08:12.51 | DoktorGreg | it looks jucy |
| 08:12.58 | op3r | all I needed to know is that is it possible to install it on an already installed centos without downloading the ISO? |
| 08:13.03 | DoktorGreg | but i like to wait for a technology shakeout to happen |
| 08:13.16 | luke-jr_ | my analog wireless phones work nicely |
| 08:13.35 | DoktorGreg | 2010 the analog phone companies are gonna be going bankrupt |
| 08:13.35 | *** join/#asterisk thx2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 08:13.57 | DoktorGreg | once the subscriber base falls below 30% of elegible |
| 08:14.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:14.07 | Alystair | I think that's why they are all trying to do that QoS bullshit |
| 08:14.15 | DoktorGreg | exactly |
| 08:14.20 | thx2000 | Anyone know how to get my sipura 3000/asterisk to display the correct caller id info when passing pstn calls to the fxs port? |
| 08:14.37 | DoktorGreg | they wana get that 50 bucks a month for a 50Kbs wire |
| 08:14.56 | DoktorGreg | this too will pass |
| 08:15.06 | DoktorGreg | the internet routes around all these types of problems |
| 08:15.36 | DoktorGreg | next year the sip phones will have vpn clients and ansterisk protocols will be blind |
| 08:15.47 | Alystair | oh... talking about VPN |
| 08:16.15 | DoktorGreg | and then thats all she wrote to bandwidth shaping |
| 08:16.21 | Alystair | does * to * over the internet use some sorta VPN system |
| 08:16.27 | DoktorGreg | no |
| 08:16.33 | Alystair | or it justworks(tm) |
| 08:16.59 | DoktorGreg | I bet you see pervasive vpn in voip solutions next year or so |
| 08:17.21 | Alystair | ok, time to name me some good companies for cheap phones |
| 08:17.28 | DoktorGreg | sipura! |
| 08:17.38 | Alystair | do they also sell secretary style ones? |
| 08:17.54 | DoktorGreg | just get everyone the 942's |
| 08:18.10 | luke-jr_ | no apostrophe! |
| 08:18.20 | luke-jr_ | apostrophe is possession! |
| 08:18.28 | luke-jr_ | the 942 doesn't possess anything in that context |
| 08:18.28 | *** join/#asterisk lionelp (n=lionel@ip-128.net-82-216-65.rev.numericable.fr) |
| 08:18.30 | DoktorGreg | then give your receptionist a gui client |
| 08:18.38 | DoktorGreg | grammer nazi! |
| 08:18.43 | DoktorGreg | er rather |
| 08:18.47 | DoktorGreg | grammer nazi's! |
| 08:18.52 | luke-jr_ | I wouldn't be if people would JUST GET IT RIGHT! |
| 08:18.55 | Alystair | HAHA |
| 08:19.21 | luke-jr_ | GCC is a grammer nazi too >.> |
| 08:19.29 | luke-jr_ | same w/ most compilers |
| 08:19.35 | luke-jr_ | AEL 1 especially ;) |
| 08:19.43 | DoktorGreg | yup |
| 08:19.55 | DoktorGreg | I am all nerd all the time |
| 08:20.06 | DoktorGreg | people come up to me in street in my small town |
| 08:20.13 | DoktorGreg | and beg me to fix their computers |
| 08:20.27 | DoktorGreg | ignoring the local computer repair store |
| 08:20.43 | Alystair | :( |
| 08:20.47 | luke-jr_ | I don't let people like that know me |
| 08:20.59 | Alystair | whole board gets me to fix their machines, or try to. |
| 08:21.22 | DoktorGreg | I have a one speed cruiser bike made out of graphite |
| 08:21.31 | DoktorGreg | just to annoy the bike jocks |
| 08:21.55 | DoktorGreg | Yah, your faster, but my bike weighs less |
| 08:22.10 | luke-jr_ | I have pi memorized to ...375 |
| 08:22.24 | Alystair | heh |
| 08:22.31 | luke-jr_ | I think it's like 30 some digits |
| 08:22.33 | Alystair | you could probably list of 3 random digits and it would be somewhere in pi |
| 08:22.50 | luke-jr_ | there's a website that does that |
| 08:22.51 | *** join/#asterisk lucifr (n=chatzill@ppp-71-135-229-57.dsl.irvnca.pacbell.net) |
| 08:23.06 | lucifr | hello |
| 08:23.09 | luke-jr_ | hi |
| 08:23.14 | lucifr | :P |
| 08:23.30 | DoktorGreg | I got in a in algebra 2 in high school because i wrote an arbitrary length calculator for the teacher so he could computer pi to whatever digit he wanted to as demonstration |
| 08:23.41 | lucifr | I hope you guys can help me with this questions... |
| 08:23.54 | DoktorGreg | repent! |
| 08:23.57 | luke-jr_ | lucifr: we can't, sorry |
| 08:24.02 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 08:24.06 | luke-jr_ | we don't know the questions |
| 08:24.08 | luke-jr_ | :( |
| 08:24.09 | lucifr | really? |
| 08:24.13 | Alystair | I'm suprised he didn't ask in the form of a question. |
| 08:24.16 | AsteriskAlbania | hi everyone |
| 08:24.19 | lucifr | hehehehehehehe |
| 08:24.27 | luke-jr_ | not more specifically than 42, no |
| 08:24.29 | AsteriskAlbania | anyone has experience with asterisk and radius |
| 08:24.41 | luke-jr_ | AsteriskAlbania: the radius of asterisk is 42 |
| 08:24.43 | Alystair | AsteriskAlbania: I... sense... that you are from Albania? |
| 08:24.49 | AsteriskAlbania | yeap |
| 08:25.05 | AsteriskAlbania | luke-jr_: is 42 ? |
| 08:25.06 | lucifr | I looking forward building my first Asterisk solution for myself... It will be an linux embedded systems based on the nano-ITX mobo... |
| 08:25.12 | luke-jr_ | AsteriskAlbania: yes |
| 08:25.17 | AsteriskAlbania | which means ? |
| 08:25.31 | luke-jr_ | lucifr: fun, might want to check out OpenWrt or OpenEmbedded |
| 08:25.37 | Alystair | that they haven't translated Douglas Adams books to Albianian, obviously :) |
| 08:25.53 | lucifr | luke-jr, ok thank you |
| 08:26.00 | luke-jr_ | AsteriskAlbania: the circumference would be 42 * 2 * pi |
| 08:26.13 | luke-jr_ | lucifr: anything more specific |
| 08:26.13 | luke-jr_ | ? |
| 08:26.17 | AsteriskAlbania | :) thats funny |
| 08:26.29 | lucifr | One question still in my mind and read will awhile ago but can't seem to remember is... How can I get my own (700) number? |
| 08:26.37 | luke-jr_ | wtf is a 700 number? |
| 08:26.58 | DoktorGreg | 700 is the voip area code |
| 08:27.07 | luke-jr_ | POTS-dialable? |
| 08:27.08 | lucifr | exactly! |
| 08:27.14 | lucifr | yes |
| 08:27.18 | luke-jr_ | oooh |
| 08:27.22 | DoktorGreg | fwd has em i think |
| 08:27.31 | luke-jr_ | long distance? or local? |
| 08:27.32 | lucifr | DoktorGreg, can you explain this, please? |
| 08:27.59 | lucifr | DoktorGreg, is there a charge for this (700) number? |
| 08:28.04 | DoktorGreg | got me |
| 08:28.09 | lucifr | i see |
| 08:28.16 | DoktorGreg | all my research has been in pri |
| 08:28.24 | DoktorGreg | as i have a pri at my disposal |
| 08:28.28 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:28.33 | op3r | pri? |
| 08:28.40 | DoktorGreg | ok im tired now |
| 08:28.43 | AsteriskAlbania | luke-jr_: I have a radius billing software can I make asterisk authenticate to it ? |
| 08:28.46 | Alystair | noooooooo |
| 08:28.48 | DoktorGreg | primary rate interface |
| 08:28.54 | op3r | ok |
| 08:29.11 | op3r | my brain is bleeding at the moment |
| 08:29.26 | DoktorGreg | its a way of breaking a t1 line up into 23 voice channels and one data channel |
| 08:29.26 | Alystair | so how hard is it to get an 800 # from a terminator? |
| 08:29.27 | op3r | Im trying to figure out how to make the supervisors hears the agents on call |
| 08:29.40 | tainted- | Alystair depending on the itsp, some - instantly |
| 08:29.45 | DoktorGreg | all the 800 numbers are gone |
| 08:29.54 | luke-jr_ | Alystair: 800 or toll free? |
| 08:30.02 | DoktorGreg | you can no longer get an 800 number unless you buy it from someone |
| 08:30.04 | tainted- | DoktorGreg not true |
| 08:30.21 | tainted- | well that's a vague statement |
| 08:30.50 | DoktorGreg | last i checked, you could sell 800 numbers for good money |
| 08:31.00 | tainted- | when did u check? |
| 08:31.04 | tainted- | 1987? |
| 08:31.08 | DoktorGreg | couple of years ago |
| 08:31.21 | luke-jr_ | I get a busy signal trying to call (700) 555-4141 |
| 08:31.27 | lucifr | luke-jr_, are "OpenWrt and OpenEmbedded" Asterisk based solutions? |
| 08:31.40 | tainted- | http://businessesales.att.com/products_services/tollfreeproduct_cataloglookup.jhtml;jsessionid=UD1O4CWIMSGRXQFIHOISKT4AVA3Y4JLW?res=y&_requestid=43851 |
| 08:31.47 | luke-jr_ | lucifr: they're generic embedded cross-compiler systems |
| 08:31.49 | tainted- | there's a block of 800 # right there |
| 08:31.59 | tainted- | snatch them up and sell them for good money |
| 08:32.32 | DoktorGreg | oh |
| 08:32.38 | lucifr | luke-jr_, i see thanx |
| 08:32.51 | DoktorGreg | well then the situation has changes since 888 877 and 866 |
| 08:33.11 | Alystair | ok so |
| 08:33.21 | tainted- | soon 855 |
| 08:33.22 | Alystair | lets say I want a local number and a toll free number, that's simple right |
| 08:33.34 | tainted- | Alystair for the most part yes |
| 08:33.39 | luke-jr_ | Alystair: I clicked the button to buy an 866 number and it instantly worked |
| 08:33.49 | luke-jr_ | same w/ my 402/Omaha # |
| 08:34.02 | Alystair | ok so now I need to find a provider, is there a list somewhere? |
| 08:34.13 | Alystair | specifically ones that deal with Canada heh |
| 08:34.14 | tainted- | luke-jr_ where |
| 08:34.20 | luke-jr_ | tainted-: SellVoIP |
| 08:34.21 | tainted- | Alystair what part of CA |
| 08:34.30 | Alystair | Ontario, specifically Toronto |
| 08:34.35 | luke-jr_ | Alystair: Note SellVoIP has higher rates for toll-free calls originating in Canada |
| 08:34.50 | tainted- | Canada toll free is a different story |
| 08:35.05 | luke-jr_ | Alystair: and also that Canada has some dumb law that can make certain VoIP usage illegal |
| 08:35.11 | Alystair | well it's still a 1800 right? |
| 08:35.15 | Alystair | :O |
| 08:35.21 | Alystair | don't ruin my dream! |
| 08:35.32 | tainted- | luke-jr_ is sellvoip your operation? |
| 08:35.37 | luke-jr_ | tainted-: nope |
| 08:35.40 | tainted- | Alystair no 1800 != 1800 |
| 08:35.47 | luke-jr_ | Alystair: 4.5 cents/minute for toll-free calls originating in Canada |
| 08:36.02 | luke-jr_ | w/ SellVoIP |
| 08:36.05 | tainted- | US 1800 != CA 1800 i should say |
| 08:37.07 | Alystair | couldn't I get a USA 1800# that provides both america and canada? |
| 08:37.13 | Alystair | or vice versa |
| 08:38.00 | Alystair | wait yeah, they sell them, nevermind |
| 08:39.44 | luke-jr_ | who does? |
| 08:40.19 | tainted- | yes |
| 08:40.23 | Alystair | I mean sellvoip has toll free numbers that provide both |
| 08:40.28 | tainted- | u only get 4.9/min if someone from canada calls u |
| 08:40.35 | tainted- | otherwise it's a lot less |
| 08:40.40 | Alystair | they don't sell 416 numbers though |
| 08:42.09 | luke-jr_ | they don't? |
| 08:42.20 | luke-jr_ | hm |
| 08:44.11 | Alystair | does vonage supply DIDs? |
| 08:44.43 | luke-jr_ | tainted-: my "operation" w/ regard to VoIP is currently reselling toll-free numbers and using reinvites so I don't need to deal with bandwidth |
| 08:44.49 | luke-jr_ | Alystair: Vonage sucks |
| 08:44.55 | DoktorGreg | vonage wants to be traditional phone service |
| 08:45.07 | DoktorGreg | like ma bell circa 1979 |
| 08:45.10 | Alystair | ew. |
| 08:45.15 | Alystair | ok that's definately bad then |
| 08:45.20 | luke-jr_ | yes |
| 08:45.22 | DoktorGreg | they want to own equiptment |
| 08:45.34 | Alystair | hmmm |
| 08:45.35 | luke-jr_ | iConnectHere does not support Asterisk, though it used to work for me |
| 08:45.54 | tainted- | luke-jr_ oh that's right.. how's that going.. i remember telling u that SER is the way to go |
| 08:45.55 | DoktorGreg | from what i am hearing htough |
| 08:46.04 | DoktorGreg | vonage is having serious scaling problems |
| 08:46.06 | luke-jr_ | tainted-: works fine now, not sure why |
| 08:46.15 | DoktorGreg | and they keep blaming the cable and dsl providers |
| 08:46.22 | *** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net) |
| 08:46.35 | tainted- | luke-jr_ asterisk stays out of media? |
| 08:46.45 | luke-jr_ | tainted-: yep, when it's supposed to |
| 08:46.53 | tainted- | how? |
| 08:47.00 | luke-jr_ | SIP reinviting |
| 08:47.19 | tainted- | i know |
| 08:47.22 | tainted- | but where |
| 08:47.26 | luke-jr_ | ? |
| 08:47.26 | tainted- | dialplan? |
| 08:47.30 | luke-jr_ | yes |
| 08:47.31 | Shaun2222 | Any of you guys good with DB? |
| 08:47.32 | tainted- | how did u implement it |
| 08:47.35 | luke-jr_ | Dial? |
| 08:47.40 | tainted- | w/ dial? |
| 08:47.45 | tainted- | what options is that!? |
| 08:47.45 | Shaun2222 | Set(AGENTEXTEN=${DB(Agent/${EXTEN})}) |
| 08:47.51 | Shaun2222 | anybody know whats wrong with that? |
| 08:47.57 | luke-jr_ | it's default if you don't use Monitor, L(x:y), tT, or some other thigns |
| 08:48.09 | luke-jr_ | and both ends are SIP on a public IP |
| 08:48.23 | tainted- | public ip required? |
| 08:48.47 | luke-jr_ | well, I imagine it could work if the two ends are on the same side of a NAT |
| 08:49.01 | tainted- | hmm |
| 08:49.03 | luke-jr_ | potentially, it might work through NAT... not sure |
| 08:49.10 | tainted- | so it must already be working |
| 08:49.13 | *** join/#asterisk pengyong (n=lala@222.188.142.216) |
| 08:49.24 | tainted- | i should take a look at the rtp stream |
| 08:49.38 | Alystair | Hrm! |
| 08:49.44 | luke-jr_ | I don't know why not many people use it |
| 08:49.57 | luke-jr_ | it saves the bandwidth of the audio |
| 08:50.02 | tainted- | how do u know it's not involved in media |
| 08:50.21 | luke-jr_ | Alystair: I have a ML post selling em in my browser, but they require 20 DIDs minimum and at a higher cost than SellVoIP |
| 08:50.28 | luke-jr_ | tainted-: Ethereal doesn't show RTP |
| 08:50.55 | luke-jr_ | or rather, the RTP stops when it's reinvited |
| 08:51.37 | tainted- | ah |
| 08:51.43 | tainted- | must be the L() options i use |
| 08:51.56 | luke-jr_ | if more than one L() option is supplied, that disables it |
| 08:52.04 | luke-jr_ | since it needs the RTP to inject the announcement |
| 08:52.10 | tainted- | right |
| 08:52.35 | luke-jr_ | though I'm sure if you wanted to, you could reinvite it back to your own * box to announce, and then reinvite back to a direct connection |
| 08:52.39 | tainted- | i could go w/o the announcement if there were another mechanism for limiting th ecall |
| 08:52.41 | Alystair | ok quick question |
| 08:52.44 | luke-jr_ | since you keep control over the call |
| 08:52.47 | Alystair | really really wierd |
| 08:53.03 | Alystair | Say I was in a colo facility and the cage next to ours had a CRT that was ALWAYS on |
| 08:53.03 | luke-jr_ | tainted-: I don't think L() with only one argument will block the reinvite |
| 08:53.20 | Alystair | and it had a very quickly scrolling list of telephone numbers on, like two numbers on each line |
| 08:53.25 | Alystair | and a running cash total, what do you think it was? |
| 08:53.31 | Alystair | calling card system or a voip system? |
| 08:53.32 | *** join/#asterisk koenvi (n=root@tech.ascom.be) |
| 08:53.40 | koenvi | hello you all! |
| 08:53.43 | luke-jr_ | maybe both |
| 08:54.12 | tainted- | Alystair could be a dialing for dollars system |
| 08:54.25 | Alystair | all I know was that they were making at least $15 every time I blinked :p |
| 08:54.42 | koenvi | this is an emergency call |
| 08:54.47 | luke-jr_ | koenvi: 911 |
| 08:54.49 | tainted- | well they'd better |
| 08:55.08 | tainted- | considering the volume wholesalers have to push |
| 08:55.11 | Alystair | koenvi: you sound like the biginning of a massive trance call |
| 08:55.11 | koenvi | right ... do they help on asterisk emergencies? |
| 08:55.13 | Alystair | er song |
| 08:55.34 | tainted- | koenvi ask for the asstricks special |
| 08:55.39 | Alystair | Hello you all! This is an emergency call! *guntz guntz guntz* |
| 08:55.53 | koenvi | does anyone here know Nortel BCM? |
| 08:55.55 | luke-jr_ | koenvi: I'll help for $500 per minute ;) |
| 08:56.08 | luke-jr_ | I don't =p |
| 08:56.25 | Alystair | ICBM over IP ;D |
| 08:56.27 | koenvi | luke-jr: I'll check with my CEO |
| 08:56.36 | Alystair | ... |
| 08:56.38 | Alystair | Wow. |
| 08:56.39 | luke-jr_ | koenvi: lol |
| 08:56.51 | Alystair | luke-jr_ he may be serious you know. |
| 08:57.03 | koenvi | I don't pay the bills, so I don't care |
| 08:57.07 | luke-jr_ | Alystair: I don't think anyone gets $30k/hr |
| 08:57.17 | koenvi | I'm just supposed to make things work |
| 08:57.18 | Alystair | heh, yeah they do |
| 08:57.24 | luke-jr_ | ... |
| 08:57.36 | Alystair | luke-jr_: lawyers ;) |
| 08:57.37 | koenvi | at fot the moment .. it doesn't work |
| 08:58.00 | luke-jr_ | Alystair: that's ~$60 mil a year... |
| 08:58.05 | koenvi | I'm trying to connect Asterisk to a Nortel BCM with a pri |
| 08:58.15 | Alystair | I didn't say they worked all day :p |
| 08:58.29 | Alystair | or all week, or month. |
| 08:58.30 | mocker | Ok, brain fart. Can someone show me how to set a priority label? |
| 08:58.40 | koenvi | tried EuroISDN and Qsig without success |
| 08:58.42 | luke-jr_ | mocker: n(label) |
| 08:58.44 | mocker | My google skills are failing during my maintenance window. |
| 08:58.47 | luke-jr_ | mocker: tho AEL is nicer |
| 08:58.50 | mocker | luke-jr_: Thanks. :) |
| 08:59.01 | luke-jr_ | with AEL it's just |
| 08:59.02 | luke-jr_ | label: |
| 08:59.03 | luke-jr_ | :) |
| 08:59.15 | koenvi | when setting up a call from Asterisk to BCM, it replies with 'release complete' |
| 09:00.40 | koenvi | what flavor of QSig does * use anyway? |
| 09:00.51 | luke-jr_ | QS? |
| 09:01.10 | koenvi | OSI? ETSI? |
| 09:01.13 | Alystair | http://en.wikipedia.org/wiki/QSIG |
| 09:01.32 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 09:04.18 | koenvi | the 'release complete' message show 'cause unkwown' and 'class=protocol error' |
| 09:05.49 | Alystair | luke-jr_: is there anything wrong with putting VOIP traffic on our standard network segment or should it be seperated all together? |
| 09:06.16 | Alystair | intranet network segment |
| 09:06.20 | luke-jr_ | Alystair: shouldn't be a problem unless you're pushing more traffic than the router can handle |
| 09:06.25 | *** join/#asterisk mozartsghost (n=dexter@dsl-145-196-98.telkomadsl.co.za) |
| 09:07.02 | mozartsghost | morning all :) |
| 09:07.19 | Alystair | hmmmmmm |
| 09:08.40 | mozartsghost | hmm, having a starange prob on a PRI, I have a schools pabx connected to a asterisk box via a pri. now the pri works fine. but when the schools pabx starts sending digits, asterisk thinks its done by the third digit and starts dialing out onto the outgoing trunks. anyone have any pointers ? as to how i can tell asterisk to wait for the rest ? |
| 09:08.51 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 09:09.04 | mozartsghost | OH, and aswell, when the schools pabx is connected to a telco pri, it works fine. |
| 09:09.11 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:09.13 | mozartsghost | anyone have any advice on that ? |
| 09:09.31 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:10.31 | mozartsghost | first time i'm having this prob. very wierd. |
| 09:10.53 | koenvi | no one knows, or no one cares |
| 09:11.10 | mozartsghost | heh |
| 09:11.53 | mozartsghost | thats rich coming from someone who irc's as root. |
| 09:15.17 | koenvi | mozartsghost: you won't find much on this piece of junk |
| 09:15.34 | mozartsghost | lol, in this channel u mean ? |
| 09:15.38 | mozartsghost | hmm |
| 09:15.48 | luke-jr_ | doubt it |
| 09:15.52 | OloBola | what is an "octel 0250" used for? We gotta couple of them sitting around upstairs. |
| 09:15.54 | luke-jr_ | he probably means the box he's IRCing from |
| 09:16.11 | mozartsghost | lol |
| 09:17.00 | mozartsghost | OloBola: isn't that a voicemail server ? |
| 09:17.45 | chris_ast | When we delete a voicemail * normally spells out options like 'press 7 to confirm delete' etc, how can we make * to play next voicemail when we delete a voicemail? Please help me |
| 09:17.54 | OloBola | I have no idea! It sure looks cool though. It's got a 24 gig tape drive in it too, so it can't be terribly old. |
| 09:18.01 | OloBola | its |
| 09:18.20 | OloBola | or maybe it is old |
| 09:18.20 | mozartsghost | lol, I know their damn expensive. |
| 09:18.28 | mozartsghost | ye, I believe its a voicemail systems thingy. |
| 09:18.37 | mozartsghost | ppl still use them tho. |
| 09:18.38 | chris_ast | mozartsghost,luke-jr_,koenvi: any ideas on my question |
| 09:18.43 | OloBola | interesting |
| 09:18.48 | mozartsghost | chris_ast reading.. |
| 09:19.23 | mozartsghost | I'm sure u can modify that somehow |
| 09:19.46 | koenvi | chris_ast: haven't played with * voicemail yet |
| 09:20.19 | koenvi | I don't even get past connecting the pri :-( |
| 09:21.29 | chris_ast | mozartsghost: I checked voip-info and some search on net but no luck |
| 09:22.37 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-168-37.hsd1.ca.comcast.net) |
| 09:23.06 | CrashHD | can anyone tell me which phones truely support the Sip BLA standard? |
| 09:24.10 | macTijn | *tumbleweed rolls by* |
| 09:24.16 | CrashHD | heh |
| 09:24.16 | mozartsghost | chris_ast: not shure, just had a squiz on google, saw on or two links. |
| 09:25.52 | koenvi | I think I'm gonna sit in a corner and start crying ... |
| 09:26.32 | mozartsghost | koenvi what are u struggling with ? |
| 09:26.46 | koenvi | pri connection ... |
| 09:26.59 | mozartsghost | ok, elaborate abit :) |
| 09:27.05 | koenvi | gives a protocol error (that's what it looks like) |
| 09:27.24 | chris_ast | mozartsghost: Even I cud not find related information |
| 09:27.39 | mozartsghost | koenvi: ok what have u got setup ? |
| 09:27.56 | koenvi | * is pri_net in EuroISDN |
| 09:28.02 | koenvi | link is up |
| 09:28.14 | mozartsghost | chris_ast: ... :/ should maybe wait for one of these techs to wake up or come back. |
| 09:28.23 | mozartsghost | koenvi: what have u got it connected to ? |
| 09:28.23 | CrashHD | keep going |
| 09:28.37 | koenvi | Norte BCM |
| 09:28.41 | CrashHD | what is it doing? |
| 09:28.46 | koenvi | Nortel that is |
| 09:29.01 | mozartsghost | ok, and the Nortel is in slave mode right ? |
| 09:29.05 | koenvi | yes |
| 09:29.07 | mozartsghost | what pinout are u using ? |
| 09:29.11 | chris_ast | mozartsghost:I will keep posting here now and then and search on net, we need this feature |
| 09:29.33 | mozartsghost | chris_ast: kewl, I'm quite interested in that aswell acctually. will keep an eye out. |
| 09:29.33 | CrashHD | ya make sure you have the right cabling |
| 09:29.40 | koenvi | no idea, but link is up, so pins should be oké |
| 09:29.44 | CrashHD | no |
| 09:29.51 | CrashHD | go look up t1 cross over cable |
| 09:29.53 | CrashHD | on google |
| 09:29.54 | CrashHD | verify |
| 09:29.57 | mozartsghost | koenvi: and also, does the Nortel side say link is up ? |
| 09:29.57 | CrashHD | I had the same problem |
| 09:30.52 | koenvi | yes |
| 09:30.52 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:31.16 | mozartsghost | a straight PRI cable is, 1-1,2-2,4-4,5-5. |
| 09:31.31 | CrashHD | you want a cross over though |
| 09:31.58 | mozartsghost | crossover PRI is, 1-4 2-1 4-5 5-2 |
| 09:32.05 | CrashHD | http://www.cisco.com/univercd/cc/td/doc/product/ipvc/ipvc20/priguide/appd.htm |
| 09:32.14 | mozartsghost | sorry |
| 09:32.17 | mozartsghost | I screwed that up |
| 09:32.33 | mozartsghost | no thats right |
| 09:32.47 | CrashHD | 1-4, 2-1, 4-5, 5-2 |
| 09:32.48 | CrashHD | ya |
| 09:33.09 | mozartsghost | koenvi: double check your connections quickly |
| 09:33.16 | papo | Hi. I'm trying to plan a tricky set up. I have an asterisk server on my gateway and want to connect with the same SIP software phone from outside and inside. I understand the problem that asterisk would reinvite to my client connecting from the internet (which is what I want), but also to the one connecting from my LAN which I don't want. Can I solve this with asterisk or should I use some sort of hack, for example running asterisk on two |
| 09:33.16 | papo | ports, configuring two servers in my client and firewall one from inside and one from outside? |
| 09:36.34 | koenvi | mozartsghost: checked and doublechecked ... link looks ok, no alarms on zap status |
| 09:37.23 | mozartsghost | ok, kewl. paste me your zapata.conf quickly. in private |
| 09:37.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:37.41 | CrashHD | when you try to send a call over those channels what happens? |
| 09:37.43 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 09:38.04 | mozartsghost | yeah, paste us the error koenvi |
| 09:39.04 | Alystair | ugh, 5:30am |
| 09:39.21 | CrashHD | and the dial line you use in your exten.conf |
| 09:42.10 | *** join/#asterisk MGSsancho (n=user@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
| 09:43.23 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 09:44.47 | koenvi | I'm an irc newbie ... how do I set up a private channel mozartsghost? |
| 09:45.00 | koenvi | btw, I'm using textbased irc (epic) |
| 09:45.01 | mozartsghost | double click on my name |
| 09:45.05 | koenvi | no fancy stuff |
| 09:45.06 | mozartsghost | oh |
| 09:45.13 | mozartsghost | /query mozartsghost |
| 09:45.16 | CrashHD | "/query $USER" |
| 09:45.39 | wasim | <PROTECTED> |
| 09:45.49 | mozartsghost | lol |
| 09:46.00 | puzzled | morning |
| 09:46.11 | mozartsghost | morning |
| 09:46.56 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 09:49.12 | L|NUX | hello |
| 09:49.19 | mozartsghost | hello |
| 09:49.25 | L|NUX | i am trying to do realtime sip |
| 09:49.37 | L|NUX | but i get this error when i try to register my extension |
| 09:49.37 | L|NUX | Apr 26 10:11:28 WARNING[11942] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 09:50.03 | puzzled | mysql is not running, username/password incorrect etc. |
| 09:50.25 | L|NUX | well mysql is running |
| 09:50.34 | L|NUX | and username and password is also corret :0 |
| 09:50.38 | puzzled | firewall preventing from getting to it |
| 09:50.39 | Nugget | or mysql silently ate the row, which mysql does sometimes. :) |
| 09:51.26 | Shaun2222 | anybody know why when i use chanisavail(sip/303) why the dialplan just exits out with a non-zero value? 303 doesnt exist, the phone unregistered... |
| 09:51.39 | Shaun2222 | i see this warning... Apr 26 02:51:25 WARNING[11297]: chan_sip.c:1973 create_addr: No such host: 303 |
| 09:51.57 | Shaun2222 | i want it to just continue on so i can check if ${AVAILCHAN} is null |
| 09:53.42 | L|NUX | Nugget : i am using 1.2.x |
| 09:58.12 | RoyK | ~nickometer L|NUX |
| 09:58.18 | RoyK | :) |
| 09:59.04 | CrashHD | ~bored |
| 09:59.06 | jbot | La ... lalalala ... beer! |
| 09:59.33 | CMike | Beer ? ohh.. .. I'll take one keg too .. |
| 09:59.34 | CMike | :) |
| 10:01.01 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 10:03.04 | CrashHD | ~beer |
| 10:03.06 | jbot | No beer and no TV make purl... something something... |
| 10:04.27 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 10:05.06 | RoyK | ~lart L|NUX |
| 10:05.07 | RoyK | zoa: hi |
| 10:05.34 | L|NUX | jbot ~kick RoyK |
| 10:05.36 | RoyK | zoa: we had some problems with latency with the jb as well |
| 10:05.39 | L|NUX | ~kick RoyK |
| 10:05.41 | jbot | bugger off sod! |
| 10:05.56 | wasim | :) |
| 10:06.05 | L|NUX | wasim : hey wasim |
| 10:06.22 | L|NUX | wasim : salam |
| 10:06.27 | wasim | L|NUX: ws |
| 10:06.55 | L|NUX | wasim you got some time ? |
| 10:07.14 | wasim | only if you have some cold beer |
| 10:07.27 | wasim | even murree |
| 10:07.39 | L|NUX | argh |
| 10:07.43 | L|NUX | i don't drink :( |
| 10:08.06 | L|NUX | wasim bahi i was trying to do realtime extension from db but its not working :( |
| 10:08.16 | L|NUX | i was following voip-info.org wiki |
| 10:08.27 | L|NUX | but its giving me error |
| 10:08.32 | L|NUX | Apr 26 10:11:28 WARNING[11942] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 10:08.54 | RaYmAn-Bx | have you checked that the realtime mysql module is loaded? |
| 10:09.03 | RoyK | L|NUX: install asterisk-addons :P |
| 10:09.53 | L|NUX | RoyK : O_o |
| 10:09.54 | L|NUX | brb |
| 10:10.18 | koenvi | mozartsghost still here? |
| 10:10.23 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 10:12.17 | mozartsghost | yeah[ |
| 10:12.21 | RoyK | zoa: ? |
| 10:12.24 | mozartsghost | just going to buy lunch quick, I will brb. |
| 10:13.19 | koenvi | I'll do the same |
| 10:13.45 | *** join/#asterisk Durk (n=Durk@dsl-58-6-123-162.qld.westnet.com.au) |
| 10:14.10 | Durk | hey all :) |
| 10:14.21 | CrashHD | hello |
| 10:14.28 | CrashHD | mozartsghost |
| 10:14.30 | CrashHD | hmmm |
| 10:14.32 | CrashHD | ohhhhhh |
| 10:14.34 | CrashHD | the net |
| 10:14.39 | CrashHD | sandra bullock |
| 10:14.41 | CrashHD | mmmmmm |
| 10:14.47 | CrashHD | hot hacker chick |
| 10:14.54 | CrashHD | guessing your a guy though? |
| 10:14.57 | CrashHD | *laugh* |
| 10:14.59 | Durk | is anyone here familiar with softhangup? I can't get it to hang up an active call... |
| 10:15.09 | CrashHD | Hangup()? |
| 10:15.12 | Durk | lol yeah i'm a guy :p |
| 10:15.15 | Durk | nah, softhangup |
| 10:15.18 | CrashHD | durk I meant mozartsghost |
| 10:15.23 | CrashHD | never used it |
| 10:15.28 | Durk | ahh kk |
| 10:15.29 | CrashHD | sorry |
| 10:15.32 | Durk | np.. ta anyway :) |
| 10:16.04 | L|NUX | RoyK : bro same error :( |
| 10:16.11 | Durk | its similar to hangup, just you can specify which channel to kill |
| 10:16.19 | Durk | having trouble getting it to hang up an active call :/ |
| 10:18.01 | Durk | woohoo! got it workin... |
| 10:18.23 | Durk | ahhhh the irony... spend hours on it, finally give up and ask for help, and i find the problem :/ |
| 10:20.35 | wasim | L|NUX: type mysql -p |
| 10:20.39 | wasim | L|NUX: can you get to mysql? |
| 10:20.59 | wasim | L|NUX: if so, check if the db and tables exist that you have defined in extconf |
| 10:21.20 | L|NUX | [root@mbox asterisk]# mysql -p |
| 10:21.21 | L|NUX | Enter password: |
| 10:22.31 | Greek-Boy | can g279 really work with only 8kbps? |
| 10:22.54 | wasim | Greek-Boy: for the codec, yes |
| 10:22.54 | L|NUX | yup |
| 10:23.04 | stoffell | I dial out through zap (bristuff, ISDN), when I hangup before remote picks up, the remote phone keeps ringing.. is this a setting in zapata.conf maybe?? |
| 10:23.15 | Durk | do you lose much quality running at 8k??? |
| 10:23.23 | wasim | Durk: nope |
| 10:23.33 | Durk | surely that can't be good... then again.. you do have to pay for it, guess it'd have to be good :) |
| 10:23.34 | wasim | but 8k is just the codec weight, you still need a protocol |
| 10:23.45 | Durk | ahhhh ok |
| 10:23.50 | wasim | like SIP or IAX that are 13 kbps each |
| 10:24.06 | Durk | 21 in total still isn't bad |
| 10:24.19 | Durk | hows ulaw/alaw hold up for kps? |
| 10:24.30 | wasim | 64 |
| 10:24.33 | Durk | ouch. |
| 10:24.57 | *** join/#asterisk backblue (n=moo@87-196-15-195.net.novis.pt) |
| 10:25.05 | Greek-Boy | worst for me is i live in a country only with sattelite connectivity |
| 10:25.21 | Durk | yeah that sucks.. latency will kill you regardless of packet size |
| 10:25.38 | Greek-Boy | well |
| 10:25.42 | Greek-Boy | it doesn't really kill me |
| 10:25.45 | Greek-Boy | but i do feel it |
| 10:25.57 | Greek-Boy | and I was thinking of starting a voip provider here |
| 10:26.09 | Greek-Boy | maybe it suits me better to get a co-located server in the us |
| 10:26.30 | Durk | surely there has to be a datacentre somewhere with a decent line.. |
| 10:26.51 | Durk | find out where your sat links terminate and rent a rack in there |
| 10:27.34 | AsteriskAlbania | radiusclient-ng and asterisk ? anyone can help |
| 10:28.27 | Greek-Boy | the sat links terminate in norway |
| 10:28.38 | wasim | aha! thats your problem, crazy vikings |
| 10:28.55 | L|NUX | wasim bro its not working no idea what happen :( |
| 10:29.12 | wasim | L|NUX: ssh? |
| 10:29.27 | Greek-Boy | wasim hows the connectivity in pakistan? I know india is connected to SAFE which goes to europe via SAT-3 through south africa |
| 10:29.34 | L|NUX | wasim : ys |
| 10:29.34 | Greek-Boy | sumbarine cables |
| 10:29.40 | L|NUX | wasim : can i give you |
| 10:29.42 | wasim | Greek-Boy: smw3 |
| 10:29.47 | L|NUX | let me give you in pvt |
| 10:30.56 | Greek-Boy | smw3, is it sub cable |
| 10:30.57 | Greek-Boy | ? |
| 10:31.08 | wasim | Greek-Boy: yeah |
| 10:31.56 | Greek-Boy | i wonder if there is a site that shows all the worlds sub cables |
| 10:32.06 | Greek-Boy | i'll google it up and see what i can find |
| 10:32.28 | tainted- | anyone work with ATAs for t.38? |
| 10:37.05 | *** part/#asterisk koenvi (n=root@tech.ascom.be) |
| 10:38.49 | *** join/#asterisk koenvi (n=kova@tech.ascom.be) |
| 10:40.11 | *** join/#asterisk Husk (n=Husk__@202.55.153.169) |
| 10:45.00 | wasim | tainted-: RoyK and coppice were |
| 10:49.36 | RoyK | tainted-: we have a few |
| 10:51.58 | *** part/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 10:55.20 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 11:00.43 | AsteriskAlbania | any one has been able to configure ASTERISK to authenticate to a radius server ? |
| 11:02.58 | AsteriskAlbania | any one has been able to configure ASTERISK to authenticate to a radius server ? |
| 11:03.39 | iDunno | a question so good it had to be asked twice, in the space of 3 minutes, with no other talk in between. |
| 11:07.42 | *** join/#asterisk duckz (n=duckz@193.192.47.26) |
| 11:08.22 | wasim | http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth |
| 11:12.29 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 11:13.03 | AsteriskAlbania | ANYONE Can you help for Radius & Asterisk |
| 11:13.47 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 11:19.11 | *** part/#asterisk koenvi (n=kova@tech.ascom.be) |
| 11:20.18 | AsteriskAlbania | ANYONE Can you help for Radius & Asterisk |
| 11:21.39 | Greek-Boy | did u check the site wasim posted? |
| 11:21.41 | Greek-Boy | its on the wiki |
| 11:24.26 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:27.26 | *** join/#asterisk pengyong (n=lala@222.185.18.211) |
| 11:28.01 | tzafrir | AsteriskAlbania, asking this again and again won't get you the answer faster |
| 11:28.53 | tzafrir | The information in that wiki page is dated and has broken links. See my comments there. |
| 11:37.36 | *** join/#asterisk syle2 (n=blah@unaffiliated/syle) |
| 11:38.31 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
| 11:42.21 | ManxPower | jbot, ECFO is Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", or other usless terms. |
| 11:42.24 | jbot | okay, ManxPower |
| 11:43.25 | *** join/#asterisk damned (n=vpol@prior.lanck.net) |
| 11:43.28 | damned | hi all |
| 11:43.46 | damned | can anybody suggest a MML client for linux? |
| 11:45.13 | ManxPower | jbot, ECFO is Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 11:45.14 | jbot | ...but ecfo is already something else... |
| 11:45.24 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 11:45.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 11:45.27 | ManxPower | jbot, no, <reply>ECFO is Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 11:45.28 | jbot | okay, ManxPower |
| 11:46.22 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 11:49.53 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 11:50.14 | RoyK | ~ecfo |
| 11:50.16 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 11:54.01 | *** join/#asterisk shadebob (n=chatzill@adsl196-197-168-206-196.adsl196-6.iam.net.ma) |
| 11:54.06 | shadebob | hi |
| 11:54.57 | shadebob | Someone known if I can make ppp a connection with an te110p? |
| 11:56.57 | RoyK | zapras perhaps |
| 11:58.19 | shadebob | I think zapras if for a pppd server? My te110p will be a client |
| 11:58.42 | RoyK | i don't want to support that :) |
| 11:58.54 | shadebob | ;) |
| 11:59.01 | RoyK | shadebob: for non-telephony, i'd rather use a sangoma |
| 12:03.15 | *** join/#asterisk I-MOD (n=opticron@gateway.digium.com) |
| 12:05.11 | shadebob | ok RoyK. I will see. |
| 12:05.45 | *** join/#asterisk papo (n=mathias@adsl-177-161-fixip.tiscali.ch) |
| 12:13.20 | RoyK | hm. anyone that knows what revision resembles 1.2.6? |
| 12:15.56 | *** join/#asterisk perlmonky (n=perlmonk@69-168-21-26.chvlva.adelphia.net) |
| 12:20.27 | op3r | voxee no more? |
| 12:26.47 | luke-jr_ | op3r: ? |
| 12:27.39 | op3r | luke-jr_: i went to voxee.com and now it seems voxee domain name expired |
| 12:27.48 | luke-jr_ | O.o |
| 12:28.01 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:32.52 | *** join/#asterisk Radude (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 12:34.13 | *** join/#asterisk ReD-MaN (i=daemon@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 12:38.02 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:40.42 | dpryo | Anyone played with chan_bluetooth? |
| 12:40.56 | dpryo | (HS) |
| 12:40.59 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 12:41.08 | dpryo | I have it all connected and paired, and calls make through.. but I can't answer them.. |
| 12:41.35 | darkskiez | its not been maintained in years |
| 12:41.40 | darkskiez | it never really worked |
| 12:41.40 | dpryo | I know ;) |
| 12:41.43 | dpryo | hehe |
| 12:41.53 | dpryo | The idea is cool though. |
| 12:42.02 | dpryo | I would like to use my handsfree in the office |
| 12:42.10 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 12:42.22 | darkskiez | dpryo, I used my headset via my laptop and x-lite instead. |
| 12:42.28 | dpryo | ;P |
| 12:42.38 | caio1982 | a guy from asteriskbrasil.org got it working using some hacks with recentes versions of asterisk, AFAIK, but with lots and lots of problems and misfunction |
| 12:46.52 | *** join/#asterisk coppice (n=chatzill@141.193.17.210.dyn.pacific.net.hk) |
| 12:52.40 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 12:55.01 | mocker | Can someone recommend software for screenpops? Basically something that just can bring up a URL when a call comes in? |
| 12:58.28 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:58.31 | Ariel_ | morning everyone |
| 13:02.03 | dpryo | darkskiez: Are you using bluetooth-alsa? |
| 13:02.18 | darkskiez | dpyro: No, OSX |
| 13:02.46 | darkskiez | although bluetooth-alsa was written by a mate at uni, he stopped development after he got a mac. |
| 13:03.08 | dpryo | hehe |
| 13:03.24 | dpryo | My handsfree won't work with my ibook... |
| 13:08.03 | *** part/#asterisk damned (n=vpol@prior.lanck.net) |
| 13:09.55 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140) |
| 13:10.52 | *** join/#asterisk Zhadnost (n=tom@cpc1-sout6-0-0-cust691.sot3.cable.ntl.com) |
| 13:11.58 | Zhadnost | is there anyone on here that uses Queues? |
| 13:13.04 | mocker | Zhadnost: I do. |
| 13:14.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:15.14 | RoyK | hm. is it possible to somehow list the current users in meetme? |
| 13:15.43 | mocker | RoyK: meetme list maybe? |
| 13:17.28 | Zhadnost | mocker> I'm trying to get the machine to divert to voicemail (well another queue, and then voicemail) if there is no-one logged into it |
| 13:17.47 | *** join/#asterisk kay2 (n=key2@LNeuilly-152-22-65-88.w193-251.abo.wanadoo.fr) |
| 13:17.55 | *** join/#asterisk Hmmhesays (n=Neg@72.24.105.126) |
| 13:18.06 | Hmmhesays | bang |
| 13:18.10 | mocker | Zhadnost: The simplest way would probably be queue timeouts.. |
| 13:18.11 | Zhadnost | in the queues.conf I have leavewhenempty = yes in the varius queue defs, |
| 13:18.28 | mocker | i.e. if nobody answers in 60 seconds in the queue, just continue down the dialplan. |
| 13:18.43 | Zhadnost | I've got timeout=30, which doesn't appear to exit the queue |
| 13:18.53 | mocker | Show me your Queue statement.. |
| 13:19.20 | AsteriskAlbania | sorry to bother again , I need a solution for ASTERISK -> RADIUS |
| 13:19.28 | AsteriskAlbania | if any one can help |
| 13:19.40 | *** join/#asterisk bigblockO2 (i=pplwro@ool-43504a4a.dyn.optonline.net) |
| 13:20.02 | AsteriskAlbania | if someone has implemented it , or have some information if it is possible or not |
| 13:20.15 | tzafrir | AsteriskAlbania, IT is known to work |
| 13:20.36 | mocker | AsteriskAlbania: I could swear I saw something about that on voip-info. |
| 13:20.57 | tzafrir | I have asked aabout this a while ago, and people refered me to JerJer. I still haven't asked him. |
| 13:20.57 | AsteriskAlbania | mocker: I have seen it also but there are a lot of broken links |
| 13:21.10 | *** join/#asterisk radude (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 13:21.16 | tzafrir | mocker, there is just one page, with outdated information |
| 13:21.19 | AsteriskAlbania | they say smth about a patch for asterisk but the link is dead |
| 13:23.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:26.23 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 13:27.37 | *** join/#asterisk pif (n=ldm@sabliere-nerim-ld.lncsa.com) |
| 13:30.35 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 13:35.55 | zoa | yo Cresl1n |
| 13:36.47 | coppice | zoa done any good talk-off tests lately? |
| 13:36.56 | zoa | yes |
| 13:37.18 | coppice | did you pass with distinction? |
| 13:37.24 | zoa | nothing happens |
| 13:37.24 | zoa | ; |
| 13:37.26 | zoa | :( |
| 13:37.39 | zoa | oh and you were right |
| 13:37.39 | coppice | what does nothing mean? |
| 13:37.43 | zoa | those people are very nice |
| 13:37.46 | zoa | they sent me a recording |
| 13:37.56 | zoa | never expected that |
| 13:37.58 | coppice | great. can you send it to me? |
| 13:38.24 | RoyK | hm. is it possible to somehow list the current users in meetme audiable to the users? |
| 13:39.16 | russellb | RoyK: no, but it wouldn't be too hard to add since there is already an option to have each user record it |
| 13:39.52 | RoyK | need to modify app_meetme, then :( |
| 13:39.54 | RoyK | oh well |
| 13:39.59 | Zhadnost | there's meetme list <confnumber> <usernumber> |
| 13:40.09 | Zhadnost | in CLI |
| 13:40.10 | RoyK | Zhadnost: console command iirc |
| 13:40.13 | kaz0358 | should there be any problem specifying one trunk group composed of two T1 spans in asterisk? in zapata.conf.. trunkgroup => 1,24,48; spanmap => 1,1,1; spanmap => 2,1,1; |
| 13:40.16 | RoyK | Zhadnost: doesn't help the users much |
| 13:42.28 | *** join/#asterisk Cubano (n=Cubano@unaffiliated/cubano) |
| 13:42.53 | russellb | RoyK: well you *could* parse that and use cepstral ... |
| 13:43.19 | RoyK | cepstral_ |
| 13:43.20 | RoyK | ? |
| 13:43.28 | RoyK | ah |
| 13:43.29 | russellb | a commercial text to speech engine |
| 13:43.35 | RoyK | yeah |
| 13:43.47 | coppice | now the big test. how to you pronounce cepstral? :-) |
| 13:44.05 | russellb | kep - strawl |
| 13:44.21 | MikeJ[Laptop] | strull? |
| 13:44.23 | coppice | give the man a cookie :-) |
| 13:44.25 | RoyK | not [sepstrahl] |
| 13:44.29 | russellb | MikeJ[Laptop]: yeah, that's better |
| 13:44.56 | RoyK | sypstreeahl |
| 13:45.10 | coppice | no. not sep |
| 13:45.16 | RoyK | why? |
| 13:45.42 | coppice | how do you pronounce spectral? |
| 13:46.10 | macTijn | spektral |
| 13:46.12 | kay2 | someone has ever used cirpack ? |
| 13:46.19 | zoa | yes |
| 13:46.25 | kay2 | zoa ? |
| 13:46.41 | zoa | for five minutes once |
| 13:46.43 | zoa | :) |
| 13:46.46 | zoa | looks cool |
| 13:46.51 | zoa | and expensive |
| 13:46.55 | kay2 | what kind of DB does it use ? |
| 13:46.59 | Damin | zoa: I'm speaking at the MySQL users conference today.. does queuestats work w/ MySQL as well as postgres? Or just postgres? |
| 13:47.08 | zoa | just postgres for the moment |
| 13:47.10 | zoa | but we will port it |
| 13:47.27 | zoa | and probably open source it |
| 13:47.32 | *** join/#asterisk inv_arp[work] (i=junya@adsl-10-153-159.mia.bellsouth.net) |
| 13:47.32 | coppice | so, rearrange spektral, and you get the answer for cepstral. its amazing how many people know what cepstral means, and still pronounce it sep... |
| 13:47.43 | ManxPower | I pronounce it the same way as the TTS engine does when I feed it "cepstral" |
| 13:48.11 | tzanger | I always thought it was sep too |
| 13:48.17 | Damin | zoa: I'll demonstrate it anyway and tell them that it is being prototyped on Postgres, but will be available for MySQL in the future.. |
| 13:48.32 | Damin | Anything else Mysql / asterisk related that I should mention? |
| 13:49.15 | zoa | amp ? |
| 13:49.22 | Damin | zoa: Yeah.. that's a good one.. |
| 13:49.39 | Damin | zoa: Althought I loathe amp... |
| 13:49.48 | zoa | i dont like it either |
| 13:49.50 | Damin | zoa: But I'll mention it anyway.. |
| 13:49.53 | *** join/#asterisk zaf (n=zaf@wsip-68-228-9-79.br.br.cox.net) |
| 13:49.53 | Cubano | what i need(hardware) for turn on VOIP with asterisk in my company? |
| 13:49.59 | Damin | Cubano: Yes. You do. |
| 13:50.00 | coppice | hey, wikipedia has a page on it - http://en.wikipedia.org/wiki/Cepstrum |
| 13:50.25 | Cubano | Damin: do u can help me? |
| 13:50.38 | kaz0358 | for testing purposes is it possible to have asterisk pickup when no extension is being sent from another switch? i'm getting Apr 26 08:49:16 WARNING[8073]: chan_zap.c:8381 pri_dchannel: Ring requested on unconfigured channel 0/1 span 1 Extension '' in context 'default' from '' does not exist. Rejecting call on channel 1/23, span 1 |
| 13:51.22 | kaz0358 | i have setup exten => s,1,Dial(500); exten => s,n,Hangup; in extensions.conf, but that doesn't seem to make much difference |
| 13:52.41 | Ariel_ | Cubano, we can help you but we need to know more on what you want to do? Also how much do you know about voip? What are you going to replace a PBX or add to it? |
| 13:53.37 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:55.02 | *** join/#asterisk pdunkel (n=pdunkel@213.235.192.21) |
| 13:55.22 | Cubano | Ariel_: i need turn on buildings of my company with asterisk.. |
| 13:56.11 | Ariel_ | Cubano, ok how much of asterisk do you know are you looking for a consultant to assist you with this? Where are you located? |
| 13:56.30 | Cubano | Ariel_: i know about voip and asterisk, but i dont know about hardwares.. |
| 13:56.39 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 13:56.45 | ghenry | That was cool |
| 13:57.02 | ghenry | just dialed in and got the nice American lady! |
| 13:57.06 | ghenry | First time setting up |
| 13:57.07 | Ariel_ | Cubano, ok hardware to connect what? to Asterisk |
| 13:57.17 | Cubano | Ariel_: yes.. |
| 13:58.47 | Cubano | Ariel_: i would like of turn on departaments in my company of different cities.. |
| 13:59.34 | Cubano | Ariel_: i need one server in each building? |
| 13:59.53 | Ariel_ | Cubano, it really depends on allot of things |
| 14:00.41 | Cubano | Ariel_: server with asterisk, its my question.. |
| 14:01.12 | Cubano | Ariel_: i think than need only one server in my matrix.. |
| 14:01.39 | jsharp | One server + a bunch of IP phones? |
| 14:01.50 | Ariel_ | Cubano, again I come up with the same reply more info is needed on what you have, what is at the other end and how you want to connect them together with. Also what pbx do they have already? Are you going to replace it? |
| 14:03.09 | Cubano | Ariel_: currently i have a frame-relay |
| 14:04.17 | marl | hi, can anyone give me some help getting a TDM400 working with CID in the UK on a BT phone line? i have followed the wiki instructions : http://www.voip-info.org/wiki/view/Asterisk+and+UK+Caller+ID : but i still dont get anything sowing in the log files sujesting a CID and i also dont get any CID forwarded to ether my SIP or my FXS extension |
| 14:04.19 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 14:05.11 | Ariel_ | marl, if you plug in a normal analog phone to the line does it get callerID? |
| 14:05.28 | Damin | <PROTECTED> |
| 14:05.29 | marl | yup, have full CID on all my lines |
| 14:05.39 | Cubano | Ariel_: i already one good infrastructure of network between my buildings.. |
| 14:06.18 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:07.36 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:10.14 | *** join/#asterisk famany (n=famany@81-208-88-100.ip.fastwebnet.it) |
| 14:10.22 | famany | hi at all |
| 14:10.28 | websae | hey famany |
| 14:11.05 | famany | hii |
| 14:11.13 | famany | i stay try to study this softwre |
| 14:11.15 | famany | :) |
| 14:11.34 | famany | . |
| 14:13.07 | famany | but i don't found a good manual for see after installation to do |
| 14:13.57 | ManxPower | ~thebook |
| 14:14.00 | jbot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 14:14.19 | famany | good let me see |
| 14:15.01 | vader-- | anyone here configure the cisco 7940G/7960G phones for asterisk? |
| 14:15.27 | vader-- | i downloaded the new sip firmware but i can't seem to find the default configuration files cisco talks about being in the zip? |
| 14:17.30 | Hmmhesays | god i love this song |
| 14:17.35 | Hmmhesays | too bad the singer is a dope |
| 14:18.04 | jsharp | I've got a bunch of 7940s talking to *. |
| 14:18.20 | famany | for read the online book i need login and password? |
| 14:18.50 | frawd | hi all, anyone has experience with Spanish E1 lines (framing, signalling, etc...) or has some clue how to configure them? |
| 14:19.14 | ManxPower | famany, no, |
| 14:19.29 | *** join/#asterisk lunk (n=lunk@negative-influence.com) |
| 14:19.31 | famany | thanks |
| 14:20.25 | lunk | is there a Postgres dial-plan function? I see many references to the MySQL one but have a pre-existing postgres DB |
| 14:21.04 | frawd | umm, nevermind for spanish stuff, but i have a simpler question: where are the dchan and bchan usually located on E1 lines? |
| 14:21.12 | Jacke | frawd: E1/crc/PRI |
| 14:21.21 | jsharp | D channel is on slot 16 |
| 14:21.29 | frawd | ony one D channel? |
| 14:21.32 | Jacke | frawd: d chan is 16, the rest is b. |
| 14:21.38 | papo | what's the difference between that password provided in the register-line of a sip proxy and the one provided with secret in the [sipproxy] section? |
| 14:21.40 | papo | of sip.conf |
| 14:21.45 | Jacke | why would you need any more? |
| 14:22.40 | ManxPower | papo, they will normally be the same. register => just tells the remote system what your IP address is. |
| 14:22.56 | frawd | Jacke: i saw in voip-info that a config (germany) would be bchan=1-15, dchan=16, bchan=17-31. Where is the 32nd channel?, is it number 0 or number 32? :) |
| 14:22.58 | papo | ManxPower: ok, thank you |
| 14:22.58 | ManxPower | the [sipproxy] section is what is used for calls. |
| 14:23.12 | papo | ManxPower: both for incoming and outgoing calls? |
| 14:23.22 | jsharp | Slot 0 on an E1 is reserved for framing & timing information. |
| 14:23.32 | jsharp | Slot 16 is ISDN signalling. |
| 14:23.33 | Jacke | frawd: in e1 you got one channel for synchronization, one d-chan and 30 bchans. |
| 14:23.40 | jsharp | The other 30 are voice channels. |
| 14:23.41 | ManxPower | papo, type=friend is incoming and outgoing calls. type=user is incoming only, type=peer is outgoing only. |
| 14:23.59 | frawd | okay! thanks, so no need to declare channel 0 then |
| 14:24.15 | jsharp | No. The framer chip on the board handles slot 0. |
| 14:24.20 | Jacke | it's inaccesible for the higher level protocols like PRI ;> |
| 14:25.21 | papo | ManxPower: oh I have type=peer. I guess that's the reason why I can't receive any calls through my sip proxy. Thanks a lot |
| 14:25.58 | frawd | okay, one more question, any weird configuration thing can happen to me when i'll connect that E1 (in spain)?, like overlap parameters or switchtypes, or can I just leave standard stuff like euroisdn and such (that i found for germany or italian configs)? |
| 14:27.11 | frawd | (sorry i really have no clue) |
| 14:28.40 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 14:30.11 | frawd | i guess i'll see once i'm there :) thanks very much anyway!! |
| 14:31.53 | coppice | without the hardware handling time slot 0 for you, there are no time slots at all :-) |
| 14:33.12 | frawd | why couldn't it have been handled by software? :p |
| 14:34.25 | coppice | bizzarely, I have actually done that |
| 14:34.40 | frawd | isn't hardware actually software (runned by the card's dsp)? |
| 14:34.46 | frawd | oh! |
| 14:34.56 | frawd | how? |
| 14:35.15 | coppice | what DSP? |
| 14:35.17 | frawd | did you build your own card? |
| 14:35.36 | coppice | many times :-) |
| 14:35.42 | frawd | i don't know, isn't there some type of DSP or CPLD in E1 cards? |
| 14:35.46 | brettnem | coppice is a wiz! |
| 14:35.52 | frawd | wow! |
| 14:36.05 | coppice | there is not DSP on most E1 cards |
| 14:36.34 | frawd | oh... some FPGA type of stuff then? |
| 14:36.38 | coppice | E1s are 1960 technology. they were designed to be ever so simple to make |
| 14:36.57 | brettnem | which is why the US doesn't use them. ;) |
| 14:36.59 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 14:36.59 | papo | ManxPower: Hm... I can't even get a very simple setup to work. my config is http://captu.lugs.ch/83 and I get http://captu.lugs.ch/82 for incoming calls. I don't see the problem since it's almost impossible to design a less complex dialplan... |
| 14:37.01 | frawd | directly with NAND doors? |
| 14:37.23 | coppice | T1s are even earlier, and were originally implemented with thermionic tubes |
| 14:37.43 | brettnem | what about N carrier? And FDM? :) |
| 14:37.51 | frawd | the things that warm your food in a microwave? |
| 14:38.06 | coppice | FDM goes back to the 30s |
| 14:38.22 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 14:38.23 | brettnem | fun |
| 14:38.29 | frawd | wow, i don't even know what it is... but how old are u coppice? |
| 14:38.53 | brettnem | just cause he knows about stuff from the 30s doesn't necessarily say anything about his age. :P |
| 14:38.54 | coppice | I didn't implement the first FDM stack :-) |
| 14:39.03 | brettnem | oh come on you where THERE |
| 14:39.05 | brettnem | ;) |
| 14:39.08 | coppice | thought I had some involvement with one of the last |
| 14:39.29 | coppice | a DSP FDM to E1 gateway |
| 14:39.43 | brettnem | I've done a bunch of WDM, but that's a different technology altogether |
| 14:40.07 | coppice | WDM is exactly the same technology, actually |
| 14:40.13 | brettnem | well different use |
| 14:40.16 | mut | wooohooo |
| 14:40.16 | brettnem | same concept |
| 14:40.50 | coppice | I love the term DWDM. its sounds so good, yet the dense part is actually a limitation, not something wonderful :-) |
| 14:40.58 | brettnem | WDM was fun to play with.. man it was expensive for a while |
| 14:41.15 | kmilitzer | Hello everyone ... how do I tell asterisk which indication to use in the dial command together with the r option? |
| 14:41.21 | frawd | we young generation started with GSM and DSL stuff :) makes me feel really unexperienced asking basic things about E1s to guys that even know older stuff |
| 14:41.35 | brettnem | coppice: I worked for a startup designing ROADMS in planer CMOS chips.. cool stuff |
| 14:42.06 | vader-- | anyone here configure the cisco 7940G/7960G phones for asterisk? |
| 14:42.07 | vader-- | i downloaded the new sip firmware but i can't seem to find the default configuration files cisco talks about being in the zip? |
| 14:42.32 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 14:42.32 | coppice | its amazing how much of this stuff exists. in the late 80s we were finding it hard to sell enoug 565M fibre systems to cover development, and nopbody wanted to invest in full scale development of the 2.4G prototypes stuff in the lab |
| 14:42.33 | jsharp | looking for Sipdefault.cnf? |
| 14:42.43 | vader-- | ya |
| 14:43.19 | brettnem | coppice: well, no one wanted a ROADM.. I don't blame them.. I built lots of FO networks with OADMS and wouldn't pay the same amount for a ROADM.. |
| 14:43.20 | frawd | anyway, thanks again for the tips old[er] folks ;) |
| 14:43.22 | vader-- | and the sipxxx.cnf |
| 14:43.32 | brettnem | frawd: thanks for making us feel old[er] |
| 14:44.00 | frawd | brettnem: it's because i'm only 21 :) |
| 14:44.22 | brettnem | frawd: so? I designed my first FO network at 19 |
| 14:44.23 | frawd | brettnem: it's a compliment being older than me |
| 14:44.30 | brettnem | oh please.. hah! |
| 14:44.40 | brettnem | look, you won't think that in a few years |
| 14:44.44 | frawd | brettnem: what's an FO? |
| 14:44.45 | kaz0358 | i finally figured out why i was getting no audio exchanged on calls being carried over the T1/PRI.. for some reason specifying a logicalspan in spanmap, hoses things up. kinda strange.. |
| 14:44.50 | frawd | jeje |
| 14:44.53 | coppice | SDH really cut the size of drop and insert kit. to do it with 565M PDH took 2 racks |
| 14:44.53 | brettnem | frawd: FO: Fiber optic |
| 14:44.57 | jsharp | vader--: Want a copy of mine? |
| 14:45.42 | brettnem | coppice: wow.. I remember pulling out our Fujitsu eqpt and Titan 5500 for 15454s and a Cienna Coredirector |
| 14:45.57 | brettnem | coppice: reduced like 40 racks of equipment to like 8. |
| 14:46.21 | brettnem | coppice: our 80 OC-12 Tellabs titan 5500 was over 27 bays |
| 14:46.23 | frawd | sounds fun |
| 14:46.36 | *** join/#asterisk Seyr (n=Seyr@grant254.grantgeo.com) |
| 14:47.12 | coppice | brettnam: did you ever play with PDH stuff? |
| 14:47.38 | brettnem | you know.. nortel responded in a RFC to us for this HDX beast of theirs, ever heard of it? The core switch was 4 like 54" bays and consumed 30KW |
| 14:47.48 | brettnem | brettnem: nah, just SONET |
| 14:48.48 | coppice | anything with SDH is tiny, cheap and simple compared to something similar with PDH :-) |
| 14:48.55 | brettnem | coppice: when they came to bid us, we said "Ok, 30KW, are you serious? Just to put that in perspective, that's about the amount of power a typical McDonalds consumes" |
| 14:49.28 | brettnem | and it needed structural reinforcement underneath it because of it's weight.. nuts.. |
| 14:49.34 | coppice | PDH was very short sighted, because the basic design never allowed for the say when everything was digital and could be sync'ed up |
| 14:49.53 | brettnem | was PDH async? |
| 14:50.03 | brettnem | it's been so long since my transport days.. :( |
| 14:50.14 | coppice | the P is for plesiochronous |
| 14:50.20 | brettnem | ah right |
| 14:50.26 | brettnem | I love that word |
| 14:51.22 | brettnem | I used to know a lot more about timing and syncronization.. use it or lose it.. |
| 14:51.29 | coppice | like asterisk is plesiofunctional :-) |
| 14:51.40 | brettnem | I worked with a guy for many years who designed the SBC sync network |
| 14:51.43 | brettnem | heh, yes it is |
| 14:51.47 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:52.26 | brettnem | I built a nice big transport network here in texas.. never seen anything like it.. |
| 14:52.28 | coppice | its a long time since i was developing that stuff |
| 14:53.39 | brettnem | that ciena coredirector was really cool |
| 14:53.52 | brettnem | 256 OC-48s in one bay.. it was revolutionary |
| 14:54.24 | coppice | revolutionary stuff quickly looks obsolete :-) |
| 14:54.28 | brettnem | haha right |
| 14:54.32 | brettnem | "bubble switch" |
| 14:54.55 | brettnem | when I went to the lab to test out the bubble switch, we performed a "sneeze test".. |
| 14:55.08 | brettnem | which, actually, it passed.. heh |
| 14:55.46 | *** join/#asterisk froguz (n=alvaro@200.104.155.95) |
| 14:55.48 | brettnem | ok, back to work.. <sigh> |
| 14:56.24 | lunk | is there a way to queue outbound calls? like an emergency weather reminder for employees? |
| 14:56.37 | Vagabond | is there a reason 'show queues' doesn't show realtime queues? |
| 14:57.24 | ManxPower | lunk, you mean like as documented in the Wiki and in the Asterisk source tree? |
| 14:57.58 | lunk | ManxPower: probably, but my inability to find it has hindered that process |
| 14:58.06 | lunk | ;) |
| 14:58.26 | ManxPower | lunk, sample.call in the source tree. |
| 14:58.29 | coppice | why do you have to remind people about bad weather? don't they notice its there? |
| 14:59.00 | lunk | coppice: to announce office closings depending on where the hurricane is going to hit |
| 14:59.51 | coppice | here typhoon warning just seem to magically permeate through society :-) |
| 15:00.25 | lunk | well yea, but we like to announce closings to keep employees safe |
| 15:00.25 | ManxPower | or use text messages |
| 15:00.25 | op3r | if I upgrade to 1.2.7 will the chanspy work properly? |
| 15:00.43 | ManxPower | op3r, what's wrong with it? |
| 15:00.55 | trelane_ | Does anyone know of a phone that supports on-hook paging? (IE I punch the extension and it activates the speakerphone) |
| 15:01.11 | ManxPower | trelane, Polycom, Cisco |
| 15:01.12 | coppice | well, the government does that for us, but how the message travels seems to be a magical process |
| 15:01.12 | op3r | ManxPower: choppy spying, sometime you can only hear 1 person on the the agent's line |
| 15:01.24 | ManxPower | op3r, what version are you using right now? |
| 15:01.32 | op3r | ManxPower:1.2.4 |
| 15:01.41 | *** join/#asterisk brockj49464 (n=brockj49@41.105.dhcp.hope.edu) |
| 15:01.44 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:01.51 | ManxPower | op3r, I believe there is a Changelog |
| 15:01.55 | trelane_ | ManxPower, 7960? and if so how do I interface to it? |
| 15:01.56 | op3r | ManxPower: I just dont want to you know upgrade the server because it is a production server |
| 15:01.59 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 15:02.16 | ManxPower | trelane, the mailing list archives are your friend. Love them. Hold them. Buy them flowers. |
| 15:02.24 | Seyr | op3r: a lot has been fixed between 1.2.4 and 1.2.7.1, memory leaks, etc |
| 15:02.38 | trelane_ | ManxPower, thought listserv preferred beer? |
| 15:02.40 | coppice | flowers over IP? |
| 15:02.41 | ManxPower | op3r, then why not download the latest tarball and read the changelog |
| 15:02.53 | jsharp | I wish you could set auto-answer for a 7940's line in the config file. |
| 15:03.06 | *** join/#asterisk axarob (n=jbash@spc1-barn1-5-0-cust117.asfd.broadband.ntl.com) |
| 15:03.08 | ManxPower | jsharp, um, I believe you can. |
| 15:03.32 | ManxPower | Granted, on the Ciscos you must dedicate a line apperance for that stuff |
| 15:03.36 | ManxPower | i.e. auto answer |
| 15:03.44 | jsharp | Not that I've seen, unless Cisco changed that in 8.0 |
| 15:04.09 | CrashHD | can you #include dir/* ? |
| 15:04.13 | op3r | ManxPower: anyway i should be fine as long as I dont do make samples right? |
| 15:04.14 | CrashHD | wildcard's etc...? |
| 15:04.15 | jsharp | You can allow auto-answer in the config file, but you still have to physically set a line appearance for auto-answer on the phone. |
| 15:04.28 | jsharp | And that setting goes away when you reboot the phone. |
| 15:04.41 | ManxPower | http://lists.digium.com/pipermail/asterisk-users/2003-October/016635.html |
| 15:05.13 | jsharp | Its available, but you have to turn it on at the phone through the menu. |
| 15:05.30 | Seyr | ManxPower: That was from 2003... and it said coming soon, lol |
| 15:06.16 | rculp | where can I find a good resource to troubleshoot inbound callerid? |
| 15:06.30 | rculp | ie, only numbers displayed |
| 15:06.33 | rculp | and no name info |
| 15:06.50 | ManxPower | rculp, what brand of phone |
| 15:07.03 | rculp | polycomm and eyebeam |
| 15:07.07 | Seyr | rculp: Did you ask your provider if they send the name? |
| 15:07.16 | ManxPower | rculp, what interface to the PSTN? |
| 15:07.16 | rculp | in the process of checking that |
| 15:07.19 | rculp | on hold with them |
| 15:07.25 | Seyr | a lot dont and force you to subscribe to the service |
| 15:07.37 | rculp | I have a pri going into a wildcard 110 |
| 15:08.08 | ManxPower | rculp, do a wait(1) as the first thing when a call comes in to make Asterisk wait for the callerid name info (it's not usually part of the setup messages) |
| 15:08.09 | kaz0358 | does anyone know if most cell phone carriers use the ANI for callerid sent to their handsets? |
| 15:08.30 | ManxPower | kaz0358, I doubt any of them do. |
| 15:08.36 | ManxPower | because you cannot hide ani |
| 15:08.46 | tzanger | kaz0358: Telus Mobility fills ANI with interesting numbers. sometimes it's the caller #, sometimes it's a number in the area the cell is calling from, sometimes it's blank |
| 15:08.59 | rculp | manx: ok, I'll go read where to put wait(1) |
| 15:09.01 | tzanger | generally speaking ANI is only ever right for incoming 800 calls, in my experience |
| 15:09.16 | tzanger | since for any other call (excepting collect calls) you aren't getting billed |
| 15:09.25 | kaz0358 | manxpower, no.. i mean on a call coming into the cell phone network.. do they ignore whatever is sent in callerid and instead use the ANI number to send to the cell phone |
| 15:09.29 | ManxPower | rculp, in the dialplan |
| 15:09.37 | rculp | ah ok |
| 15:09.47 | rculp | so at the top of the extensions.conf |
| 15:09.50 | trelane_ | ManxPower, could you point me to documentation on the polycoms? |
| 15:10.06 | ManxPower | kaz0358, no, because cell companies will get yelled at by the govt if they override the caller requested hiding of callerid |
| 15:10.24 | kaz0358 | tzanger, ahh okay.. well i'm getting something strange now. |
| 15:10.34 | ManxPower | trelane, www.polycom.com download the admin guide for your version of the firmware. Or USE GOOGLE |
| 15:10.51 | trelane_ | google? now why would I use THAT?! :0 |
| 15:10.54 | trelane_ | ;) even |
| 15:11.26 | ManxPower | Here is the super secret URL: http://www.google.com/search?hl=en&lr=&q=site%3Alists.digium.com+polycom+auto+answer&btnG=Search |
| 15:11.37 | ManxPower | but don't tell anyone else. |
| 15:12.00 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:12.38 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 15:12.48 | kaz0358 | tzanger, my setup--asterix over T1/PRI to avaya switch over DS3 to SBC to something to Sprint.. anyway, when i make a phone call from the asterisk box to my sprint cell phone, my cell phone is display some random number that is in this area instead of the number i'm sending in callerid. however, when i watch the call records stream by on the avaya switch i'm seeing the right callerid number.. strange |
| 15:13.20 | kaz0358 | err phone displays some random number |
| 15:14.02 | *** join/#asterisk Hali_303 (n=surfk@dsl5402ACB4.pool.t-online.hu) |
| 15:14.41 | kaz0358 | tzanger, when i make a call from asterisk box to a phone hanging off the avaya switch, callerid seems to be right.. although in the caller id name there is a weird character at the start.. maybe that has something to do with it |
| 15:14.53 | ManxPower | kaz0358, What is the callerid setting in Asterisk |
| 15:15.21 | ManxPower | i.e. what is the exact callerid= line in zapata.conf or sip.conf or whatever.conf |
| 15:15.21 | kaz0358 | callerid=Kurt Zoglmann <7855327001> in sip.conf |
| 15:15.40 | ManxPower | kaz0358, Hmm. That is actually correct format. |
| 15:15.47 | rculp | manx: my provider is saying that they send number first and then name |
| 15:15.49 | ManxPower | Lots of people use quotes, which is bad bad bad |
| 15:16.01 | ManxPower | rculp, so a Wait(1) should fix it. |
| 15:16.09 | coppice | can I quote you on that? |
| 15:16.18 | ManxPower | coppice, nope. |
| 15:16.27 | kaz0358 | manxpower, this is kinda odd.. maybe it is something odd on the avaya switch |
| 15:16.44 | stoffell | hm, has x-lite changed ports to 100XX ? (I opened up 800x but doesn't seem to work) |
| 15:16.46 | ManxPower | kaz0358, BTW, what cell company provides callerid name? |
| 15:17.18 | kaz0358 | manxpower, none that i know of.. but it is wierd that it is replacing 7855327001 with 7853132023 |
| 15:17.43 | ManxPower | kaz0358, is 7853132023 a number on your PRI? |
| 15:17.55 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 15:18.18 | kaz0358 | manxpower, nope.. we only have two exchanges.. 785-532 and 785-395 |
| 15:18.45 | rculp | Manx: I have this in my calltree |
| 15:18.47 | rculp | exten => s,2,Wait(1) |
| 15:18.56 | ManxPower | rculp, what is s,1 ? |
| 15:19.04 | *** join/#asterisk kay2 (n=key2@LNeuilly-152-22-65-88.w193-251.abo.wanadoo.fr) |
| 15:19.06 | rculp | exten => s,1,Answer() |
| 15:19.21 | ManxPower | Also exten => s will NEVER be called on a PRI |
| 15:19.27 | kay2 | someone has ever used wipphone's services |
| 15:19.27 | kay2 | ? |
| 15:19.34 | ManxPower | Also I told you to make Wait(1) the FIRST priority. |
| 15:19.45 | rculp | oh |
| 15:19.52 | rculp | I'm sorry, I missed that |
| 15:20.02 | ManxPower | The wait should be done before ANYTHING else. But as I said, on a PRI exten => s should never be run |
| 15:20.14 | kaz0358 | manxpower, thanks for the tid bit on extension s.. as a test i was trying to make that work. we were temporarily having an issue with our avaya switch sending anything as the destination number. |
| 15:20.25 | ManxPower | because exten => s means "I don't know what number the caller dialed". On a pri you always know what number the caller dialed. |
| 15:20.36 | brettnem | exten "s" is something the asterisk folks made up |
| 15:20.50 | ManxPower | they should rename it to exten => unknown |
| 15:20.59 | kaz0358 | maxpower, and if it doesn't know.. asterisk automatically tells the other side it won't accept it |
| 15:21.02 | brettnem | that "sounds" like exten => i |
| 15:21.14 | brettnem | shoulds return CC 3 |
| 15:21.22 | brettnem | which is like a SIP 404 |
| 15:21.33 | ManxPower | kaz0358, no. |
| 15:21.48 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 15:21.58 | ManxPower | If asterisk knows what number the user dialed, but doesn't have the number in the dialplan, then the call will go to exten => i |
| 15:22.22 | kaz0358 | manxpower, 785-313 is owned by united states cellular corp of kansas.. so probably not a sprint number nor ours |
| 15:22.31 | brettnem | could be ported. |
| 15:22.48 | ManxPower | If asterisk does not know what number was dialed, then exten => s will be run. IN FACT, in the real world, the only time exten => s is run is 1) when you use immediate=yes 2) when using analog POTS FXO or 3) using CT1 without DID service. |
| 15:23.10 | kaz0358 | manxpower, ahh okay.. |
| 15:23.29 | ManxPower | I wish I knew what everyone's fetish for the 's' extension is. |
| 15:23.47 | mut | ooo 9 & 10 news is covering the wireless in the cop cars we're doing in west branch |
| 15:23.49 | mishehu | "ooooh baby! want to see my s extension??" |
| 15:24.05 | ManxPower | Everyone things it's a catch all exten, or an invalid extension, or a don't catch anything exten, I think someone even said it cures cancer. |
| 15:24.24 | mishehu | ManxPower: I thought it actually caused cancer. |
| 15:24.28 | Seyr | the s extension cured my baldness |
| 15:24.38 | ManxPower | exten => s will never be used if you have DID service. |
| 15:24.50 | mishehu | "exten => s will get you laid" |
| 15:24.52 | kaz0358 | manxpower, i was just basing on this message i was getting earlier 'Apr 26 08:55:24 WARNING[8073]: chan_zap.c:8381 pri_dchannel: Ring requested on unconfigured channel 0/1 span 1 Extension '' in context 'default' from '' does not exist. Rejecting call on channel 1/23, span 1' |
| 15:25.15 | ManxPower | kaz0358, call your carrier. Report a problem |
| 15:25.26 | ManxPower | extension should NEVER EVER be '' |
| 15:25.29 | kaz0358 | manxpower, heheh.. the carrier is us |
| 15:25.43 | ManxPower | that indicates a problem on the PRI |
| 15:25.47 | kaz0358 | manxpower, it was a problem with the configuration of the avaya switch |
| 15:25.52 | ManxPower | Ah! UNCONFIGURED channel. |
| 15:25.53 | mishehu | kaz0358: you're not posting the did. |
| 15:27.10 | kaz0358 | mishehu, it was a random did.. which might or might not be valid |
| 15:29.52 | kaz0358 | manxpower, i'm don't think it was an unconfigured channel. it's just that they setup the switch to eat a certain number of numbers before handing it off to the asterisk trunk. they just happened to specify eating all of the numbers |
| 15:31.08 | brettnem | mmm.. me like eating numbers |
| 15:31.28 | darkskiez | Aww crap, even with dnsmgr on, hosts in your iax.conf register=> lines dont get refreshed. |
| 15:32.47 | rculp | ok, changed s,1 to be Wait(1) |
| 15:32.53 | rculp | and s,2 to be answer |
| 15:34.08 | *** part/#asterisk Seyr (n=Seyr@grant254.grantgeo.com) |
| 15:34.33 | ManxPower | rculp, Do you have analog FXO or CT1? |
| 15:35.42 | ManxPower | rculp, Perhaps you missed this: If asterisk does not know what number was dialed, then exten => s will be run. IN FACT, in the real world, the only time exten => s is run is 1) when you use immediate=yes 2) when using analog POTS FXO or 3) using CT1 without DID service. |
| 15:36.52 | rculp | ok, wasn't sure earlier if that was directed at someone else since you were helping multiple people |
| 15:36.55 | rculp | :) |
| 15:37.38 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:38.01 | rculp | we're not running fxo |
| 15:38.31 | rculp | our switchtype is ni1 due to a hardware compatibility issue |
| 15:38.34 | rculp | with our carrier |
| 15:38.41 | rculp | which we're considering leaving |
| 15:38.42 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 15:38.56 | rculp | since they have not been able to assist us with outbound callerid |
| 15:39.04 | rculp | even with them managing it |
| 15:39.07 | ManxPower | ni1 does not support calleridname in Asterisk |
| 15:39.13 | rculp | ok |
| 15:39.19 | rculp | that was my main suspicion |
| 15:39.23 | ManxPower | You should double check, but I think that is the case. |
| 15:39.28 | *** join/#asterisk brockj49464_ (n=brockj49@41.105.dhcp.hope.edu) |
| 15:39.59 | rculp | and to comment on what you said earlier |
| 15:40.03 | rculp | I have a line |
| 15:40.08 | rculp | that directs all calls to s,1 |
| 15:40.17 | rculp | based on the inbound # |
| 15:40.27 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:40.41 | rculp | exten => _302X.,1,Goto(calltree,s,1) |
| 15:41.26 | blebleble | i have a main asterisk box connected to my carrier and another asterisk box that is put in place at home just to connect my sips and soft phones directly to it to avoid nat issues, what is the best way to connect that asterisk box to that of my main one that is actually connected to the carrier, i have done some reading on iax2 trunks and or sip ones, anyone have any suggestions or can point me in the right direction |
| 15:42.06 | Ahrimanes | blebleble: i prefer IAX2 for that sort of connection |
| 15:42.31 | kaz0358 | manxpower, well i figured out why callerid wasn't being correctly sent out to my cell phone.. it turned out it was another avaya switch configuration problem. i was dialing all 10digits to reach my local cell phone number and for some reason that caused the avaya switch to send out a special trunk to sbc that is not callerid complinent.. which causes some strangeness |
| 15:42.58 | kaz0358 | err go out on a special trunk |
| 15:45.01 | blebleble | Ahrimanes: is http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers is that what i should be looking for / following for my situation |
| 15:45.44 | *** join/#asterisk mrq1 (n=mrq1@M499P015.dipool.highway.telekom.at) |
| 15:46.13 | *** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net) |
| 15:46.20 | Ahrimanes | blebleble: yes, that's a good place to start |
| 15:47.39 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 15:48.26 | MGSsancho | I wonder, is there like a profesional servie that will set up an Asterisk network? Cuz iif some admin needs to make a transition from some old crumy system to VOIP, large networks (1K - 20K phones) wont be for the faint of heart |
| 15:48.33 | MGSsancho | *service |
| 15:49.29 | darkskiez | digium do that :) |
| 15:49.49 | MGSsancho | hmmmm cool |
| 15:49.54 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 15:50.15 | jaiger | MGSsancho, I'm sure there are plenty in this channel and others that will do that for you |
| 15:50.25 | MGSsancho | lol |
| 15:50.31 | darkskiez | 20k phones tho, that'd probably need a SER expert too |
| 15:50.35 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 15:50.42 | *** join/#asterisk drfoomod2 (i=DrFooMod@ool-43501d9f.dyn.optonline.net) |
| 15:51.04 | jsharp | There's lots of Asterisk consultants out there. |
| 15:51.10 | MGSsancho | well not for me, i jus run asterisk on my linksys router. not like 1 call or 2 phones istaxing it |
| 15:52.51 | drfoomod2 | is anyone using dialogic boards w/ *? |
| 15:53.11 | MGSsancho | jsharp haahaha |
| 15:53.14 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 15:53.21 | snitt | lol |
| 15:54.07 | MGSsancho | put casters on it, use it as a footstool for the server room |
| 15:54.37 | MGSsancho | or door stop |
| 15:57.10 | MGSsancho | never ventered into the solutions of digiums site before. http://www.digium.com/en/products/software/abe.php thats cool. a whole binder with goodies |
| 15:57.40 | mut | yesss |
| 15:57.47 | mut | the mouse pad seals the deal |
| 15:58.00 | *** join/#asterisk saftsack (n=saftsack@p54A7CDB0.dip.t-dialin.net) |
| 15:58.34 | ManxPower | paying that kind of money for the manual kind of sucks |
| 15:58.59 | MGSsancho | includes spport and lience |
| 15:59.19 | MGSsancho | or i could just hangout here more |
| 15:59.26 | MGSsancho | rtfm works too |
| 16:01.00 | ManxPower | *nod* But most people just need a good manual, |
| 16:01.04 | mut | i like the feature list |
| 16:01.08 | mut | it's just every app it has |
| 16:02.30 | MGSsancho | i just looked at voip-info.org and clicked around |
| 16:03.17 | *** join/#asterisk saftsack (n=saftsack@p54A7CDB0.dip.t-dialin.net) |
| 16:03.20 | MGSsancho | then "oh snaps thats my setup" then i copied and pasted. then modified the relevent info |
| 16:03.37 | mut | yea it is |
| 16:03.43 | mut | all you need are like.. 2 pages |
| 16:03.47 | mut | document of applications |
| 16:03.50 | MGSsancho | haha |
| 16:03.53 | mut | and document of config files |
| 16:04.05 | ManxPower | mut, Yeah. Every link on the site brings you to the same 2 pages. |
| 16:04.17 | Ahrimanes | anyone know who hosts and runs voip-info.org ? |
| 16:04.27 | mut | some dude |
| 16:04.36 | coppice | the voip elves |
| 16:04.39 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:04.42 | Ahrimanes | ooh the elves |
| 16:04.56 | Ahrimanes | damnit.. told them to stay in the sweatshop.. |
| 16:04.59 | MGSsancho | cold just do a whois |
| 16:05.18 | Ahrimanes | whois gives domain owner, not sure to be the admin.. |
| 16:05.21 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:05.29 | mut | i offered to mirror it |
| 16:05.29 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:05.31 | MGSsancho | but i dun feel like opening putty |
| 16:05.34 | coppice | the spin gold into straw; make string from the straw; then use it to connect pairs of cans |
| 16:05.36 | mut | but he said he didn't want anyone to |
| 16:05.41 | mut | cause it was 'hard to do with wiki' |
| 16:05.44 | mut | o_O |
| 16:05.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:05.48 | DoktorGreg | if you go with default install of asterisk there is pretty thurough documentation right in the config files |
| 16:05.59 | MGSsancho | werd |
| 16:06.16 | ManxPower | pretty much ALL docs needed are included with Asterisk. The Book will help you even further. |
| 16:06.38 | ManxPower | The only time I use the Wiki is when I'm wanting to do something weird, or use it as a reference (tellabs, t-1 crossover pinouts, etc |
| 16:06.41 | *** join/#asterisk Nuttah (n=andrew@amber2.interdart.co.uk) |
| 16:06.46 | coppice | darkskiez: then you'll find serious management takes too much of your time, and you'll end up with the same thing :-) |
| 16:07.23 | Nuttah | has anyone had any issues with IAX between 1.2.6 and 1.0.7/9? |
| 16:07.25 | MGSsancho | tho i am looking into voip serviceproviders |
| 16:07.41 | MGSsancho | i just use voip phones internaly at my house |
| 16:07.44 | mut | I'm working on an upgrade -- hopefully will improve speed and ease of mirroring. |
| 16:07.44 | mut | Jim |
| 16:07.53 | darkskiez | coppice: even allowing renaming of pages and better tables i'd be happier |
| 16:07.55 | mut | that was back in september of 05 |
| 16:07.59 | Ahrimanes | hehe |
| 16:09.32 | MGSsancho | not a bad idea. do you we know te setup? just it just need another tincan, more ram, upgrade cpu? |
| 16:09.33 | Nuttah | bah this is driving me up the wall |
| 16:11.04 | Ahrimanes | MGSsancho: that's why i wanted to know who to contact :) |
| 16:11.29 | mut | jht@lj.net |
| 16:11.31 | *** join/#asterisk thock (n=thock@216.119.93.253) |
| 16:11.40 | Ahrimanes | mut: thanks :) |
| 16:12.57 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 16:14.44 | Nuttah | anyone here willing to lend their brain power on a iax registration problem? |
| 16:15.23 | MGSsancho | i duno |
| 16:15.51 | Nuttah | fair enough sancho :) Hoping someone can |
| 16:16.44 | MGSsancho | http://www.voip-info.org/wiki/view/Asterisk+readme+iax |
| 16:16.46 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 16:17.01 | Nuttah | thanks sancho.. but its not a common thing |
| 16:17.28 | MGSsancho | oh =( |
| 16:17.33 | ManxPower | OR /path/to/src/asterisk/docs/README.iax which will always be as up to date as the Wiki, and usually more up to date than the Wiki |
| 16:20.23 | Nuttah | i'll check out you latter link, I doubt it will help. at this point in time the iax client registers. however wont route more than 1 call to the external PBX. Personally think its a version problem. External PBX is 1.2.6, central PBX is 1.0.9 |
| 16:21.00 | MGSsancho | iax or iax2? something diffrent that they have diff ports |
| 16:21.06 | Nuttah | it lets one call in, then wont again until the external iax client re-registers |
| 16:21.08 | Nuttah | iax2 |
| 16:21.27 | Nuttah | as far as i'm aware asterisk only uses iax2 since 1.0 |
| 16:21.33 | Nuttah | or thereabouts |
| 16:21.43 | MGSsancho | IAX runs on port 5036 |
| 16:21.43 | MGSsancho | IAX2 runs on port 4569 |
| 16:21.43 | MGSsancho | IAX is NOT compiled in by default unless you edit channels/Makefile |
| 16:21.43 | MGSsancho | gnophone and tkPhone use IAX on port 5036 so will fail unless you add IAX in the Makefile |
| 16:22.08 | MGSsancho | from botom of wiki |
| 16:22.25 | Nuttah | heh yeah I know, but I appreciate it anyway :) |
| 16:22.31 | MGSsancho | :) |
| 16:22.38 | file | the old IAX isn't in SVN anymore |
| 16:23.50 | *** join/#asterisk Timmins (n=ryan_tim@64.201.63.41) |
| 16:23.54 | MGSsancho | 09/22/2003 |
| 16:23.58 | MGSsancho | bit outdated |
| 16:24.18 | MGSsancho | ya powers' suggestion is a good idea |
| 16:24.25 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 16:25.35 | Nuttah | its just so weird. its not a NAT problem, the 1.0.9 * box does not seem to attempt to call the remote 1.2.6 box even though there is an active session through NAT and it has the correct ports |
| 16:26.00 | Nuttah | does it once, then never again until it re-registers |
| 16:26.09 | Nuttah | which it also does not seem to do |
| 16:27.34 | Nuttah | cheers for the pointer Manx, but readme.iax contains nowt I have not already tried |
| 16:27.46 | MGSsancho | if it works oncehenits highly unlikely a network poblem |
| 16:28.00 | Nuttah | agreed |
| 16:28.01 | MGSsancho | stupid space bar >.< |
| 16:28.03 | thock | Can anyone help me out with some zapata.conf help |
| 16:28.25 | justinu|laptop | MGSsancho: you in LA? |
| 16:28.49 | MGSsancho | thock http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
| 16:28.54 | Nuttah | this is why I think its a 1.0.9 problem with 1.2.6 |
| 16:28.57 | MGSsancho | justinu|laptop yes |
| 16:29.38 | MGSsancho | looked at my hostmask? >_> |
| 16:29.45 | justinu|laptop | yeah |
| 16:30.13 | *** join/#asterisk Sandra78 (n=aerae@200.106.40.154) |
| 16:30.45 | thock | MGSsancho: yeah, the problem is i've got it configured but when i dial an external number all i get is "channel unavailable" |
| 16:30.52 | MGSsancho | then you must be from the aera the know that the entire LA aeraconects the the irvine node thingy (what ever the ppoe server is called) |
| 16:31.06 | MGSsancho | hmm |
| 16:31.43 | justinu|laptop | DSLAM |
| 16:33.05 | MGSsancho | ahh cool |
| 16:35.05 | *** join/#asterisk docelmo (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 16:35.08 | MGSsancho | thock, i dunno :( |
| 16:35.11 | thock | :/ |
| 16:35.14 | thock | It's cool |
| 16:35.23 | thock | most people apparently don't use sangoma cards |
| 16:35.55 | justinu|laptop | what's the sangoma problem? |
| 16:36.02 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:36.05 | thock | I can't get a channel to call out |
| 16:36.08 | mut | i jus got mine put in last night |
| 16:36.13 | mut | works great |
| 16:36.20 | mut | probly put it in play friday morning |
| 16:36.22 | justinu|laptop | mine works fine too |
| 16:36.33 | thock | okay, i'm only using one FXO card |
| 16:36.48 | thock | can i see one of your guys zaptel/zapata.conf? |
| 16:36.52 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 16:36.53 | justinu|laptop | mine's a T1 card |
| 16:36.58 | mut | yea, ditto, all i have is the quad pri a104d |
| 16:36.59 | justinu|laptop | so it won't apply |
| 16:37.01 | qseek | hey all |
| 16:37.01 | thock | i'm just using an A200 |
| 16:37.01 | thock | heh |
| 16:37.08 | mut | bbmin |
| 16:37.10 | mut | smoke break |
| 16:38.02 | Sandra78 | hi |
| 16:38.23 | Sandra78 | does anybody has worked with openbox? cards? |
| 16:38.25 | Sandra78 | http://www.openvox.com.cn |
| 16:39.12 | Sandra78 | <thock> how is A200? it has echo? |
| 16:39.21 | thock | Sandra78: wish i can tell you, i can't get it configured. |
| 16:39.42 | Sandra78 | <thock> why? what is your prob? |
| 16:39.45 | justinu|laptop | have you tried talking with sangoma? |
| 16:39.57 | thock | Yeah |
| 16:40.06 | thock | but they don't give out zaptel/zapata.conf support i wouldn't imagine |
| 16:40.12 | thock | and that's where the problem lies, i'm sure. |
| 16:40.13 | justinu|laptop | yeah they will |
| 16:40.33 | Sandra78 | do you get the lights on? |
| 16:40.41 | thock | on the back of the card? yeah |
| 16:41.21 | Sandra78 | maybe you have misconfigured something |
| 16:41.51 | thock | Most likely |
| 16:42.07 | Sandra78 | i recommend you that you can test it with AAH firt to know if the card works |
| 16:42.18 | thock | that's what i'm using right now actually |
| 16:42.27 | Sandra78 | ahh |
| 16:42.47 | Sandra78 | did you put genzaptel command? |
| 16:43.08 | thock | that's only for digium hardware |
| 16:43.33 | Sandra78 | but should work |
| 16:43.45 | thock | i'll give it a shot |
| 16:43.57 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 16:48.21 | *** join/#asterisk papo (n=mathias@adsl-177-161-fixip.tiscali.ch) |
| 16:48.35 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 16:49.39 | [TK]D-Fender | A@H won't work with an A200 out of the box... infact it'd take quite a big just to gett all the other bits in place to MAKE it work... |
| 16:52.04 | devel | hey all, i startup v 1.2.6 with 'asterisk -cvvvg' and get "asterisk ready", but never get the "prompt", is that normal, or is something awry? |
| 16:52.51 | [TK]D-Fender | devel : think so. Hit enter, do you get it then? |
| 16:52.52 | tzafrir | hmmm, try pressing Enter |
| 16:53.05 | tzafrir | Or writing a command, such as "show uptime" |
| 16:53.14 | *** join/#asterisk Nuxi (n=david@eder.us) |
| 16:53.21 | devel | nope. |
| 16:53.34 | devel | but i see stuff happen (like phone registering, etc.) |
| 16:53.50 | devel | and i can 'asterisk -r' in another console and works fine |
| 16:53.54 | Nuxi | How big of a configuration change is needed to upgrade from 1.0.9 to 1.2.7? (should my existing configs work?) |
| 16:53.55 | justinu|laptop | weird |
| 16:53.58 | devel | just not in my 'daemon' window.... |
| 16:54.07 | [TK]D-Fender | Nuxi : Depends on your configs... |
| 16:54.19 | devel | i am running as user 'asterisk', let me try as root. |
| 16:54.21 | [TK]D-Fender | Nuxi : Only real threat might be extensions.conf |
| 16:54.49 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 16:54.55 | Nuxi | hmmm. that's a pretty important file. |
| 16:55.00 | devel | nope, same thing. |
| 16:55.30 | [TK]D-Fender | Nuxi : Pastebin it and I'll take a look if you want |
| 16:55.37 | Nuxi | ok |
| 16:56.16 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@m810f36d0.tmodns.net) |
| 16:56.55 | Nuxi | http://pastebin.ca/51548 |
| 16:57.32 | [TK]D-Fender | Nuxi..... thats IT? |
| 16:57.55 | [TK]D-Fender | And oh boy that AGI may be screwed.... |
| 16:58.03 | Nuxi | yup. most stuff AGI. |
| 16:58.55 | [TK]D-Fender | Nuxi : no idea how much trouble that will cause... clearly your problems are well buried... |
| 16:59.52 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 16:59.59 | Nuttah | Nuxi, the main reason I havnt upgraded my 1.0.9 server is because of potential AGI issues |
| 17:00.21 | Nuttah | best bet is to build a 1.2.6/7 box and try your scripts on em |
| 17:00.31 | Nuxi | We are looking for sound quality improvements. Will an upgrade help? |
| 17:00.42 | Nuxi | Nuttah, which AGI toolkit are you using? |
| 17:00.43 | [TK]D-Fender | Nuxi : Doubtful... |
| 17:00.45 | Nuttah | I doubt it |
| 17:01.01 | Nuttah | its mostly functionality improvments |
| 17:01.20 | coppice | Sandra78: openvox were e-mailing all the well known developers a few months ago offering free cards. I didn't take any, and I don't know anyone else who did. I've never heard of anyone actually running them |
| 17:01.52 | Nuttah | although I know of a IAX issue between the 2 * versions.. because I've got one! :P |
| 17:02.39 | Nuttah | nuxi, good question, not one I can answer as I didnt do the AGI parts |
| 17:04.41 | devel | well, i'm stymied. i wouldn't be concerned, but safe_asterisk doesn't think asterisk starts, so just keeps looping (but asterisk is clearly running, can connect with 'asterisk -r') |
| 17:05.48 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 17:06.16 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 17:06.57 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 17:07.35 | *** part/#asterisk Samoied (n=Samoied@201-25-253-22.fnsce703.dsl.brasiltelecom.net.br) |
| 17:07.52 | Assid | is it me.. or does asterisk not have a number that you can call them on |
| 17:08.14 | Assid | errr |
| 17:08.14 | Assid | sorry |
| 17:08.16 | Assid | voipjet |
| 17:08.35 | file | they do not. |
| 17:09.31 | Assid | err.. any clue on how much moneybookers charges? |
| 17:10.58 | Assid | how can a phone co. have no way to contact them |
| 17:11.19 | file | it's not a phone company... it's a VoIP carrier who consists of a single person who does it as a hobby |
| 17:11.27 | justinu|laptop | heh |
| 17:12.03 | Assid | well.... these guys like jets' quality |
| 17:12.11 | Assid | so... i gotta get some credit in there |
| 17:12.27 | Nuxi | Is there a good list of AGI changes somewhere? |
| 17:15.10 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:15.15 | generalhan | hey all ! |
| 17:20.50 | *** part/#asterisk thock (n=thock@216.119.93.253) |
| 17:24.17 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 17:26.30 | Timmins | so can you use VoipJet to call pstn lines ? |
| 17:26.40 | xachen | Of course |
| 17:26.42 | xachen | thats what it is for |
| 17:26.47 | [TK]D-Fender | Nuxi : it will most likely have changed with the * diaplan commands they are related to and I wouldn't bet on good documentation |
| 17:27.00 | xachen | I'm dropping Voipjet though |
| 17:27.02 | xachen | too unstable |
| 17:27.10 | Timmins | what are you going with instead ? |
| 17:27.20 | xachen | I work for a voip company now |
| 17:27.22 | xachen | so go figure :) |
| 17:30.04 | [TK]D-Fender | My office has become one for me unknowingly :D |
| 17:32.33 | Timmins | do you need a did to make voip calls to pstn numbers ? |
| 17:32.45 | [TK]D-Fender | Timmins : nope... |
| 17:34.22 | *** join/#asterisk op3r (n=op3r@202.71.189.66) |
| 17:34.56 | op3r | does anyone know how to get a clear sip registration and to avoid getting unauthorized shit when you turn on ip phones? |
| 17:36.55 | Assid | i need to get some money into voipjet |
| 17:37.03 | Assid | any other carrier around the same rate.. with good quality? |
| 17:37.19 | Assid | maybe i can ask them to try another carrier for tme being |
| 17:38.41 | justinu|laptop | nufone! |
| 17:38.43 | justinu|laptop | :P |
| 17:38.51 | nahirean | hahah.. yeah right |
| 17:39.03 | op3r | voxee no more |
| 17:39.11 | Assid | somehow from their box to nufone isnt going too well |
| 17:39.22 | *** join/#asterisk dsfr_ (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 17:39.42 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 17:40.29 | fourcheeze | what's the correct way to return more than one variable from agi? |
| 17:40.47 | fourcheeze | I don't seem to be able to run set_variable() twice |
| 17:42.39 | *** join/#asterisk bsdfreak (n=alex@breakbeats.okkernoot.net) |
| 17:42.50 | bsdfreak | http://www.bsdfreak.org/modules/news/article.php?storyid=486 |
| 17:44.07 | fourcheeze | interesting |
| 17:44.08 | zaf | op3r: i'm having good luck with voicepulse connect right now |
| 17:44.09 | *** join/#asterisk ToTo (n=ToTo@host24-208.pool879.interbusiness.it) |
| 17:44.43 | vader-- | sip.conf is where i wanna register all my hard sip phones? |
| 17:44.44 | vader-- | correct |
| 17:45.01 | generalhan | yes |
| 17:45.54 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 17:46.01 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 17:46.35 | eKo1 | Has anyone had any problems getting Cisco 7960 phones registering with Asterisk? Mine doesn't register at all. |
| 17:48.54 | *** join/#asterisk zx225 (n=root@65.183.42.3) |
| 17:50.14 | *** join/#asterisk poisoner (i=poisoner@tauceti.poisoner.de) |
| 17:53.25 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:53.48 | generalhan | eKo1: i had issues with that a LONG time ago i MAY be of some help to you |
| 17:54.27 | shido6 | heh |
| 17:54.27 | generalhan | eKo1: can i see the sip.conf entries for those phones ? |
| 17:54.45 | jsharp | Make sure they're configured to register. |
| 17:54.58 | generalhan | lol |
| 17:55.20 | generalhan | my issue was that i set them up the same as my other sip phones, but for some reason they need a split entry in sip.conf |
| 17:55.27 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:55.28 | generalhan | one for peer and one for friend |
| 17:55.41 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 17:56.04 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:56.26 | Juggie | i never had that problem |
| 17:56.31 | Juggie | one entry was enough |
| 17:59.48 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 17:59.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:00.17 | jsharp | All mine are "friends", but I had forgotten to set 'proxy_register' in the sip.cnf file. |
| 18:00.26 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 18:00.59 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 18:02.03 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 18:02.10 | *** join/#asterisk jarrod (i=jarrod@juniperyour.net) |
| 18:02.30 | jarrod | HELLLLLLLLOOOOOOOOOOOOOOOOOOOOOO |
| 18:03.01 | file | yes? |
| 18:03.16 | jarrod | whars the t38 support! |
| 18:03.27 | DoktorGreg | how many digits are in an international number from US? |
| 18:03.38 | jsharp | DoktorGreg: Any number. |
| 18:03.45 | file | jarrod: on the bug tracker, in a branch, and hopefully soon to be in trunk |
| 18:03.52 | justinu|laptop | 011+country code+number |
| 18:03.57 | justinu|laptop | country code can be 1-3 digits |
| 18:04.01 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 18:04.02 | jarrod | file: for SIP ? |
| 18:04.07 | file | jarrod: yes |
| 18:04.10 | jarrod | nice |
| 18:04.31 | DoktorGreg | so my mask would be like, _011. for dialing international numbers? |
| 18:04.35 | sevard | DoktorGreg: exten = _011.,1,DIAL() |
| 18:04.35 | file | we talked about it on the developer call yesterday |
| 18:04.43 | file | and a recording of it will soon be available |
| 18:05.14 | *** join/#asterisk BrianR___ (i=brianr@setient-sucks.978.org) |
| 18:05.31 | *** join/#asterisk skydog (n=skydog@72.18.13.34) |
| 18:06.19 | skydog | hello. Somehwhere is a "fix" of sorts for changing kernel parameters for a CentOS 4.3 install of Asterisk. Can someone point me to it? |
| 18:06.30 | *** join/#asterisk nortex (n=nortex@snapper.titanspecialties.com) |
| 18:06.32 | DoktorGreg | what would be the 911 mask look like? |
| 18:06.33 | file | darn what is it... |
| 18:06.38 | jsharp | ~bug |
| 18:06.40 | jbot | bug is, like, n: A son of a glitch. An error in design or programming in hardware or software. Effects range from cosmetic errors to system crash and loss of data. See also Feature. |
| 18:06.43 | jsharp | Nopr. |
| 18:06.46 | file | ~centosbug |
| 18:06.47 | jbot | centosbug is probably a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 18:07.04 | skydog | thats it! thanks. |
| 18:07.29 | BrianR___ | SW echo cancellation never seems to keep up on my te406. Any adivce besides getting a te411p or sangoma a104d? |
| 18:07.36 | skydog | Does the so-called "patc" for latest version of asterisk achieve the same thing? |
| 18:07.36 | DoktorGreg | exten => _911,1,Dial(Zap/G1/${EXTEN}) look right? |
| 18:07.41 | jarrod | anyone want to by a TE405P card? |
| 18:07.46 | skydog | patch...sorry |
| 18:07.54 | tzanger | DoktorGreg: why have the _ there? you're not pattern matching |
| 18:08.09 | justinu|laptop | it'll still work, i think |
| 18:08.26 | DoktorGreg | I put the underscore there becaues i dont knwo what im doing of course! |
| 18:08.26 | tzanger | justinu|laptop: yes I think it will too |
| 18:08.33 | justinu|laptop | but it's superfluous |
| 18:08.48 | tzanger | DoktorGreg: I'd suggest a Playback(about-to-call-911) and Wait(5) in there myself, but it's up to you |
| 18:08.49 | nortex | Anybody using FOP with the astmanproxy? |
| 18:08.50 | justinu|laptop | the underscore is for pattern matching, doktorgreg |
| 18:08.54 | *** join/#asterisk volkerli (n=volkerli@port-212-202-198-104.dynamic.qsc.de) |
| 18:09.20 | BrianR___ | A lot of folks here accidentally call 911... :( |
| 18:09.34 | BrianR___ | I have a rule in my dialplan that emails the it department and receptionist whenever it happens.. |
| 18:09.39 | justinu|laptop | heh |
| 18:09.42 | websae | <-------is 911, makes me leave everytime someone does that |
| 18:10.28 | tzanger | websae: eh? |
| 18:10.45 | jsharp | paramedic/firefighter? |
| 18:10.52 | websae | both |
| 18:11.02 | websae | *volunteer |
| 18:11.10 | nortex | ? |
| 18:12.13 | websae | I am up til 4:00am in the morning, so that helps offset the crazy shock of my pager going off |
| 18:12.53 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 18:12.58 | websae | that way I'm always alert....and ready |
| 18:13.17 | jsharp | Mine would always go off about 30 minutes after I fell asleep. |
| 18:13.55 | websae | that's why I stay up now hehehe |
| 18:14.01 | websae | always went off around 2-3am |
| 18:14.38 | websae | them fire alarms will get you everytime |
| 18:14.45 | Hmmhesays | I kill the customers that call me late and bury their bodies |
| 18:14.54 | Hmmhesays | I don't get many calls that late anymore |
| 18:15.02 | websae | don't do that...then I'll get called out for that too lol |
| 18:16.23 | nahirean | why not create one |
| 18:16.31 | Hmmhesays | volunteer for what? |
| 18:17.10 | websae | emt/paramedic/firefighter/hazmat team |
| 18:17.17 | jsharp | Cause around here its an all paid county department. |
| 18:17.25 | jsharp | No volunteer reserves. |
| 18:17.53 | websae | jsharp: where are you at? |
| 18:18.35 | jsharp | Just outside of Atlanta, GA. I used to be in a department about 40 miles northwest of San Antonio, TX. |
| 18:19.09 | justinu|laptop | adrenaline junkies |
| 18:20.00 | jsharp | Family affair. My dad was the department chief. |
| 18:20.04 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 18:20.19 | ManxPower | I like vendors. You can be mean to them. http://pastebin.ca/51564 |
| 18:20.19 | websae | my grandpa was chief as well |
| 18:21.03 | brif8 | what would cause the odd phone to drop a call? it is random and seems unpredictable |
| 18:21.08 | justinu|laptop | ManxPower: is that for home sweet home? |
| 18:21.18 | ManxPower | justinu, yup. |
| 18:21.33 | justinu|laptop | nice |
| 18:21.43 | ManxPower | brif8, busydetect or callprogress enabled would do it. |
| 18:21.58 | brif8 | ManxPower: neither are enabled |
| 18:22.01 | ManxPower | justinu, also going to sell internet and phone service to others here. |
| 18:22.07 | justinu|laptop | good deal |
| 18:22.25 | justinu|laptop | you need that payphone too |
| 18:22.27 | *** join/#asterisk gursikh (n=guriskh1@adsl-209-30-245-180.dsl.hstntx.swbell.net) |
| 18:22.54 | sevard | Man, ControlPlayback is awesome. |
| 18:24.32 | brif8 | ManxPower: any other ideas? |
| 18:24.43 | ManxPower | brif8, no |
| 18:25.17 | brif8 | ManxPower: any suggestions what to monitor or look for |
| 18:26.34 | ManxPower | brif8, no |
| 18:26.45 | ManxPower | Other than maybe IRQ sharing |
| 18:28.13 | Hmmhesays | ok i seriously need some tunes... I'm thinking a little "avenged sevenfold" |
| 18:28.58 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-10-217.37-151.net24.it) |
| 18:29.04 | BladeRunner05 | hola all |
| 18:29.37 | vader-- | anyone know what setting would be off that is create this error in asterisk |
| 18:29.38 | vader-- | Apr 26 08:28:54 NOTICE[7183]: chan_sip.c:10886 handle_request_register: Registration from 'sip:272@172.13.0.2' failed for '172.13.0.100' - Not a local SIP domain |
| 18:29.58 | vader-- | im trying to athuenticate a cisco 7940G phone |
| 18:30.00 | *** join/#asterisk thock (n=thock@216.119.93.253) |
| 18:30.27 | ManxPower | vader--, what is the phone registering as? |
| 18:30.50 | vader-- | the user? |
| 18:30.56 | btm | Is there an asterisk function to send a SIP FLASH event (to a sipura 3k)? |
| 18:31.13 | ManxPower | btm, not that I know of. |
| 18:31.33 | brif8 | ManxPower: I would doubt IRQ sharing, most calls are between IP Phone and VoIP provider, they do the PSTN terminate side |
| 18:31.33 | vader-- | in the phone's config it's registering as line1_authname: "001759E558CE01" |
| 18:32.01 | ManxPower | vader--, And you have a [001759E558CE01] section in sip.conf? |
| 18:32.22 | vader-- | ya i did |
| 18:32.27 | vader-- | i changed it to the [272] |
| 18:32.34 | vader-- | but i did have that before |
| 18:32.36 | *** part/#asterisk volkerli (n=volkerli@port-212-202-198-104.dynamic.qsc.de) |
| 18:32.39 | ManxPower | well 272 != 001759E558CE01 |
| 18:32.45 | gursikh | what does it mean when the SIP phone has two ethernet jacks and they call it a switch? Why would i want that functionality? |
| 18:32.53 | ManxPower | [thestuffinhereiswhatasteriskexpectstheuseridtobe] |
| 18:33.15 | BladeRunner05 | what is this rtc: lost some interrupts at 1024Hz ? |
| 18:33.15 | ManxPower | gursikh, do you don't have to run one ethernet cable for the phone and one for the PC. |
| 18:34.24 | thock | Anyone here running an A200 Sangoma card? |
| 18:34.28 | thock | PM if you do~ |
| 18:34.28 | gursikh | Does it have any side effects doing this? |
| 18:34.39 | jsharp | Just male pattern baldness. |
| 18:35.35 | gursikh | Well, that may be an issue with my installation then jsharp |
| 18:36.08 | ManxPower | BladeRunner05, you didn't even to a simple google search about this problem, did you? |
| 18:36.28 | BladeRunner05 | ManxPower no I don't |
| 18:36.37 | ManxPower | gursikh, some phones only provide 10Mbps, not 100Mbps ports. |
| 18:36.56 | ManxPower | BladeRunner05, since you don't even do the basic google searches, I will have to put you on my /ignore list. |
| 18:37.28 | gursikh | ManxPower: thanks for the headsup, I'll keep that in mind while looking for phones |
| 18:37.28 | Hmmhesays | Oh be nice ManxPower: you know most people that wander in here can't read |
| 18:37.38 | LostFrog | lol |
| 18:38.33 | *** part/#asterisk Timmins (n=ryan_tim@64.201.63.41) |
| 18:38.40 | Hmmhesays | xxxxxxxxxxxxxxxxxxxx |
| 18:39.05 | justinu|laptop | hah |
| 18:40.56 | ManxPower | *grumble* My hands have blisters. |
| 18:41.21 | Hmmhesays | yeah mine are all callused up from all those dead hookers i've been chopping up |
| 18:42.05 | eKo1 | I got my Cisco 7960 to register with Asterisk but when I dial a number, I just get a fast busy and I don't see anything on the Asterisk CLI. What could be the problem? |
| 18:43.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-225-31.red.bezeqint.net) |
| 18:43.31 | jaiger | Hmmhesays, using this should cut down on the calluses... http://cuoz.org/gallery/main.php?g2_view=core.DownloadItem&g2_itemId=551&g2_serialNumber=2&g2_GALLERYSID=8cafc7caebfcd1a4c7acd52687196328 |
| 18:43.57 | justinu|laptop | anyone here using areski's asterisk-stat? |
| 18:44.05 | vader-- | ManxPower can you take a look at this |
| 18:44.10 | vader-- | http://pastebin.ca/51566 |
| 18:44.58 | ManxPower | vader--, "not a local sip domain" isn't the classic message, it's something different. |
| 18:45.12 | Hmmhesays | eko1 you are dialing a non existant extension |
| 18:45.14 | ManxPower | Perhaps you put in some other stuff in the phone config about domains or proxy realm or something |
| 18:45.21 | gursikh | can anyone suggest a good, affordable 24 port switch that works well with * ? I don't know If i will need POE or not because it seems to jack up the price quite a bit, and I don't know if the phones I want (polycom 301/501) even support it. |
| 18:45.30 | generalhan | eKo1: do you have the dial plan setup in the tftp directory ? |
| 18:45.56 | Hmmhesays | jaiger: fantastic |
| 18:46.10 | eKo1 | generalhan: Yes I do. |
| 18:46.20 | Hmmhesays | but I have to make sure the arms are intact, I like to sew them to my clothes and pretend i'm a spider |
| 18:46.42 | generalhan | what happens if you try and dial that phones extension ? |
| 18:46.43 | Hmmhesays | sip debug and you'll likely see the call is trying to go to an extension that doesn't exist in that context eko1 |
| 18:46.47 | vader-- | what goes into that dialplan.xml file? |
| 18:46.49 | generalhan | in the CLI i mean |
| 18:46.57 | eKo1 | generalhan: I get a fast busy. |
| 18:47.29 | generalhan | eKo1: Hmmhesays is right ... sip debug and ill bet you figure it out fast |
| 18:47.46 | Hmmhesays | yes my knowledge is vast, lol |
| 18:47.52 | generalhan | haha ! |
| 18:47.58 | Hmmhesays | now where the hell did I put that speak n spell |
| 18:48.04 | eKo1 | Hmmhesays: I get SIP/2.0 488 Not Acceptable Here. |
| 18:48.14 | generalhan | awww ... i wish i still had a speak and spell |
| 18:48.17 | jsharp | Codec issue, most likely. |
| 18:48.17 | justinu|laptop | codec mismatch |
| 18:48.23 | eKo1 | and SIP/2.0 407 Proxy Authentication Required as well. |
| 18:48.29 | Hmmhesays | oh man I have a treat for you generalhan |
| 18:49.08 | Hmmhesays | generalhan you on a windoze machine? |
| 18:49.15 | ManxPower | eKo1, "not acceptable" means the phone wants to use a codec asterisk is not allowing |
| 18:49.53 | justinu|laptop | or vice versa |
| 18:50.39 | ManxPower | You would think that after all this time eKo1 could diagnose this stuff on his own. |
| 18:50.46 | ManxPower | Maybe he's not done the VoIP piece before. |
| 18:51.08 | Hmmhesays | generalhan: http://www.zophar.net/Files/usass1.3b.zip |
| 18:51.30 | justinu|laptop | ManxPower: *shrug*, some people have it, some don't |
| 18:51.35 | eKo1 | ManxPower: But how come I don't get a message on the CLI that tells me there is no translation path? |
| 18:51.40 | justinu|laptop | no comment on this particular person tho :) |
| 18:51.42 | Assid | how do i kill 1 iax2 path? |
| 18:51.51 | ManxPower | eKo1, because it's failing LONG before then. |
| 18:52.12 | ManxPower | no translation path: Asterisk accepts the call and the codec, then makes a call and codec and cannot translate between them. |
| 18:52.20 | eKo1 | But the phone is set to ulaw and I have ulaw as one of the codecs to use in sip.conf |
| 18:52.32 | eKo1 | I don't get this. |
| 18:52.48 | ManxPower | not acceptable: Asterisk rejects the inbound call because all the codecs the phone is requesting are not allowed by asterisk's config. |
| 18:52.49 | nortex | I'm trying get Flash Operator panel to connect via the astmanproxy and having problems, anybody got this working? |
| 18:53.14 | ManxPower | eKo1, A sip debug will tell you what codecs the phone us requesting |
| 18:54.55 | *** join/#asterisk julien[re] (n=julien[r@AStDenis-103-1-13-159.w80-8.abo.wanadoo.fr) |
| 18:55.03 | julien[re] | hi there |
| 18:55.19 | julien[re] | i've got a small prob when MOH while dialing |
| 18:55.55 | eKo1 | Hmm...I don't see any requests for codecs!? |
| 18:56.05 | eKo1 | I guess that is the problem. |
| 18:56.08 | justinu|laptop | it's in the SDP |
| 18:56.14 | justinu|laptop | you need to learn to read the spec |
| 18:56.35 | justinu|laptop | i think asterisk will reject the call if the SDP doesn't have all the required elements |
| 18:56.39 | justinu|laptop | s/spec/protocol |
| 18:57.36 | eKo1 | I've never had so much trouble with a phone like I'm having with the Cisco phone. Maybe it is flawed. |
| 18:58.11 | julien[re] | is it possible to enable music on hold for every call made in a given context, without having to use the 'm' option |
| 18:58.13 | julien[re] | ? |
| 18:58.22 | justinu|laptop | the cisco phones work |
| 18:58.33 | justinu|laptop | yours is possibly broken, but I doubt it |
| 18:58.48 | vader-- | hmmm i wish i could figure mine out justinu |
| 18:58.57 | vader-- | it's saying it's not registering for some reason |
| 18:59.03 | vader-- | and then when i try to make a call from it |
| 18:59.13 | justinu|laptop | i'm not a cisco expert, but I know a lot of people who use them w/ asterisk |
| 18:59.13 | vader-- | asterisk does some stuff but i don't hear anything on the phone |
| 18:59.17 | vader-- | ya |
| 18:59.33 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 19:00.21 | ManxPower | Ahrimanes, yes, Corporate phone politics. |
| 19:00.59 | ManxPower | stupid auto complete |
| 19:01.53 | justinu|laptop | any PHP gurus around? |
| 19:02.08 | Nuttah | well I may have just found my first bug |
| 19:02.29 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 19:02.30 | ManxPower | One of our offices is having problems with their Toll Free (they just discovered it stopped working during Katrina). The Wicked Witch of the phone closet said all our Toll Frees are from BellSouth, so I contact our BellSouth rep and he said that the number is handled by MCI. So I toss the issue back to The Wicked Phone Witch for her to get me the required info for MCI to even find the account. |
| 19:02.57 | Nuttah | i.e. i think I know whats causing the iax reregistration problem between 1.2.7 and 1.0.9 |
| 19:03.02 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 19:03.19 | ManxPower | Nuttah, It's not a BUG it's a FEATURE! |
| 19:03.35 | Nuttah | heh I'm well aware of the 'Feature" aspect :P |
| 19:04.13 | Nuttah | its 1.0.9 asterisk scheduler changing the REGREQ from having a refresh of 60 |
| 19:04.17 | Nuttah | to having a refresh of 0 |
| 19:04.19 | brettnem | wicked witch of the phone closet eh? |
| 19:04.21 | paryl | hi everyone... i'm trying to install the zaptel drivers on an ubuntu system, and after compile i always get "FATAL: Error inserting zaptel (/lib/modules/2.6.12-9-k7-smp/misc/zaptel.ko): Invalid module format" |
| 19:04.35 | Nugget | http://zgp.org/~dmarti/images/feature.jpeg |
| 19:04.46 | brif8 | anyone using * in a production env. +/- 40 IP Phones |
| 19:04.49 | paryl | zaptel and libpri are the only problems i have :\ |
| 19:04.51 | Nuttah | that links bound to be naughty :P |
| 19:05.00 | brettnem | brif8: I max my boxes out around 200 phones |
| 19:05.28 | brettnem | Nugget: that car is facing the wrong way? |
| 19:05.37 | Nuttah | anyone know how to fiddle iax REGREQ refresh timeouts? |
| 19:05.43 | brif8 | brettnew: wooah. what are your specs if I may ask |
| 19:05.44 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 19:06.03 | *** join/#asterisk hinckc (n=hinckc@ool-43522ae9.dyn.optonline.net) |
| 19:06.12 | brettnem | brif8: nothing special.. P4 something or other.. meg of memory.. typical load has about 150 registered and 15-20 calls up |
| 19:06.21 | brettnem | all G.711.. it's really nothing special |
| 19:06.27 | brettnem | all SIP <-- key |
| 19:06.43 | *** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net) |
| 19:07.22 | tier_1 | is asterisk ever going to move to sqlite or some other db or is the plan to stay with db1 ? |
| 19:07.28 | brettnem | NO |
| 19:07.40 | russellb | brettnem: you don't know that |
| 19:07.45 | justinu|laptop | i vote for DB/2 |
| 19:07.51 | russellb | we probably will make that interface pluggable. |
| 19:07.53 | brettnem | I thought they didn't like the licencing model for sqlite |
| 19:07.57 | brif8 | ok I'm close 2 x Dual Core 2.8 GHz, 9G Ram , 50 Registered and 20 calls concurrent, all alaw and SIP |
| 19:07.58 | russellb | it's on the long list of things to do |
| 19:08.16 | justinu|laptop | pluggable interfaces... now there's a concept |
| 19:08.25 | brettnem | justinu: no, that's just plain crazy |
| 19:08.27 | brif8 | brettnem: how long has it been running |
| 19:08.31 | brettnem | russellb: well that's good to hear |
| 19:08.36 | tier_1 | do users can choose thier db |
| 19:08.55 | brettnem | brif8: pfftt. it only runs for like 20 hours at a time before it begins to deadlock.. haha.. but I don't think tat's because of the registrations.. |
| 19:09.01 | generalhan | if im on a single line phone WITH call waiting and i have the busy priority set up, will i even hear the callwaiting beep before it goes to that priority ? |
| 19:09.35 | brettnem | brif8: seriously, an older server running 1.0.9 runs for weeks just fine.. but 1.2.0 chokes pretty frequently if you ask it to do anything special |
| 19:10.01 | Nuttah | russell have you ever seen asterisk scheduler change a iax2 regreq refresh value from 60 to 0, mainly because between 1.0.9 and 1.2.6/7 it seems to occur |
| 19:10.04 | brif8 | brettnew: I'm using 1.2.6 and btw what is pfftt |
| 19:10.16 | russellb | Nuttah: no |
| 19:10.16 | generalhan | lol |
| 19:10.18 | brettnem | brif8: it's a sound I make. |
| 19:10.22 | generalhan | sound it out ! |
| 19:10.34 | Nuttah | arse, probably a good idea to log a bug fault |
| 19:10.43 | skydog | i dont know all the hoopla about pluggable interfaces..take a look at what it did to apache 2...what a convoluted mess... |
| 19:11.12 | brif8 | any ideas why the deadlock ? would it perhaps be wise to reload * asterisk nightly ? |
| 19:11.13 | Nuttah | unfortuntly a huge show stopper |
| 19:11.40 | brettnem | brif8: no, the whole "Avoided Deadlock" issue can be caused by anything and no one has any advice on it. |
| 19:12.06 | brettnem | "avoided deadlock".. what a joke |
| 19:12.14 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 19:12.19 | brettnem | it should say "I'm totally choking, help me die" |
| 19:12.31 | tier_1 | me thinks it should all move to postgress |
| 19:12.39 | fourcheeze | how do I return more than one variable from AGI? |
| 19:12.47 | brif8 | brettnew: ok. do you have the odd echo problems. I often get on a call and the other side says I'm fine but they sound real bad echo. (It is random not all calls) |
| 19:12.50 | Dr-Linux | many time i saw this "Avoided Deadlock" on my CLI, how can i avoid this stuff error? |
| 19:13.12 | brettnem | brif8: You hearing echo is typically a problem on the other side.. |
| 19:13.37 | *** join/#asterisk Jan_FL (n=trekkie@relativity.mediaweb-it.net) |
| 19:13.40 | brettnem | remember, the sound has to propogate.. typically |
| 19:13.40 | generalhan | can anyone help me figure out what to do to stop callwaiting on some phones now that incominglimit is depreciated ? |
| 19:13.50 | brettnem | call limit? |
| 19:13.55 | generalhan | yea |
| 19:13.56 | brif8 | right: I got that from the great book. I'm just wondering why the random side and what I can do about it |
| 19:14.05 | brettnem | or turn off call wait on terminating UA |
| 19:14.25 | brif8 | random side -> random one sided |
| 19:14.36 | generalhan | i cant disable callwaiting from the phone or phone cfg files. so i used to use incominglimit=1 in sip.conf and that stopped the callwaiting. but now i cant use it anymore so i need to find a new solution |
| 19:14.51 | nahirean | yo, anyone here ever use a Sonicwall - is there a way to disable bloody SPI? |
| 19:15.07 | nahirean | i recently aquired one, but it's dropping chunks of audio during calls |
| 19:16.00 | fourcheeze | am I right in thinking that it's only possible to return a single variable from AGI? |
| 19:16.02 | fourcheeze | if so how do people return more? |
| 19:16.04 | brettnem | generalhan: the offical way to do it is with callgroups |
| 19:16.07 | brif8 | nahirean: what's SPI I have a sonicwall |
| 19:16.16 | brettnem | stateful packet inspection |
| 19:16.17 | generalhan | brettnem: what do you mean ? |
| 19:17.00 | brettnem | generalhan: show function group |
| 19:17.25 | brettnem | oops. group needs to be capitalized |
| 19:17.31 | brettnem | show function GROUP |
| 19:17.35 | brettnem | and show function GROUP_COUNT |
| 19:17.38 | generalhan | brettnem: heres my issue. i have a call queue that dials all my reps' phones. but if they are on the phone they get to listen to the call waiting beep until someone answers the call form the queue. so i need to turn it off so they dont have to listen to it anymore |
| 19:17.39 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:17.50 | brettnem | oh I see. |
| 19:18.13 | harryvv | is it possible with a windows client to send a fax though a asterisk box and out a pstn line? |
| 19:18.18 | brif8 | nahirean: what about firewall -> advanced -> Connections ? |
| 19:18.22 | brettnem | you should be able to disable call waiting on the UA |
| 19:18.33 | brettnem | or use the local channel driver with GROUP and GROUP_COUNT |
| 19:18.34 | generalhan | so i was thinking that if in the dialplan i put the 101 priority in there for the busy respons and just said Hangup then it would shut it off |
| 19:18.38 | brettnem | or call-limit I think still works |
| 19:18.45 | brettnem | no that won't work.. |
| 19:18.54 | brettnem | if the UA doesn't disable call waiting, then it won't 487 you |
| 19:19.07 | generalhan | VA ? |
| 19:19.11 | brettnem | and you should avoid using n+101 priorities |
| 19:19.14 | brettnem | FWIW |
| 19:19.45 | brettnem | the way the asterisk developers like to totally deprecate functions and make them backwards incompatible, you're likely to break your scripts by upgrading someday |
| 19:20.20 | eKo1 | What should be used instead of n+101 priorities then? |
| 19:20.22 | *** join/#asterisk dVoka (n=dVoka@CPE-69-76-38-12.wi.res.rr.com) |
| 19:21.02 | generalhan | brettnem: i dont know what to do ... my reps are yelling at me becuase of this beep and i have to figure out a way to stop it. whatever it takes |
| 19:21.35 | brettnem | generalhan: I have a contract rate |
| 19:22.19 | generalhan | brettnem: my boss is the cheapest man you will EVER meet. there is no way i can convince him to pay someone to look into this for me |
| 19:22.26 | dVoka | Do either of you host multiple VoIP clients with Asterisk? My business partner is considering leasing me a rack unit his web hosting company isn't using at ANet in Chicago but ... kinda pricey - I think they're paying around $4000/mo for four servers. |
| 19:22.29 | brettnem | generalhan: well, sounds like your in a pickle!!! |
| 19:22.35 | *** join/#asterisk fusie (n=fusie@gmt-8.dsl.intention.nl) |
| 19:22.36 | generalhan | brettnem: i am |
| 19:22.49 | brettnem | generalhan: look, read my messages, the answer is in there...wooOOOOooOOOooOooOo |
| 19:23.17 | brettnem | dVoka: sounds like a redicuous ripoff |
| 19:23.43 | brettnem | eKo1: most new fuctions have some sort of status variable that is set.. like ${DIALSTATUS} or ${QUEUESTATUS} |
| 19:23.44 | dVoka | brettnem: well ... they "used to have" some high end clients - web hosting stuff, lots of hits. |
| 19:23.45 | devel | well, i looked through the strace output of my no-prompt-console, and still have no clue what's going on... |
| 19:23.53 | brettnem | eKo1: see asterisk/docs/README.variables |
| 19:24.00 | dVoka | but yes ... dang pricey!! Am thinking it may be overkill. |
| 19:24.10 | brettnem | not overkill, overcost |
| 19:24.32 | eKo1 | brettnem: so what you're basically saying is that I should set a status variable. |
| 19:24.48 | brettnem | eKo1: no, check a status variable.. many functions set them already. |
| 19:25.10 | eKo1 | Can I have my own custom status variable? |
| 19:25.18 | brettnem | exten => _XXX,1,Dial(SIP/${EXTEN})... exten=> _XXX,1,Goto(${DIALSTATUS},1) |
| 19:25.40 | Nuttah | ok, well hopefulyl I havnt made any faux pas, no bug reported under IAX that matches mine asfaik. Bugg filed |
| 19:25.51 | brettnem | eKo1: you miss the point.. Dial will someday NOT goto pri+101 for failure. it'll just leasurly goto the next pri and set ${DIALSTATUS} |
| 19:26.01 | dVoka | brettnam: Good point ... I suppose 1&1, Dreamhost and others may offer same performance/bandwidth at lower cost. |
| 19:26.34 | brettnem | dVoka: I imagine you can probably order your own DS3s to your house (multiple) install a Leibert cooling system and a couple of racks yourself for that cost. :) |
| 19:26.55 | eKo1 | brettnem: yes, I understand that point. What I want to know is, if I make a custom module, should it do priority jumping or set a custom status variable? |
| 19:27.09 | brettnem | custom module? like app_myapp.so? |
| 19:27.21 | eKo1 | Yes, exactly. |
| 19:27.40 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 19:27.49 | brettnem | I think if you are making your own apps, the typical convention today is to NOT do 101 jumping unless option 'j' is specified, and definately set a status variable -> ${MYAPP_STATUS} |
| 19:28.10 | brettnem | and of course, document it in the app, much like app_dial and app_queue |
| 19:28.37 | eKo1 | That is exactly what I needed to know. Thanks. |
| 19:28.55 | brettnem | np |
| 19:29.00 | ManxPower | I believe priority jumping will be removed in 1.4. At least that's what I've heard. |
| 19:29.02 | dVoka | brettnem: thanks for the recommendation on the multi-DS3s to the house ... would be nice! ;) |
| 19:29.03 | brettnem | what app are you maing? |
| 19:29.11 | brettnem | s/maing/making/ |
| 19:29.21 | brettnem | ManxPower: I hate deprecation |
| 19:29.29 | brettnem | deprecate(deprecation) |
| 19:29.43 | ManxPower | brettnem, I don't. Keeps things cleaner. |
| 19:29.46 | brettnem | dVoka: I miss the days of having a T1 at home.. |
| 19:30.02 | brettnem | ManxPower: the option should be there I think.. having code that isn't backwards compatible sucks |
| 19:30.31 | ManxPower | brettnem, the policy I've heard is depricate a feature in a major release, remove it in the next major release. |
| 19:30.31 | brettnem | dVoka: when you build the transport network yourself, having a T1 is a nice perk. of course, we needed it at home to test with, right? :) |
| 19:30.44 | brettnem | ManxPower: barf |
| 19:30.54 | brettnem | I understand why it's done.. but still |
| 19:31.15 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-235-102.telkomadsl.co.za) |
| 19:31.16 | dVoka | brettnem: right ;) |
| 19:31.41 | brettnem | dVoka: SBC had to trench my backyard to add more wire into my house. :) |
| 19:31.43 | eKo1 | brettnem: I have a whole bunch of custom apps. |
| 19:31.52 | brettnem | eKo1: fun :) |
| 19:33.13 | *** join/#asterisk twilson (n=terry@69.17.122.227) |
| 19:34.32 | brettnem | anything good? |
| 19:35.21 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 19:35.22 | *** join/#asterisk stoffell_h (n=stoffell@d5153F9E0.access.telenet.be) |
| 19:36.54 | ManxPower | BellSouth added another telephone pole to support the 25 pair cable they brought in when I ordered my 2 lines. |
| 19:36.57 | eKo1 | I have two apps right now that do authentication and authorization and they're using priority jumping if something goes wrong. |
| 19:37.52 | eKo1 | I guess I'll have to recode them so that they use a status variable. |
| 19:39.12 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 19:39.43 | rculp | manxpower: just wanted to say thanks earlier for confirming the ni1 vs ni2 callerid issue |
| 19:39.54 | rculp | my vendor is working to fix that with me now |
| 19:42.35 | harryvv | manx, asterisk is not capable reciving fax request from a windows fax client then sending the fax out? |
| 19:44.13 | jaiger | harryvv, what do you mean? asterisk isn't a fax server |
| 19:44.19 | harryvv | yea okay |
| 19:44.27 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 19:44.37 | harryvv | wonder if i can load hylafax onto the asterisk box and it will work. |
| 19:44.45 | harryvv | got this in my email |
| 19:44.48 | harryvv | Dear Customer, |
| 19:44.48 | harryvv | <PROTECTED> |
| 19:44.48 | harryvv | their carrier has terminated their service. |
| 19:44.48 | harryvv | <PROTECTED> |
| 19:44.48 | harryvv | fill out the port requrest in the LNP section of the control panel and |
| 19:45.03 | harryvv | I wonder who that may be. |
| 19:45.39 | brettnem | hmmmmmmmmmmmmmm oh I wonder.. |
| 19:46.11 | brettnem | it's only a matter of time.. people should be asking why the service was terminated.. |
| 19:46.38 | fourcheeze | ok, who is an AGI god around here |
| 19:46.41 | justinu|laptop | i've been asking |
| 19:46.42 | fourcheeze | c'mon own up |
| 19:46.59 | fourcheeze | or even average at AGI :-) |
| 19:46.59 | justinu|laptop | brettnem: i'd love to know the answer |
| 19:47.22 | *** join/#asterisk IceManRISK (n=kart@201.66.56.199) |
| 19:47.43 | brettnem | justinu: I'm sure it's about access compensation.. it's going to happen to everyone.. so lookout! |
| 19:48.23 | fourcheeze | I just need to return more than one variable from my AGI script |
| 19:48.28 | fourcheeze | it can't be that hard can it? |
| 19:48.43 | justinu|laptop | i know about recip comp |
| 19:48.45 | brettnem | fourcheeze, return it in a parsable format.. use cut |
| 19:48.57 | fourcheeze | brettnem: you are kidding me? |
| 19:49.06 | fourcheeze | you mean it really can't return more than one variable |
| 19:49.09 | brettnem | justinu: it's similar to recip comp... but a different kind of intercarrier compensation |
| 19:49.13 | justinu|laptop | oh |
| 19:49.19 | brettnem | fourcheeze: I don't know if it can,I was just making a recommendation |
| 19:49.32 | fourcheeze | well at the moment it's looking like I might need it |
| 19:49.52 | brettnem | fourcheeze: why not just set a bunch of variables? |
| 19:50.03 | fourcheeze | because it craps out when I try to set the second variable |
| 19:50.25 | tier_1 | res_sqlite wont build on centos/asterisk@home/freepbx |
| 19:50.29 | tier_1 | grrrr |
| 19:50.36 | fourcheeze | unless there's a different command to return multiple variables |
| 19:50.36 | brettnem | justinu: basically the incumbants want to charge CLECs access rates for terminating calls from the "internet".. And they threaten CLECs with huge bills when they (the CLECs) attempt to termiante such traffic locally. |
| 19:50.43 | tier_1 | make: execvp: /usr/src/asterisk-1.2/contrib/scripts/astxs: Permission denied |
| 19:50.44 | tier_1 | make: *** [install] Error 127 |
| 19:51.38 | brettnem | CLECs don't want to mess with SBC and rather tell the ITSPs to get lost than deal with it.. |
| 19:51.53 | brettnem | er s/SBC/the incumbants/ |
| 19:52.07 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:52.14 | brettnem | anyway, it'll happen to you someday.. thank your local PUC |
| 19:52.32 | brettnem | and I mean the royal you.. I'm pointing to #asterisk |
| 19:53.24 | tier_1 | grrr |
| 19:53.28 | websae | where's a good place to download a country code list? |
| 19:53.37 | mercestes | brettnem: Great.... |
| 19:53.41 | brettnem | ~google country code list |
| 19:53.59 | mercestes | I shoulda went for the business degree...lol |
| 19:54.04 | *** part/#asterisk thock (n=thock@216.119.93.253) |
| 19:54.16 | brettnem | mercestes: heh, yeah, were all F-D! wahoo! |
| 19:54.18 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:57.17 | ketanp | Has anyone encountered problems using DBput to store 20,000 entries? |
| 19:58.24 | paryl | why would i get "Invalid module format" after compiling the zaptel drivers? |
| 19:58.24 | brettnem | HAHAHAHA |
| 19:58.39 | brettnem | I can't imagine doing 20,000 of anything with asterisk |
| 19:58.58 | brettnem | althougt.. I have done over 50,000 CNAM queries aparently.. |
| 19:59.14 | tier_1 | grrr |
| 19:59.19 | tier_1 | frick |
| 19:59.29 | tier_1 | res_sqlite wont build |
| 19:59.32 | jaiger | ketanp, 20,000 in what timeframe? |
| 20:00.44 | ketanp | jaiger: Load them upon asterisk starting, for example |
| 20:00.59 | Hmmhesays | damn |
| 20:01.26 | eKo1 | tier_1: check the permissions on astxs |
| 20:01.37 | sevard | Does anyone know the cache flag for when MoH plays shoutcast streams? I have broken content and need to cache it |
| 20:01.42 | ManxPower | I'm starting to think it would be easier just to get a VGA->RCA/Composit converter than try to find a cheap video card with RCA out |
| 20:01.43 | *** part/#asterisk dVoka (n=dVoka@CPE-69-76-38-12.wi.res.rr.com) |
| 20:02.06 | sevard | Actually it sounds fine for about 3 seconds then starts breaking up and skipping around |
| 20:02.24 | tier_1 | make install |
| 20:02.26 | tier_1 | /usr/src/asterisk-1.2/contrib/scripts/astxs -autoload -install res_sqlite.so |
| 20:02.26 | tier_1 | make[1]: *** No rule to make target `apps_env'. Stop. |
| 20:03.49 | tier_1 | ok fixed |
| 20:04.05 | tier_1 | I had to link asterisk-1.2 to asterisk |
| 20:04.10 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 20:04.21 | Ariel_ | Afternoon all |
| 20:04.26 | SplasPood | Anyone here been hit by the nufone 866 .... fuckary? |
| 20:04.33 | tier_1 | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 20:04.34 | tier_1 | See `config.log' for more details. |
| 20:04.34 | tier_1 | make: *** [.sqlite] Error 1 |
| 20:04.43 | op3r | hahahahahah 60% of the call centers here cant dial US |
| 20:04.48 | op3r | at the moment |
| 20:04.54 | brettnem | son of a..... http://go.connect.yahoo.com/go/tv |
| 20:05.32 | ManxPower | I don't suppose anyone has a couple of test numbers in Europe I can dial from the USA? The Wicked Phone Witch claims international dialing isn't working. |
| 20:06.00 | snitt | ManxPower: i can setup one for you |
| 20:06.10 | ManxPower | snitt, what country? |
| 20:06.15 | snitt | hungary |
| 20:06.19 | ManxPower | I would prefer a telco one. |
| 20:06.28 | snitt | its a pstn |
| 20:07.41 | jaiger | ManxPower, doesn't suse have offices in europe? |
| 20:07.50 | ManxPower | no idea. |
| 20:07.50 | jaiger | ManxPower, http://www.novell.com/offices/emea/ |
| 20:07.53 | jsharp | +44 20 8734 2600 That's the sales number for a web hosting company I worked for. |
| 20:07.56 | ManxPower | jaiger, thanks |
| 20:08.38 | snitt | :P |
| 20:08.41 | jaiger | ManxPower, +353 1 605 8000 is the dublin office number for novell |
| 20:09.05 | snitt | +36 1 4474000, a telco's IVR |
| 20:09.30 | brettnem | I always look for holiday inns in other countries... that typically gives you an ACD |
| 20:11.16 | tier_1 | shit |
| 20:11.32 | tier_1 | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 20:11.44 | tier_1 | this is on centos |
| 20:16.57 | *** part/#asterisk BrianR___ (i=brianr@setient-sucks.978.org) |
| 20:21.41 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 20:23.13 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
| 20:23.37 | *** join/#asterisk luzeefa (n=luzeefa@p54AE92D8.dip0.t-ipconnect.de) |
| 20:23.46 | *** part/#asterisk luzeefa (n=luzeefa@p54AE92D8.dip0.t-ipconnect.de) |
| 20:26.24 | sevard | I'm streaming MoH off a shoutcast server and it plays fine for about 3 seconds but then starts skipping around everywhere and I get res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! all over the place |
| 20:27.14 | Jacke | sevard: ztdummy loaded? |
| 20:27.15 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
| 20:28.04 | *** join/#asterisk redondos (n=redondos@190.48.41.29) |
| 20:28.21 | redondos | If I add "channel => 1" to zapata.conf, asterisk doesn't start. Why? |
| 20:28.44 | redondos | Oh, RTFLogs! |
| 20:28.49 | redondos | Sorry :/ |
| 20:29.06 | sevard | Jacke: I have a TDM400P w/ 2.6.x iirc I don't need ztdummy |
| 20:29.09 | ketanp | Is there some mechanism to run either an external program or call an asterisk app when the server is started or reloaded? |
| 20:29.25 | sevard | Jacke: also regular mp3s work great, streams are the ones that are fuxx0red |
| 20:29.47 | redondos | Apr 26 17:29:16 ERROR[13009] chan_zap.c: Signalling requested on channel 1 is FXO Kewlstart but line is in FXS Kewlstart signalling |
| 20:29.50 | redondos | Apr 26 17:29:16 ERROR[13009] chan_zap.c: Unable to register channel '1' |
| 20:29.58 | redondos | What does that mean? |
| 20:30.24 | sevard | redondos: exactly what it says. |
| 20:30.46 | redondos | sevard: I can't be more precise than the logs :/ |
| 20:30.58 | ketanp | redondos: it's saying the line is set to fxO but is using fxS |
| 20:31.04 | sevard | redondos: that's pretty much all you need |
| 20:31.51 | redondos | ok, asterisk starts now. thank you both :D |
| 20:33.41 | *** join/#asterisk MGSsancho (n=weee@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
| 20:33.44 | *** join/#asterisk MGSsancho (n=weee@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
| 20:35.48 | Dr-Linux | sevard: do you have any problems with your TDM interfaces? i.e. noise/echo/horrible ? |
| 20:36.18 | sevard | Dr-Linux: It's crystal clear. |
| 20:36.37 | *** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net) |
| 20:36.45 | *** join/#asterisk ToTo (n=ToTo@host24-208.pool879.interbusiness.it) |
| 20:37.10 | ManxPower | Does anyone know much about the ! pattern match? |
| 20:37.18 | ManxPower | The extensions.conf seems rather unnclear |
| 20:40.20 | *** join/#asterisk kimosabe (n=kimosabe@dsl-201-133-195-203.prod-infinitum.com.mx) |
| 20:40.33 | kimosabe | does anyone have asterisk running over freebsd yet ? |
| 20:41.34 | mog_work | yes kimosabe |
| 20:41.37 | mog_work | its in ports |
| 20:41.52 | file | an older version mind you |
| 20:42.05 | mog_work | use the linux |
| 20:42.06 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 20:42.12 | kimosabe | mog work what version compiles it good without flaws ?? |
| 20:42.21 | mog_work | i think up to trunk doe |
| 20:42.22 | mog_work | s |
| 20:42.24 | mog_work | and 1.2 |
| 20:42.48 | kimosabe | mog work what version of freebsd with ?? |
| 20:43.39 | kimosabe | 4.11 ? |
| 20:43.58 | mog_work | all we really require is gcc 3 or newer, and some other minor stuff |
| 20:44.05 | mog_work | it should build on just about anything |
| 20:44.08 | eKo1 | Anyone here has a Uniden UIP200 phone? I've configured mine with the latest firmware. I can call it, but I can call from it. |
| 20:44.10 | mog_work | just try it |
| 20:45.05 | kimosabe | oki thanks man |
| 20:45.08 | redondos | What does this mean, guys? Apr 26 17:44:56 NOTICE[13069] chan_sip.c: Unable to create/find channel |
| 20:45.39 | mog_work | ill give you a hint |
| 20:46.00 | mog_work | <PROTECTED> |
| 20:46.13 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 20:46.32 | redondos | mog_work: Come on, if I asked it's because I don't know what I'm doing wrong. :) |
| 20:46.48 | mercestes | <PROTECTED> |
| 20:46.49 | jpablo | hey people, any idea where can i actually download the grandstream firmware? in their page i just see a tftp address, but i need to download the files cause i want to upgrade a lot of telephones |
| 20:46.59 | mog_work | it didnt find the channel you were trying to connect to or it couldnt create a channel |
| 20:47.15 | redondos | Where are channels created, mog? |
| 20:47.31 | mog_work | in the channel core |
| 20:47.40 | mog_work | what were you trying to do |
| 20:48.11 | pigpen | anyone using iaxmodem with asterisk for faxing? |
| 20:49.50 | Shoragan | how can i select a context based on callerid without having asterisk wait for more digits? |
| 20:50.37 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 20:51.02 | ManxPower | Shoragan, exten _NXXNXXXXXX/5045551212,1,Goto(realcontext,${EXTEN},1) |
| 20:51.03 | *** join/#asterisk Abydos313 (i=abydos31@ppp-71-138-7-212.dsl.irvnca.pacbell.net) |
| 20:51.07 | ManxPower | where 504 is the callerid |
| 20:51.59 | Shoragan | ManxPower, i used _X./_531349[234].,1,Goto(default,${EXTEN},1) but it waits after receiving digits before executing the goto |
| 20:52.03 | SplasPood | Would anyone mind taking a look at a GotoIfTime of mine and telling me what, if anything, is wrong? |
| 20:52.18 | Shoragan | if i use include => default it works without a delay |
| 20:52.52 | ManxPower | Shoragan, . will wait until DigitTimeout before proceeding. That's why I don't recommend using . in a pattern |
| 20:53.01 | *** join/#asterisk Assid (n=assid@59.183.26.213) |
| 20:53.20 | Shoragan | but i don't know how many digits need to follow |
| 20:53.32 | ManxPower | Shoragan, then you'll have to put up with the delay |
| 20:53.52 | ManxPower | If you don't know how many digits follow, how do you expect Asterisk to know that. |
| 20:54.09 | Shoragan | hmmm, i seemed simple enough, because i only need to look at the callerid (which is know immediately) |
| 20:54.16 | DoktorGreg | try pressing the # key |
| 20:55.07 | ManxPower | DoktorGreg, Do you really think that will work? # is not part of the pattern match. |
| 20:55.15 | ManxPower | Shoragan, I assume you are using a ZAP channel? |
| 20:55.18 | Shoragan | yes |
| 20:55.29 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 20:55.32 | Shoragan | and the callerid can be trusted |
| 20:55.34 | DoktorGreg | on ata # key means send the dial string right now |
| 20:55.42 | ManxPower | DoktorGreg, he's not using an ATA |
| 20:55.54 | Shoragan | # does not work with my older pbx |
| 20:56.03 | ManxPower | and no, it does not. ATAs can be configured to either use # as a regular digit or use it as an end of dial indicator |
| 20:56.48 | Shoragan | is there a way to begin executing GotoIf immediatley after entering the context? |
| 20:57.01 | ManxPower | Shoragan, you can use immediate=yes in zapata.conf. That will send the call directly to exten => s,1,whatever when the person picks up the handset, then you can use Goto a DISA extension in the context you need. |
| 20:57.18 | Shoragan | hmm |
| 20:57.23 | DoktorGreg | by default on every ata i have tried so far, # sends the dial string... |
| 20:57.31 | ManxPower | Shoragan, NOTHING happens until a pattern of some kind is matched in the dialplan unless you have immediate=yes |
| 20:57.53 | dlynes | DoktorGreg: that's just the default, and it may not be the default on all atas...just the ones you're familiar with |
| 20:58.02 | Shoragan | :( |
| 20:58.04 | ManxPower | DoktorGreg, Wben was the last time you tried to dial #72 as a special services code on an ATA. |
| 20:58.40 | Shoragan | in that context the dialed number is irrlevant |
| 20:58.49 | Shoragan | only the callerid |
| 20:58.56 | DoktorGreg | I just use ## to transfer on an ata that # also sends dial string about 5 minutes ago.. |
| 20:59.21 | ManxPower | Shoragan, to you, but Asterisk always considers the dialed number first, before everything else. UNLESS you have immediate=yes or the call is coming in on a port that CANNOT get digits, like an analog FXO |
| 20:59.31 | Alystair | Hi DoktorGreg :D |
| 20:59.39 | *** join/#asterisk thock (n=thock@216.119.93.253) |
| 20:59.41 | DoktorGreg | heya |
| 21:00.13 | Shoragan | ok, i'll try with immediate=yes |
| 21:00.13 | Alystair | I ended up making a nice layout diagram for gursikh: http://techmight.com/office-pbx-layout.png |
| 21:00.44 | Alystair | he decided against PoE/UPS |
| 21:01.05 | DoktorGreg | heya |
| 21:01.12 | DoktorGreg | my son wants to make a movie |
| 21:01.22 | ManxPower | Alystair, other than the NetGear, it looks fine, as long as you don't expect the VoIP over the internet to be reliable. |
| 21:01.30 | DoktorGreg | what is a good plot for a 4 year old boy, a dog and a big yard as the sound stange? |
| 21:01.42 | redondos | I've got an FXS card. I use it for testing purposes, so when I'm not using it I don't want it to answer when an incoming connection is detected. What would you suggest? I tried using a non-existant context in zapata.conf, that didn't work: it always falls back to the 'default' context. |
| 21:02.13 | ManxPower | redondos, Don't you mean an FXO card? |
| 21:02.27 | jsharp | exten => s,1,Wait(99999999) |
| 21:02.29 | Alystair | ManxPower: it's not my design, just did him a favour and made a pretty visio design |
| 21:02.34 | jsharp | It'll never answer. |
| 21:02.50 | Alystair | er not my network design |
| 21:02.55 | ManxPower | jsharp, it will never answer if you unplug the phone line from the server either. |
| 21:02.55 | redondos | ManxPower: Yes, FXO. |
| 21:03.07 | ManxPower | ~fxofxs |
| 21:03.10 | jbot | it has been said that fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 21:03.10 | redondos | jsharp: good idea. thanks. |
| 21:03.16 | ManxPower | ~ecfo |
| 21:03.18 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 21:03.23 | redondos | I wanted a software solution, though ManxPower. |
| 21:03.27 | jsharp | ~eieio |
| 21:03.31 | jsharp | Darn. |
| 21:03.38 | redondos | jsharp: Oh, but that affects the whole contextr. |
| 21:03.44 | ManxPower | redondos, then use jsharp's idea |
| 21:03.46 | redondos | jsharp: I still want to be able to use the context via SIP. |
| 21:03.59 | justinu|laptop | interesting... i hadn't heard of ECFO before, but I've experienced it |
| 21:04.08 | ManxPower | redondos, Asterisk cannot take a line out of service on a temporary basis. |
| 21:04.19 | ManxPower | You can take a line out of service, but you have to restart asterisk to get it back again |
| 21:04.44 | redondos | ManxPower: Restarting asterisk is bareable. |
| 21:04.50 | ManxPower | juanjoc, I made up the term, but not the symptop. |
| 21:04.51 | redondos | ManxPower: How'd I got abou ttaking it out of service? |
| 21:04.58 | ManxPower | redondos, zap destroy 1 |
| 21:04.58 | redondos | s/abou t/about / |
| 21:05.02 | ManxPower | or something like that. |
| 21:05.22 | juanjoc | ManxPower: Excuse me? What term? |
| 21:05.23 | redondos | And how do I add it back? |
| 21:05.29 | ManxPower | "zap destroy channel X" |
| 21:05.51 | ManxPower | redondos, 1) stop and start asterisk or 2) unload chan_zap.so then load chan_zap.so |
| 21:06.07 | Dr-Linux | ManxPower: can you tell me how can i send some DTMF digits after the call bridged? |
| 21:06.15 | ManxPower | Dr-Linux, no. |
| 21:06.16 | Alystair | DoktorGreg: I figured out VoIP but now I have to learn about asterisk :( |
| 21:06.21 | Dr-Linux | like after 10 seconds |
| 21:06.31 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 21:06.49 | ManxPower | BUT, the amazing command "SHOW APPLICATION DIAL" can! It's faster than a speeding armadillo! |
| 21:07.20 | Dr-Linux | ManxPower: i tried that, but that didn't work as i want |
| 21:07.40 | ManxPower | In fact, do a "show applications" at the Asterisk CLI some time and you'll stumble on the secret treasrue trove of docs, know only to the few, the proud, the Astmasters! |
| 21:08.05 | eKo1 | assmasters.... |
| 21:08.37 | Dr-Linux | ManxPower: i'm asking here and directly you, bcoz i already made some search on it, but no luck so far |
| 21:08.48 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 21:09.05 | SexyKen | Hey guys -- does anyone here have experiencing with getting queue statuses to a web page? |
| 21:09.14 | dlynes | Dr-Linux: saydtmf? |
| 21:09.18 | ManxPower | Dr-Linux, I don't know why it doesn't work for you, but I use D() all the time. |
| 21:09.42 | ManxPower | many times you need to wait .5 or 1 second before dialing your digits, of course. |
| 21:09.57 | *** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net) |
| 21:10.03 | redondos | ManxPower: very well, that works like a charm! thank you :) |
| 21:10.22 | Dr-Linux | i know its Dial command, but in same pirority i want to send some DTMF digits after the call bridged |
| 21:10.34 | *** join/#asterisk hal1on (i=hal@shut.your.yapzap.com) |
| 21:10.36 | *** join/#asterisk adept_ (i=adept@hektik.org) |
| 21:11.19 | adept_ | gooooooooo nufone :( |
| 21:11.31 | *** join/#asterisk popvoxdave (n=popvoxda@c-68-49-123-118.hsd1.md.comcast.net) |
| 21:11.36 | *** part/#asterisk popvoxdave (n=popvoxda@c-68-49-123-118.hsd1.md.comcast.net) |
| 21:12.02 | Dr-Linux | dlynes: i don't want my caller to enter their code when IVR asks, i want their code should be sent after the 10 seconds wait as call bridged |
| 21:12.13 | redondos | What ports should I forward on a router in which my asterisk server is behind from so that I can connect to it using SIP over the Internet? |
| 21:12.16 | vader-- | <PROTECTED> |
| 21:12.24 | redondos | Can the RDP port range be restricted? |
| 21:12.28 | vader-- | i need some help getting mine connected to asterisk |
| 21:12.32 | redondos | s/RDP/RTP/ |
| 21:13.01 | Dr-Linux | vader--: ask your question |
| 21:13.02 | dlynes | Dr-Linux: yeah...i just didn't know whether saydtmf would work after the call was bridged or not...i was just offering it up as an alternative |
| 21:13.23 | ManxPower | Dr-Linux, And D() dial option does not work for you? |
| 21:13.28 | *** join/#asterisk AK01 (n=none@nv-65-173-88-46.sta.sprint-hsd.net) |
| 21:13.36 | AK01 | If I have a IP system that does not provide a SIP trunk am I going to need a PRI? |
| 21:13.43 | vader-- | i have a cisco 7940G phone running sip 8.2 firmware when it boots up i get an error in the asterisk console that says Apr 26 11:11:14 NOTICE[7551]: chan_sip.c:10886 handle_request_register: Registration from 'sip:272@172.13.0.2' failed for '172.13.0.100' - Username/auth name mismatch |
| 21:13.48 | vader-- | i am able to dial with the phone |
| 21:13.57 | dlynes | vader--: no |
| 21:13.57 | Dr-Linux | dlynes: but that can be done in next prirorty, so if call is bridged then that will never go for next priroty |
| 21:14.00 | *** part/#asterisk thock (n=thock@216.119.93.253) |
| 21:14.01 | vader-- | and asterisk then says -- Executing Playback("SIP/172.13.0.2-0818fe48", "demo-abouttotry") in new stack |
| 21:14.01 | vader-- | <PROTECTED> |
| 21:14.09 | vader-- | but i don't hear anything on the phone |
| 21:14.09 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 21:14.24 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 21:14.52 | Dr-Linux | vader--: check your username/password |
| 21:14.54 | dlynes | Dr-Linux: ah...that's what i thought, but i was grasping at straws |
| 21:15.39 | dlynes | Dr-Linux: i've never actually tried to do anthing with asterisk after the call was bridged |
| 21:15.46 | vader-- | i have but i don't seem to see any problem |
| 21:16.11 | adept_ | any voip providers that are a bit more stable than nufone? I need a toll free number port :( |
| 21:16.11 | dlynes | vader--: if there wasn't a problem with the username/password, it wouldn't be giving you that error |
| 21:16.12 | Dr-Linux | dlynes: i found a way, that works but don't know how to put specific time |
| 21:16.51 | vader-- | here are my configs |
| 21:16.52 | vader-- | http://pastebin.ca/index.php |
| 21:16.56 | Dr-Linux | dlynes: simple, when you dial my company DID number and you want to talk to me, so you dial my extension |
| 21:16.59 | dlynes | Dr-Linux: and manx's D() method in the dial command doesn't work? |
| 21:17.05 | vader-- | http://pastebin.ca/51606 |
| 21:17.06 | ManxPower | you cannot use SendDTMF to do what you want, you can only use D() |
| 21:17.22 | Dr-Linux | what could be done if you directly reach to me after dialing DID |
| 21:17.26 | ManxPower | If it's an analog FXO or CT1 port, then you can just append waits and more digits on the dial string. |
| 21:17.50 | dlynes | ManxPower: is there a way to lengthen the 'w' pause in the dial string? |
| 21:18.01 | ManxPower | dlynes, put more of them in |
| 21:18.09 | dlynes | ManxPower: ah...yeah...never worked for me |
| 21:18.13 | dlynes | ManxPower: tried that |
| 21:18.21 | Dr-Linux | dlynes: yes that's why i tried, i put there "ww" |
| 21:18.44 | Dr-Linux | but i don't how much time wait in "ww" |
| 21:18.55 | ManxPower | English: Asterisk dials 555-1212, waits for 4 seconds, then dials 666. extensions.conf Dial(Zap/g1/5551212,,D(wwww666)) |
| 21:19.01 | vader-- | ok i think i figured that out |
| 21:19.06 | ManxPower | sorry, that would be. |
| 21:19.11 | ManxPower | English: Asterisk dials 555-1212, waits for 4 seconds, then dials 666. extensions.conf Dial(Zap/g1/5551212,,D(wwwwwwww666)) |
| 21:19.14 | dlynes | ManxPower: Was an issue for me at one site because i needed to increase the amount of time 'w' waited because the pause needed after dial tone (from *72) was quite large at one site |
| 21:19.18 | vader-- | right now im still using the demo context for testing |
| 21:19.31 | vader-- | when i dial extension 500 asterisk saysing it's executing playback |
| 21:19.37 | vader-- | but i don't hear anything on the phone |
| 21:19.51 | dlynes | ManxPower: Oh...I was trying Dial(SIP/*72w6042663152) |
| 21:20.10 | ManxPower | dlynes, that would ONLY work on an analog FXO |
| 21:20.14 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 21:20.23 | dlynes | ManxPower: ah...your method works on sip as well? |
| 21:20.24 | adept_ | and is anybody here a nufone user who may have asked more recently about service restoration? |
| 21:20.28 | Dr-Linux | ManxPower: but how many seconds wait in per "w" ? |
| 21:20.34 | vader-- | any ideas what would cause that? |
| 21:20.35 | ManxPower | dlynes, D() should work on all technologies. |
| 21:20.40 | ManxPower | Dr-Linux, w = .5 second |
| 21:20.41 | dlynes | ManxPower: cool...thanks |
| 21:20.48 | Dr-Linux | if i want 10 seconds wait, how many w i need |
| 21:20.49 | Dr-Linux | aaa |
| 21:20.51 | Dr-Linux | nice |
| 21:21.03 | sevard | Does anyone know how the hell to cache with mpg123? I think that's the stutter I'm having with my shoutcast MoH |
| 21:21.26 | ManxPower | sevard, what makes you think you can cache with mpg123? |
| 21:21.34 | MikeJ[Laptop] | sevard, try native moh. |
| 21:21.58 | sevard | MikeJ[Laptop]: is that the difference between mp3: and moh: ? |
| 21:22.49 | MikeJ[Laptop] | and don't use mp3's, use raw files or pre-encoded to the codecs you use |
| 21:22.59 | MikeJ[Laptop] | mp3 is an audio encodin format |
| 21:23.05 | MikeJ[Laptop] | moh is music on hold |
| 21:23.06 | ManxPower | MikeJ[Laptop], he's using an network shoutcast stream |
| 21:23.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-225-31.red.bezeqint.net) |
| 21:23.21 | sevard | shoutcast streams mp3s |
| 21:23.39 | MikeJ[Laptop] | using mp3 based anything will always suck |
| 21:26.43 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 21:28.47 | Katty | Ariel_: you around? |
| 21:29.56 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 21:29.56 | Katty | yay |
| 21:29.56 | dlynes | Good afternoon, Katty |
| 21:29.56 | *** join/#asterisk h3x0r (i=hex@ip68-224-57-17.lv.lv.cox.net) |
| 21:29.56 | Katty | hiya dlynes |
| 21:30.33 | dlynes | Have a good catnap? |
| 21:32.34 | jarrod | everyone liking e164 ? |
| 21:32.57 | adept_ | are there any iax providers that can port a toll free that don't suck? |
| 21:33.25 | *** join/#asterisk Z0m81e (n=pault@85-210-55-152.dsl.pipex.com) |
| 21:33.48 | Z0m81e | Hey all does anyone know why our IAX server spams sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 0! ? Cheers |
| 21:33.53 | vader-- | really weird |
| 21:34.09 | vader-- | this phone will not play back anything asterisk is doing |
| 21:34.20 | vader-- | asterisk is saying it's playing the demo but nothing is comming out of the phone |
| 21:34.51 | vader-- | anyone seen that happen before? |
| 21:35.37 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:36.21 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 21:37.34 | vader-- | hey tk |
| 21:37.46 | *** join/#asterisk newmember[laptop (n=username@ptr-66-11-81-65.ptr.terago.ca) |
| 21:37.56 | [TK]D-Fender | y0 |
| 21:38.00 | vader-- | hows it going |
| 21:38.05 | stack_ | I have a dialplan that does Answer -> Set(TIMEOUT(response)=3) -> Background(menu), I have a 't' extension for the context. If I run through the Background and don't hit anything, the Auto fallthrough fails... if I put a Goto after the background, there is no timeout... what am I missing? |
| 21:38.40 | Z0m81e | Does anyone know why our IAX server spams "sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 0!" ? |
| 21:39.29 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 21:39.47 | [TK]D-Fender | stack_ : pastebin that entire section of your dialplan and I'll take a look |
| 21:40.04 | stack_ | doing that right now |
| 21:40.05 | [TK]D-Fender | vader-- : Not too bad.. today is "down time" and the rest doing ok |
| 21:40.11 | vader-- | thats cool |
| 21:40.16 | vader-- | i got one of my phones flashed |
| 21:40.19 | adept_ | has anybody had a good experience with an IAX provider porting a number? the nufone sucks ass, I'm down right now :( |
| 21:40.24 | vader-- | and i finally got it registered with asterisk |
| 21:40.25 | [TK]D-Fender | vader-- : What kind? |
| 21:40.29 | vader-- | cisco 7940G |
| 21:40.34 | stack_ | http://pastebin.com/683841 |
| 21:40.37 | vader-- | im having a problem with the demo dialplan |
| 21:40.46 | vader-- | when i dial asterisk says everything is cool |
| 21:40.48 | [TK]D-Fender | vader-- : Congrats... they can be finnicky |
| 21:40.51 | vader-- | but i don't hear anything on the phone |
| 21:41.04 | *** part/#asterisk Hali_303 (n=surfk@dsl5402ACB4.pool.t-online.hu) |
| 21:41.12 | vader-- | this is what asterisk says in the console |
| 21:41.13 | vader-- | Executing Playback("SIP/001759E558CE01-0129", "demo-abouttotry") in new stack |
| 21:41.13 | vader-- | <PROTECTED> |
| 21:41.20 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 21:41.23 | [TK]D-Fender | stack_ : Did I ever help you with your diaplan design? |
| 21:41.24 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:41.42 | stack_ | [TK]D-Fender, you might have, why do you ask? |
| 21:41.51 | [TK]D-Fender | stack_ : in CLI does it hangup as soon as "s" runs out? |
| 21:42.03 | stack_ | [TK]D-Fender: yep |
| 21:42.06 | [TK]D-Fender | stack_ : Just noticing a bit of my style in it... |
| 21:42.26 | [TK]D-Fender | stack_ : make sure you have "autofallthrough=no" in [general] |
| 21:42.32 | vader-- | tk any ideas what would cause that? |
| 21:42.38 | stack_ | [TK]D-Fender: just trying to make use of whitespace :) |
| 21:42.40 | stack_ | [TK]D-Fender: ok |
| 21:42.41 | vader-- | or where i could look |
| 21:42.58 | [TK]D-Fender | stack_ : Its also your capitalization, etc |
| 21:43.31 | stack_ | [TK]D-Fender: just following along on the websites |
| 21:43.32 | [TK]D-Fender | vader-- : not sure.... NO sound huh? And the phone is local to *? |
| 21:43.35 | *** join/#asterisk rowter (n=Silver@201.138.157.112) |
| 21:43.45 | vader-- | ya |
| 21:43.49 | vader-- | same switch |
| 21:44.08 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:44.27 | [TK]D-Fender | vader-- : Ok, not sure what to say... start a call and verify your codecs. Try to force G711. |
| 21:44.38 | stack_ | [TK]D-Fender: I had autofallthrough=yes... that took care of it, thanks |
| 21:44.55 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 21:44.59 | [TK]D-Fender | np... autofallthrough is an ASSY idea... |
| 21:45.02 | kink0 | good night |
| 21:45.04 | [TK]D-Fender | (imo) |
| 21:45.23 | vader-- | im trying to force ulaw |
| 21:45.33 | Z0m81e | Fender, do you know what causes "sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 0!" ? Previously i've been told loading ztdummy woudl fix the issue but I have modprobe ztdummy and and I still get that spammed down the console |
| 21:46.15 | kink0 | there any way to log in Asterisk REAL public IP instead private NAT ip for peers that are nated ? |
| 21:46.20 | [TK]D-Fender | Z0m81e : not a clue... |
| 21:46.28 | kimosabe | does any one here asterisk on there freebsd box ?? |
| 21:46.54 | Nugget | I do. |
| 21:47.17 | kimosabe | nugget what version freebsd are you using ? |
| 21:47.32 | Nugget | 6.mumble, but I've been running asterisk on it since 5.21 or so |
| 21:47.42 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 21:47.50 | Nugget | I had asterisk on a 4.x box last year |
| 21:47.54 | kimosabe | nugget you compilled direct from ports |
| 21:47.59 | vader-- | [TK]D-Fender in my sip.conf i forced disallow=all and then allow-ulaw and on my phone it's set to g711ulaw |
| 21:48.03 | brif8 | understanding that echo I hear is caused by the other end. Is there anything I can do to get rid of it on my side? Ever odd call (now and again) I hear real bad echo |
| 21:48.05 | Nugget | no, I compiled the tarball. |
| 21:48.09 | Nugget | the port is full of cruft |
| 21:49.15 | kimosabe | nuget ok what do i have 2 add 2 the box so that i can compile in asterisk i have a base install |
| 21:49.35 | Nugget | gmake |
| 21:49.52 | dlynes | Nugget...what's the problem with asterisk on freebsd? I've had it running on FreeBSD 6.0, without any problems other than zaptel problems |
| 21:49.56 | Nugget | I don't think there's anything else you *have* to have, but I might be wrong. |
| 21:50.11 | Nugget | zaptel is the source of all the problems i've had too. |
| 21:50.20 | Nugget | I don't run zaptel any more and it's stable |
| 21:50.29 | [TK]D-Fender | vader-- : ok, no clue what to suggest for no audio... I'm not particularly experienced with Cisco's quirks |
| 21:50.30 | Alystair | HAHA! |
| 21:50.37 | Alystair | I just realized something really stupid/funny |
| 21:50.49 | Alystair | http://www.digium.com/en/products/hardware/tdm400p.php <-- if you look at the top left where the shopping cart it, it looks like an evil ghost from mario |
| 21:50.55 | vader-- | here is a copy of a sip debug from the call |
| 21:50.56 | vader-- | http://pastebin.ca/51610 |
| 21:51.25 | dlynes | Nugget: ah...it's kimosabe that's having problems with FreeBSD then? |
| 21:51.43 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 21:52.00 | blaylock | anyone here have an * PBX behind a DSL conn? |
| 21:52.11 | Nugget | dlynes: I wouldn't say problems, just asking how to compile it. |
| 21:52.14 | Z0m81e | blaylock yes |
| 21:52.14 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 21:52.16 | jaiger | blaylock, I do |
| 21:52.23 | jaiger | blaylock, I'm sure many here do |
| 21:52.26 | Nugget | I do too. |
| 21:52.31 | blaylock | any wierdness with congestion problems? |
| 21:52.32 | ManxPower | One way audio is almost always a NAT or Firewall problem |
| 21:52.42 | Nugget | he didn't say NAT. |
| 21:52.57 | Nugget | oh, n/m. |
| 21:53.02 | Nugget | I missed that question |
| 21:53.07 | jaiger | blaylock, congestion in what way? my allergies have been acting up lately |
| 21:53.20 | blaylock | i keep getting this error: auto_congest: Auto-congesting call due to slow response |
| 21:53.26 | dlynes | Nugget: ah |
| 21:53.30 | ManxPower | kink0, Yes. nat=yes |
| 21:53.31 | jaiger | blaylock, I haven't seen that |
| 21:53.36 | blaylock | hmm |
| 21:53.47 | blaylock | this is when Im trying to do IAX2 calls |
| 21:53.48 | Nugget | I run my home phone as VoIPoI on consumer dsl. 98% of the time it's great, and 2% of the time it totally sucks. |
| 21:53.54 | kimosabe | nuget yeah i been running it on linux just trying 2 switch it 2 linux box my only linuxes are asterisk all otheres are freebsd |
| 21:53.56 | Nugget | that's with traffic shaping |
| 21:54.19 | Z0m81e | blaylock I can't say i've ever seen that i'm afraid |
| 21:54.26 | blaylock | darn |
| 21:55.05 | Z0m81e | blaylock do you have a particularly bad ping to the iax server? |
| 21:55.14 | stack_ | Has anyone here ever used iaxmodem? How reliable is it? |
| 21:55.16 | blaylock | Z0m81e, nah the ping is great |
| 21:55.24 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 21:55.30 | blaylock | less than a ms actually |
| 21:55.49 | ruza | chan_iax2.c: Unable to create translator path for unknown to gsm on IAX2/bill-11 |
| 21:55.50 | bsdfreak | yeah |
| 21:55.54 | bsdfreak | it's lr3 |
| 21:55.56 | ruza | what does it mean ? |
| 21:56.01 | bsdfreak | but you gotta compile it with -O3 |
| 21:56.32 | mog_work | o3 is for children us ricers use o7 |
| 21:56.57 | Nugget | -funroll-loops -zomg-optimize! |
| 21:57.59 | Z0m81e | blaylock if your ping is that good i'd have to guess you ahve a firewall thats eating the data |
| 21:58.36 | blaylock | perhaps |
| 21:58.37 | kink0 | ManxPower, hmmm that is what I had nat=yes, now I am trying nat=never, I want to avoid peers are behind nated and I need to log the public IP in CDR instead a private IP |
| 21:59.39 | *** join/#asterisk Stp1800 (n=Stp1800@68-235-251-98.atlsfl.adelphia.net) |
| 21:59.55 | kink0 | ManxPower, my question is about a configuration where there no callid:password, so the system works based on publics IP allowed or disallowed, then I am not able to trace CDR and so, if the peers uses NAT, because only private ip are shows |
| 22:00.06 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 22:00.14 | tehdely | HI |
| 22:00.43 | tehdely | what could be causing horrible jitter and garbly sound on the inbound leg of a PRI zap channel |
| 22:00.52 | bsdfreak | tehdely |
| 22:00.55 | tehdely | calls that come in, the caller's voice is being destroyed |
| 22:00.57 | bsdfreak | you fucking buttsniffer! |
| 22:00.58 | tehdely | but the outbound leg of it soundsfine |
| 22:00.59 | bsdfreak | :) |
| 22:01.01 | tehdely | and if we call them back |
| 22:01.02 | tehdely | it is fine |
| 22:01.04 | tehdely | only when they call |
| 22:01.05 | tehdely | *whew* |
| 22:01.08 | tehdely | bsdfreak: lick me |
| 22:01.29 | bsdfreak | is metawire a lost cause? |
| 22:02.00 | tehdely | no. |
| 22:02.08 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 22:02.09 | bsdfreak | hasn't it been down for over a month? :( |
| 22:02.13 | tehdely | though i would prefer that you refer to it from this time forward as Schiavowire |
| 22:02.13 | dlynes | kink0: Use account codes |
| 22:02.18 | kink0 | ManxPower, I used ${SIPURI} to detect the IP, but if the peer is nated, the IP that $SIPURI takes is the private |
| 22:02.22 | _Thor | Hello everybody |
| 22:02.22 | bsdfreak | oh ok |
| 22:02.23 | bsdfreak | lol |
| 22:02.43 | bsdfreak | you should come back to irc.bsdgeek.com |
| 22:02.44 | bsdfreak | :P |
| 22:02.47 | ManxPower | kink0, I know of no way to log the public IP |
| 22:03.00 | Z0m81e | wouldn't the public IP be on the sip contact? |
| 22:03.44 | tehdely | are you still linked to that shitty foxchat net |
| 22:03.45 | kink0 | ManxPower, yes , as I search for info, you are right not a direct way to log the public IP, but in this scenario I prefered to refuse nated peers |
| 22:03.55 | kink0 | Z0m81e, no |
| 22:04.45 | kink0 | dlynes, that was I did fews weeks ago, but my peers send me differents accounts codes, imprevisible, because he takes the account code from the ends users, and may be any arbitrary string |
| 22:05.34 | dlynes | kink0: imprevisible? |
| 22:05.41 | kink0 | while I have not way to log public IP, I prefered to refuse peers or phones who are behind a NAT |
| 22:05.48 | _Thor | guys I have a question... On those split T1s (half data, half voice channels), how in the world is it physically done (assuming I have a sangoma card) |
| 22:06.15 | kink0 | dlynes, yes, because they send me the original account name, the name they receives from her phones and other peers |
| 22:06.43 | dlynes | kink0: What are you referring to as an account name? |
| 22:06.55 | kink0 | dlynes, was not sending the account name I assigned them, they was really sending me a name that was comming from her users. |
| 22:07.22 | dlynes | kink0: But what are you calling an 'account name'? |
| 22:07.25 | kink0 | ${CDR(clid) |
| 22:07.33 | dlynes | kink0: that's your account name? |
| 22:07.58 | dlynes | kink0: CallerID(num)? |
| 22:08.03 | kink0 | that is what I got from my peer, and no other fields who referers inequivoque to that peer |
| 22:08.23 | dlynes | kink0: SetAccountCode(thisuserisadweeb) |
| 22:08.25 | kink0 | no, is not callerID, but many times is the same as callerID |
| 22:09.02 | kink0 | dlynes, hmmmm but how I difference one peer from other peer ? |
| 22:09.23 | kink0 | if I set an accountcode, will be the same for both, no ? |
| 22:09.34 | dlynes | kink0: They're authenticating, with a username and password, are they not? |
| 22:09.37 | kink0 | in the event I have not added "friends" to my sip.conf |
| 22:09.47 | kink0 | ahhh no !! that is the question !! |
| 22:10.04 | kink0 | they are not authenticated with user and pass, I have not added so to my sip.conf |
| 22:10.11 | Dr-Linux | hey kink0 :) |
| 22:10.24 | dlynes | So they're just calling, without registering, authenticating, or anything else? |
| 22:10.27 | blaylock | if i change iax.conf do i need to restart asterisk totally or just do a reload in the console? |
| 22:10.37 | kink0 | because when they routes me a call, they sends me accountcode from her users, and not what I assigned to them |
| 22:10.52 | kink0 | dlynes, public IP basis |
| 22:10.58 | kink0 | Dr-Linux, heyyy !! |
| 22:10.59 | dlynes | kink0: gimme a few |
| 22:11.05 | kink0 | dlynes, sure |
| 22:11.36 | *** part/#asterisk newmember[laptop (n=username@ptr-66-11-81-65.ptr.terago.ca) |
| 22:11.45 | *** join/#asterisk MGSsancho (n=weee@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
| 22:12.07 | kink0 | anywise I know is a large issue, and little documented, in the event is very complex to really logs the public IP, would nat=never refuses all nated peers ? |
| 22:12.09 | mercestes | blaylock: jsut to a reload in the console. |
| 22:13.31 | blaylock | k thanks |
| 22:14.58 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 22:17.36 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:17.38 | brif8 | are there any self contained pre-built * pbx boxes/units ? |
| 22:17.52 | shmaltz | I need a bit of advice with a large project, anybody interessted? |
| 22:19.11 | kimosabe | schmalt come on with it |
| 22:20.28 | shmaltz | ok, kimosabe, here are the details, a DMS 500 with 4 asterisk boxes that each contain 4 quad span cards, with DID configured on it. |
| 22:20.47 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 22:21.02 | shmaltz | the DMS 500 is dumb, it has no clue where the next call has to go, all it does is it starts a PRI 1 on asterisk1 and tries the next channel. |
| 22:21.35 | kimosabe | hosnestlly sorry im not familiar with did config |
| 22:21.40 | shmaltz | until it gets to the 48th PRI on asterisk 4, and if all channels are busy there it just returns a busy signal |
| 22:21.41 | eKo1 | brif8: yes there are |
| 22:21.47 | brif8 | eko1 yes |
| 22:21.54 | shmaltz | thank kimosabe |
| 22:22.17 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 22:22.32 | brif8 | eKo1: can you give a URL or more info |
| 22:22.38 | brif8 | shmaltz: keep going |
| 22:22.58 | shmaltz | the 4 asterisk boxes will be used for conferencing |
| 22:23.15 | hads | brif8: Xorcom make one |
| 22:23.40 | shmaltz | the provisioning for setting up the conference will be done is such a way that the each conference is assigned a DID |
| 22:23.42 | Igbothom_III | the xorcom hardware looks schweeet |
| 22:23.59 | shmaltz | each asterisk knows, which DID goes to wich conf |
| 22:24.00 | vader-- | any of you guys ever see this error: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 22:24.12 | shmaltz | each asteirsk knows which of the 4 hosts which conf |
| 22:24.18 | brif8 | shmaltz: so it stops after the 48th PRI and doen't see the other PRI ? |
| 22:24.34 | shmaltz | brif8, there are only 48 PRIs |
| 22:24.51 | shmaltz | since 4 * asterisk, each asterisk with 12 PRIs |
| 22:24.57 | shmaltz | 3 quad cards |
| 22:25.02 | Igbothom_III | OUCH! |
| 22:25.30 | shmaltz | so as far as the DMS is concerned these are all the same or are all 20 30 it doesn' care, |
| 22:25.36 | Igbothom_III | running on more than an EPIA CPU, I gather :) |
| 22:25.42 | shmaltz | it has all the PRIs in one hunt for the DIDs |
| 22:25.56 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 22:25.59 | shmaltz | lgbothom_lll |
| 22:26.03 | shmaltz | sorry |
| 22:26.58 | shmaltz | what I need to know is what would be the best way to route a DID thru an asterisk server to another one, just because the conf is hosted on another box, best being defined with the least performance hit |
| 22:28.56 | mercestes | Vader: That means your trying to dial a phone that is not registered. |
| 22:29.12 | mercestes | Vader: Could be a phone that's gone offline..(usually is) |
| 22:30.09 | skydog | bandwidth question - assuming G.711 one call is 82.1 kpbs, if ADSL supports 128K then its about 15 calls comapred to 2 over standard telephony. What do you need to calculate approximately say 20000 calls/24hr period with calls lasting for about 60s? |
| 22:30.23 | vader-- | here is my dial plan for it |
| 22:30.23 | vader-- | exten => _2XX,1,NoOp(.call for .${EXTEN}) |
| 22:30.23 | vader-- | exten => _2XX,2,Dial(SIP/${EXTEN}) |
| 22:30.23 | vader-- | exten => _2XX,3,VoiceMail(u${EXTEN}) |
| 22:30.23 | vader-- | exten => _2XX,4,Congestion |
| 22:30.32 | dlynes | kink0: No idea, unless you actually modify some of the asterisk C code |
| 22:30.33 | vader-- | does that look right? |
| 22:30.38 | Igbothom_III | vader - dude, use a pastebin |
| 22:30.40 | dlynes | kink0: Or keep a tighter wrap on your security |
| 22:30.46 | vader-- | sorry i figured it was only 4 lines |
| 22:30.55 | [TK]D-Fender | skydog : You math = bad. tha means your ADSL supports *1* call |
| 22:31.13 | Igbothom_III | eggs zacchery - 128 divided by 82.1 = 1.something |
| 22:31.40 | [TK]D-Fender | Igbothom_III : lol... I type like shit and you raise it to an art form :) |
| 22:31.51 | [TK]D-Fender | askl;djfkl;asdfjkl;asdfasdf |
| 22:31.52 | Igbothom_III | hehe |
| 22:31.58 | OloBola | how can I tell if my server is registering with my new provider's server? |
| 22:32.18 | kink0 | dlynes, well, I think I prefered try to use an external application capable to extract ip |
| 22:32.58 | vader-- | you guys see anything wrong in that dial plan? |
| 22:34.01 | [TK]D-Fender | vader-- : Yeah, fisrt it should have been pastbin'd, second it allows no-valid entries to dial and ASSUMES there is voicemail to back it as well..... |
| 22:34.26 | *** join/#asterisk wolfson (n=wolfson@nc-71-2-31-90.dyn.sprint-hsd.net) |
| 22:34.34 | [TK]D-Fender | vader-- : 3rd, if someone ends up there on a 1-800 call, they could hang around in Congestion INDEFINATELY costin you a fortune :) |
| 22:34.40 | OloBola | iax2 show peers |
| 22:34.49 | [TK]D-Fender | vader-- : ... for starters :) |
| 22:34.53 | skydog | [TK]D-Fender: just curious as to "how" to arrive at the # ....Is there a base calculation or formula? |
| 22:35.03 | vader-- | hehe |
| 22:36.36 | [TK]D-Fender | skydog : 82kbps vs 128kbps... do the math.... |
| 22:37.11 | [TK]D-Fender | vader-- : you are in dire need of a STDEXTEN style macro... |
| 22:37.22 | skydog | got it :) |
| 22:37.51 | vader-- | im just trying to get communication between two phones |
| 22:38.03 | vader-- | because the demo context doesn't seem to be producing any audio to the phones |
| 22:38.11 | *** join/#asterisk nvrs (n=RUR@HSE-Montreal-ppp3469401.sympatico.ca) |
| 22:38.11 | vader-- | so i was going to try phone to phone communication |
| 22:38.26 | [TK]D-Fender | the * sample configs aren't worth shit.... |
| 22:38.58 | Hmmhesays | They're ok |
| 22:39.00 | fourcheeze | well, they could certainly be better |
| 22:39.05 | skydog | [TK]D-Fender: The question here is that the assumption is that "silence" if you will does not take up packets, and the assumption being made is that there is voice both ways in relating to the 82kpbs. Also is the curve linear for the number of calls that increase on * machine for bandwidth? |
| 22:39.10 | [TK]D-Fender | I should publish my samples.... |
| 22:39.15 | fourcheeze | please do |
| 22:39.22 | brif8 | yeah |
| 22:39.25 | vader-- | ditto |
| 22:39.31 | [TK]D-Fender | skydog : And what you fail to realize is * doesn't SUPPORT slience detection :) |
| 22:39.46 | vader-- | ya i was just trying to setup a plan that would dial an extension |
| 22:39.48 | vader-- | something simple |
| 22:39.52 | vader-- | to see if the phone responded |
| 22:40.26 | *** join/#asterisk zeppelin_ (n=zeppelin@201-34-96-165.paemt700.dsl.brasiltelecom.net.br) |
| 22:40.39 | skydog | [TK]D-Fender: Is this wrong info then - ? - http://www.voip-info.org/wiki-Bandwidth+consumption |
| 22:40.39 | [TK]D-Fender | vader-- : Well you fail to get audio to play back from * period... an exceptionally bad sign to start... |
| 22:41.08 | [TK]D-Fender | skydog : that is +/- right, just no benifit of silence detection. |
| 22:41.34 | vader-- | ya |
| 22:41.36 | [TK]D-Fender | skydog : And then you can take OTHER factors into account like IAX trunking where applicable, RTP packet size, etc... |
| 22:41.46 | vader-- | not sure where to even look |
| 22:42.12 | [TK]D-Fender | vader-- : Do you head DIALTONE on the handset? |
| 22:42.21 | vader-- | ya |
| 22:42.39 | vader-- | i can dial the extension |
| 22:42.47 | vader-- | and all i hear is dead air |
| 22:42.49 | [TK]D-Fender | And it rings, and you answer? |
| 22:42.58 | vader-- | i dial the demo extension 500 |
| 22:42.59 | [TK]D-Fender | Can you hear dial-tone on both sides? |
| 22:43.07 | vader-- | yes |
| 22:43.18 | [TK]D-Fender | Ok ell at least your handsets are plugged in right then :) |
| 22:43.23 | vader-- | hehe |
| 22:43.26 | vader-- | ya they are registering |
| 22:43.36 | [TK]D-Fender | vader-- : I didn't even care about that part yet :) |
| 22:43.39 | skydog | [TK}D-Fender: Good enough..:) Just need some clarity..which youve provided...thnx. |
| 22:43.51 | [TK]D-Fender | skydog : np |
| 22:44.10 | [TK]D-Fender | skydog : Lots of little things go into finding out what you can do based on HOW you want to do it. |
| 22:44.25 | vader-- | ya i get this in the asterisk cli |
| 22:44.25 | vader-- | <PROTECTED> |
| 22:44.55 | [TK]D-Fender | skydog : For instance switching to G.729, changing the packet size to 30ms, and using IAX trunking has a DRAMATIC effect on usage |
| 22:45.11 | [TK]D-Fender | vader-- : if you don't even hear that file being played, you're DOA |
| 22:46.20 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 22:47.10 | ManxPower | *grumble* I never thought building a campground wide cable system would be so hard. Ground Loops: The Bane of Us All. |
| 22:47.48 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 22:47.54 | [TK]D-Fender | ManxPower : Oh... and can you rach over to that 3rd rail if you please! Thanks!!! |
| 22:48.11 | [TK]D-Fender | bbbzzzzzzzZZZZZZZZZZORTCH!!!!! |
| 22:48.28 | ManxPower | 1 common satallite dish, multiple receivers, some of them served off a different mains transformer |
| 22:49.05 | ManxPower | .3 volts between mains ground and cable ground |
| 22:49.24 | ManxPower | at the receiver where the USB network adapter on the TiVo keeps blowing up |
| 22:50.32 | sevard | people don't need tv in a campground anyway. |
| 22:51.27 | sevard | if i ran a campground all you could bring in would be a big piece of cloth, the clothes on your back and one knife for your whole group |
| 22:51.40 | vader-- | would asterisk tell me if those sounds weren't there? |
| 22:51.52 | sevard | vader--: it should |
| 22:51.58 | [TK]D-Fender | vader-- :yes, and those are defaults... |
| 22:52.30 | vader-- | hmmm wtf would it not work then ;( |
| 22:52.31 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 22:52.44 | [TK]D-Fender | vader-- : RTP being blocked..... |
| 22:52.57 | vader-- | RTP? |
| 22:52.58 | [TK]D-Fender | got firewall of any kind on that box? |
| 22:53.00 | vader-- | na |
| 22:53.11 | vader-- | the server and both phones are on the same switch |
| 22:54.43 | ManxPower | vader--, you're not doing something stupid like using G723.1 or G729 in passthru-mode, are you? |
| 22:54.45 | [TK]D-Fender | vader-- : Doesn't mean the * server isn't blocking it somehow |
| 22:54.59 | [TK]D-Fender | ManxPower : No supposedly its G711 end-to-end |
| 22:55.19 | vader-- | na i forced everything to be G711ulaw |
| 22:55.26 | ManxPower | [TK]D-Fender, odd. You have him run iptables -l or ipchains -l ? |
| 22:55.57 | [TK]D-Fender | ManxPower : I though I'd ask the "generic" way first :) |
| 22:56.09 | vader-- | neither are running |
| 22:56.14 | ManxPower | vader--, you don't have a bindaddr= in sip.conf or multiple ip addresses on the server, do you? |
| 22:56.25 | vader-- | na |
| 22:56.50 | vader-- | bindaddr=0.0.0.0 |
| 22:56.56 | ManxPower | vader--, now you are beyond even my help, son. |
| 22:57.03 | ManxPower | vader--, comment the bindaddr out. |
| 22:57.28 | vader-- | nope that didn't work |
| 22:57.36 | sevard | did you reload? |
| 22:57.42 | vader-- | ya |
| 22:57.45 | ManxPower | add context=INVALID in sip.conf [general] to make sure the phones are matching a [blah] entry |
| 22:58.20 | vader-- | manx they are dialing fine |
| 22:58.26 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 22:58.32 | vader-- | and asterisk is responding to their requsts properly |
| 22:58.38 | vader-- | but it's just not producing any audio |
| 22:59.06 | sevard | vader--: are you dialing sip phone to sip phone or are you dialing an extension that you've set up to play a file for you |
| 22:59.34 | vader-- | im trying to dial the demo extension |
| 22:59.35 | vader-- | 500 |
| 22:59.50 | sevard | and 500 consists of |
| 22:59.52 | vader-- | when i dial 500 asterisk console says -- Executing Playback("SIP/001759E558FE01-dd60", "demo-abouttotry") in new stack |
| 22:59.53 | vader-- | <PROTECTED> |
| 23:00.33 | sevard | is like... the volume up on your handset? ;/ |
| 23:00.38 | vader-- | ya |
| 23:00.50 | sevard | what happens when you call the other phone |
| 23:00.59 | vader-- | i don't have a dial plan setup for that |
| 23:01.20 | sevard | try it |
| 23:01.33 | vader-- | i tried to create one and it didn't work |
| 23:01.45 | OloBola | Teliax question: is that just a one time setup fee for toll free numbers (4.99), or is it monthly? |
| 23:03.16 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36) |
| 23:03.16 | sevard | OloBola: it's a 10.00 fee from that is deducted 4.99(your monthly fee) then you have the left overs to make calls, as soon as you dip below 5.00 you are charged another 10.00 |
| 23:03.42 | blaylock | can anyone tell me which port IAX uses? |
| 23:04.21 | file | 4569 UDP |
| 23:04.46 | *** join/#asterisk MGSsancho (n=weee@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
| 23:06.02 | MoutaPT | HI any one could advice me the best way or any opensource project to sipphone for call centers based on asterisk |
| 23:06.08 | OloBola | sevard: thanks. So toll-free numbers are 4.99 a month or is that just an account fee? I'd like to have multiple toll-free numbers. |
| 23:06.29 | *** join/#asterisk Maxxed (n=whyman@65.59.245.122) |
| 23:06.32 | sevard | OloBola: I don't know anything about the other plans, i'm just talking about the pay as you go plan |
| 23:06.34 | Maxxed | yo :) |
| 23:06.40 | MoutaPT | I mean an inbound call when rings in the Agent SIp phone and he answers the call, automaticaly opens a website in IE or mozilla |
| 23:06.41 | MoutaPT | = |
| 23:06.43 | MoutaPT | ? |
| 23:06.50 | sevard | OloBola: you can add numbers under your account |
| 23:07.03 | blaylock | thanks file |
| 23:08.07 | OloBola | sevard: under my account it says "Get A Toll Free Number @ $4.99". I'm guessing that's just a one time setup fee, not sure. |
| 23:08.27 | *** part/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 23:08.31 | [TK]D-Fender | Olo : or a monthly DID fee, or who knows.... |
| 23:08.34 | file | OloBola: why don't you ask them? |
| 23:08.35 | sevard | OloBola: Monthly |
| 23:08.56 | sevard | OloBola: all of this is explaine don the website and if you call Dave will tell you everything |
| 23:09.41 | OloBola | sevard: thanks. I just didn't want to bother anyone, I know they are probably really busy right now. |
| 23:10.05 | sevard | yeah, but Dave is awesome. Jus t4ell him he's the best * guru on the planet |
| 23:11.34 | sevard | actually they're probably closed by now |
| 23:11.39 | sevard | 10 minutes ago |
| 23:12.37 | Maxxed | hey |
| 23:12.56 | Maxxed | im trying to goto a context useing a feature |
| 23:12.57 | Maxxed | sales-ringnspy => *6,caller,GoTo,sales-ringnspy,s,1 |
| 23:13.05 | Maxxed | can i use the goto during a call? |
| 23:14.05 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 23:16.52 | MoutaPT | Any one knows how to trigger IE with a Specific URI when asterisk dials a user? |
| 23:17.07 | sevard | cat |
| 23:17.08 | MoutaPT | I mean i receive the call and at the same time IE launches a Website |
| 23:17.59 | MoutaPT | sevard: talking to me? |
| 23:18.21 | ManxPower | BTW, this campground will have at least 5 people living there full time, and will have at least 5 cabins where the people will be here most weekends |
| 23:18.27 | *** part/#asterisk Jan_FL (n=trekkie@relativity.mediaweb-it.net) |
| 23:18.56 | shmaltz | MoutaPT, look at FOP |
| 23:19.00 | shmaltz | ~fop |
| 23:19.01 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 23:19.11 | MoutaPT | thks |
| 23:21.17 | shmaltz | ManxPower, if I need to route thru an asterisk box up to 200 calls, but I want to keep some calls on there for app_meetme, and others I want to route to other asterisk boxes, all calls come in as PRI, waht would be the best way, best defined as the best for performance? |
| 23:22.23 | ManxPower | shmaltz, the best way is for you to start handing me large piles of cash to do performance benchmarks and system design for you. |
| 23:22.37 | shmaltz | ManxPower, lol |
| 23:22.47 | sevard | I'll take some milk money. |
| 23:22.52 | shmaltz | I honestly belive you about that offer |
| 23:23.46 | ManxPower | and Montgomery. |
| 23:24.11 | sevard | I am now accepting clients everywhere. Send me your money with a picture of your woman. |
| 23:24.29 | shmaltz | ManxPower what about irc://irc.freenode.net/#asterisk ? accepting clients from there? |
| 23:25.08 | ManxPower | shmaltz, if they are located in Birmingham, Atlanta, Huntsville, Chattanooga, or Montgomery, yes. |
| 23:25.22 | shmaltz | sevard, I'm now accepting clients anywhere just send me the money, no pics needed |
| 23:25.24 | ManxPower | or in that area. |
| 23:26.08 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:26.25 | ManxPower | For stuff as complicated as Asterisk, and since clients seldom have any idea of what they really want or what it will really cost, I don't feel I can provide good services to clients that are too far away, unless they want to pay for my travel expenses in addition to my consulting feew. |
| 23:27.07 | tainted- | i do incalls as well as outcalls but i don't do greek |
| 23:27.15 | tainted- | anywhere near your house |
| 23:27.27 | ManxPower | LOL! |
| 23:27.28 | shmaltz | ManxPower, you remember that chat we had a while ago about *breeders*? |
| 23:27.37 | ManxPower | shmaltz, vaguely |
| 23:27.41 | ManxPower | was I drunk at the time? |
| 23:28.32 | shmaltz | ManxPower you used a word that was new to me, the definition of that word is about raising/producing *perfect* humans, I have since forgotten that word, what is it? |
| 23:28.32 | Alystair | argh |
| 23:28.33 | Alystair | CEO asked me a good question |
| 23:28.41 | shmaltz | ManxPower, don't remember if you were drunk or not |
| 23:28.51 | shmaltz | Alystair, no he didnt' |
| 23:29.01 | ManxPower | Was the word eugenics? |
| 23:29.03 | *** join/#asterisk CpuID (n=nathan@gentoo/contributor/cpuid) |
| 23:29.09 | shmaltz | you got it |
| 23:29.09 | RoyK | <PROTECTED> |
| 23:29.18 | Alystair | recieving faxs is exactly the same if you plug in a fax with an adapter to the VOIP system that has a DID? |
| 23:29.22 | CpuID | hmm, have queues changed at all in the past 2-3 versions? |
| 23:29.28 | CpuID | (eg. prior to 1.2.7.1 |
| 23:29.37 | shmaltz | Alystair, no in fact it doesn't work with asterisk |
| 23:29.48 | shmaltz | ManxPower thanks |
| 23:30.17 | ManxPower | Alystair, no. |
| 23:30.28 | Alystair | How would I get faxes to work then |
| 23:30.36 | ManxPower | shmaltz, only 1 person on it right now. |
| 23:30.39 | shmaltz | Alystair, by *NOT* using VoIP |
| 23:30.42 | tainted- | wow ManxPower teaches eugenics? |
| 23:30.49 | ManxPower | tainted-, no, I do not. |
| 23:30.52 | Alystair | also, we have a credit/debit swipe system, could that work over VOIP |
| 23:30.57 | shmaltz | ManxPower, but fax is in there as wll, no? isnt' it? |
| 23:31.00 | sevard | Alystair: A friend of mine uses a SIP 2002 on his fax machine and gets faxes through 50-70% of the time |
| 23:31.04 | shmaltz | Alystair, nope |
| 23:31.11 | ManxPower | Alystair, anything that does modem like stuff is problematic to do over VoIP. |
| 23:31.18 | shmaltz | Alystair, neither will ADSI, or Alarm systems |
| 23:31.31 | tainted- | Alystair no |
| 23:31.38 | RoyK | ~foip |
| 23:31.39 | jbot | well, foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject |
| 23:31.52 | sevard | It's stupid anyway, why do analog shit over your digital lines, upgrade your hardware. |
| 23:31.52 | RoyK | Alystair: read that |
| 23:32.03 | tainted- | are there digital card swipes? |
| 23:32.07 | ManxPower | or just get an analog line for fax, what's what we do. |
| 23:32.19 | shmaltz | tainted-, of course |
| 23:32.22 | ManxPower | analog lines are also good for 911 and CC machines |
| 23:32.29 | tainted- | analog shit to digital line is hardly an 'upgrade' in some cases |
| 23:32.29 | RoyK | sevard: some people need backward compatibility........ |
| 23:32.34 | tehdely | anybody here in japan? |
| 23:32.49 | tainted- | i've got a song called 'made in japan' |
| 23:32.50 | ManxPower | And any company that does all VoIPoInternet with no local lines is a moron anyway. |
| 23:33.08 | shmaltz | tainted- your local ShopRite, WalMar, Target and who not uses a digital card swipe |
| 23:33.08 | [TK]D-Fender | tehdely : I hear ther are THOUSANDS actually, perhaps more still! |
| 23:33.21 | CpuID | my bad, queues are fine :) |
| 23:33.24 | justinu|laptop | big in japan |
| 23:33.26 | tehdely | i know, damn people are everywhere in japan |
| 23:33.28 | tehdely | problem is |
| 23:33.36 | tehdely | it's so tough finding japanese people willing to test your new toll free DID |
| 23:33.50 | tehdely | they have to satisfy the special criteria of being japanese AND being in japan |
| 23:34.01 | shmaltz | tehdely, just advertise that it offers sex girls |
| 23:34.06 | justinu|laptop | why do they have to be japanese? |
| 23:34.08 | sevard | are all the white people back in japan? |
| 23:34.21 | tehdely | justinu|laptop: because it's a japanese toll free # |
| 23:34.24 | tainted- | tehdely start a cult.. u'll have people in no time |
| 23:34.24 | tehdely | i have no way of testing it outside of japan |
| 23:34.32 | tehdely | tainted-: i'd be going up against aum, and they have lots of money |
| 23:34.34 | RoyK | wow. i just got an email from someone who could answer all my questions about economy :) |
| 23:34.34 | tehdely | also nerve gas |
| 23:34.39 | tehdely | RoyK: lyndon larouche? |
| 23:34.46 | tehdely | ;) |
| 23:34.46 | tainted- | yea but u got voip and tf |
| 23:34.50 | RoyK | tehdely: que? |
| 23:35.01 | tehdely | obviously not an american :P |
| 23:35.10 | RoyK | not at all |
| 23:35.12 | sevard | tehdely: I heard in japan you can get 48mbit fiber to your house for like 20 USD |
| 23:35.23 | tehdely | sevard: you can also buy girls' used panties in vending machines |
| 23:35.25 | tehdely | strange country... |
| 23:35.35 | tainted- | according to mark @ digium, iax2 traverses many type of nerve gases |
| 23:35.41 | RoyK | proud to be unamerican :) |
| 23:35.42 | sevard | tehdely: awesome + disease center |
| 23:36.05 | sevard | tainted-: wtf? |
| 23:36.30 | tainted- | nerve gas + iax2 = crazy delicious |
| 23:36.32 | tehdely | tainted-: nerve gas causes an explosion in the local weasel population |
| 23:36.35 | tehdely | i'm sure you can imagine why that's bad |
| 23:36.42 | tehdely | first they start gnawing on your PRI |
| 23:36.49 | tehdely | then they crack open the rack and start going after the server |
| 23:36.56 | tehdely | soon weasels are shitting out half-digested dialplans |
| 23:37.04 | tehdely | horrifying, really |
| 23:37.08 | tainted- | fucking weasels |
| 23:37.21 | sevard | Kids, take note. This is what happens to you if you're a telco junkie cooped up in a server room all day. |
| 23:37.34 | [TK]D-Fender | ... |
| 23:37.42 | Maxxed | so yeah bah |
| 23:37.48 | [TK]D-Fender | :/ |
| 23:37.48 | Maxxed | can i use a goto in the features.conf |
| 23:37.49 | [TK]D-Fender | :| |
| 23:37.50 | justinu|laptop | can't a person in japan, who speaks japanese, but may not be a native japanese test your number? |
| 23:37.53 | [TK]D-Fender | :{ |
| 23:37.54 | tehdely | justinu|laptop: yeah |
| 23:37.57 | tehdely | they just have to be in japan |
| 23:37.57 | ManxPower | that reminds me to check the weather for tomorrow |
| 23:38.12 | justinu|laptop | so why do they have to be japanese? |
| 23:38.16 | tehdely | justinu|laptop: sarbanes-oxley |
| 23:38.33 | tainted- | sarbanes-oxley can kiss my ass |
| 23:38.43 | sevard | tehdely: you meant to say they just have to be in japan, not be in japan and also be japanese |
| 23:38.47 | tehdely | yes sevard |
| 23:39.04 | tainted- | i had no idea that sox had juridiction over your ethnicity |
| 23:39.05 | tehdely | but i would like 'justinu' to believe that a bunch of government regulations requires an authentic japanese on the other end of the line for auditing purposes |
| 23:39.21 | tehdely | imean cmon, it seems plausible, right? |
| 23:39.31 | justinu|laptop | lol |
| 23:39.45 | ManxPower | I'm getting REALLY sick of this "storms every wekeend" stuff. |
| 23:39.50 | Maxxed | sales-ringnspy => *6,caller,GoTo,sales-ringnspy,s,1 |
| 23:39.54 | Maxxed | will that work? |
| 23:40.00 | tainted- | ever have to make something sox compliant? |
| 23:40.03 | tainted- | it's sweet |
| 23:40.04 | justinu|laptop | ManxPower: maybe you should move |
| 23:40.04 | Maxxed | like, goto conext s 1 |
| 23:40.14 | tehdely | tainted-: i have to imagine it is the most horrible skullduggery ever |
| 23:40.19 | ManxPower | Maxxed, Try it and let us know. |
| 23:40.23 | Maxxed | well, no |
| 23:40.24 | Maxxed | heh |
| 23:40.28 | tainted- | it's gobs of money |
| 23:40.31 | [TK]D-Fender | I get sox complaints all the time... like myfile.doc and myotherfile.xls not valid source.... |
| 23:40.38 | ManxPower | justinu|laptop, I just moved. For the 5th time since Katrina |
| 23:40.40 | tainted- | y2k bug type magnitude in money |
| 23:40.54 | ManxPower | I'm not moving again. |
| 23:41.00 | justinu|laptop | ManxPower: on the mesa right? |
| 23:41.04 | Maxxed | <PROTECTED> |
| 23:41.04 | Maxxed | Apr 26 18:07:37 NOTICE[19479]: pbx.c:1750 pbx_extension_helper: No such label 'sales-ringnspy' in extension '8088' in context 'houstonsip' |
| 23:41.04 | Maxxed | Apr 26 18:07:37 WARNING[19479]: pbx.c:6510 ast_parseable_goto: Priority 'sales-ringnspy' must be a number > 0, or valid label |
| 23:41.05 | Maxxed | <PROTECTED> |
| 23:41.11 | ManxPower | I'm going to buy my cabin in the woods. |
| 23:41.15 | ManxPower | justinu, yup. |
| 23:41.21 | tainted- | Maxxed u brokeback sob |
| 23:41.23 | justinu|laptop | how kazinsky of you |
| 23:41.24 | Maxxed | lol |
| 23:41.25 | tainted- | ~pb |
| 23:41.27 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 23:41.39 | Maxxed | tainted- likes the cawk |
| 23:41.57 | justinu|laptop | uh |
| 23:42.06 | ManxPower | Maxxed, looks like the syntax for goto is different |
| 23:42.12 | Maxxed | is it? |
| 23:42.32 | Maxxed | not 100% on how to work the features crap |
| 23:42.43 | tehdely | [TK]D-Fender: lol |
| 23:42.51 | ManxPower | I'll bet a you can only specify a priority in the current extension, based on the error message |
| 23:42.59 | ManxPower | well, a priority or a label |
| 23:43.13 | Maxxed | eh |
| 23:43.22 | ManxPower | but I'm only guessing based on the error message. |
| 23:44.53 | *** join/#asterisk twilson (n=terry@69.17.122.227) |
| 23:45.00 | *** join/#asterisk hydride (n=nathan@HSE-Montreal-ppp140758.sympatico.ca) |
| 23:45.09 | ManxPower | Roxio Easy Media Creator is neither easy, nor does it seem to create media. It only leads you along, trying to be as confusing as possible, then finally you realisze you paid $80 for a problem that isn't useful to you. |
| 23:45.25 | ManxPower | problem == program |
| 23:46.32 | sevard | ManxPower: That's why you 1) don't run windows 2) don't pay for shit ass software |
| 23:46.56 | ManxPower | sevard, ffmpeg seems to create pretty crappy video, even at 8Mbps |
| 23:47.05 | tainted- | ManxPower what would u charge to put together a 25 extension pbx with 6 analog lines |
| 23:47.06 | justinu|laptop | sevard: lol |
| 23:47.12 | sevard | ManxPower: mencoder is the best. |
| 23:47.29 | ManxPower | tainted-, located where? |
| 23:47.38 | tainted- | near ssh |
| 23:47.41 | justinu|laptop | bolivia |
| 23:47.57 | ManxPower | tainted-, I'm trying to avoid analog lines. I'm still in therapy from my last run in with them. |
| 23:48.02 | [TK]D-Fender | lol |
| 23:48.02 | redondos | Can you please tell me what this means? I'm using debian testing, haven't touched anything refering to the logs: Apr 26 20:47:14 ERROR[14279]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : No such file or directory |
| 23:48.18 | justinu|laptop | it means it can't open the CDR file |
| 23:48.54 | ManxPower | tainted-, give me a real list of requirements and I can give you a more accurate quote. |
| 23:49.02 | redondos | Yes, but why isn't it there? Should it? |
| 23:49.27 | ManxPower | redondos, are you running asterisk as non-root? |
| 23:49.52 | redondos | ManxPower: nope. |
| 23:49.53 | ManxPower | or more specifically does the user Asterisk is running as have permission to create files/directories in /var/log ? |
| 23:50.02 | redondos | oh, perhaps I am, let me check |
| 23:50.18 | redondos | You are right. |
| 23:50.19 | redondos | asterisk 14095 0.0 1.4 17132 7316 ? Ssl 18:08 0:04 /usr/sbin/asterisk -p -U asterisk |
| 23:50.45 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 23:51.00 | redondos | we'll see now. |
| 23:54.33 | Maxxed | damnit |
| 23:54.44 | Maxxed | god damn my ass kiss it |
| 23:54.48 | Maxxed | heh |
| 23:54.59 | *** join/#asterisk BadPacket (n=BadPacke@67.132.43.8) |
| 23:55.10 | Maxxed | so..any idea how i get a feature to jump to a context |
| 23:55.15 | Maxxed | cuz eruah, im lost |
| 23:55.16 | Maxxed | as useual |
| 23:56.48 | Apach | Hey, where can I find a manual for linking asterisk with a termination and orgination service? |
| 23:56.54 | Apach | I can't find it on asteriskguru |
| 23:57.14 | [TK]D-Fender | Apach : Depends which, using what tech, etc.... |
| 23:57.47 | Apach | I don't know exactly what I need to use or what my options are, thats what information I am trying to find with no luck. |
| 23:59.06 | redondos | ManxPower: you were right again. thank you very much. :) |
| 23:59.49 | ManxPower | redondos, My grandmother always said I was smarter than I look. |
