| 00:07.21 | jaytee | chesstrain, here's a link to the spandsp page by the guy who wrote spandsp. |
| 00:08.16 | *** join/#asterisk jhirley (n=jhirley@adsl-161-132-200.mia.bellsouth.net) |
| 00:08.23 | jaytee | http://www.soft-switch.org/ |
| 00:10.49 | jhirley | anyone have an opinion on IPitomy ? Good , Bad, Ugly, better the a cuban cigar ? |
| 00:16.53 | *** join/#asterisk SomethingISODD (n=Dan@h208-70-61-141.pcccinc.net) |
| 00:17.09 | SomethingISODD | hello all question is there anyway i can find out what context a call is coming in on.. with sip? |
| 00:19.19 | kam187 | hmm definatly ooh323 breaking |
| 00:20.33 | SomethingISODD | ooh323 doesnt work 90% of the time use chan_h323 it works more stable i find |
| 00:21.16 | kam187 | is that included in asterisk/addons? |
| 00:22.05 | SomethingISODD | yes but you also need to install pwlib and openh323 look in your source directory |
| 00:22.21 | kam187 | sure thats easy enough |
| 00:22.58 | SomethingISODD | been running it for af ew years no real issues |
| 00:23.06 | kam187 | cool thanks |
| 00:23.08 | kam187 | will try it |
| 00:23.10 | SomethingISODD | so far i have to say asterisk 1.6.2 isnt great |
| 00:23.15 | *** join/#asterisk jaskew (n=jdaskew@netblock-66-159-217-102.dslextreme.com) |
| 00:23.47 | SomethingISODD | can anyone tell me how to confirm, what extension a guest is coming in on, via sip |
| 00:24.09 | kam187 | i'm using trunk because i need some of the subs in it |
| 00:24.22 | SomethingISODD | oh ok |
| 00:24.23 | kam187 | hmm arent there some variables that get set for that |
| 00:24.46 | kam187 | http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List |
| 00:25.35 | SomethingISODD | my issue is, call is coming in and i have set the context to demo and it keeps saying rejected even know Guess is enabled |
| 00:26.01 | kam187 | hmm wierd |
| 00:26.21 | kam187 | turn on debug in logger.conf and start it with -vvvvvvdc |
| 00:26.25 | kam187 | to see what its trying to do |
| 00:27.08 | SomethingISODD | good idea thanks |
| 00:27.43 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 00:29.17 | SomethingISODD | hrm not giving any information |
| 00:30.16 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 00:32.23 | jaskew | Anyone have a recommendation for QoS configuration on a Linux router (iptables / iproute2)? |
| 00:32.52 | jaskew | I sent a large e-mail today and my outbound audio started breaking up... |
| 00:34.32 | [TK]D-Fender | SomethingISODD: go look at the SIP DEBUG for the call. |
| 00:38.23 | SomethingISODD | <PROTECTED> |
| 00:39.26 | SomethingISODD | i have it doesnt show the context |
| 00:40.51 | kam187 | debug didnt show u anything? |
| 00:41.04 | kam187 | it should tell what exactly it failed on |
| 00:41.59 | kam187 | did u switch the commend on the console => line in logger.conf before running with option d ? |
| 00:42.18 | ChannelZ | it is: there is no extension called 4035369052 |
| 00:42.54 | SomethingISODD | yes and ChannelZ i sent an extension under [demo] with exten => _X.,1,answer |
| 00:43.04 | SomethingISODD | and it keeps saying rejected |
| 00:44.14 | ChannelZ | then perhaps this is coming into a different context based on the sip peer in sip.conf |
| 00:44.25 | ChannelZ | or you didn't reload extensions or sip.conf or something |
| 00:44.59 | SomethingISODD | i restarted asterisk in general. |
| 00:46.02 | ChannelZ | sip show users |
| 00:46.36 | ChannelZ | oops wrong window |
| 00:48.04 | ChannelZ | Something: is this coming in from an ITSP or.. |
| 00:48.27 | SomethingISODD | yes it is |
| 00:48.35 | SomethingISODD | he is also running asterisk I believe |
| 00:48.56 | ChannelZ | ok so then you are registering with him or vice-versa? |
| 00:49.27 | *** join/#asterisk joako (n=Owner@opensuse/member/joak0) |
| 00:49.36 | SomethingISODD | he send to me as a guest |
| 00:49.38 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:49.58 | ChannelZ | And you're sure you don't have anything in sip.conf that would match his host? |
| 00:52.00 | SomethingISODD | no nothing |
| 00:52.04 | SomethingISODD | thats whats confusing me |
| 00:55.03 | [TK]D-Fender | SomethingISODD: Problem is you aren't looking at what PEER its matching, and what CONTEXT * is looking in. |
| 00:58.25 | ChannelZ | pastebin your sip.conf because something is awry |
| 01:02.49 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 01:04.05 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 01:05.06 | kam187 | hmm how do i allow sip from an ip without any auth? |
| 01:05.38 | voipmonk | guest, autocreate peer, no user or pass or openser/kamailio |
| 01:05.52 | kam187 | ok |
| 01:05.54 | voipmonk | use a host=ip of the device |
| 01:06.13 | voipmonk | look at some examples in sip.conf.sample |
| 01:06.30 | voipmonk | or they may already be in your sip.conf |
| 01:06.55 | voipmonk | what are you attempting, kam187 ? :) |
| 01:07.16 | *** join/#asterisk joesuffceren (n=chatzill@ip68-104-167-226.ph.ph.cox.net) |
| 01:07.32 | kam187 | i have a local machine doing sip<->h323 |
| 01:07.39 | joesuffceren | anyone have any tips for troubleshooting oneway audio with Skype for Asterisk? |
| 01:07.53 | kam187 | so i just want to allow it to send to the asterisk without having to register etc |
| 01:08.00 | voipmonk | hows that working for you so fa,r kam187 ? :) |
| 01:08.12 | kam187 | works fine :) |
| 01:08.15 | ChannelZ | joesuffceren: which way? None from Skype->* ? |
| 01:08.28 | kam187 | i was using h323 into it, but i'm having wierd problems with chan_ooh323 |
| 01:08.40 | voipmonk | is there nat involved, kam187 ? |
| 01:08.40 | kam187 | i suspect its the patch i made to it :P |
| 01:08.46 | ChannelZ | joesuffceren: and is your * behind a firewall or have multiple interfaces? |
| 01:08.47 | voipmonk | ooh a patch |
| 01:08.48 | kam187 | nope |
| 01:09.02 | joesuffceren | call from asterisk to skype via custom extension. skype client can hear asterisk user, but asterisk user cannot hear skype client |
| 01:09.11 | carrar | kam187, is that Asterisk box on the internet? |
| 01:09.28 | carrar | cause then people could make calls via your server |
| 01:09.32 | carrar | without auth |
| 01:09.39 | kam187 | carrar: its behind a firewall |
| 01:09.56 | joesuffceren | ChannelZ: yes, * is behind a firewall. only one NIC. |
| 01:10.06 | kam187 | so i'll use an onbscure port that i'll never need just to sip between those two, and of course firewall it on the machine and with the hw firewall |
| 01:10.09 | joesuffceren | when placing call from skype to asterisk, audio works both ways |
| 01:10.23 | ChannelZ | joesuffceren: set rtp_address=x.x.x.x - where x.x.x.x is the LAN address of your * box and see if that helps |
| 01:10.34 | ChannelZ | (LAN, not WAN! I know it sounds backwards) |
| 01:11.06 | ChannelZ | Oh and bind_address=x.x.x.x to the same |
| 01:11.08 | joesuffceren | ChannelZ: what conf file does that go in? or is that just from the CLI |
| 01:11.09 | ChannelZ | both in chan_skype.conf |
| 01:11.25 | joesuffceren | will do and report back. one sec |
| 01:11.29 | ChannelZ | I had an issue where SFA was binding to localhost only |
| 01:12.19 | SomethingISODD | ChanServ i got it working thanks anyway it seems my carrier id can not match the number i am calling thats whats causing the issue |
| 01:12.43 | SomethingISODD | joesuffceren how did you get skype+ asterisk to work at all |
| 01:12.51 | SomethingISODD | i cant even figure out how to get it to interconnect |
| 01:13.11 | joesuffceren | SomethingISODD: you can only use it with accounts you created using the Skype business account manager |
| 01:13.19 | joesuffceren | so you can't just login with your existing account |
| 01:13.30 | joesuffceren | also, you have to have > 1.4.25 |
| 01:13.44 | SomethingISODD | joesuffceren can you use 1.6.2? |
| 01:13.47 | *** join/#asterisk youngproguru (n=youngpro@cpe-76-180-188-78.buffalo.res.rr.com) |
| 01:13.53 | SomethingISODD | and what does it cost for the business package do u know? |
| 01:14.16 | ChannelZ | $0 |
| 01:14.35 | SomethingISODD | really ok thanks |
| 01:15.20 | ChannelZ | just nornmal skype fees if you're dialing real numbers or need SkypeIn |
| 01:15.36 | SomethingISODD | ChannelZ how many channels does skype support to and from asterisk just the one?? |
| 01:15.40 | joesuffceren | not sure on 1.6.2. listed in readme are: Asterisk 1.4 versions >= 1.4.25, Asterisk 1.6.0 versions >= 1.6.0.6, Asterisk 1.6.1 versions >= 1.6.1.5 |
| 01:15.49 | joesuffceren | SomethingISODD: you pay per channel |
| 01:16.00 | joesuffceren | channel = concurrent call in this case |
| 01:16.36 | SomethingISODD | joesuffceren if youdont mind me asking whats the price per seat? |
| 01:16.51 | SomethingISODD | if you dont know off the top of your head i can wait till i sign up for a business account |
| 01:17.03 | joesuffceren | really reasonable. it's right at $70 per concurrent call |
| 01:17.23 | SomethingISODD | Ok thank you |
| 01:17.55 | joesuffceren | SomethingISODD: and it includes g.729 |
| 01:18.49 | SomethingISODD | oh perfect ok thank you i think i will give it try |
| 01:20.48 | joesuffceren | ChannelZ: I implemented those changes, but same problem |
| 01:21.20 | joesuffceren | any other thoughts? |
| 01:23.28 | ChannelZ | hmm.. you reloaded the module? |
| 01:23.39 | joesuffceren | restarted asterisk |
| 01:25.13 | ChannelZ | are you port-forwarding a port to the * box? |
| 01:25.24 | ChannelZ | as per bind_port in chan_skype.conf |
| 01:26.27 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 01:28.01 | *** join/#asterisk Katty (n=User@adsl-75-5-180-58.dsl.stlsmo.sbcglobal.net) |
| 01:28.10 | Katty | hello. |
| 01:28.18 | carrar | HARRO |
| 01:28.36 | carrar | I just made Costco Pumpkin Bread |
| 01:28.43 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:28.54 | joesuffceren | ChannelZ: doh, no, I'm not. missed that. BRB |
| 01:29.15 | Katty | carrar: oooh |
| 01:29.18 | Katty | carrar: recipe and photo? |
| 01:29.19 | ChannelZ | well I'm not positive it matters, I'm trying to test here. I did it by default.. |
| 01:29.41 | carrar | Added walnuts, oh I should go take a phote. It's the box thing were you add water, oil and the MIX |
| 01:29.52 | Katty | i don't suppose anyone knows where a calorie sheet for Logan's Roadhouse might be. |
| 01:30.29 | *** part/#asterisk jaskew (n=jdaskew@netblock-66-159-217-102.dslextreme.com) |
| 01:30.41 | titter | Remote host can't match request NOTIFY to call -- any ideas |
| 01:30.56 | titter | Katty: call them and ask, most of the time they have one |
| 01:31.26 | ChannelZ | joesuffceren: might not be necessary.. I just turned off the port forward on my firewall and it seems to be working still |
| 01:31.29 | ChannelZ | hmmmm |
| 01:31.48 | Katty | titter: well i asked at the resturant, they didn't know of any... |
| 01:31.52 | Katty | titter: none on their website either )= |
| 01:32.12 | titter | Katty: they suck lol, just go by what you at |
| 01:32.12 | titter | e |
| 01:32.45 | Katty | hrmm |
| 01:32.54 | Katty | i don't know the ammount of calories in a long island iced tea |
| 01:33.06 | *** join/#asterisk NicoleMun (n=ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
| 01:33.20 | coppice | it depends how long it is |
| 01:33.31 | Katty | oh it was pretty long |
| 01:34.25 | Katty | we'll just call it a bajillion calories >.< |
| 01:35.56 | *** join/#asterisk Ferrenrock (n=sampo@tri0702.urh.uiuc.edu) |
| 01:36.24 | Katty | i have a terrible habit of eating every /but/ the entree |
| 01:38.04 | Ferrenrock | hey, would it be possible to use my laptop's dial-up jack and a microphone to make phone calls with asterisk? |
| 01:38.20 | Katty | laptop, yes. |
| 01:38.22 | Katty | modem, no |
| 01:38.43 | Ferrenrock | what would you suggest instead? |
| 01:38.54 | Katty | Asterisk doesn't support modems. |
| 01:39.07 | Katty | if you want a cheap way to make calls, you might check into skype |
| 01:39.25 | Katty | that borrow a friend's server who has unlimited long distance |
| 01:39.34 | Ferrenrock | Katty: well my dorm has this phone jack with a number I get for free, including free long distance |
| 01:39.43 | Ferrenrock | but I don't have an actual landline telephone |
| 01:39.51 | Katty | then go buy one |
| 01:39.57 | Katty | they are like 10 bucks |
| 01:40.14 | Ferrenrock | I know, but I have to take the bus to get to a place that sells phones |
| 01:40.17 | carrar | cheap at road side sales |
| 01:40.18 | Katty | aww |
| 01:40.20 | carrar | cheaper |
| 01:40.25 | Katty | would you like some cheese with your whine? ;) |
| 01:40.52 | Ferrenrock | Katty: point is, I won't bother with that, I was just wondering if it was possible with my laptop |
| 01:41.01 | Ferrenrock | should I be looking into something other than asterisk? |
| 01:41.03 | Katty | you can load asterisk on your laptop |
| 01:41.10 | Katty | and connect sip channels. |
| 01:41.21 | voipmonk | sure can. |
| 01:41.22 | Katty | i'm not sure if there's any external equipment made or not |
| 01:41.54 | carrar | he sould buy a ATA and go from his laptop SIP to the ATA over ethernet then from the ATA to the jack in the wall |
| 01:41.56 | Ferrenrock | Katty: but there's no way to use this landline connection? |
| 01:41.58 | carrar | could |
| 01:42.04 | carrar | heh |
| 01:42.06 | Katty | Ferrenrock: Asterisk does not support modems. |
| 01:42.12 | Katty | Ferrenrock: Asterisk does not support modems. |
| 01:42.14 | Ferrenrock | ok |
| 01:42.15 | Katty | Ferrenrock: one more time! |
| 01:42.19 | Katty | Ferrenrock: Asterisk does not support modems. |
| 01:42.49 | Katty | seriously just go buy a phone |
| 01:42.57 | Ferrenrock | alright |
| 01:42.58 | Ferrenrock | thanks |
| 01:43.00 | carrar | order one off of Ebay |
| 01:43.07 | Katty | BYE |
| 01:43.07 | voipmonk | wow |
| 01:43.11 | Katty | HAVE A NICE EVENING |
| 01:43.16 | Katty | oh, hi monk |
| 01:43.20 | Katty | i was meaning to ask how your daughter is |
| 01:43.46 | voipmonk | she's kicking right now - wife is playing montezuma 2 on the iphone. |
| 01:43.56 | voipmonk | which happens to be my favorite iphone app right now |
| 01:44.00 | Katty | well tell her to be nicer to her mom! |
| 01:44.10 | voipmonk | heh |
| 01:44.22 | carrar | http://cgi.ebay.com/Bell-System-Cream-Rotary-Telephone-Vintage-Western-Elec_W0QQitemZ350305571605QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item518fd6d315 |
| 01:44.26 | carrar | there is a phone forhim |
| 01:44.30 | carrar | 8.99 |
| 01:44.31 | Katty | he already left |
| 01:44.33 | voipmonk | the guy signed off , carrar |
| 01:44.35 | carrar | doh |
| 01:44.42 | Katty | yeah he was all KTHXBAI |
| 01:44.43 | voipmonk | dont you love that? |
| 01:44.43 | carrar | I might order that for myself |
| 01:44.59 | Katty | i have zoiper on my laptop |
| 01:45.04 | Katty | don't use it tho |
| 01:47.27 | titter | Remote host can't match request NOTIFY to call -- I hate thie error -.- |
| 01:49.25 | joesuffceren | ChannelZ: had to install g729 |
| 01:49.57 | joesuffceren | digium tech said that is negotiates 729 on outbound but not on inbound |
| 01:50.06 | joesuffceren | life is good now |
| 01:50.36 | *** join/#asterisk Caplain (i=shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
| 01:52.54 | titter | Katty: http://www.thedailyplate.com/nutrition-calories/food/generic/long-island-iced-tea |
| 01:53.56 | Katty | thanks. |
| 01:54.41 | titter | thats a good site btw |
| 01:54.57 | Katty | i usually use nutritiondata |
| 02:00.46 | ChannelZ | joesuffceren: huh that's wierd. |
| 02:02.08 | ChannelZ | mine seems to use alaw |
| 02:02.15 | ChannelZ | err ulaw |
| 02:08.44 | *** part/#asterisk chesstrian (n=chesstri@186.83.99.12) |
| 02:11.06 | ChannelZ | hmm I seem to have toasted my g729 codec actually during one asterisk reinstall or another. |
| 02:13.24 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 02:16.48 | hardwire | hmm.. I have an older TE110P and a newer TE122P |
| 02:17.05 | Katty | is 8 too early to go to bed? |
| 02:17.09 | hardwire | the TE110P chucks HDLC abort errors.. the TE122P works flawlessly |
| 02:17.24 | hardwire | I wonder if I'm missing something obvious other than "maybe your TE110P is bad" |
| 02:17.49 | Katty | rma it |
| 02:19.38 | hardwire | TE110P is way old.. I'll have to get a new TE122P |
| 02:20.33 | Katty | ahh |
| 02:20.47 | Katty | well i'mma go nap on the couch, me thinks. my energies have left me |
| 02:21.34 | *** join/#asterisk minotaur01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 02:22.00 | coppice | Katty: 8 is a perfect time to go to bed, if you are in the early stages of a relationship |
| 02:23.18 | fenrus | =)) |
| 02:23.53 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 02:25.07 | voipmonk | n0 m0ez has caffeine, eh? |
| 02:28.00 | *** join/#asterisk jmcdowell (n=airmadne@174-154-12-142.pools.spcsdns.net) |
| 02:28.29 | jmcdowell | Anyone have any experience with Polycom phones? |
| 02:29.01 | jmcdowell | I cleared the dial patterns from the phone, but it keeps stripping off the "9" to get out. |
| 02:29.16 | *** part/#asterisk dpisites (n=cheng@dsl-67-204-18-213.acanac.net) |
| 02:29.19 | leifmadsen | jmcdowell: yes. |
| 02:29.32 | jmcdowell | Any suggestions, or good reads? |
| 02:29.52 | jmcdowell | I have read the manual, and I just can't quite grasp what the hell they are talking about. |
| 02:31.54 | *** join/#asterisk ManxPower-work (n=ewieling@216.186.151.147) |
| 02:31.55 | jmcdowell | From what I read, if I understand it. I may have to PXE boot the phone to get it to stop doing that. |
| 02:32.17 | ManxPower-work | jmcdowell: Polycom? |
| 02:35.35 | jmcdowell | yes |
| 02:35.41 | jmcdowell | Polycom 601 |
| 02:39.27 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 02:41.51 | *** join/#asterisk jakent (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
| 02:41.56 | *** join/#asterisk etnos (n=etnos@adsl-2-215-86.mia.bellsouth.net) |
| 02:47.52 | kam187 | hmm is it possible to run an app or macro in a new thread? |
| 02:48.57 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:51.03 | jmcdowell | Well.. |
| 02:51.08 | jmcdowell | Guess no luck there.. |
| 02:51.12 | voipmonk | new thread? |
| 02:51.18 | voipmonk | kam187: ? |
| 02:51.19 | jmcdowell | Thread? |
| 02:51.44 | kam187 | from what i can tell its blocking any events like hangup |
| 02:51.58 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 02:51.58 | kam187 | this seems to be a 'feature' of a macro it seems |
| 02:52.51 | kam187 | basically if i dial with a macro, neither extension will hangup untill the macro returns, and even then only sometimes |
| 02:53.07 | kam187 | on a code level, the channel state hasnt been changed either, so i cant catch that |
| 02:56.59 | jmcdowell | Is there a "dial" plan to keep polycom from stripping off the 9 at dial time? The route uses the 9 to determine where to send the call. |
| 02:58.16 | jmcdowell | I sure don't want that controlled at the phone. |
| 02:58.29 | jaytee | modify the dialplan.digitmap to include a 9 then with however many other digits you need, like 9XXXXXXX and 9XXXXXXXXXXX |
| 02:59.09 | jmcdowell | will try.. |
| 03:02.29 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:05.03 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 03:05.09 | *** join/#asterisk jakent (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
| 03:08.57 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 03:12.17 | jmcdowell | hmmmm |
| 03:12.28 | jmcdowell | It still strips off the 9 |
| 03:13.05 | jmcdowell | If I dial 913143212222 it send 3143212222 to the routes which fails because there is no match. |
| 03:13.42 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:19.33 | beek | Good evening jaytee |
| 03:19.43 | jaytee | hi beek |
| 03:19.54 | beek | Working late tonight or is this just recreational IRC use? |
| 03:20.16 | jmcdowell | wait a minute? |
| 03:20.19 | jaytee | just hanging |
| 03:20.24 | jmcdowell | <PROTECTED> |
| 03:20.45 | jaytee | jmcdowell, do you have a Polycom phone? |
| 03:20.55 | jmcdowell | Yes |
| 03:21.01 | jaytee | how did you provision it? |
| 03:21.10 | jmcdowell | I didn't i set everything and it connected |
| 03:21.16 | jmcdowell | I was reading about PXE provisioning. |
| 03:21.34 | jaytee | in the SIP Admin guide or the whitepaper from Polycom? |
| 03:21.35 | jmcdowell | But wanted to aviod if possible because I can't find good documentatin and example templates. |
| 03:21.50 | jmcdowell | The Polycom admin guide |
| 03:22.03 | jaytee | hang on a second, help is on the way |
| 03:22.14 | jmcdowell | sweet |
| 03:22.22 | beek | Don't avoid it -- embrace it. Makes provisioning so much easier. I use FTP though. |
| 03:22.29 | jmcdowell | I am very familar with PXE and ftp |
| 03:22.34 | jmcdowell | so either is no biggie |
| 03:23.05 | jaytee | yes, vsftp is far superior and using it with DHCP with option 66 makes things alot more flexible |
| 03:23.08 | jmcdowell | I guess I would have to use a domain name for an external phone and it could still provision? |
| 03:23.56 | beek | jmcdowell: Check this out: http://kfife.com/voip/ |
| 03:24.45 | jmcdowell | Wow those are small config files |
| 03:24.49 | jmcdowell | I thought they would be much larger. |
| 03:25.12 | beek | There is a default provided by Polycom. You only put into what yours what you want to override. |
| 03:25.31 | beek | I use Karl's system and it works well. |
| 03:25.54 | jaytee | jmcdowell, here's Polycom's white paper on provisioning, it includes a good fairly easy setup process. I used a combination of the section of the "the book" on setting up provisioning and this guide. |
| 03:25.57 | jaytee | http://www.polycom.com/global/documents/whitepapers/vlans_and_polycom_soundpoint_ip_desktop_ip_telephones.pdf |
| 03:26.27 | jmcdowell | .net has to be installed for xml notepad. |
| 03:26.29 | jmcdowell | grrr |
| 03:27.47 | jaytee | I use Polycom 330's and Polycom550's at work and I setup dummy config files as templates and the shell scripts copy them to files named with the new phone's mac address and SIP account ID. I match my 4 digit internal extensions to the ID in my dialplan in Asterisk |
| 03:28.36 | jaytee | so all I have to do is set the phone to boot with DHCP with Option 66, have the Option 66 string in my DCHP server pass the provisioning server's address |
| 03:29.14 | jmcdowell | hmmm |
| 03:29.17 | jmcdowell | I have never heard of that, |
| 03:29.19 | jaytee | then I just run a one of the two scripts like: ./prepphone330.sh 0004f21a1234 5555 |
| 03:29.33 | jmcdowell | I have to install .joke |
| 03:29.45 | jaytee | and that generates the config files for the phone with extension 5555 |
| 03:29.54 | jmcdowell | nice |
| 03:30.48 | jaytee | the shell scripts are simple and just use sed to match and replace strings in the template files after their copied to their MAC address replacements |
| 03:31.55 | jmcdowell | So this enormous file sip.cfg |
| 03:32.02 | fenrus | jaytee, i've built the same for Cisco 7940/7960 |
| 03:32.06 | jmcdowell | When I open it in xml 2007 or what ever.. |
| 03:32.11 | jmcdowell | What is that going to look like? |
| 03:32.12 | jaytee | but the provisioning method is the best. you can't modify things like custom alert info to modify the ringtone based on a dialed number etc. or the nightmare of using the web gui to change 200 phones to change the registration expiration interval |
| 03:32.13 | fenrus | and it adds to sip-config and voicemail :) |
| 03:32.46 | dlynes_laptop | Does anyone know of a stripped down version of voicemail for asterisk? i.e. one with very simplistic prompts that old fogeys can understand? |
| 03:32.55 | jaytee | fenrus, sure beats editing each config file one at a time doesn't it? :-) |
| 03:33.01 | fenrus | jaytee, well - my script use a database as data-source, not CLI-parameters.. so it's really easy to rebuild and change alot of stuff fast. |
| 03:33.30 | fenrus | i'll be migrating to MySQL soon, to support my linksys pap2t from the same provisioning-tool |
| 03:33.59 | jaytee | dlynes_laptop, all the voicemail prompts are in /var/lib/asterisk/sounds and have a vm- prefix, like vm-goodbye.wav |
| 03:34.11 | dlynes_laptop | jaytee, and? |
| 03:34.13 | jaytee | so you could record your own and replace them |
| 03:34.30 | dlynes_laptop | jaytee, how does that mitigate all the bs with all the other folders, and what-not? |
| 03:34.49 | beek | Gotta run gang... CU and GN |
| 03:35.00 | dlynes_laptop | jaytee, then they accidentally hit '3' to save it in the friends and family folder, and can't figure out where that voicemail is, because that folder doesn't exist on the prompts |
| 03:35.18 | dlynes_laptop | jaytee, i need more than just the prompts changed |
| 03:35.49 | dlynes_laptop | jaytee, that's the solution i'm looking at right now...changing prompts and changing the C code...I just don't want to go down that road if I don't have to |
| 03:36.24 | jaytee | dlynes_laptop, not sure what you want to accomplish but you can modify some of it. I don't use Asterisk's voicemail. I use Exchange Unified Messaging as a back end voicemail system that integrates with my email so messages can be played by calling the main voicemail number or from my Outlook inbox when they show up as a new voicemail message attachment. |
| 03:36.28 | dlynes_laptop | jaytee, my life is hell trying to support dementia patients as it is...I don't want it anymore hellish :) |
| 03:36.51 | dlynes_laptop | jaytee, yeah...if that's all i needed, i'd just use app_minivm.so |
| 03:37.14 | dlynes_laptop | jaytee, however, old fogeys have enough trouble with the 'beeping' in the background (call waiting)...they can't fathom using internet |
| 03:38.08 | dlynes_laptop | jaytee, we do have some assisted living residents that have internet, and even most of them know what call waiting is, but hate it, and hate the confusing voicemail menus even more |
| 03:38.25 | jaytee | dlynes_laptop, yeah I can imagine. Not sure what options in Comedian Mail that you can tweak without messing with the C code. |
| 03:38.44 | dlynes_laptop | jaytee, i can modify the c code...it's not an issue |
| 03:38.54 | dlynes_laptop | jaytee, just a hell of a lot of work that I'd rather not do, if I don't have to |
| 03:40.23 | voipmonk | dementia patients.. |
| 03:40.44 | voipmonk | what do you want to tweak? |
| 03:40.46 | dlynes_laptop | alzheimers, dementia, schizophrenia, |
| 03:42.00 | dlynes_laptop | voipmonk, get rid of all folders so that only inbox and saved still exist, get rid of unavailable and temporary voicemail greetings |
| 03:42.08 | voipmonk | no problem |
| 03:42.23 | voipmonk | u have to remove those options in the code |
| 03:42.27 | voipmonk | easily commented out |
| 03:42.27 | dlynes_laptop | voipmonk, and then change the key codes for mailbox options so that they're exactly the same as telus |
| 03:42.35 | voipmonk | oh good lord |
| 03:42.37 | voipmonk | ok |
| 03:42.43 | voipmonk | that can be changed, too in the source |
| 03:42.48 | dlynes_laptop | yeah |
| 03:42.50 | voipmonk | anything else? |
| 03:42.56 | dlynes_laptop | but now you see my dilemna :) |
| 03:43.00 | voipmonk | ive only done it in 1.4.x |
| 03:43.11 | voipmonk | not a dilemna |
| 03:43.15 | coppice | do schizophrenia sufferers with alzheimers forget about the voices in their heads? |
| 03:43.21 | voipmonk | u have to dive in in your dev platform and get er done |
| 03:43.46 | dlynes_laptop | voipmonk, yeah...just thought if there was a precanned solution, i could save the company some money |
| 03:44.29 | dlynes_laptop | voipmonk, maybe what I can do is customize it for telus |
| 03:44.36 | dlynes_laptop | voipmonk, charge that out to the company |
| 03:45.03 | dlynes_laptop | voipmonk, and then customize it a bit more so that it's tailorable by config file, and then commit that back to the asterisk project |
| 03:45.12 | voipmonk | ok |
| 03:46.25 | *** join/#asterisk seanjohn (n=john@static-173-50-101-14.nrflva.east.verizon.net) |
| 03:47.18 | seanjohn | with the dial command, how do you execute commands after you have connected the call, such as dial more numbers into a conference? |
| 03:50.24 | jaytee | seanbright, either using features.conf or using the features built into most voip phones but most voip phones only let you conference 3 people at a time. |
| 03:53.09 | dlynes_laptop | seanjohn, core show application dial will show you the D(...) option to dial() |
| 03:53.26 | seanjohn | G(context^exten^pri): If the call is answered, transfer both parties to the specified context and extension. The calling party is transferred to priority x, and the called party to priority x+1. |
| 03:53.40 | dlynes_laptop | seanjohn, assuming you mean send additional dtmf tones after it's connected |
| 03:54.02 | dlynes_laptop | seanjohn, or do you mean execute additional code in the dialplan after it's connected? |
| 03:54.30 | seanjohn | so if I want the calling party to go to called,s,1 the callee will go to exten => called,s,101 ???????? |
| 03:54.31 | *** join/#asterisk Znuff (n=ibm86@2001:0:53aa:64c:2c26:2f82:a65a:7c98) |
| 03:54.44 | jaytee | seanjohn, if you're looking to create a way to dial a number and have several phones called into a conference you could use call files with local channels to dial the phone and transfer the call to a meetme conference |
| 03:55.09 | dlynes_laptop | seanjohn, the M(...) option to dial, then |
| 03:56.29 | seanjohn | or would the callee go to s,1+1 ? |
| 03:57.04 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 03:58.56 | seanjohn | ? |
| 04:00.27 | dlynes_laptop | seanjohn, they'll go to whatever you specify in M(...) |
| 04:01.10 | dlynes_laptop | seanjohn, you can also use G(...) if you want a goto, instead of a macro |
| 04:03.16 | *** join/#asterisk s0lid (n=s0lid@122.55.59.247) |
| 04:03.24 | seanjohn | i'm referring to G |
| 04:03.37 | seanjohn | I want the legs to still be separated |
| 04:05.28 | dlynes_laptop | seanjohn, G(mycontext^xxx^nnn), where mycontext is your dialplan context, xxx is your extension, and nnn is your priority |
| 04:07.33 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 04:14.11 | jmcdowell | omg |
| 04:14.21 | jmcdowell | this polycom config file is dumb |
| 04:22.55 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 04:32.14 | Katty | hi |
| 04:33.09 | jmcdowell | Katty hi |
| 04:33.16 | Katty | jmcdowell: hi. |
| 04:33.32 | Katty | i got a charlie horse. |
| 04:34.27 | Katty | :< |
| 04:37.18 | jmcdowell | Wait a minute... |
| 04:37.22 | jmcdowell | Katty, that sux.. |
| 04:37.26 | jmcdowell | Wait a minute.. |
| 04:37.34 | jmcdowell | I can't run a DHCP server on the phone systemm.. |
| 04:37.39 | jmcdowell | That would clash with the router. |
| 04:38.01 | Katty | well |
| 04:38.03 | Katty | yes, and no |
| 04:38.15 | jmcdowell | St. Louis mo? |
| 04:38.17 | jmcdowell | Really? |
| 04:38.17 | Katty | you can hand out one range with your router, and another range with your box. |
| 04:38.25 | jmcdowell | Hmmm... |
| 04:38.25 | Katty | not exactly, but close. |
| 04:38.34 | jmcdowell | St. Clair, MO |
| 04:38.42 | Katty | idk where that is |
| 04:38.55 | jmcdowell | About 30 minutes past six flags |
| 04:39.02 | Katty | ohhh that's way up north |
| 04:39.14 | jmcdowell | No, WEST |
| 04:39.18 | jmcdowell | 44 west |
| 04:39.20 | Katty | it's north |
| 04:39.22 | Katty | from me. |
| 04:39.25 | Katty | 2.5hrs north |
| 04:39.25 | jmcdowell | Ahhh |
| 04:39.27 | Katty | as i am in cape. |
| 04:39.37 | jmcdowell | Ahh most of my family is from Cape.. |
| 04:39.47 | jmcdowell | The Seyers and McDowells |
| 04:39.55 | Katty | never heard of them |
| 04:40.00 | Katty | but i'm not a native cape person |
| 04:40.03 | jmcdowell | So is that POS Rush Limbash |
| 04:40.07 | jmcdowell | Ahh |
| 04:40.12 | Katty | hehe |
| 04:40.13 | Katty | yesh. |
| 04:40.13 | jmcdowell | Eeeew, must be hard then. |
| 04:40.18 | Katty | nope |
| 04:40.21 | Katty | i am well liked. |
| 04:41.01 | jmcdowell | I am pretty sure that running 2 dhcp server will cause issues, despite breaking the networks apart. |
| 04:41.23 | jmcdowell | And, I need them to be on the same subnet because the box will route external extensions in from the outside. |
| 04:41.41 | p3nguin | WHAT DOES THAT MEAN? |
| 04:41.59 | Katty | jmcdowell: well. |
| 04:42.02 | Katty | jmcdowell: you could fake it. |
| 04:42.21 | Katty | jmcdowell: but it would probably be infinately easier if you just set the few boxes static. |
| 04:42.21 | jmcdowell | Fake it? |
| 04:42.39 | jmcdowell | I have 14 phones |
| 04:42.41 | Katty | you can put it on another network...with 255.0.0.0 or somethin |
| 04:43.09 | jmcdowell | I could do that, but when using dhcp, the client uses 0.0.0.0/0.0.0 |
| 04:43.13 | p3nguin | Why such a large subnet? |
| 04:43.22 | Katty | p3nguin: so he can see other subnets. |
| 04:43.52 | jmcdowell | Hmmmmm... |
| 04:44.18 | p3nguin | I don't get it. |
| 04:44.41 | Katty | 255.255.255.0 can only see other networks where the last digit is different |
| 04:44.42 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 04:44.50 | Katty | so 192.168.0.1-254 |
| 04:45.10 | jmcdowell | but all clients I have ever seen, search ip 0.0.0.0 netmask 0.0.0.0 |
| 04:45.17 | Katty | 255.255.0.0 would be 0.1 - 254.254 |
| 04:45.19 | jmcdowell | Which would traverse that |
| 04:45.44 | jmcdowell | Can I specify the subnet in the Polycom phone. |
| 04:45.46 | jmcdowell | I wonder.. |
| 04:46.03 | Katty | yes. you can |
| 04:46.04 | jmcdowell | All of this, because of a damn dial plan that can't be bipassed. |
| 04:46.20 | dlynes_laptop | Katty, where the last 8 bits are different |
| 04:46.31 | jmcdowell | I am perfectly fine setting them up manually |
| 04:46.41 | jmcdowell | but it doesn't seem to work in regard to the dial plan. |
| 04:47.27 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
| 04:47.28 | p3nguin | dlynes_laptop: I'm waiting on her to start ANDing for us. |
| 04:47.40 | dlynes_laptop | p3nguin, ? |
| 04:47.50 | p3nguin | binary conversion |
| 04:48.04 | dlynes_laptop | p3nguin, you haven't converted to the binary religion yet? |
| 04:48.14 | p3nguin | chokes |
| 04:51.15 | dlynes_laptop | jmcdowell, sounds like you've got a bogus phone |
| 04:51.29 | dlynes_laptop | jmcdowell, maybe you should've pitched in the extra money to buy a grandstream? |
| 04:52.01 | dlynes_laptop | jmcdowell, i know their dialplans work |
| 04:55.07 | jmcdowell | You know they work? |
| 04:55.20 | jmcdowell | So do I, so well in fact that I cannot turn it off. |
| 04:55.52 | jmcdowell | dlynes_laptop : Grand stream? They were cheaper than the Polycom. |
| 04:56.24 | jmcdowell | There is nothing wrong with Polycom, the thing sounds outstanding, I just have to figure out how to disable that damn dial plan. Even if it means that I have to provision them. |
| 04:57.15 | jmcdowell | I could always change the trunk to append the nine on it's own and not worry about it. |
| 04:57.19 | Katty | polycoms are my favorite |
| 05:00.57 | dlynes_laptop | jmcdowell, it was a bit of wry sarcasm...sorry...should've put a smiley face winking there, i guess |
| 05:01.59 | dlynes_laptop | Katty, considering there's so many of you polycom users in here, i thought someone would have had a solution to jmcdowell's extremely simple problem by now? |
| 05:02.25 | Katty | well sure, if people were around |
| 05:02.32 | Katty | but it's friday night. what do you really expect. |
| 05:02.35 | dlynes_laptop | i would think his problem would be fairly common |
| 05:02.52 | dlynes_laptop | i.e. probalby in the list of top ten faq's, for that matter |
| 05:03.04 | dlynes_laptop | well |
| 05:03.15 | dlynes_laptop | i expect people would think asterisk was more important than a social life |
| 05:03.18 | dlynes_laptop | don't you? |
| 05:03.49 | dlynes_laptop | I think even your squirrels are nuts about it |
| 05:03.56 | dlynes_laptop | :) |
| 05:07.30 | Katty | i'm playing STO with ryan |
| 05:08.02 | Katty | we already went out for dinner. |
| 05:08.07 | Katty | etc. |
| 05:08.46 | Katty | and no, i think a social life is a very healthy part of a person's life. |
| 05:09.36 | coppice | social lives are people who lack the intellect to spend their Friday evenings developing better technology :-\ |
| 05:10.31 | Katty | a lack of regular human social contact can bring on lots of things, like depression |
| 05:12.29 | jmcdowell | Yeah |
| 05:12.34 | jmcdowell | I have 5 kids, I have no life |
| 05:12.59 | Katty | sure you do |
| 05:13.01 | Katty | they ARE your life (= |
| 05:13.05 | coppice | Katty: social contact can also cause depression |
| 05:13.07 | jmcdowell | Yeah right |
| 05:13.13 | Katty | coppice: doesn't in me :P |
| 05:13.22 | jmcdowell | they are my high blood pressure, nose bleeds and source of drama |
| 05:13.22 | Katty | coppice: i find it very theraputic |
| 05:13.42 | Katty | jmcdowell: and you would no doubt give your life for them |
| 05:13.56 | coppice | he has already |
| 05:15.23 | *** join/#asterisk RonaldRaygun (n=RonaldRa@d174h72.resnet.uconn.edu) |
| 05:15.30 | RonaldRaygun | Hi everyone |
| 05:15.48 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 05:16.31 | RonaldRaygun | Hello? |
| 05:16.39 | Katty | shh, everyone's napping |
| 05:18.02 | RonaldRaygun | Oh well |
| 05:18.28 | RonaldRaygun | I'm curious, my MagicJack subcription expired. Is there any way I can repurpose the hardware for asterisk? |
| 05:20.03 | ChannelZ | It might work as a sacrafice.. |
| 05:23.16 | Katty | i don't really get "repurpose" |
| 05:25.32 | RonaldRaygun | Well, the magicjack subscription ended, so I can't call the traditional way. I was thinking, using the magicjack hardware to connect a regular landline phone (or maybe a whole house system) to a computer with asterisk installed, so I can at least play around with the PBX possibliities. |
| 05:25.48 | RonaldRaygun | Or maybe I don't really understand how these fancy applications work. |
| 05:28.30 | p3nguin | You want to try to make a Magic Jack ATA. |
| 05:28.34 | dlynes_laptop | RonaldRaygun, what is the 'magicjack' hardware, specifically? |
| 05:28.44 | RonaldRaygun | the usb dongle. |
| 05:28.54 | p3nguin | dlynes_laptop: You haven't seen MagicJacks? |
| 05:29.05 | Pan3D | magic jack constists of the device and the app. You'd have to know how the app communicates with the dongle |
| 05:29.25 | p3nguin | Picture a USB-to-RJ45 adapter. |
| 05:29.31 | RonaldRaygun | I think it's RJ-11 |
| 05:29.49 | Pan3D | it'll be over USB, which is an easy format for basic drivers -- but the format of the actual commumnications, you'd have to figure out. |
| 05:30.07 | RonaldRaygun | As I envision it, the magicjack dongle (tigerjet sip card or something like that) will replace this: http://store.digium.com/telephony_card_selector.php |
| 05:30.33 | dlynes_laptop | RonaldRaygun, have you tried 'http://www.magicjacksupport.com/magicjack-patch-for-asterisk-updated-t7243.html'? |
| 05:30.38 | dlynes_laptop | p3nguin, nope |
| 05:30.40 | p3nguin | Trust me, you aren't the first person to want to make his Magic Jack into an ATA to use with Asterisk. There is probably a forum out there with hundreds of people just like you that can help you. |
| 05:30.55 | dlynes_laptop | p3nguin, i think i seen some magicjack store a little while ago |
| 05:31.25 | p3nguin | http://honestinfomercialreviews.com/wp-content/uploads/2009/08/magic_jack_review.jpg |
| 05:31.50 | dlynes_laptop | p3nguin, i c |
| 05:31.56 | dlynes_laptop | p3nguin, and is it a hunk of junk? |
| 05:32.00 | RonaldRaygun | I know I'm not the first. The other solutions I've seen between MJ and asterisk involve just using the MJ hardware so I can connect my phone to my computer, and then make/take calls through a different VOIP service |
| 05:32.06 | p3nguin | dlynes_laptop: That's what I hear, anyway. |
| 05:32.09 | dlynes_laptop | p3nguin, i.e. akin to an x101p, or an x100p? |
| 05:32.14 | RonaldRaygun | MJ is an okay service |
| 05:32.16 | dlynes_laptop | p3nguin, or maybe even worse? |
| 05:32.19 | RonaldRaygun | but not nearly as useful as ooma |
| 05:32.32 | Pan3D | the big thing is this new device they've announced |
| 05:32.33 | p3nguin | dlynes_laptop: I figure it's really no different than an HSP 56k modem. |
| 05:32.41 | dlynes_laptop | RonaldRaygun, did you or did you not click on that link i gave you? |
| 05:32.43 | Pan3D | for VoIP over cell |
| 05:33.16 | RonaldRaygun | I don't understand what I'm looking at dlynes_laptop |
| 05:33.26 | dlynes_laptop | RonaldRaygun, hire a techy then |
| 05:33.33 | Pan3D | heh |
| 05:33.40 | RonaldRaygun | I will once I know what it is I want lol |
| 05:34.05 | dlynes_laptop | RonaldRaygun, it's instructions that tell you how to patch the asterisk source to be able to use magicjack with asterisk as a trunk |
| 05:34.18 | p3nguin | dlynes_laptop: I think that uses Asterisk on MJ's service. He's wanting to use the MJ dongle as an ATA for Asterisk. |
| 05:34.29 | dlynes_laptop | p3nguin, oh |
| 05:34.38 | dlynes_laptop | p3nguin, what the hell for? |
| 05:34.46 | Pan3D | RonaldRaygun: what the hell for? |
| 05:34.55 | RonaldRaygun | Because my magicjack subscription ended and I already have a VOIP solution |
| 05:35.01 | p3nguin | 'Cause he has the dongle and a real ATA will cost another $30. |
| 05:35.02 | Pan3D | ummm.... |
| 05:35.08 | Pan3D | lol |
| 05:35.15 | dlynes_laptop | p3nguin, and $30 will break his bank account, or something? :) |
| 05:35.20 | p3nguin | perhaps |
| 05:35.23 | Pan3D | that is... ass backwards |
| 05:35.26 | p3nguin | Let is take up a collection. |
| 05:35.29 | RonaldRaygun | Rather not buy new hardware |
| 05:35.30 | RonaldRaygun | haha |
| 05:35.31 | p3nguin | Let us, rather |
| 05:35.38 | dlynes_laptop | RonaldRaygun, dood |
| 05:35.51 | RonaldRaygun | But I guess there are more options than what I see here? http://store.digium.com/telephony_card_selector.php |
| 05:35.52 | dlynes_laptop | RonaldRaygun, you hang out at the 56th st bridge? |
| 05:36.15 | p3nguin | A PAP2T-NA will run you $34 on ebay (with free shipping). |
| 05:36.17 | dlynes_laptop | RonaldRaygun, yeah...way more |
| 05:36.53 | RonaldRaygun | I'm guessing all PCI-based |
| 05:36.55 | dlynes_laptop | RonaldRaygun, and if you're super cheap, search for the term 'x100p' on ebay |
| 05:37.02 | p3nguin | The PAP2T will be almost exactly the same as MJ's dongle, but will give you POTS phone to Ethernet rather than to USB. |
| 05:37.05 | RonaldRaygun | As in, I can't repurpose an aging dell inspiron 5150 |
| 05:37.08 | dlynes_laptop | RonaldRaygun, you can pick those up for about $7 plus shipping |
| 05:37.37 | RonaldRaygun | I guess it sucks being a college student lol. |
| 05:37.50 | dlynes_laptop | RonaldRaygun, dood...which class? |
| 05:37.56 | dlynes_laptop | RonaldRaygun, electronics or double e? |
| 05:37.58 | p3nguin | ronaldraygun: I run a Gateway 3400 PIII 933MHz box... I'm sure your hardware is better than mine. |
| 05:38.01 | RonaldRaygun | Finance |
| 05:38.16 | dlynes_laptop | RonaldRaygun, ah....was going to say you could etch your own telephony card |
| 05:38.26 | dlynes_laptop | RonaldRaygun, if you were electrically inclined |
| 05:38.33 | RonaldRaygun | p3nguin, specs-wise yes. But I don't think this laptop has a PCI slot =) |
| 05:38.52 | p3nguin | ronaldraygun: You really only need one Ethernet port. |
| 05:39.13 | p3nguin | Why did you want a PCI card, anyway? |
| 05:39.18 | RonaldRaygun | I didn't |
| 05:39.32 | RonaldRaygun | I was under the assumption that the vast majority of these ATAs (I think) are PCI-based |
| 05:39.33 | p3nguin | If you have Ethernet, that's good enough. |
| 05:39.43 | RonaldRaygun | But apparently there are ethernet versions |
| 05:39.47 | p3nguin | There are zero PCI ATAs that I am aware of. |
| 05:39.52 | RonaldRaygun | Uhh |
| 05:40.02 | RonaldRaygun | These things: http://store.digium.com/telephony_card_selector.php |
| 05:40.21 | p3nguin | ATAs are normally a small standalone box. |
| 05:40.37 | p3nguin | http://www.google.com/products/catalog?hl=en&source=hp&q=pap2t-na&um=1&ie=UTF-8&cid=16639585396118339203&ei=y4taS_y0OIG-NqLfpJEP&sa=X&oi=product_catalog_result&ct=result&resnum=3&ved=0CB0Q8wIwAg#ps-sellers |
| 05:41.03 | dlynes_laptop | RonaldRaygun, you might be able to find a digium s100u kicking around somewhere |
| 05:41.09 | dlynes_laptop | RonaldRaygun, it's a usb fxs port device |
| 05:41.41 | p3nguin | Seriously, the PAP2T is $34 on ebay. |
| 05:42.05 | RonaldRaygun | Okay. I obviously have a lot to learn. Where should I start reading on PBX technilogies with regard to Asterisk? |
| 05:42.54 | dlynes_laptop | RonaldRaygun, /usr/local/src/asterisk-1.6.1.13/configs/*.sample |
| 05:42.59 | RonaldRaygun | What I had originally intended to do was run the asterisk server in a VM, and then connect peripherals to that VM. That way, I don't tie up potentially usable resources on that laptop. |
| 05:43.09 | dlynes_laptop | RonaldRaygun, also voip-info.org and the book |
| 05:43.10 | dlynes_laptop | ~thebook |
| 05:43.11 | infobot | i guess thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 05:43.28 | p3nguin | So you'll add overhead and use more resources by installing virtual machine software... good plan. |
| 05:43.29 | dlynes_laptop | RonaldRaygun, you were going to run the asterisk server in a voicemail? |
| 05:43.40 | RonaldRaygun | VM => Virtual machine |
| 05:43.51 | dlynes_laptop | RonaldRaygun, ok...when you're on a channel talking about phone systems |
| 05:43.52 | RonaldRaygun | and p3nguin, fileservers are important too =D |
| 05:43.58 | dlynes_laptop | RonaldRaygun, VM means voicemail, not virtual machine |
| 05:44.02 | RonaldRaygun | Got it |
| 05:44.11 | dlynes_laptop | RonaldRaygun, VPS/VPE/... all mean virtual machine |
| 05:44.17 | voipmonk | or VE |
| 05:44.21 | dlynes_laptop | or that |
| 05:44.30 | voipmonk | now im interested... |
| 05:44.32 | voipmonk | scrolls up |
| 05:44.48 | p3nguin | ronaldraygun: What OS is on your laptop? |
| 05:45.12 | RonaldRaygun | The host is win xp, guest 1 is freebsd and guest 2 would be asterisk |
| 05:45.26 | p3nguin | that... doesn't make sense to me. |
| 05:45.29 | dlynes_laptop | RonaldRaygun, you mean guest 2 would be linux? |
| 05:45.35 | dlynes_laptop | RonaldRaygun, asterisk isn't an OS |
| 05:45.35 | p3nguin | Asterisk is an application, not an OS. |
| 05:45.42 | p3nguin | And you can run Asterisk ON FreeBSD. |
| 05:45.43 | p3nguin | I do. |
| 05:45.44 | RonaldRaygun | okay, linux running asterisk |
| 05:46.07 | RonaldRaygun | Hmmm, maybe it might be worth putting it all in one machine |
| 05:46.28 | dlynes_laptop | RonaldRaygun, well, seeing as how you're in finance |
| 05:47.02 | dlynes_laptop | RonaldRaygun, it might be better to take windows off of there, so your spreadsheets don't get lost someday, when windows is feeling drunk |
| 05:47.12 | dlynes_laptop | RonaldRaygun, just put everything on freebsd or linux, and use openoffice, instead |
| 05:47.24 | p3nguin | I'll assume you mean OpenOffice.org. |
| 05:47.46 | dlynes_laptop | semantics...sheesh |
| 05:47.57 | p3nguin | Not really, open office is something else... different. |
| 05:48.13 | dlynes_laptop | how so? |
| 05:48.16 | RonaldRaygun | I'm not sure if you're always like this or what. dlynes, if there is a newbie chat elsewhere, let me know |
| 05:48.26 | p3nguin | http://www.openoffice.org/about_us/summary.html |
| 05:48.31 | p3nguin | http://www.openoffice.org/FAQs/faq-other.html#4 |
| 05:48.35 | p3nguin | http://www.rehuel.com/2007/04/25/openofficeorg-vs-open-office/ |
| 05:48.39 | dlynes_laptop | RonaldRaygun, i was just being facetious :) |
| 05:48.45 | RonaldRaygun | and OpenOffice is garbage for what I do. VBA is mighty useful when working with large 100MB+ spreadsheets |
| 05:48.49 | dlynes_laptop | RonaldRaygun, i'm bored |
| 05:48.53 | dlynes_laptop | RonaldRaygun, and it's friday night |
| 05:49.02 | RonaldRaygun | O_o... |
| 05:49.13 | RonaldRaygun | Okay, you're in the states |
| 05:49.18 | dlynes_laptop | RonaldRaygun, no |
| 05:49.19 | RonaldRaygun | See here, it's Saturday noon |
| 05:49.23 | dlynes_laptop | RonaldRaygun, but nice try |
| 05:49.31 | RonaldRaygun | well that timezone at least |
| 05:49.35 | dlynes_laptop | yes |
| 05:49.38 | RonaldRaygun | North/south america |
| 05:49.55 | dlynes_laptop | and if you're at noon |
| 05:50.00 | dlynes_laptop | it sounds like you're in asia |
| 05:50.06 | RonaldRaygun | Yes |
| 05:50.30 | dlynes_laptop | more specifically taiwan, china, hong kong, malaysia, or australia |
| 05:50.39 | RonaldRaygun | Singapore, but you were close enough |
| 05:50.56 | dlynes_laptop | Yeah...just know you're on Chinese time zone |
| 05:51.19 | dlynes_laptop | 15 hours ahead of Vancouver time |
| 05:51.25 | dlynes_laptop | 12 hours ahead of Toronto time |
| 05:51.31 | RonaldRaygun | I get it =D |
| 05:51.34 | p3nguin | In Singapore, but bouncing off University of Connecticut's network. Nice. |
| 05:51.52 | dlynes_laptop | p3nguin, to defeat the great Internet firewall, probably |
| 05:51.53 | RonaldRaygun | Mhmm |
| 05:52.11 | dlynes_laptop | erm |
| 05:52.12 | dlynes_laptop | nvm |
| 05:52.18 | dlynes_laptop | he's in Singapore, not China |
| 05:52.23 | p3nguin | :) |
| 05:52.27 | RonaldRaygun | Easy to tunnel to Uconn when you know people who can help provide an end. |
| 05:53.04 | dlynes_laptop | RonaldRaygun, so you can watch boxee, or something? |
| 05:53.12 | RonaldRaygun | Hmm? |
| 05:53.26 | dlynes_laptop | RonaldRaygun, some service that insists that you're in the US, or denies you access |
| 05:53.39 | dlynes_laptop | I really hate boxee |
| 05:53.47 | RonaldRaygun | Hulu for me. |
| 05:53.49 | dlynes_laptop | and hulu |
| 05:53.52 | dlynes_laptop | they both suck |
| 05:53.58 | RonaldRaygun | Nothing better |
| 05:53.59 | dlynes_laptop | can't watch them from Canada |
| 05:54.30 | nix8n82 | tunnel |
| 05:54.35 | *** join/#asterisk joako (n=Owner@opensuse/member/joak0) |
| 05:54.47 | dlynes_laptop | nix8n82, yeah...would have to tunnel through my account on 1 and 1 |
| 05:54.57 | dlynes_laptop | nix8n82, such a pain in the ass, thoguh |
| 05:56.06 | nix8n82 | yeah, but not hard to do if you really feel the urge to watch |
| 05:57.57 | dlynes_laptop | yeah...but the work server |
| 05:58.06 | dlynes_laptop | so, if the bandwidth goes over our allotment |
| 05:58.17 | dlynes_laptop | there'll be a witchhunt for me :) |
| 05:58.50 | RonaldRaygun | Ok, lemme start over. There was an interesting article I read sometime ago about Asterisk being a business phone for the consumer. Specifically, creating extensions for callers to dial through. At least that way, I can better keep on top of the stuff I'm dealing with. I have a MagicJack that recently expired. The software is now useless, but the hardware is still viable. As I understand Asterisk, there is some sort of VoIP connection c |
| 05:58.50 | RonaldRaygun | onnecting the computer to the telephony world. The computer would handle the voicemails, and any other fun feature I come across. The magicjack would simply allow me to connect a regular RJ-11 phone and everything else would be normal. How best can I achieve this? |
| 05:59.32 | nix8n82 | right and it's probably not worth paying for your own server in the us |
| 06:00.35 | Pan3D | ? |
| 06:00.36 | p3nguin | ronaldraygun: As I already mentioned, spend $34 on a PAP2T and scrap the idea of using your MJ dongle. |
| 06:00.45 | Pan3D | nix8n82: how do you figure? |
| 06:01.04 | RonaldRaygun | What I'm looking for is the intelligent voicemail. |
| 06:01.18 | voipmonk | build it |
| 06:01.19 | RonaldRaygun | Needn't be with the MJ dongle, just need a way to access it |
| 06:01.33 | RonaldRaygun | Access the voicemail that is |
| 06:01.42 | p3nguin | PAP2T? |
| 06:01.50 | p3nguin | Maybe a PAP2T would be of some use. |
| 06:01.53 | p3nguin | perhaps. |
| 06:01.57 | nix8n82 | well to him, I would set up an AWS cloud server and use them, if I really had the need or desire |
| 06:01.58 | voipmonk | the MJ hardware is not viable without an account with MJ |
| 06:02.29 | voipmonk | its done - go burn it , put it on youtube and buy a pap2tna or spa941 or polycom 330 w/AC adapter |
| 06:04.32 | Pan3D | yeah, that's sort of the irony in this... RonaldRaygun, you want to salvage the propriatary hardware when instead you could buy something that would be open to multiple systems moving forward. |
| 06:04.35 | p3nguin | ronaldraygun: It is really as easy and signing up with the ITSP of your choice, obtaining a phone number, setting up Asterisk on an internet-enabled computer, and connecting the computer and a PAP2T to a switch/hub. |
| 06:05.16 | p3nguin | If you don't want to get an ATA such as the PAP2T, you can buy an IP phone instead of reusing your analog phones on an ATA. |
| 06:05.19 | RonaldRaygun | I suppose it's ironic. I guess I'm not familiar with all the technologies involved with this yet. |
| 06:05.36 | Pan3D | where's that video? |
| 06:05.43 | Pan3D | anyone got the link? |
| 06:05.45 | p3nguin | ~itsp |
| 06:05.46 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 06:05.52 | dlynes_laptop | hrm...guess squid can't proxy hulu traffic, either |
| 06:06.08 | Pan3D | RonaldRaygun: watch this... http://revision3.com/systm/asterisk |
| 06:06.27 | Pan3D | very informative about the wholve process |
| 06:12.19 | *** part/#asterisk kam187 (n=kam187@81.179.8.102) |
| 06:15.58 | *** join/#asterisk zamba (i=marius@flage.org) |
| 06:19.26 | nix8n82 | ~itsplist |
| 06:19.34 | nix8n82 | ~itsplist-us |
| 06:19.35 | infobot | hmm... itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 06:21.41 | p3nguin | No Flowroute nor VoIP.ms? |
| 06:22.32 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 06:28.54 | *** join/#asterisk aceking5 (n=aceking5@71-94-132-102.static.mtpk.ca.charter.com) |
| 06:54.55 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 06:55.40 | sun28 | moin \o/ |
| 06:56.38 | ChannelZ | infobot ~itsplist-us is also http://flowroute.com , http://voip.ms |
| 06:56.38 | infobot | ChannelZ: okay |
| 07:03.48 | profxavier | Pan3D>RonaldRaygun: watch this... http://revision3.com/systm/asterisk |
| 07:03.49 | profxavier | <Pan3D>very informative about the wholve process |
| 07:03.51 | profxavier | I must say, that is an excellent link |
| 07:06.48 | p3nguin | ~itsplist-us |
| 07:06.49 | infobot | methinks itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 07:06.49 | ChannelZ | I get a big error here. |
| 07:07.05 | p3nguin | It makes my browser die when I try to play it. |
| 07:08.54 | ChannelZ | oh.. oops, these stupid ~ triggers make the syntax weird |
| 07:09.00 | ChannelZ | infobot forget ~itsplist-us |
| 07:09.00 | infobot | i forgot ~itsplist-us, ChannelZ |
| 07:09.08 | ChannelZ | infobot itsplist-us is also http://flowroute.com , http://voip.ms |
| 07:09.09 | infobot | ChannelZ: okay |
| 07:09.42 | p3nguin | ~itsplist-us |
| 07:09.43 | infobot | well, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 07:10.08 | p3nguin | yayz |
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| 09:18.04 | seanjohn | with the dial command, how do you execute commands after you have connected the call, such as dial more numbers into a conference? |
| 09:19.53 | ChannelZ | If you want a bunch of people in a conf, wouldn't you just transfer them into one, and then place another call, transfer them in, etc ? |
| 09:23.40 | *** join/#asterisk toppe (n=root@cs181001225.pp.htv.fi) |
| 09:23.47 | *** join/#asterisk sergey (n=sergey@ua0zeh.iks.ru) |
| 09:27.10 | aceking5 | is vtwhite a good/reliable DID provider? |
| 09:27.28 | aceking5 | i don't wanna lose DIDs if the company goes bankrupt |
| 09:27.52 | drmessano | Who? |
| 09:28.57 | aceking5 | vtwhite.com |
| 09:29.00 | aceking5 | viatalk |
| 09:39.27 | seanjohn | is there anyway to identify channels using variables asterisk automatically sets on the DIal command? |
| 09:46.42 | ChannelZ | like ${CHANNEL} ? |
| 09:52.20 | seanjohn | is this legal in asterisk? ExecIf($["${SPOOF}" = "1"],Sub,spoof^1) |
| 09:55.04 | ChannelZ | that's not the syntax, no |
| 09:57.44 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 10:02.37 | *** join/#asterisk dennis00 (i=dennis00@unaffiliated/dennis00) |
| 10:02.53 | dennis00 | How can I set up that multiple devices are logged in on the SIP account at a time? |
| 10:03.05 | dennis00 | <PROTECTED> |
| 10:04.29 | ChannelZ | you can't |
| 10:05.01 | dennis00 | I want to be available at a phone number on multiple devices. |
| 10:05.14 | ChannelZ | Each device must be unique |
| 10:05.19 | ChannelZ | but you can Dial multiple devices |
| 10:05.30 | ChannelZ | Dial(SIP/WorkPhone&SIP/HomePhone) |
| 10:07.29 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 10:08.31 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:19.45 | dennis00 | Great! :) |
| 10:28.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 10:59.34 | aceking5 | how do i disable asterisk from requiring registration/authentication from a certain IP? |
| 10:59.47 | aceking5 | insecure= doesnt seem to be working anymore |
| 11:04.57 | *** join/#asterisk dennis00 (i=dennis00@unaffiliated/dennis00) |
| 11:05.14 | dennis00 | Can extensions be both numbers and characters? Or is this not a FreePBX limit only? |
| 11:19.05 | *** join/#asterisk RypPn (n=TuMbL@rosscom.co.uk) |
| 11:22.03 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 11:24.25 | *** join/#asterisk fofware (n=chatzill@190.139.79.206) |
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| 11:45.16 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.172.179.dsl.dyn.forthnet.gr) |
| 11:54.40 | dennis00 | lol, my asterisk asked my name and I had not configured that. It happened after I changed my extern to exten => 3110number,n,Dial(SIP/1001,25&SIP/x-lite,25&SIP/iphone3gs,25&SIP/profoon,25). |
| 11:54.48 | dennis00 | Also, it only calls the 1001 number... |
| 11:58.25 | dennis00 | ÃŒs there a default timeout for dial? |
| 12:18.49 | *** join/#asterisk Akiraaa (n=Akiraaaa@79.112.36.132) |
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| 12:31.31 | Gido-E | dennis00 in the cli, core show applicaion dial |
| 12:31.37 | Gido-E | dennis00 in the cli, core show application dial |
| 12:50.56 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 12:53.39 | leifmadsen | dennis00: ya, your formatting in Dial() is wrong |
| 12:54.48 | dennis00 | @leifmadsen: Can you please be more clear? |
| 12:55.10 | leifmadsen | dennis00: what does the syntax output of 'core show application dial' tell you? |
| 12:55.22 | dennis00 | @leifmadsen: It gives a list of instructions. |
| 12:55.36 | leifmadsen | dennis00: at the top -- it shows you how to format the Dial() line |
| 12:55.44 | leifmadsen | above all the flags |
| 12:56.06 | *** join/#asterisk aceking5 (n=aceking5@71-94-132-102.static.mtpk.ca.charter.com) |
| 12:56.14 | dennis00 | It uses brackets... |
| 12:56.32 | dennis00 | the line you are refering to: Dial(Technology/resource[&Tech2/resource2...][,timeout][,options][,URL]) |
| 12:56.33 | aceking5 | how come DTMF is all broken up and jiterred in a SIPtoPSTN call? |
| 12:56.59 | leifmadsen | dennis00: right -- now look at your line, and look at how the first option in Dial() is structured |
| 12:57.56 | dennis00 | leifmadsen: So, do I need to use brackets in my dial()? |
| 12:58.16 | leifmadsen | dennis00: no.... those brackets just mean "optional" |
| 12:58.31 | leifmadsen | dennis00: Technology/resource[&Tech2/resource2...] <-- do you see any commas in there? |
| 12:58.32 | dennis00 | Dial(SIP/1001&1002&1003) |
| 12:58.43 | dennis00 | <PROTECTED> |
| 12:58.44 | leifmadsen | dennis00: no, look again |
| 12:58.53 | leifmadsen | dennis00: Technology/resource[&Tech2/resource2...] |
| 12:59.00 | leifmadsen | Tech2 |
| 12:59.12 | dennis00 | Dial(SIP/1001&SIP/1002&SIP/1003) |
| 12:59.28 | leifmadsen | dennis00: yes, now you can add your second argument, which in this case is a timeout |
| 12:59.44 | dennis00 | Dial(SIP/1001&SIP/1002&SIP/1003,25) XD |
| 12:59.55 | leifmadsen | correct |
| 13:00.40 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 13:00.40 | leifmadsen | aceking5: how do you have your dtmfmode setup in sip.conf ? |
| 13:01.03 | aceking5 | all rfc2833 |
| 13:01.16 | aceking5 | including itsp's and internal sip clients |
| 13:02.30 | aceking5 | and im pretty sure the call is ulaw from asterisk to the itsp |
| 13:02.31 | leifmadsen | how are you calling the pstn? |
| 13:02.44 | leifmadsen | sounds like a problem at the itsp |
| 13:02.46 | aceking5 | actually, it happens on an inbound call |
| 13:03.06 | aceking5 | RTP between asterisk and itsp is ulaw |
| 13:03.10 | leifmadsen | because rfc2833 is out of band, so the audio for the dtmf gets created after the fact |
| 13:03.13 | leifmadsen | ok... |
| 13:03.14 | aceking5 | idk what it is between asterisk and sip client |
| 13:03.34 | leifmadsen | either do I :) |
| 13:03.48 | aceking5 | how do i find out what codecs are being used? |
| 13:04.05 | leifmadsen | look at the SDP |
| 13:04.15 | aceking5 | right |
| 13:04.16 | leifmadsen | runs off to do some more phoneprov testing with older versions to see if it's just trunk |
| 13:04.16 | aceking5 | hold on |
| 13:04.24 | *** join/#asterisk aliverius (n=quassel@athedsl-387963.home.otenet.gr) |
| 13:04.53 | aliverius | is anyone here experienced with mISDN and LCR ? |
| 13:06.20 | aceking5 | a=rtpmap:0 PCMU/8000 |
| 13:06.25 | aceking5 | is between asterisk and ITSP |
| 13:07.41 | aceking5 | im pretty sure its all ulaw |
| 13:07.52 | aceking5 | is rfc2833 supposed to work? |
| 13:07.56 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 13:08.12 | *** join/#asterisk mrchrisadams_ (n=mrchrisa@82.45.160.216) |
| 13:08.12 | dennis00 | I have ulaw problems too. |
| 13:08.29 | aceking5 | hmm should i try inband |
| 13:08.40 | aceking5 | since it's ulaw |
| 13:09.00 | dennis00 | [Jan 23 14:07:16] NOTICE[3465] channel.c: Dropping incompatible voice frame on SIP/x-lite-00000001 of format alaw since our native format has changed to 0x4 (ulaw) |
| 13:09.00 | dennis00 | [Jan 23 14:07:18] WARNING[3465] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 13:09.00 | dennis00 | I have set allow ulaw, alaw and gsm in sip.conf, but the call still does not get through my PAP2T adapter. What could be the issue? |
| 13:09.18 | aceking5 | what did you do to see that |
| 13:09.20 | aceking5 | just verbosity? |
| 13:15.43 | dennis00 | No, I did not change to verbosity. |
| 13:17.09 | dennis00 | with verbosity: http://pastebin.ca/1762806. |
| 13:19.04 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 13:20.02 | aceking5 | is the call being established? |
| 13:26.50 | *** join/#asterisk kisom (n=kisom@c-9fdce155.648-1-64736c11.cust.bredbandsbolaget.se) |
| 13:32.12 | *** join/#asterisk kinkyboy (n=T3T4R@rue92-8-88-177-204-146.fbx.proxad.net) |
| 13:34.42 | aceking5 | are there 800 numbrs i can test dtmf on |
| 13:36.44 | dennis00 | Is there any reason I should not update to 1.6.2? |
| 13:41.50 | *** join/#asterisk retentiveboy (n=pdugas@adsl-221-170-128.pns.bellsouth.net) |
| 13:47.03 | *** join/#asterisk Cain` (n=Geek@unaffiliated/cain) |
| 13:50.26 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 13:53.59 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-18-213.acanac.net) |
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| 14:02.09 | *** join/#asterisk droemel (n=droemel@p548EFA05.dip.t-dialin.net) |
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| 14:09.36 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 14:15.28 | dlynes_laptop | dennis00, it was released recently, and so there's probably still security bugs in it |
| 14:16.47 | dlynes_laptop | aliverius, what's the problem with mISDN and LCR? |
| 14:19.15 | aliverius | i am still at the learning stage... that is my main problem it seems! atm i am trying to see if lcr receives an external ring, to verify i have done everything ok so far |
| 14:20.25 | aliverius | "23.01.10 16:06:00.683 CH: PH_ACTIVATE INDICATION U<-N port 0" tells me my isdn card is connected to my tel co's NT which is a relief |
| 14:21.45 | *** join/#asterisk yoshx (n=yoshx@38.115.119-80.rev.gaoland.net) |
| 14:36.37 | *** join/#asterisk Heretic (n=fallen@dsl-246-66-63.telkomadsl.co.za) |
| 14:36.49 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:12.26 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 15:15.13 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:15.56 | benngard | fun moment 22, i f i keep my settings "allow=alaw:40" i cam make calls but got fax problems "Only generating 240 samples, where 320 requested", when i change to "allow=alaw:20" (on both sides ofc) i get rid of the fax problems but cant place calls :( |
| 15:17.35 | *** join/#asterisk jmcdowell (n=airmadne@174-154-12-142.pools.spcsdns.net) |
| 15:17.38 | jmcdowell | hola all |
| 15:18.02 | jmcdowell | is looking for suggested hardware for a 3 line asterisk setup |
| 15:18.02 | *** join/#asterisk NicoleMun (n=ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
| 15:18.18 | voipmonk | explain 3 lines |
| 15:18.20 | voipmonk | 3 pstn lines? |
| 15:18.24 | voipmonk | or 3 sip lines ( voip ) ? |
| 15:18.32 | jmcdowell | 3 SIP lines |
| 15:18.41 | voipmonk | ok - and what do you want to do with them? |
| 15:18.48 | jmcdowell | Make phone calls |
| 15:18.49 | voipmonk | how do you want to use them? |
| 15:18.51 | jmcdowell | back and forth |
| 15:19.03 | jmcdowell | Sip lines. |
| 15:19.13 | jmcdowell | Tie each of them to a did |
| 15:19.26 | jmcdowell | and use them pretty regularly |
| 15:19.35 | voipmonk | excellent |
| 15:19.36 | jmcdowell | in one box with voice mail |
| 15:19.40 | voipmonk | so 1 number 3 lines |
| 15:19.44 | jmcdowell | IVR etc |
| 15:19.52 | voipmonk | do you have a preference for the location of the didi? |
| 15:19.54 | voipmonk | did... |
| 15:20.05 | jmcdowell | No, they have 3 existing #'s that they are porting over |
| 15:20.24 | jmcdowell | they bought a dual machine that turned out to be a POS and crashed last night during integration |
| 15:20.40 | jmcdowell | I told them to get a dell optiplex and call it done |
| 15:20.49 | voipmonk | for 3 lines? |
| 15:20.50 | voipmonk | wow |
| 15:20.50 | jmcdowell | but wanted to look for suggestions |
| 15:20.53 | voipmonk | :) |
| 15:20.58 | jmcdowell | Yes for 3 lines. |
| 15:21.19 | voipmonk | you could sit a watch on solar panels running linux and run 3 lines |
| 15:21.24 | jmcdowell | Wow as in dell not good enough or wow the dual 3.2 ghz with 4 gig ramn was an over kill kind of wow? |
| 15:21.32 | voipmonk | yes overkill |
| 15:21.40 | voipmonk | but you use what you got |
| 15:21.45 | jmcdowell | That's what I thought.. NEVER use shuttle based products |
| 15:21.47 | jmcdowell | they SUCK |
| 15:22.06 | voipmonk | if you want to squash a fly with the empire state building , go ahead :) |
| 15:22.17 | voipmonk | well back up now |
| 15:22.18 | jmcdowell | So I am thinking about getting a small dual p4 or dual core from CL and using that. |
| 15:22.26 | jmcdowell | What ever make more $ sense |
| 15:22.35 | voipmonk | shuttle based products dont suck - the systems integrator does |
| 15:22.42 | jmcdowell | voipmonk |
| 15:22.44 | jmcdowell | I don't suck |
| 15:22.53 | jmcdowell | the system CRASHED as in hardware failure. |
| 15:23.08 | jmcdowell | Locked up then lost CMOS crc wouldn't post, lost CMOS again etc. |
| 15:23.18 | jmcdowell | The hardware FAILED and I don't want to mess with it anymore. |
| 15:23.23 | voipmonk | it happens |
| 15:23.39 | jmcdowell | Oh and it won't boot now as the hard drive also has failed in some way. |
| 15:23.45 | jmcdowell | I hear ya |
| 15:24.02 | voipmonk | we could go along the very long list of ...... ambient temperatures, power, what did you do in the dev lab to test your system before deploying, how long didi you run the tests, blah blah |
| 15:24.17 | voipmonk | but right now the system is done - so moving on :) |
| 15:24.32 | voipmonk | so ur getting a dell, eh? |
| 15:24.36 | jmcdowell | Voipmonk is the dood you were talking to the other night that is dropping a donation in your box soon. |
| 15:24.52 | jmcdowell | This is the same box that you were on, it just failed. |
| 15:25.01 | jmcdowell | I am doing more experiementing right now than anything else. |
| 15:25.06 | voipmonk | well that sux |
| 15:25.09 | voipmonk | :) |
| 15:25.23 | jmcdowell | But that POS failed out of no where, in a perfectly suitable environement. |
| 15:25.38 | jmcdowell | Your right it happens, but I am testing, I have no time for failures. |
| 15:27.06 | jmcdowell | I am also exploring forking asterisknow |
| 15:27.07 | voipmonk | so when do u get the new box? |
| 15:27.19 | dmast | g'morning all |
| 15:27.22 | jmcdowell | I have no idea, I am hoping today. |
| 15:27.34 | jmcdowell | But if we just RMA the existing box, there is no telling when. |
| 15:27.42 | voipmonk | good lord |
| 15:28.10 | jmcdowell | good lord? |
| 15:32.12 | jmcdowell | Ok.. |
| 15:33.09 | jmcdowell | You know, it woul be nice to fork asterisknow into a more viable project focused around something like Ubuntu |
| 15:33.30 | *** join/#asterisk af_ (n=getsmart@88-149-240-203.dynamic.ngi.it) |
| 15:33.46 | jmcdowell | Bring it up to Asterisk 2.6 and not let it fall behind the curve, create some really intuitive interace enhancements while leaving the option for the old. |
| 15:33.52 | jmcdowell | One could almost not go wrong. |
| 15:36.57 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 15:37.46 | *** join/#asterisk timholum_ (n=chatzill@88.sub-97-143-84.myvzw.com) |
| 15:38.12 | dlynes_laptop | timholum_, still cruising around the mountain in virginia, while irc'ing and ssh'ing on your iphone? |
| 15:38.41 | timholum | nope, now im in a hotel in cary north carolina |
| 15:39.15 | timholum | still using my phone internet thought, due to the hotels internet sucking :) |
| 15:40.35 | jmcdowell | nice |
| 15:41.17 | timholum | but i have verizon so my speed's and coverage is good |
| 15:41.52 | *** join/#asterisk Alagar (n=Administ@122.164.32.148) |
| 15:42.53 | jmcdowell | But your bill probably sucksk, and that cap really sucks. |
| 15:42.57 | dlynes_laptop | timholum, so did you ever get your problem fixed? |
| 15:43.25 | timholum | no I didnt work on it after I disconnected yesterday |
| 15:43.33 | dlynes_laptop | ah |
| 15:45.34 | timholum | im googleing for the same issue it looks like a few people have had the issue but i have not seen a solution yet. I am sure it is just that i am missing something simple |
| 15:49.43 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
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| 15:52.46 | dlynes_laptop | timholum, i can't even remember what the issue was anymore |
| 15:53.36 | timholum | I am trying to configure voicemail to use mysql to store the messages. I keep getting a res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified |
| 15:53.57 | timholum | I have my config's here http://pastebin.com/m2320e3b4 |
| 15:54.04 | timholum | if you wanted to take a look |
| 15:54.58 | dlynes_laptop | timholum, i've got a few questions for you |
| 15:55.02 | *** join/#asterisk dennis00 (n=dennis@unaffiliated/dennis00) |
| 15:55.12 | timholum | ok |
| 15:55.29 | dennis00 | "[Jan 23 16:51:54] WARNING[25579] file.c: File please-enter-the does not exist in any format" < could it be i am missing sounds? |
| 15:55.30 | dlynes_laptop | timholum, Have you checked to make sure both shared objects actually exist where your odbcinst.ini says they live? |
| 15:55.44 | dlynes_laptop | dennis00, that's exactly the problem |
| 15:55.54 | dennis00 | dlynes_laptop: great :) |
| 15:56.12 | dlynes_laptop | timholum, also have you checked to make sure the mysql.sock file exists where your odbc.ini file says it does? |
| 15:56.27 | dennis00 | I am very tired and I want the wakeup call to be a 2nd alarm in case my phone fails. |
| 15:57.02 | dlynes_laptop | dennis00, what do you want us to do about it? |
| 15:57.24 | dennis00 | dlynes_laptop: nothing, I just wanted people to know what I am doing :) |
| 15:57.24 | dlynes_laptop | dennis00, put your alarm clock further away, so you have to drag your ass out of bed to shut it off |
| 15:57.28 | timholum | well isql astrealtime works. and I can use the database and table that I have configured |
| 15:57.42 | dennis00 | dlynes_laptop: I will do that too :) i guess that' s what went wrong lsat time. |
| 15:58.52 | dlynes_laptop | timholum, one other issue I see immediately, too |
| 15:59.02 | timholum | ok |
| 15:59.10 | dlynes_laptop | timholum, you created these files when you were running on little or no sleep |
| 15:59.23 | dlynes_laptop | timholum, so you might want to take the same suggestion I gave dennis00 |
| 16:00.20 | dlynes_laptop | timholum, i bet you're wondering why I'm saying that, right? |
| 16:00.38 | timholum | how is that :) |
| 16:01.00 | dlynes_laptop | timholum, hehe...your 'dsn' in res_odbc.conf is pointing to a non-existent dsn |
| 16:02.07 | dlynes_laptop | timholum, I'm guessing it works now? |
| 16:02.34 | timholum | its pointing to astrealtime which is how i have it configured in /etc/odbc.ini |
| 16:05.53 | dlynes_laptop | timholum, no...look again |
| 16:05.59 | timholum | wow :) |
| 16:06.38 | timholum | I think it was that i looked at it so many times that my mind made it look correct to me :) thanks :) |
| 16:06.52 | dlynes_laptop | yeah...that's happened to me before, too |
| 16:06.56 | dlynes_laptop | several times |
| 16:07.02 | dlynes_laptop | usually with C code, though :) |
| 16:07.07 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:07.34 | dlynes_laptop | timholum, so it's working now? |
| 16:08.37 | timholum | I am no longer getting the errors in my console, I am just about to give my vm a call |
| 16:09.45 | timholum | it works :) |
| 16:10.00 | timholum | thanks i have been working on that off and on for a week |
| 16:10.59 | dennis00 | dlynes_laptop: thanks, got it working! bye |
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| 16:18.46 | dlynes_laptop | timholum, wow....a week? |
| 16:19.50 | timholum | ya, :( not the same exact error the whole time but trying to get voicemail in a db |
| 16:20.18 | dlynes_laptop | timholum, well, i think your biggest problem was trying to use the mysql method instead odbc |
| 16:21.06 | timholum | ya :) and spelling :) |
| 16:23.07 | timholum | but now I can work on the webapp that I am integrating my phone system into :) |
| 16:23.32 | carrar | a web page for Asterisk, Thats crazy talk |
| 16:24.30 | dlynes_laptop | timholum, you do know about vmail.cgi, right? |
| 16:26.21 | timholum | dlyness_laptop: nope never heard of it. but the webapp is written in php and mysql. so my programing team will have an easyer time integrating with mysql. |
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| 16:30.33 | *** join/#asterisk kam187 (n=kam187@81.179.8.102) |
| 16:30.38 | kam187 | hey guys |
| 16:32.15 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 16:37.52 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:37.52 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:38.12 | dlynes_laptop | timholum, ah...yeah...the vmail.cgi app is written in perl |
| 16:43.10 | timholum | that is one of the bennefits of asterisk thought. if you want a different way to do anything. more often then not you can :) |
| 16:44.09 | dlynes_laptop | timholum, not even so much asterisk, than linux in general and in the greater microcosm, unix |
| 16:44.14 | aliverius | dlynes_laptop: what does ' l1 link = unknown' mean? before it was 'up' |
| 16:44.28 | dlynes_laptop | aliverius, huh? |
| 16:44.43 | dlynes_laptop | aliverius, you grab three words out of a log file and expect people to interpret it? |
| 16:44.51 | aliverius | lcradmin portinfo says that |
| 16:45.02 | aliverius | sorry i thought you remember i was asking about lcr before |
| 16:45.09 | dlynes_laptop | aliverius, oh...you're talking about the mISDN issue |
| 16:45.11 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 16:45.26 | aliverius | yeah, sorry i didnt mention that |
| 16:45.26 | timholum | true :) my office is run almost entirely off open source stuff, samba for my domain controler zimbra for mail and asterisk for the phone system |
| 16:45.30 | dlynes_laptop | aliverius, no idea...I just thought you might be running across the same issue as another guy the other day |
| 16:45.59 | dlynes_laptop | aliverius, someone else was working with the same stuff you're working with, but linux was completely locking up on them |
| 16:46.05 | aliverius | dlynes_laptop: ok. do you remember his nick? |
| 16:46.12 | dlynes_laptop | aliverius, not offhand, no |
| 16:46.23 | dlynes_laptop | aliverius, but he was european, fwiw |
| 16:46.29 | dlynes_laptop | aliverius, i think he was from norway |
| 16:46.30 | *** part/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 16:46.52 | aliverius | i guess misdn is useful only for europe :> |
| 16:47.12 | dlynes_laptop | aliverius, not necessarily so |
| 16:47.23 | dlynes_laptop | aliverius, it's just that in north america, it's almost non-existent |
| 16:47.35 | aliverius | in case you meet him again tell him to lower the priority of lcr |
| 16:47.35 | dlynes_laptop | aliverius, the telcos here don't even want to admit they have the service |
| 16:47.50 | dlynes_laptop | aliverius, ah, ok...so you had the same problem, then? |
| 16:48.12 | aliverius | no bad i have been reading the guides over and over hehe |
| 16:48.17 | aliverius | but* |
| 16:48.18 | dlynes_laptop | ah |
| 16:49.33 | *** join/#asterisk droemel (n=droemel@p548ED033.dip.t-dialin.net) |
| 16:50.14 | aliverius | in theory isdn should me more comfortable to work with than pots but the implementation is somewhat obscure |
| 16:50.30 | aliverius | i dont even know where to start from :p |
| 16:51.11 | *** join/#asterisk markwaters (n=mark@weloveit.info) |
| 16:51.17 | *** part/#asterisk droemel (n=droemel@p548ED033.dip.t-dialin.net) |
| 16:51.28 | aliverius | brb |
| 16:51.46 | kam187 | hmm asterisk keeps seg faulting after a few hours |
| 16:51.56 | kam187 | but when it restarts it wont accept any sip calls |
| 16:52.05 | timholum | well I am going to head out, thanks again dlynes_laptop for your help. |
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| 17:51.07 | elitemx | Hi everyone. I have two basic questions. |
| 17:51.59 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 17:52.46 | elitemx | If I get t1 pri would it be possible for me to call |
| 17:53.18 | elitemx | one of the external numbers (and instruct it to make a four-way five-way call) |
| 17:54.04 | p3nguin | You should be able to have 23 calls on a PRI, if I remember right. |
| 17:58.02 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 17:58.06 | elitemx | what really confused me is looking up some products on digium (specifically the TE122B which cost about $700) ... because when I took a look at their "switchbox soho" prodcut it says "up to 10 concurrent calls" (and that costs over $2000) |
| 17:59.27 | elitemx | also the overview for the TE122 card says "the TE122 can be used to deliver ... conferencing, three-way calling, ..." |
| 17:59.51 | elitemx | does that mean four-way, five-way, six-way etc is not supported |
| 18:01.02 | p3nguin | That's the way I would interpret it, but I am not familiar with those hardwares. |
| 18:05.02 | elitemx | thanks |
| 18:05.20 | elitemx | I will try a little more research |
| 18:12.19 | carrar | elitemx, with a PRI you could have 23 people on one call |
| 18:12.23 | carrar | oh he left |
| 18:12.42 | carrar | more if they had sip |
| 18:12.44 | carrar | heh |
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| 19:00.05 | aliverius | ok i need a softphone for linux to help me learn asterix |
| 19:00.33 | aliverius | recomendations please? i dont care about the look and stuff, i just wanna connect it easily to an asterix |
| 19:01.48 | ChannelZ | Zoiper.. Twinkle.. Ekiga.. X-lite.. KPhone? |
| 19:02.03 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 19:02.21 | aliverius | okkkk i will pick the k thing |
| 19:02.21 | aliverius | ty |
| 19:03.00 | ChannelZ | good luck |
| 19:03.07 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 19:03.27 | chuckf | aliverius: what distro are you using? |
| 19:03.47 | aliverius | archlinux and apparently kphone is not available :( |
| 19:04.35 | chuckf | I don't run arch but I like twinkle or ekiga |
| 19:04.50 | chuckf | in that ordedr |
| 19:04.53 | zamba | i dislike ekiga |
| 19:04.58 | zamba | it doesn't handle roaming very well |
| 19:05.05 | zamba | that is, at all |
| 19:05.25 | chuckf | no, but for basic testing its not bad as many distros include it |
| 19:06.12 | aliverius | ok twinkle is my first bet |
| 19:06.13 | *** join/#asterisk aceking5 (n=aceking5@71-94-132-102.static.mtpk.ca.charter.com) |
| 19:06.18 | aliverius | ekiga may look ugly |
| 19:08.13 | aliverius | twinkle is qt3.. it wont be great either |
| 19:11.33 | *** join/#asterisk Buklov (n=buklov@mail.abcmebel.ru) |
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| 19:24.32 | p3nguin | I use twinkle on Arch. |
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| 19:27.32 | aliverius | i put twinkle here, ekiga on the other pc |
| 19:28.24 | aliverius | so basically i should now connect the two to asterisk and let them talk through it? |
| 19:32.27 | aliverius | asterisk-gui is cycling this message: Creating a config file to store GUI Preferences |
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| 19:35.21 | p3nguin | We don't do GUIs. |
| 19:37.04 | kam187 | how do i specfify the ring cadence in asterisk? |
| 19:37.28 | p3nguin | That might be in indications.conf |
| 19:38.56 | aliverius | ok p3nguin |
| 19:39.33 | p3nguin | I don't know if anyone is in #Asterisk-GUI or not. |
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| 19:42.49 | aliverius | i could edit plain files or use the cli but asterisk is so vast |
| 19:43.28 | aliverius | could you please at list indicate a very introductory tutorial? those on asteriskguru dont follow a straight line |
| 19:43.39 | p3nguin | ~book |
| 19:43.40 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:46.16 | aliverius | ty |
| 19:53.26 | ChannelZ | did linuxdoc.org die some time ago? |
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| 21:21.52 | circut | afternoon all |
| 21:22.35 | circut | is there a reputable service provider that replaced nufone |
| 21:22.36 | circut | ? |
| 21:22.42 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:23.16 | drmessano | There's hundreds |
| 21:25.25 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
| 21:25.34 | ChannelZ | ~itsplist-us |
| 21:25.35 | infobot | i heard itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 21:25.54 | ChannelZ | there's a few less than hundreds, but GIYF |
| 21:26.19 | circut | thanks friend |
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| 21:28.37 | drmessano | ChannelZ: Just how many ITSPs do you think exist? |
| 21:28.58 | seanbright | 42 |
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| 21:30.17 | ChannelZ | I have no idea |
| 21:30.40 | ChannelZ | I meant "there, above what infobot said, are a few less than hundreds.." |
| 21:31.04 | *** part/#asterisk kam187 (n=kam187@81.179.8.102) |
| 21:31.44 | drmessano | Hundreds would be more more than 200, and you can find at least 120 listed on the voip wiki |
| 21:32.05 | drmessano | I'd say that number is small compared to the actual count |
| 21:33.26 | ChannelZ | you're missing what I meant. I didn't mean to say there are less than hundreds in existance |
| 21:33.26 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.250.dyn.user.ono.com) |
| 21:33.41 | Godfather_ | o/ |
| 21:33.43 | ChannelZ | Only that "there, the ones infobot just listed, are less than hundreds of choices" |
| 21:47.08 | ruben23 | hi |
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| 22:32.16 | youngproguru | So.. What do you all think... Is it safe to move to 1.5 for full production use? |
| 22:32.22 | youngproguru | 1.6 |
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| 22:33.17 | bmoraca_work | yes |
| 22:34.45 | youngproguru | neat |
| 22:41.30 | p3nguin | I've always ran straight asterisk (no FreePBX add-on, no PiaF, no asterisk-gui, no Trixbox, etc.). Would it be foolish for me to go with AsteriskNOW for my next deployment? |
| 22:44.36 | ManxPower-work | p3nguin: you'll have to learn everything over again. |
| 22:45.06 | bmoraca_work | p3nguin, stick with straight asterisk. there are some minor benefits to running something like freepbx...but in general, it's not worth it |
| 22:45.50 | bmoraca_work | i've gone through a lot of trouble to provide customers with individual GUIs where they can modify find-me-follow-me and forwarding and read voicemails...and they never use them after the first week or so...it's just not worth the headache |
| 22:47.45 | *** join/#asterisk loceur (i=37e5687e@204.15.179.182) |
| 22:48.02 | loceur | anyone work with a hosted PBX provider that'd be willing to provide a DID, auto attendant, and call forwarding for a month or two for a haiti relief program? |
| 22:48.06 | loceur | http://jetsupport.com/home/2010/01/20/jssi-makes-strong-commitment-to-haiti-relief-efforts-2/ |
| 22:50.03 | bmoraca_work | i'm skeptical of all these haiti releif organizations |
| 22:50.32 | loceur | well, this isn't an org. It's just a business company doing the work. |
| 22:50.41 | loceur | but there should be skepicism |
| 22:50.49 | loceur | there's plenty of room for fraud |
| 22:51.49 | loceur | I'd also be more than willing to go through any verification, as that would be understandable |
| 22:53.19 | bmoraca_work | well, unfortunately, i don't do international termination or origination, and i couldn't really authorize it anyway (well, i probably could, but it'd be hard to justify it to the highers-up) |
| 22:53.28 | loceur | ohh it's all local |
| 22:53.41 | loceur | no international calls, I don't think |
| 22:53.53 | loceur | though it's like 16h/day of calls |
| 22:54.38 | loceur | they're in Florida and it'd be call forwarding to cell phones there |
| 22:54.51 | loceur | no outbound, just inbound. |
| 22:55.35 | bmoraca_work | call forwards are outbound :) |
| 22:55.46 | loceur | good point :/ |
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| 23:01.58 | loceur | if we setup sip phones there, would that change anything? |
| 23:03.30 | bmoraca_work | like i said, it's ultimately not my decision...i was more curious than anything else |
| 23:03.59 | loceur | of course. glad you asked, maybe someone else will meander in and be interested |
| 23:04.45 | loceur | also, asking on a Saturday is a seemingly large barrier. Decision makers don't make decisions till Monday :) |
| 23:05.09 | bmoraca_work | if it were up to me and i had DIDs where you needed them and the cellphone termination rates were't terrible, i would probably be willing to spare a few channels |
| 23:05.56 | p3nguin | You want to originate calls in FL and terminate them to cell phones in Haiti? |
| 23:06.22 | loceur | no no, originate around US, terminate in FL. This is mostly logistic work |
| 23:06.50 | loceur | doesn't have to be toll free, though it'd be nice |
| 23:06.55 | p3nguin | What is in FL that needs all these calls to be routed through a PBX? |
| 23:07.04 | loceur | http://jetsupport.com/home/2010/01/20/jssi-makes-strong-commitment-to-haiti-relief-efforts-2/ |
| 23:07.09 | loceur | airport |
| 23:07.24 | loceur | and about 15 volunteers on cell phones (3-4 onsite at any given time) |
| 23:07.38 | loceur | shipping everything from gause to orphans |
| 23:09.34 | loceur | bmoraca_work; should I have an npa-nxx of where they're at for info? What else should I have to help get a grasp on estimated costs? |
| 23:10.26 | bmoraca_work | that would be it...that'll determine the direct costs of terminating to those cell phones. i looked up the number you PMed me in Texas and my rate was ~$0.018/min |
| 23:10.29 | p3nguin | If you had a good internet connection for data and IP phones on location, that would simplify things and reduce costs. |
| 23:10.55 | loceur | cell phone numbers are different area codes, but they'll all be in florida. would their location or their cell's npa-nxx be the deciding factor? |
| 23:11.39 | p3nguin | Flowroute offers $0.0098/minute to all US numbers. |
| 23:12.02 | p3nguin | But if you are using IP phones on-site, the termination rate is 0. |
| 23:12.13 | bmoraca_work | flowroute also charges inbound, don't they? |
| 23:12.30 | p3nguin | They have various "plans" for inbound. |
| 23:12.37 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 23:12.55 | p3nguin | How many calls do you expect to receive? |
| 23:13.30 | loceur | they're averaging 2-3 minute calls every 10 minutes times 3-4 phones for 16 hours a day |
| 23:13.38 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 23:13.40 | loceur | +/- |
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| 23:14.23 | p3nguin | Your numbers have caused confusion. |
| 23:14.29 | bmoraca_work | you're looking at roughly 2000 minutes per day |
| 23:15.15 | loceur | bmoraca_work; I calculated 1000 minutes, but I guess call forwarding is 2x? |
| 23:15.20 | bmoraca_work | no |
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| 23:15.47 | loceur | but yeah, 1000-2000 minutes a day is a good estimate |
| 23:16.07 | bmoraca_work | well, i rounded up to 5 minutes per 10 minutes (30 minutes per hour) * 4 phones = 120 minutes /hr * 16 hours /day = 2000 minutes per day |
| 23:19.06 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 23:19.47 | p3nguin | Who is going to be paying for the costs on VoIP channels? |
| 23:20.07 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 23:20.08 | Katty | hi |
| 23:20.15 | bmoraca_work | p3nguin, he's looking for them to be donated |
| 23:20.26 | p3nguin | I see. |
| 23:20.36 | Katty | i put up 'Fruit Baskets' for the birds. |
| 23:21.11 | Katty | nailed 1 inch of a 6 inch nail into each tree...and then hung a small wire basket on the nail. |
| 23:21.14 | p3nguin | I would be more apt to donate my internet and clock cycles than my money to pay for channels. |
| 23:21.37 | loceur | p3nguin; actually, may have just found $200 to do this. |
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| 23:22.22 | bmoraca_work | loceur, what's the NPA-NXX for this? |
| 23:22.45 | loceur | bmoraca_work; their cell phone's npa-nxx or the physical location's npa-nxx? |
| 23:22.52 | bmoraca_work | cell phones |
| 23:23.04 | loceur | crap, they're prolly all over |
| 23:23.09 | loceur | I could get them |
| 23:23.22 | loceur | might take an hour or two, but most of these guys aren't local |
| 23:23.24 | bmoraca_work | just an example...they're likely the same provider, right? |
| 23:23.29 | Katty | http://ecx.images-amazon.com/images/I/41eianzQSWL._AA260_.jpg |
| 23:23.35 | Katty | ^- that's the basket i hung on the nail. |
| 23:23.43 | loceur | bmoraca_work; no, probably all over |
| 23:23.46 | Katty | and then i put slices of peach in the basket. |
| 23:23.47 | bmoraca_work | ahh |
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| 23:24.39 | p3nguin | Flowroute provides a "virtual PRI" for the incoming calls -- $17.95/month/channel, unlimited minutes. You'll probably want at least 1.5x as many channels as you have people to receive calls. |
| 23:24.49 | loceur | the goal is to frequently rotate numbers around, as well p3nguin, cell phone 1 answers while he's onsite, then to cell 2 when he leaves, so it'd take frequent administration |
| 23:25.02 | loceur | p3nguin; great price |
| 23:25.07 | loceur | reliable enough? |
| 23:25.13 | loceur | get what you pay for of course |
| 23:25.26 | p3nguin | It is only as reliable as the internet connection where the PBX is connected. |
| 23:25.33 | loceur | ohh right |
| 23:25.35 | p3nguin | I would trust Flowroute 99%. |
| 23:26.34 | p3nguin | As far as users going on-site and needing calls to come in, then no more calls when they go off site... that is simple with Queues and Agent logins. |
| 23:26.44 | bmoraca_work | i generally charge $40/mo/channel (includes local and longdistance and inbound) for hosted pbx origination/termination. for non-hosted PBX, I charge $30/mo for same. the reason i charge more for hosted pbx is because i don't actually charge for the hosting of the pbx |
| 23:26.52 | p3nguin | You go on-site, you login to your queue. You logout before you leave. |
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| 23:31.30 | loceur | bmoraca_work; have a management tool to add/remove numbers? |
| 23:32.46 | loceur | or a website I can go look at? Seems I have funding now |
| 23:33.33 | loceur | p3nguin; I'll check out flowroute as well, looks like it may be a good choice as well |
| 23:33.54 | p3nguin | DIDs - 20-number blocks start at $14.95/month with a $19.95 setup fee. |
| 23:34.04 | p3nguin | Do you need 20 numbers? |
| 23:34.16 | p3nguin | single DIDs at bargain price of $1.39/month with a $1.00 setup fee! |
| 23:34.39 | loceur | single did. 20 users/call forwarded numbers |
| 23:34.51 | p3nguin | 1 DID? |
| 23:35.03 | p3nguin | interesting |
| 23:35.04 | dlynes_laptop | p3nguin, at the price flowroute charges, you may as well just get a pri |
| 23:35.19 | loceur | yeah. 1-800 or general call-in number, then an auto-attendant for call routing |
| 23:35.37 | loceur | dlynes_laptop; lol, where to then? |
| 23:35.37 | p3nguin | Yeah, for $375/month, I'll go ahead and run a T1. *sigh* |
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| 23:39.36 | dlynes_laptop | loceur, just talk to your telco, and steal a pri off of them for $300-450/mo |
| 23:40.25 | loceur | dlynes_laptop; this is a 1 -2 month deal and I need 3-4 lines (6 would be perfect) and only call forwarding... |
| 23:40.42 | loceur | getting mine own pri and setting up my own * box would be a bit overkill |
| 23:41.34 | dlynes_laptop | loceur, yeah...that'd be overkill for 3-4 lines |
| 23:41.40 | dlynes_laptop | or even 6 for that matter |
| 23:41.54 | dlynes_laptop | loceur, and you only get the good rates on pri's, if you sign a 5 yr commit |
| 23:42.02 | loceur | hehe |
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| 23:47.50 | loceur | bmoraca_work; I'd be interested in your hosted pbx. have any details you can pass my way? |
| 23:57.42 | jhirley | quick question, what the name of the sip provider starts with a V ? |
