| 00:00.03 | footnote | And the benefits here... |
| 00:00.03 | Apple | you're fired! |
| 00:00.21 | footnote | crap |
| 00:00.21 | footnote | fired again |
| 00:00.21 | Apple | every time I tried to grep the logs from my cell phone it froze :( |
| 00:00.22 | footnote | haha |
| 00:00.49 | Apple | not to mention I still can't make calls |
| 00:01.13 | hardwire | can anybody else here not reach the wiki? |
| 00:01.28 | Apple | hardwire: I guess you missed the news... |
| 00:01.32 | Apple | it's been down for several hours |
| 00:01.37 | hardwire | I probably wasn't paying attention |
| 00:01.39 | hardwire | ok |
| 00:01.48 | p1tst0p | hmm, my friend is trying to dial me, ver my asterix at home, using SIP ( X-Lite ) he is getting Call not approved, do i have to allow him or something ? |
| 00:02.01 | footnote | X-Lite setup |
| 00:02.04 | zyke | anyone using asterisk realtime? |
| 00:02.13 | PTG123 | anyone here know polycom, my phone isn't trying to register and not sure why |
| 00:02.29 | Chuji | PTG123 : what model? |
| 00:02.32 | PTG123 | p1st: you have a guest context? |
| 00:02.36 | PTG123 | chuji: ip500 |
| 00:02.46 | Chuji | PTG123 I have one on my desk yeah |
| 00:02.55 | PTG123 | chuji: ok any idea why its not trying to register? |
| 00:03.09 | PTG123 | can you look at your webpage |
| 00:03.14 | PTG123 | and tell me what your settings are |
| 00:03.15 | Chuji | Doing it from the webpage |
| 00:03.16 | PTG123 | actually |
| 00:03.20 | p1tst0p | PTG123, what do you mean, im new to this mate, im using the Asterisk@Home |
| 00:03.30 | Chuji | Yeah, I'll look at it |
| 00:03.34 | threat|BX | modulus_ hmm 3 on one side, 1 on the other, I am guessing I times everything by 3 on the right hand side / :) |
| 00:03.41 | p1tst0p | At least im seeing this --> Registered SIP '667' |
| 00:03.46 | KalD|Work | can anyone tell me why Asterisk resets SPANS? |
| 00:03.54 | PTG123 | well i see register packets |
| 00:04.08 | Chuji | PTG123 : What's the admin password? |
| 00:04.12 | PTG123 | but yet it shows as not registered |
| 00:04.13 | Chuji | I always forget and have to look it up |
| 00:04.17 | PTG123 | chuji: 1234 or 123 |
| 00:04.22 | p1tst0p | ah PTG123, under his extention in context it says from-internal |
| 00:04.22 | PTG123 | Polycom for username |
| 00:04.24 | zyke | anyone using asterisk realtime here? |
| 00:04.34 | PTG123 | zyke: i use my own realtime |
| 00:04.42 | MikeJ[Jayden] | no.. no one uses realtime... |
| 00:04.44 | PTG123 | p1tst0p: you need to add a guest account with no password |
| 00:04.57 | zyke | PTG123: you are not using the one from the CVS head? |
| 00:05.19 | p1tst0p | PTG123, as in a unix guest account, i dont get you sorry. |
| 00:05.29 | Chuji | hmm, it's 456 on mine |
| 00:05.58 | roamer323 | is the wiki gonzo or just busy? |
| 00:06.04 | MikeJ[Jayden] | busted... |
| 00:06.56 | threat|BX | PTG123 are u crazy? |
| 00:07.07 | Chuji | PTG123 : All I have is the IP address in Address under server 1 |
| 00:07.20 | Chuji | PTG123 : I have to run or I would give you more info |
| 00:07.25 | Chuji | should be easy |
| 00:07.34 | *** join/#asterisk Inv_arp (junya@adsl-3-245-189.mia.bellsouth.net) |
| 00:07.37 | PTG123 | zyke: no my own |
| 00:07.44 | PTG123 | p1tst0p: to sip.conf |
| 00:07.51 | PTG123 | Chuji: ah yah it could be that too :) |
| 00:07.54 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 00:07.58 | Apple | argg, I swear, there's no help for me :( |
| 00:08.00 | PTG123 | threat|BX: crazyy like a fox |
| 00:08.05 | roamer323 | anyone heard of or know the admin password for vonage linksys pap2s? |
| 00:08.10 | PTG123 | Chuji: ok |
| 00:08.17 | threat|BX | roamer323 admin / admin ? |
| 00:08.18 | PTG123 | roamer323: they aren't usable |
| 00:08.22 | threat|BX | admin / password ? |
| 00:08.34 | zyke | PTG123: is it good? any reason why u didn't want to use the main one? |
| 00:08.40 | roamer323 | threat|BX - I wish :-) |
| 00:08.45 | threat|BX | roamer323 :P |
| 00:08.58 | zyke | i'm looking to use the realtime stuff but don't know where to get started |
| 00:08.58 | PTG123 | zyke: yah i do lots of fancy things, mine is alot better.. but doesn't use mysql, it uses my own db |
| 00:08.59 | *** join/#asterisk skeezer (~crozierm@64.65.134.42) |
| 00:09.26 | threat|BX | PTG123 DB4 ? :) |
| 00:09.31 | roamer323 | PTG123 - they will be once the 'admin' passwd is released, the 'user' passwd is already out, and looks like they're moving wholesale to Motorola gear |
| 00:09.31 | zyke | i have all my stuff on mysql :) |
| 00:09.35 | p1tst0p | PTG123, in what syntax, is there a doc, i dont wanna hassle u |
| 00:10.05 | threat|BX | SELECT * FROM zyke; oh yeah |
| 00:10.05 | PTG123 | roamer323: if that ever happens let me know, i gto 4 useless ones laying around |
| 00:10.11 | PTG123 | not pap2s though |
| 00:10.33 | PTG123 | threat|BX: its like berkely db.. its a local hashed based db, for extreme performance |
| 00:10.50 | PTG123 | i am actually posting on #asterisk-dev right now about how no one wants to implement my patches to make it possible :) |
| 00:10.53 | threat|BX | PTG123 uber! |
| 00:10.59 | roamer323 | PTG123 - I'm sure you'll hear through the grapevine when that happens :-) hope the wiki gets healed soon. |
| 00:11.10 | threat|BX | PTG123 how does it compare to other databases? |
| 00:11.12 | PTG123 | who hosts the wiki anyhow? |
| 00:11.21 | PTG123 | threat|BX: its created just for asterisk, so it can't get any faster |
| 00:11.29 | threat|BX | heh |
| 00:11.31 | threat|BX | nice |
| 00:11.40 | zyke | PTG123: can your stuff be configered to use mysql? |
| 00:11.45 | threat|BX | PTG123 you believe in opensource dont you? :P |
| 00:11.59 | roamer323 | PTG123 - not pap2s? you mean you have the Cisco black boxes? |
| 00:12.11 | PTG123 | zyke: no you don't udnerstand going to an external db has problems |
| 00:12.28 | PTG123 | zyke: think aboutg it, all the communication that has to be done with every call, everytime someone registeres, etc |
| 00:12.41 | PTG123 | threat|BX: yah but if it won't get implemented whats the point in releasing it |
| 00:12.47 | PTG123 | roamer: i have some ciscos and some motorolas |
| 00:12.54 | PTG123 | roamer323: managed to unlock one of my ciscos, thats it |
| 00:13.04 | threat|BX | so whats the worst that can happen in Asterisks? endless menu option loops? drop outs midway? |
| 00:13.49 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.184) |
| 00:13.50 | zyke | PTG123: i'm looking for a solution to support large number users |
| 00:13.56 | darius_ | PTG123: I'm ok with the load .. that's what replication nodes are for :) |
| 00:13.58 | PTG123 | zyke: thats why this was created |
| 00:14.02 | Apple | argg |
| 00:14.16 | Apple | so, I can still receive calls from asterisk to kphone (via fwd) |
| 00:14.17 | PTG123 | darius_: just the communcation between realtime and asterisk itself is a major problem |
| 00:14.22 | Apple | but can't make them! |
| 00:14.29 | darius_ | probably BerkelyDB would be a good fit |
| 00:14.30 | PTG123 | darius_: all link list based, very slow, lots of string compares, i mean milliosn per minute |
| 00:14.31 | *** join/#asterisk rious (~rious@adsl-69-208-72-102.dsl.klmzmi.ameritech.net) |
| 00:14.35 | PTG123 | darius_: on a load server |
| 00:14.43 | PTG123 | darius_: well this basically does what berkely does, except better |
| 00:14.57 | zyke | PTG123: is the realtime part of the main cvs head? does it get compiled with asterisk? |
| 00:14.59 | p1tst0p | PTG123, is there a document anywher that talks about externel sip users, and adding this guest account ? |
| 00:14.59 | darius_ | people want something they can hook into |
| 00:15.00 | PTG123 | darius_: if you understyand how a db stores its info, i do it firect myself |
| 00:15.09 | zyke | or u need to compile it separelty? |
| 00:15.15 | PTG123 | p1st0p: just add an account, as guest, with no password |
| 00:15.15 | Apple | anybody else have an idea about why I can send but not receive with kphone and asterisk? |
| 00:15.19 | PTG123 | zyke: seperate |
| 00:15.25 | zyke | oh i see |
| 00:15.31 | PTG123 | darius_: and this makes a direct hook my mod :) |
| 00:15.43 | PTG123 | darius_: so someone could implemet berkely db, their own sql, whatever |
| 00:15.49 | darius_ | if an asterisk optimized datastore could be accessed through odbc, it would be nice |
| 00:15.51 | p1tst0p | PTG123, i dont know what to add to sip.conf, this is my first install of Asterisk. |
| 00:16.02 | PTG123 | p1st0p: ok you owe me one second |
| 00:16.08 | zyke | PTG123: is it locate in the addons section? |
| 00:16.15 | p1tst0p | yeh man, |
| 00:16.31 | PTG123 | <PROTECTED> |
| 00:16.31 | PTG123 | [guest] |
| 00:16.31 | PTG123 | type=user |
| 00:16.31 | PTG123 | context=incoming |
| 00:16.31 | PTG123 | auth=none |
| 00:16.47 | threat|BX | isn't ODBC evil? |
| 00:16.49 | p1tst0p | then change his context to incoming in the user panel |
| 00:16.58 | darius_ | why would it be evil? |
| 00:17.12 | darius_ | it's not a windows only thing |
| 00:17.15 | PTG123 | p1s:no idea know nothing about @home asterisk |
| 00:17.21 | threat|BX | darius_ ok :) |
| 00:17.49 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 00:18.02 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
| 00:18.50 | zyke | PTG123: do you know where i can download the realtime stuff? |
| 00:19.05 | PTG123 | zyke: it comes with it |
| 00:19.10 | PTG123 | look for realtime whent he wiki comes up |
| 00:19.17 | PTG123 | its a bitch to get installed |
| 00:19.19 | PTG123 | so good luck |
| 00:19.22 | PTG123 | it took me 2 days :) |
| 00:19.25 | zyke | really |
| 00:19.27 | PTG123 | which is another reason i wrote my own |
| 00:19.36 | zyke | will come back for help :) |
| 00:19.39 | Sedorox | Does Anyone use Link2voip (aka freeworldtel)? |
| 00:19.54 | modulus_ | jbot fwd? |
| 00:19.55 | jbot | i guess fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 00:20.07 | darius_ | I haven't used asterisk for quite a while .. but I assume that there's still not 100% database config support? |
| 00:20.10 | Beirdo | I'm trying to, Sedorox |
| 00:20.15 | Sedorox | Beirdo: ok |
| 00:20.30 | Sedorox | they finally gave us our toll free number.. but I still can't dial it... |
| 00:20.41 | MikeJ[Jayden] | ~rtfw |
| 00:20.42 | jbot | i heard rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 00:20.43 | Sedorox | "beep.. we're sorry.. the number cannot be completed as dialed... " |
| 00:20.46 | MikeJ[Jayden] | hehe |
| 00:20.55 | Sedorox | or "beep.... we're sorry.. that number is no longer in service" |
| 00:21.18 | Beirdo | Sedorox: mine's been like that for almost a week now, it takes the telco 4-6 days apparently after you get the number |
| 00:21.25 | Sedorox | hmmmmm |
| 00:21.26 | Beirdo | and I'm bitchy at them |
| 00:21.38 | Sedorox | that'll be well over a week |
| 00:21.45 | Sedorox | is it that way for all providers? |
| 00:22.12 | MikeJ[Jayden] | no, some providers actually service their customers.. but very few |
| 00:22.31 | Sedorox | hmm |
| 00:22.52 | MikeJ[Jayden] | that is true of voip and traditional telco both |
| 00:22.56 | MikeJ[Jayden] | and clecs |
| 00:23.20 | MikeJ[Jayden] | pure and simple, assume they are crooks, unless they do a lot to prove otherwise |
| 00:23.31 | MikeJ[Jayden] | big and small |
| 00:24.22 | jehunter | how do you tell asterisk to handle a fax in the extensions.conf file? |
| 00:24.23 | MikeJ[Jayden] | wow, that shut everyone up :D |
| 00:24.31 | Sedorox | lol |
| 00:24.34 | MikeJ[Jayden] | handle a fax? |
| 00:24.41 | MikeJ[Jayden] | send? receive? what? |
| 00:24.59 | jehunter | detect a fax and re-direct to external fax machine |
| 00:25.01 | threat|BX | modulus_ G'Day, quick question, [(k³ + 5k² + k² + 5k)/3] + [(k + 1)]/1 :P |
| 00:25.29 | MikeJ[Jayden] | there is fax detect in zap, and I think somone did it in iax as well |
| 00:25.40 | Sedorox | threat|BX: the answer you seek.... 36 |
| 00:25.41 | Sedorox | :-p |
| 00:25.52 | MikeJ[Jayden] | there is an app to check |
| 00:25.59 | MikeJ[Jayden] | it's on the wiki :) |
| 00:26.07 | Sedorox | which is down :-p |
| 00:26.11 | MikeJ[Jayden] | I know |
| 00:26.12 | threat|BX | Sedorox :P |
| 00:26.31 | Sedorox | is there any licensing issues which will prevent me from using stuff from *@Home for my own use? |
| 00:26.32 | MikeJ[Jayden] | show apps, it is there |
| 00:26.34 | criptos | wikki.................. |
| 00:26.40 | jehunter | I have autodetect set in the zap file, now i need to know what the syntax is to handle it in the extensions.conf file |
| 00:27.09 | MikeJ[Jayden] | jehunter, look at how *@home handles it in their sample configs |
| 00:27.19 | jehunter | will do, thanks |
| 00:28.23 | Sedorox | jehunter: I pulled it.. mounted the Iso on loopback... and pulled the config files out.. so no need to burn it or anything |
| 00:28.30 | *** part/#asterisk CoderCR (~creyna@adsl-67-112-135-29.dsl.sndg02.pacbell.net) |
| 00:28.32 | *** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 00:29.02 | MikeJ[Jayden] | jehunter.. you don't even need to dl the cd, you can dl the tar file too if all you wanna do is peek at configs... |
| 00:29.13 | Sedorox | well yea.... |
| 00:30.17 | MikeJ[Jayden] | if you are new.. you will think @home is great for about 3 days, learn a ton, then rebuild your box from scratch the following weekend |
| 00:30.30 | Silik0n | heh |
| 00:31.13 | MikeJ[Jayden] | but I still suggest it for every newcomer through the door... it answers sooooo many of the first 50 quesitons you have automatically, and you can dig in to somthing working from the start, and figure out how latter... |
| 00:33.13 | *** join/#asterisk cmslaght (~chris-jun@cblmdm69-45-209-31.buckeye-express.com) |
| 00:33.29 | Silik0n | the answer to all noob questions is unlink /dev/zero |
| 00:33.37 | cmslaght | ok.. whats up with the wiki? |
| 00:33.46 | Silik0n | its sick today |
| 00:33.52 | modulus_ | [root@asterisk asterisk]# perl -lne 'print if /\d{10}/' extensions.conf |sed 's/\,/\ /g'|awk '{print $3}'|uniq|wc -l |
| 00:33.52 | modulus_ | <PROTECTED> |
| 00:33.52 | modulus_ | [root@asterisk asterisk]# |
| 00:33.59 | modulus_ | 39 DIDs |
| 00:34.02 | modulus_ | not bad |
| 00:34.43 | threat|BX | DIDs? |
| 00:34.52 | modulus_ | jbot did? |
| 00:34.53 | jbot | it has been said that did is Direct Inward Dialing |
| 00:35.06 | MikeJ[Jayden] | it's just a number :) |
| 00:35.28 | modulus_ | age? |
| 00:35.35 | cmslaght | why is 39 not bad? |
| 00:35.35 | Sedorox | oh? |
| 00:35.36 | Sedorox | of? |
| 00:35.57 | modulus_ | cmslaght, went with voip in begining of jan. |
| 00:37.20 | Sedorox | 10 more days and it'll be a month since I ordered it |
| 00:38.04 | Sedorox | jbot: wiki-status is Current Down |
| 00:38.05 | jbot | Sedorox: okay |
| 00:38.11 | Sedorox | wiki-status? |
| 00:38.48 | threat|BX | oh nice |
| 00:38.50 | Silik0n | anyone doneany real load testing on app_meetme? |
| 00:39.02 | modulus_ | i once had 3 whole people on meetme |
| 00:39.03 | Silik0n | ~wiki-status |
| 00:39.04 | jbot | somebody said wiki-status was Current Down |
| 00:39.05 | modulus_ | lol |
| 00:39.11 | Silik0n | hah |
| 00:39.21 | Silik0n | yeah well I had more then that in there today |
| 00:39.50 | Silik0n | i hope the box can hand with 15 or 20 people... i mean hell its only a hyperthreaded dual 3ghz xeon |
| 00:39.51 | threat|BX | jbot G'Day |
| 00:39.53 | jbot | G'day mate, hazzagarn? |
| 00:39.57 | Silik0n | w/ 4G ram |
| 00:40.05 | threat|BX | hehe |
| 00:40.28 | threat|BX | Silik0n is that all? |
| 00:40.30 | Sedorox | Silik0n: I'll be surprized if it'll support one person |
| 00:40.31 | Sedorox | :-p |
| 00:40.36 | Silik0n | hah |
| 00:40.38 | threat|BX | Silik0n my calculator is faster then that |
| 00:40.44 | Silik0n | yeah then it falls over dead |
| 00:40.50 | Silik0n | that swapping is a bitch |
| 00:40.55 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 00:40.55 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:41.49 | *** join/#asterisk seanegan (~sean@cpe-66-24-230-237.stny.res.rr.com) |
| 00:42.28 | *** part/#asterisk seanegan (~sean@cpe-66-24-230-237.stny.res.rr.com) |
| 00:42.34 | *** join/#asterisk kernel3 (kernel@netblock-66-245-242-106.dslextreme.com) |
| 00:42.54 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 00:44.18 | Sedorox | anyone ever notice the girl in the "your call cannot be completed as dialed" sounds like the computer from Star Trek TNG? |
| 00:46.17 | *** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net) |
| 00:46.34 | KalD|Work | Sedorox, nope - two completely different people. |
| 00:47.30 | Sedorox | hmmm |
| 00:48.04 | KalD|Work | Sedorox, I think the voice of the computer was Roddenberry's wife. |
| 00:48.21 | Sedorox | ahhh ok |
| 00:48.35 | KalD|Work | Sedorox, or the lady that played diana's mother... (or maybe same person!?) |
| 00:49.26 | Sedorox | thats true too... |
| 00:49.26 | Sedorox | dunno |
| 00:50.02 | Silik0n | KalD|Work: the computers voice and the woman that played diana troy's mother were both 1 in the same and she also played nurse chappel on the original series |
| 00:50.02 | Eight | It's the same woman. |
| 00:50.16 | Sedorox | hmmmm |
| 00:50.19 | Sedorox | interesting |
| 00:50.20 | Silik0n | and yes it was roddenbury's wife |
| 00:50.23 | Eight | *and* she's-- |
| 00:50.24 | Eight | ya. |
| 00:50.25 | KalD|Work | =) Ok sorry I haven't watched TNG for years =) |
| 00:50.28 | Eight | But it's Troi =p |
| 00:50.29 | Sedorox | hehe |
| 00:50.35 | Shido6 | troi used to behto |
| 00:50.36 | Sedorox | I love TNG |
| 00:50.38 | Shido6 | err hot |
| 00:50.39 | KalD|Work | wohoo... where was that question on jerpordy? |
| 00:50.42 | Sedorox | ahah Shido6 |
| 00:50.46 | Silik0n | i had a room mate that was a super trek geek |
| 00:50.53 | Shido6 | had a crusher on her in high school |
| 00:50.58 | KalD|Work | lol |
| 00:51.08 | Sedorox | she's a real ditz tho.. both on and off set.. |
| 00:51.10 | KalD|Work | we dont wanna know about you and wesley |
| 00:51.12 | Eight | Shido6: nice freudian slip =p |
| 00:51.23 | Eight | KalD|Work: hey, there was another Crusher, give him SOME credit. |
| 00:51.30 | Sedorox | ahah |
| 00:51.34 | KalD|Work | =) true but I forgot about her |
| 00:51.39 | Sedorox | 0_o |
| 00:51.52 | KalD|Work | ... nm that sounds worse |
| 00:52.00 | Sedorox | yes |
| 00:52.01 | Sedorox | very |
| 00:52.51 | Sedorox | ahahahha |
| 00:53.24 | Bentley | hi all, I'm try to configure zaptel.conf for an E1 in argentina. Anyone here in argentina using E1? |
| 00:55.13 | *** join/#asterisk easydone (~notdone@eksel.demon.nl) |
| 01:00.03 | KalD|Work | ok so this is starting to feel like galaticomm teleconference... |
| 01:03.06 | Druken | ? |
| 01:03.09 | KalD|Work | So... why does asterisk reset the spans on the digium PRI cards? - and is there a way to disable this? |
| 01:03.30 | KalD|Work | Druken, old school BBS =) |
| 01:03.46 | Druken | ahh, ok... |
| 01:04.02 | Druken | been a long time since i've heard anyone mention bbs's :( |
| 01:04.03 | terrapen | galacticom hhahaha |
| 01:04.18 | KalD|Work | Druken, MajorBBS =) |
| 01:04.18 | terrapen | i'll keep my Maximus BBS, thank you |
| 01:04.21 | terrapen | and my BinkleyTerm |
| 01:04.28 | KalD|Work | terrapen, Spitfire man! |
| 01:04.31 | terrapen | That's BinkleyTerm/2 |
| 01:04.39 | terrapen | but I started with FrontDoor |
| 01:04.45 | terrapen | and RemoteAccess |
| 01:04.46 | KalD|Work | WWIV <-- WTF? |
| 01:04.49 | terrapen | heh |
| 01:04.51 | terrapen | Waffle |
| 01:04.53 | KalD|Work | lol |
| 01:04.55 | KalD|Work | GAP |
| 01:04.57 | KalD|Work | Wildcat |
| 01:04.59 | Silik0n | <-- 50line G'Comm |
| 01:05.00 | Druken | hehe renegade, frontdoor and gecho |
| 01:05.01 | terrapen | AdeptXBBS |
| 01:05.09 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 01:05.13 | KalD|Work | Silik0n, FOOL =) You paid way too much... did you get the galatibox too? |
| 01:05.17 | Silik0n | complete with IP access |
| 01:05.31 | Silik0n | KalD|Work hell no and I made a living off that thing for years |
| 01:05.37 | KalD|Work | lol |
| 01:05.46 | Silik0n | bocaboards and Vircom's DMA server split the board into multiple boxes |
| 01:06.20 | terrapen | desqview haha |
| 01:06.24 | Druken | i like schronet for that... it spanned it'sself across multipul boxen |
| 01:06.27 | terrapen | Desqview/X |
| 01:06.31 | terrapen | how obscure am i now |
| 01:06.32 | KalD|Work | it was all fine until one power spike took out all 4 of my USR 14.4s =( |
| 01:06.35 | Silik0n | i ran a 4node RemoteAccess w/Frontdoor under desqview on a 386 hah |
| 01:07.08 | Druken | 2 node... front door, win95 on a p75 hehe |
| 01:07.15 | KalD|Work | I miss it... like was simple - you could kick a node off to play tradewars... esp after you see where he/she had the home planet |
| 01:07.17 | terrapen | i mostly ran Maximus/BinkleyTerm/GoldED |
| 01:07.25 | terrapen | anyone remember GoldED? |
| 01:07.28 | KalD|Work | s/like/life/g |
| 01:07.34 | terrapen | I ported it to run under Mac OS X one time for the hell of it |
| 01:07.44 | Mavvie | terrapen: golded sucked. |
| 01:07.47 | KalD|Work | Q-Modem was the best! |
| 01:07.50 | terrapen | whatever :P |
| 01:07.54 | Silik0n | KalD|Work tradewars 2002 made me tons of cash hah |
| 01:08.07 | Silik0n | people would pay 25cents per hour to play that bitch heh |
| 01:08.18 | Mavvie | terrapen: actually, every mail reader sucked until they invented the YAM messagebase. |
| 01:08.20 | Druken | hehehe |
| 01:08.28 | terrapen | i used Squish |
| 01:08.31 | terrapen | which was ok |
| 01:08.36 | Silik0n | damn lines were packed at nite and ran 60% capacity during max low peek |
| 01:08.39 | KalD|Work | fidonet =) |
| 01:08.41 | Mavvie | I used FNNR. |
| 01:08.51 | terrapen | fidonet was so fun...yet so lame |
| 01:08.57 | Silik0n | fidonethaha... i was 1:123/49 |
| 01:08.59 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:09.02 | *** join/#asterisk xfr (~xfrosch@69-2-83-42.wan.networktel.net) |
| 01:09.15 | KalD|Work | I paid about 300 bucks a month in LD to seattle from AK cuz everyone decided to use my board for fido traffic instead of my friends =( |
| 01:09.16 | Druken | my god... hehehe |
| 01:09.37 | Druken | ya know we ar a bunch of geeks... hehe when we all brag about our bbses.. ehehe |
| 01:09.38 | *** join/#asterisk ioerror (~ioerror@mail.lostinthenoise.net) |
| 01:09.44 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-39-201.no.no.cox.net) |
| 01:09.56 | Silik0n | hah |
| 01:10.03 | Silik0n | oh like we have something better to do |
| 01:10.23 | Druken | very true |
| 01:10.24 | KalD|Work | Druken, well... the next thing we're gonna do is make a tw2002.agi script to play tw2002 over dtmf w/ festival |
| 01:10.34 | Druken | hahahaha |
| 01:10.51 | KalD|Work | lol |
| 01:10.51 | Mavvie | only a couple of old simtel and garbo archives have it, for the rest it's gone from the internet. |
| 01:10.51 | Silik0n | KalD|Work: gotta make it read the old DB files too |
| 01:11.01 | KalD|Work | hey--- anyone got any NIGHTOWL cds around??? |
| 01:11.03 | Druken | i'm supprised the boards didn't play more of a role in telnet |
| 01:11.20 | KalD|Work | Silik0n, that is easy =) it is all pascal until v3 |
| 01:11.22 | ioerror | If I want to make a call plan in extensions.conf to call europe, what's a good way to make sure the number is semi-valid like I can with _1NXXNXXXXXX,2,Dial() |
| 01:11.34 | ioerror | Or really for any international call that doesn't fit that format? |
| 01:11.55 | KalD|Work | ioerror, from US? 0X.,1,Dial(${EXTEN},,r) should do it |
| 01:12.05 | KalD|Work | with a _ in front |
| 01:12.25 | Silik0n | actually from the US _011X. |
| 01:12.27 | ioerror | Assuming I am using voicepulse, would that change it? |
| 01:12.44 | Silik0n | askvoicepulse |
| 01:12.46 | *** join/#asterisk xfr (~xfrosch@69-2-83-42.wan.networktel.net) |
| 01:12.52 | ioerror | I basically just want to have the user type in any number and have it relay |
| 01:13.14 | ioerror | Well it's very specific to asterisk in this case, it's a normal us number, so i believe the 011 is correct |
| 01:13.16 | KalD|Work | Druken, the bbs era ended when local colleges start offering gopher and muds =( |
| 01:13.48 | Silik0n | KalD|Work not true |
| 01:13.58 | Druken | but muds suck... hehehe |
| 01:14.23 | Silik0n | KalD|Work we morphed from a bbs into a isp until someone put land.c into their cron and aimed it at out main AAA server |
| 01:14.33 | ioerror | :-( |
| 01:14.57 | KalD|Work | Silik0n, yeah but a true bbs was not an isp =) |
| 01:15.23 | KalD|Work | =) BBS (w/ doors etc) |
| 01:15.23 | Druken | i used to offer internet email on my board |
| 01:16.43 | KalD|Work | I have AOL v1 for dos =) |
| 01:16.51 | KalD|Work | wait... why would I smile about that? |
| 01:17.47 | Druken | pfft, AOL |
| 01:18.11 | KalD|Work | or would you prefer qlink? |
| 01:18.31 | Silik0n | <PROTECTED> |
| 01:20.50 | Druken | i'm not at that level... i never had a c64 |
| 01:20.57 | Druken | i started off with a 286 |
| 01:21.08 | KalD|Work | Compuserve then |
| 01:22.09 | KalD|Work | I just realized I have no way to power up my old 286... I have no ISA vga card or a cga monitor =( |
| 01:22.12 | modulus_ | damn |
| 01:22.15 | Silik0n | hah |
| 01:22.18 | modulus_ | who remembers compuserve? |
| 01:22.27 | modulus_ | jesus i remember when they were AOL's only competitor |
| 01:22.35 | Silik0n | 147435,1252@compuserve.com |
| 01:22.38 | KalD|Work | lol =) |
| 01:22.42 | KalD|Work | I KNEW SOMEONE WOULD BITE! |
| 01:23.01 | Druken | KalD|Work: i bet i have an ISA videocard and an EGA monitor around somewhere |
| 01:23.14 | Druken | you pay shipping, i'll give ya free :) |
| 01:23.22 | Silik0n | i never had a real compuserve account tho... i always ued the local university's VAX cluster dailups |
| 01:23.26 | KalD|Work | Druken, =) let me check the local value-village |
| 01:23.42 | modulus_ | yeah universities are gold mines for connectivity |
| 01:23.58 | Druken | KalD|Work: hehe probably a better idea :) |
| 01:24.02 | KalD|Work | I used to use work's to goto ibm to download os/2 drivers for my PAS16 and Aztec cdrom |
| 01:24.08 | Silik0n | you could dial in w/ no password and use their "phone book" server which showed last logins (or No Last Login) then pick accounts that have never been logged int and use them... (default passwd was the username) |
| 01:24.28 | *** join/#asterisk Kokey (~ubunturer@dsl-200-78-65-251.prod-infinitum.com.mx) |
| 01:24.56 | Druken | hahaha |
| 01:25.07 | Druken | now that's security |
| 01:25.19 | KalD|Work | ok was it just me or was TD3 just the best game of all time? |
| 01:25.19 | modulus_ | yeah and the email server used the same passwords using LDAP info |
| 01:25.36 | modulus_ | i used to crack email accounts @genericuniv.edu |
| 01:25.47 | modulus_ | then change their passwords for them |
| 01:25.56 | modulus_ | b/c they had brain-dead passwords |
| 01:25.58 | KalD|Work | I miss Novell 3.12 |
| 01:26.11 | KalD|Work | rock solid |
| 01:26.14 | mgth | I know people that still run it |
| 01:26.14 | modulus_ | yes |
| 01:26.27 | Sedorox | 6.5 |
| 01:26.45 | KalD|Work | i'm getting netware 6.5 atm from novell |
| 01:27.05 | KalD|Work | sconsole send all was my friend |
| 01:27.07 | Sedorox | hehe |
| 01:27.23 | Sedorox | you can always get the eval |
| 01:27.26 | modulus_ | i miss sunos too |
| 01:27.28 | Sedorox | unlimited licensing |
| 01:27.29 | KalD|Work | fire phaser 5 =) |
| 01:27.32 | Sedorox | lol |
| 01:27.41 | KalD|Work | login scripts rocked |
| 01:27.53 | KalD|Work | people would look over and be like - wow your admin |
| 01:27.53 | modulus_ | i switched to bsd after sunos move to sys V |
| 01:27.54 | KalD|Work | lol |
| 01:28.26 | KalD|Work | I burned my copy of xenix =) |
| 01:28.45 | KalD|Work | kinda on accident... but mostly on purpose |
| 01:29.05 | Sedorox | :-p |
| 01:29.16 | Sedorox | I need to redo my login scripts |
| 01:29.21 | Sedorox | when I moved from 5.1 to 6.5 |
| 01:29.23 | Sedorox | I didn't upgrade |
| 01:29.25 | Sedorox | :/ |
| 01:29.25 | KalD|Work | wow we should change the topic to "Memory Lane" |
| 01:29.28 | Sedorox | lost a lot... |
| 01:29.30 | Sedorox | lol |
| 01:29.41 | SexyKen | Is voip-info down |
| 01:29.48 | modulus_ | voip sucks |
| 01:29.49 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 01:29.58 | Sedorox | ~wiki-status |
| 01:29.59 | jbot | from memory, wiki-status is Current Down |
| 01:30.19 | Druken | god that wiki goes down alot |
| 01:30.27 | modulus_ | reminds me of voip |
| 01:30.33 | KalD|Work | it dont 'work on my box' |
| 01:30.38 | cpatry | voip sucks? stop crack smokin' :) |
| 01:30.44 | modulus_ | i hate voip |
| 01:30.47 | modulus_ | i hate asterisk |
| 01:30.57 | Druken | and we all hate modulus_ :) |
| 01:31.03 | cpatry | but ya like h323 and gatekeeper huh? |
| 01:31.13 | modulus_ | no those are more braindead |
| 01:31.37 | KalD|Work | Why wont asterisk dial pulse over my PRI? |
| 01:31.47 | modulus_ | b/c it sucks |
| 01:32.16 | KalD|Work | I hooked up my 1977 rotary dial phone to Iaxy and it dont work =( |
| 01:32.24 | KalD|Work | j/c - I dont have that anymore |
| 01:32.30 | KalD|Work | ok - well I dont use it anymore |
| 01:32.41 | KalD|Work | ok - well I dont have it in this room... |
| 01:32.48 | modulus_ | on that note, i'm gonna go take a dump |
| 01:32.51 | modulus_ | laters |
| 01:32.52 | KalD|Work | lol |
| 01:33.01 | mog_home | ytes |
| 01:33.05 | ioerror | How can I make a call plan in my extensions.conf file that decides if it's a call to the USA or to another country? |
| 01:33.24 | ioerror | Can I just have two lines in a stanza and it will match against one of them depending on what type of number it is? |
| 01:33.27 | modulus_ | exten => _011.,1,Dial() |
| 01:33.29 | ioerror | Ie: 011 or 1 |
| 01:33.34 | ioerror | Ah. |
| 01:33.40 | ioerror | that's quite simple |
| 01:34.29 | modulus_ | exten => _011.,1,Dial(IAX2/sukyVOIPgeneriCprovider.com/${EXTEN}) |
| 01:36.50 | *** join/#asterisk justinnnn (~dsf@solid.mpa.net.au) |
| 01:36.52 | justinnnn | hey |
| 01:36.57 | Druken | exten => _011.,1,Dial(let them fucken worry about it) :) |
| 01:37.01 | justinnnn | does anyone no howto setup the cisco 7914 expansion thingy |
| 01:37.04 | justinnnn | does it work with sip ??? |
| 01:37.08 | justinnnn | she doesnt look like she does.. ? |
| 01:37.16 | ioerror | thanks modulus_ |
| 01:38.22 | afe | Darn... I just learned the sad fact that neither SJPhone or X-pro works well with my PocketPC/smartphone, while the icky Skype does :/ |
| 01:38.43 | afe | (as well as Skype can work that is) |
| 01:38.56 | KalD|Work | odd.. I call this damn number w/ asterisk and it kills the call after 11 seconds everytime?! |
| 01:39.26 | afe | I was hoping to be able to use the phone with * |
| 01:40.49 | ioerror | modulus_, exten => _011.,2,Dial(IAX2/username:key@gwiaxt01.voicepulse.com/${EXTEN}) |
| 01:40.56 | justinnnn | anyone |
| 01:40.59 | justinnnn | expansion thingy ?? |
| 01:41.05 | ioerror | modulus_, does that seem correct? |
| 01:42.41 | Druken | looks about right |
| 01:43.54 | modulus_ | looks about right |
| 01:44.43 | Eight | I haven't used IAX yet, just SIP... but shouldn't the auth info go into IAX.conf, not the dial command? |
| 01:45.13 | Eight | Dial(IAX2/${EXTEN}@gwiiax01.voicepulse.com), for example. |
| 01:45.23 | justinnnn | anyone setup a cisco 7970 with asterisk ??? |
| 01:45.28 | Eight | That's how it is with SIP and Broadvoice. |
| 01:45.49 | *** join/#asterisk sergiovel (sergiovel@194-167-126-200.fibertel.com.ar) |
| 01:45.52 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 01:46.00 | sergiovel | hello everyone |
| 01:47.10 | r0d3nt|m | Hi. |
| 01:47.30 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 01:47.53 | r0d3nt|m | Why would asterisk stop taking calls, T1 just rings and rings and rings.. all channels.. system ran GREAT for 49 days.... But I had to reboot it to get it to start taking calls again....... |
| 01:48.07 | r0d3nt|m | Anyone care to speculate ?? |
| 01:48.27 | *** join/#asterisk letherglov (~robbie@128.54.170.88) |
| 01:48.30 | Eight | r0d3nt|m: did you login to the machine before you rebooted it? |
| 01:48.31 | Druken | welcome to the zaptel drivers |
| 01:48.35 | r0d3nt|m | Eight, yes... |
| 01:48.38 | Eight | or did it lock up entirely? |
| 01:48.41 | r0d3nt|m | Eight, nooo |
| 01:48.43 | r0d3nt|m | the box was AOK |
| 01:48.47 | r0d3nt|m | asterisk was OK |
| 01:48.51 | *** part/#asterisk letherglov (~robbie@128.54.170.88) |
| 01:48.53 | r0d3nt|m | you could call out No problem |
| 01:48.54 | Eight | did you restart ast'? |
| 01:48.56 | r0d3nt|m | yes |
| 01:48.58 | r0d3nt|m | i did |
| 01:49.00 | r0d3nt|m | same results |
| 01:49.05 | Eight | sounds like drivers =) |
| 01:49.05 | r0d3nt|m | i restarted everything |
| 01:49.35 | r0d3nt|m | then I had to reboot the whole box |
| 01:49.35 | Sedorox | even zaptel? |
| 01:49.35 | r0d3nt|m | yes |
| 01:49.35 | Druken | try reloading the drivers before rebooting? |
| 01:49.35 | r0d3nt|m | even zaptel |
| 01:49.35 | r0d3nt|m | hmm.... |
| 01:49.43 | Druken | rmmod zaptel; modprove zaptel |
| 01:49.52 | r0d3nt|m | ok.. i'll note that.... |
| 01:49.54 | r0d3nt|m | thank you sir... |
| 01:49.55 | Druken | modprobe... |
| 01:49.58 | *** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org) |
| 01:49.58 | r0d3nt|m | any other ideas ?? |
| 01:50.00 | r0d3nt|m | aye.. ya.. |
| 01:50.02 | r0d3nt|m | probe* |
| 01:50.26 | Mavvie | r0d3nt|m: 49 days is 4233600000 milliseconds. |
| 01:50.32 | Druken | no, that's it... the drivers i find die about every 22 days... i'm impressed you got 40+.. |
| 01:50.34 | r0d3nt|m | ya so ? |
| 01:50.42 | Mavvie | up to you to convert it to hex and wonder why it is so close to an variable overflow. |
| 01:50.42 | r0d3nt|m | Druken, really ? |
| 01:51.03 | r0d3nt|m | Mavvie, i care not to get into things that deep.. |
| 01:51.32 | Mavvie | r0d3nt|m: *shrug* that is where your problem lays anyway. |
| 01:51.41 | r0d3nt|m | If it's a problem with the software, I report it and wait for a fix.. if my vendor fucked up.. i go kill them. |
| 01:51.51 | r0d3nt|m | Mavvie, oh really... ? |
| 01:52.03 | Eight | just to be clear. |
| 01:52.08 | r0d3nt|m | Eight, wise. |
| 01:52.31 | r0d3nt|m | eheheh |
| 01:52.35 | r0d3nt|m | well i appreciate the speculation guys |
| 01:52.37 | r0d3nt|m | for real... |
| 01:52.48 | Eight | r0d3nt|m: thanks for asking, I learned some stuff myself =) |
| 01:53.05 | Druken | that the zaptel drivers are a pain in the ass? |
| 01:53.05 | r0d3nt|m | oh really / |
| 01:53.10 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
| 01:53.11 | r0d3nt|m | Eight, what did you learn ? |
| 01:53.26 | Eight | ya, rmmod, and 22 days is what Druken usually has his zap' drivers bail on him at =) |
| 01:53.34 | r0d3nt|m | aye |
| 01:53.43 | ioerror | modulus_, exten => _1NXXNXXXXXX,1,Playback(beep) |
| 01:53.43 | ioerror | exten => _011.,2,Dial(IAX2/username:key@gwiaxt01.voicepulse.com/${EXTEN}) |
| 01:53.45 | r0d3nt|m | i have another system that has been up much long then that.... |
| 01:53.49 | Eight | don't look at me like that... I'm not a unix newbie... just BSD =p (re: rmmod) |
| 01:54.15 | ioerror | modulus_, should i modify the first line to match the second one? |
| 01:54.22 | Mavvie | 22 days is 1900800000 ms, which is close to an signed variable overflow. |
| 01:54.31 | r0d3nt|m | which means ? |
| 01:54.38 | Brixius | hello |
| 01:54.39 | Eight | r0d3nt|m: something might be counting. |
| 01:54.42 | Qwell | Are you joking? heh |
| 01:54.54 | r0d3nt|m | <PROTECTED> |
| 01:54.55 | justinnnn | anyone use cisco 7970 or 7914 ???? |
| 01:54.58 | r0d3nt|m | that is another system |
| 01:55.01 | r0d3nt|m | never needed to restart it. |
| 01:55.06 | *** join/#asterisk JamesDotCom (~james@sweep.bur.st) |
| 01:55.12 | Mavvie | that's your OS, not your application. |
| 01:55.27 | r0d3nt|m | asterisk restarts every night @ 3am |
| 01:55.30 | r0d3nt|m | but not zaptel |
| 01:55.33 | ioerror | Druken, ? |
| 01:55.38 | Druken | yeah ? |
| 01:56.14 | r0d3nt|m | System uptime: 13 hours, 55 minutes, 56 seconds |
| 01:56.14 | r0d3nt|m | Last reload: 3 hours, 35 minutes, 11 seconds |
| 01:56.31 | JamesDotCom | does someone have any idea of why a recent cvs version of asterisk (yesterday) seems to kill the network including my current ssh sessions to the machine when it tries to place some iax calls |
| 01:56.33 | ioerror | Druken I am trying to get my extentions.conf to work for interntaional calls exten => _1NXXNXXXXXX,1,Playback(beep) |
| 01:56.33 | ioerror | exten => _011.,2,Dial(IAX2/username:key@gwiaxt01.voicepulse.com/${EXTEN}) |
| 01:56.39 | JamesDotCom | should i even be looking at asterisk to blame? |
| 01:56.43 | ioerror | Druken, The second line was more like the first to start |
| 01:56.56 | ioerror | Should both lines have the same _011. ? |
| 01:57.10 | Druken | well, those are two diffrent dial strings, they have to match |
| 01:57.22 | Druken | if you want international calls, the line 1 must match line 2 |
| 01:57.32 | Druken | for the dial string |
| 01:57.41 | ioerror | Alright, I need to fix that |
| 01:57.45 | ioerror | I thought so. |
| 01:57.53 | ioerror | What's the best place to read the documentation for this? |
| 01:58.02 | Eight | ioerror: voip-info |
| 01:58.05 | Druken | ~voip-info |
| 01:58.06 | jbot | i guess voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 01:58.07 | Eight | ~docs |
| 01:58.08 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:58.09 | ioerror | it's seemingly down for me |
| 01:58.21 | ioerror | wow, thanks! |
| 01:58.22 | Eight | ioerror: it's seemingly down, altogether. |
| 01:58.34 | Sedorox | ~wiki-status |
| 01:58.35 | jbot | hmm... wiki-status is Current Down |
| 01:58.35 | Sedorox | :-p |
| 01:58.41 | Eight | Sedorox: hah. |
| 01:59.02 | Brixius | That answers my question |
| 01:59.17 | Sedorox | someone needs to mirror it... |
| 01:59.21 | Eight | inded |
| 01:59.21 | Sedorox | who controls it? |
| 01:59.25 | Eight | erm, indeed, rather. |
| 01:59.32 | Eight | If it were lost entirely we'd be screwed =p |
| 01:59.41 | Sedorox | yea... |
| 01:59.42 | Brixius | so much for doing much on * tonight :( |
| 01:59.46 | Silik0n | who runs the wiki? |
| 01:59.47 | Druken | no.... YOU'D be screwed.... |
| 01:59.48 | shmaltz | anybody know what's wrong with the wiki? |
| 01:59.48 | Druken | hehehe |
| 01:59.51 | Sedorox | I should contact the owner and see if I could mirror it... |
| 01:59.52 | Eight | Druken: pfff. |
| 01:59.56 | Qwell | shmaltz: It's down. |
| 02:00.02 | Silik0n | shmaltz it fell over dead |
| 02:00.07 | Eight | Druken: this channel would go to heck in a coin purse =p |
| 02:00.15 | shmaltz | why? |
| 02:00.26 | Qwell | because it chose to |
| 02:00.34 | Druken | someone stuck their dick in the wiki's ear... |
| 02:00.38 | Apple | argg, if only the site would be fixed :( |
| 02:00.46 | Qwell | Druken: My apologies. |
| 02:01.00 | Druken | haha |
| 02:01.16 | Silik0n | mirror it and put that thing on roaud robbin |
| 02:01.37 | shmaltz | Qwell, how old are you? |
| 02:01.37 | Apple | Silik0n: give me the db |
| 02:01.39 | Mavvie | hope it's back tomorrow before I have to do the move. |
| 02:01.47 | Silik0n | if I had the DB i would have it online hah |
| 02:02.00 | shmaltz | Mavvie, you might have to push off the move |
| 02:02.23 | Silik0n | Mavvie: or you could pay me to do the configs for you hah |
| 02:02.47 | Silik0n | nd if you were really smart just google for what you are looking for... google does have voip-infocached |
| 02:03.13 | Eight | aye, good point. |
| 02:03.16 | KalD|Work | when it comes up someone wget it =) |
| 02:03.32 | Eight | I nominate KalD|Work |
| 02:03.53 | KalD|Work | Eight, ok let me know when it is back... I've got just the site for it =) |
| 02:04.24 | Silik0n | no wonderits down... his DNS changed... whois voip-info.org and you'll see NS2.MYDYNDNS.ORG for name servers hah |
| 02:04.39 | Eight | you mean it's on a dynamic IP?! |
| 02:04.44 | Eight | no wonder it flakes out. |
| 02:04.48 | Silik0n | whois it and you tell me |
| 02:05.17 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 02:05.48 | Apple | who runs the site? |
| 02:06.02 | Silik0n | whois it and email him to find out |
| 02:06.14 | Apple | :/ |
| 02:06.17 | Silik0n | geez do you people never do whois lookups on websites? |
| 02:07.01 | Mavvie | whois only gives the person who owns/runs the domain, it doesn't say anything about websites. |
| 02:07.21 | Eight | nor what their IRC nick on freenode is =p |
| 02:07.40 | Qwell | It should list all of that |
| 02:07.53 | Eight | Qwell: And I thank $diety it doesn't =p |
| 02:08.03 | Eight | oh wait, wrong syntax for this channel. |
| 02:08.10 | Eight | ${DIETY} |
| 02:08.27 | Qwell | ${DIETY:1} |
| 02:08.34 | Eight | od? |
| 02:08.35 | Eight | =p |
| 02:08.43 | Qwell | Well, what if they dialed a 9? |
| 02:08.55 | *** join/#asterisk xilch (~xilch@66.239.17.226.ptr.us.xo.net) |
| 02:09.40 | mog_home | use google cache |
| 02:10.59 | KalD|Work | Eight, when it is up let me know I'll put it on www.asterisk.kicks-ass.org |
| 02:11.42 | Eight | I'm probably not the most reliable source of information on that subject atm. |
| 02:12.35 | t3t | Commpartners runs the site... it doesn't appear to be dynamic. I'm sure that they are working hard to get it back up again |
| 02:13.03 | *** join/#asterisk Shido6 (~greg@d57-87-253.home.cgocable.net) |
| 02:18.09 | Shido6 | hrmm |
| 02:20.17 | Beirdo | yes? |
| 02:20.33 | Beirdo | grr. stupid autoaway script |
| 02:23.05 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 02:23.05 | shepherd | apple: why you ask? |
| 02:24.31 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 02:25.48 | *** join/#asterisk azzazz|n (~3d5f1499@ns1.webonce.com) |
| 02:27.00 | *** part/#asterisk azzazz|n (~3d5f1499@ns1.webonce.com) |
| 02:28.32 | spackle | . |
| 02:28.38 | AgiNamu | / |
| 02:28.38 | robtro | omfg |
| 02:28.38 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 02:30.11 | ioerror | How can I loop a part of my extensions.conf stanza until the user says that the number they have input is correct? |
| 02:30.19 | ioerror | In pseudo code here: |
| 02:31.04 | ioerror | while (1) do: enter $number, is $number correct? Press 1 for yes, 2 for no; |
| 02:31.14 | ioerror | So I don't have to keep hanging up and calling back basically. |
| 02:31.37 | Silik0n | with is gotoif |
| 02:31.49 | Silik0n | with a gotoif that is |
| 02:31.58 | shepherd | ioerror: see agi |
| 02:32.14 | Silik0n | gotoif would be less costly thena agi for that |
| 02:32.49 | shepherd | ph |
| 02:32.54 | shepherd | oh |
| 02:32.55 | shepherd | heh |
| 02:32.56 | ioerror | Silik0n, where can i see the documentation on that? |
| 02:33.25 | Qwell | ioerror: show application gotoif |
| 02:33.57 | ioerror | Qwell, I mean, how can i see an example of how to use it in my extensions.conf |
| 02:34.34 | *** join/#asterisk stevecayona (~neondog@adsl-065-006-194-171.sip.pns.bellsouth.net) |
| 02:34.39 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) |
| 02:34.44 | nestAr | ioerror: www.google.com, search for openclose.agi site:voip-info.org |
| 02:34.53 | *** part/#asterisk zignig (~simon@203.217.15.10) |
| 02:34.54 | nestAr | you'll find an example of gotoif there |
| 02:35.01 | Qwell | except the wiki is dead |
| 02:35.04 | nestAr | o |
| 02:35.10 | shepherd | what happened to it? |
| 02:35.10 | nestAr | well, then |
| 02:35.37 | nestAr | it dies from time to time |
| 02:35.39 | Inv_arp | hmm yep wiki not up |
| 02:35.41 | shepherd | oh.. haha |
| 02:35.56 | shepherd | they need to just cluster that |
| 02:36.00 | ioerror | seriously. |
| 02:36.14 | ioerror | It would be awesome if extensions.conf just had a man page |
| 02:36.19 | ioerror | Perhaps I will write one |
| 02:36.31 | Qwell | That'd be one long ass manpage |
| 02:36.33 | nestAr | ioerror: http://pastebin.ca/7528 |
| 02:36.38 | nestAr | there's the example |
| 02:36.45 | shepherd | ioerror: it's pretty much a programing language in itself |
| 02:36.51 | shepherd | programming |
| 02:37.09 | ioerror | shepherd, examples are still quite useful |
| 02:37.22 | shepherd | heh |
| 02:37.37 | shepherd | maybe i should go through pastebin |
| 02:37.45 | shepherd | and collect some examples, lol |
| 02:38.35 | Inv_arp | shepherd: programing, we knew what ya meant :) |
| 02:39.13 | stevecayona | I've been messing with AAH cause I'm to lazy to do it the CLI way. |
| 02:39.15 | nestAr | anyway.. the example is.... on 1100,3.. if status = closed, go to 1100,7.. if it's not true.. goto 1100,4 |
| 02:40.13 | ioerror | nestAr, how can i prompt the user to confirm though? I see that you can say if $VAR is such and suck. |
| 02:40.21 | shepherd | omg.. google is down :( |
| 02:40.34 | shepherd | now it's up |
| 02:40.51 | nestAr | ioerror: probably need an AGI to do that |
| 02:40.53 | nestAr | i'm not sure |
| 02:41.14 | ioerror | Necko, I need to learn about AGI I guess |
| 02:41.41 | ioerror | nestAr, even |
| 02:41.45 | shepherd | oh wait.. google is down |
| 02:41.48 | shepherd | wtf |
| 02:41.55 | shepherd | NO!!!!!!!! |
| 02:41.59 | Qwell | shepherd: Its all you |
| 02:42.04 | Inv_arp | shepherd: your down |
| 02:42.05 | shepherd | try searching |
| 02:42.19 | Qwell | woah |
| 02:42.26 | Inv_arp | ok fine |
| 02:42.27 | Qwell | They're putting images in search results now? |
| 02:42.38 | Qwell | search for "wtf". That scared the shit out of me |
| 02:42.47 | techie | imagine searching for pr0n |
| 02:42.50 | shepherd | i can't search :( |
| 02:43.05 | Inv_arp | Qwell: lol heh sean connery yea |
| 02:43.16 | Qwell | Is that new, or what? |
| 02:43.33 | nestAr | i don't see any images |
| 02:43.39 | nestAr | unless i click on "images" |
| 02:43.43 | Qwell | odd |
| 02:43.44 | Druken | Qwell: www.tubgirl.com |
| 02:43.47 | Qwell | its only coming up for "wtf" |
| 02:43.54 | techie | wtf.org |
| 02:43.56 | shepherd | qwell: it's some windows program on your box |
| 02:44.00 | shepherd | heh |
| 02:44.02 | Inv_arp | yep wtf brings up iages for me |
| 02:44.04 | Qwell | shepherd: I'm on Linux. :p |
| 02:44.06 | Inv_arp | err images |
| 02:44.23 | shepherd | i've seen it before in windows |
| 02:44.25 | shepherd | never linux |
| 02:44.46 | *** join/#asterisk IQ (~IQ@63-230-46-90.omah.qwest.net) |
| 02:44.57 | shepherd | and i still can't search :( |
| 02:48.22 | stevecayona | well google is ther so the problem is on your end |
| 02:48.55 | Apple | would anybody here be willing to give me a copy of their aix.conf, extensions.conf, and sip.conf that uses FWD and/or nufone? |
| 02:49.02 | Apple | so I can look at something to get a grasp of them |
| 02:51.10 | JamesDotCom | there's so many of those around it's not funny |
| 02:52.00 | JamesDotCom | http://www.google.com.au/search?q=asterisk+example+configs |
| 02:55.12 | Darwin35 | is wiki down |
| 02:55.55 | t3t | no, it's just you... and thousands of others |
| 02:56.17 | Darwin35 | ok |
| 02:56.21 | t3t | it's strange... it's accepting some port 80 connections, but not letting data through |
| 02:56.33 | Darwin35 | yeah |
| 02:57.03 | justinnnn | ppls |
| 02:57.10 | justinnnn | ive got sccp... when i install it.. i get thid |
| 02:57.38 | justinnnn | sccp_channel.c:49: error: structure has no member named `callerid' |
| 02:57.55 | t3t | that's supposed to happen SCCP = sisco's crappy communication protocol |
| 02:58.30 | justinnnn | is there anyway to get past it.. i need sccp for these stupid 7970 and 7914 things |
| 02:58.50 | t3t | looks like sccp isn't synced with the rest of the source on the release that you are using |
| 02:58.53 | t3t | which one is it? |
| 02:59.02 | justinnnn | im using cvs from like 15/01 |
| 02:59.10 | justinnnn | sccp i just downloaded off source-forge |
| 02:59.45 | t3t | try a more recent cvs and see what happens |
| 03:00.15 | *** join/#asterisk hurryupandwait (~costasmen@m815f36d0.tmodns.net) |
| 03:00.20 | hurryupandwait | hello |
| 03:00.27 | Darwin35 | ok who screwd with wiki |
| 03:00.33 | *** join/#asterisk dontmsgme (~none@207.215.252.80) |
| 03:00.34 | t3t | justinnnn: looks like someone else had a similar problem |
| 03:00.34 | t3t | http://lists.digium.com/pipermail/asterisk-users/2005-January/083623.html |
| 03:00.41 | Darwin35 | put wiki back where he belongs |
| 03:00.53 | justinnnn | ya whys the wiki down :( its resticting my voiping :P |
| 03:01.00 | t3t | Darwin35: I swear, i barely touched it... |
| 03:01.06 | hurryupandwait | is anyone here used Switchvox software? |
| 03:01.12 | dontmsgme | Xlite was loggin in now (though nothing has changed) it says "Login Timed out", if I changed no settings at all is the only option that the wireless network is flakey? |
| 03:01.14 | Darwin35 | stop abusing my wiki |
| 03:01.35 | *** join/#asterisk mishehu (~mishehu@adsl-65-42-150-153.dsl.chcgil.ameritech.net) |
| 03:01.41 | justinnnn | hmm they dont say howto fix it :( |
| 03:01.41 | t3t | Darwin35: I just asked the wiki what the last digit of Pi was... it will be done in a little while |
| 03:01.47 | t3t | justinnnn: nope |
| 03:01.56 | Darwin35 | hehe |
| 03:02.00 | justinnnn | :~( |
| 03:02.11 | t3t | justinnnn: maybe you can use stable |
| 03:02.15 | Darwin35 | this just means I can go to bed and deal wit hit in the am |
| 03:02.42 | t3t | Darwin35: google's cache is your friend... |
| 03:02.55 | t3t | A few of the pages I looked at were < 10 days old |
| 03:03.10 | shepherd | bah, i can't get google to work today |
| 03:03.19 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 03:03.44 | t3t | shepherd: huh? the mighty search thingie isn't working for you? |
| 03:03.46 | t3t | try google.ca |
| 03:04.15 | *** join/#asterisk lately (~dougb@chi.econ.usyd.edu.au) |
| 03:04.18 | stevecayona | well pick another search engine and be done with it |
| 03:04.19 | shepherd | hehe.. that works :) |
| 03:05.02 | t3t | you'll get some wierd canadian stuff, but hey, they are good entertainment sometimes ;) |
| 03:05.13 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 03:05.42 | mishehu | hmm seems like voip-info.org is down... anybody remember how to convert wav49 files to standard wav format with sox? |
| 03:05.50 | mishehu | my mind is drawing a blank |
| 03:06.19 | t3t | what's a wav49? |
| 03:07.11 | Corydon76-home | WAV format #49... aka MS-GSM |
| 03:07.37 | t3t | but of course it is |
| 03:08.36 | shepherd | http://lists.digium.com/pipermail/asterisk-users/2002-April/002020.html |
| 03:08.45 | mishehu | oh wait, think I might have it |
| 03:08.49 | shepherd | that's it |
| 03:08.52 | mishehu | -t wav -b -g |
| 03:09.27 | shepherd | oh wait, you're trying to do reverse |
| 03:09.29 | shepherd | heh |
| 03:09.32 | ioerror | Does anyone know how to change the number of threads asterisk has running? |
| 03:09.39 | mishehu | shepherd: yes, reverse. but I think I remembered it. |
| 03:12.25 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 03:12.39 | *** join/#asterisk _santiago_ (~santiago@201.245.167.104) |
| 03:12.59 | justinnnn | hey |
| 03:13.03 | justinnnn | does sccp work over nat ? |
| 03:14.06 | mishehu | justinnnn: just speculation that it is like SIP. |
| 03:14.10 | mishehu | but don't quote me on that |
| 03:14.20 | *** join/#asterisk loud (~ariel@datamarkets.com.mx) |
| 03:14.44 | neopher | i don't beleive you can, however, i have gotten it to work through vpn tunnels |
| 03:15.16 | *** part/#asterisk _santiago_ (~santiago@201.245.167.104) |
| 03:15.21 | neopher | whats up with the wiki tonight |
| 03:17.21 | neopher | quite tonight |
| 03:17.43 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 03:18.31 | ioerror | Asterisk is the sendmail of PBX software. |
| 03:18.38 | shepherd | ugh! |
| 03:18.48 | shepherd | postfix of PBX software :) |
| 03:18.52 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 03:18.53 | loud | * is not shit |
| 03:18.57 | loud | riight. |
| 03:18.59 | loud | :) |
| 03:19.06 | ioerror | It's like m4 without the pleasure ;-) |
| 03:19.36 | t3t | can someone change the channel info to add 'yes, you're not dreaming. The wiki is down..." |
| 03:19.59 | *** join/#asterisk hacim (micah@micha.hampshire.edu) |
| 03:20.15 | *** join/#asterisk Beirdo_ (~gjhurlbu@beirdo.user) |
| 03:20.30 | shepherd | loud: you have something against asterisk? |
| 03:21.29 | *** join/#asterisk hurryupandwait (~costasmen@m815f36d0.tmodns.net) |
| 03:21.29 | t3t | I think he has something against sendmail |
| 03:21.43 | hurryupandwait | hi |
| 03:21.56 | shepherd | oh.. haha |
| 03:22.30 | hurryupandwait | Anyone here using Switchvox? |
| 03:22.42 | Darwin35 | night kids |
| 03:22.58 | *** part/#asterisk stevecayona (~neondog@adsl-065-006-194-171.sip.pns.bellsouth.net) |
| 03:23.04 | neopher | night old man |
| 03:23.11 | neopher | hehe |
| 03:23.21 | ioerror | Does anyone know how to change the number of threads asterisk uses during runtime? |
| 03:23.31 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) |
| 03:23.38 | ioerror | Also, is there a way to limit who can connect to my asterisk server for IAX? |
| 03:23.42 | ioerror | Otherthan a firewal? |
| 03:24.40 | shepherd | ioerror: mark is working on the threading problem |
| 03:24.47 | loud | of course, access lists. |
| 03:24.58 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) |
| 03:24.59 | shepherd | and you can individualy add iax users yes :) |
| 03:25.00 | ioerror | shepherd, what problem? |
| 03:25.08 | loud | shepherd, no, absolutely not, i do against sendmail tho :> |
| 03:25.17 | shepherd | io: it only uses like 2500 threads or something |
| 03:25.22 | shepherd | sorry.. 250 |
| 03:25.33 | shepherd | which is small |
| 03:25.35 | shepherd | hah |
| 03:25.37 | ioerror | shepherd, i have ten processes of asterisk running, is it possible to make it 5? |
| 03:25.46 | ioerror | Is it compile time or is it in a config somewhere? |
| 03:25.57 | ioerror | shepherd, what I am trying to do is basically only allow voicepulse to connect to me and no one else. |
| 03:25.58 | shepherd | compile time and config from what i know |
| 03:28.27 | *** join/#asterisk scythelx (~wow@pc-24-151-59-224.newt1.ct.charter.com) |
| 03:29.14 | ioerror | shepherd, do you know where i can configure asterisk to only allow the voicepulse part at least? |
| 03:29.22 | ioerror | I guess the number of processes doesn't matter |
| 03:29.57 | scythelx | hello all, anyone know if this is possible: plug in a linksys voip router somewhere out on the internet, tell asterisk to look for the MAC address and configure it, or know which ext it is based on the MAC address? trying to avoid configuring each piece of equiptment |
| 03:31.12 | nestAr | mac addresses don't cross the internet |
| 03:31.22 | nestAr | only a LAN |
| 03:32.15 | scythelx | ahh ok, someone told me you can just buy a linksys voip router, and use ATT's callvantage service, give them yer mac address and thats it |
| 03:32.40 | scythelx | so thats not possible then right |
| 03:33.00 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:33.46 | heison | is the wiki down? |
| 03:33.48 | *** join/#asterisk dontmsgme (~none@207.215.252.80) |
| 03:34.00 | neopher | heison: yes |
| 03:34.09 | dontmsgme | What else can I use with Nufone besides Xlite (software phone wise) |
| 03:34.16 | scythelx | sjphone |
| 03:34.26 | dontmsgme | URL? |
| 03:34.41 | scythelx | http://www.sjlabs.com/ |
| 03:35.08 | AgiNamu | FireFly |
| 03:35.11 | heison | does anyone know if I can run * on Windows? with hardware support on the Digium cards? |
| 03:35.17 | AgiNamu | heison no |
| 03:35.17 | file[laptop] | heison: no. |
| 03:35.26 | ioerror | heison, why window? |
| 03:35.31 | Beirdo | sigh |
| 03:35.36 | AgiNamu | Cause Microsoft's CLR is better than Mono. |
| 03:35.43 | Beirdo | OK, this is odd. |
| 03:35.50 | Beirdo | just upgraded to 1.0.6 |
| 03:35.51 | AgiNamu | and SQL Server 2000/2005 kicks the shit out of MySQL? ;) |
| 03:35.52 | dontmsgme | AgiNamu when I installd firefly it kept asking me for my "Firefly Phone #" |
| 03:35.53 | heison | because the MS folks does not believe their OS cannot handle the load |
| 03:35.58 | dontmsgme | How do you use that with Nufone?? |
| 03:36.02 | *** join/#asterisk qwerp (~abc@219.93.57.58) |
| 03:36.04 | AgiNamu | then you installed the wrong version |
| 03:36.07 | qwerp | harlo |
| 03:36.08 | AgiNamu | get the version for 3rd party networks |
| 03:36.15 | AgiNamu | querp hyalo |
| 03:36.26 | qwerp | i was wondering, |
| 03:37.01 | qwerp | is there a application which is a combination of MusicOnHold + Background? |
| 03:37.19 | dontmsgme | Its weird I can Ssh into my box, I can ping my box, but Xlite won't connect it says "login timeout" I'm going to try with SJphone |
| 03:37.41 | qwerp | meaning when i run the application, it will listen to the music on hold but still able to capture input? |
| 03:37.59 | qwerp | dontmsgme : check ur firewall... |
| 03:38.01 | neopher | dontmsgme: is port 5060 and your sip data ports open? |
| 03:38.19 | qwerp | dontmsgme : firewall must be blocking it.. |
| 03:38.55 | dontmsgme | Ive got port 5060 and 8000 forwarded |
| 03:40.07 | PatrickDK | not nearly enough |
| 03:40.26 | dontmsgme | What else? |
| 03:40.34 | PatrickDK | every udp port sjphone uses |
| 03:40.37 | PatrickDK | not sure what it uses |
| 03:40.44 | PatrickDK | asterisk normally uses 10000-20000 |
| 03:40.53 | dontmsgme | Can you forward that in one block |
| 03:41.00 | dontmsgme | Someone told me you got to do it one at a time |
| 03:41.03 | qwerp | if u do that, even though u got registered, u might not be able to talk.. |
| 03:41.04 | PatrickDK | ya |
| 03:41.37 | mog_home | ???? |
| 03:41.37 | mog_home | what |
| 03:42.21 | PTG123 | anyone know much about configuring cisco 7960gs? |
| 03:42.59 | dontmsgme | Is there 2 versions of SJphone like there are two versions of Firefly |
| 03:43.49 | *** part/#asterisk lately (~dougb@chi.econ.usyd.edu.au) |
| 03:44.18 | qwerp | any application that able to play musiconhold and capture inputs? |
| 03:46.02 | Beirdo | and now when I call out the zaptel, and then hangup, it sees a busy signal after the call, the call progress doesn't seem to behave anymore |
| 03:46.54 | dontmsgme | Wheres the official firefly download webstie |
| 03:47.04 | Beirdo | qwerp: your best bet may be an empty call queue, but I haven't tried that |
| 03:51.25 | justinnnn | hats this mean |
| 03:51.25 | justinnnn | Dead SCCP client: SEP0013196EBF2F |
| 03:51.31 | justinnnn | i got sccp working :)) i can recieve calls now yay :) |
| 03:52.28 | Silik0n | any as5300 fax experts about tonight? |
| 03:52.45 | dontmsgme | Is wiki down because I can't get the link for Firefly's website |
| 03:53.03 | t3t | justinnnn: how did you do it? |
| 03:53.21 | Sedorox | ~firefly |
| 03:53.22 | jbot | it has been said that firefly is http://virbiage.com/firefly/download/firefly-thirdparty.exe |
| 03:53.33 | Sedorox | and yes |
| 03:53.36 | Sedorox | ~wiki-status |
| 03:53.37 | jbot | i guess wiki-status is Current Down |
| 03:53.56 | *** join/#asterisk KirkL (~me@c-24-22-57-111.client.comcast.net) |
| 03:54.06 | Qwell | Hmm. That firefly one needs to be better. heh |
| 03:54.08 | Qwell | save me a google |
| 03:54.13 | Sedorox | lol |
| 03:54.27 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 03:54.36 | Qwell | The entire first page of "firefly" on google is related to the show |
| 03:54.44 | implicit | hi |
| 03:54.47 | Qwell | softphone or something? |
| 03:54.47 | *** join/#asterisk jmhunter (~jacob@wire3-148.razzolink.com) |
| 03:54.47 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 03:55.18 | *** join/#asterisk _6Flamez_ (lklk@00045a809589.click-network.com) |
| 03:56.03 | Beirdo | argh argh argh |
| 03:56.15 | Beirdo | and after reboot, my zaptel's not working? |
| 03:58.34 | Beirdo | haha |
| 03:58.49 | Beirdo | somehow not modprobing wcfxo |
| 03:58.51 | Silik0n | we dont need no stinking zaptel |
| 03:59.07 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 03:59.22 | niZon | aww man no wiki today :( |
| 03:59.32 | Silik0n | use google and that caches |
| 03:59.35 | niZon | yeah |
| 03:59.41 | *** join/#asterisk jey (~jey@221.219.230.31) |
| 03:59.41 | techie | use your brain |
| 03:59.45 | dontmsgme | Mar 15 19:50:55 NOTICE[-235193424]: chan_iax2.c:5183 socket_read: Rejected connect attempt from 207.215.252.80 |
| 03:59.50 | `Sauron | what if its broken? |
| 03:59.54 | Silik0n | site:voip-info.org whatever |
| 03:59.57 | dontmsgme | Why is asterisk rejecting the attempted connection from my softphone? |
| 04:00.57 | *** join/#asterisk jey (~jey@221.219.230.31) |
| 04:01.03 | PTG123 | anyone here use a cisco 7960? |
| 04:01.04 | justinnnn | hey |
| 04:01.04 | justinnnn | Mar 16 15:18:26 ERROR[26539]: sccp_actions.c:252 sccp_handle_line_number: SCCP device SEP0013196EBF2F requested a line configuration for unknown line 3 |
| 04:01.07 | justinnnn | what does that mean ? |
| 04:01.16 | justinnnn | its on a 7960 |
| 04:01.19 | niZon | PTG123: I'm trying to get one :P |
| 04:01.39 | PTG123 | nizon: heh i got one, but i can't figure out something |
| 04:01.46 | Beirdo | anyone have any hints for fixing call progress on zaptel? |
| 04:01.47 | KirkL | anyone know where to get those discontinued single port fxs and fxo cards? |
| 04:02.01 | PTG123 | justinnnn: hey mine sits there saying configuring ip.. how do i make it allow me to unlock it so i can change the settings? |
| 04:02.25 | niZon | PTG123: I'm trying to learn all I can about them before I start using one, don't want to waste $250 or so :P |
| 04:02.28 | dontmsgme | Mar 15 19:53:56 NOTICE[-235156560]: chan_iax2.c:5183 socket_read: Rejected connect attempt from 207.215.252.80 |
| 04:02.35 | PTG123 | nizon: got mine for $132 |
| 04:02.40 | dontmsgme | What does that mean? |
| 04:02.41 | niZon | PTG123: ebay? |
| 04:02.44 | PTG123 | nizon: yah |
| 04:02.53 | niZon | Did you get the SIP firmware for it? |
| 04:03.23 | dontmsgme | Who me? |
| 04:03.26 | b4nkrup7 | its old... http://web.archive.org/web/20040216055911/www.voip-info.org/tiki-index.php |
| 04:03.27 | niZon | PTG123 |
| 04:03.31 | PTG123 | someone emailed it to me yes |
| 04:03.36 | PTG123 | but i can't get it without getting in |
| 04:03.37 | niZon | hmm |
| 04:03.42 | PTG123 | can i reset to factory defaults on this thing? |
| 04:03.54 | niZon | there's probablly a way... |
| 04:04.02 | niZon | does it get an IP via DHCP? |
| 04:04.04 | techie | the hard way |
| 04:04.19 | niZon | (talking to PTG123) :P |
| 04:05.03 | PTG123 | nizon: yes |
| 04:06.03 | niZon | PTG123: did you try configuring your DHCPd to send it a TFTP server IP? |
| 04:06.17 | PTG123 | no idea how to do that |
| 04:06.23 | niZon | hm |
| 04:07.32 | niZon | PTG123: what do you use for a DHCP server? |
| 04:07.40 | PTG123 | cisco wrt54g |
| 04:08.49 | niZon | ah, I have a WRT54GS, i think the DHCP server is pretty limited |
| 04:09.01 | niZon | you'll have to setup one on a spare box |
| 04:15.34 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 04:19.00 | *** join/#asterisk Trepalium (~chadk@wnpgmb02dc1-59-91.dynamic.mts.net) |
| 04:19.12 | dontmsgme | Anyone want to login to my router setup page so they can see if I have something forwarded wrong if this is the reason why Xlite won't connect |
| 04:19.21 | *** join/#asterisk Blah- (~nunya@ip68-227-198-28.dc.dc.cox.net) |
| 04:19.38 | Blah- | Ahoy. |
| 04:21.12 | Blah- | So, someone thrust this link at me, I liked it, downloaded the iso, burned it to disk, but I still have no idea how to set it up, or what it does, or what else I need other than a run down p3. Thoughts? |
| 04:21.13 | Beirdo | hmph |
| 04:21.39 | niZon | Blah-: asterisk@home? |
| 04:21.54 | Blah- | This kinda fits in the glittery things category. |
| 04:22.05 | Blah- | niZon: yeah. |
| 04:22.34 | niZon | to set it up you pop the cd in your machine, let it boot to the cd, then hit enter a bunch of times :P |
| 04:22.40 | *** join/#asterisk Veto (mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 04:22.41 | mikegrb | Blah-: first thing you need to do is throw the cd away |
| 04:22.49 | niZon | it will install linux, asterisk and lots of fun stuff |
| 04:22.51 | mikegrb | Blah-: it is crap |
| 04:22.56 | Blah- | Is there a 'for-dummies' manual, for those who just want to tinker with it? |
| 04:23.06 | mikegrb | download a debian netinstall iso |
| 04:23.11 | *** join/#asterisk Damin (~damin@nucleus.nacs.net) |
| 04:23.12 | mikegrb | or ubuntu install cd |
| 04:23.23 | niZon | Blah-: http://asteriskathome.sourceforge.net/ |
| 04:23.36 | t3t | Does anyone know how I can capture digits that are sent within a few hundred miliseconds of a connection to an analog (zap) line? |
| 04:23.37 | *** join/#asterisk Redb3ard (~oylerj@24.125.144.228) |
| 04:23.40 | Redb3ard | hi guys |
| 04:23.57 | t3t | hi Redb3ard |
| 04:23.59 | Redb3ard | im still trying to figure out what hardware i want to get |
| 04:24.07 | `Sauron | t3t: Why so quickly? |
| 04:24.21 | t3t | it's a PBX sending call status |
| 04:24.26 | `Sauron | oh |
| 04:24.28 | Blah- | See, I went there, and perhaps I'm blind, but I didn't see what it's for, just a bunch of buzzwords about voip. If I wanted mumbojumbo, I'd ask a ccna for help :) *ducks* |
| 04:24.36 | Redb3ard | im really thinking i want to get a TDM400P |
| 04:24.41 | Redb3ard | but theyre kinda pricey |
| 04:24.57 | ioerror | Anyone here know why the Maikefile for Asterisk is hardcoded to use bison? |
| 04:24.59 | t3t | `Sauron: if I get this to work, I may have a cool replacement for standard VM systems |
| 04:25.06 | `Sauron | thats cool |
| 04:25.18 | Redb3ard | anyone have any recommendations? |
| 04:25.34 | t3t | Redb3ard: what are you using it for? |
| 04:25.47 | Blah- | What I'm after is, what hardware do I need for making 2 aterisk pbxs talk to each other, from one analog phone, to another? |
| 04:25.56 | Redb3ard | t3t, nothing yet... but i can think of a 1000 things |
| 04:26.01 | Blah- | ...or am I missing the point? |
| 04:26.12 | t3t | How much money is burning a hole in your pocket then? |
| 04:26.15 | Redb3ard | blah, an ethernet cable |
| 04:26.16 | Sedorox | Blah-: you want them to talk over a analog line? |
| 04:26.21 | Blah- | yeah. |
| 04:26.24 | Sedorox | why? |
| 04:26.25 | Blah- | analog line. |
| 04:26.42 | Blah- | Um, no idea. it just popped in my head. |
| 04:26.43 | Damin | Blah? |
| 04:26.48 | Sedorox | like... dial asterisk from asterisk over analog.. or provide analog access |
| 04:26.57 | Sedorox | from=to |
| 04:27.29 | Redb3ard | ive got a hell of an idea for a project, myself, but i dunno how many people dont mind if theyre landline is tied up 24/7 |
| 04:27.45 | Sedorox | Redb3ard: ? |
| 04:27.45 | t3t | `Sauron: I can successfully detect the digits with a few "background(25ms sound file)" statements, but I'd like something more elegant |
| 04:27.47 | *** join/#asterisk jjg (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net) |
| 04:27.49 | jjg | hi |
| 04:27.49 | Redb3ard | if i could find a few like-minded people in other cities, we may start offering free long distance |
| 04:28.01 | Redb3ard | not sure if just among the group, or if to the general public |
| 04:28.02 | jjg | anyone got any advice on debugging IAX2 dtmf? |
| 04:28.10 | mikegrb | Redb3ard: been done |
| 04:28.13 | Sedorox | Redb3ard: thats already being done... |
| 04:28.19 | jjg | i'm not seeing any dtmf reception in the CLI after i terminate through nufone |
| 04:28.21 | t3t | Redb3ard: a la Bellster^w FWDOut? |
| 04:28.29 | Redb3ard | cool |
| 04:28.53 | Sedorox | since wiki is down.. I don't have a link.. but there is a place that whoever lnks to it it providers their landline for access |
| 04:29.14 | Redb3ard | does anyone have it set up as a phone gateway? |
| 04:29.19 | Sedorox | oo |
| 04:29.22 | Redb3ard | so you can literally just call a local number? |
| 04:29.28 | mikegrb | yes |
| 04:29.28 | Blah- | Ok, Sedorox: assume I'm a complete idiot, (I am, at the moment, when it comes to voip), how would I go about asterisk to asterisk over analog, and has someone already drawn a how-to on this? |
| 04:29.33 | t3t | Redb3ard: hundreds of people |
| 04:29.35 | Sedorox | jbot: no, wiki-status is Up and Running |
| 04:29.36 | jbot | okay, Sedorox |
| 04:29.37 | mikegrb | it is excactly what your great idea was |
| 04:29.37 | Damin | Redb3ard: dundi |
| 04:29.39 | Redb3ard | cool |
| 04:29.42 | Sedorox | ~wiki-status |
| 04:29.43 | jbot | somebody said wiki-status was Up and Running |
| 04:29.43 | ariel_ | Redb3ard, go to bellster from FWD. But be careful if you let your pots line to be used by others you might just get crank calls via your line. |
| 04:29.44 | t3t | http://www.fwdout.net/web/ |
| 04:29.46 | Sedorox | :):) |
| 04:29.52 | jjg | heh |
| 04:29.54 | jjg | hear that |
| 04:30.10 | Sedorox | Blah-: there isn't really a point to have asterisk connect to another asterisk box over a analog line |
| 04:30.12 | jjg | do you have to have a x100p or can you use a winmodem? |
| 04:30.14 | Sedorox | since it would only support one call |
| 04:30.28 | *** join/#asterisk jey (~jey@221.219.230.31) |
| 04:30.48 | Redb3ard | nah, no pstn access |
| 04:31.08 | ariel_ | only reason to connect one asterisk box to another via analog line is for testing only. But it sure makes no sence. |
| 04:31.22 | Redb3ard | in any event, if i wanted each phone in my house to be an extension, not directly connected to the landline... |
| 04:31.35 | Redb3ard | what do i want? |
| 04:31.49 | Blah- | This is where I beat my head into the wall, and repeat "why did I click the link, why did I..." |
| 04:31.50 | ariel_ | Redb3ard, have you see the project called Asterisk@home? |
| 04:31.57 | PTG123 | anyone know how to make it stop saying configuring ip, and let me unlock the settings on my cisco 7960? |
| 04:32.11 | Redb3ard | ariel, ive seen lots of stuff, trying to make sense of it |
| 04:32.14 | ariel_ | PTG123, take a gun and shoot it. |
| 04:32.34 | Apple | yay, wiki is back up :) |
| 04:32.36 | Sedorox | Redb3ard: hardware wise... depends how many phones you want.. either you can get a few ATA's.. run the lines back to the asterisk back and have a TDM card or two.. or get a channel bank... |
| 04:32.37 | ariel_ | Redb3ard, what then do you really want to do. Set up an asterisk box for your home? |
| 04:33.10 | *** join/#asterisk dontmsgme (~none@207.215.252.80) |
| 04:33.22 | Redb3ard | ariel, i have no specific goals, im a hobbyist, not some managerial asshat jumping on a bandwagon |
| 04:33.26 | dontmsgme | I got $50 egold to someone who can help me get Xlite->ASterisk workin again msg me |
| 04:33.30 | Redb3ard | i have no clue, other than to do cool stuff |
| 04:33.50 | Blah- | What Redb3ard said :) |
| 04:33.53 | Redb3ard | and id like to start by having 3 or 4 phones on seperate extensions, not directly attached to the landline |
| 04:34.01 | Damin | Redb3ard: Yes. Cool stuff. Good. Now get to work and shut the hell up. ;) |
| 04:34.02 | ariel_ | Redb3ard, example my home one asterisk@home box. One TDM11b card. One inbound PSTN line for 911 and other local calls, 1 of the fxs ports on the TDM is for my fax machine. A siupra 2100 with 2 fxs for 2 phones in the house. |
| 04:34.32 | Silik0n | god damnit |
| 04:34.38 | Silik0n | who uses a as5300? |
| 04:34.40 | ariel_ | The other phones I have is in my office a sipuar 841 works really nicely. |
| 04:34.41 | Silik0n | for faxing |
| 04:34.41 | Blah- | ariel_: all in all, with just the cards, how much, and where can I get a bundle of the same config? |
| 04:34.52 | Damin | Silik0n: Not me. |
| 04:34.57 | Damin | Silik0n: They suck. |
| 04:34.58 | Redb3ard | ariel, so you have 2 inbound, 2 outbound? |
| 04:34.58 | dontmsgme | $100 egold? |
| 04:35.04 | Blah- | as5300 is a dialin router... |
| 04:35.10 | Silik0n | Damin yeah well its what I have to playwith in my lab |
| 04:35.22 | Damin | Silik0n: Stop bitching and make it work then. ;) |
| 04:35.25 | ariel_ | Blah-, go to digium and get the dev kit. That is what it comes with. Get your self an Dell 400 dollar server and your set. |
| 04:35.27 | Silik0n | Blah- it can be a VoIP gateway or a fax on/off-ramp too |
| 04:35.39 | Blah- | I've never heard of it being used for that, neat :) |
| 04:35.48 | ariel_ | or go to voipsupply and get the items there. I got the sipura phones and ata 2100 from them. |
| 04:35.55 | *** join/#asterisk Koshatul (~evangelio@CPE-138-217-191-19.qld.bigpond.net.au) |
| 04:35.57 | Silik0n | Damin: i'm working on it |
| 04:35.57 | Blah- | we just threw 2 pri lines at it, and viola :) |
| 04:36.16 | Silik0n | Blah-: gotta have the DSP cards in it... the mmic cards wont do it |
| 04:36.21 | Blah- | ah. |
| 04:36.36 | Blah- | ariel_: thanks. |
| 04:36.43 | Silik0n | so I have the right IOS and the right cards, not just getting it to play nice |
| 04:37.47 | Blah- | so, 200$ is cheap, then? |
| 04:37.48 | Damin | Silik0n: What are you attempting to do with it? |
| 04:38.23 | dontmsgme | What ports do you usually forward for Xlite? |
| 04:38.38 | ariel_ | Blah-, it's a good price. |
| 04:39.02 | ariel_ | Silik0n, you have the one with the h323 gateway or with the sip gateway? |
| 04:39.46 | ariel_ | dontmsgme, I have 5060 and 10,000 to 20,000 open udp |
| 04:39.59 | Silik0n | SIP |
| 04:40.15 | ariel_ | Silik0n, then it's allot easyer to configure and get working. |
| 04:40.24 | Silik0n | but I'm not doing that... the 5300 rcvs the faxes and emails them over to me via smtp |
| 04:40.48 | dontmsgme | Ariel thanks |
| 04:41.19 | MajestiK | I'm trying to upgrade firmware on a Cisco 7960 phone, and I'm having some troubles with the tftp, it seems to be adding ^Js after the name, even if I echo into the OS79XX.TXT file.. anyone have any ideas on what I should try? |
| 04:41.34 | dontmsgme | How is it possible that Xlite would be working 3 hours ago and now it won't login? |
| 04:41.59 | Blah- | cool, a fxo card for asterisk goes for 7$ :) |
| 04:42.23 | *** join/#asterisk CoderCR (~creyna@adsl-67-112-135-29.dsl.sndg02.pacbell.net) |
| 04:42.30 | CoderCR | hey all |
| 04:43.03 | CoderCR | i wanted to know how to debug what command was been sent to asterisk from a zaptel driver to see asterisk may not be handling it. |
| 04:43.04 | ariel_ | Blah-, that is a clone just becareful of them. |
| 04:43.15 | datareactor | dontmsgme is asterisk is working fine ? |
| 04:43.15 | CoderCR | what is the best way to find out what the driver is sending to asterisk |
| 04:43.28 | dontmsgme | Datareactor it was |
| 04:43.40 | Blah- | ariel_: gotcha |
| 04:43.42 | ariel_ | CoderCR, turn debug on |
| 04:43.44 | dontmsgme | So I switched phones, I used IAXphone and then it said |
| 04:43.55 | CoderCR | with this command /usr/sbin/asterisk -vvvvvv -g  -dddddd -c? |
| 04:44.05 | dontmsgme | "connection refused from IP-with-IAX-phone" |
| 04:44.09 | CoderCR | or on zaptel itself |
| 04:45.39 | `Sauron | CoderCR: Don't think that's a bit overkill? :) |
| 04:45.48 | CoderCR | No.. |
| 04:45.53 | CoderCR | i need to know this is happening |
| 04:45.55 | ariel_ | CoderCR, debug channel |
| 04:46.05 | dontmsgme | Can you use nufone directly with firefly |
| 04:46.11 | Blah- | So, with the dev kit, I can use an analog line to send/receive, then place my phones whereever, plug into the asterisk unit, and I'm done? |
| 04:46.19 | CoderCR | my channel bank is sending commands over the zaptel driver and asterisk is not handling certain disconnects |
| 04:46.24 | *** join/#asterisk SuperBawlz (user@24.247.116.57.kzo.mi.chartermi.net) |
| 04:46.30 | ariel_ | Blah-, yes |
| 04:46.33 | Blah- | Nice . |
| 04:46.47 | CoderCR | ariel: hehehe sorry forgot about that |
| 04:46.55 | SuperBawlz | I have a couple questions if anyone has a moment of time |
| 04:47.09 | CoderCR | o |
| 04:47.11 | CoderCR | wait |
| 04:47.11 | ariel_ | SuperBawlz, ask way |
| 04:47.15 | ariel_ | away..... |
| 04:47.16 | CoderCR | i dont see a zap debug option |
| 04:47.29 | SuperBawlz | OK. |
| 04:47.32 | ariel_ | CoderCR, debug channel XXXX |
| 04:47.37 | Redb3ard | hmm |
| 04:47.40 | ariel_ | do help |
| 04:47.41 | SuperBawlz | I'm setting up my first Asterisk pure SIP server |
| 04:47.54 | ariel_ | SuperBawlz, there is nothign pure about sip |
| 04:47.58 | Redb3ard | ok, so FXS is for attaching phones to, and FXS will generate dialtone/ringtone? |
| 04:47.59 | SuperBawlz | LOL |
| 04:48.02 | SuperBawlz | Tru dat |
| 04:48.09 | `Sauron | yes |
| 04:48.19 | ariel_ | Redb3ard, yes fxo is for office or to connect the incoming pots line |
| 04:48.31 | Redb3ard | we make do with 3 phones in the house now, but i can see me wanting to do 5 or 6 in some cases |
| 04:48.55 | Redb3ard | and ideally, id want each on its own extension |
| 04:48.58 | ariel_ | Redb3ard, yes so |
| 04:49.04 | dontmsgme | Simple question: How do you show all IAX phones registered in your box |
| 04:49.08 | Redb3ard | trying ot figure out what hardware i want |
| 04:49.14 | dontmsgme | Or can you only do Sip show peers |
| 04:49.15 | `Sauron | iax2 show peers |
| 04:49.17 | SuperBawlz | ok. I need to know in the sip.conf what "progressinband" does |
| 04:49.22 | ariel_ | iax2 show registry or peers |
| 04:49.44 | Apple | ok, I got further! |
| 04:49.55 | Redb3ard | dumb question, will an FXS generate callerid signal? whateever thats called, i forget |
| 04:50.01 | Apple | Calls are ended with "404 not found" in sjphone |
| 04:50.04 | `Sauron | yes |
| 04:50.06 | Apple | and they still don't show up in asterisk |
| 04:50.08 | ariel_ | SuperBawlz, I have not seen that one are you running head from cvs? |
| 04:50.11 | Apple | but I know my phone is registered! |
| 04:50.12 | Apple | why? |
| 04:50.20 | *** join/#asterisk jedaustin (~chatzilla@host5.twingeckos.net) |
| 04:50.30 | SuperBawlz | now. |
| 04:50.32 | SuperBawlz | oops |
| 04:50.34 | SuperBawlz | no |
| 04:50.42 | jedaustin | voip-info.org down I suppose? |
| 04:50.43 | SuperBawlz | I have a DL version |
| 04:50.47 | SuperBawlz | yes |
| 04:51.01 | SuperBawlz | voip-info is down. |
| 04:51.33 | Apple | no |
| 04:51.35 | Apple | it's back up! |
| 04:51.36 | dontmsgme | I've tried 3 different softphones and they all say: Mar 15 20:42:47 NOTICE[-234665040]: chan_iax2.c:5183 socket_read: Rejected connect attempt from 207.215.252.80 |
| 04:51.37 | Apple | I'm on it :) |
| 04:51.42 | SuperBawlz | ok. BRB |
| 04:51.53 | jedaustin | I took the plunge today.. ordered Broadvoice, downloaded asterisk@home, installed it.. now trying to get the setup working, The things on the Broadvoice site don't seem to completely jive with what I see in the settings |
| 04:51.55 | SuperBawlz | I will look there for the progressinband thing. |
| 04:52.05 | ariel_ | dontmsgme, your using iax2 phones or sip? |
| 04:52.21 | `Sauron | jedaustin: follow the instructions here: |
| 04:52.21 | jedaustin | Whoa, now theres timing.. voip-info up :) |
| 04:52.28 | `Sauron | http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup |
| 04:52.29 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 04:52.56 | dontmsgme | Xlite/IAXphone/now Firefly |
| 04:52.57 | ariel_ | jedaustin, try the #amportal they might be able to help with those settings it's done via amp |
| 04:53.14 | SuperBawlz | bacl |
| 04:53.16 | SuperBawlz | back |
| 04:53.26 | SuperBawlz | Studio POS MS KB |
| 04:53.28 | ariel_ | dontmsgme, xlite is only sip. |
| 04:53.33 | SuperBawlz | stupid |
| 04:53.35 | dontmsgme | Ok.. |
| 04:53.39 | SuperBawlz | OK. |
| 04:53.44 | dontmsgme | Xlite is the one that wont login at all |
| 04:53.47 | SuperBawlz | I could find in on the site there. |
| 04:53.52 | dontmsgme | Firefly and IAXphone say "rejected" |
| 04:54.05 | dontmsgme | Is it possibly some firewall blocking mechanism in the linux box |
| 04:54.17 | dontmsgme | Though everythin was working 4 hours ago |
| 04:54.21 | ariel_ | dontmsgme, you will need to make sure your settings are correct in iax.conf |
| 04:54.27 | harryvv | arial, I guess the linux beta trial for the xlite is up. Xten is not responding. Is there a softphone that will work with linux in a 32/64 enviroment? |
| 04:54.31 | ariel_ | for the xlite set it up in sip.conf |
| 04:54.31 | SuperBawlz | If you guys don't know I will just leave it off. |
| 04:54.34 | CoderCR | well the debug gave me nothing |
| 04:54.50 | ariel_ | xlite works jut fine. |
| 04:54.57 | harryvv | dont do a show sip peers |
| 04:55.01 | ariel_ | sorry I use the one for windows |
| 04:55.05 | harryvv | i use xlite all the time. |
| 04:55.12 | harryvv | for windows |
| 04:55.18 | harryvv | but want something for this machine. |
| 04:55.30 | ariel_ | SuperBawlz, good Idea |
| 04:55.45 | harryvv | Seems the supply of a open source softphone that looks half way decent is scarce. |
| 04:55.54 | dontmsgme | I dont see any errors in sip.conf that would stop Xlite from connecting |
| 04:56.25 | ariel_ | dontmsgme, post your settings on http://pastebin.ca and let us see them. |
| 04:57.17 | ariel_ | CoderCR, what is happening? |
| 04:57.19 | SuperBawlz | Do you know where I can download an open source win32 SIP software for placing calls into the asterisk system? |
| 04:57.28 | `Sauron | sjphone |
| 04:57.31 | ariel_ | www.xten.com |
| 04:57.34 | SuperBawlz | thanks |
| 04:57.36 | ariel_ | get xlite it works. |
| 04:57.46 | `Sauron | xlite sucked when I tried |
| 04:57.49 | SuperBawlz | this place looks nice |
| 04:57.59 | ariel_ | `Sauron, sorry to hear that. |
| 04:58.20 | SuperBawlz | What would you suggest Sauron? |
| 04:58.29 | `Sauron | shrug |
| 04:58.30 | dontmsgme | Ok hold on |
| 04:58.31 | CoderCR | ariel_: my channel bank is either no sending a disconnect command to asterisk or asterisk is not understanding the channel getting sent by the channel bank |
| 04:58.34 | `Sauron | I don't use softphones |
| 04:58.48 | dersteer | softphones suck |
| 04:58.51 | dersteer | imho |
| 04:59.01 | _6Flamez_ | they're fine for testing.. |
| 04:59.02 | ioerror | What's the best softphone for linux? |
| 04:59.03 | Apple | in my config, I have a @S$, (part of a users passwd), how can I make it work? |
| 04:59.05 | CoderCR | ariel: the problem occurs when the lines are bad only. |
| 04:59.05 | ariel_ | CoderCR, which channel bank and how do you have it connected to the asterisk box. pri t1 em&wink? |
| 04:59.13 | jedaustin | Better than no phone.. I'd use it on my laptop for 'emergencies' only. |
| 04:59.13 | Apple | when it tries, it says it can't find that network or some BS |
| 04:59.23 | spackle | Agreed, with softphones the effort is finding the one that sucks least. |
| 04:59.36 | Apple | spackle: sjphone works fairly well |
| 04:59.43 | Apple | kphone isn't too bad, but sjphone is better |
| 04:59.49 | Apple | and I want to try gnophone |
| 04:59.56 | harryvv | xlite is the best so far. |
| 04:59.58 | SuperBawlz | where is sjphone? |
| 04:59.59 | CoderCR | ariel: i have a rhino and it is connected correctly. it has to do with poor lines from pac bell.. the stuff works on good lines perfectly.. just not on this site. |
| 05:00.06 | Apple | harryvv: for winbloze, yes |
| 05:00.06 | Blah- | quick question, about how much bw is used, during a voip call, over a network, and is it configurable? I'd like to cram as many calls as possible with 384k. |
| 05:00.13 | Apple | ~google sjphone |
| 05:00.23 | shepherd | Apple: did you try \@\$ |
| 05:00.34 | dontmsgme | Ok Ariel I found out in iax.conf I simply had a typo on top of the file it was little " ' " |
| 05:00.36 | Apple | shepherd: didn't like it :/ |
| 05:00.38 | shepherd | or is /@/$ |
| 05:00.42 | shepherd | heh |
| 05:00.43 | shepherd | yeah |
| 05:00.44 | ariel_ | Blah-, that really depends on which codec your using. |
| 05:00.46 | Apple | hrmm, will try the / |
| 05:00.47 | dontmsgme | But now when I call my DID it still doesnt ring it just says ring/connected/time out |
| 05:01.00 | SuperBawlz | is this Soft Joy labs? |
| 05:01.11 | Blah- | So, let's say, whatever's default, 32k, 24k? |
| 05:01.12 | ariel_ | dontmsgme, your have configured something incorrectly. |
| 05:01.14 | jedaustin | maint password in amp, where do I change it? |
| 05:01.18 | dontmsgme | In extensions? |
| 05:01.48 | CoderCR | ariel_: so at this point i am trying to find out why the channel bank is closing the audio channel and asterisk is not doing anything. Either the rhino channel is sending a message asterisk does not understand and asterisk is just leaving alone or the rhino channel bank is not sending anything at all because it does not handle the error properly. |
| 05:02.03 | CoderCR | ariel_: i need to find out to help debug the issue. |
| 05:02.13 | ariel_ | Blah-, best sound is ulaw it's 64k plus ethernet overhead so about 80k. GSM is 32k plus ethernet overhead. g729 is around 13 to 16k plus you got it ethernet overhead. |
| 05:02.17 | SuperBawlz | Yeah. |
| 05:02.20 | SuperBawlz | SJLabs |
| 05:02.21 | Beirdo | my firewall in colo GAKED today |
| 05:02.32 | CoderCR | so i need to know how to make zaptel debug the information it received over a T1 card |
| 05:02.32 | dontmsgme | Ive seen something like this before it seems to dial/connect/then hang up after 10 seconds with my verbosity set high it says "no rule for t timeout" |
| 05:02.33 | SuperBawlz | They have an NDA with our company for something. |
| 05:02.34 | Blah- | yikes, that much. |
| 05:02.39 | Blah- | ok, thanks! |
| 05:02.46 | SuperBawlz | I think its some of our compression or something. |
| 05:03.05 | CoderCR | ariel_: any ideas? |
| 05:03.09 | Beirdo | anyone a zaptel call progress expert? |
| 05:03.36 | ariel_ | CoderCR, unless I can see it and how it's setup no I don't. |
| 05:04.45 | CoderCR | ariel_; what do you need to see,. i ma happy to give you access to the machine if you want |
| 05:04.53 | CoderCR | this is going to help me out alot |
| 05:04.53 | dontmsgme | If you got the command Dial(IAX2/12345) in extensions |
| 05:04.58 | dontmsgme | where do you set the parameters for that? |
| 05:05.00 | dontmsgme | Iax.conf? |
| 05:07.24 | ariel_ | dontmsgme, yes both have to get setup extensions.conf and iax.conf like if you use exten => 100,1,Dial(iax2/DevcieName) in the iax.conf you have there your device [devicename] |
| 05:08.03 | dontmsgme | So what does in the iax portion under [devicename] |
| 05:08.34 | Blah- | ok, folks, this has been quite informative. Thanks a bunch, and good night :) |
| 05:08.52 | Apple | shepherd: didn't work :( |
| 05:09.59 | dontmsgme | I dont see any typos yet when I call my DID is auto connects/hangs for 10 seconds then disconnects |
| 05:09.59 | Apple | ok, my problem, I have a @ in my nufone passwd, problem is, I can't change it on nufone.net, and I also can't login because it tries to treat the rest of the password as a domain |
| 05:10.04 | Apple | so, how can I fix that? |
| 05:10.23 | Apple | I tried using \@ but it doesn't like it |
| 05:10.28 | Apple | same for /@ |
| 05:11.50 | jjg | is there anything wrong with writing a server for an agi app that sits in a loop and listens for packets on a port then executes commands based on that packet |
| 05:11.53 | Silik0n | fuck it |
| 05:11.58 | Silik0n | this 5300 can wait |
| 05:12.02 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
| 05:12.08 | *** join/#asterisk bobessutio (~bobessuti@c-67-180-96-152.client.comcast.net) |
| 05:12.36 | bobessutio | What is the best provider for telemarketing? |
| 05:12.43 | Sedorox | ..... |
| 05:12.53 | *** join/#asterisk PBXtech (~upirc@wirelessdata-167-248.mycingular.net) |
| 05:12.58 | bobessutio | Voip provider |
| 05:12.59 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 05:13.01 | AgiNamu | hey there |
| 05:13.04 | Qwell | eww |
| 05:13.07 | AgiNamu | I'm making an app for asterisk |
| 05:13.20 | AgiNamu | how do I tell it where to look for linked libraries |
| 05:13.33 | PBXtech | is a quad t1 and quad fxo to much for 1 pc? |
| 05:13.41 | AgiNamu | it's giving me an error that it cant' find the file, but its right in /usr/local/lib/ |
| 05:14.00 | syslod | PBXtech: Depends on what you are doing. |
| 05:14.00 | Apple | anybody? |
| 05:14.07 | justinnnn | ppls |
| 05:14.08 | justinnnn | with sccp |
| 05:14.12 | Apple | PBXtech: no |
| 05:14.17 | Apple | I need more bandwidth then that |
| 05:14.52 | PBXtech | usage volume is low, outbound calling is all |
| 05:15.14 | syslod | PBXtech: Ulaw? |
| 05:15.22 | PBXtech | getting echo thinking hardware |
| 05:15.26 | PBXtech | yea ulaw |
| 05:15.30 | Trepalium | Is it normal for zaptel to cause a kernel panic if the system is shut down without asterisk being stopped (seems like a bug to me)? |
| 05:15.32 | syslod | U'll be fine |
| 05:15.53 | PBXtech | all zap to zap connections |
| 05:16.03 | syslod | Trepalium: I've never seen that and I kill system with * running all the time. |
| 05:16.04 | AgiNamu | <PROTECTED> |
| 05:16.04 | AgiNamu | <PROTECTED> |
| 05:16.38 | syslod | PBXtech: I have many of those doing meet-me with 4 port cards and a 2.8 gig processor. No probs here. |
| 05:16.39 | PBXtech | slow bus will cause echo right? |
| 05:16.43 | Trepalium | For some reason my init scripts are not killing asterisk during shutdown properly, and when zaptel tries to unload, the kernel panics repeatedly. :-( |
| 05:16.50 | justinnnn | ppls with sccp, how do i setup my 7960 to have multiple lines |
| 05:17.05 | `Sauron | follow the sccp configuration examples |
| 05:17.07 | jmhunter | why even bother |
| 05:17.17 | jmhunter | sccp is awesome.. but just use sip |
| 05:17.22 | `Sauron | actually |
| 05:17.26 | AgiNamu | Ahh... asterisk is looking in /usr/lib instead of /usr/local/lib/ |
| 05:17.28 | `Sauron | I use chan_skinny |
| 05:17.28 | AgiNamu | how can I change that? |
| 05:17.33 | `Sauron | cuz it works better than chan_sccp |
| 05:17.35 | Apple | so, anybody have an idea how to fix it? |
| 05:17.44 | PBXtech | hmm why the echo prob? |
| 05:17.56 | jmhunter | ~seen damin |
| 05:17.57 | jbot | damin is currently on #asterisk (54m 46s). Has said a total of 7 messages. Is idling for 40m 9s |
| 05:17.57 | syslod | PBXtech: U have echo cancel on? |
| 05:18.04 | PBXtech | yea |
| 05:18.42 | jedaustin | When you do multiple lines on asterisk how does that work with voip; for example If you have a 4 line voip phone, an asterisk box, and voip connection with a single primary number |
| 05:18.50 | harryvv | anyone here use asterisk-stat-v2 |
| 05:18.55 | syslod | I've seen some echo but I can't relate it to load. It sometimes happens when just one call. |
| 05:19.06 | PBXtech | harryvv I do |
| 05:19.08 | PBXtech | love it |
| 05:19.34 | SuperBawlz | Thanks for you help. |
| 05:19.44 | SuperBawlz | I'll be back tomorrow I'm sure. |
| 05:19.47 | harryvv | pbx, this is my first time working with php do i just point the brouser to one of those files? does not seem to work. php and the associated libs are installed. |
| 05:19.54 | *** part/#asterisk SuperBawlz (user@24.247.116.57.kzo.mi.chartermi.net) |
| 05:20.21 | syslod | harryvv: U'll need to tell httpd that php is to be used. |
| 05:20.29 | PBXtech | latency from zap card right? |
| 05:20.45 | harryvv | you mean it needs to be edited/added in httpd |
| 05:20.48 | `Sauron | Sigh. |
| 05:20.50 | syslod | PBXtech: Where u hear echo? |
| 05:20.50 | `Sauron | No msgs |
| 05:20.57 | PBXtech | is there much speed dif from v5 vs v3.3 cards? |
| 05:21.07 | jedaustin | Survey, best voip phone for the money? |
| 05:21.09 | syslod | harryvv: if its apache its in httpd.conf |
| 05:21.16 | harryvv | it is |
| 05:21.19 | syslod | PBXtech: I use 3.3v cards |
| 05:21.53 | syslod | jedaustin: Poly 500 |
| 05:22.00 | PBXtech | wonder if the v5 is the issue. have 2 other v3.3 and don't hear complaining |
| 05:22.21 | Shido6 | poly 500 are 190 |
| 05:22.24 | Shido6 | 195 i think |
| 05:22.25 | syslod | PBXtech: I have single port 5v and have no additonal echo |
| 05:22.52 | PBXtech | mine is v5 quad |
| 05:22.52 | PBXtech | hmm |
| 05:23.01 | PBXtech | I should change it out |
| 05:23.07 | PBXtech | for fun |
| 05:23.29 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 05:23.54 | PBXtech | damn my treo is almost dead. |
| 05:24.29 | jedaustin | syslod: that a speaker phone? |
| 05:25.06 | jedaustin | Are Grandstream phones any good? |
| 05:25.18 | nestAr | GHETTONE! |
| 05:25.23 | nestAr | :) |
| 05:25.35 | PTG123 | anyone here familiar with upgrading a cisco 7960 to the sip image |
| 05:25.43 | *** join/#asterisk santiago (~santiago@63.245.86.89) |
| 05:25.45 | spackle | jedaustin: not even for boat anchors ;-) |
| 05:25.57 | jedaustin | :) |
| 05:26.03 | syslod | jedaustin: Yea its speaker. I like grandstream too but not for a business phone. |
| 05:26.37 | spackle | jedaustin, Grandstreams are like VOIP training wheels. |
| 05:27.17 | *** part/#asterisk PBXtech (~upirc@wirelessdata-167-248.mycingular.net) |
| 05:27.50 | Essobi | Baah |
| 05:28.08 | jedaustin | So lets say I get a phone like the Sipura SPA-841 that is a 2 line phone. Can you configure it so the lines are extensions/etc? |
| 05:28.33 | Essobi | Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
| 05:28.34 | Essobi | :| |
| 05:29.14 | spackle | jedaustin: yes, you can do that. |
| 05:29.48 | jedaustin | If you have 10 extensions and 1 voip connection you can dial out on more than one extension as long as you have the bandwidth right? |
| 05:29.49 | spackle | Jedaustin, for the record, I'd take the Sipura over the Grandstream too. |
| 05:30.10 | dersteer | sipura works AWESOME here |
| 05:30.14 | dersteer | i'm impressed with it |
| 05:30.23 | dersteer | a tad loud.. but otherwise good |
| 05:30.30 | spackle | dersteer, have you used the speakerphone much? |
| 05:30.43 | AgiNamu | so anyone know how I can change how Asterisk looks for modules? |
| 05:30.45 | dersteer | no |
| 05:30.49 | AgiNamu | i.e., linked libraries |
| 05:30.53 | AgiNamu | to look in /usr/local/lib/ |
| 05:31.23 | spackle | dersteer, it has volume problems, fades up and down. |
| 05:31.43 | spackle | dersteer, I think it is a few firmware upgrades away from being really good. |
| 05:32.16 | spackle | dersteer, and the next model should be a contender. With POE, and built in switch. |
| 05:32.41 | Sedorox | what does Billed 6/6 mean? |
| 05:32.46 | dersteer | I've been impressed with it :) |
| 05:32.47 | harryvv | syslod yea i did a grep php httpd.conf and got only DirectoryIndex index.php index.html index.htm index.shtml |
| 05:33.02 | syslod | I think its AddType. |
| 05:33.05 | spackle | dersteer, did it take a while to get used to the keys? |
| 05:33.14 | syslod | Anyone have a rateengine? |
| 05:33.27 | harryvv | Addtype? okay will grep that btw..is there a way to make vi goto one particular string? |
| 05:33.58 | harryvv | syslod nope comes up with nothing. |
| 05:33.58 | jedaustin | How can I tell in asterisk if it's talking with the voip provider? |
| 05:34.22 | harryvv | sip debug sip show registry |
| 05:34.22 | spackle | jedaustin, sip or iax? |
| 05:34.28 | harryvv | if its sip |
| 05:34.30 | CoaxD | hail from dallas, TX. (I'm here to fix my aunt's computers.) |
| 05:34.37 | jedaustin | spackle: sip I believe (Broadvoice) |
| 05:34.40 | CoaxD | Well, technically i'm in irving. |
| 05:34.44 | spackle | sip show peers |
| 05:35.10 | hardwire | years after downloading the worst mpeg of it. ever.. |
| 05:35.15 | spackle | if there are calls in progress, sip show channels |
| 05:35.16 | harryvv | syslog brb |
| 05:35.17 | syslod | harryvv: U install php with apache? |
| 05:35.32 | harryvv | yea..php echos a responce on the console |
| 05:35.48 | PTG123 | ok can anyone tell me why my 7960G won't upgrade to the sip firmware? :) |
| 05:36.35 | syslod | harryvv: u do cgi binary or apache mod? |
| 05:37.10 | justinnnn | ptg.. |
| 05:37.11 | harryvv | Dont know what you mean by that. apache is installed. It installed with the server install. |
| 05:37.13 | justinnnn | try a diff tftp server |
| 05:37.19 | justinnnn | sometimes i have to use windows.. sometimes unix |
| 05:37.22 | justinnnn | i find.. |
| 05:37.23 | jedaustin | hmm.. sip isn't in the path |
| 05:37.35 | justinnnn | to upgrade from h323 to 6.3 i use windows (unix doesnt work for some reason...?) |
| 05:37.43 | justinnnn | then from 6.3 - 7.4 on the unix server |
| 05:38.08 | *** join/#asterisk KirkL (~me@c-24-22-57-111.client.comcast.net) |
| 05:38.19 | PTG123 | ~Seen moc |
| 05:38.20 | jbot | moc <~Moc@modemcable212.49-80-70.mc.videotron.ca> was last seen on IRC in channel #asterisk, 16h 17m ago, saying: 'vaewyn, that sound very cool'. |
| 05:38.33 | harryvv | syslod I am asuming you are bussy if so then could spend some time with my wife. |
| 05:38.44 | syslod | harryvv: u can compile php to run standalone or as a module. If addtype line didn't get added I'm guessing php didn't get installed as apache module. |
| 05:39.09 | harryvv | ohh php is installed it does echo a responce on the command line |
| 05:39.28 | syslod | Yea that sounds like standalone. |
| 05:39.32 | harryvv | actually it echos nothing but does not return to the command line. |
| 05:39.38 | syslod | You need to either do apache mod or do php as cgi. |
| 05:39.39 | harryvv | like its waiting |
| 05:39.59 | harryvv | you mean edit httpd.conf |
| 05:40.19 | *** join/#asterisk Nethab (~Anon3042@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 05:40.29 | syslod | Well you could do a standalone cgi but I recommend recompile with php as apache mod. |
| 05:40.41 | harryvv | have a wiki on this? |
| 05:41.00 | Nethab | yawn |
| 05:41.12 | *** part/#asterisk CoderCR (~creyna@adsl-67-112-135-29.dsl.sndg02.pacbell.net) |
| 05:41.13 | harryvv | yea its getting late |
| 05:41.14 | harryvv | :) |
| 05:41.16 | syslod | I think its still in README or INSTALL that comes with PHP. You can copy and paste the commands to build |
| 05:41.25 | harryvv | k |
| 05:41.54 | jedaustin | I'm unclear how to tell if asterisk@home is actually working as far as asterisk <-> Broadvoice Can't find the sip command |
| 05:41.55 | *** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz) |
| 05:42.04 | Shido6 | jeez |
| 05:42.07 | Shido6 | wjats up jedaustin ? |
| 05:42.14 | Shido6 | where'd that "j" come from |
| 05:42.21 | Shido6 | stupid eluminix keyboard |
| 05:42.27 | harryvv | syslog anyway thanks its enough for me to go on |
| 05:42.28 | Shido6 | im s upposed to "see" the keys |
| 05:42.33 | harryvv | im off thanks :) |
| 05:42.34 | Shido6 | but typing is just another skill alltogether |
| 05:42.48 | Delmar | anyone have any ideas why I am getting a console warning message saying.. unable to request echo training on channel 1 .. etc ? both incomming and outgoing via Zap. |
| 05:43.51 | jedaustin | Shido6: Im doing a fresh setup of asterisk/broadvoice and I'm a complete newbie. I followed the directions here http://foo.robotics.net/mediawiki-1.3.10/index.php/Talk:Asterisk_Setup but not sure how to tell if it actually works |
| 05:45.22 | `Sauron | make a call to/from it |
| 05:45.35 | jedaustin | Calling to it doesnt work, get the No answer message |
| 05:45.36 | jedaustin | :( |
| 05:45.39 | PTG123 | does anyone have old sip images for the 7960? |
| 05:46.23 | spackle | jedaustin: are you at the asterisk CLI? |
| 05:46.30 | hardwire | PTG123: whats wrong with new ones? |
| 05:46.31 | jedaustin | spackle: yes |
| 05:46.48 | spackle | what do you get from typing sip show peers |
| 05:46.59 | PTG123 | hardwire: i am reading something thats says you need to step through the versions to upgrade, which could be my problem |
| 05:47.06 | jedaustin | spackle: -bash: sip: command not found |
| 05:47.29 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 05:47.29 | PTG123 | hardwire: do you use the ciscos? |
| 05:47.29 | hardwire | I am about too |
| 05:47.29 | spackle | jedustin, try asterisk -vvvvvvr |
| 05:47.39 | spackle | now you are at the asterisk cli |
| 05:48.26 | jedaustin | spackle: Ah.. see Im an idiot :) |
| 05:48.28 | `Sauron | you forgot the -c |
| 05:48.37 | PTG123 | hardwire: that mean you have one? |
| 05:48.43 | `Sauron | -vvvvc |
| 05:48.57 | spackle | 'Sauron, I was guessing he already had it running. |
| 05:49.00 | jedaustin | what are all the v |
| 05:49.03 | jedaustin | v's for? |
| 05:49.11 | spackle | more troubleshooting info. |
| 05:49.12 | shepherd | different levels of verbosity :) |
| 05:49.24 | shepherd | the more v's the more verbrose |
| 05:49.32 | shepherd | only there is only 3 leves |
| 05:49.37 | shepherd | levels |
| 05:49.42 | shepherd | but 255 set |
| 05:50.13 | spackle | jedaustin, now try the sip show peers |
| 05:50.49 | *** join/#asterisk bjohnson_ (~bjohnson@66.11.188.184) |
| 05:51.33 | *** join/#asterisk tessier (~treed@203.113.174.33) |
| 05:51.42 | IQ | Hi, Anyone know default username/password for Asterisk Management Portal? |
| 05:52.14 | jedaustin | spackle: I think I need to build it again.. in the haste of playing around I screwed it up :) |
| 05:53.05 | spackle | jedaustin: it happens. |
| 05:53.57 | jedaustin | spackle: like dejavu when I first started playing with Linux in 93 or so :) |
| 05:54.35 | spackle | jedaustin: much less painful. No floppy disks! |
| 05:54.49 | jedaustin | spackle: or 24 hour ftp installs ;) |
| 05:54.57 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 05:55.03 | spackle | jedaustin: ugh. |
| 05:56.42 | jedaustin | Hunt groups.. is that where it dials several numbers simultaneously and the first to answer gets the call? |
| 05:57.10 | Essobi | Ain't that a motherfucker. |
| 05:57.15 | jedaustin | ? |
| 05:57.37 | Essobi | oh323 works perfectly fine, where chan_h323 acts like a retard. |
| 05:58.14 | Essobi | jedaustin 93? Pssh, late bloomet. |
| 05:58.17 | Essobi | bloomer, even. |
| 05:58.42 | spackle | Essobi, have you been pissed at h323 all day? |
| 05:58.48 | jedaustin | Late bloomer ?? |
| 05:59.22 | Essobi | spackle Hell yea. |
| 05:59.32 | jedaustin | Maybe.. most guys my age have given up on learning anything new if that's what you mean |
| 05:59.45 | Essobi | jedaustin Straight from 88 I was banging on unix boxes.. |
| 06:00.08 | Essobi | They were not exactly my boxes thou.. if you catch my drift. :) |
| 06:00.11 | jedaustin | Essobi: Oh.. I was playing with BSD a few years before that, not that I knew what i was doing :) |
| 06:00.34 | Essobi | Ahhh... the good ole days.. Tymnet, and telenet... |
| 06:00.52 | jedaustin | Essobi: yeah, I have a few friends that were into the same thing. I used to play around on telenet a bit |
| 06:01.04 | Essobi | >:) |
| 06:01.35 | Essobi | and the old school cocot tricks.. and pbx muckery. |
| 06:01.39 | jedaustin | Essobi: I still remember one of the places I used to go to : 0302057400022 (McGill University) |
| 06:01.50 | Essobi | Imagine I get payed to do that now.. |
| 06:01.50 | jedaustin | Essobi: I was into phreaking back then too |
| 06:01.52 | Essobi | Mehe. |
| 06:02.01 | Essobi | I remeber 03020 well.. |
| 06:02.03 | Shido6 | i still have my redbox around here somewhere |
| 06:02.12 | spackle | here we go. . . |
| 06:02.17 | Essobi | ;) |
| 06:02.22 | jedaustin | I have one in my garage.. they worked into the early 90's |
| 06:02.27 | jedaustin | :) |
| 06:02.39 | Essobi | some you can still drop a nikel and blow tones on |
| 06:02.43 | Shido6 | I used to go to sleep to "quarter tones" ( nickel tones) |
| 06:02.47 | Essobi | if you catch a town with old phones still |
| 06:02.58 | jedaustin | We used to social engineer the ani number back before caller id came out |
| 06:03.01 | Shido6 | now I just dial my 8xx number to get to my ast box |
| 06:03.10 | Shido6 | press a button to record a tv show |
| 06:03.15 | Shido6 | press another button to flush the toilet |
| 06:03.21 | Essobi | I had some friends there were really into grey boxing, and conf calls.. and vm/pbx hacking.. |
| 06:03.46 | Shido6 | im not giving up anymore phreaking info :) |
| 06:03.50 | Shido6 | it was fun |
| 06:04.06 | jedaustin | Essobi: yeah I built a few divertor boxes that used audio transformers and dtmf chips.. until I found out my friends were using them to rack up huge phone bills on innocent people |
| 06:04.09 | Shido6 | hah |
| 06:04.15 | Shido6 | all that stuff died on a zip drive |
| 06:04.16 | Essobi | maha |
| 06:04.22 | Essobi | I may a gold box or two |
| 06:04.23 | Shido6 | from clicks of death |
| 06:04.33 | Essobi | tick-tick-tick-tick |
| 06:04.37 | Essobi | made |
| 06:05.01 | *** join/#asterisk KirkL (~me@c-24-22-57-111.client.comcast.net) |
| 06:05.03 | PTG123 | anyone here with a 7960 or 7940? |
| 06:05.05 | jedaustin | I still use my 'beige' box.. to test phone lines |
| 06:05.06 | Essobi | I used to just bust up on the cocots to make phone calls.. |
| 06:05.09 | KirkL | guess someone didsn't like my question :-) |
| 06:05.15 | Essobi | PTG several 1000.. |
| 06:05.31 | PTG123 | Essobi: any idea why mine won't fetch the new image from tftp and upgrade |
| 06:05.59 | Essobi | umm.. busted tftp? files? dhcp? |
| 06:06.00 | Essobi | :) |
| 06:06.08 | Essobi | OS79XX? |
| 06:06.14 | Essobi | Did you read the cisco docs? |
| 06:06.15 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 06:06.22 | Essobi | BORK BORK BORK BORK! |
| 06:06.30 | PTG123 | Essobi: yes my file has the version number in it, yet it doesn't try and fetch the image |
| 06:06.41 | PTG123 | Essobi: you by chance have some old sip images, that may be my problem according to one thing ir ead |
| 06:06.45 | Essobi | MM... whip out a sniffer and see what it's doing. |
| 06:06.45 | MuppetMaster | B) |
| 06:06.50 | PTG123 | Essobi: i need to upgrade to 4, then 5 then 6 tghen 7 |
| 06:06.51 | t3t | PTG123: the ciscos are all but impossible to update |
| 06:07.00 | Essobi | 4-5-6-7.. correct |
| 06:07.04 | PTG123 | Essobi: its not requesting the file |
| 06:07.10 | PTG123 | Essobi: do you have 4, 5, 6 sip image? |
| 06:07.16 | Essobi | Ehh.. you got something fuxored.. |
| 06:07.18 | t3t | PTG123: some of the earlier revisions don't like tftp files that are over 8.3 characters |
| 06:07.20 | *** join/#asterisk DyOS (~me@ip68-2-145-171.ph.ph.cox.net) |
| 06:07.25 | Essobi | true that |
| 06:07.30 | PTG123 | t3t: its not |
| 06:07.30 | jedaustin | So once I get my asterisk box setup.. can I use a soft phone to dial from my # from across the net? |
| 06:07.34 | Essobi | did you check case sensitivity? |
| 06:07.35 | PTG123 | Essobi: what do you mean? |
| 06:07.44 | Essobi | upper and lower |
| 06:07.48 | PTG123 | Essobi: its not trying to fetch anything after it fetches the .txt file except conf files |
| 06:07.53 | PTG123 | so it must not think it needs to upgrade |
| 06:08.01 | Essobi | did you put the version in the .cnf? |
| 06:08.11 | t3t | PTG123: is it sccp or sip now? |
| 06:08.11 | PTG123 | No do i need to? |
| 06:08.11 | Essobi | that matched the os79XX? |
| 06:08.14 | Essobi | yes |
| 06:08.16 | spackle | jedaustin: probably. |
| 06:08.17 | PTG123 | t3t: sccp |
| 06:08.21 | Essobi | otherwise it won't try to read it |
| 06:08.22 | t3t | ooohhh |
| 06:08.27 | PTG123 | Essobi: do you have a sample cnf? |
| 06:08.32 | Essobi | one |
| 06:08.39 | PTG123 | i can't find a SEP conf |
| 06:08.40 | t3t | I think sccp looks for an xml file for the config which contains an update file |
| 06:08.42 | PTG123 | can i rename a SIP conf? |
| 06:08.52 | t3t | there's a file with the list of valid images... |
| 06:08.53 | PTG123 | t3t: it looks fore xml too, don't have any examples of that i can find either |
| 06:09.01 | t3t | ...then there's the file that tells it which image to use |
| 06:09.09 | PTG123 | t3t: any idea where to find these files? |
| 06:09.09 | t3t | just a sec... |
| 06:09.42 | jedaustin | Is it fairly trivial to set it up so you can dial in and then dial out from your number. Im thinking of changing my 'home' number on my cell phone and using my * box to make free calls |
| 06:10.35 | spackle | jedaustin: yes. However it is apparently not trivial to call asterisk, hang up, and have it call you back. grrrr |
| 06:11.08 | Essobi | PTG123 do you got a sample SIPDefault.cnf? |
| 06:11.17 | Essobi | I'll just give you the line if so.. |
| 06:11.22 | PTG123 | Essobi: no |
| 06:11.25 | PTG123 | Essobi: yah give me the one line |
| 06:11.39 | jedaustin | spackle: cron maybe? |
| 06:12.04 | Essobi | there's docs telling exactly how to do this on cisco's site..it's what I used. |
| 06:12.17 | *** join/#asterisk clive- (~pirch@rndf-146-22-79.telkomadsl.co.za) |
| 06:12.20 | PTG123 | Essobi: couldn't find it, but whats the one line? |
| 06:12.23 | Shido6 | whats up |
| 06:12.25 | jedaustin | night all.. I'll be back.. tons to learn |
| 06:12.25 | Shido6 | whats wrong |
| 06:12.29 | Essobi | It's in a pm |
| 06:13.06 | Shido6 | dialplan.xml |
| 06:13.17 | Shido6 | SIPDefault.cnf |
| 06:13.22 | Shido6 | SIP<macaddress>.cnf |
| 06:13.25 | t3t | PTG123: yuck.. my conf directory for cisco updates is a mess. Versions everywhere. I think that I had to update to 06-3-00 first, then to 7 |
| 06:13.35 | Shido6 | OS79XX.TXT |
| 06:13.45 | Shido6 | RINGLIST.DAT |
| 06:13.54 | brc_ | »»»» |
| 06:14.04 | Essobi | I never setup a dialplan.xml |
| 06:14.06 | Shido6 | and your firmware files |
| 06:14.13 | t3t | make sure that SIPDefault.xml is under a few Kb or it'll fail |
| 06:15.58 | t3t | I have a question for anyone willing to wrap their brain around it... |
| 06:16.08 | t3t | Does anyone have any pointers for doing inline voicemail integration with an existing PBX... capturing the DTMF digits immediately after connect and just after disconnect has me stumped. |
| 06:16.13 | Shido6 | lay out your dirty voip laundry |
| 06:16.18 | Shido6 | bleh |
| 06:16.24 | Shido6 | what does your legacy pbx talk? |
| 06:16.27 | spackle | night all. |
| 06:16.27 | AgiNamu | Hey, can anyone here recommend a good SOAP client that's suitable to use from inside Asterisk? |
| 06:16.33 | PTG123 | t3t: you have those xml files? |
| 06:16.36 | t3t | Shido6: It speaks crap |
| 06:16.37 | Shido6 | fxo, fxs, t1, e1, mr2 , |
| 06:16.40 | Shido6 | sip |
| 06:16.42 | Shido6 | crap? |
| 06:16.44 | Shido6 | ok |
| 06:16.49 | Shido6 | if we can listen to the crap |
| 06:16.59 | Shido6 | and decode it we can deal with it |
| 06:17.02 | Shido6 | and code around it |
| 06:17.09 | Shido6 | i think its called a wrapper |
| 06:17.17 | Shido6 | what kind of interface you have on this thing? |
| 06:17.24 | t3t | Shido6: fxs ports sending DTMF before and after the station connection ABC1234567890*# |
| 06:17.30 | Shido6 | not a problem |
| 06:17.32 | Shido6 | how many |
| 06:17.33 | Shido6 | ? |
| 06:18.04 | Shido6 | how many fxs ports? |
| 06:18.04 | t3t | PTG123: they aren't that interesting... I'll post the one with the files for ya |
| 06:18.12 | t3t | Shido6: 8 ports |
| 06:18.17 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:18.20 | t3t | for VM |
| 06:18.54 | *** join/#asterisk juan_pablo (~no@69.39.14.142) |
| 06:19.05 | juan_pablo | wadup all |
| 06:19.29 | Essobi | my bloodpressure |
| 06:19.30 | Essobi | :) |
| 06:19.38 | juan_pablo | lots of people, little typing..? |
| 06:20.33 | juan_pablo | soooo.... whats going on tonight |
| 06:20.54 | dontmsgme | When I make a call it passes it along and says "releasin" when it gets to that part it dsiconnects me! |
| 06:21.01 | *** part/#asterisk KirkL (~me@c-24-22-57-111.client.comcast.net) |
| 06:21.20 | *** join/#asterisk KirkL (~me@c-24-22-57-111.client.comcast.net) |
| 06:21.24 | juan_pablo | anyone now hot to perform a reverse lookup on a broadvoice customer? |
| 06:21.33 | juan_pablo | that would be "how" of course |
| 06:21.49 | juan_pablo | or if BV shares their directory |
| 06:22.05 | implicit | social engineering? |
| 06:22.06 | implicit | :-P |
| 06:22.19 | juan_pablo | yeah, and crisp $100's... |
| 06:22.37 | juan_pablo | what i need as a littl more privacy than BV offers, as they require a credit card for payment... |
| 06:22.45 | dontmsgme | Why is my call disconnecting whenever it gets to "releasing so and so" |
| 06:23.11 | implicit | juan_pablo, do you mind paying a little more? |
| 06:23.19 | juan_pablo | is there pehaps a "list" of ppl providing gateway services in recip.. im willing toi share |
| 06:23.19 | implicit | juan_pablo, for more privacy? |
| 06:23.23 | juan_pablo | of course... |
| 06:23.34 | juan_pablo | i realyt a need a list of proxy's im willing to set some up |
| 06:23.34 | implicit | msg me then |
| 06:26.18 | dontmsgme | Has anyone ever seen that happening? |
| 06:26.59 | t3t | PTG123: http://www.pastebin.com/257607 |
| 06:28.09 | t3t | PTG123: that's XMLDefault.cnf.xml |
| 06:28.38 | t3t | PTG123: I guess I went to P003-07-3-00 right away |
| 06:30.40 | dontmsgme | SOMEBODY HELP ME |
| 06:32.33 | PTG123 | i did it |
| 06:32.35 | PTG123 | i did it :) |
| 06:32.46 | PTG123 | you fucker |
| 06:32.50 | PTG123 | you pasted me what i needed |
| 06:32.54 | PTG123 | right after i figured it out |
| 06:32.55 | PTG123 | :) |
| 06:33.08 | PTG123 | ProtocolApplicationInvalid |
| 06:33.10 | PTG123 | is what i get now |
| 06:33.11 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 06:34.20 | PTG123 | oh now its saying verifying load |
| 06:34.22 | PTG123 | perhaps its working |
| 06:35.40 | PTG123 | t3t: you awake? |
| 06:36.41 | *** join/#asterisk SwedMiro (~swed@330a.licentiaten.umea.hsb.se) |
| 06:38.13 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 06:43.35 | t3t | PTG123: barely :) |
| 06:44.11 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:44.22 | *** join/#asterisk zippp (~zip@c66.190.110.64.ts46v-01.rckprt.tx.charter.com) |
| 06:44.37 | justinnnn | can anyone help me with my 7914 :) ? |
| 06:44.37 | t3t | PTG123: so did it work or what? |
| 06:44.40 | justinnnn | its kinda working now |
| 06:44.55 | t3t | justinnnn: sure. just toss it out a window greater than 10 stories up |
| 06:45.10 | t3t | that should take care of any problem you were having |
| 06:45.11 | justinnnn | ive made like 10 users.. |
| 06:45.21 | justinnnn | one is on the 7960.. one is on the 7914 |
| 06:45.21 | PTG123 | t3t: no |
| 06:45.23 | justinnnn | its kinda working.. |
| 06:45.25 | PTG123 | t3t: now new problems |
| 06:45.28 | justinnnn | just wont add more the 7914 |
| 06:45.31 | t3t | PTG123: what's that? |
| 06:46.23 | t3t | justinnnn: what do you mean it won't add it? what's the error? |
| 06:46.28 | justinnnn | nah like |
| 06:46.33 | justinnnn | i dunno.. |
| 06:46.35 | PTG123 | t3t: its looping fetching config files |
| 06:46.38 | justinnnn | i cant get it to have 2 ppl on the 7914 |
| 06:46.41 | justinnnn | it will come up with 1... |
| 06:46.43 | justinnnn | but no more.. |
| 06:46.46 | *** join/#asterisk bsdfreak (ninja@enterthebass.com) |
| 06:46.55 | t3t | typical cisco... the loader thing is stuck... |
| 06:47.21 | t3t | PTG123: sometimes if you power off the phone for a few secs it will load it properly |
| 06:47.48 | t3t | PTG123: otherwise the three-finger salute on the phone might make it happen |
| 06:48.37 | t3t | PTG123: is it cycling between detecting network settings, fetching the image, and reloading itself? |
| 06:48.50 | PTG123 | t3t: |
| 06:48.57 | PTG123 | t3t: found a hint about this, trying it now |
| 06:49.03 | PTG123 | http://www.voip-info.org/tiki-index.php?page=Firmware%20issues%20on%207940%20-%207960 |
| 06:50.13 | PTG123 | this is like a nightmare |
| 06:50.16 | PTG123 | i thought my polycom was tough |
| 06:50.58 | t3t | no kidding.. it was enough to make me standardize on polycom instead of cisco |
| 06:51.02 | PTG123 | now it says |
| 06:51.06 | PTG123 | Upgrading Application |
| 06:51.10 | t3t | That's good |
| 06:51.18 | PTG123 | Oh Configuring VLAN |
| 06:51.21 | PTG123 | we can only hope and pray |
| 06:51.21 | PTG123 | :) |
| 06:51.29 | PTG123 | Requesting config |
| 06:51.36 | PTG123 | YAYAYA |
| 06:51.40 | PTG123 | its requesting sip configs now |
| 06:51.41 | PTG123 | :) |
| 06:51.44 | t3t | excellent |
| 06:51.52 | PTG123 | ok |
| 06:51.53 | PTG123 | now |
| 06:51.56 | PTG123 | i got to find these things |
| 06:52.26 | t3t | so what ended up being the thing that did it? |
| 06:52.38 | PTG123 | um |
| 06:52.40 | *** join/#asterisk FryGuy- (fryguy@c-24-23-19-33.client.comcast.net) |
| 06:52.43 | PTG123 | that xml file in the url i pasted you |
| 06:52.47 | PTG123 | i think that was the main thing that fixed it |
| 06:53.18 | t3t | oh... I had one of those in my config dir, but it didn't have anything in it that seemed relevant |
| 06:53.25 | PTG123 | yah i saw it too |
| 06:53.26 | t3t | the contents were different from the URL |
| 06:53.29 | PTG123 | but didn't look like anything special |
| 06:53.35 | PTG123 | do you have a sipdefault.cnf |
| 06:53.36 | PTG123 | ? |
| 06:53.43 | t3t | yep |
| 06:53.45 | PTG123 | and the one with the mac address |
| 06:53.48 | PTG123 | can you paste them to pastebin? |
| 06:53.50 | t3t | yep |
| 06:53.58 | t3t | one for each phone |
| 06:54.14 | t3t | sure, then I've got to get to sleep... |
| 06:54.46 | PTG123 | ok cool |
| 06:54.51 | PTG123 | well that should be all i need now right? :) |
| 06:55.08 | PTG123 | do these things have a backlight? |
| 06:55.35 | *** join/#asterisk shaZwaz (~Adnan@203.81.196.167) |
| 06:56.20 | shaZwaz | hi all |
| 06:56.21 | bobessutio | are there any voip providers that permit autodialers? |
| 06:56.43 | PTG123 | bobessutio: probably most, how many minuteds bobe? |
| 06:57.18 | bobessutio | 60000 a month |
| 06:57.35 | PTG123 | bobessutio: pm me |
| 06:57.51 | t3t | http://www.pastebin.com/257609 |
| 06:57.56 | t3t | that's the one with the mac |
| 06:59.09 | PTG123 | why the php tags? |
| 07:00.51 | t3t | this is the default |
| 07:00.52 | t3t | http://www.pastebin.com/257610 |
| 07:01.17 | PTG123 | ok |
| 07:01.19 | hardwire | PTG123: for highlighting |
| 07:01.19 | t3t | php? oh, cause pastebin is broken. |
| 07:01.20 | PTG123 | do i remove the php part? |
| 07:01.22 | PTG123 | k |
| 07:02.02 | t3t | oh, it's actually php pastebin... strange. |
| 07:02.15 | *** join/#asterisk `Kirk (~me@c-24-22-57-111.client.comcast.net) |
| 07:02.15 | PTG123 | what do you mean |
| 07:02.31 | t3t | I used pastebin.com which is a php-centric bin |
| 07:02.38 | t3t | pastebin.ca is the agnostic one |
| 07:02.51 | t3t | anyway, good luck and good night |
| 07:03.06 | hardwire | how often can the logo url change? |
| 07:03.13 | hardwire | or can you constantly publish one to the phones? |
| 07:03.19 | PTG123 | thanks for your help |
| 07:03.19 | PTG123 | cya |
| 07:03.23 | t3t | sure thing |
| 07:03.26 | hardwire | I would like to have a screen of a sort for showing network graphs |
| 07:03.32 | t3t | hardwire: once per boot I think |
| 07:03.39 | hardwire | ah |
| 07:03.52 | t3t | hardwire: there is a crappy xml browser built into the phone though |
| 07:03.58 | justinnnn | damn this crap |
| 07:04.00 | hardwire | thats what I have heard. |
| 07:04.06 | t3t | 'night. now I'm really leaving |
| 07:05.22 | *** part/#asterisk justinnnn (~dsf@solid.mpa.net.au) |
| 07:05.24 | *** join/#asterisk justinnnn (~dsf@solid.mpa.net.au) |
| 07:05.30 | justinnnn | why is my stupid 7914 only coming up with 1 line |
| 07:05.36 | hardwire | indeed |
| 07:05.43 | hardwire | its stupid. |
| 07:05.48 | shaZwaz | hi justinnnn |
| 07:05.58 | justinnnn | its half working :) /me is so close |
| 07:06.11 | justinnnn | its got 1 user on the 7960, 1 on the 7914.. |
| 07:06.16 | justinnnn | but it should have another 5.. on the 7914 |
| 07:06.37 | hardwire | are you showing line status on it at all? |
| 07:07.18 | shaZwaz | justinnnn, anything new about ur outgoing fax thing ? |
| 07:07.24 | _Vile | yeah once per boot |
| 07:07.41 | hardwire | _Vile: any way to push more images at it.. using SendImage or anything? |
| 07:07.49 | _Vile | but you could probably do something with the services button |
| 07:08.18 | _Vile | I think it's a pull thing, not really a push thing |
| 07:08.23 | hardwire | doheth |
| 07:08.24 | *** join/#asterisk marc32344 (~marc32344@69-90-241-15.dsl.teksavvy.com) |
| 07:08.32 | _Vile | i'd go services |
| 07:08.34 | hardwire | starting to see the benifits of really complicated ADSI programming :) |
| 07:08.41 | hardwire | but oh well |
| 07:08.43 | _Vile | i'd avoid adsi |
| 07:08.46 | hardwire | I know |
| 07:08.49 | _Vile | check out polycom |
| 07:08.53 | hardwire | well |
| 07:09.05 | hardwire | we were going to go snom-360 and 190 |
| 07:09.07 | _Vile | I think that might be a pull only too |
| 07:09.08 | _Vile | hmm |
| 07:09.11 | hardwire | and then some duuuude |
| 07:09.15 | _Vile | but you can do animated there |
| 07:09.23 | hardwire | offered me a realyl cheap price on some 7960's |
| 07:09.24 | _Vile | i haven't seen animated on cisco |
| 07:09.32 | _Vile | well |
| 07:09.35 | _Vile | if you code it right |
| 07:09.35 | hardwire | from some bankrupt company |
| 07:09.39 | _Vile | you could hit the services button |
| 07:09.43 | _Vile | and get a new graph |
| 07:09.47 | _Vile | every time you hit it |
| 07:09.49 | hardwire | _Vile: yeh |
| 07:09.53 | hardwire | I bet there is an xml refresh |
| 07:09.59 | _Vile | could be, good thought |
| 07:10.12 | hardwire | who knows if it caches or not :) |
| 07:10.22 | SuperPhone | justinnnn : I have Cisco 7960 with all 6 lines working |
| 07:10.24 | _Vile | I doubt it, I think it's live pulls |
| 07:10.32 | hardwire | yeh |
| 07:10.38 | hardwire | not sure I like using cisco equipment |
| 07:10.38 | hardwire | heh |
| 07:10.38 | _Vile | not enough memory to cache I would suspect the programmers to code it that way |
| 07:10.41 | hardwire | its so.. odd.. |
| 07:10.53 | hardwire | but for the price we are getting for 35 of them |
| 07:10.58 | hardwire | its sorta hard to pass up |
| 07:10.58 | _Vile | $150? |
| 07:11.04 | hardwire | yup |
| 07:11.10 | _Vile | good price |
| 07:11.15 | hardwire | no books |
| 07:11.16 | hardwire | no box |
| 07:11.19 | _Vile | no need |
| 07:11.20 | _Vile | :) |
| 07:11.23 | hardwire | no kiddin |
| 07:11.32 | hardwire | just hard to resell if these are shit |
| 07:11.35 | hardwire | w/o that |
| 07:11.39 | _Vile | not really |
| 07:11.43 | hardwire | I would like to atleast resell for 180 a pop |
| 07:11.49 | _Vile | ebay is ~$200 |
| 07:11.55 | hardwire | I suppose |
| 07:11.55 | _Vile | more including the boxes etc |
| 07:12.01 | justinnnn | hmmm |
| 07:12.02 | hardwire | hmm |
| 07:12.18 | hardwire | I found a neat place on ebay today |
| 07:12.36 | hardwire | http://www.empc2.com/ |
| 07:12.45 | hardwire | I am going to buy some more toughbooks (I halready have a very current one) from them |
| 07:12.54 | hardwire | for some of our service men |
| 07:13.00 | _Vile | i'd check out xml refresh (if possible), services button for that functionality... and go with the 7960s, easy platform to work with w/ *.. |
| 07:13.14 | _Vile | and cheap it appears |
| 07:13.22 | _Vile | going to bed |
| 07:13.23 | _Vile | later |
| 07:13.27 | hardwire | I should too |
| 07:13.29 | hardwire | winter ale kicking in |
| 07:14.40 | Pumpkin_Pie | huh |
| 07:18.02 | *** join/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net) |
| 07:24.05 | jesster | zaptel.conf question - i have 2 TE410P cards, 1 receives clock from telco trunks, the 2nd card connects to channel banks. How should card two be setup to use clock from the first card? |
| 07:24.40 | Mavvie | jesster: I didn't know the answer neither, so I just plugged them one after the other. |
| 07:24.52 | Mavvie | telco - channel - telco - channel - telco - channel |
| 07:25.01 | *** join/#asterisk SkySky (~Miranda@69.156.127.119) |
| 07:25.09 | SkySky | wow.. asteriskwin32 is cool |
| 07:25.16 | SkySky | takes up very little ram |
| 07:25.17 | jesster | Mavvie: hrm ok |
| 07:25.36 | jesster | Mavvie: you understand what I'd like to do though? use clock from card1 for card2 |
| 07:25.46 | Mavvie | jesster: I understand your probelm. |
| 07:25.59 | Mavvie | but I couldn't find out the answer neither. |
| 07:26.07 | Mavvie | so I resolved it in the way of the least pain. |
| 07:26.30 | Mavvie | I however would put money in it that the clock is delivered to it via the driver. |
| 07:28.12 | drumkilla | asteriskwin32? |
| 07:28.20 | drumkilla | astwind? |
| 07:29.21 | kernel3 | http://www.asteriskwin32.com/ |
| 07:29.44 | Mavvie | I truely believe that some people are wasting money on domain names. |
| 07:29.48 | Mavvie | asteriskdocs.org |
| 07:29.51 | Mavvie | asteriskwin32.com |
| 07:31.45 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 07:32.15 | Mavvie | <PROTECTED> |
| 07:32.15 | Mavvie | <PROTECTED> |
| 07:33.38 | drumkilla | what should it be |
| 07:33.58 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 07:36.58 | Mavvie | I would go for the info tone. |
| 07:37.16 | *** join/#asterisk terrapen_ (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 07:39.51 | *** part/#asterisk terrapen_ (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 07:42.35 | *** join/#asterisk Tarox (someone@pD9E7A847.dip.t-dialin.net) |
| 07:44.42 | Mavvie | the congestion indication says "there is a problem with reaching it", while the info indication would say "there is something wrong with this number" |
| 07:48.26 | *** join/#asterisk pascals (~248d34d6@ip503c8584.speed.planet.nl) |
| 07:52.17 | pascals | I am looking for suggestions for hardphones for office use. |
| 07:53.02 | pascals | A speaker phone option would be nice. |
| 07:53.17 | modulus_ | polycom |
| 07:53.23 | modulus_ | ip500 |
| 07:53.39 | Mavvie | laptop with xlite. |
| 07:53.44 | Mavvie | hey, a latop is hard! |
| 07:53.46 | Mavvie | latop |
| 07:53.48 | Mavvie | laptop |
| 07:53.50 | Mavvie | geez. |
| 07:53.53 | Mavvie | that joke turned out bad. |
| 07:53.57 | pascals | :D |
| 07:54.16 | pascals | I never seem to get xlite working properly. |
| 07:54.50 | modulus_ | yeah xlite sucks |
| 07:54.58 | Mavvie | is sucks, but it looks nice. |
| 07:55.09 | modulus_ | after i finally got it to work figuring out its stupid interface it sounded like crap |
| 07:55.14 | pascals | Which softphone gives people the least problems on win32? |
| 07:55.27 | pascals | Looks nice?!? |
| 07:55.44 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 07:55.52 | pascals | xlite breaks every UI convention ever conceived |
| 07:56.11 | pascals | modulus_ what does the ip500 cost? |
| 07:56.19 | Mavvie | I bet you haven't seen Lotus Notes? |
| 07:56.23 | modulus_ | what do i look like? priceguide.com? |
| 07:56.43 | pascals | modulus_ no ;-) |
| 07:57.05 | pascals | But usually when someone suggests something, they bought one at some point in time ;-) |
| 07:57.13 | modulus_ | i've bought so much voip crap (shit too) that i forgot how much everything costs |
| 07:57.27 | modulus_ | and now i have all this stupid voip hardware and voip still sucks |
| 07:57.27 | pascals | Mavvie: yes, i've seen notes - and I don't really like it |
| 07:57.46 | pascals | except the ip500? |
| 07:57.53 | pascals | That doesn't suck? |
| 07:57.53 | Mavvie | anyway, you'll get used to it when you have bumped your nose for the umpthied time. |
| 07:58.40 | pascals | My first bump is not being able to use my hfc-4s card to connect to two isdn lines and then the pbx. |
| 07:58.43 | modulus_ | and back to that xlite thing, i'm a master of figuring out interfaces to _everything_ |
| 07:58.55 | modulus_ | but xlite is just braindead |
| 07:59.07 | pascals | modulus_ :) Yea, but my users likely are not. |
| 07:59.25 | modulus_ | i don't recommend xlite for you my friend |
| 07:59.36 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 07:59.56 | pascals | No, I know |
| 08:00.51 | modulus_ | i was extremely intrigued by the ip500 functionality/quality -wise |
| 08:00.52 | pascals | the ip500 looks professional |
| 08:01.20 | pascals | can the display be remotely controlled? |
| 08:02.04 | modulus_ | you can but logos on the screen |
| 08:02.37 | modulus_ | s/but/put |
| 08:02.57 | modulus_ | goddamnit i'm out of cigs |
| 08:03.14 | modulus_ | i have to go all the way to the garage and all the way to 7-11 |
| 08:03.15 | modulus_ | grrr |
| 08:03.21 | pascals | Actually - I need a central directory, and I'd like the ability to make the phone call a specific number |
| 08:04.41 | modulus_ | you can do that with asterisk my friend |
| 08:05.33 | pascals | I know I like asterisk, I just have to figure out what I want to do with it all :) |
| 08:06.07 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 08:06.13 | modulus_ | this is great |
| 08:06.21 | modulus_ | you're making me sound smart: S-M-R-T |
| 08:09.15 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 08:10.52 | *** join/#asterisk felipex (~dsfdsf@85.33.91.162) |
| 08:16.35 | *** join/#asterisk acek (~acek@p508708D1.dip0.t-ipconnect.de) |
| 08:18.31 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:24.10 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:27.08 | *** part/#asterisk dontmsgme (~none@207.215.252.80) |
| 08:30.46 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:38.24 | *** join/#asterisk mayco (~mayco@hideout.rega.khleuven.be) |
| 08:38.50 | mayco | i have a small question, i want to use festival to generate some speech in asterisk |
| 08:38.56 | mayco | but it doesn't work |
| 08:39.03 | mayco | i get this "error": |
| 08:39.11 | mayco | <PROTECTED> |
| 08:39.11 | mayco | <PROTECTED> |
| 08:39.11 | mayco | <PROTECTED> |
| 08:39.11 | mayco | <PROTECTED> |
| 08:39.12 | Zeeek | you mean like a few fixed files or dynamically? |
| 08:39.20 | mayco | dynamic |
| 08:39.29 | *** join/#asterisk afe ([fLx8Ssgp4@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 08:39.36 | mayco | but once i have it working, it may be fixed files |
| 08:39.42 | mayco | but for now, it just doesn't work |
| 08:40.04 | mayco | on my sip-client, it just disconnects after the error |
| 08:40.14 | Zeeek | I was just going to point out that unless the texts must be dynamic you can generate them on the demo sites free and not use CPU |
| 08:40.46 | mayco | hmm, ok, but then i have to regenerate them after i change the text |
| 08:40.52 | Zeeek | yup |
| 08:41.08 | Zeeek | if you already have festival installed there must be an easy way to do that |
| 08:42.07 | mayco | ok, i can generate the files, and play them with background, but i would rather have them be processed on the fly |
| 08:42.52 | mayco | exten => 1,1,Answer |
| 08:42.52 | mayco | exten => 1,2,Festival(Welcome) |
| 08:42.52 | mayco | exten => 1,3,Hangup |
| 08:42.58 | mayco | that is my extension |
| 08:43.02 | mayco | that's correct, not? |
| 08:44.49 | mayco | the festival server looks ok also: |
| 08:44.50 | mayco | client(6) Wed Mar 16 09:44:08 2005 : accepted from localhost |
| 08:44.50 | mayco | client(6) Wed Mar 16 09:44:09 2005 : disconnected |
| 08:48.28 | Zeeek | ironcally I got a phone call just then |
| 08:49.17 | Zeeek | I don't use festival myself so I can't really help. I have used another TTS but I do not generate on the fly |
| 08:51.06 | mayco | ah, ok |
| 08:52.24 | *** join/#asterisk MacDeath (david@196.22.239.13) |
| 08:52.28 | MacDeath | morning all |
| 08:53.01 | Zeeek | MacDeath hello - mayco, there is probably oodles of stuff on the wiki and the mailing list about festival |
| 08:53.08 | *** join/#asterisk isam (~isam@81.10.126.2) |
| 08:53.14 | MacDeath | hey Zeeek |
| 08:53.27 | MacDeath | soz, i got disconnected on monday when you were speaking to me |
| 08:53.40 | mayco | Zeeek, yes, there is, and that's what i did, but it doesn't work :) |
| 08:53.58 | *** join/#asterisk shaZwaz (~Adnan@203.81.196.167) |
| 08:54.23 | Zeeek | festival is variously appreciated in comments here, going from it sucks to it's ok ( but what isn't) |
| 08:54.51 | MacDeath | I have a question with regards to b/w |
| 08:54.58 | MacDeath | got a grandstream 102 |
| 08:55.01 | Zeeek | you can name a distro, a program, a softphone, a hardphone or a programming languge that isn't |
| 08:55.14 | MacDeath | and an asterisk box, with a 64kb line |
| 08:55.18 | Zeeek | MacDeath me too |
| 08:55.34 | MacDeath | what protocols should i be using |
| 08:55.48 | Zeeek | it depends a little on who you are connecting to I would say |
| 08:55.49 | MacDeath | it really breaks up badly |
| 08:56.04 | MacDeath | grandstream is on the other side of the 64kb line |
| 08:56.05 | Zeeek | FWD uses ulaw only but will pass anything |
| 08:56.19 | Zeeek | who are you calling/being called by mostly? |
| 08:56.27 | MacDeath | umm, let me rephrase, im bad at wording things |
| 08:56.33 | MacDeath | asterisk box with 1 phone |
| 08:56.39 | MacDeath | 64kb line with another phone |
| 08:56.41 | *** join/#asterisk Dibbler (~Dibbler@zidane.pi-net.net) |
| 08:57.03 | MacDeath | so phone 1 will be calling phone 2 |
| 08:57.06 | *** join/#asterisk widowlicker (~Naturalbl@62.77.178.121) |
| 08:57.17 | Zeeek | the BT102 ahs ilbc - some say it's good |
| 08:57.27 | Zeeek | has ilbc |
| 08:57.32 | Zeeek | iLBC? |
| 08:57.37 | Zeeek | whatever it is |
| 08:57.37 | shaZwaz | it also has 729 zeeek |
| 08:57.54 | Zeeek | yes but that won't help with no license from digium |
| 08:58.09 | MacDeath | i will buy license if thats what i need |
| 08:58.15 | shaZwaz | if both are BT then it does |
| 08:58.16 | Zeeek | MacDeath do the phones hook up directly after the initial session? |
| 08:58.38 | MacDeath | i dont have invite on, but I can change it? |
| 08:58.40 | Zeeek | or is there NAT |
| 08:58.44 | MacDeath | no nat |
| 08:58.59 | Zeeek | no NAT you might be able to try 729 |
| 08:59.08 | Zeeek | with canreinvite=yes |
| 08:59.17 | shaZwaz | if both of ur phones have G729 than I think u sh'd use it |
| 08:59.31 | MacDeath | failing that? |
| 08:59.37 | Zeeek | iLBC |
| 09:00.22 | MacDeath | and in sip.conf? |
| 09:00.58 | Zeeek | try the codecs one at a time |
| 09:01.04 | Zeeek | disallow=all |
| 09:01.11 | Zeeek | allow=ilbc |
| 09:01.17 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 09:01.17 | Zeeek | or whetever it is |
| 09:01.29 | Zeeek | and set both phones to do ilbc obviously |
| 09:01.36 | MacDeath | and would that be 729.1 or 729a/b? |
| 09:01.37 | Zeeek | it's a laborious process |
| 09:02.36 | shaZwaz | Zeeek BT has no iLBC |
| 09:03.02 | shaZwaz | instead there is G723.1 |
| 09:03.26 | MacDeath | it has ilbc in the list? |
| 09:03.35 | Zeeek | mine does |
| 09:03.38 | shaZwaz | not in mine |
| 09:03.44 | Zeeek | what firmware? |
| 09:03.58 | Zeeek | it came in last year |
| 09:04.01 | shaZwaz | 1.0.4.39 |
| 09:04.06 | Zeeek | errrrrr |
| 09:04.12 | Zeeek | that's so 1990's! |
| 09:04.34 | Zeeek | move up to 5.12 at least - the current beta is like .22 |
| 09:04.36 | *** join/#asterisk teemu-x (~tnurmine@tuomi.oulu.fi) |
| 09:05.11 | shaZwaz | I tried once but it doesn't update |
| 09:05.51 | Zeeek | shame |
| 09:06.25 | Leland | morning peeps |
| 09:06.59 | teemu-x | On SIP<->PSTN call, everything sounds fine on SIP end but PSTN ends hears lots of popping noise and other distortion... where should I start looking the problem? |
| 09:07.14 | *** join/#asterisk cgeek (~cgeek@pl6.lawrence.org.uk) |
| 09:07.20 | teemu-x | same asterisk does H323<->PSTN calls, and they sound fine on both ends |
| 09:08.03 | Leland | jluk: most people do |
| 09:08.05 | Leland | :P |
| 09:08.11 | teemu-x | and sip-sip calls sound fine too, so it's just sip->pstn |
| 09:08.11 | *** join/#asterisk olivier_ (~olivier_@82.127.99.32) |
| 09:08.50 | shaZwaz | teemu-x, u using echo can ? |
| 09:09.30 | teemu-x | I guess they are there, yes |
| 09:09.42 | teemu-x | no echo problems here |
| 09:09.55 | *** join/#asterisk markak2 (~twist@ndn-165-130-201.telkomadsl.co.za) |
| 09:10.02 | markak2 | hia ll |
| 09:10.10 | shaZwaz | teemu-x, which * ver |
| 09:10.29 | teemu-x | several, now 1.0.6 but the problem has persisted for months |
| 09:10.54 | shaZwaz | whats on SIP end har /soft phone ? |
| 09:10.55 | Leland | teemu-x: what FXO card are you using? (out of curiosity) |
| 09:10.56 | MacDeath | is there anyway in asterisk to see what protocol they are uising? |
| 09:11.08 | Zeeek | MacDeath sip show channels |
| 09:11.27 | Zeeek | oops |
| 09:11.31 | Zeeek | I read codec! |
| 09:11.35 | Zeeek | codec on the brain |
| 09:11.47 | Leland | I suspect that my echo problem might be impedence related to be honest. |
| 09:11.52 | markak2 | i have changed all my dial extension to include t and T which has sloved all the transfer problems. however on our board when no one answers it rings all extensions, but this does not allow a transfer. ie: current => exten => 0,1,Dial(iax2/2001&iax2/2002&iax2/2003&iax2/2006&iax2/2007) should this be exten => 0,1,Dial(iax2/2001&iax2/2002&iax2/2003&iax2/2006&iax2/2007,) |
| 09:12.04 | teemu-x | I think Wildcard TE410P |
| 09:12.09 | markak2 | sorry exten => 0,1,Dial(iax2/2001&iax2/2002&iax2/2003&iax2/2006&iax2/2007,tTm) or exten => 0,1,Dial(iax2/2001&iax2/2002&iax2/2003&iax2/2006&iax2/2007,20,Ttm) |
| 09:12.26 | Leland | teemu-x: you in the US or elsewhere ? |
| 09:12.32 | *** join/#asterisk cjk (~cjk@80.92.75.100) |
| 09:12.34 | teemu-x | finland, europe |
| 09:12.42 | shaZwaz | and u are using G711 I suppose ? |
| 09:12.48 | Leland | I think the TE4xx have adjustable impedence settings for the PSTN |
| 09:13.20 | teemu-x | I've used G729 mainly and G711 and some others too, they all have the same noise |
| 09:13.41 | shaZwaz | u didn't tell what are u using on the SIP side ? |
| 09:13.51 | *** join/#asterisk pascals (~248d34d6@ip503c8584.speed.planet.nl) |
| 09:14.17 | MacDeath | Zeeek |
| 09:14.23 | MacDeath | yeah, you read right |
| 09:14.27 | MacDeath | i wrote wrong |
| 09:14.28 | MacDeath | lol |
| 09:14.31 | Zeeek | heh |
| 09:14.31 | teemu-x | on sip side G729 now, if you asked me |
| 09:14.41 | *** join/#asterisk wbzz (~3ed4e70d@grosha.majordomo.ru) |
| 09:14.48 | shaZwaz | no I meant a soft /hard phone ? |
| 09:15.31 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 09:15.36 | teemu-x | right now I have here Snom190 hardware phone with G729 and some Sipura hw phone with G729 |
| 09:16.04 | shaZwaz | are u using them on the LAN ? |
| 09:16.16 | MacDeath | Zeeek |
| 09:16.18 | teemu-x | I've used several (~10) sip phones and they all sound alike (well, sipura spa-841 sounds shit but other than that) |
| 09:16.19 | MacDeath | im getting |
| 09:16.23 | MacDeath | Mar 16 11:15:47 WARNING[2046]: chan_sip.c:2776 process_sdp: No compatible codecs! |
| 09:17.08 | teemu-x | all of these in good intranet lan and next to me is connection to pstn via that wildcard |
| 09:17.47 | Zeeek | MacDeath what? |
| 09:17.57 | MacDeath | i set the phones to ilbc |
| 09:18.02 | MacDeath | and allow=ilbc |
| 09:18.07 | pascals | Anyone know why iaxclient mutes my logitech usb headset when it starts up? |
| 09:18.09 | teemu-x | it must be configuration issue somewhere that causes this, but I'm more of businees application developer and dont know where to start lookig :) |
| 09:18.20 | MacDeath | set canreinvite=yes |
| 09:18.28 | MacDeath | and when i try call the other phone |
| 09:18.35 | pascals | using iaxcomm now, but all clients seem using iaxclient seem to do that |
| 09:18.38 | MacDeath | chan_sip.c:2776 process_sdp: No compatible codecs! |
| 09:19.19 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:19.21 | shaZwaz | well codec is of most importance on the sip side |
| 09:19.30 | *** join/#asterisk mad_router (~ornate@84.246.176.66) |
| 09:19.37 | shaZwaz | but noise is something else |
| 09:19.53 | MacDeath | shaZwaz : all im am looking for now is a more efficeient codec to use |
| 09:20.05 | MacDeath | cause i only have 64kb of b/w |
| 09:20.13 | MacDeath | and im getting a very bad breakup |
| 09:20.19 | Zeeek | MacDeath are you at the asterisk box location with a console? (not ssh) |
| 09:20.27 | MacDeath | nope :( |
| 09:20.37 | shaZwaz | teemu-x, u using T1s ? |
| 09:20.47 | teemu-x | E1 |
| 09:20.52 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 09:21.47 | Zeeek | If you were at the console you could monitor the rates of various codecs. I tried that once |
| 09:21.48 | r0d3nt|m | what is the name of the dummy zaptel driver if you don't have any digium hardware installed... ? |
| 09:22.00 | shaZwaz | teemu-x, can u paste u zapata.conf |
| 09:22.24 | p1tst0p | hi, is it possible with Asterisk@Home, to run a server in one location, and have my friends coneect thru the internet with Sip phones / softphones |
| 09:22.49 | shaZwaz | r0d3nt|m, its ztdummy |
| 09:23.11 | teemu-x | but if all combinations of (sip,h323,pstn) <--asterisk--> (sip,h323,pstn) sound fine, except sip->pstn, shouldnt it be easy to track and narrow down where the problem is? :/ |
| 09:23.20 | teemu-x | I'll put the zapata.conf to pastebin, sec.. |
| 09:23.50 | r0d3nt|m | shaZwaz, thank you sir |
| 09:24.27 | teemu-x | http://pastebin.ca/7539 |
| 09:25.18 | MacDeath | Zeeek : do i have to compile anything again for ilbc |
| 09:25.29 | MacDeath | i cant call at all now |
| 09:26.23 | Zeeek | not that I know of |
| 09:26.41 | Zeeek | I've used ilbc it's not bad |
| 09:26.50 | Zeeek | you have a recent version of * ? |
| 09:26.59 | MacDeath | december 31 |
| 09:27.11 | cjk | hi is there a var in * defining the date? |
| 09:27.11 | MacDeath | everything I do i get |
| 09:27.15 | shaZwaz | teemu-x, strange gains values |
| 09:27.19 | MacDeath | process_sdp: No compatible codecs! |
| 09:27.24 | MacDeath | handle_request: Unable to create/find channel |
| 09:27.40 | shaZwaz | why are u keeping them so low ? |
| 09:28.43 | teemu-x | I cant really answer that since that conf is written by cow-orker, but it was changed recently and voice quality improved a bit |
| 09:29.47 | shaZwaz | teemu-x, u are not using a sip jitter buffer are u ? |
| 09:30.11 | teemu-x | hmm.. I think I am, why? |
| 09:30.28 | shaZwaz | disable it |
| 09:30.56 | shaZwaz | its not final yet, still needs a lot of work |
| 09:31.16 | p1tst0p | hi peeps, im using Asterisk@Home, i have my friend connected from his house with X-Lite to my box, problem is he can only hear me, i can t hear him, hmm, any ideas? |
| 09:32.00 | shaZwaz | may be he has a bad mic :) |
| 09:32.13 | sambal | :D |
| 09:32.14 | teemu-x | shaz: by putting jitterbuffers=0 to zapata.conf? |
| 09:32.33 | shaZwaz | no not there |
| 09:32.51 | p1tst0p | shaZwaz, nah man, he uses it for other voip services, i cant hear him, he can hear me. |
| 09:32.59 | shaZwaz | it must be in ur sip.conf |
| 09:33.13 | shaZwaz | p1tst0p, what codec ? |
| 09:33.26 | shaZwaz | and u are not behind nat ? |
| 09:33.36 | cjk | p1tst0p, you sure its a codec problem? maybe a nat problem. set canreinvite to no helps quite often |
| 09:33.40 | p1tst0p | erm, were both using X-Lite, and its selected G711u, |
| 09:34.09 | p1tst0p | canreinvite is set to no on both enxtentions cjk dude. |
| 09:34.34 | cjk | p1tst0p, for me it sounds like a nat problem |
| 09:34.40 | shaZwaz | check the IP and ports |
| 09:35.09 | shaZwaz | Xlite sh'd find the nat if there is one |
| 09:35.30 | cjk | i tried cisco, zyxel, smc, linksys, dlink, 9box .... portforwarding was never the solution |
| 09:35.35 | p1tst0p | shaZwaz, i forwarded, about 8 ports i think to my asterisk box. |
| 09:35.47 | shaZwaz | and on what IP are u binding SIP |
| 09:35.49 | teemu-x | shaz: thanks, I'll try to alter the jitter settings |
| 09:36.13 | p1tst0p | 0.0.0.0 i think, as in any on the box is that correct ? |
| 09:36.35 | p1tst0p | there is only one nic in the box thats it |
| 09:36.59 | p1tst0p | he can hear me perfect, which is good cuz i can abuse him without him responding back ! ahah |
| 09:37.41 | shaZwaz | but he is connecting to ur box from outside on some IP |
| 09:37.43 | Alexis | ok i have a snom220 and i want it to use its led for indicate the state of other phones |
| 09:38.03 | p1tst0p | shaZwaz, yep |
| 09:38.11 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:38.14 | Alexis | someone know the solution ? |
| 09:38.16 | p1tst0p | shaZwaz, hes conneting on my hostname |
| 09:38.18 | Leland | this is weird... |
| 09:38.25 | *** join/#asterisk dubphil (~corinne@dyn-83-156-81-130.ppp.tiscali.fr) |
| 09:38.44 | Leland | I just deleted and re-created the PSTN trunk in Asterisk .. exactly the same config that I previously had, and now my echo problem is gone |
| 09:38.48 | Leland | weeeeeeeeeird :S |
| 09:39.30 | Alexis | because on the snom man they say use the Line field and on * they say use the Destination field !!! |
| 09:40.10 | shaZwaz | p1tst0p, is ur friend behind nat ? |
| 09:40.46 | shaZwaz | may be a proxy |
| 09:41.04 | cjk | p1tst0p, you friend should disbale somehting like "udp flood attack" on this router,nat,firewall |
| 09:41.15 | cjk | maybe disablen the windwos firewall or similar |
| 09:41.50 | shaZwaz | or may be he has a personal firewall or something |
| 09:42.11 | shaZwaz | which is restricting the outgoing UDP /RTP |
| 09:43.22 | MacDeath | Zeeek |
| 09:43.23 | p1tst0p | shaZwaz, he uses sipgate.co.uk ok, he has loads of ports forwarded too. i noticed this in my logs, after he called me, Ooh, format changed from unknown to ulaw, |
| 09:43.28 | MacDeath | doesnt matter what I do |
| 09:43.36 | MacDeath | I get no compatible codecs |
| 09:43.52 | p1tst0p | shaZwaz, and WARNING[1591]: Maximum retries exceeded on call |
| 09:44.11 | p1tst0p | shaZwaz, and RFC3389 support incomplete. Turn off on client if possible |
| 09:44.21 | p1tst0p | thats all i can see on the log around the call time at the moment |
| 09:44.54 | shaZwaz | the RFC thing is becoz of the Transmit Silence settings, it must be Yes |
| 09:45.02 | shaZwaz | on Xlite |
| 09:45.47 | shaZwaz | do this in ur sip.conf disallow = all allow = gsm |
| 09:46.45 | shaZwaz | in both the sections definitions |
| 09:47.49 | *** join/#asterisk caesar2 (caesar@p54800431.dip.t-dialin.net) |
| 09:47.54 | shaZwaz | in turn off silence suppersion on both Xlites Advance Settings > Audio > Silence > Transmit Silence = Yes |
| 09:50.12 | p1tst0p | ok, after changing the settings in the sip.conf do i need to restart anything |
| 09:50.37 | shaZwaz | sip reload on cli |
| 09:50.59 | *** join/#asterisk RoyK (~roy@80.239.107.121) |
| 09:51.33 | widowlicker | can someone tell me some cheap ip phones that works well with asterisk |
| 09:52.33 | shaZwaz | widowlicker, Grand Stream Budge Tone |
| 09:53.01 | widowlicker | thankx |
| 09:55.17 | p1tst0p | shaZwaz, same results with those chages dude. |
| 09:55.23 | p1tst0p | he hears me, i dont hear him |
| 09:55.55 | shaZwaz | what codec u see on ur Xlite as selected ? |
| 09:56.24 | p1tst0p | they both have GSM selected now, |
| 09:56.31 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
| 09:58.40 | r0d3nt|m | Anyone have any luck with Asterisk@Home ?? |
| 09:58.51 | shaZwaz | what are his network settings in Xlite , listening on specific IP or auto selected ? |
| 09:59.09 | r0d3nt|m | sounds like it really doesn't work straight off the CD/ISO without a bunch of fiddeling and tweaking.... |
| 10:03.54 | p1tst0p | hello, am i still connected in here, think my internet just glyched. |
| 10:05.40 | p1tst0p | r0d3nt|m, im using it now |
| 10:06.24 | p1tst0p | altho, im having issues with one way audio atm |
| 10:06.40 | p1tst0p | shaZwaz, hes just nipped out for 10 mins, ill get him to check when he comes back |
| 10:06.48 | *** join/#asterisk LorenzoMarouani (~LorenzoMa@i01v-50-28.d4.club-internet.fr) |
| 10:06.51 | shaZwaz | k |
| 10:08.34 | r0d3nt|m | p1tst0p, i see... |
| 10:08.43 | r0d3nt|m | i'm helping someone install it.. and it's wierd... |
| 10:08.48 | r0d3nt|m | seemed like stuff was missing.. |
| 10:08.54 | r0d3nt|m | what user should it be running as ?? |
| 10:09.21 | p1tst0p | dunno man, i just put the cd in, left it 30 mins, when i came back it was installed |
| 10:09.39 | *** join/#asterisk brc-tux (~brc-tux@pD9E9A219.dip0.t-ipconnect.de) |
| 10:09.45 | r0d3nt|m | no missing files or directories on bootup ? |
| 10:09.48 | p1tst0p | just logged in as root ( password = password ) and checked ip address ( ifconfig ), then connected to it from a web browser |
| 10:09.48 | p1tst0p | no dude |
| 10:09.52 | p1tst0p | nothing missing my end |
| 10:09.56 | r0d3nt|m | hmm.. |
| 10:10.14 | r0d3nt|m | 1st run thru it.. all the web stuff was missing... |
| 10:10.56 | r0d3nt|m | "Asterisk died with code 1" |
| 10:11.01 | r0d3nt|m | is all he gets now out of it... |
| 10:13.13 | p1tst0p | is the cd ok, isnt there a checkmedia switch for bootup ? |
| 10:14.13 | r0d3nt|m | ok... |
| 10:14.33 | r0d3nt|m | Erick says: |
| 10:14.33 | r0d3nt|m | yah.. it's fine.. install was flawless... it's probably just a config problem |
| 10:15.09 | r0d3nt|m | so he got rid of the install problems... |
| 10:15.09 | p1tst0p | awsome |
| 10:15.33 | r0d3nt|m | ok folks.. thanks for the information .. i'm off to bed.. i think erick is too.. |
| 10:15.46 | r0d3nt|m | i'll be back in the "morning" |
| 10:19.13 | *** join/#asterisk cgeek (~cgeek@pl6.lawrence.org.uk) |
| 10:22.03 | *** join/#asterisk _THEEND_ (~DrEaM@host57-13.pool80182.interbusiness.it) |
| 10:23.50 | p1tst0p | shaZwaz, he has Auto detect ip |
| 10:24.12 | shaZwaz | change it to listen on his local IP |
| 10:24.26 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:27.18 | *** join/#asterisk cgeek (~cgeek@pl6.lawrence.org.uk) |
| 10:29.11 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-3-169.w81-249.abo.wanadoo.fr) |
| 10:29.11 | p1tst0p | shaZwaz, still no joy mate, its oddness. |
| 10:29.28 | jhiver | guys do you know what this AGI error means? |
| 10:29.31 | jhiver | AGI Rx << Urgent handler |
| 10:29.31 | jhiver | AGI Tx >> 510 Invalid or unknown command |
| 10:29.39 | *** join/#asterisk Tjardick (~tjardick@238.114-136-217.adsl.skynet.be) |
| 10:30.24 | shaZwaz | p1tst0p, thats all I c'd do , I think it a nat problem see the viki for connecting behind nat |
| 10:31.12 | p1tst0p | shaZwaz, wheres the wiki man, also just a quick one, do u know what ports need to be forwarded from my router to Asterisk, also what ports the clients need for X-Lite to be forwarded in etc |
| 10:31.49 | shaZwaz | there is a tool on Xten site called Net Check run it and it will tell u |
| 10:32.40 | p1tst0p | hmm i cany find that |
| 10:33.18 | *** join/#asterisk feral_kid (~not@209.205.207.130) |
| 10:34.18 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 10:34.23 | feral_kid | Anyone fluent with iax2 problem? |
| 10:34.23 | Mother__ | greetings |
| 10:34.26 | shaZwaz | u can do one more thing ....get the xlite log from ur friend and pastebin it |
| 10:34.41 | *** join/#asterisk rcsw (~rcsw@host213-123-195-3.in-addr.btopenworld.com) |
| 10:34.44 | p1tst0p | ok |
| 10:34.45 | p1tst0p | good plan |
| 10:35.13 | shaZwaz | that will show the IPs and ports |
| 10:35.54 | rcsw | anyone had experience in connecting asterisk to NTL in the UK, I'm moving from a BT to NTL line. Any pitfalls / gotchas I need to be aware of? |
| 10:36.28 | feral_kid | I have two fwd iax2 proxies... Both are registered and both work fine on out-dials... But I am not able to get anything coming in... I am behind two NAT'd firewalls, but that shouldn't be an issue, or so I am to understand... |
| 10:37.16 | Mother__ | unless the firewalls are blocking traffic, no |
| 10:37.21 | Mother__ | also look at your port mappings |
| 10:37.59 | *** join/#asterisk Rick_Hunter (~rhunter@02-096.008.popsite.net) |
| 10:40.10 | *** join/#asterisk feral_kid (~not@209.205.207.130) |
| 10:40.56 | jhiver | DAMN!!! |
| 10:41.01 | jhiver | AGI Rx << Urgent handler |
| 10:41.03 | jhiver | AGI Tx >> 510 Invalid or unknown command |
| 10:41.10 | jhiver | means: this script does not exist |
| 10:41.21 | jhiver | bloody hell it should be *more* obscure! |
| 10:45.33 | *** join/#asterisk Abbas (Abbas@203.81.194.242) |
| 10:45.35 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 10:49.09 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:49.21 | threat|BX | G'day, is 3 X100P clones for $127.50 AU a good or bad price? |
| 10:56.36 | RoyK | sounds reasonable |
| 10:59.29 | threat|BX | fair enough |
| 10:59.36 | threat|BX | would you like to go halfs with me ? :P |
| 11:00.34 | zyke | can someone recommend which method to use for realtime? odbc or mysql? |
| 11:01.41 | threat|BX | out source to india and get them to do it for you :) |
| 11:01.52 | threat|BX | thats what everyone else is doing :P |
| 11:02.14 | *** join/#asterisk afe ([PUcQXLMYO@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 11:02.39 | threat|BX | is $60 too steep for one X100P clone? |
| 11:03.36 | RoyK | zyke: doesn't matter, really |
| 11:03.44 | RoyK | threat|BX: the clones are usually no good, though |
| 11:04.18 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 11:06.15 | Mother__ | can I issue a System() command via the manager API? |
| 11:06.41 | Mother__ | I have tried but it returns that command System is unknown |
| 11:13.20 | *** join/#asterisk Bob_ (~icechat5@196.7.14.183) |
| 11:16.40 | zyke | RoyX: is reatime stable now? that you can use it in production? |
| 11:17.07 | *** join/#asterisk justinnnnnn (~justinnnn@c211-28-200-67.eburwd1.vic.optusnet.com.au) |
| 11:17.40 | RoyK | zyke: no. you can't |
| 11:17.50 | RoyK | well, of course you _can_ |
| 11:17.54 | RoyK | but then |
| 11:18.02 | RoyK | you prolly don't want to |
| 11:18.15 | stustu | Can Asterisk send CallerID to Zap phones on fxs interfaces? I can not get it to work. I'm in Sweden, so my phones expect the CallerID as DTMF after a polarity switch, before the first ring. |
| 11:18.19 | RoyK | realtime is in CVS HEAD only |
| 11:19.24 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 11:19.47 | *** join/#asterisk dan2 (~beta3@dan2.active.supporter.pdpc) |
| 11:19.57 | ManxPower | ~docs |
| 11:19.58 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 11:20.41 | *** join/#asterisk p1tst0p (~will@82-38-104-189.cable.ubr03.donc.blueyonder.co.uk) |
| 11:24.06 | *** join/#asterisk crich1999 (~crich@pD9E77F1B.dip.t-dialin.net) |
| 11:24.55 | crich1999 | ohla gringos ;) |
| 11:25.12 | crich1999 | what do you say to bug # 0003764 |
| 11:28.56 | Zeeek | ManxPower you're up early/late |
| 11:30.06 | ManxPower | Zeeek, early |
| 11:30.21 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:32.05 | p1tst0p | hes just gone offline i think |
| 11:32.14 | Zeeek | who's on first |
| 11:32.52 | *** join/#asterisk gonzo- (~gonzo@SIRIUS-ats227-UTC.ukrtel.net) |
| 11:37.37 | *** join/#asterisk lidl (~little@213-140-6-96.fastres.net) |
| 11:47.26 | zyke | RoyX: are you using it yourself? |
| 11:50.16 | *** join/#asterisk jhiver (~jhiver@ABesancon-102-1-2-200.w80-11.abo.wanadoo.fr) |
| 11:50.56 | jhiver | guyz... How can I tell asterisk to do stuff after a channel has been hungup? I've tried the 'g' option but it doesn't seem to do anything |
| 11:51.37 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 11:51.45 | PoWeRKiLL | Hi |
| 11:51.53 | jhiver | hi... |
| 11:52.03 | jhiver | I can repeat my question to you I guess :) |
| 11:52.06 | jhiver | guyz... How can I tell asterisk to do stuff after a channel has been hungup? I've tried the 'g' option but it doesn't seem to do anything |
| 11:52.21 | Zeeek | what about 'h' |
| 11:52.26 | *** join/#asterisk KLiMaX (~michael@adsl-203-15-140-20.arach.net.au) |
| 11:53.02 | jhiver | <PROTECTED> |
| 11:53.11 | jhiver | won't do much for what I wanna do... |
| 11:53.15 | Zeeek | no |
| 11:53.17 | jhiver | Basically I have: |
| 11:53.23 | Zeeek | it's the h extension |
| 11:53.29 | Zeeek | I'm leaving sorrry |
| 11:53.35 | jhiver | exten => s,5,Dial(${YKOZCC_DIAL}|180|gS(${YKOZCC_TIMEOUT})) |
| 11:53.35 | jhiver | exten => s,6,SetVar(YKOZCC_TIME_SPENT=ANSWEREDTIME) |
| 11:53.35 | jhiver | exten => s,7,AGI(ykozcc.postcall.agi) |
| 11:53.44 | Zeeek | when the channel hangs up you'll go to the 'h' |
| 11:53.57 | jhiver | But s,6, is never reached when I hang the call |
| 11:54.01 | jhiver | aaah I see |
| 11:54.07 | jhiver | so h,1,... ? |
| 11:54.24 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 11:54.24 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 11:55.08 | *** join/#asterisk Pj386 (~pj@fernande.happycoders.org) |
| 11:55.24 | Pj386 | Hello |
| 11:56.01 | jhiver | but it says this: |
| 11:56.03 | jhiver | Warning |
| 11:56.04 | jhiver | There are currently some problems to be aware of when using the 'h' extension. Specifically, the variables about the call are lost as the information is destroyed with the channel. |
| 11:56.11 | ManxPower | Post call processing is handled in two ways. What what way it's called can depend on which side of the call hangs up. |
| 11:56.24 | ManxPower | Most people do their post call processing using exten => h,1,Blah(... |
| 11:56.36 | jhiver | since ANSWEREDTIME is given my Dial, how do I grab this variable? |
| 11:56.58 | ManxPower | jhiver, Are you trying to use AGI for post call processing? |
| 11:57.02 | jhiver | yeah |
| 11:57.16 | ManxPower | jhiver, Well then you need to use the DeadAGI rather than AGI. |
| 11:57.31 | ManxPower | jhiver, And yes, many variables are not available after a call is disconnected. |
| 11:58.44 | jhiver | So how would you do it? I can't really look at the CDRs and I need to know how long the call has been (excluding ring time...) |
| 11:59.01 | ManxPower | jhiver, You may want to download the asterisk calling card app (astcc?) and look at the docs to see how the author of astcc suggests post call processing is done. |
| 11:59.15 | ManxPower | jhiver, What type of interface are your calls going thru? |
| 11:59.18 | jhiver | Or should just I move the Dial() withing the AGI script? |
| 11:59.29 | jhiver | VoIP |
| 11:59.32 | ManxPower | jhiver, Moving Dial into the AGI will NOT accomplish what you want. |
| 11:59.49 | ManxPower | jhiver, You have confirmed that you cannot access ANSWERED time via DeadAGI? |
| 12:00.09 | jhiver | Well I don't know yet I need to try |
| 12:00.25 | jhiver | so basically, h,1,DeadAGI(...) ok? |
| 12:01.57 | ManxPower | jhiver, Yes. |
| 12:02.29 | jhiver | ok I'll give that a go... |
| 12:03.06 | shadebob | hi, does anyone use exten => s/incoming number,1,xxxxxx in his dialplan? |
| 12:03.13 | zyke | RoyX: i have compiled asterisk-addons directory but there was no res_config_mysql.so, any idea? |
| 12:03.39 | zyke | RoyX: or is it app_addon_sql_mysql.so that i need for the realtime? |
| 12:03.55 | ManxPower | zyke, What version of Asterisk are you using? |
| 12:04.16 | ManxPower | shadebob, Your statement makes no sense. |
| 12:04.17 | zyke | ManxPower: i'm using from the CVS head |
| 12:04.28 | ManxPower | shadebob, unless you are trying to match when you are calling yourself. |
| 12:05.02 | zyke | downloaded the asterisk-addons folder from cvs head yesterday |
| 12:06.21 | ManxPower | zyke, I cannot help you. I don't run the developement version of Asterisk. |
| 12:06.37 | ManxPower | zyke, You are on the asterisk-cvs mailing list, of course? |
| 12:06.54 | shadebob | hi ManxPower : no i have already my same problem. An E1 line with SDA numbers... I have to send s/number to SIP phones and all others call to my old PBX... |
| 12:07.12 | zyke | no i am not .. may be i should be thanks. |
| 12:07.16 | shadebob | manxpower : but it seem s/number not working |
| 12:07.24 | ManxPower | shadebob, Do you understand what s/number is telling Asterisk? |
| 12:07.48 | ManxPower | s/number means "if you don't know the number dialed and the callerid is number, then match" |
| 12:07.48 | shadebob | ManxPower : s/number : number is callee number? |
| 12:08.09 | ManxPower | shadebob, "s" matches ONLY if we do not know what number was dialed. |
| 12:08.15 | shadebob | ManxPower : hmmmm... callerid, not calle... |
| 12:10.02 | shadebob | ManxPower : sorry for my newbie questions.... I haven't E1 line in my office and my asterisk server is for my next office so I work.. in theory |
| 12:10.39 | ManxPower | If you want to match the callED number and Asterisk knows the callED number then exten => number,1,Blah(... |
| 12:11.11 | ManxPower | shadebob, Have you confirmed that the telco is sending you the number dialed? |
| 12:11.43 | shadebob | ManxPower : yes |
| 12:12.06 | ManxPower | then exten => number,1,Blah(.. should work. |
| 12:12.48 | ManxPower | If exten => s,1,Blah(... is run, then Asterisk is NOT receiving the dialed number. |
| 12:13.24 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 12:13.34 | shadebob | ManxPower : thanks for your patience |
| 12:13.46 | ManxPower | shadebob, you can thank me by finding me a job in Europe. |
| 12:14.03 | shadebob | manxpower : sorry, i'm in Morocco :s |
| 12:14.06 | lidl | ManxPower, where in Europe? |
| 12:14.49 | ManxPower | lidl, I would prefer somewhere in Benelux, but all serious offers will be considered. |
| 12:16.13 | lidl | i'm in italy, if i hear something interesting i'll give you an advice |
| 12:16.30 | ManxPower | lidl, Thank you. |
| 12:17.23 | ManxPower | Unfortunatly, I am not an EU citizen, but I feel that my skills are unique enough that a company can justify hireing a non-EU citizen |
| 12:17.25 | shadebob | ManxPower : where are you actually? Can you move for consulting mission? |
| 12:17.44 | *** join/#asterisk d00gster (~doughant@toronto-HSE-ppp4046057.sympatico.ca) |
| 12:17.49 | ManxPower | shadebob, I am currently located near New Orleans, Louisiana USA. |
| 12:18.06 | ManxPower | shadebob, as I said, all serious offers will be considered. |
| 12:19.49 | ManxPower | shadebob, I want to live in Europe. I do not want to live in a country that would elect George Bush for a SECOND term. |
| 12:19.50 | *** join/#asterisk Bob_ (~icechat5@196.7.14.183) |
| 12:20.17 | shadebob | ManxPower : idea of your day cost for a consulting mission in Casablanca or Rabat (Morocco) during 2-3 months? |
| 12:21.17 | ManxPower | shadebob, That would depend on many things, but it would have to be enough to cover my costs and make a decent profit. |
| 12:21.39 | ManxPower | Also for only 2-3 months I would not be able to give up my current apartment. |
| 12:21.52 | shadebob | ManxPower : ok :s |
| 12:22.09 | Bob_ | Hi there. I am trying to install Asterisk. The file I downloaded from the Asterisk website is asterisk-1.0.6.tar.tar. How does one unpackage this file type - should it not be .tar.gz? Any assistance would be greatly appreciated |
| 12:22.19 | shadebob | ManxPower : French speaking? |
| 12:22.20 | ManxPower | shadebob, I will consider sort term contracts, but I would prefer a long term or perm contract. |
| 12:22.33 | ManxPower | shadebob, Like most USAians I speak/write only english. |
| 12:23.03 | ManxPower | My Resume/CV should be finished in a couple of days |
| 12:23.38 | ManxPower | shadebob, Morocco would be an interesting place to live for a few months. |
| 12:24.05 | ManxPower | shadebob, Does Morocco have a secular govt or a religious govt? |
| 12:24.24 | shadebob | ManxPower : religious govt |
| 12:24.52 | shadebob | ManxPower : I'm french and Morocco is for me a land of dream ;) |
| 12:24.52 | lidl | ManxPower, i suggest you sending your CV to telcos directly |
| 12:24.54 | ManxPower | That makes it somewhat less interesting, but that by itself would not keep me from considering a contract there. |
| 12:25.20 | ManxPower | lidl, I'm MOST interested in Asterisk related consulting. |
| 12:25.30 | shadebob | ManxPower : Telco liberalization is in progress here... I will see opportunity for you if you send me a CV |
| 12:25.45 | ManxPower | shadebob, /msg me your e-mail address |
| 12:26.17 | lidl | gotta go |
| 12:29.44 | *** join/#asterisk Zaffa (~mjauhola@addr-212-50-156-26.suomi.net) |
| 12:32.30 | justinnnnnn | hey |
| 12:32.35 | justinnnnnn | anyone played with sccp speed dials ? |
| 12:32.44 | justinnnnnn | specificialy with 7914's |
| 12:32.54 | justinnnnnn | when we press a speed dial... it goes to 's' extension |
| 12:33.17 | ManxPower | ~google sccp speed dial |
| 12:33.29 | ManxPower | drat |
| 12:33.42 | ManxPower | ~google site:lists.digium.com sccp "speed dial" |
| 12:33.48 | justinnnnnn | oo thanks :) |
| 12:35.49 | justinnnnnn | speeddial = 4,John Doe |
| 12:36.02 | justinnnnnn | but alas.. i dont get how u put the actual number in.. |
| 12:36.18 | ManxPower | justinnnnnn, very few people use SCCP with Asterisk |
| 12:36.29 | justinnnnnn | is that cause they dont use cisco ? |
| 12:36.33 | justinnnnnn | or theres something better to use ? |
| 12:36.36 | justinnnnnn | besides sip i mean |
| 12:36.45 | ManxPower | Most people use SIP. |
| 12:37.12 | ManxPower | nomad_, most people make sure the phone they use supports SIP. Many Cisco phones support SIP. |
| 12:37.15 | justinnnnnn | we got those gay 7914's to try |
| 12:37.16 | justinnnnnn | and a 7970 |
| 12:37.36 | justinnnnnn | i dont think 7970 does sip either :( |
| 12:37.42 | justinnnnnn | its working gr8 so far.. sccp |
| 12:37.48 | justinnnnnn | using nat and everything etc |
| 12:40.26 | justinnnnnn | hmmm ses u need callmanaged to set up speed dials... |
| 12:40.27 | justinnnnnn | hmmmm |
| 12:41.00 | Bob_ | any ideas about why when I download Asterisk from asterisk.org it comes as a .tar.tar file, which I don't know how to deal with? |
| 12:41.52 | ManxPower | Bob_, Unfortunatly, you MUST be familiar with Linux and building software in order to use Asterisk. |
| 12:42.36 | pascals | Bob_ suse comes with precompiled astersik |
| 12:42.40 | pascals | asterisk. |
| 12:42.51 | Bob_ | Should it not be a ".tar.gz" file and not ".tar.tar" |
| 12:43.17 | Bob_ | I am using Redhat 9, but downloading the file from windows xp |
| 12:43.26 | pascals | That is perhaps your webbrowser doing something funny |
| 12:43.36 | Bob_ | could be |
| 12:44.04 | pascals | try tar xf, not xzf to see if it has been decompressed on the fly |
| 12:44.05 | Bob_ | but I am often able to download .tar.gz files |
| 12:44.10 | Bob_ | ok |
| 12:44.28 | pascals | and otherwise just try xzf. |
| 12:45.19 | Bob_ | xzf did work - thanks |
| 12:46.52 | ManxPower | Recently a television show had the word "breast" deleted because they were afraid they would be fined by the FCC for indecency. |
| 12:46.58 | ManxPower | (in the USA, of course) |
| 12:47.59 | ManxPower | pascals, It doesn't, but neither does half the stuff we talk about here. 8-) |
| 12:48.17 | pascals | Oh, I hadn't been paying much attention - |
| 12:48.25 | pascals | We do that over on #tcl too |
| 12:49.29 | pascals | There is a breast for every person on the planet, how on earth can something so common be indecent. |
| 12:50.32 | ManxPower | pascals, Remember the USA was originally founded by massive numbers of European prudes. |
| 12:51.01 | pascals | Hmmm. Good point. |
| 12:51.30 | ManxPower | During the 1960's and 1970's the USA became quite open minded, but since about 1980 things have been getting pretty bad. |
| 12:51.53 | ManxPower | Which is why I want to get out of the USA. |
| 12:52.29 | ManxPower | pascals, I would love to live in the netherlands. |
| 12:52.38 | ManxPower | My home town is Holland, Michican 8-) |
| 12:52.44 | ManxPower | Michigan too. |
| 12:52.53 | Zaffa | I have a weird problem with calls related with asterisk (1.0.1) and snom220. Some of the incoming calls comes to phone with no alerting sounds and after answering no audio is transfered. Phone screen shows incoming call normally... |
| 12:53.06 | pascals | Ah, we drove by Holland a couple of years ago. |
| 12:53.06 | ManxPower | Zaffa, You need to upgrade. |
| 12:53.08 | Zaffa | Anyone experienced similar problems? |
| 12:53.59 | *** join/#asterisk __Sparks_ (ringding@bb-194-6-115-252.ukonline.co.uk) |
| 12:54.02 | Zaffa | ManxPower, i thought so, but i like to be sure that it helps cause the system is in production use... =/ |
| 12:54.57 | pascals | If you live in the Netherlands *and* are offended by bare breasts, you're going to have a hard time getting by - too many adds have almost naked women |
| 12:55.47 | __Sparks_ | I have a little problem with calls not routing to voicemail form Sipgate - Can anyone here lend me a hand! - I havre the following in my extensions.conf - exten => SIPGATENUMBER,1,Dial(SIP/503,2,tr) - but I get the following error "pbx.c:1934 ast_pbx_run: Timeout, but no rule 't' in context 'trunk' |
| 12:55.55 | pascals | I don't quite get why firefly wont show clid, and iaxclient/iaxcomm does. |
| 12:57.00 | Bob_ | Sorry if this is a stupid question, but what is the function of libpri? |
| 12:57.59 | ManxPower | Bob_, It's for use with PRI lines. |
| 12:58.11 | RoyK | and BRI with bristuff..... |
| 12:58.32 | ManxPower | RoyK, Thanks. I always forget about that stuff. |
| 12:58.48 | RoyK | should prolly be called libisdnsignallingsomething |
| 12:59.32 | Bob_ | do I need that - I will just be using VoIP softphones |
| 13:00.28 | RoyK | ~seen zoa |
| 13:00.41 | jbot | zoa <~zoa@dD577788C.access.telenet.be> was last seen on IRC in channel #asterisk, 14h 8m 9s ago, saying: 'and now im super active'. |
| 13:00.50 | RoyK | Bob_: you'll only need it to connect to a pri or an hfc-pci bri |
| 13:01.31 | justinnnnnn | grrr stupid speed dials :((( |
| 13:01.34 | justinnnnnn | stupid call manager |
| 13:01.53 | Bob_ | Thanks - and I guess I won't need Zaptel either then |
| 13:02.16 | RoyK | correct |
| 13:04.23 | shadebob | Can i control IP phones (7940) from a PC? |
| 13:05.22 | __Sparks_ | if a sip phone has na extantion number of 503, and it's voicemail is 403 - in the dialplan, how do you send to voicemail (I tried (SIP/403) but it didnt work! |
| 13:06.22 | __Sparks_ | (yes I am a Noob!!!) |
| 13:07.09 | oej | Voicemail referes to the voicemailbox id, not the extension, so SIP/403 is wrong. 403 is a voicemailbox id, not a SIP peer |
| 13:07.26 | oej | Try voicemail(b403) |
| 13:09.44 | __Sparks_ | :-) excelent it works! - thanks! |
| 13:10.48 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 13:13.20 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 13:14.01 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 13:14.17 | *** join/#asterisk caesar2 (caesar@p54800BE6.dip.t-dialin.net) |
| 13:16.53 | pascals | can I do auto-dial out so the local phone rings first, and then starts trying the remote? |
| 13:17.48 | *** join/#asterisk w0w0 (~w0w0@80.26.165.152) |
| 13:18.36 | ManxPower | pascals, that is a complicated thing to do. |
| 13:18.45 | ManxPower | ~docs |
| 13:18.46 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:18.51 | ManxPower | see .call files and sample.call |
| 13:19.20 | *** join/#asterisk ws9455 (~ws9455@adsl-68-94-61-239.dsl.rcsntx.swbell.net) |
| 13:20.07 | pascals | Yes, I found that and know how to do it |
| 13:20.16 | *** part/#asterisk widowlicker (~Naturalbl@62.77.178.121) |
| 13:20.52 | pascals | The trick is that I want our helpdesk person to pick up his/her phone before the other side rings |
| 13:22.51 | pascals | I guess I could solve it by using the manager api, calling the helpdesk extension and then dialing out. |
| 13:23.06 | Leland | pascals: are these calls coming in from the PSTN ? |
| 13:23.33 | pascals | No, from the office (SIP phones) out to the pstn |
| 13:24.03 | ManxPower | pascals, You said "auto dialout" that means "dialing without a call coming in" |
| 13:24.23 | ManxPower | If you just want calls to exten => 611,1,Dial(Zap/1/5551212) that's easy to do. |
| 13:25.43 | Zeeek | pascals where are the numbers called coming from? A file or db? |
| 13:25.47 | pascals | What I want it that the helpdesk worker clicks a button on the computer screen, picks up phone, and asterisks dials out |
| 13:26.08 | ManxPower | pascals, then you need call files. |
| 13:26.16 | pascals | Zeeek: database, but I wrote the durned thing, so I can write a file wherever I want |
| 13:26.17 | Zeeek | or queues maybe |
| 13:26.36 | Zeeek | pascals yeah that's kind of what I was getting at |
| 13:26.46 | pascals | ManxPower: the way I read it - it will always call out customer before ringing the helpdesk phone. |
| 13:27.05 | Zeeek | for example: agent dials **1 and your yet to be written connects the next number with your guy |
| 13:27.27 | Zeeek | yet to be written application |
| 13:27.44 | Zeeek | ddoesn't that do what you describe? |
| 13:27.55 | pascals | Yea, mostly. |
| 13:27.59 | Zeeek | cool, no? |
| 13:28.17 | Zeeek | then your app marks a flag that that number has been called |
| 13:28.29 | pascals | The most convenient way would be: click button in app, phone rings, pick up, asterisks dials out. |
| 13:28.33 | Zeeek | or your agent hits a digit at the end of the call to indicate status |
| 13:28.59 | Zeeek | what phones? |
| 13:29.12 | Zeeek | seems easy enough to pick one up and hit one button |
| 13:29.13 | pascals | the phone on the desk of the helpdesk worker |
| 13:29.19 | Zeeek | what kind I mean |
| 13:29.46 | pascals | Oh, I have to decide, but the polycom ones look good |
| 13:30.12 | pascals | the ip300/500 or something |
| 13:30.29 | pascals | They might be able to take directions from the pc too. |
| 13:31.16 | *** join/#asterisk bjohnson_ (~bjohnson@jecinc.tor.istop.com) |
| 13:32.32 | Zeeek | why not just show a scren "Pick up phone and idal *11" |
| 13:33.03 | pascals | I am just trying to save user work |
| 13:33.35 | Zeeek | that have to pick after the ring, so why not before and signle button dial? |
| 13:33.42 | Zeeek | that's one finger stab |
| 13:34.07 | pascals | They just clicked a button on screen first, that is why. |
| 13:34.13 | pascals | I know I am nitpicking |
| 13:34.14 | Zeeek | my own fingers aren't working so great |
| 13:34.36 | Zeeek | I'd rather hit a button on the phone, personally |
| 13:34.42 | Zeeek | very subjective that |
| 13:35.00 | Zeeek | too much clicking on screen is bad for you |
| 13:35.05 | pascals | Yes, probably very subjective :) |
| 13:36.32 | Tjardick | pascals: you can just generate a call file in the asterisk spool dir that will call the user and then will call the destination (or initiate a call via the manager). |
| 13:37.42 | pascals | Tjardick: but I want our people to pickup the phone first, I don't want the customer we're calling to hear tones before talking to our people |
| 13:38.13 | pascals | It seems odd call files only work like that |
| 13:39.06 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 13:39.21 | ManxPower | Create a .call file that uses Local/1234@blah. Then in [blah] put in exten => 1234,1,Dial(SIP/supportphone,,M(calloutmacro)) and in [macro-calloutmacro] dial the second destination. |
| 13:39.28 | ManxPower | See? Easy! |
| 13:40.06 | Mother_ | ;) |
| 13:40.20 | pascals | ManxPower: that is what I mean :) |
| 13:40.37 | ManxPower | pascals, I DID say what you want to do is complicated. |
| 13:40.54 | pascals | I don't care about complicated |
| 13:40.58 | ManxPower | You could also generate TWO .call files and point them to the same meetme conference. |
| 13:41.16 | ManxPower | one .call to dial the support agent, one .call to dial the luser |
| 13:41.18 | *** join/#asterisk juanjoc (~jcomellas@200.73.189.82) |
| 13:41.59 | pascals | I like the macro one better. |
| 13:42.29 | *** join/#asterisk Zgarbi (~my@212.58.125.68) |
| 13:42.33 | Zgarbi | hi |
| 13:44.42 | Zgarbi | I want to add MySQL addon to my asterisk, I already configure MySQL server and cdr_mysql.conf but can't understand how to say system to work with mysql ??? what do I need? |
| 13:45.42 | Zgarbi | I already compile asterisk_addon with - make clean, make install as described in manual |
| 13:45.47 | Zgarbi | but what more? |
| 13:46.02 | sambal | modules |
| 13:47.24 | Zgarbi | ? |
| 13:47.30 | Zgarbi | where and how? |
| 13:48.16 | *** join/#asterisk Hairlocks (~Hairlocks@82-70-162-85.dsl.in-addr.zen.co.uk) |
| 13:49.06 | *** part/#asterisk Hairlocks (~Hairlocks@82-70-162-85.dsl.in-addr.zen.co.uk) |
| 13:49.15 | *** join/#asterisk Hairlocks (~Hairlocks@82-70-162-85.dsl.in-addr.zen.co.uk) |
| 13:49.42 | Zgarbi | [root@ms asterisk-addons-1.0.6]# make |
| 13:49.42 | Zgarbi | ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` |
| 13:49.42 | Zgarbi | app_addon_sql_mysql.c:31:25: mysql/mysql.h: No such file or directory |
| 13:49.42 | Zgarbi | cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory |
| 13:49.42 | Zgarbi | cdr_addon_mysql.c:33:19: mysql.h: No such file or directory |
| 13:49.42 | Zgarbi | cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory |
| 13:49.43 | Hairlocks | Where can I buy a TDM04B that will ship to the UK? |
| 13:50.13 | Zgarbi | what it wants? |
| 13:50.24 | Zeeek | Hairlocks telepliant |
| 13:50.30 | *** join/#asterisk TheEmperor (TheEmperor@j129.dms40.jaring.my) |
| 13:50.32 | Zeeek | they're IN the UK |
| 13:50.39 | Bob_ | I am not a linux expert, but I have successfully set up SIP Express Router, with SEMS for conferencing - I now want to migrate to Asterisk, because I believe that it is more stable... |
| 13:50.40 | ManxPower | Zaffa, It's pretty obvious that you don't have mysql-devel installed and it looks like you don't have glibc-headers nor asterisk installed. |
| 13:50.48 | Bob_ | I have just installed asterisk, but all the functionality wich asterisk provides makes it seems rather complex ... |
| 13:50.56 | Bob_ | Now, I am wanting to use a very small amount of the functionality. I need Asterisk to support softphones dialing into a conference, and be able to support multiple conferences concurrently... |
| 13:51.03 | Bob_ | The way that SER works is a client will dial a conference id - sip:<conf_id>@domain and then be routed to the conference module if the conference id starts with x (which is specified in the configuration file)... |
| 13:51.13 | Bob_ | Does Asterisk work in a similar fashion? |
| 13:51.42 | Zgarbi | ok, I will try to install mysql-devel |
| 13:52.28 | ManxPower | Bob_, Forget EVERYTHING you learned when using SER. |
| 13:52.55 | ManxPower | Bob_, Asterisk pretty much considers EVERYTHING an extension |
| 13:53.11 | *** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 13:53.39 | pascals | I like the general-ness of everything being an extension |
| 13:53.39 | crich1999 | who kram |
| 13:53.56 | pascals | Very Tcl. |
| 13:54.02 | pascals | Or zen, if you will |
| 13:54.11 | Bob_ | ok, so basically would specify an extension to route to conference |
| 13:54.22 | ManxPower | Bob_, yes. |
| 13:54.24 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 13:54.41 | Hairlocks | Zeeek: did you spell it right, a google for telepliant came up with www.TelePliant.com |
| 13:54.54 | Bob_ | Does the conference work the same way as SER & SEMS, in that they conference acts like a "client" on the server and receives and broadcasts packets? |
| 13:55.06 | ManxPower | Bob_, Forget EVERYTHING you learned when using SER. |
| 13:55.34 | ManxPower | Bob_, Pretty much any question like "Does Asterisk do X like SER does X?" has the answer of "NO!" |
| 13:55.43 | Bob_ | So how does conferencing work with Asterisk? |
| 13:55.56 | ManxPower | Bob_, exten => 666,1,MeetMe |
| 13:56.12 | Zeeek | try voiptalk.org |
| 13:56.26 | ManxPower | Bob_, Asterisk takes the audio from each client, converts it to SLINEAR, mixes it, then converts it back to the codec each client uses and sends it off. |
| 13:56.30 | ManxPower | ~docs |
| 13:56.31 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:56.49 | Zeeek | Hairlocks yea, that's it http://voiptalk.org |
| 13:56.59 | ManxPower | Of course you need zome kind of zaptel timer to use MeetMe. |
| 13:58.00 | *** join/#asterisk nix000 (~nix000@ip194-172.dsl.istop.com) |
| 13:58.53 | Bob_ | so I would need to install the zaptel module then? |
| 13:59.38 | *** join/#asterisk RPH (~ron_hartm@ppp-69-213-251-59.dsl.dytnoh.ameritech.net) |
| 14:00.03 | RPH | i have lost my mind and can not seam to figure out what i am missing on sip registration |
| 14:00.22 | *** join/#asterisk coppice (~chatzilla@111.196.17.210.dyn.pacific.net.hk) |
| 14:00.31 | RPH | I would be most appreciative if someone can give me a good swift kick in the ass and tell me why i am so stupid |
| 14:00.51 | RPH | ne1 able to look at my debug http://pastebin.com/257718 |
| 14:00.59 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
| 14:02.11 | *** part/#asterisk rious (~rious@adsl-69-208-72-102.dsl.klmzmi.ameritech.net) |
| 14:02.18 | Hairlocks | Zeeek sorry for bugging you about this, www.voiptalk.org has the FXS cards but none of the FXO cards or modules. I wish to buy a card with 4 FXO ports, I read somewhere the they do not have the CE mark yet? |
| 14:02.59 | coppice | not entirely stupid, then :-) |
| 14:03.02 | Zeeek | Hairlocks I think you might be able to order from telappliant - try emailing or calling them |
| 14:03.05 | ManxPower | I don't help anyone that uses the "word": ne1 |
| 14:03.09 | bjohnson_ | can the phones call into * at all? Find or make an extension that will play a file and try calling it |
| 14:03.26 | RPH | yeah i can hit the voicemail |
| 14:03.29 | *** join/#asterisk ascvdf (~1234@bzq-218-184-235.red.bezeqint.net) |
| 14:03.33 | nix000 | anyone can tell me what is the difference between the X100P FXO PCI and the fxo modules on the quad pci card ? are they identical ? whats the deal ? |
| 14:03.56 | coppice | ManxPower: ! ne 1 at all? |
| 14:03.58 | Zeeek | they do the same thing but are different |
| 14:04.04 | *** part/#asterisk Zaffa (~mjauhola@addr-212-50-156-26.suomi.net) |
| 14:04.24 | ManxPower | nix000, They are significantly different from a chip standpoint, but not from an Asterisk standpoint. |
| 14:04.28 | bjohnson_ | nix000: they are different |
| 14:04.32 | nix000 | Zeeek: do they have the same "features" , is the x100p discontinued ? |
| 14:04.42 | bjohnson_ | yes discontinued |
| 14:04.47 | Zeeek | it is officially discontinued |
| 14:04.48 | bjohnson_ | same * config |
| 14:04.51 | Zeeek | I hear |
| 14:04.57 | RPH | it seams that when i do a show sip registry i should see my phones that have registered |
| 14:05.09 | bjohnson_ | RPH: that is something different |
| 14:05.12 | ManxPower | nix000, The FXO modules have more features than the X100P |
| 14:05.21 | bjohnson_ | but you CAN use voicemail? |
| 14:05.33 | RPH | let me see |
| 14:05.34 | RPH | brb |
| 14:05.38 | nix000 | ManxPower: what features are missing on the x100p ? |
| 14:05.42 | ManxPower | RPH: NO NO NO! "sip show registry" shows what servers Asterisk is registered to using a register => line in sip.conf. |
| 14:05.52 | bjohnson_ | RPH: that would confirm that your phones are talking to * |
| 14:05.57 | Zeeek | ne how |
| 14:06.01 | ManxPower | nix000, X100P only has 600Ohm impedence, it has no echo cancel features on the card. |
| 14:06.17 | ManxPower | "sip show peers" and "sip show users" is what you want. |
| 14:06.22 | Zeeek | the module is $10 cheaper |
| 14:06.39 | bjohnson_ | sounds like he has them talking to * .. he just needs to make extensions for each of the phones |
| 14:06.57 | nix000 | ManxPower: what about country capabilities .. would it work in europe, us ? |
| 14:07.00 | ascvdf | hi =) i want to install asterisk on a pc ... what do i need to do first? |
| 14:07.13 | Zeeek | nix000 not guaranteed to work in Europe (where I am) |
| 14:07.18 | bjohnson_ | ascvdf: a new, blank pc? |
| 14:07.23 | ManxPower | nix000, Which one X100P or TDM400P FXO? |
| 14:07.23 | ascvdf | yeap |
| 14:07.37 | ManxPower | ascvdf, Read the docs |
| 14:07.39 | Hairlocks | nix000 the module has different country support in the hardware |
| 14:07.39 | ManxPower | ~docs |
| 14:07.40 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:07.40 | Zeeek | nix000 for example, some of my phones will not ring on the FXS card |
| 14:07.46 | bjohnson_ | ascvdf: easiest is to find and download asterisk@home which will do the full system install from CD |
| 14:07.49 | ManxPower | Did slashdot do another article on Asterisk??? |
| 14:07.57 | *** join/#asterisk kensuke (~bryan@rrba-146-118-90.telkomadsl.co.za) |
| 14:07.58 | nix000 | ManxPower: ia mworried about the the x100p. |
| 14:08.05 | Zeeek | nix000 sometimes little patches are needed or even (gasp) hardware adapters |
| 14:08.14 | Hairlocks | www.theregister.co.uk did an articule today |
| 14:08.22 | kensuke | <PROTECTED> |
| 14:08.25 | ManxPower | nix000, Don't expect the X100P to work correctly outside the USA. |
| 14:08.29 | Zeeek | nix000 and if you are by any chance a BT customer, look at the mailing list |
| 14:08.30 | ascvdf | asterisk@home . where can i find it plz? |
| 14:08.30 | ManxPower | It MAY,but don't expect it to. |
| 14:08.41 | Zeeek | depends on the country Manx |
| 14:08.52 | bjohnson_ | ascvdf: it installs Centos (a RHE clone) and asterisk. You can choose whether to use the * gui included or config the text files |
| 14:08.52 | Zeeek | and the amount of dicking around you are willing to do |
| 14:09.01 | bjohnson_ | ascvdf: google |
| 14:09.31 | bjohnson_ | ~theusualsuspects |
| 14:09.33 | ascvdf | ok thanx ppl |
| 14:09.36 | RPH | i think it was a typo in my include on extensoins.conf |
| 14:09.39 | RPH | Grrrr |
| 14:09.44 | RPH | thanks for the kick in the ars |
| 14:10.11 | ManxPower | Did you hear about the vampire typesetter? All his mistakes were Type O's. |
| 14:10.20 | bjohnson_ | I thought some were A- |
| 14:10.24 | nix000 | Zeeek: i am in canada. but i was dreaming about setting up stuff in africa where a $20 pbx system makes a lot of sense to home users. |
| 14:10.25 | Hairlocks | nix000: the X100P can be had for £10 the FXO module I can't find in the UK, I did read somewhere that it doesn't have the CE mark yet, so can't be sold in europe. |
| 14:10.33 | ManxPower | RPH, "show dialplan" may have let you see the error |
| 14:10.42 | kensuke | please /// what does FXO stand for ? |
| 14:10.49 | Zeeek | I thought they mostly had cells in those countries |
| 14:10.50 | bjohnson_ | ~fxo |
| 14:10.51 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 14:11.00 | kensuke | yeah ! |
| 14:11.00 | ManxPower | Hairlocks, That is totally incorrect. X100P CLONE can be had for $10 |
| 14:11.26 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 14:11.37 | nix000 | Hairlocks: i have seen 10 of them for $10 in ebay |
| 14:11.38 | bjohnson_ | nix000: a $20 pbx system will be hard to find/make |
| 14:11.51 | ManxPower | But those clones will be very very hard to find soon. |
| 14:12.33 | Hairlocks | How can you tell it is a clone? |
| 14:12.38 | kensuke | What software can be used to connect as an extention to asterisk pbx debian server ? |
| 14:12.42 | Zeeek | lift the tail feathers |
| 14:12.48 | nix000 | bjohnson_: this http://techdatapros.com/asterisk/ claims otherwise |
| 14:12.51 | ManxPower | Hairlocks, If it costs $10 then it's a clone. |
| 14:12.53 | pascals | I'd only spend it on man-hours getting it to work |
| 14:12.57 | bjohnson_ | Hairlocks: they are selling it for less than $100 |
| 14:13.09 | ManxPower | Hairlocks, Digium no longer sells the X100P. |
| 14:13.38 | ManxPower | Since the chip used on the X100P is no longer being made by the OEM (intel) |
| 14:14.08 | Zeeek | <PROTECTED> |
| 14:14.09 | Zeeek | Not quite |
| 14:14.09 | pascals | Zeeek: it should be, but not quite indeed |
| 14:14.10 | nix000 | ManxPower: what is the chip ? |
| 14:14.16 | Zeeek | because of lack of consistent up to date docs |
| 14:14.23 | *** join/#asterisk prh (~paul@212.13.203.69) |
| 14:14.43 | ManxPower | nix000, I don't know. Pry the heatsink off an X100P and look. |
| 14:14.48 | pascals | Zeeek: that, and the hardware required is, um, troublesome |
| 14:14.49 | bjohnson_ | nix000: that starts with a free computer. You got many of those in your Africa example? |
| 14:14.53 | ManxPower | It might be the Ambient chipset. |
| 14:15.15 | Zeeek | pascals well the statement would need to compare Apache and * with no hardware to be fair |
| 14:15.20 | Leland | I have an X100P sitting in front of me and there is no heat sink on it at all |
| 14:15.27 | nix000 | ManxPower: i find there could be quite a good use in canada even |
| 14:15.33 | ManxPower | Leland, If it doesn't have a heatskink then it's a CLONE |
| 14:15.39 | Leland | It's the ambient |
| 14:15.42 | Leland | MD3200 |
| 14:15.47 | Zeeek | Canada has some anti-voIP legislation, do they not? |
| 14:15.53 | ManxPower | The clones use the same chipset |
| 14:16.00 | pascals | Zeeek: unfortunately, there aren't enough voip->pstn providers yet |
| 14:16.08 | nix000 | Zeeek: never heard .. what have it booming in fact |
| 14:16.08 | Zeeek | something about taxing calls that start and terminate in the country |
| 14:16.34 | Zeeek | pascals I think there are almost enough |
| 14:16.43 | Zeeek | not enough in Europe yet though |
| 14:16.52 | pascals | Not in the netherlands |
| 14:16.53 | nix000 | pascals: all the voip terminators i have seen always claim us and canada ! |
| 14:16.59 | Leland | it might be a clone.. guess that's why I got a deal for 10 of them for cheap bux |
| 14:17.00 | Zeeek | but that's because all the telcoms are offering unlimited national calling themselves |
| 14:17.17 | ManxPower | If the card is much less than US$100 then it's a clone. |
| 14:17.26 | Zeeek | or used? |
| 14:17.28 | Leland | heh.. I got 10 of them for less than $100 |
| 14:17.29 | Leland | :P |
| 14:17.31 | Leland | brand new |
| 14:17.32 | pascals | Zeeek: long distance? I thought it was only local calls |
| 14:17.34 | RPH | give me a sec and i will get the print out |
| 14:17.49 | coppice | min x100p was much less than $100... $0 :-) |
| 14:17.51 | Leland | works fine though |
| 14:18.10 | Zeeek | in France people get a free box (called "Freebox") and for $37/mo make unlimited national calls and have 20Mbit DSL |
| 14:18.11 | ManxPower | RPH, If you paste more than a couple of lines to the channel you'll get bitchsmacked into next week. use pastebin.ca |
| 14:18.24 | Leland | though I don't like the price of the TDM400P |
| 14:18.36 | RPH | yeah i will |
| 14:18.51 | nix000 | ManxPower: i see Digium Wildcard X100P OEM FXO PCI for 6.95 on ebay ! |
| 14:18.59 | Leland | Zeeek: the 20Mbps dsl (and television) is only available with an unbundled line though |
| 14:19.07 | Zeeek | Correct sir |
| 14:19.30 | Zeeek | and there are issues about signing for a year winth a documents raise in very fine print |
| 14:19.31 | Leland | with an FT line they can still get voice and 8Mbps dsl, but no television |
| 14:19.46 | RPH | http://pastebin.com/257721 |
| 14:19.58 | RPH | extensions are towrds the bottom |
| 14:20.01 | Zeeek | We have 4Mb (supposedly) for $18/mo |
| 14:20.24 | Zeeek | actually it's 2Mb but we got a nice letter months ago that it was double for the same price |
| 14:20.52 | Leland | the UK is about the only main European country left which still doesn't have an infrastructure for combined voice, video, and data over high-speed DSL |
| 14:20.55 | nix000 | Zeeek: i wish i was in france ! |
| 14:21.11 | Zeeek | there are othjer less convenient things about it but there are also advantages |
| 14:21.16 | ManxPower | BTW, how much is a GSM SIM card + service (no contract) in Europe? |
| 14:21.23 | Zeeek | 20eu |
| 14:21.30 | Zeeek | plus mileage |
| 14:21.37 | ManxPower | What's the per min cost? |
| 14:21.42 | Zeeek | high! |
| 14:21.43 | coppice | nix000: there are nicer places with the same low prices :-) |
| 14:21.45 | ManxPower | Free incoming, I assume. |
| 14:21.48 | Zeeek | yup |
| 14:21.53 | Zeeek | and free SMS email alerts |
| 14:21.56 | Leland | would also depend on where you're calling .. different tarrifs |
| 14:22.01 | nix000 | coppice: and the speed ? |
| 14:22.04 | Zeeek | and you can read your email w GPRS |
| 14:22.06 | pascals | And you need a sim card for each country- |
| 14:22.16 | ManxPower | Ick! |
| 14:22.25 | coppice | you can have 100M for $25 in japan |
| 14:22.26 | Zeeek | it's about 35c/minute IIRC |
| 14:22.34 | ManxPower | What do people do that travel thru .eu very much? |
| 14:22.37 | pascals | for prepaid, you'd pay the premium if you roam across borders |
| 14:22.38 | Leland | I stick to my contract-based GSM ... |
| 14:22.38 | ManxPower | Buy 15 SIM cards? |
| 14:22.58 | pascals | ManxPower: have a contract |
| 14:23.00 | ManxPower | Leland, I will be happy to have a contract if I'm in the EU for 12 months. |
| 14:23.03 | pascals | Not prepayed |
| 14:23.22 | Zeeek | ManxPower no if you don't call much you can still roam on prepaid - I do it on vacation |
| 14:23.38 | ManxPower | Zeeek, So incoming calls are free, even when roaming? |
| 14:23.42 | Zeeek | the thing is we do not call that much - the trick is to get people to call you :) |
| 14:23.46 | pascals | If you're called - you still have to pay for the incomming call |
| 14:24.12 | pascals | at least, that is what all dutch providers do |
| 14:24.13 | Zeeek | I'd have to go look at my Orange account |
| 14:24.17 | ManxPower | I plan on being in the EU for 5 weeks this summer. |
| 14:24.17 | RaYmAn-Bx | ManxPower: the prices are very different depending on which country |
| 14:24.21 | Zeeek | le tme see |
| 14:24.36 | ManxPower | Flying into Stockholm, then to Brussels, then to Amsterdam, then to Madrid. |
| 14:24.44 | pascals | ManxPower: all dutch providers charge for receiving calls while roaming |
| 14:25.29 | RaYmAn-Bx | ManxPower: also remember that you need a phone that can do GSM 900/1800 (either is fine) |
| 14:25.44 | ManxPower | RaYmAn-Bx, That is what ebay is for. |
| 14:26.00 | RaYmAn-Bx | indeed, but it's still a good thing to remember :P (in case you didn't know) |
| 14:26.17 | Leland | Orange give you free cinema tickets |
| 14:26.19 | RaYmAn-Bx | most phones sold here these days have 900/1800 and 1900 builtin though |
| 14:26.27 | Leland | not much use if you work on a tuesday afternoon though |
| 14:26.34 | _THEEND_ | someone uses musiconhold with cisco phones?! |
| 14:26.35 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 14:26.47 | Leland | _THEEND_: I have MoH with cisco 7960's |
| 14:26.56 | spackle | A good morning to all. |
| 14:27.11 | RPH | any further thoughts on my phone issue |
| 14:27.16 | *** join/#asterisk cgeek (~cgeek@pl6.lawrence.org.uk) |
| 14:27.41 | Leland | though MoH itself is not a function of the phone |
| 14:28.23 | RaYmAn-Bx | The only good thing about Orange is that they are present in most fairly large european countries...(and hence you pay their roaming charges which might be lower than using a diff company) |
| 14:28.34 | ManxPower | Networks GSM 1900 |
| 14:28.35 | ManxPower | Frequency band 1850 - 1910 MHz (TX) |
| 14:28.35 | ManxPower | 1930 - 1990 MHz (RX) |
| 14:28.48 | RaYmAn-Bx | no chance of that working here then |
| 14:28.52 | Zeeek | still haven't found out how to access the account I'm looking for :) |
| 14:28.55 | ManxPower | I didn't think so. |
| 14:29.56 | *** join/#asterisk X-Gen (~x-gen@rrba-146-121-118.telkomadsl.co.za) |
| 14:30.27 | ManxPower | Are there any pan-european providers? |
| 14:31.18 | nix000 | has anyoone done call support center setup with asterisk yet ? |
| 14:32.00 | Leland | Orange, T-Mobile |
| 14:32.01 | Zeeek | only a tri-band phone will work here, but it's cheaper to buy a $100 phone |
| 14:32.45 | ManxPower | Zeeek, not cheaper to buy a $10 phone on ebay? |
| 14:33.20 | *** join/#asterisk Aviator (~ask@ip-65-111-77-10.customer.accelacom.net) |
| 14:33.39 | spackle | LeLand: I like the fisher price phones, nobody can call me on them! |
| 14:33.44 | Leland | yea |
| 14:33.56 | Leland | though my son seems to think that I always have a phone call on his |
| 14:34.19 | Leland | not bad considering he's only 7 months old |
| 14:34.51 | bjohnson_ | I just gave my kids my old ones |
| 14:35.08 | jluk | wait another year and he'll happily chat away to the asterisk box :) |
| 14:35.09 | ManxPower | I sure would like to know why using ZapScan randomly disconnects my call, but no other calls are disconnected. |
| 14:35.43 | Leland | jluk: will give him a Grandstream to play with ... set up asterisk to respond using an auto-attendant no matter what he dials ;) |
| 14:35.51 | Leland | he'll get a kick out of that |
| 14:36.58 | Leland | hmm.. the Mrs. just brought me a big bowl of ice cream :) |
| 14:37.32 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-39-201.no.no.cox.net) |
| 14:38.02 | *** join/#asterisk Beirdo_ (~gjhurlbu@beirdo.user) |
| 14:40.21 | Zeeek | ManxPower - after a long and complete investigation, here is the answer: Yes, a 2m26s call received in Spain cost the cell user $2.10 ! |
| 14:40.33 | jhiver | eeeeewwww |
| 14:40.53 | Leland | I like being able to phone france for $0.0002 / min |
| 14:40.58 | jhiver | I've just realized that Digium's astcc app is written _without_ use warnings; and use strict; |
| 14:41.07 | Zeeek | IOW, it would be best to enter into a contact with attractive ceiling and roaming and then default on that contract |
| 14:41.32 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 14:41.35 | Zeeek | I phone FRanbce every day free |
| 14:42.02 | Leland | also like having a french (paris area) phone number even though I'm in the UK :D |
| 14:42.18 | Zeeek | what are you using for that? |
| 14:42.21 | jhiver | so you can impress your friends :) |
| 14:42.22 | Leland | so the other half's family can phone her local rate for them |
| 14:42.36 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 14:43.01 | LorenzoMarouani | Zeeek : quelle est ta société ? |
| 14:43.04 | Leland | Zeeek: not sure who my ITSP used to get the french local number .. but it's a "follow me" service which directs the call to my DID in the UK |
| 14:43.15 | Zeeek | k |
| 14:43.42 | Leland | though there's another company in france who claim to be able to do it as well.. |
| 14:43.50 | Zeeek | wengo is currently doing SIP it works with asterisk and for 8eu/mo unlimited national and a geo number |
| 14:43.54 | ManxPower | Zeeek, I would like to avoid screwing over a company on my first trip to Europe. |
| 14:44.04 | ManxPower | It's just not very nice. |
| 14:44.09 | Zeeek | then don't work for Enron |
| 14:44.13 | Leland | directcentrex they're called I think |
| 14:44.28 | afe | I was in Russia for 3 days, and when I got back and received my bill I found that my phone bill was about $500 for two hours of calling |
| 14:44.35 | Zeeek | ManxPower that was irony, not a true bit of advice |
| 14:44.44 | ManxPower | Zeeek, *nod* |
| 14:44.57 | Zeeek | if I were on my cell for two hours it would be at least that! |
| 14:44.57 | Leland | anyway..smoke break then back to work... |
| 14:45.11 | Zeeek | you're right to leave the room |
| 14:45.42 | Zeeek | I have to call the US later and talk to someone who will be paying minutes on their cell |
| 14:46.00 | afe | I pay less than 10c per minute for domestic calls on my cellular here |
| 14:46.15 | ManxPower | Zeeek, Pretty much all cell phones in the USA have to pay for incoming. |
| 14:46.31 | ManxPower | There are now some unlimited incoming plans, but they are still pretty uncommon |
| 14:46.34 | jhiver | you guys pay for incoming calls? how weird? |
| 14:46.50 | ManxPower | jhiver, it costs the same to call a landline as a cell phone in the USA. |
| 14:46.58 | jhiver | aaah I see |
| 14:47.03 | pascals | jhiver: that would be roughly equivalent to us paying for roaming within the eu... |
| 14:47.04 | ManxPower | i.e. if the cell number is local, the caller pays nothing |
| 14:47.26 | jhiver | mind you there's really no reason for which the price should be any different... |
| 14:47.33 | pascals | Oh. Calling cellphones is *much* more expensive than landlines |
| 14:47.44 | ManxPower | pascals, not in the USA. |
| 14:47.58 | `Sauron | Here it's the opposite |
| 14:48.03 | pascals | Somehow, we're both getting screwed :D |
| 14:48.05 | jhiver | I'm sure there is less capital invested in mobile phones than originally spent on 'traditional' phone lines... |
| 14:48.37 | ManxPower | The whole "caller pays" is what caused cell phone usage to be so popular in the EU and dialup internet so unpopular in the EU |
| 14:48.39 | pascals | Most traditional lines were installed with tax payers money... |
| 14:49.04 | jhiver | yep |
| 14:49.34 | jhiver | so maybe the cell infrastructure should have been done the same way :) |
| 14:50.16 | jluk | jhiver: wot and have to go through LLU with the cell infra as well ? |
| 14:50.31 | SwedMiro | In sweden there are even companies that pay you for getting phonecalls |
| 14:50.31 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) |
| 14:50.31 | jhiver | I really fail to see why cells should be more expensive than landline... |
| 14:51.06 | jluk | jhiver: cos they're newer and the intrastructure's still being paid for. |
| 14:51.06 | ManxPower | jhiver, Well part of it might be that you can just run more cable for landlines, but there's a limited amount of radio spectrum. |
| 14:51.26 | jhiver | yeah right |
| 14:51.40 | jhiver | that's why they have 3G superfast wireless data in Japan |
| 14:52.00 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 14:52.01 | jhiver | it's well known that Japanese radio spectrum is broader than the rest of the world's |
| 14:52.13 | SwedMiro | hehe..the word "superfast" is a bit of an exhageration |
| 14:52.18 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 14:52.34 | ManxPower | "Not quite as dog slow as the old service" is better? |
| 14:52.40 | jhiver | lol |
| 14:52.43 | SwedMiro | hehe |
| 14:52.48 | SwedMiro | yupp |
| 14:53.45 | LorenzoMarouani | is there a devel channel ? |
| 14:54.08 | jhiver | that's the problem when you have only a few big companies with just a few very expensive licenses |
| 14:54.15 | jluk | asterisk-dev but it tends to be quiet |
| 14:54.26 | LorenzoMarouani | ok thanks ! |
| 14:54.28 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 14:54.35 | Nugget | I wish there was an #asterisk-common-guys-cant-you-just-tell-me-what-to-type and an #asterisk-too-lazy-to-read-the-wiki to take this channel's flotsam. |
| 14:54.46 | jhiver | lol |
| 14:54.49 | ManxPower | at least in the USA the govt sells the spectrum for LOTS of money. |
| 14:55.07 | jhiver | #asterisk-do-my-work-and-order-beer-and-pizza would be nice |
| 14:55.15 | jhiver | in france it's been ridiculous |
| 14:55.23 | jluk | ManxPower: No where near as much as the UK sold the 3G stuff for. |
| 14:55.29 | jhiver | they have done it "on study" rather than auction |
| 14:55.29 | SwedMiro | Same in sweden..or..they gave them away for a promise that the company would cover 98% of swedens population |
| 14:55.50 | jhiver | results: the germans made lots of money, the french govt basically gave its ass |
| 14:55.59 | Beirdo | Nugget: how about a #asterisk-the-wiki-is-down for yesterday? |
| 14:56.20 | Bob_ | I am using sip and softphones with Asterisk. I want any user on the internet to be able to dial in. What do I specify in the sip.conf file to allow this/ |
| 14:56.34 | jhiver | I think it would be simpler to go for 'free for all' style and then let operators sort themselves out |
| 14:57.09 | `Sauron | Nugglet |
| 14:57.17 | Nugget | Moo |
| 14:58.01 | Beirdo | Moo |
| 14:58.22 | `Sauron | Moof |
| 14:58.26 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 14:58.27 | eKo1 | Everyone is turning into a cow!? |
| 14:58.31 | Nugget | ]:8) |
| 14:58.34 | jhiver | moo? |
| 14:58.38 | `Sauron | not a cow |
| 14:58.42 | `Sauron | a DOGcow |
| 14:58.49 | Essobi | meeeeow |
| 14:58.58 | Essobi | ahem.. I mean, moooooo. |
| 14:59.17 | *** join/#asterisk santiago (~santiago@63.245.86.89) |
| 14:59.23 | `Sauron | http://developer.apple.com/technotes/tn/tn1031.html |
| 14:59.29 | *** part/#asterisk santiago (~santiago@63.245.86.89) |
| 15:02.09 | spackle | Essobi: working on H323? |
| 15:02.27 | Essobi | ehh |
| 15:02.49 | spackle | Essobi: yesterday you said you had oh323 working, did you find significant differences in scalability? |
| 15:03.26 | *** join/#asterisk scrubb (~scrubb@OCI-19-41.onecall.net) |
| 15:05.04 | *** join/#asterisk Flash_ (~Flash_@pc-80-235-133-235-na.blueyonder.co.uk) |
| 15:05.08 | jhiver | okay guys, silly n00bish question time |
| 15:05.22 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 15:05.35 | jhiver | is there a way to know, in a given context or macro, what the user is? |
| 15:05.49 | jhiver | you know like a cool ${CURRENT_USER} variable |
| 15:06.06 | jontow | (huh?) |
| 15:06.24 | eKo1 | jhiver: What do you mean by user? |
| 15:06.34 | spackle | jhiver, does cid count? |
| 15:06.42 | jhiver | well say in your iax.conf you have |
| 15:06.46 | jhiver | [my-mum] |
| 15:06.50 | jhiver | (some config) |
| 15:06.53 | jhiver | context => foo |
| 15:07.01 | jhiver | and in your extensions.conf |
| 15:07.07 | jhiver | [foo] |
| 15:07.33 | jontow | http://www.voip-info.org/wiki-Asterisk+variables |
| 15:07.42 | jhiver | GotoIf(${CURRENT_USER} = my-mum? <something> : <something-else>) |
| 15:07.58 | jontow | look there and see if you can find what you want |
| 15:08.07 | jhiver | ok |
| 15:08.28 | jhiver | at the moment I do something horrible: I write a context for each user |
| 15:08.33 | jontow | perhaps you want accountcode ? |
| 15:08.49 | jhiver | mind you it's not too much work since I have a template doing that job for me |
| 15:09.44 | *** join/#asterisk lidl (~little@213-140-6-96.fastres.net) |
| 15:09.58 | *** join/#asterisk DT-V (~sjaaknabu@fia254-108-100.dsl.mxposure.nl) |
| 15:10.05 | jhiver | mhhh I wonder if ${CALLERID} or ${CALLERIDNAME} would do it |
| 15:10.14 | jhiver | can you set those in iax.conf and sip.conf? |
| 15:10.22 | lidl | the wildcard x100p works as an FXO (let's say as a TDM400 with 1 fxo), right? |
| 15:10.50 | jhiver | lidl, yes |
| 15:10.52 | DT-V | how can i set h.323 tracing |
| 15:11.12 | DT-V | asked for a number but 1 till 10 I tryed but nothing happend |
| 15:11.23 | jhiver | in other word, you plug in in your phone plug |
| 15:11.28 | lidl | is the x100p so suitable for europe? |
| 15:11.35 | jhiver | yeah I'm using one |
| 15:11.39 | DT-V | also h.323 debug, with no results |
| 15:11.41 | jhiver | well I'm using a clone actually |
| 15:11.52 | jhiver | (I know, bad me) |
| 15:11.57 | lidl | jhiver, i could get 1 for 7$ |
| 15:12.13 | jhiver | oh that's pretty cool :) |
| 15:12.40 | jluk | jhiver: since oyu can't buy X100p from digium you've no choice |
| 15:12.46 | jhiver | I'm bought my X100P clones like 50$ for 2 |
| 15:12.56 | lidl | +25$ of shipping -.- |
| 15:13.04 | jhiver | why can't you? they don't do them ne more? |
| 15:13.18 | jluk | jhiver: nope no longer made/sold |
| 15:13.35 | lidl | jluk, but working anyway with recent asterisk, right? |
| 15:13.35 | jhiver | wow so now it's like TDM or nuthin' ? |
| 15:13.55 | jluk | lidl: yep |
| 15:14.19 | jluk | jhiver: yep, but still not authorised for connection to the uk pstn |
| 15:14.40 | jhiver | well screw it and plug it already :) |
| 15:14.58 | Flash_ | Mar 16 15:07:25 NOTICE[8434]: chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to '<sip:626311@fwd.pulver.com>;tag=as34564a99' |
| 15:15.00 | lidl | jluk, how about italy? do you know if they're accepted? |
| 15:15.09 | spackle | anybody floating around here get a mediatrix product working with * ? |
| 15:15.55 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 15:15.58 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 15:16.23 | jhiver | Flash_ are u behind NAT? |
| 15:16.25 | jluk | lidl: don't know the rules in Italy |
| 15:16.33 | Hairlocks | lidi I read somewhere that the TDM FXO does not have the CE mark yet |
| 15:17.01 | nix000 | is asterisk part of debian//ubuntu yet ? |
| 15:17.10 | jhiver | yep |
| 15:17.13 | eKo1 | Is there a way to make * not challenge SIP invites from my provider? |
| 15:17.17 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 15:17.18 | Flash_ | jhiver: I am |
| 15:17.18 | jhiver | apt-get install asterisk and whamo! |
| 15:17.32 | jluk | Hairlocks: that makes it illegal for use in EU I tihnk. |
| 15:17.35 | CosmicRay | nix000: asterisk was even in debian *stable* |
| 15:17.38 | jhiver | you might have to recompile zaptel drivers though |
| 15:17.49 | jhiver | Flash_ what SIP phone are you using? |
| 15:17.51 | lidl | Hairlocks, this is strange, since without the CE mark digium TDM cards couldn't be sold at all in europe |
| 15:17.55 | jhiver | has it got options for NAT? |
| 15:17.58 | TheBear | is it wise/safe to remove all non us from the indications.conf file ? |
| 15:18.02 | jhiver | like keep-alive, nat mapping, etc |
| 15:18.14 | jluk | lidl: they can be sold but not used :) |
| 15:18.28 | jhiver | Ah bollocks it just works fine |
| 15:18.30 | lidl | jluk, :) |
| 15:18.40 | Hairlocks | I read it was just the FXO module the FXS have passed |
| 15:18.41 | nix000 | CosmicRay: cool ... but the latest release is not in there i would imagine |
| 15:18.48 | CosmicRay | no, of course not :-) |
| 15:18.49 | jhiver | it's 1.05 i think |
| 15:18.54 | CosmicRay | sid has 1.0.6 |
| 15:18.57 | CosmicRay | testing has 1.0.5 |
| 15:18.58 | DT-V | H323 geeks here ??? |
| 15:19.01 | jhiver | still not too bad... |
| 15:19.29 | jhiver | never did anything with h323 apart for when I was using netmeeting :) |
| 15:19.36 | TheBear | is it wise/safe to remove all non [us] from the indications.conf file? I only intend to call and receive US calls no international |
| 15:19.48 | nix000 | CosmicRay: is in it the other way ? testing should be at a higher version |
| 15:19.56 | CosmicRay | nix000: sid is unstable |
| 15:20.10 | nix000 | CosmicRay: ok |
| 15:20.12 | Flash_ | jhiver: not got a phone just yet.... just trying to get * to connect to the remote sip service.. |
| 15:20.12 | Hairlocks | I have been looking for days for somewhere I can buy the FXO in the UK with no luck! |
| 15:20.13 | PatrickDK | thebear, indications doesn't do that |
| 15:20.16 | DT-V | let's try how to set trace on |
| 15:20.21 | DT-V | or debug |
| 15:20.22 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 15:20.24 | PatrickDK | all it does it make different sounds for dialtone ringtone and stuff |
| 15:20.24 | jhiver | aaah well then I don't know man |
| 15:20.45 | jhiver | maybe you should try to google site:voip-info.org asterisk sip.conf nat |
| 15:20.46 | jluk | Hairlocks: direct from digium os the only way I know of to get a TDO |
| 15:21.04 | jluk | TDM FXO I meant |
| 15:21.11 | TheBear | PatrickDK: the indications.conf file has [us] [fr] [au] etc... |
| 15:21.12 | Hairlocks | jluk do they ship to the UK? |
| 15:21.24 | PatrickDK | thebear, like I said, that doesn't do anything about calls |
| 15:21.31 | Essobi | I need a poe switch |
| 15:21.31 | PatrickDK | all it does it make different sounds |
| 15:21.31 | jluk | Hairlocks: yes, I know a few people that have bought direct. |
| 15:21.33 | Essobi | :| |
| 15:21.46 | PatrickDK | essobi, hmm, I have two poe switchs, they are nice |
| 15:21.49 | Essobi | I need a 12 port POE.. |
| 15:21.50 | TheBear | PatrickDK: so I can remove them being in the US I don't need [fr] or [au] sounds |
| 15:21.55 | PatrickDK | yep |
| 15:21.58 | TheBear | thanks |
| 15:22.01 | jhiver | hey, the sipuras do have the CE logo on them |
| 15:22.09 | *** part/#asterisk LorenzoMarouani (~LorenzoMa@i01v-50-28.d4.club-internet.fr) |
| 15:22.12 | Essobi | Or something moderately small.. how much you pay PatrickDK ? |
| 15:22.13 | jhiver | u can use them as FXO if it's a sipura 3k |
| 15:22.18 | jluk | jhiver: yep. |
| 15:22.22 | PatrickDK | essobi, $1400 |
| 15:22.29 | Essobi | how many ports? |
| 15:22.32 | jhiver | and then it's nice 'cause you get to keep a nice pure voip asterisk install |
| 15:22.32 | PatrickDK | 24 |
| 15:22.34 | Essobi | Baaah. |
| 15:22.37 | Essobi | a piece? |
| 15:22.43 | PatrickDK | heh? |
| 15:22.48 | PatrickDK | ya, $1400 each switch |
| 15:22.55 | Essobi | 24 ports each or total |
| 15:23.02 | PatrickDK | 24ports per switch |
| 15:23.07 | Essobi | tehe.. Damn. |
| 15:23.16 | Essobi | I need like 12 ports for my Lab. |
| 15:23.32 | PatrickDK | hmm, you shouldn't need a poe switch, unless you are using 2pair 100mb |
| 15:23.36 | Essobi | I've ran out of power strips and plugs.. and POE would make setting up a lab much easier. |
| 15:23.42 | PatrickDK | you can use an injector if you use 4pair |
| 15:23.51 | Essobi | Ehh? |
| 15:24.00 | Essobi | How much are injectors? |
| 15:24.04 | PatrickDK | dunno |
| 15:24.10 | TheBear | Using a snom 200 which has multiple lines, is it a good thing to have pedantic = yes in the sip.conf file? |
| 15:24.37 | jluk | TheBear: don't know - what does pedantic=yes do ? |
| 15:24.38 | Essobi | I got a ton of 7960's I need to pull out of the graveyard and cannabalize.. |
| 15:24.56 | PatrickDK | hmm, 7960 doesn't do poe |
| 15:25.05 | PatrickDK | it uses a backwards cisco standard |
| 15:25.06 | Shido6 | err |
| 15:25.20 | Essobi | PatrickDK An inverted cable takes care of that. |
| 15:25.34 | ManxPower | 7960 doesn't do IEEE PoE, it does Cisco PoE |
| 15:25.35 | *** join/#asterisk syslod (~yurplsl@65.114.15.70) |
| 15:25.48 | syslod | Hello |
| 15:25.53 | Essobi | http://www.voip-info.org/wiki-Cisco%20POE |
| 15:25.54 | Essobi | :) |
| 15:26.03 | jhiver | hi |
| 15:26.08 | PatrickDK | heh, 4 pair, I'm to used to using 2 pair poe |
| 15:26.42 | Essobi | Right right.. this is 4 pair poe.. so I can swap polarity without blinking. |
| 15:27.00 | PatrickDK | I doubt that would work with my poe switch at all |
| 15:27.02 | Bob_ | Is it correct that if I want to use Meetme I either need a piece of hardware (Zaptel car) or I need to enable ztdummy? |
| 15:27.23 | syslod | Anyone know how to fix * side echo only in the following setup * -> Digium T1 card -> channelbank -> analog line -> CO -> analog line -> human? |
| 15:27.32 | Essobi | Anyways.. 568b<POE> to 4/7/5/8 reversed on 7960 side.. and you're money. |
| 15:27.48 | PatrickDK | essobi, only if you use an injector |
| 15:27.50 | Essobi | PatrickDK 4 port or 2 port POE switch? |
| 15:27.54 | PatrickDK | if you used a poe swich, it wouldn't work |
| 15:28.17 | Flash_ | jhiver: did you say somthing in reply to my comment on phone ? |
| 15:28.19 | PatrickDK | heh? 4port 2port? |
| 15:28.20 | Essobi | Ahh, true enough.. or I could just use a Cisco POE switch. ;) |
| 15:28.28 | jhiver | yeah... |
| 15:28.31 | jhiver | aaah well then I don't know man |
| 15:28.33 | Essobi | s/port/wire |
| 15:28.34 | jhiver | maybe you should try to google site:voip-info.org asterisk sip.conf nat |
| 15:28.55 | PatrickDK | a poe switch can do poe over 4pair or 2pair, but it autodetects |
| 15:28.56 | Flash_ | kk |
| 15:29.08 | PatrickDK | injectors can only do 4pair, and or normally dumb, always injecting power |
| 15:29.52 | PatrickDK | my poe switch turns off the power, like if you have a computer plugged into it |
| 15:30.05 | *** join/#asterisk Lairsdragon (~steve@80.146.165.65) |
| 15:30.42 | Essobi | I'll have to check, but I thought they switched to standard poe on the 7960G models, and the non G's were the inverted POEs.. |
| 15:31.08 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 15:31.23 | PatrickDK | ya, G are the good ones for poe |
| 15:31.33 | PatrickDK | did you read the link someone just posted about it? |
| 15:32.01 | Essobi | Anope. |
| 15:32.11 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:32.12 | Essobi | So G's support standard POE and non's dont? |
| 15:32.22 | Essobi | Half my test phones are Gs. |
| 15:32.35 | Bob_ | Does anyone know if asterisk can be used with the Microsft RTC core? |
| 15:32.39 | Essobi | they quit making standards.. everything is a G now.. |
| 15:32.45 | Essobi | Bob A what what? |
| 15:32.48 | _Brian | Does anyone know of a card that is available for a standard 4 wire T1 DID covering start protocols: immediate, ground, loop, wink options, framing d4/ESF, encoding AMI/B8ZS.....all DID information would be sent 48V |
| 15:33.06 | Essobi | ?! |
| 15:33.09 | Essobi | 48V? |
| 15:33.14 | Bob_ | It is basically the control library which MSN messenger is built on |
| 15:33.16 | ManxPower | Bob_, No. |
| 15:33.25 | Essobi | AMI is a T1.. |
| 15:33.30 | Bob_ | not - are you sure - even if using sip |
| 15:33.33 | ManxPower | _Brian, none that I know. |
| 15:33.34 | _Brian | Yes, 48VDC |
| 15:33.37 | Essobi | Bob Old versions of MSN do sip thou.. |
| 15:33.56 | ManxPower | _Brian, none that I know of for Analog. Any Digium T-1/E-1 card will support all those. |
| 15:34.11 | _Brian | sad part is that this is analog :( |
| 15:34.32 | Essobi | Yea, with winking.. what is it? A pbx trunk? |
| 15:34.37 | ManxPower | _Brian, Then someone is lieing to you. analog doesn't do framing d4/ESF, encoding AMI/B8ZS |
| 15:34.38 | syslod | Anyone else have any echo problems when going T1 to analog? |
| 15:34.43 | Bob_ | Because I have built an app which uses sip and would need to dial into a conference extension - it works with SER, but not sure if it would work with Asterisk |
| 15:34.49 | *** join/#asterisk r3d5un (~r3d5un@83-64-243-19.dynamic.xdsl-line.inode.at) |
| 15:35.15 | Essobi | Well, of course it would. |
| 15:35.18 | Essobi | Sip is SIP. |
| 15:35.30 | Bob_ | I suppose you are right |
| 15:35.38 | _Brian | he is standing next to me......let me get this 100% correct.... |
| 15:36.11 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 15:36.12 | _Brian | the DID and the framing on the T1 are two different things, if it can handle both, then it would be the best of both worlds. |
| 15:36.14 | *** join/#asterisk fcsirc (~Administr@cvl232019.columbus.rr.com) |
| 15:36.27 | ManxPower | _Brian, Is the interface a T-1/E-1 |
| 15:36.46 | _Brian | ManxPower: yes, that would be the interface |
| 15:36.58 | ManxPower | _Brian, then any Digium T-1/E-1 card will support all that. |
| 15:37.06 | _Brian | ManxPower: sorry for being unclear...... |
| 15:37.22 | *** part/#asterisk fcsirc (~Administr@cvl232019.columbus.rr.com) |
| 15:37.23 | TheBear | jluk: sorry got a call. they say it helps with multilines |
| 15:37.27 | *** join/#asterisk fcsirc (~Administr@cvl232019.columbus.rr.com) |
| 15:37.27 | Essobi | Now if the winking isn't a standard.. you might have to do some hacking to get all the winking you want supported. |
| 15:37.35 | _Brian | ManxPower: does it source the 48V? |
| 15:37.39 | ManxPower | E&M/Wink works just fine. |
| 15:37.48 | Essobi | I know there's been some questions reguarding E&M lately. |
| 15:37.50 | Essobi | :) |
| 15:38.05 | TheBear | jluk: Enable slow, pedantic checking for Pingtel and multiline fotmatted headers for strict SIP compatibility defaults to no |
| 15:38.05 | Bob_ | I want any client on the internet to be able to dial in using sip - what do I specify in the sip.conf file? I also do not want users to be authenticated |
| 15:38.13 | Essobi | E&M originating has a bug currently.. went up on mantis 2 days ago I think.. |
| 15:38.19 | ManxPower | _Brian, I don't know the voltage, but it works with pretty much any T-1/E-1 line except for R2 and you can get R2 support for some countries via a 3rd party. |
| 15:38.30 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 15:38.36 | ManxPower | Essobi, bug #? |
| 15:38.54 | *** join/#asterisk sariabod (~sariabod@66.238.64.6) |
| 15:38.57 | _Brian | ManxPower: thanks for your help..... |
| 15:39.07 | *** part/#asterisk fcsirc (~Administr@cvl232019.columbus.rr.com) |
| 15:39.07 | Essobi | I don't know.. Km- and ... someone else in here were discussing it.. and |
| 15:39.15 | ManxPower | BrianR___, I have T-1 -> Asterisk -> Channel bank -> PBX |
| 15:39.28 | Essobi | Umm.. Tzanger maybe documented it with km I'm thinking... |
| 15:39.37 | Essobi | but don't quote me on that.. |
| 15:39.46 | tzanger | eh? |
| 15:39.48 | cjk | hi, what is mutlicast transfer in a tftpd configuration? |
| 15:39.51 | cjk | anyone an idea? |
| 15:40.01 | ManxPower | cjk, nohing you have to worry about. |
| 15:40.05 | Essobi | was that you working on e&m with km? |
| 15:40.07 | tzanger | yes |
| 15:40.10 | spackle | anybody floating around here get a mediatrix product working with * ? |
| 15:40.16 | Essobi | did that bug get filed? |
| 15:40.17 | cjk | ManxPower, so i can set any ip fuck i like? |
| 15:40.21 | tzanger | asterisk does not originate calls with E&M wink trunks properly as far as I can tell |
| 15:40.24 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 15:40.28 | tzanger | there is certainly no code to look for the terminating side wink |
| 15:40.35 | tzanger | asterisk does correctly wink when terminating though |
| 15:40.37 | Essobi | wink wink. |
| 15:40.41 | ManxPower | cjk, Well, since neither Asterisk, not any phone I know of suports Multicast..... |
| 15:41.16 | ManxPower | tzafrir, I have Telco (E&M/Wink) -> Asterisk -> Cannel Bank (E&M/Wink) |
| 15:41.25 | cjk | ManxPower, yeah but this shitty install scripts asks me for it and i do not know what it is and havent found much details either |
| 15:41.29 | TheBear | in extensions.conf if I have [line1] _1XXXXXXXXXX,1,Dial(.... and then in [line2] _1XXXXXXXXXX,1,Dial(..... and I include line1 and line2, will it try the call on line1 and then if that doesn't work then try on line2 ? |
| 15:41.30 | ManxPower | tzanger, I have Telco (E&M/Wink) -> Asterisk -> Cannel Bank (E&M/Wink) |
| 15:41.46 | *** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net) |
| 15:41.48 | ManxPower | TheBear, no. You cannot have overlapping exten/priorities |
| 15:42.01 | tzanger | ManxPower: when asterisk makes a call to the telco what state is the zap channel in after you dial? |
| 15:42.28 | TheBear | ManxPower: how then can I set it up to roll over should it not go through on line1 |
| 15:42.30 | tzanger | ManxPower: I may be blind but there is *no* code in chan_zap.c or zaptel.c to deal with waiting for far-end wink |
| 15:42.39 | eKo1 | Is there a way to make * not challenge SIP invites from my provider? |
| 15:42.56 | ManxPower | tzanger, our DID trunks are incoming only. |
| 15:42.56 | Lairsdragon | cjk: set the mcast ip to 239.239.239.0-255 |
| 15:42.59 | Essobi | JerJer[mobile] Heh.. I never did get chan_h323 to let me playback, or Moh yesterday. :| Allow=gsm, or all, or anything of the like.. Just got those "Asked to transmit frame " non-native format errors.. |
| 15:43.05 | tzanger | ManxPower: ahh well you don't see the problem then :-) |
| 15:43.11 | ManxPower | But when sending a call to the channel bank using E&M/Wink nothing weird happens |
| 15:43.31 | tzanger | and I am willing to bet that your Asterisk-ChannelBank is fine for some toher reason |
| 15:43.34 | tzanger | brb |
| 15:43.35 | ManxPower | Essobi, allow=all will break lots of stuff. |
| 15:44.16 | Essobi | I tried disabling it too.. and just turning on ulaw,alaw,729,and gsm.. still no playback/moh in chan_h323.. |
| 15:44.40 | ManxPower | Essobi, When you troubleshoot it is stupid to enable many codecs. disallow=all and allow=ulaw. |
| 15:44.46 | Essobi | *SHRUG* I'll piss with it some more after hours... or if I get a dev box up given to me anything.. |
| 15:45.16 | Essobi | ManxPower I did that and had no MOH or playback() |
| 15:45.31 | Essobi | ManxPower Jerjer recommended allow=gsm.. *SHRUG* |
| 15:45.34 | *** join/#asterisk ronn (ronn@host217-46-199-165.in-addr.btopenworld.com) |
| 15:46.00 | eKo1 | Wow, I have 46 active channels and 40 of them are ghosts. |
| 15:46.00 | ManxPower | Essobi, The key is to only enable ONE codec when trying to troubleshoot. |
| 15:46.36 | Damin | disallow=all |
| 15:46.57 | Essobi | 711 phone to phone with chan_h323 worked,, inbound and outbound.. passed dtmf, etc... just had native format warnings with app_playback, and moh. |
| 15:47.01 | Damin | allow=lpc10 |
| 15:47.22 | Essobi | Damin: Ehh? |
| 15:47.24 | Damin | That's it.. |
| 15:47.33 | Damin | I'm using lpc10 for all my IAX trunks! :) |
| 15:47.42 | Essobi | Oh. ;) |
| 15:48.30 | *** join/#asterisk ronn (ronn@host217-46-199-165.in-addr.btopenworld.com) |
| 15:49.24 | TheBear | ManxPower: how then can I set it up to roll over should it not go through on line1 |
| 15:50.58 | Lairsdragon | Could I use a Cisco AS5300 aus Dialin/Dialout GW for * |
| 15:51.05 | Bentley | hi all. on a TE410P which span is closest to the motherboard: #1 or #4? |
| 15:54.09 | *** join/#asterisk k_bot (~mejii@hosted-pat.7ticks.com) |
| 15:54.32 | Darwin35 | ~kiss k_bot |
| 15:54.34 | jbot | ACTION smooches k_bot on the neck |
| 15:54.47 | k_bot | oh jbot |
| 15:54.53 | k_bot | down down |
| 15:55.08 | riksta | mount /jbot |
| 15:55.09 | k_bot | what's with all the friendlies |
| 15:55.22 | Darwin35 | ~mount k_bot |
| 15:55.27 | k_bot | frisky in here this morning |
| 15:56.00 | Darwin35 | hehhe |
| 15:56.18 | Lairsdragon | hot show here :-) |
| 15:56.34 | k_bot | since we are all in such a great mood, I was wondering if some could recommend a great SIP provider |
| 15:56.54 | *** join/#asterisk phocus (~phocus@67.32.20.65) |
| 15:56.55 | k_bot | somewhere in chicago |
| 15:57.29 | Darwin35 | flatrate or per min |
| 15:57.34 | Darwin35 | with free incoming |
| 15:57.46 | Darwin35 | sip or iax |
| 15:57.48 | k_bot | sip |
| 15:57.50 | MikeJ[Laptop] | I know of a great dev conference called clue con, that will be in chicago Aug 3rd-5th |
| 15:57.55 | k_bot | could be flatrate or per min |
| 15:58.12 | phocus | I know this is really really stupid question, but i dont know much about voip workings yets.....is it possible, to use a single regluar modem, and and itnernet connection, and asterisk to build a voip phone, with no other hardware needed? |
| 15:58.20 | MikeJ[Laptop] | but that has nothing to do with an isp, I was just saying |
| 15:58.36 | Darwin35 | they might no be locacted in CHi but have did/pstn nmbrs for chicago |
| 15:58.41 | eKo1 | phocus: What do you need a modem for? |
| 15:58.53 | Darwin35 | broadvoice voicepulse teleiax |
| 15:58.55 | jontow | phocus; if you plan to make phone calls over the dialup, it'll be difficult.. |
| 15:59.01 | k_bot | just concerned about latency |
| 15:59.02 | phocus | to hook the phone to |
| 15:59.05 | Darwin35 | faxing in some cases |
| 15:59.08 | *** join/#asterisk j_berry (~jberry@acs-24-154-150-174.zoominternet.net) |
| 15:59.15 | phocus | i have 8meg cable modem |
| 15:59.24 | k_bot | anybody had any experience with broadvoice, voicepulse or teleiax? |
| 15:59.33 | eKo1 | phocus: OK, that would require some major h4x0ring |
| 15:59.41 | *** part/#asterisk j_berry (~jberry@acs-24-154-150-174.zoominternet.net) |
| 15:59.43 | phocus | eKo1 what would? |
| 16:00.14 | phocus | eKo1 Maybe you can explain what all hardware is needed to build a system |
| 16:00.20 | *** join/#asterisk j_berry (~jberry@acs-24-154-150-174.zoominternet.net) |
| 16:00.38 | eKo1 | phocus: Maybe you should just buy an ATA. |
| 16:00.46 | *** join/#asterisk VerCom (~ron_hartm@ppp-69-213-251-59.dsl.dytnoh.ameritech.net) |
| 16:00.47 | phocus | ata? |
| 16:00.55 | jontow | yeah, for $75-$100 you can get a single or dual port analog telephone adapter (ATA) |
| 16:00.57 | eKo1 | ~ATA |
| 16:00.58 | jbot | from memory, ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
| 16:01.11 | jontow | maybe cheaper, but thats the range i've been shopping in |
| 16:01.33 | phocus | ok, is this better than using a pre-built servers like vonage? |
| 16:01.36 | VerCom | good day all.... Getting ready to fire up a PRI and provider said i need to set the isdn flagg to unkown... any info on how to do this I have googled |
| 16:01.42 | *** part/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 16:02.05 | jluk | phocus: what is it you actually want to do ? |
| 16:02.34 | phocus | jluk i have no clue, i use vonage, but thought it would be cool if i could built my own system, so i started searching and found http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Ada |
| 16:02.42 | phocus | sorry ,i mean i found astrick |
| 16:02.42 | VerCom | they said that when i send them the 10 digit number i need to set the ISDN Flag to "unknown" |
| 16:03.14 | VerCom | so their switch determines how to proceed with the number |
| 16:03.39 | jluk | It will be kind of difficult to advise you on hardware when you don't know what you want it to do. |
| 16:03.39 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmu6.dialup.mindspring.com) |
| 16:03.59 | jluk | Start at the wiki and decide what you want. |
| 16:04.02 | jluk | ~wiki |
| 16:04.34 | jluk | ~voip-info |
| 16:04.35 | jbot | voip-info is, like, the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 16:04.35 | VerCom | does anyone understand what this person is asking me to do |
| 16:05.07 | shaZwaz | :) |
| 16:05.21 | jhiver | is there a way I can tell asterisk to lauch the Perl debugger when I'm on the CLI for AGI scripts? |
| 16:05.23 | jontow | vercom; sounds like they do NOT want you doing PRI_CPE |
| 16:05.32 | jontow | meaning you do not do the signalling, they do |
| 16:05.46 | VerCom | what should i change it to |
| 16:06.03 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 16:06.11 | jontow | signalling=... is set in zapata.conf |
| 16:06.14 | cjk | damn this atftpd sucks. im able to confgure * but not that stupif tftpd deamon |
| 16:06.21 | riksta | anyone in the uk using call1899 ? |
| 16:06.33 | phocus | thanks guys, i have a lot of reading to do |
| 16:06.42 | jontow | another thing: |
| 16:06.43 | jontow | ; PRI Dialplan: Only RARELY used for PRI. |
| 16:07.27 | jontow | if its PRI signalling and they don't want you doing it.. then it is pri_net |
| 16:07.32 | jontow | ; pri_cpe: PRI signalling, CPE side |
| 16:07.32 | jontow | ; pri_net: PRI signalling, Network side |
| 16:07.46 | jontow | but if its part of the dialplan, you need to view earlier in the sample file |
| 16:08.27 | TheBear | I'm trying to get out going calls to rollover from teliax, to iaxtel or then zap/1. I have pasted my extensions.conf at http://pastebin.ca/7571 if someone could please help, thanks |
| 16:10.38 | VerCom | its pri dialplan |
| 16:11.05 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 16:12.28 | VerCom | jontow would PRI Dialplan be a zap..conf or exten...conf |
| 16:12.46 | j_berry | Is there a way I can make asterisk not pick up on the first ring? I'm using it at home with a single x100p card. |
| 16:13.16 | *** join/#asterisk enemy^x (lkqw@c213-158-233-150.sdsl.no) |
| 16:13.22 | j_berry | I want to give someone the chance to pick up an anolog phone somewhere in the house |
| 16:13.27 | shaZwaz | use wait(secs) |
| 16:13.28 | jontow | zapata.conf |
| 16:13.52 | j_berry | thank you |
| 16:13.55 | enemy^x | Im trying to set up musiconhold using shoutcast, but it seems like I only get alot of noise. is my mpg123 broken or something? |
| 16:14.02 | shaZwaz | I am not sure ... but try |
| 16:14.07 | VerCom | thanks i go read |
| 16:14.37 | eKo1 | It appears that the SIP stack on these Clipcomms suck big-floppy-donkey dick. |
| 16:15.06 | enemy^x | it's the same if I try to play regular mp3 files actually |
| 16:16.48 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 16:17.43 | Jearil | hmm, anyone ever had a Polycom SoundPoint IP-500 freeze up when you dialed it via asterisk? |
| 16:18.21 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 16:18.41 | Lairsdragon | Could I use a Cisco AS5300 aus Dialin/Dialout GW for asterisk? |
| 16:20.04 | r3d5un | are you sure you are using mpg123 and not mpg321 |
| 16:21.05 | Darwin35 | <PROTECTED> |
| 16:21.06 | Darwin35 | <PROTECTED> |
| 16:21.06 | Darwin35 | <PROTECTED> |
| 16:21.06 | Darwin35 | Mar 16 11:20:35 WARNING[901]: chan_sip.c:1400 create_addr: No such host: |
| 16:21.06 | Darwin35 | Mar 16 11:20:35 NOTICE[901]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' |
| 16:21.14 | bkw_ | no such host |
| 16:21.16 | bkw_ | NEXT!!! |
| 16:21.23 | Darwin35 | wich host |
| 16:21.31 | bkw_ | its null |
| 16:21.33 | bkw_ | @ |
| 16:21.38 | bkw_ | you don't have anything after the @ |
| 16:21.39 | Lairsdragon | cjk, why not using the plain tftpd? I had also trouble with atftpd because of inresponsive after a while |
| 16:21.40 | bkw_ | NEXT!!! |
| 16:21.44 | Skysky | I wonder if anyone are managing more than 1 asterisk server here? |
| 16:21.57 | bkw_ | Skysky, yes why? |
| 16:22.13 | r3d5un | enemy, are you using mpg123 or mpg321 |
| 16:22.38 | Darwin35 | grrr |
| 16:22.48 | Darwin35 | this who thing is going south fast |
| 16:23.39 | Darwin35 | ;US & Canada |
| 16:23.39 | Darwin35 | exten => _NXXNX.,1,SetCallerID(${BVLINE}) |
| 16:23.39 | Darwin35 | exten => _NXXNX.,2,Dial(SIP/${EXTEN}@${BVCOMPANY}) |
| 16:23.39 | Darwin35 | exten => _NXXNX.,3,Congestion() |
| 16:23.39 | Darwin35 | exten => _NXXNX.,4,Hangup |
| 16:23.51 | tzanger | ?? |
| 16:23.55 | tzanger | why are you using NXXNX.? |
| 16:24.11 | enemy^x | r3d5un: what do you recommend? |
| 16:24.12 | Darwin35 | have to use 10 digit dialing in PA |
| 16:24.13 | tzanger | what's wrong with NXXNXXXXXX... why variable? |
| 16:24.18 | tzanger | Darwin35: so spell out 10 digits |
| 16:24.23 | *** join/#asterisk |Vulture| (~Vulture@c-24-129-110-157.se.client2.attbi.com) |
| 16:24.24 | tzanger | not "5 or more" |
| 16:24.47 | Darwin35 | thats not the issue at the min |
| 16:24.50 | |Vulture| | hey guys anyone here know about PRIs? |
| 16:24.54 | tzanger | yeah what's up |
| 16:25.05 | Darwin35 | the issue is hostname |
| 16:25.14 | Skysky | bkw_: may i ask a question? let say i have 2 asterisk server, both has peer and user to each other. I wonder if i should start up one b4 the other. because sometimes when i call from 1 to the other, the other side seems not responding |
| 16:25.32 | |Vulture| | I have a frac, 12 lines being set up and the sales guy is asking me about setting up the Provisioning Worksheet |
| 16:25.36 | Darwin35 | Mar 16 11:20:35 WARNING[901]: chan_sip.c:1400 create_addr: No such host: |
| 16:25.36 | Darwin35 | Mar 16 11:20:35 NOTICE[901]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' |
| 16:25.38 | r3d5un | you must not use mpg321 it doesnt work with asterisk (thats what the doku tells). If you installed it on debian there is only mpg321 but with a symbolic link to mpg123. So make sure you are really using mpg123 |
| 16:25.39 | Darwin35 | that is |
| 16:26.02 | Skysky | bkw_: this really troubles me, because i would expect to restart one of them periodically |
| 16:26.04 | *** join/#asterisk BurnedOutGeek (~BurnedOut@216.215.202.4.nw.nuvox.net) |
| 16:26.09 | |Vulture| | tzanger: I was under the belief that you just set DIDs to the PRI and when people dialed in it looked for the first avail. open channel |
| 16:26.20 | tzanger | yes |
| 16:26.24 | Darwin35 | this all worked 3 days ago |
| 16:26.34 | Darwin35 | grrr |
| 16:26.38 | tzanger | Darwin35: it's saying on such host |
| 16:26.40 | tzanger | check your DNS |
| 16:26.42 | |Vulture| | tzanger: so what could he be talking about when he says Provisioning Worksheet? |
| 16:26.48 | tzanger | what's your ${BVOICE} or whatever varibale |
| 16:27.31 | enemy^x | r3d5un: is mpg123 ver 0.59r a good choice? |
| 16:27.58 | r3d5un | its the only choice (no just joking, but its the rcommended version) |
| 16:27.58 | tzanger | |Vulture|: it's just a way to spell out everything. it's a GOOD thing |
| 16:28.26 | |Vulture| | tzanger: so what should I tell them, I have 1 fax DID and 1 main # DID |
| 16:28.31 | tzanger | no |
| 16:29.07 | enemy^x | r3d5un: that is the version which I am currently using. But it still seems to give noise only, could there be an encoding issue (bitrate stuff) or maybe asterisk doesnt automaticly link to mpg123? I havent configured anything for mpg123 (asterisk wise). |
| 16:29.12 | BurnedOutGeek | hey guys. Has anybody been experienceing strange echo and call quality problems with the newer versions of CVS HEAD? |
| 16:30.02 | enemy^x | r3d5un: Im running asterisk 1.0.5 btw. |
| 16:30.12 | r3d5un | hmm,did you check if there is another mpg123 on your system, becuase i had the same problems and with the correct mpg123 it worked |
| 16:30.14 | *** join/#asterisk xilch (~xilch@66.239.17.226.ptr.us.xo.net) |
| 16:30.21 | tzanger | you tell then you're getting 23B+D, calls from them should use the B channels in ascending order, you want B8ZS/ESF with National2 or dms100 signaling (or whatever), that you want NXXNXXXXXX as the failover number (a number that is NOT a DID on the PRI where all calls to the PRI go if it's down) or if you just want their switch to report congestion.. and then also give them a list of all DIDs you want on the PRI, tell them you want the ability to set CID |
| 16:30.34 | r3d5un | sorry have to leave now i hope anybody else can help you |
| 16:31.06 | tzanger | you might also want to ask them to make sure you are able to receive RDNIS information and that you also want 7 or 10 digits for your PRI DIDs |
| 16:31.06 | enemy^x | ok, thanks anyways |
| 16:31.12 | tzanger | 4 digits suck |
| 16:31.25 | |Vulture| | THANX TZANGER! |
| 16:31.41 | Darwin35 | I hate when they play with the dns server at work |
| 16:31.44 | Darwin35 | they put ? infront of all my sip.bv.com |
| 16:31.52 | tzanger | |Vulture|: but yes, you just have DIDs on the PRI and it 'just works' -- but the provisioning sheet spells out exactly what works. |
| 16:33.03 | *** join/#asterisk mbaron (~mbaron@AVelizy-154-1-37-253.w82-124.abo.wanadoo.fr) |
| 16:33.22 | |Vulture| | tzanger: thank you for your help this is my first install |
| 16:33.33 | tzanger | |Vulture|: no worries, PRI is very cool |
| 16:33.35 | Skysky | can anyone pls give me some guideline on when and how to use register in iax.conf |
| 16:33.40 | *** join/#asterisk gr8nash (~andy@mamabear.si-forest.com) |
| 16:33.57 | JerJer[mobile] | register => bob:secret@host.tld |
| 16:34.36 | Darwin35 | ok this pisses me off I just lost my sip.conf on a reboot |
| 16:34.43 | Darwin35 | that sucks |
| 16:34.53 | JerJer[mobile] | a reboot did not cause the loss |
| 16:34.54 | Darwin35 | glad I keep a .bak |
| 16:35.01 | JerJer[mobile] | unless you are runnng TMPFS |
| 16:35.09 | Darwin35 | I am on ffs |
| 16:37.15 | Skysky | JerJer: i just wanna know y do we need register, because couldn't u set host=host.tld, username=bob, secret=secret under a peer? |
| 16:37.37 | Essobi | FFS? That doesn't stand for what I think it does.. Tehe. |
| 16:37.42 | *** join/#asterisk jdb1968 (~jdb1968@S010600045af29653.cg.shawcable.net) |
| 16:38.22 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 16:38.23 | Skysky | JerJer: or should i use both? a peer and a register line? |
| 16:38.33 | jdb1968 | hi all, quick ?, how are the spans on a TE410P numbered? i.e. Is span 1 closest to the motherboard? |
| 16:38.51 | enemy^x | I`ve got the mpg123 working now with regular mp3 files, but while trying shoucast streaming I get no sound at all..... Im using in musiconhold.conf (default => mp3:/usr/share/asterisk/mohmp3-empty,http://mfm.powershout.com:8700/listen.pls) Have also tried without listen.pls |
| 16:39.14 | JerJer[mobile] | http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ |
| 16:39.50 | *** join/#asterisk jeffik (~jeffik@208-41-192-106.client.dsl.net) |
| 16:39.53 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 16:39.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:40.02 | JerJer[mobile] | Skysky: the host.tld box needs a type=peer (or type=friend if you like hartburn) with a host=dynamic |
| 16:40.04 | Essobi | Hey JerJer.. I never did get chan_h323 MOH/playback working.. I'll have to wait till non-peak hours to play with it s'more.. but you think allow=slin would let my moh play? I was getting a non-native playback on from error message, and it was 64 to 4/4 since it was using 711u on the call in progress.. |
| 16:40.07 | VerCom | thanks for the help jon |
| 16:40.15 | JerJer[mobile] | slin?! |
| 16:40.17 | JerJer[mobile] | are you nutz |
| 16:40.20 | VerCom | pridialplan is what fixed my day |
| 16:40.21 | Essobi | lol |
| 16:40.24 | JerJer[mobile] | put the crack pipe down |
| 16:40.37 | Essobi | That's the format MOH was trying to use to play on the 711 streamm. |
| 16:41.07 | VerCom | starting at 23, 22, 21 etc on my pri |
| 16:41.21 | Essobi | *SHRUG* You suggested gsm when my playback wouldn't work.. thought maybe you knew something about the h323 internals reguarding codes, I didn't. and the crackpipe is put away. ;) |
| 16:41.31 | Essobi | codecs, even. |
| 16:42.08 | Essobi | I wonder why it's not trying to transcode the slin from MOH to 711 then.. Hmm. |
| 16:42.09 | gr8nash | im trying to page using overhead spearkers out /dev/dsp "exten => 1236,1,Dial(Console/dsp)" is there something wrong that makes this not work? |
| 16:42.16 | anthm | did you try the [moh_files] way? |
| 16:42.40 | Essobi | You talking to me or enemy^x ? |
| 16:42.44 | anthm | you |
| 16:42.57 | Essobi | Umm. No, I didn't mess with any of the moh settings.. |
| 16:43.09 | Essobi | what's the moh_files way? |
| 16:43.57 | anthm | [moh_files] |
| 16:43.57 | anthm | default => /var/lib/asterisk/holdmusic |
| 16:44.09 | anthm | a diff full of files |
| 16:44.11 | anthm | dir |
| 16:44.21 | anthm | can be in any format |
| 16:44.28 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 16:44.42 | anthm | if you need mp3 then you need the format_mp3 from asterisk-addons |
| 16:44.49 | *** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
| 16:44.50 | Essobi | Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4).. that's the h323 debug line I'm seeing when it tries to Moh the h323... 64 is slin and 4 is the 711u it was using.. |
| 16:44.53 | isamar | hi folks |
| 16:44.54 | ManxPower | !docs |
| 16:44.56 | ManxPower | ~docs |
| 16:44.57 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 16:44.58 | jontow | woah |
| 16:45.07 | ManxPower | Essobi, "show codecs" |
| 16:45.07 | jontow | got my sipura spa-2000 |
| 16:45.12 | jontow | overnight, and i paid for ground shipping |
| 16:45.13 | isamar | having "no authority found" issues with IAX2 |
| 16:45.23 | Skysky | wow. Mark Spencer is a very smart person~ (just read the article) |
| 16:45.27 | VerCom | can anyone give me a key search word for specifing the outbound dialing order for channels in a pri |
| 16:45.40 | Essobi | ManxPower What about it? |
| 16:46.06 | ManxPower | VerCom, see group= in /etc/asterisk/zapata.conf and use Dial(Zap/g1/blah) or Zap/G1 |
| 16:46.18 | ManxPower | Essobi, that tells you what number each codec is. |
| 16:46.20 | Essobi | 64 is slin and G711U is 4 in show codecs. |
| 16:46.25 | VerCom | I need to make sure when i dial out it starts at channel 23, then 22 etc rather than channel 1, 2, 3 |
| 16:46.31 | Essobi | I already stated that. :) |
| 16:46.35 | ManxPower | 64 is slinear. No idea why that is enabled. |
| 16:46.40 | anthm | G instead of g |
| 16:46.42 | Essobi | It's not. |
| 16:46.48 | Essobi | That's the whole point.. it's not enabled. |
| 16:47.01 | ManxPower | Essobi, How about on the OTHER side? |
| 16:47.02 | Essobi | but MOH is attempting to use it reguardless. |
| 16:47.05 | anthm | you cant have sound with our slinear that's why |
| 16:47.07 | Essobi | Anope. |
| 16:47.10 | anthm | without |
| 16:47.14 | *** join/#asterisk kahuna_ (~sootroom@rtl-2.i2k.com) |
| 16:47.19 | Essobi | G711u, 711a, and g729. |
| 16:47.24 | *** part/#asterisk kahuna_ (~sootroom@rtl-2.i2k.com) |
| 16:47.27 | ManxPower | anthm, There's no reason Asterisk can't transcode. |
| 16:47.28 | Essobi | is the only thing enabled on both sides. |
| 16:47.35 | isamar | having "no authority found" issues with IAX2 |
| 16:47.44 | ManxPower | Essobi, What IS your fetish for enableing ulaw and G729???? |
| 16:47.46 | isamar | normally.. what's the common reason for that? |
| 16:47.47 | enemy^x | does anyone here stream hold music from a shoucast server? |
| 16:47.51 | Essobi | lol |
| 16:47.52 | ManxPower | don't enable both ulaw and alaw for testing. |
| 16:48.05 | ManxPower | and for the love of all that's good don't enable G729 for testing! |
| 16:48.12 | Essobi | You asked about the other side.. that's hardcoded. I can't change it. |
| 16:48.27 | ManxPower | Essobi, What codecs are set on the far side? |
| 16:48.37 | Essobi | G711u, 711a, and g729. |
| 16:48.45 | *** join/#asterisk DJ-Pyro_ (~DJ-Pyro@firewall.brevient.net) |
| 16:48.47 | JerJer[mobile] | Speex or nothing! |
| 16:48.54 | Essobi | lol |
| 16:49.11 | ManxPower | JerJer[mobile], did you see Essobi's current error message? |
| 16:49.23 | ManxPower | Essobi Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4).. that's the h323 debug line I'm seeing when it tries to Moh the h323... 64 is slin and 4 is the 711u it was using.. |
| 16:49.24 | JerJer[mobile] | niet |
| 16:49.24 | anthm | mpg123 flavored moh runs in slinear because that is the format the player renders it to |
| 16:49.42 | Damin | JerJer[mobile]: LPC10! :) It rocks for IAX trunking over DialUp! :) |
| 16:49.45 | anthm | so in turn anybody listening to sets its readformat to slinear |
| 16:49.53 | anthm | which will build a translation path |
| 16:49.59 | ManxPower | anthm, as does MeetMe, but I've never seen an error with a translation path for that |
| 16:50.58 | isamar | I have a point A and a point B both set as friend in iax.conf |
| 16:51.15 | Essobi | Well, I assumed the translation path was getting broken, for some reason. |
| 16:51.20 | isamar | I call IAX2/testb/s@home-in and gives me no authority found... |
| 16:51.24 | isamar | what the hell is that? |
| 16:51.43 | anthm | manx, what was the arguement again i cant remember? |
| 16:51.58 | Essobi | if I drop an in bound call in queue from chan_h323.. no moh and no playback.. playback does the same thing.. |
| 16:52.18 | stevekstevek | Totally off-topic question: Anyone have any wiring guys/shops in NYC area they recommend? |
| 16:52.21 | Essobi | Hmm. Maybe there's a slip up with progress not getting set on the call leg.. Hmm. |
| 16:52.40 | isamar | I solved that issue in the past but cannot remember the reason.. |
| 16:52.48 | ManxPower | isam, Try Dial(IAX2/remoteiaxconf@locaiaxconf/s) |
| 16:53.14 | anthm | if you are using ulaw anyway that's why i suggested the moh_files you can make the music in ulaw and deliver it w/o transcoding |
| 16:53.23 | ManxPower | isam, The error message means (usually) on the remote side "I can't match the user/secret" |
| 16:53.35 | anthm | if yo ureally want you can make each file in every format and it will always pick the one that matches |
| 16:54.13 | mikegrb | stevekstevek: I prefer that question to some of the ontoppic but obviously not researched crap questions in here |
| 16:54.25 | mikegrb | stevekstevek: unfortunatly, I don't have a recommendation for you |
| 16:54.45 | stevekstevek | thx anyway :) |
| 16:56.00 | VerCom | <manxPower> if i have channel=>1,23 and then group=0 how do i tell outbound dialing to start at channel 23 |
| 16:56.21 | ManxPower | VerCom, Maybe you want channel -> 1-23 |
| 16:56.41 | VerCom | that is what i meant sorry for typo |
| 16:56.47 | ManxPower | You do also want group=1 |
| 16:56.57 | ManxPower | Dial(Zap/G1/666) |
| 16:57.26 | ManxPower | anthm is not just spouting words, he's trying to help you. |
| 16:57.36 | VerCom | problem is that when i do the dial(Zap/G1/999) it always picksup and dials out channel 1 first |
| 16:58.12 | anthm | unless someone broke it, gx is top down and Gx is bottom up |
| 16:58.13 | ManxPower | VerCom, It's not supposed to do that. |
| 16:58.24 | ManxPower | anthm, What is Rx and rx anyway? |
| 16:58.39 | ManxPower | VerCom, BTW, why do you care about the order? |
| 16:58.41 | VerCom | let me verify case of Gx versus gx |
| 16:59.02 | anthm | never heard of R |
| 16:59.07 | ManxPower | anthm, it's there. |
| 16:59.15 | anthm | random maybe ? |
| 16:59.17 | Darwin35 | ok fixed |
| 16:59.19 | Darwin35 | yes |
| 16:59.28 | ManxPower | anthm, That's what I would assume, but it didn't appear to do random. |
| 16:59.36 | anthm | yah and you wouldnt need 2 |
| 17:00.12 | anthm | maybe it changes to be case insensitive g and r |
| 17:01.25 | anthm | according to the code |
| 17:01.29 | anthm | r is round robin |
| 17:01.32 | *** join/#asterisk NicolasC (~nico@logilab.net2.nerim.net) |
| 17:01.40 | *** part/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 17:01.52 | anthm | so r R must be fwd or reverse round robin |
| 17:01.59 | Darwin35 | now to figure out why my phone is not getting a time update |
| 17:02.07 | anthm | f (dest[0] == 'G') { |
| 17:02.07 | anthm | <PROTECTED> |
| 17:02.18 | anthm | if (dest[0] == 'R') { |
| 17:02.18 | anthm | <PROTECTED> |
| 17:02.19 | anthm | <PROTECTED> |
| 17:03.14 | ManxPower | What would be the difference between "round robin" and plain G/g? |
| 17:03.56 | JerJer[mobile] | uses all channels |
| 17:04.07 | JerJer[mobile] | not just the top or bottom few |
| 17:04.14 | anthm | use 1, next time use 2 even if 1 is free |
| 17:04.17 | ManxPower | So G/g won't us all the channels in the group |
| 17:04.20 | anthm | 3, 4 ,so on |
| 17:04.41 | ManxPower | Ah. |
| 17:04.55 | ManxPower | I think the guy has a PRI and is confuzzled about how PRIs work. |
| 17:05.04 | *** join/#asterisk outtolunc (~me@66.218.53.170) |
| 17:05.16 | AgiNamu | Wow, gSOAP flies |
| 17:06.07 | AgiNamu | I ran a test, calling from linux to windows (.NET). I was able to sustain 476 requests/sec, with an average single request taking 0.005 seconds. |
| 17:06.12 | AgiNamu | doing DID -> Codec lookups. |
| 17:06.23 | AgiNamu | on a database of 100,000 DIDs. |
| 17:06.35 | *** join/#asterisk jedaustin (~chatzilla@austin-j.its.dist.maricopa.edu) |
| 17:06.57 | AgiNamu | And that's against Windows XP, my personal desktop |
| 17:07.15 | AgiNamu | Windows 2003 would provide even more performance (the bottleneck is the windows soap code) |
| 17:07.50 | AgiNamu | I'd like to see locked linked lists get that perf!!! |
| 17:07.53 | *** part/#asterisk Flash_ (~Flash_@pc-80-235-133-235-na.blueyonder.co.uk) |
| 17:08.08 | AgiNamu | I wrote it up here: www.atrevido.net |
| 17:13.32 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 17:14.27 | |Vulture| | tzanger: thanx man I just had the conversation and everything is set :) |
| 17:14.32 | jedaustin | Setting up broadvoice.. register => <accountid>@sip.broadvoice.com:<password>:<account id>@sip.broadvoice.com/<extension> is accountid my phone number or their account number? |
| 17:14.32 | tzanger | good good |
| 17:14.39 | jhiver | nyeaaaah it works |
| 17:14.57 | jhiver | i've got like my _own_ prepaid solution... well... embryo of... |
| 17:15.09 | jhiver | Perl is cool |
| 17:15.14 | jhiver | AGI is cool |
| 17:15.14 | |Vulture| | tzanger: I went with 7 digits since were only in a single lata |
| 17:15.18 | jhiver | Asterisk is cool |
| 17:15.23 | tzanger | good :-) |
| 17:15.27 | jedaustin | jhiver: its the duck tape of the internet :) |
| 17:15.28 | tzanger | I asked for 7 as well |
| 17:15.47 | jhiver | yeah and I've used like strict and warnings throughout :) |
| 17:15.54 | jhiver | because astcc doesn't :( |
| 17:16.04 | |Vulture| | tzanger: he said he had to ask about the fallover to "All circuits are busy |
| 17:16.11 | |Vulture| | " and the RDNIS info |
| 17:17.15 | jhiver | It's true Perl is awesome |
| 17:17.21 | tzanger | not all circuits busy (although yuouc an do that too) but rather if the PRI is down what happens to clals |
| 17:17.30 | jhiver | it's got bazillion of modules on CPAN and it works with Asterisk too :) |
| 17:19.31 | *** join/#asterisk riksta (~rick@81-178-173-194.dsl.pipex.com) |
| 17:19.54 | *** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 17:20.54 | jedaustin | ok.. how do I tell if asterisk is connected to broadvoice? |
| 17:20.55 | anthm | now if you only could embed perl in asterisk and natively expose it's api to perl... hmm |
| 17:21.07 | Pkunk | jedaustin: BV is fucked |
| 17:21.12 | Darwin35 | res_perl is in addons |
| 17:21.42 | jedaustin | Pkunk: what about em? |
| 17:21.50 | Darwin35 | man issues |
| 17:21.52 | ManxPower | jedaustin, "iax2 show registry" shows what places Astrisk is registered TO |
| 17:21.56 | Darwin35 | but it is working again |
| 17:22.21 | Pkunk | ManxPower: BV supports iax2 ? |
| 17:22.50 | |Vulture| | doubt it |
| 17:23.06 | ManxPower | jedaustin, "sip show registry" shows what places Astrisk is registered TO |
| 17:23.13 | jedaustin | I show this line when I do sip show peers: sip.broadvoice. 147.135.8.128 255.255.255.255 5060 Unmonitor |
| 17:23.14 | jhiver | BTW, what other IAX termination telcos would you recommend? I currently use Nufone + VoIPJet |
| 17:23.15 | *** join/#asterisk Inv_arp (junya@adsl-3-245-189.mia.bellsouth.net) |
| 17:23.27 | |Vulture| | voicepulse |
| 17:23.31 | Pkunk | Darwin35: i haven't been able to get it working for > a week . with nat=yes i can hear no ringing sound , and the call gets disconnected within a few seconds |
| 17:23.34 | jhiver | it'd be nice to have a third inexpensive one |
| 17:23.35 | |Vulture| | Nufone and VPC are my favs |
| 17:23.37 | jedaustin | nothing for sip show registry |
| 17:23.43 | jhiver | vpc? |
| 17:23.48 | |Vulture| | voice pulse connect |
| 17:23.48 | jhiver | voice pulse connect? |
| 17:24.04 | ManxPower | jedaustin, then yu are not registered to bv and cannot receive calls. |
| 17:24.12 | bkw_ | AsterLink!!! |
| 17:24.13 | jhiver | isn't voicepulse a bit expensive for international? |
| 17:24.13 | Pkunk | Darwin35: with nat=never or no , all i get is "Got a response on a call we don't know about" for outgoing |
| 17:24.21 | bkw_ | jedaustin, asterlink.com |
| 17:24.36 | Darwin35 | well have it working now in and out |
| 17:26.27 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 17:26.46 | jedaustin | Does the tftpboot stuff need to be set with Broadvoice? |
| 17:27.48 | Darwin35 | ok inbound is not working at the mn |
| 17:27.51 | Darwin35 | min |
| 17:27.56 | Darwin35 | outbound is working |
| 17:28.22 | _THEEND_ | someone could tell me if i put my own mp3 for musiconhold when i reload asterisk it crash? |
| 17:28.33 | NicolasC | hello, anyone willing to help me debug ring detection problem ? |
| 17:28.37 | _THEEND_ | someone could tell me why if i put my own mp3 for musiconhold when i reload asterisk it crash? |
| 17:28.55 | ManxPower | _THEEND_, What moron told you that? We need to beat them senseless. |
| 17:29.16 | _THEEND_ | ManxPower ??? |
| 17:29.30 | riksta | _THEEND_: what version of mpg123 are you using |
| 17:29.35 | JerJer[mobile] | rpm -e openofficeorg-core01 openofficeorg-core02 openofficeorg-core03 openofficeorg-core04 openofficeorg-core05 openofficeorg-core06 openofficeorg-core07 openofficeorg-core08 openofficeorg-core09 openofficeorg-spellcheck openofficeorg-writer openofficeorg-gnome-integration openoffice-i18n openofficeorg-math |
| 17:29.39 | JerJer[mobile] | damnit |
| 17:29.42 | JerJer[mobile] | grr |
| 17:29.51 | sivana | heh |
| 17:30.14 | ManxPower | _THEEND_, Now, mpg123 does not handle VBR files very well and does not handle IDv3 tags very well when it has to do rate resampleing (as required by asterisk) |
| 17:30.16 | _THEEND_ | mpg123-pre0.59s. |
| 17:30.23 | bkw_ | 0.59r BABY |
| 17:30.30 | ManxPower | _THEEND_, ONLY 0.59r works |
| 17:30.33 | Darwin35 | 412-238-1009 some one drop a line pls |
| 17:30.41 | _THEEND_ | so what i have to do? |
| 17:30.53 | _THEEND_ | i have to install the newer one? |
| 17:30.54 | ManxPower | In fact you can do a "make mpg123" and Asterisk will download and install the correct version of mpg123 for you. |
| 17:30.55 | Darwin35 | use madplay |
| 17:31.11 | jedaustin | Setting up broadvoice.. register => <accountid>@sip.broadvoice.com:<password>:<account id>@sip.broadvoice.com/<extension> is accountid my phone number or their account number? |
| 17:31.12 | JerJer[mobile] | or simply don't use mp3s for MOH |
| 17:31.13 | jhiver | damn, voicepulse *is* kind of expensive |
| 17:31.17 | ManxPower | ("make mpg123" in the asterisk source directory) |
| 17:31.24 | NicolasC | my ring detection problem is as follows |
| 17:31.25 | bkw_ | Ten Out of Ten For Style, But Minus Several Million For Good Thinking!!! |
| 17:31.32 | jhiver | for international, compared with voipjet... but I guess the reliability is different? |
| 17:31.34 | ManxPower | Darwin35, Than you for volunteering to step him thru setting up and installing madplay |
| 17:31.35 | bkw_ | NEXT!!! |
| 17:31.45 | NicolasC | ztmonitor shows Rx pulsing |
| 17:31.54 | Darwin35 | madplay is easy |
| 17:31.56 | _THEEND_ | it install 0.59 r |
| 17:31.56 | NicolasC | but asterisk does not take the call |
| 17:32.24 | ManxPower | NicolasC, What card do you have? |
| 17:32.37 | NicolasC | TDM400P + 2FXO + 2 FXS |
| 17:32.41 | NicolasC | i'm in france |
| 17:32.54 | JerJer[mobile] | NicolasC: have you loaded the appropriate country ? |
| 17:32.58 | Darwin35 | can some one drop me a line to test incoming |
| 17:33.11 | ManxPower | NicolasC, Remember if you accidently plug a phone line into an FXS module it will blow up the module and you'll have to replace it. |
| 17:33.30 | Darwin35 | 412-238-1009 |
| 17:33.33 | JerJer[mobile] | well if you plug it in and the line rings |
| 17:33.35 | NicolasC | defaultzone and loadzone are set to fr in /etc/zaptel.conf |
| 17:33.37 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 17:33.52 | JerJer[mobile] | i smell bacon |
| 17:34.12 | Hmmhesays | mmmm bacon |
| 17:34.12 | Darwin35 | I smell spam |
| 17:34.20 | Hmmhesays | i smell a mc chicken coming on |
| 17:34.36 | Hmmhesays | possibly 2 |
| 17:34.37 | NicolasC | ztmonitor shows Rx pulsing when I call in |
| 17:34.48 | ManxPower | NicolasC, Can you dial out on the line? |
| 17:35.12 | NicolasC | have not tried yet. |
| 17:35.12 | ManxPower | NicolasC, You didn't listen to a thing I said, did you? |
| 17:35.30 | ManxPower | NicolasC, the port at the TOP of the card is the first port. |
| 17:35.35 | NicolasC | i did listen to the "plug a phone line into an FXS module it will blow up" |
| 17:35.44 | ManxPower | Check the card to confirm that you have the modules plugged into the ports you expect. |
| 17:36.08 | *** join/#asterisk ClayReiche123 (~creiche@199-112.35-65.tampabay.res.rr.com) |
| 17:36.09 | NicolasC | my card has two red modules (FXO) then two green modules (FXS), meaning i have FXO as 1-2 |
| 17:36.22 | ManxPower | NicolasC, How do you know that? |
| 17:36.33 | ManxPower | Did you look on the card and see the port numbers printed next to the connectors? |
| 17:36.45 | ManxPower | They are the reverse of what many people expect. |
| 17:36.46 | ClayReiche123 | would it be crazy of me to say that the HEAD version works better than the STABLE? |
| 17:37.03 | ManxPower | ClayReiche123, We would have to beat you senseless |
| 17:37.07 | NicolasC | that's what "Installing the TDM400" pdf says on figure 1 |
| 17:37.25 | Beirdo | anyone know of debian/ubuntu packages for asterisk that are newer than 1.0.4? |
| 17:37.44 | NicolasC | the cable is plugged on port 1 ( 1 as written on the card) |
| 17:37.45 | ManxPower | Digium has docs now? Wow! |
| 17:37.55 | NicolasC | on the web :-) |
| 17:37.57 | DJ-Pyro_ | anyone else having a problem with CVS-HEAD-03/13/05-17:37:09 and audio for call parking? the parked user can hear the voice of the person who picks up, but the user can't hear the parker's voice |
| 17:38.01 | *** join/#asterisk smash3 (~smash@198.107.16.189) |
| 17:38.15 | smash3 | Hello |
| 17:38.24 | jhiver | yeah, on /testing you have 1.0.5 |
| 17:38.49 | smash3 | i got asterisk setup with network support and running but i dont know a whole lot about how to use it. |
| 17:38.58 | ManxPower | ClayReiche123, Occasionally CVS-HEAD doesn't even COMPILE. |
| 17:39.05 | smash3 | can someone help me with any documentation |
| 17:39.15 | Beirdo | jhiver: can you msg me the requisite sources line so I don't screw it up? |
| 17:39.15 | ManxPower | CVS-HEAD is USUALLY pretty stable, but you don't want to count on it. |
| 17:39.19 | ManxPower | ~docs |
| 17:39.20 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:39.55 | NicolasC | is there any special configuration to put into zapata.conf to detect (or at least debug) ringing ? |
| 17:39.56 | Essobi | Beirdo nope |
| 17:40.05 | ManxPower | I forgot the -r v1-0 on a cvs checkout a few weeks ago. |
| 17:40.27 | ManxPower | HARD locked the machine when it tried to load the zaptel drivers on one system. On another system Asterisk just crashed after a few days. |
| 17:40.52 | gr8nash | is there a term for when a park light flashes if someone is on hold with asterisk.. i dont know the proper term to search google for |
| 17:40.52 | Essobi | didn't update libpri? ;) |
| 17:41.16 | Essobi | "park light"? |
| 17:41.20 | ManxPower | Essobi, I never use CVs without updating asterisk asterisk-sounds zaptel libpri |
| 17:41.22 | gr8nash | lol |
| 17:41.25 | Essobi | Hehe. |
| 17:41.28 | ClayReiche123 | ManxPower:I'm just having a hard time with the current STABLE. I'm getting poor audio quality. Using the same TA, same gateway, same calling number... only thing I change is the softswitch. |
| 17:41.55 | Essobi | ClayReiche123 might be your codecs you're using |
| 17:41.58 | ClayReiche123 | ManxPower:I swap asterisk versions and audio cleans up on HEAD. |
| 17:42.24 | ClayReiche123 | ManxPower:Only variable is asterisk versions. |
| 17:42.26 | Essobi | -head has better preferencing for codecs I'm told |
| 17:42.46 | ManxPower | ClayReiche123, as Essobi said it might be a codec issue. |
| 17:42.58 | Essobi | NEXT! |
| 17:43.12 | ManxPower | ClayReiche123, you have do NOT have a bandwidth= line and you have disallow=all and only have allow= lines for the 1 or 2 codecs you want? |
| 17:43.31 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 17:43.37 | *** join/#asterisk ronn (~ronn@host217-46-199-165.in-addr.btopenworld.com) |
| 17:44.01 | ClayReiche123 | ManxPower:I don not have a bandwidth= line. I have disallow=all allow=g729 allow=ulaw. |
| 17:44.18 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 17:44.28 | ManxPower | ClayReiche123, I have never heard of differences in audio quality between stable and head |
| 17:44.44 | Essobi | so is alaw europe/asia and ulaw is US PSTN? |
| 17:44.58 | *** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net) |
| 17:45.04 | Essobi | ClayReiche123 What g729 driver are you using? |
| 17:45.18 | ManxPower | Essobi, yes |
| 17:45.26 | ClayReiche123 | ManxPower:I pretty certain it is a codec issue. And I was contemplating going to HEAD.... |
| 17:45.29 | smash3 | how many voip phones can you hookup to a asterisk serveR? |
| 17:45.34 | ClayReiche123 | Essobi: I'm doing passthru. |
| 17:45.38 | ManxPower | smash3, 14 million |
| 17:45.41 | smash3 | hrm |
| 17:45.44 | Inv_arp | heh |
| 17:45.53 | Essobi | smash3 depends on the gear and bandwidth involved. |
| 17:46.00 | smash3 | i got a burstable ds3 |
| 17:46.08 | smash3 | 20mbit burstable |
| 17:46.08 | ManxPower | smash3, Once you understand Asterisk a little better you'll realize how silly that question is. |
| 17:46.18 | smash3 | yeah i got it installed |
| 17:46.18 | Essobi | ClayReiche123 Have you wipped out a sniffer and verified you're doing pass through? |
| 17:46.22 | smash3 | but im trying to figure out the big picture |
| 17:46.39 | Essobi | smash3 http://www.voip-info.org |
| 17:46.41 | _THEEND_ | ManxPower what i have to do to play an my own mp3 |
| 17:46.42 | smash3 | i installed the windows version on colinux for a trial |
| 17:46.47 | _THEEND_ | in musiconhold? |
| 17:46.49 | ManxPower | your question is much like "how fast can my car go". Without knowing the type of car, engine, tires, age, etc et can't tell you. |
| 17:46.53 | Essobi | Partake in our bible, and ye shall see. |
| 17:47.00 | _THEEND_ | i only put it in the default dir |
| 17:47.03 | ManxPower | _THEEND_, install mpg123r |
| 17:47.13 | denon | I wonder why people call them "Burstable T3s" instead of "throttled T3s" :) |
| 17:47.21 | Essobi | tehe |
| 17:47.33 | _THEEND_ | ManxPower i have it installed |
| 17:47.41 | ManxPower | denon, because the sales people are afraid of giving customers ideas when the customer buys the service and doesn't understand the service. |
| 17:47.48 | ManxPower | _THEEND_, what does mog123 -v" show? |
| 17:47.50 | Essobi | denon: why are 1000meg HD sold as a gig? It's all pr and spin. |
| 17:47.53 | johnnyb | If Asterisk is in the media path, but the codecs on both sides are the same, does asterisk have to decode/re-encode each frame, or does it just copy the packets in and out? |
| 17:48.05 | ManxPower | jonas, yes and no. |
| 17:48.13 | ClayReiche123 | Essobi: I've been sniffing all day... it's using g729, just sounds like crap when using stable. If passthru wasn't working, I would hear nothing, correct? |
| 17:48.16 | ManxPower | jonas, depends on a lot of things, but mostly on the dial options. |
| 17:48.22 | _THEEND_ | the help |
| 17:48.30 | _THEEND_ | ManxPower |
| 17:48.30 | ManxPower | _THEEND_, and the VERSION number? |
| 17:48.46 | Darwin35 | ok this is a killer |
| 17:48.47 | _THEEND_ | azz |
| 17:48.48 | _THEEND_ | s |
| 17:48.52 | _THEEND_ | non r |
| 17:48.52 | _THEEND_ | soryr |
| 17:48.54 | johnnyb | ManxPower: it's johnnyb, not jonas :) |
| 17:48.54 | _THEEND_ | now i update |
| 17:48.57 | Essobi | ClayReiche123 Umm. passthrou just means the endpoints are talking RTP to each other.. bridged they are both talking to the * server. |
| 17:49.09 | Darwin35 | outbound is working but inbound is not |
| 17:49.10 | smash3 | Hey am i going to need a VOIP provider to make longdistance phone calls? |
| 17:49.12 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 17:49.19 | Essobi | Use tethereal/ethereal and sniff them to see what it's doing and where your rtp is going. |
| 17:49.28 | ManxPower | Essobi, Um, passthru means Asterisk is not touching the audio, it still may be going thru Asterisk, You are thinking of reinvites. |
| 17:49.42 | Essobi | Oh, yea.. what he said. ;) |
| 17:49.45 | ManxPower | smash3, No. |
| 17:49.59 | smash3 | even to non voip phones? |
| 17:50.00 | ClayReiche123 | ManxPower:It is going through asterisk. and reinvites are off. |
| 17:50.08 | ManxPower | smash3, Correct. |
| 17:50.17 | ManxPower | If you install local phone lines you don't need a VoIP provider. |
| 17:50.35 | smash3 | local phone lines to that area? |
| 17:50.39 | ManxPower | smash3, please step away from the channel and go read the docs. |
| 17:50.42 | smash3 | in which im calling |
| 17:50.44 | Essobi | you still get a phone bill either way. ;) |
| 17:50.45 | smash3 | ok thanks |
| 17:50.49 | Essobi | MEhe. |
| 17:50.59 | ManxPower | ~docs |
| 17:51.00 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:51.01 | ManxPower | ~mailinglist |
| 17:51.02 | jbot | somebody said mailinglist was Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 17:51.03 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
| 17:51.14 | Essobi | someone should draw a basic voip diagram and stick it on the docs links.. |
| 17:51.22 | Essobi | been a lot of newb these days |
| 17:51.32 | mAsH` | anyone can help me? |
| 17:51.33 | ManxPower | Essobi, nother asterisk article came out today |
| 17:51.50 | ManxPower | mAsH`, Only if you ask an intellegent question. |
| 17:51.54 | Tall-guy | Hey lads, I just got an snom 190 (sip phone), and it works great internally.....when I take it external (ie: at home), I can't make it work to my asterisk box (which is reachable on the internet, but behind a firewal of course)....yet softphones from Xlite can hit asterisk just fine in this fashion....any clues??? |
| 17:52.00 | mAsH` | i installed 1.0.6 and the config of 1.0.5 file doesn't work |
| 17:52.05 | mAsH` | it could be? |
| 17:52.17 | ManxPower | I said an intellegent question. |
| 17:52.26 | ManxPower | Like how about an error message to start out with. |
| 17:52.41 | Essobi | (Or an exact description of a problem) |
| 17:52.44 | ManxPower | mAsH`, Regardless you should not use 1.0.x |
| 17:52.48 | ManxPower | ..er..1.0.6 |
| 17:52.57 | ManxPower | 1.0.6 had a significant problem with musiconhold |
| 17:52.59 | JerJer[mobile] | no stick with your first statement |
| 17:53.05 | JerJer[mobile] | cvs -head or nothing |
| 17:53.07 | ManxPower | JerJer[mobile], Hush you! |
| 17:53.28 | ManxPower | JerJer[mobile] is one of those crazy people that run CVS-HEAD in production. Other than that he's a pretty nice guy. |
| 17:53.33 | Essobi | Tall-guy Um. nat problem? |
| 17:53.52 | ClayReiche123 | I'm considering it too! |
| 17:53.55 | mAsH` | if i try to call a sip client in the same extension, i got this errore message |
| 17:53.56 | mAsH` | NOTICE[20927]: pbx.c:1330 pbx_extension_helper: Cannot find extension context 'local' |
| 17:54.04 | Essobi | ManxPower He lives and dies by the sword. ;) |
| 17:54.08 | mAsH` | is the same config file of 2 hours ago |
| 17:54.10 | ManxPower | JerJer[mobile], I accidently installed CVS-HEAD a couple of weeks ago. THAT was a disaster. |
| 17:54.18 | ManxPower | (forgot the -r 1-0) |
| 17:54.30 | JunK-Y | ya didnt check the asterisk-cvs? :P |
| 17:54.36 | ManxPower | mAsH`, move back to 1.0.5 until 1.0.7 comes out. |
| 17:54.54 | ClayReiche123 | JerJer: What -head version do you use in production? |
| 17:55.00 | ClayReiche123 | recent? |
| 17:55.06 | mAsH` | ok :/ |
| 17:55.38 | ManxPower | JunK-Y, it hardlocked the server when wcfxs loaded. |
| 17:55.42 | ManxPower | on the other server it just crashed every once in a while. |
| 17:56.19 | rvhi | anyone knows how to rotate /var/log/asterisk/*? |
| 17:56.26 | rvhi | looks like it is only kept one day |
| 17:56.32 | ManxPower | rvhi, asterisk -rx "logger rotate" |
| 17:56.34 | Tall-guy | Essobi: well, I would think NAT, except that my softphones work fine.... |
| 17:56.36 | JunK-Y | rvhi: logger rotate |
| 17:56.38 | mAsH` | sorry, cani use chan sip at the same time with H323 ? |
| 17:56.44 | *** join/#asterisk cgeek (~cgeek@pl6.lawrence.org.uk) |
| 17:56.56 | ManxPower | mAsH`, Yes |
| 17:57.12 | mAsH` | tnx ManxPower ;) |
| 17:57.20 | ClayReiche123 | exit |
| 17:57.21 | JerJer[mobile] | this is not Bayonne |
| 17:57.25 | ClayReiche123 | oops... |
| 17:57.25 | ManxPower | Assuming you really meant: Can I use chan_sip and chan_h323 on the same system |
| 17:57.32 | rvhi | is it able to specify logger rotate in logger.conf? |
| 17:58.05 | ManxPower | mAsH`, You can thank me by finding me a job in Europe. |
| 17:58.43 | Essobi | Man.. Don't have verbose > 0 if you fire up IPSwitchBoard |
| 17:58.56 | Essobi | those mesages will beat your server and console to death |
| 17:59.52 | Essobi | mash` If youcan get chan_h323 to work. ;) |
| 18:00.03 | mAsH` | can to work with me ;) |
| 18:00.13 | rvhi | manxpower, why europe? |
| 18:00.46 | ManxPower | rvhi, Because I don't want to live in a country that would elect George Bush for a SECOND term. |
| 18:00.54 | Essobi | MAha |
| 18:01.03 | jedaustin | ManxPower: were happy to see you go :) |
| 18:01.07 | ManxPower | But more seriously things have been going downhill since Regan took office. |
| 18:01.35 | jedaustin | ManxPower: I'd have to disagree, but this is not a forum for such a conversation |
| 18:01.48 | ManxPower | And it would be nice to live in a place that doesn't consider breasts to be obscene. |
| 18:02.10 | Qwell | ManxPower: ethiopia? |
| 18:02.12 | *** join/#asterisk Samurai_ka (~Samurai_k@ad96edd8b.dsl.de.colt.net) |
| 18:02.16 | ManxPower | Qwell, europe |
| 18:02.21 | jedaustin | Qwell: hahaha :) |
| 18:02.21 | Qwell | That works too |
| 18:02.44 | ManxPower | jedaustin, I ranted enough on #asterisk-drinkers |
| 18:03.03 | Qwell | So, when you moving? |
| 18:03.12 | ManxPower | Qwell, when I find a job. |
| 18:03.32 | *** join/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 18:03.45 | jedaustin | ManxPower: Breasts are great but where do you draw the line.. I don't want to see 80 year old ones.. theres enough of that on cable and the internet to satisfy me :) |
| 18:03.48 | Corydon-w | ManxPower: what are you talking about, Bush only got elected to a single term so far... |
| 18:03.50 | ManxPower | Qwell, That's kind of why I just said "mAsH`, You can thank me by finding me a job in Europe." |
| 18:04.01 | ManxPower | jedaustin, then don't look. |
| 18:04.03 | km- | Good Afternoon everyone! |
| 18:05.05 | jedaustin | ManxPower: I don't want to see people taking a dump in the streets either. There are places in America where you can be as nude as you want, that's good enough for me. |
| 18:05.12 | ManxPower | Corydon-w, As I said things started going downhill in the 1980's |
| 18:05.38 | *** join/#asterisk afe ([9Us9SnKMQ@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:06.10 | ManxPower | jedaustin, Did you listen to NPR this morning? |
| 18:06.17 | *** join/#asterisk awannabe (~vdub@h-64-105-157-50.phndaz91.covad.net) |
| 18:06.26 | Corydon-w | Nah, it all goes back to 1972, when lying, cheating, and stealing was used to game an election |
| 18:06.41 | Jearil | ah, the good ol' days |
| 18:06.43 | ManxPower | There was a story about a television script that had the *word* "breast" removed from it by the network censors because the network was afraid of FCC fines? |
| 18:06.52 | jedaustin | ManxPower: I remember when Carter was president.. LIFE SUCKED. Regan brought home and prosperity |
| 18:06.57 | jedaustin | NPR is blatantly anti bush |
| 18:07.08 | mAsH` | ManxPower:what do u want to do? :P |
| 18:07.20 | ManxPower | mAsH`, /msg me for a mini resume |
| 18:07.21 | jedaustin | I quit listening to them when their bias went too far for me. |
| 18:07.28 | CosmicRay | jedaustin: perhaps, but he did so only by using irresponsible fiscal policies that we are still paying for |
| 18:07.37 | awannabe | hello...i am having a problem connecting to txlink.net..i keep getting a INVITE not authticated...it seems to be passing the wrong info, its like its not using my settings in extensions.conf |
| 18:07.51 | rvhi | manxpower, if you hate him so much, you got to move, i guess, they are talking about another Bush for 8 years |
| 18:07.58 | CosmicRay | jedaustin: I've never detected any bias either way with NPR; if anything, I fault them for failing to push hard enough against the propoganda spread by presidents, both clinton and bush |
| 18:08.05 | jontow | awannabe; your settings in ..extensions.conf ? wouldn't it be sip.conf? or am i mislead?:) |
| 18:08.20 | CosmicRay | however that is a criticism that I would also level at most media outlets these days |
| 18:08.24 | awannabe | txlink doesnt use a peer entry |
| 18:08.28 | ManxPower | rvhi, There's a lot of great things about the USA, there are also a lot of really bad things about the USA |
| 18:08.31 | jontow | aha.. then i am mislead. |
| 18:08.32 | jedaustin | I think 'trickle down economics' worked. |
| 18:08.42 | jedaustin | Cosmic Ray: if you agree with their bias you won't see it |
| 18:08.53 | rvhi | talking about bias, fox must no 1 |
| 18:09.01 | rvhi | must be |
| 18:09.02 | CosmicRay | jedaustin: such an assertion is a cop-out from someone that can't prove their case |
| 18:09.07 | CosmicRay | jedaustin: give me an example of bias |
| 18:09.14 | CosmicRay | rvhi: indeed |
| 18:09.23 | JerJer[mobile] | rvhi; its 4 more years.... learn our system before bitching about it |
| 18:09.23 | awannabe | jontow, does that make since now? heh |
| 18:09.24 | ManxPower | A flame war on #asterisk? My work here is done! |
| 18:09.28 | _THEEND_ | ManxPower now it works |
| 18:09.28 | _THEEND_ | tnx |
| 18:09.30 | CosmicRay | rvhi: I have trouble believing fox is "news" though |
| 18:09.32 | _THEEND_ | :) |
| 18:09.38 | CosmicRay | it's more talk |
| 18:09.52 | jedaustin | Cosmic Ray: Flat out bush bashing every day for the past 5 years |
| 18:10.01 | rvhi | jerjer, your system? is it open source, did you write it? |
| 18:10.08 | jedaustin | Npr was different when Clinton was president |
| 18:10.09 | CosmicRay | jedaustin: I listen to NPR every day and don't recall any such incident. please give me a specific example. |
| 18:10.32 | jedaustin | CosmicRay: that's because you don't like Bush. You agree so you don't notice the bias. |
| 18:10.35 | ManxPower | NPR IS liberal, but I think they do TRY to be objective. |
| 18:10.39 | CosmicRay | jedaustin: I don't think so, they pulled about the same number of punches then. provided live coverage of the senate and house hearings, too |
| 18:10.43 | jontow | what about asterisk? |
| 18:10.45 | JerJer[mobile] | rvhi: I am a proud, gun carrying american |
| 18:10.50 | CosmicRay | jedaustin: why can you not provide even one specific example of bias? |
| 18:10.56 | jedaustin | ManxPower: yes they do try. |
| 18:11.11 | JerJer[mobile] | rvhi: i am a proud, gun carryng, war mongering american |
| 18:11.14 | JerJer[mobile] | there we go |
| 18:11.15 | Tall-guy | Is an Snom 190 a democratic or republican phone? :) |
| 18:11.25 | CosmicRay | jedaustin: transcripts of NPR programs are available online; if it is as pervasive as you assert, you should have no trouble pointing me to an example |
| 18:11.29 | ManxPower | Tall-guy, It's obviously an independent phone. |
| 18:11.31 | jedaustin | CosmicRay: Dianne Rheim |
| 18:11.43 | Tall-guy | manx: but would ralph nader be able to configure it to work behind a nat firewall? |
| 18:12.05 | Samurai_ka | Can someone give me a hint if a Fritz!PCI works with * |
| 18:12.07 | Jearil | I'm starting to think that my Polycom IP-500 belongs to the Nazi party |
| 18:12.09 | ManxPower | Tall-guy, no, but he would start complaining to anyone that would listen how bad the docs are. |
| 18:12.13 | JerJer[mobile] | hint |
| 18:12.18 | *** join/#asterisk Tili (~Tili@202-133-65-39-dialup.sat.net.pk) |
| 18:12.25 | Tall-guy | manx: :) |
| 18:12.28 | ManxPower | ~google Fritz site:lists.digium.com |
| 18:12.35 | CosmicRay | jedaustin: yes, it is clear that she is not a bush fan, but 1) hers is a talk program, and 2) I don't see her bias showing through in terms of cutting people off like one often sees in conservative talk. in fact, you'd have to listen to the program for quite a while to get this idea. |
| 18:12.49 | jedaustin | CosmicRay: I think I'd have as much trouble convincing you even with a clear example. |
| 18:12.50 | CosmicRay | jedaustin: and she is just as intolerant of lying democrats as she is of lying republicans |
| 18:13.03 | CosmicRay | jedaustin: hard to know, since you haven't come up with any. |
| 18:13.06 | Samurai_ka | thx |
| 18:13.17 | JerJer[mobile] | i am not biased or racist, I hate every one equally |
| 18:13.36 | ManxPower | We just need balance. Some ultraliberals to balance out the ones that carry on Srtom Thurmand's legacy. |
| 18:13.46 | CosmicRay | heh |
| 18:13.52 | ManxPower | JerJer[mobile], There's something we can agree on! |
| 18:14.08 | CosmicRay | actually, I think that this whole liberal vs. conservative thing is a false distinction foisted upon us by both political parties for their own gain |
| 18:14.12 | Tall-guy | I think Ross Perot would like the Snom 190 :) |
| 18:14.36 | CosmicRay | I am socially liberal and fiscal conservative. Oddly enough, the democrats are displaying both qualities more than the republicans these days. |
| 18:14.44 | ManxPower | "My tribe good! All other tribes bad!" |
| 18:14.48 | jedaustin | CosmicRay: Diane Rheim just hangs up on people 'Thank you for your call' and moves on if she disagrees with you. I like Morning Edition though, they tend to stay more distanced and aren't so obviously anti-republican like Dianne Rheim, Fresh Air, and a few other shows. |
| 18:15.21 | ClayReiche123 | NPR is unquestionably liberal... and it's very hard to give specific examples... it's more about the programming. They have daily stories about the misery in Iraq but never tell 1 story about something positive that may have come out of the Iraqi war. |
| 18:15.41 | *** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 18:15.44 | *** join/#asterisk Dark40rce (~vince@dsl-17-106.cofs.net) |
| 18:15.47 | CosmicRay | jedaustin: well, like I said, please find me an example. I've known her to tolerate even extreme nutcases from both the left and the right. I know that she does hang up on conspiracy theorist nutcases, regardless of political persuasion |
| 18:15.52 | jedaustin | ClayReiche123: Right. It's a lot in the side hand things they say. |
| 18:15.53 | *** join/#asterisk ChulJin (~chuljin@65.211.236.166) |
| 18:15.58 | ManxPower | and none of the media show the body bags coming home. |
| 18:16.25 | jedaustin | ManxPower: the point of that would be??? |
| 18:16.31 | CosmicRay | ClayReiche123: just this morning, they ran a story about the first ever freely-elected parliament in iraq |
| 18:16.40 | CosmicRay | ClayReiche123: and I have been hearing about the run-up to this for some time |
| 18:16.41 | Redb3ard | clayreich, its not an easy comparison... so what if some school was rebuilt, if 3 of the children that would have attended it were smeared with smartbombs a year ago? the "oops are weapons arent perfect" excuse wears a little thin, id think |
| 18:16.45 | Tall-guy | Hey, did you know that if you listen to a conference one day early...no one is there.???? :) |
| 18:17.10 | ManxPower | jedaustin, I'm not sure, but it's certinally as newsworthy as Bernie ebbers going to prison. |
| 18:17.10 | *** join/#asterisk machinehd (~machinehd@storm.bcgroup.net) |
| 18:17.12 | JerJer[mobile] | did i join #politics this morning? |
| 18:17.16 | CosmicRay | ClayReiche123: they also frequently interview regular iraqis. It's not NPR's fault that they tend to want Americans out. |
| 18:17.16 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:17.54 | ManxPower | JerJer[mobile], It all started with rvhi asking me why I want to move to Europe. |
| 18:18.00 | G0shen | JerJEr: I was aking myself the same question |
| 18:18.24 | rvhi | actually i am thinking that too |
| 18:18.29 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 18:18.31 | ClayReiche123 | okay... my asterisk is broke |
| 18:18.38 | Redb3ard | its hard to know what to think, if we just wanted to spread freedom, there were 20 other countries just as bad... im supposed to believe its random coincidence that iraq was singled out? |
| 18:18.47 | CosmicRay | ManxPower: not that things are any better over there. have you seen about 50% of the british press? or have to read burlosconi's papers? :-) |
| 18:18.52 | Jearil | ClayReiche123: join the club ;) |
| 18:18.56 | JerJer[mobile] | they drive on the wrong side of the road |
| 18:19.01 | Tall-guy | manx: come to canada, the beef is safe!!!! |
| 18:19.02 | ClayReiche123 | Mar 16 12:56:51 WARNING[10157]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) |
| 18:19.02 | G0shen | ClayReich: Get a new keyboard :) |
| 18:19.09 | ClayReiche123 | hehe |
| 18:19.19 | JerJer[mobile] | ClayReiche123: allow=g729 |
| 18:19.29 | JerJer[mobile] | and buy a license |
| 18:19.37 | CosmicRay | Redb3ard: I also hear a lot of comments from iraqis like, "well, we're free, but my uncle was killed in the war and my wife was killed by insurgents. at least we were alive with saddam." |
| 18:19.39 | ClayReiche123 | why won't it pass thru? |
| 18:19.50 | ManxPower | ClayReiche123, either buy a license or don't put t/T on the Dial line |
| 18:19.57 | ClayReiche123 | ahhhhhh |
| 18:20.02 | JerJer[mobile] | because somethng wants ulaw |
| 18:20.08 | ManxPower | ClayReiche123, You cannot pass thru if you need Asterisk to listen to DTMF OR play OR record any audio. |
| 18:20.16 | Redb3ard | if theyre so free, why doesnt the US listen when those people say "get the hell out of my country" ? |
| 18:20.24 | ManxPower | OR if all legs of the call are not the same codec. |
| 18:20.34 | Redb3ard | theyre not free, not in any real sense |
| 18:20.52 | JerJer[mobile] | the word 'free' is dramatically abused |
| 18:21.13 | Corydon-w | Heh, we've been telling Bush to get out of our country for years... he hasn't paid any attention... |
| 18:21.13 | Samurai_ka | jerjer: like free software :-) |
| 18:21.16 | ManxPower | CosmicRay, Because if the usa packed up and left at this point things would be far worse than they were. |
| 18:21.19 | Corydon-w | Guess I'm not really free |
| 18:21.20 | *** join/#asterisk afe ([Rzn0rtQc6@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:21.42 | Redb3ard | yeh, but i do mean it in the real meaning of the word, that is, free to live their lives how they want, without fear of systematic or institutional abuse |
| 18:22.04 | Redb3ard | manxpower, i dont even dispute that it could become worse |
| 18:22.17 | ManxPower | Anyway, I've lived in the USA for 35 years, I want to try living in Europe for 35 years. Then I can compare. |
| 18:22.25 | Redb3ard | but when the people chafe at their new "democratic government" and start doing things that upset the balance |
| 18:22.49 | ManxPower | Canada would be a good option too, of course. |
| 18:22.51 | ClayReiche123 | crap! I don't have t or T in my dial plan... |
| 18:22.53 | Redb3ard | are they going to be allowed to turn it into whatever they want, or will the provisional government start locking them up, with US backing, just to "maintain order" ? |
| 18:22.58 | Qwell | ManxPower: Save that for 70-105 |
| 18:23.02 | JerJer[mobile] | ClayReiche123: something wants 4 which == ulaw |
| 18:23.04 | ManxPower | ClayReiche123, one of the devices is not using g279 |
| 18:23.18 | ClayReiche123 | ok... could be my gateway? |
| 18:23.36 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 18:23.44 | ManxPower | ClayReiche123, don't allow=ulaw |
| 18:23.57 | ClayReiche123 | ok I'll try that |
| 18:24.09 | ManxPower | ClayReiche123, Trust me, just spend the money on Digium G729 licenses and stop dealing with passthru problems |
| 18:24.19 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 18:25.14 | ClayReiche123 | I have 50, but I'm using them on my voicemail server. |
| 18:25.42 | rephorm | is there a way to dial an outside line, and bridge them to an inside extension from the manager interface? |
| 18:26.04 | ClayReiche123 | I got em in 25 packs... maybe I'll take 25 and put them on this server to test. |
| 18:26.07 | Juggie | ClayReiche123, whats the benefit of g729 on your voicemail server |
| 18:26.38 | ClayReiche123 | Juggie: that's where it HAS to be in order to decode g729 and write to disk. |
| 18:26.45 | Tall-guy | good lord...I never used that "site:" option in google before....man that's handy. |
| 18:26.47 | christo | guys, I have a number routed to an E1 terminating at one system we have installed. It works. I can call it, asterisk answers the call and dives into the dial plan. Great. I have a similar system elsewhere, another E1 plugged in, a routed number to play with, but where do I start in the configs at the lowest level to get * to answer the call as described on the first server? :S |
| 18:26.53 | *** part/#asterisk BurnedOutGeek (~BurnedOut@216.215.202.4.nw.nuvox.net) |
| 18:26.55 | Juggie | do sip/iax clients connect direct to your vm server? or do they go to another * server first? |
| 18:27.10 | Corydon-w | Tall-guy: wait until you see what the plus sign does... |
| 18:27.22 | Corydon-w | and the minus sign.. |
| 18:27.36 | ClayReiche123 | juggie: they go to another * server first. |
| 18:27.47 | ClayReiche123 | the one I'm having pasthru problems with. |
| 18:28.00 | Juggie | i dont see the point to that config |
| 18:28.05 | Juggie | you've broken a ton of stuff now |
| 18:28.11 | Juggie | moh, conferencing etc |
| 18:28.12 | ClayReiche123 | no sir |
| 18:28.13 | Tall-guy | corydon: ok, now yer bein mean :) |
| 18:28.15 | *** join/#asterisk jeffik (~jeffik@208-41-192-106.client.dsl.net) |
| 18:28.32 | Juggie | you would be better off putting the g729 licenses on the first * box |
| 18:28.41 | ChulJin | tall-guy: no, + and - do things |
| 18:28.45 | ClayReiche123 | juggie: I'm ok with that. I'm building a carrier-type system. Not a PBX |
| 18:28.47 | Juggie | and let vm server just answer in ulaw transcoded |
| 18:29.11 | ClayReiche123 | juggie: That's a good Idea... never thought of that.... |
| 18:29.25 | ManxPower | Tall-guy, Um, you've never read the google docs page? |
| 18:29.37 | Juggie | ClayReiche123, that way you get control of the call at least.. you can play a messsage or do something... |
| 18:29.40 | ClayReiche123 | Will 1st server switch to ulaw when sending to vmail? |
| 18:29.42 | Juggie | with passthrough you cant. |
| 18:29.56 | Tall-guy | I'll rephrase: I thought I knew ALL the options...but didn't know "site" yeah, I know + and ~ :) |
| 18:30.03 | Juggie | yes, the 1st server will transcode from g729 to ulaw |
| 18:30.37 | Juggie | ClayReiche123, if you also want that first server to have any telephone lines in it |
| 18:30.54 | Juggie | or connect via/sip or iax to any non g729 sites |
| 18:31.01 | heison | ~seen JerJEr |
| 18:31.14 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 17d 2h 12m 19s ago, saying: 'mrgoby: sure'. |
| 18:31.14 | Juggie | you will need your g729 licenses on there. |
| 18:31.22 | ManxPower | Tall-guy, http://www.google.com/help/features.html |
| 18:31.42 | Tall-guy | manx: yeah, I'm all up to speed now... |
| 18:32.06 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 18:33.41 | ClayReiche123 | juggie:this is a straight SIP environment with no phone lines in the asterisk servers. I will scale this to about 6 main servers all writing vmail to 1 server. It's not very cost effective for me to buy g729 codecs for 6 servers when, theoretically I should only need them when writing to disk on the vmail server. |
| 18:34.17 | Juggie | ClayReiche123, if this is only straight SIP why not use SER as your sip server with * as your vm server. |
| 18:34.45 | ClayReiche123 | juggie: my TAs have g729 license and my Cisco gateway has g729 license... there should be no need for me to but them for asterisk. |
| 18:35.19 | ClayReiche123 | buy |
| 18:35.23 | Juggie | ClayReiche123, as i said check out SER, you would be much better off registering your phones to that and using * for just VM |
| 18:35.40 | ClayReiche123 | Why is that? |
| 18:35.53 | Juggie | SER is much higher performance for just passing around sip packets |
| 18:35.55 | Juggie | then * is. |
| 18:36.25 | SwedMiro | Is SER harder to set up than * ? |
| 18:36.33 | Juggie | its not an asterisk replacement, but in large instalations and 6+ servers sounds like 1000+ users |
| 18:36.43 | Juggie | SER removes * from the RTP path |
| 18:37.02 | ClayReiche123 | man... that sounds nice... |
| 18:37.09 | rob- | win prev |
| 18:37.21 | SwedMiro | ahh..then no prob,..my biggetst installation is 25 extensions :) |
| 18:37.50 | Juggie | ClayReiche123, http://www.voip-info.org/wiki-SIP+Express+Router |
| 18:38.14 | Tall-guy | or you can google "SER site:www.voip-info.org" : |
| 18:38.15 | Tall-guy | :) |
| 18:38.16 | ClayReiche123 | I thought canreinvite=yes removed asterisk from the TRP path? |
| 18:38.20 | ClayReiche123 | RTP |
| 18:38.33 | Juggie | Not that i am aware... RTP still remains with * |
| 18:38.37 | Juggie | the voice goes phone to phone |
| 18:38.43 | ClayReiche123 | ahhh |
| 18:38.52 | Juggie | but the control path for the call goes phone->*->phone |
| 18:40.31 | *** part/#asterisk Samurai_ka (~Samurai_k@ad96edd8b.dsl.de.colt.net) |
| 18:41.03 | ClayReiche123 | Juggie:Sounds like I should have talked to you several months ago.....or done a little more homework. I've already spent much resources scaling asterisk and building a rating engine... If I could just get this codec problem fixed, I would be fine. |
| 18:41.42 | SwedMiro | Clay..on the other hand you have learnt a lot by doing it..knowledge is an easy burdon to bear :) |
| 18:41.50 | SwedMiro | except in the form of books |
| 18:41.58 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-25-189.d4.club-internet.fr) |
| 18:42.00 | SwedMiro | bare |
| 18:42.02 | SwedMiro | beere |
| 18:42.07 | SwedMiro | bair |
| 18:42.09 | Juggie | ClayReiche123, i can give you no advise on passthrough |
| 18:42.11 | Juggie | i dont use it |
| 18:42.19 | Juggie | i am on a high speed network so i use g711 |
| 18:42.28 | Juggie | unless your phones are on a slow remote lan |
| 18:42.34 | Juggie | i suggest you stick to g711 as well |
| 18:43.02 | Juggie | it would also be worth noteing that with g729 those cisco phones are unable to conference |
| 18:43.39 | sudhir492 | anyone using te110p card? |
| 18:43.47 | ClayReiche123 | Not phones. Cisco 5350 gateway and WorldAccxx TA200s. |
| 18:44.19 | Juggie | ClayReiche123, you must have phones as voip clients? |
| 18:44.31 | ClayReiche123 | Telephone aapters. |
| 18:44.36 | ClayReiche123 | adapters |
| 18:44.50 | Juggie | they too may be unable to conference |
| 18:45.05 | JerJer[mobile] | just buy a few G.729 licenses, yo |
| 18:45.05 | Juggie | i'm pretty sure the 7960 cannot |
| 18:45.05 | ClayReiche123 | I will check. |
| 18:45.34 | PatrickDK | heh, I just use my cisco router as clients |
| 18:45.48 | Juggie | JerJer[mobile], what he says hes doing makes sense, he doesnt need licenses on the servers that should just pass the data |
| 18:45.56 | Juggie | its the vm that should need to decode |
| 18:46.35 | ClayReiche123 | My company manufactures the WorldAccxx TA200 tephone adapters. We spend money on g729 when manufacturing the product, and we spend money on g729 when purchasing our Cisco gateways.... |
| 18:46.40 | JerJer[mobile] | then the box that does vm will need licenses |
| 18:46.52 | Juggie | jer, he has them installed |
| 18:47.00 | ClayReiche123 | ... I would be fired if I had to spend money on g729 for the softswitch... :) |
| 18:47.01 | *** join/#asterisk implicit (~implicit@dhcp-118-228.mobile.uci.edu) |
| 18:47.02 | Juggie | but he is having problems with passthrough |
| 18:47.08 | JerJer[mobile] | then alllow=g729 |
| 18:47.14 | JerJer[mobile] | someone wants ulaw |
| 18:47.19 | JerJer[mobile] | very simple |
| 18:47.23 | PatrickDK | passthrough won' work if you set almost any options to dial |
| 18:47.32 | PatrickDK | youcan't use rtTm... |
| 18:47.43 | implicit | or rtFm |
| 18:47.44 | PatrickDK | you can't use music on hold, or any menus |
| 18:47.54 | JerJer[mobile] | codecpriority=disable |
| 18:47.55 | Juggie | PatrickDK, i already told him all that. |
| 18:47.57 | JerJer[mobile] | disallow=all |
| 18:47.59 | JerJer[mobile] | allow=g729 |
| 18:48.37 | ClayReiche123 | My dial looks like this exten => 1234,1,Dial(SIP/1234,20) |
| 18:48.53 | JerJer[mobile] | then whoever is calling 1234 wants ulaw |
| 18:49.11 | Juggie | ClayReiche123, whats the error on the console? |
| 18:49.20 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 18:49.27 | enemy^x | what hardware phones do you recommend? I would like transfer, conference, hold etc. |
| 18:49.34 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 18:49.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:49.39 | JerJer[mobile] | 7960 or nothing |
| 18:49.49 | ClayReiche123 | I'm no longer getting error on the console. I'm just getting very garbled incoming audio. |
| 18:49.50 | Juggie | 7960 are ok, but not admin friendly |
| 18:49.56 | enemy^x | isn't the 7960 fairly old? |
| 18:50.01 | Juggie | yes |
| 18:50.05 | Qwell | Doesn't mean its not good |
| 18:50.08 | Juggie | enemy^x, depends on your budget |
| 18:50.09 | JerJer[mobile] | more like mature |
| 18:50.12 | ClayReiche123 | Sorry, I should have kept you up on my progress. |
| 18:50.16 | Qwell | JerJer[mobile]: like Debian |
| 18:50.19 | Juggie | try the mitel 5215/5220 |
| 18:50.35 | Juggie | ClayReiche123, garbled audio shouldnt be *'s fault |
| 18:50.38 | Juggie | its just passing it |
| 18:51.00 | enemy^x | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61840&item=5760741054&rd=1 |
| 18:51.04 | enemy^x | is that a good offer? |
| 18:51.17 | JerJer[mobile] | if there is no nat then the rtp should be going endpoint to endpoint |
| 18:51.42 | Qwell | enemy^x: I've seen better, honestly |
| 18:51.44 | JerJer[mobile] | provided you have not dsabled that feature |
| 18:52.02 | enemy^x | qwell: thank you |
| 18:52.06 | afe | ClayReiche123: What codecs do you use in both ends of the call? |
| 18:52.07 | Qwell | but it looks decent. ~$250 each. If it doesn't come with the power adapters, don't bother |
| 18:52.17 | JerJer[mobile] | or user license |
| 18:52.27 | JerJer[mobile] | and if they are not a cisco authorized reseller |
| 18:52.33 | Qwell | enemy^x: It'll probably be a little more difficult to get 25 at a "good" price. |
| 18:53.00 | Juggie | ClayReiche123, when a call is live, do 'sip show channels' to see what * thinks the codec inuse is |
| 18:53.05 | enemy^x | qwell, heh, I'll surf some more on ebay then :) |
| 18:53.22 | Qwell | enemy^x: Try some real retailers first...get a feel for the price |
| 18:53.45 | enemy^x | qwell: already did that, the norwegian retailers prices are waste. |
| 18:53.54 | enemy^x | sorry for bad english anyways |
| 18:55.14 | ariel_ | hello everyone |
| 18:55.23 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 18:55.40 | SwedMiro | I managed to talk a company into using only softphones..but it made sense..they didnt have a set workspace..so they got their phone with their logon to the net |
| 18:55.48 | SwedMiro | they are very satisfied |
| 18:56.37 | wildcard0 | so anyone have any luck using a stun server to traverse a linksys embedded router? |
| 18:57.11 | wildcard0 | i have 4 devices behind one and it seems to be losing some |
| 18:57.31 | wildcard0 | also, if i have stun set up, should i still set nat=1? |
| 18:58.27 | ManxPower | wildcard0, Almost NOBODY need STUN, except maybe serrvice providers. |
| 18:59.02 | ManxPower | SwedMiro, Must be nice. Many of the people at my largest customer don't even have a computer. |
| 18:59.21 | wildcard0 | ok so what would you suggest then? i have 4 devices behind a linksys box. the server is off across the net with a routable ip. it's losing the phones after a short time |
| 18:59.24 | ManxPower | wildcard0, make sure you put qualify=yes. |
| 18:59.29 | wildcard0 | it's there already |
| 18:59.43 | ManxPower | wildcard0, yes, that is common if you don't have qualify=yes. |
| 18:59.53 | ManxPower | you can, of course, set the registration time on the phones to be 60 seconds |
| 18:59.54 | wildcard0 | yes qualify=yes already |
| 19:00.16 | wildcard0 | then the phones will be out for a max of 59 seconds. that's still not working... |
| 19:00.23 | Juggie | mitel 5215 crashes if qualify=yes isnt on |
| 19:00.31 | PatrickDK | wildcard, out for a max? |
| 19:00.35 | PatrickDK | no that isn't the point |
| 19:00.42 | PatrickDK | the point it to keep the firewall link active |
| 19:00.50 | PatrickDK | it timesout after 3min I think |
| 19:00.54 | Juggie | er, sorry.. mitel 5055 |
| 19:00.57 | Juggie | their first sip phone |
| 19:01.02 | Juggie | crashes without qualify=yes |
| 19:01.32 | *** join/#asterisk omarc55 (~omarc55@adsl-2-210-106.mia.bellsouth.net) |
| 19:02.04 | *** join/#asterisk Hydroponx (hidden-use@nat.wwisp.com) |
| 19:02.12 | afe | anyone can recommend good headsets for use with a pc and softphone (X-lite probably)? |
| 19:02.13 | Hydroponx | anyone tell me what ports skinny uses ?/ |
| 19:02.15 | SwedMiro | ManxPower..well..the company is a company that teaches computer knowledge..so..ehh..everybody..even the customers gets to sit by a puter |
| 19:02.21 | Hydroponx | or more specifically rtp |
| 19:02.27 | bannerman | Um.. can I use like.. any music that I own, for hold music? |
| 19:02.27 | wildcard0 | afe, plantronics ones sound good |
| 19:02.50 | afe | wildcard0: they are available with USB, right? |
| 19:02.50 | bannerman | Or do I have to purchase a special license, or royalty free music |
| 19:02.54 | wildcard0 | ManxPower, so you're saying that a stun server doesn't solve that problem? then what problem does it solve? |
| 19:02.59 | SwedMiro | Afe..when it comes to headsets buy the cheapest ones..the sound is really good enough for phone use in those.. |
| 19:03.10 | wildcard0 | afe, i believe they're available in both USB and mic/speaker |
| 19:03.14 | ManxPower | Hydr0p0nX, Welcome to one of the big issues with VoIP. RTP ports are dynamically determined by each side. |
| 19:03.23 | ManxPower | wildcard0, I strongly doubt STUN will help |
| 19:03.37 | wildcard0 | ok. so what then does stun do if not that? |
| 19:03.42 | afe | SwedMiro: I suppose, but I'd prefer USB so people can still use their speakers for other purposes (and to hear the ring signal) |
| 19:03.51 | PatrickDK | wildcard, setup static port redirects |
| 19:04.01 | wildcard0 | i've been reading the rfc and it seems to try to solve exactly this problem |
| 19:04.01 | PatrickDK | and have each phone use their own range |
| 19:04.21 | SwedMiro | Afe..that can be solved with a mediahub-switch |
| 19:04.23 | ManxPower | wildcard0, STUN makes SIP clients behind NAT work correctly. However, nat=yes + canreinvite=no does something very similar |
| 19:04.37 | wildcard0 | ya i already have that set |
| 19:04.51 | wildcard0 | i'll try shortening the registration time |
| 19:05.02 | ManxPower | wildcard0, All bets are off if you start doing port forwarding on your router. turn port forwarding off and let nat=yes do it's work. That's what it's there fore. |
| 19:05.03 | wildcard0 | actually, i might as well do both that and a stun server |
| 19:05.15 | Hydroponx | so there's no easy way to do qos based on port number/protocol huh ? |
| 19:05.19 | wildcard0 | im not doing any port forwarding. tho it did cross my mind |
| 19:05.43 | wildcard0 | Hydr0p0nX, http://lartc.org/howto/ |
| 19:05.49 | Hydroponx | thanks |
| 19:05.59 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 19:06.02 | ManxPower | Hydr0p0nX, I configure all my SIP and IAX devices to set the DSCP value in the packet header to "EF", but before that I configured all the devices to use pirts 16384-32768 (which is what Cisco SIP defaults to) |
| 19:06.34 | ManxPower | (thats the TOS value ox 0x6b aka 184) |
| 19:06.44 | Hydroponx | i see |
| 19:06.56 | afe | SwedMiro: true :) |
| 19:07.15 | SwedMiro | I tend to go for the cheap solutions :) |
| 19:07.30 | ManxPower | Now if my ISP can get the damn QoS guy to call me back I'll be much happier. |
| 19:07.51 | SwedMiro | the customer is happy..but when they get to understand what they need..i can sell them more expensive stuff |
| 19:07.52 | SwedMiro | :) |
| 19:09.02 | sudhir492 | What kind of cable to use with te110p? |
| 19:09.15 | ChulJin | a friend whom I introduced to * has started selling calling cards. |
| 19:09.15 | Essobi | JerJer[mobile] So when you going to pass around that mini linux install you're hoarding? :) |
| 19:09.24 | ChulJin | leaving the nest...brings a tear to one's eye. |
| 19:09.25 | ChulJin | :P |
| 19:09.26 | ManxPower | sudhir492, ethernet straight thru will work for short distances |
| 19:09.58 | JerJer[mobile] | Essobi: when its ready for distro |
| 19:10.09 | sudhir492 | ManxPower: Are you sure it is ethernet straight for te110p? |
| 19:10.11 | ManxPower | Fir a T-1 crossover you need to custom make it, diagrams are....quess where...on the wiki |
| 19:10.13 | PatrickDK | short being <10feet |
| 19:10.30 | ManxPower | sudhir492, you can use an etuerhet straight thru in pace of a T-1 straight thru |
| 19:10.37 | Essobi | JerJer[mobile] You going to have a prebuilt chan_h323? ;) |
| 19:10.57 | Essobi | Hmm.. Busybox based? |
| 19:11.04 | sudhir492 | ManxPower: I am currently using t1 crossover, and get the alarm BAD FCS(8) on Primary D channel |
| 19:11.11 | ManxPower | sudhir492, i.e. telco -> asterisk = straight thru. Asterisk -> channel bank = crossover |
| 19:11.27 | bkw_ | really? |
| 19:11.28 | bkw_ | how mini? |
| 19:11.30 | ManxPower | sudhir492, Didn't I just say that ethernet cable would not work for a T-1 crossover |
| 19:11.39 | *** join/#asterisk miguel3239 (~myoung@ip66-105-121-66.z121-105-66.customer.algx.net) |
| 19:11.49 | Essobi | bkw_ Ask jer jer.. I think he said 30megs last time he mentioned it. |
| 19:12.05 | spackle | bkw_ do you know of a telecom-based mini iso? |
| 19:12.07 | ManxPower | "Mouse Linux: The Distro With The Black Plague!" |
| 19:12.10 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 19:12.18 | Essobi | ManxPower Haha. |
| 19:12.21 | ChulJin | I thought it was rats. |
| 19:12.22 | sudhir492 | ManxPower: thanks. I am connecting telco to asterisk, no channel bank. Hence should ask those guys to connect t1 staright, or rathere Ethernet straight? |
| 19:12.28 | bkw_ | spackle, why yes |
| 19:12.29 | heison | JerJer: check your email... |
| 19:12.30 | bkw_ | yes I do |
| 19:12.32 | bkw_ | its almost done |
| 19:12.37 | bile_one | Is Eric alive? |
| 19:12.43 | ManxPower | sudhir492, i.e. telco -> asterisk = straight thru. Asterisk -> channel bank = crossover |
| 19:12.46 | Essobi | bkw_ You're making one too? Sweet. |
| 19:12.48 | bkw_ | its was going to be asterlinux.. but we renamed it PBlX |
| 19:12.52 | ChulJin | though I guess, anything that can carry fleas. |
| 19:12.55 | ManxPower | bile_one, Which Eric |
| 19:12.59 | PoWeRKiLL | Hi bkw |
| 19:13.09 | PoWeRKiLL | You release a new version of asterlinux ? |
| 19:13.12 | Essobi | You should call it MOAV |
| 19:13.13 | Essobi | :) |
| 19:13.14 | bkw_ | -rw-r--r-- 1 root root 161808384 Mar 16 23:36 pblx-dev.iso |
| 19:13.14 | bkw_ | -rw-r--r-- 1 root root 116555776 Mar 16 23:57 pblx-usr.iso |
| 19:13.15 | spackle | anybody use Digium G729 with a sipura SPA-841 yet? |
| 19:13.22 | bile_one | Eric W, the Cepstral guru! |
| 19:13.39 | sudhir492 | ManxPower: thx. Just wanted to confirm that is the case with te110p card too. I have no problem with t100p, or te405p cards. |
| 19:13.43 | bkw_ | mine is kinda larger.. but i'm not going for micro distro |
| 19:13.53 | bkw_ | i'm going for featureful |
| 19:14.03 | Essobi | :) |
| 19:14.06 | bkw_ | oh I can make it smaller |
| 19:14.10 | bile_one | ManxPower I haven't seen you in quite some time. |
| 19:14.17 | ManxPower | sudhir492, the wireing is the same for ANY T-1 device |
| 19:14.32 | Essobi | bkw_ I was going to make a joke on that, but I'll just leave it alone. |
| 19:14.32 | Juggie | bkw, is that a custom distro you made? |
| 19:14.33 | ManxPower | bile_one, That's because I got tired of answering stupid questions all day. |
| 19:14.56 | bile_one | Aye! I sent you an email did you not get it? |
| 19:15.16 | ManxPower | bile_one, Was the from address "bile_one"? |
| 19:15.22 | Essobi | mehe |
| 19:15.49 | bile_one | ManxPower, no it was from my real addy. |
| 19:16.35 | Damin | bkw_: I've got about 30 boxes sitting here doing nothing and 4 guys from the VA that I'm being paid to assess their tech skills. Want them to install Asterlink on these boxes.. |
| 19:16.48 | ManxPower | bile_one, Which is...... |
| 19:17.31 | *** join/#asterisk r3d5un (~r3d5un@83-64-243-19.dynamic.xdsl-line.inode.at) |
| 19:17.42 | sudhir492 | I get the error PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 19:17.51 | *** join/#asterisk twilson (~terry@63.77.68.11) |
| 19:18.10 | sudhir492 | Using euroisdn, on te110p card |
| 19:18.41 | spackle | anybody here familiar with cluecon? |
| 19:18.48 | ManxPower | sudhir492, That means "I got corrupted data" That's USUALLY an interrupt beig lost, but COULD be a telco problem. |
| 19:19.01 | heison | just want to confirm - it is possible to use SayDigit() to repport callerID number on the systems' sound card, right? |
| 19:19.54 | ManxPower | heison, no. |
| 19:20.10 | heison | no? |
| 19:20.21 | ChulJin | no. |
| 19:20.25 | jontow | SayDigit takes a single digit? |
| 19:20.32 | jontow | SayNumber() may work, but may sound funny.. |
| 19:20.34 | ManxPower | SayDigit(${CALLERIDNUM}) will say the callerid number on the channel it's connected to, not the system sound card. |
| 19:20.47 | jontow | oh, on the soundcard |
| 19:20.49 | jontow | wtf.. no :( |
| 19:20.52 | heison | what if i have CONSOLE=/dev/dsp |
| 19:21.44 | heison | someone challenged me that Asterisk can't replace his off-the-shelf callerID announcement thing he picked up from Radio Shack |
| 19:21.45 | ChulJin | then it would say it on the console only if the call originated on the console |
| 19:21.57 | *** join/#asterisk omarc55 (~omarc55@adsl-2-210-106.mia.bellsouth.net) |
| 19:22.18 | ChulJin | it can, but not quite in the same way you're describing. |
| 19:22.27 | spackle | you could set the console to autoanswer, call the console and play the callerid |
| 19:23.06 | ManxPower | Only if you do one of the tricks. |
| 19:23.08 | r3d5un | why not use some AGI Prog to play the callerid directly with i.e festival |
| 19:23.21 | ChulJin | or not even festival... |
| 19:23.30 | sudhir492 | ManxPower: Instead of telco switch, when we connect the analyzer, we still see the same problem |
| 19:23.34 | ManxPower | Once you Dial the dialplan STOPS until the call ENDS |
| 19:23.41 | ChulJin | just play the digits files already included with * |
| 19:23.47 | ManxPower | This is for you, sudhir492 |
| 19:23.55 | heison | if i do that (call console with auto answer), can the call continue to ring until someone picks up a phone? |
| 19:24.05 | ManxPower | ~google HDLC Bad FCS site:lists.digium.com |
| 19:24.09 | ChulJin | heison: <ManxPower> Once you Dial the dialplan STOPS until the call ENDS |
| 19:25.08 | spackle | there should be a "spawn" command in the dial plan. |
| 19:25.22 | jontow | there is a System() ...? |
| 19:25.31 | ManxPower | you can get around that by using the M(macro) option to Dial or using .call files. |
| 19:25.34 | *** join/#asterisk afe ([lnZDTJsRl@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 19:25.38 | Inv_arp | hmm i wanna dial 100 numbers in a DB agi can do that right? |
| 19:25.53 | heison | so... it's doable... that Asterisk can replace one of those callerID announcement device, using AGI and plays it to a sound card |
| 19:25.54 | *** join/#asterisk RUok (~mike@host-69-19-166-10.rev.o1.com) |
| 19:26.01 | ChulJin | heison: yes. |
| 19:26.06 | RUok | anyone here having problems with chan_sip's outboundproxy support? |
| 19:26.28 | ChulJin | inv_arp: you wouldn't really even need agi...just make some little perl script that reads from your DB and makes .call files |
| 19:26.57 | ManxPower | YES! The replacement motherboard for my main workstation arrived! |
| 19:27.01 | Nugget | yay |
| 19:27.08 | tzanger | ManxPower: what happened to the original |
| 19:27.16 | ManxPower | Soon I won't have to use this *gagging noises* Windows machine. |
| 19:27.29 | ManxPower | tzanger, It never really worked right. |
| 19:27.39 | ChulJin | heison: with a little more coding, you could implement those boxes' other feature, of playing back a name that you've recorded |
| 19:27.44 | ManxPower | tzanger, random hard locks with Win98, Win2k, and Mandrake Linux |
| 19:27.47 | jontow | anyone played with ICD ? :) |
| 19:27.57 | spackle | ManxPower, tell us of your motherboard so we may pass judgement on it and deterimine if it is worthy of you. |
| 19:27.57 | tzanger | ManxPower: you ran memtest for a few hours? |
| 19:28.00 | heison | ChulJin: yup |
| 19:28.08 | ChulJin | manx: your machine makes gagging noises? you must have cheetah drives |
| 19:28.21 | tzanger | hahaha |
| 19:28.26 | Qwell | ChulJin: No, it was running Windows |
| 19:28.32 | jontow | btw, manxpower.. useful info on the HDLC Bad FCS :) |
| 19:29.07 | ManxPower | tzanger, Um, EVERYTHING has been replaced EXCEPT for the motherboard. |
| 19:29.09 | modulus_ | hi chuljin |
| 19:29.15 | ChulJin | hi modulus_ |
| 19:29.36 | modulus_ | los lonely boys - i walk the line |
| 19:29.41 | modulus_ | good song |
| 19:29.41 | tzanger | ManxPower: gotcha |
| 19:29.49 | ChulJin | make it your hold music |
| 19:29.58 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E4E7.versanet.de) |
| 19:30.00 | Nebukadneza | hi dudes |
| 19:30.03 | ChulJin | er, why am I telling you? I'll just do it. |
| 19:30.14 | ManxPower | tzanger, My significant other always has computer problems, always hardware. I replaced/rebuilt his machines several times, always something else wrong. Finally I gave him my Dell and decided to run his old machine myself. |
| 19:30.22 | omarc55 | Hi all. I just got my TE110P and I am trying to get it to work with my TSU 600 but I can't get the FXS ports to give me a dial-tone, I already enabled dial-tone on the Adtran but I don't know if I need to use another option to reload the configuration. Does anybody know what I could be doing wrong? |
| 19:30.23 | modulus_ | chuljin, got dual xeons for asterisk |
| 19:30.31 | ChulJin | nice |
| 19:30.53 | ChulJin | is that new? if memory serves, you used to have something not much better than my paperweight |
| 19:30.53 | modulus_ | and another dual xeon for the mysql |
| 19:31.02 | ManxPower | tzanger, The last problem was that the damn power switch on the front was broken and so causing a short and so the system powered off 2 seconds after it was powered on. THAT took me a week to figure out. |
| 19:31.07 | modulus_ | chuljin yeah it's on a 6 year old machine right now |
| 19:31.17 | spackle | chuljin, got dual xeons for asterisk |
| 19:31.21 | ManxPower | You never think of a powerswitch causing a problem |
| 19:31.30 | jontow | on ATX i do :( |
| 19:31.30 | spackle | is asterisk a holiday like Christmas? |
| 19:31.39 | Pkunk | i've seen reset switches getting shorted |
| 19:31.40 | tzanger | ManxPower: *nods* -- almost all of my computer problems end up being RAM. Memtest86 identifies it usually within 1 minute but I run it overnight in the extended tests just to really test the shit out of it. The only other problem I've really noticed was poor thermal circuit on the AMD chips... using a good heatsink and properly installing it solved that |
| 19:31.44 | sudhir492 | ManxPower: thx. I had checked the voip wiki, and did everything accordingly. from google, looks like te110p card could have a problem. My other box with te405 runs smoothly under similar circumstances |
| 19:31.53 | Pkunk | they're even harder to detect |
| 19:31.57 | ChulJin | spackle: congrats. I have a 500Mhz PIII |
| 19:31.58 | ChulJin | :P |
| 19:32.07 | modulus_ | lol |
| 19:32.20 | tzanger | Pkunk: that's why I never hook that shit up. power switch on the power supply and BIOS set to always come on after failure |
| 19:32.37 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 19:32.40 | Nebukadneza | i got a problem with iaxcomm ... i got a es-1938 (solo1) and got really _heavy_ noise problems with iaxcomm, asterisk and this soundcard :) |
| 19:32.42 | ChulJin | modulus_: promise you'll make a cleaner, prettier dialplan :P |
| 19:32.45 | Eight | ~wiki-status |
| 19:32.46 | jbot | well, wiki-status is Up and Running |
| 19:32.52 | enemy^x | Im trying to connect to asterisk using a softphone, sjphone in this test, but I don't see the softphone registering in the asterisk logs? |
| 19:33.06 | modulus_ | chuljin, it's going into the database |
| 19:33.10 | ChulJin | ah |
| 19:33.12 | ChulJin | that works |
| 19:33.13 | spackle | Eight: that's spiffy. |
| 19:33.14 | modulus_ | asterisk config is braindead |
| 19:33.22 | ChulJin | will |
| 19:33.23 | ChulJin | oh my |
| 19:33.26 | afe | anyone here with pocketpc experience that could help me with a quick off topic question? |
| 19:33.29 | Skysky | does anyone know wat this warning means? "WARNING[4735]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 0b2dcbbb32d407256ddf257e59233693@myip for seqno 102 (Critical Request)" |
| 19:33.30 | ChulJin | will you continue to use dundi? |
| 19:33.30 | Nebukadneza | are there any solutions to this? |
| 19:33.41 | ChulJin | afe: I can certainly try... |
| 19:33.55 | Nugget | jeez |
| 19:33.57 | modulus_ | chuljin it'll stay on current machine |
| 19:34.05 | ChulJin | ah |
| 19:34.11 | *** join/#asterisk dave_mwi_ (~dave_mwi@harpo.dreamhost.com) |
| 19:34.23 | Apple | does anybody here know how to get jack working with kphone? |
| 19:34.26 | Nugget | I got an email that basically says "hey nugget, I saw your asterisk pages. I have a quick question: what's a dialplan? thanks" |
| 19:34.38 | ManxPower | ChulJin, That's just sick. |
| 19:34.52 | ChulJin | manx: tehee |
| 19:35.05 | spackle | ~dialplan |
| 19:35.08 | jbot | well, dialplan is something that is planned to be rewritten in VB.net, or no! it's gonna be rewritten in brainf!(k |
| 19:35.08 | ManxPower | Nugget, donate your asterisk docs to the asteriskdocs.org people and help consolitdate the asterisk documentation sites. |
| 19:35.33 | Nugget | my docs aren't really docs, though, they're just a journal of what I've done. |
| 19:36.10 | ChulJin | jbot: that, too, is just sick, I'm guessing ManxPower will say. |
| 19:36.11 | jbot | ChulJin: I think you lost me on that one |
| 19:36.37 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 19:36.46 | ChulJin | modulus: so you're going to have your current * do what it does now, and use these new machines for that new endeavour? |
| 19:36.53 | RoyK | ~seen kpfleming |
| 19:36.54 | jbot | kpfleming is currently on #asterisk (4h 20m 56s) |
| 19:37.31 | *** part/#asterisk Alexis (~alexis@www.trim.it) |
| 19:37.36 | dave_mwi_ | hi...I've been looking for a couple hours now for some info on asterisk call files and haven't had much luck...I'm trying to find out if you can write call files that basically do the same thing as extensions like in the extensions .conf file...like play one extension, followe by another - all specified in the call file. The reason behind this is I'm going to have to generate call files on the fly...based on conditional checking in a php script.. |
| 19:38.18 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 19:38.36 | tzanger | dave_mwi_: doesn't work that way |
| 19:38.52 | heison | ~seen JerJer |
| 19:38.53 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 17d 3h 19m 58s ago, saying: 'mrgoby: sure'. |
| 19:39.02 | heison | ~seen JerJer[mobile] |
| 19:39.03 | jbot | jerjer[mobile] is currently on #asterisk (3h 58m 39s). Has said a total of 51 messages. Is idling for 29m 5s |
| 19:39.22 | PTG123 | hey anyone know how rto make my cisco 7960 unlock so i can change net settings? it was **# but since i upgraded that doesn't work |
| 19:39.38 | Nugget | PTG: tftpd |
| 19:39.45 | tzanger | dave_mwi_: call files take a "source" for the call (usually Local/SOME_NUMBER@some_context) and dial a number in a specified context. That's it. |
| 19:39.48 | PTG123 | Nugget: what? |
| 19:39.56 | dave_mwi_ | tzanger: we are using call files right now for simpler stuff, like use this context, play this file, then the context handles touch tone responses...but I need to have a lot more control over what would happen from the context standpoint than I think the context logic can handle itself |
| 19:39.58 | Nebukadneza | hm ... noone with iaxcomm noise problems? :( |
| 19:40.23 | Juggie | PTG123, press settings, press 9 password is cisco |
| 19:40.25 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 19:40.37 | Nugget | PTG123: suck it up and set up a tftpd server. I can't imagine trying to maintain even one cisco "by hand" |
| 19:40.41 | *** join/#asterisk {^_^}leeb (~Gleeb@81-178-169-240.dsl.pipex.com) |
| 19:41.00 | PTG123 | invalid password |
| 19:41.05 | dave_mwi_ | anyone know of some really good call file tutorials? I'm just having a hard time finding any. |
| 19:41.09 | PTG123 | Nugget: what i am trying to set is the tftp server |
| 19:41.12 | ManxPower | Nebukadneza, Those problems are almost always the fault of the soundcard. |
| 19:41.18 | Nugget | PTG123: set it with your dhcp server. |
| 19:41.26 | Juggie | PTG123, unless you changed it cisco is the default password |
| 19:41.34 | tzanger | dave_mwi_: I'm not following... can you give an example? All the logic is in the dialplan, the call file just allows you to easily "trigger" dialplan logic from an external source |
| 19:41.35 | Juggie | small letters |
| 19:41.42 | PTG123 | ah its the console pwrd which i set |
| 19:41.47 | PTG123 | k thanks :) |
| 19:41.48 | afe | anyone know of any PocketPC IAX client? I tried SJPhone and X-Pro, but neither works good enough on my device (skype works, but who wants to use that?) |
| 19:41.48 | Nebukadneza | ManxPower: hm ... its a es-1938 (ensonique solo1) and a via onboard soundcard |
| 19:41.52 | Nebukadneza | noise problems with both |
| 19:42.04 | Nebukadneza | ManxPower: is there any solution to this? |
| 19:42.27 | ManxPower | Nebukadneza, What OS? |
| 19:42.36 | Nebukadneza | linux (2.6.10 gentoo) |
| 19:42.48 | ManxPower | Nebukadneza, I have no suggestions. |
| 19:42.51 | Nebukadneza | and unable to build iaxcomm due to wxwindows problems |
| 19:43.16 | Nebukadneza | ManxPower: :( |
| 19:43.25 | ManxPower | All Softphones Suck. |
| 19:43.27 | Nebukadneza | ManxPower: is there any other good iax client out there |
| 19:43.30 | Nebukadneza | i know -.- |
| 19:43.32 | dave_mwi_ | tzanger, for example, I might create a call file with a series of texts to be said, which I would convert to sound files. Now, I want to separate those sound files with generic sound files sitting on the asterisk server. However the number of texts to be said in any one call file will vary |
| 19:43.37 | Nebukadneza | but if there were a good one it would be godlike :) |
| 19:43.58 | Hydroponx | ptg, try hitting **# from the main settings menu then go in and hit again on the tftp server optin |
| 19:44.04 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 RC - bug #3746 || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ |
| 19:44.09 | afe | ManxPower: I use both X-Pro and Diax on my PC:s and both work pretty fine :) |
| 19:44.29 | bannerman | I'm getting this weird notice res_musiconhold.c:472 monmp3thread: Request to schedule in the past?!?! |
| 19:44.38 | afe | assuming you happen to run Windows that is :D |
| 19:44.41 | dave_mwi_ | what I really need is a tutorial on call files...I'm having a hard time finding any |
| 19:44.42 | bannerman | and music on hold doesn't seem to be working, but I'm not certain that I know how to set up moh |
| 19:44.55 | tzanger | dave_mwi_: I haven't seen any myself |
| 19:45.10 | tzanger | but for that you could have the call file set a bunch of variables which are paths and filenames to play and have the context play it |
| 19:45.50 | bannerman | Anyone know what I might have broken to cause music on hold to not work? |
| 19:45.58 | dave_mwi_ | tzanger: yes I see that now....however....hmmmm |
| 19:46.00 | tzanger | bannerman: do you have a timing source? |
| 19:46.07 | bannerman | tzanger: no, probably not |
| 19:46.12 | tzanger | bannerman: do you have the right version of mpg123 |
| 19:46.30 | tzanger | asterisk is VERY Specific about it (I've never used MOH so I don't have many details... apparently the mpg123 output is very finicky) |
| 19:46.32 | bannerman | tzanger: I believe I do have the right version of mpg123 |
| 19:46.49 | bannerman | I compiled and installed it according to some instructions somewhere |
| 19:47.02 | bannerman | so about this timing source.. |
| 19:47.08 | bannerman | I don't have any digium hardware |
| 19:47.12 | tzanger | what are you using ot try and listen to it? |
| 19:47.22 | dave_mwi_ | tzanger: I guess my question is the call file won't know all the names of the variables I'm sending, because at one time there could be 5 at another 20 - and the context has to get to them by name right? |
| 19:47.23 | bannerman | just dialing my nufone number from a pots phone |
| 19:47.32 | bannerman | and then parking it |
| 19:47.39 | dave_mwi_ | tzanger: or do I just check for all 20 each time? |
| 19:47.48 | tzanger | dave_mwi_: you might have some luck with concatenating everything and have the dialplan loop, not really sure |
| 19:47.55 | pigpen | Not so sound like a moron, but If I have a single pri, and lets say 50 inbound did's...can I over subscribe the pri with voice services? |
| 19:48.08 | `Sauron | Yes |
| 19:48.11 | tzanger | it'd be far simpler to just cut the variable at a token, play the piece you ripped out, then loop back until no more tokens are found |
| 19:48.18 | *** join/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 19:48.21 | tzanger | pigpen: you can oversubscribe anything |
| 19:48.23 | `Sauron | make more than 23 calls at one time, and it's oversubscribed |
| 19:48.23 | pigpen | That is what I thought... |
| 19:48.26 | tzanger | but when you hit 23B you will get busies |
| 19:48.28 | dave_mwi_ | tzanger: ah... |
| 19:48.38 | pigpen | cool... |
| 19:48.42 | dave_mwi_ | tzanger: that's an interesting approach |
| 19:48.46 | pigpen | I am helping an isp with ideas... |
| 19:49.00 | tzanger | bannerman: hmm... I *think* you need a timing source for MOH but I thought MOH was timed from the audio stream |
| 19:49.06 | dave_mwi_ | tzanger: so play the file, rip that part out of the string and loop back and play the next file |
| 19:49.14 | dave_mwi_ | until there is no more to rip out? |
| 19:49.20 | tzanger | dave_mwi_: it's just an idea |
| 19:49.22 | *** part/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 19:49.47 | rvhi | is there a way to ring 2 numbers in a dial plan? |
| 19:49.56 | rvhi | e.g. call 700, ring both 701 and 702 |
| 19:50.14 | bannerman | tzanger: Is there such a thing as software timing? I thought I remembered reading something about that.. |
| 19:50.21 | *** join/#asterisk mrgoby (~mrgoby@exchange.synergybroadband.com) |
| 19:50.46 | tzanger | bannerman: yes there's zaprtc which uses the RTC (and fucks up your time I think) and ztdummy which requires a specific "type" of USB host controller |
| 19:51.50 | PTG123 | Juggie: is there an easy way to reboot these things without yanking power? |
| 19:52.28 | Moc____ | tzanger: ztdummy doesnt need a specific usb host controler IF you have kernel 2.6 |
| 19:52.45 | tzanger | Moc____: ooh this is news to me... I did not know that |
| 19:52.49 | afe | rvhi: exten => 700,1,Dial(SIP/701&SIP/702) |
| 19:52.54 | bannerman | so ztdummy is what I need to figure out |
| 19:52.57 | bannerman | cool |
| 19:53.08 | bannerman | or, a problem with the wrong version of mpg123.. most likely. |
| 19:53.18 | Moc____ | bannerman: , just remove the comment in the makefile |
| 19:53.24 | Moc____ | mpg123 doesnt need timing |
| 19:54.07 | Shido6 | can we get user specific iax2 debugs worked into the cvs somehow |
| 19:54.26 | tzanger | Shido6: eh? |
| 19:54.27 | tzanger | oh |
| 19:54.30 | tzanger | yes that would be nice |
| 19:54.36 | tzanger | speicfic calls too would be nice |
| 19:55.04 | bannerman | so.. if mpg123 doesn't need timing.. I don't really need ztdummy, eh? |
| 19:55.22 | tzanger | correct |
| 19:55.28 | `Sauron | you only need ztdummy for meetme and one other thing, IIRC |
| 19:55.34 | tzanger | I apologize for not being more helpful, I really don't know much about MOH |
| 19:55.43 | G0shen | yea, Broadvoice just picked up four more calling areas in my state for DIDs |
| 19:55.54 | bannerman | it's ok, I really appreciate the help anyhow tzanger |
| 19:56.04 | CosmicRay | G0shen: livevoip.com has quite a few too |
| 19:56.20 | CosmicRay | dammit, i need some plugin to fiew broadvoice's availability |
| 19:56.24 | G0shen | livevoip.com doesn't have my state, and no LMP right? |
| 19:56.46 | CosmicRay | G0shen: really? what state? (and what's LMP) |
| 19:56.55 | G0shen | Local number portability |
| 19:57.01 | CosmicRay | oh, lnp |
| 19:57.04 | CosmicRay | no, I don't think they do |
| 19:57.05 | SexyKen | Anyone here have experience with the PolyCom IP phones? |
| 19:57.14 | Inv_arp | i used livevoip but i dont trust that it will last long for incoming |
| 19:57.14 | *** join/#asterisk Voicelynx (VoiceLynx@user-0cdv656.cable.mindspring.com) |
| 19:57.26 | *** join/#asterisk infernall-work (~Infernall@d221-71-10.commercial.cgocable.net) |
| 19:57.35 | G0shen | livevoip.com says they only allow two calls at a time, Broadvoice does 4 |
| 19:57.40 | CosmicRay | in my area code, broadvoice has nothing and livevoip had about half a dozen things |
| 19:57.47 | CosmicRay | Inv_arp: why is that? |
| 19:57.55 | CosmicRay | G0shen: not quite true, livevoip has other plans |
| 19:58.00 | G0shen | Broadvoice is porting my landline over...so I can kiss Qwest goodbye |
| 19:58.06 | CosmicRay | G0shen: nice. |
| 19:58.18 | `Sauron | G0shen: How soon 'till they're done? |
| 19:58.19 | CosmicRay | Goshen: I wish I could get rid of my landline. still must have it since I have dsl |
| 19:58.21 | G0shen | and not have to change my number with tons of people |
| 19:58.22 | Inv_arp | CosmicRay: hmm doesnt seem professional enough like BV or VP |
| 19:58.48 | CosmicRay | Inv_arp: I dunno, I'd think the fact that they offer million-minute discounts is a good sign. |
| 19:58.49 | G0shen | Sauron: I just signed up for their service, so it is going to be some time, because they are processing their old customers first |
| 19:59.01 | `Sauron | right |
| 19:59.11 | `Sauron | I'm hoping I'm in the "old customers" group |
| 19:59.16 | G0shen | But I sent my paperwork in :) |
| 19:59.17 | CosmicRay | Inv_arp: they are going after a different market, certainly. livevoip is giong after more carriers and corporate types, and happen to support end users with a clue as an accident |
| 19:59.30 | G0shen | you submit your paperwork? |
| 19:59.37 | G0shen | a copy of your bill and such? |
| 20:00.10 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 20:00.25 | km- | woot |
| 20:00.35 | km- | ~seen jer[mobile] |
| 20:00.37 | jbot | i haven't seen 'jer[mobile]', km- |
| 20:00.49 | `Sauron | yeah |
| 20:00.55 | km- | ~seen jerjer[mobile] |
| 20:00.55 | jbot | jerjer[mobile] is currently on #asterisk (4h 20m 31s). Has said a total of 51 messages. Is idling for 50m 57s |
| 20:00.56 | tzanger | km-: |
| 20:01.04 | km- | tzanger: sup dude? |
| 20:01.13 | tzanger | km-: Digium contacted me about the E&M Wink problem, I told htem I wasn't who they were looking for |
| 20:01.20 | km- | tzanger: huh, that's od |
| 20:01.22 | greg_work | anyone know why i'd get this when trying to put in cdr's? Mar 16 14:54:09 ERROR[29895]: Failed to insert into database: Unknown column 'endDate' in 'field list' I can't find any reference to "endDate" (case insensitive) in /usr/src/asterisk, /usr/src/asterisk-addons, /etc/asterisk, .. even tried strings cdr_addon_mysql.so .. this is asterisk 1.0.6 |
| 20:01.29 | km- | tzanger: I e-mailed them the support issue a few days ago |
| 20:01.37 | km- | tzanger: wonder how they got that confused |
| 20:01.41 | tzanger | not sure :-) |
| 20:01.43 | greg_work | or sorry, Asterisk CVS-v1-0-03/09/05-14:46:34 |
| 20:01.46 | km- | tzanger: can you forward me the e-mail? pgrace@rttx.com |
| 20:01.52 | km- | tzanger: that way I can get in touch with them |
| 20:02.10 | jontow | score.. got mysql working as the CDR storage :) |
| 20:02.23 | km- | ~seen shido6 |
| 20:02.24 | jbot | shido6 is currently on #asterisk (17h 49m 21s). Has said a total of 45 messages. Is idling for 8m 17s |
| 20:02.25 | tzanger | km-: sent |
| 20:02.52 | ClayReiche123 | greg_work:something is probably not quoted that should be. |
| 20:03.04 | km- | tzanger: awesome. |
| 20:03.16 | greg_work | ClayReiche123: from where? |
| 20:03.27 | sivana | hey Moc? |
| 20:03.41 | heison | question about AGI... |
| 20:03.55 | ClayReiche123 | I'm sorry... that's an Asterisk raw CDR.... I've seen that in PostgreSQL when I didn't quote the field name. |
| 20:04.19 | heison | if i have exten =>s,1,Wait, exten => s,2,AGI(script.php), exten => s,3,Answer |
| 20:04.34 | heison | does Asterisk wait until script.php is finished executing before answering? |
| 20:05.09 | *** part/#asterisk dave_mwi_ (~dave_mwi@harpo.dreamhost.com) |
| 20:05.26 | enemy^x | while calling out using a softphone, I get Unable to create channel of type 'SIP' .... could anyone give me a hint? |
| 20:06.59 | Skysky | by using qualify=yes, does it take a lot of bandwidth? |
| 20:07.04 | jontow | km; if you find anything about that E&M wink issue .. let me know, have a potential situation where its going to be used |
| 20:07.11 | Moc____ | dont use qualify with SIP |
| 20:07.19 | Moc____ | you will have problems for sure |
| 20:07.28 | Moc____ | but it work fine with IAX |
| 20:07.30 | Moc____ | hi sivana |
| 20:08.02 | Juggie | Moc____, qualify with sip works perfect here |
| 20:08.02 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 20:08.05 | Juggie | with a number of phones |
| 20:08.46 | Skysky | Moc: but i have "Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk".. wat kind of settings should i do for this stituation? |
| 20:08.52 | heison | if i have exten =>s,1,Wait \ exten => s,2,AGI(script.php) \ exten => s,3,Answer |
| 20:09.54 | km- | "Your Queue Is Tier 2 Support!" |
| 20:09.55 | km- | rock on |
| 20:10.03 | km- | I'm all about "Tier 2" whatever the hell that makes me |
| 20:10.04 | km- | :) |
| 20:10.16 | Qwell | km-: means you're special |
| 20:10.18 | *** join/#asterisk brettnem (~mive29@208.54.232.29) |
| 20:10.22 | brettnem | hello all |
| 20:10.40 | brettnem | hey anyone work with chan_mgcp at all? |
| 20:11.23 | brettnem | anyone here?? heh |
| 20:11.27 | Essobi | 've heard of some people using cisco ata's with mgcp |
| 20:12.01 | brettnem | I got it working with an Adit 600.. but looks like I have a IAX->MGCP connection that is hung and will not hangup |
| 20:12.42 | brettnem | getting a lot of: Mar 16 14:12:12 WARNING[8904]: channel.c:523 ast_channel_walk_locked: Avoided deadlock for 'MGCP/aaln/1@192.168.10.5-1', 10 retries! |
| 20:13.06 | Juggie | brettnem, i get those on local channel. |
| 20:13.06 | brettnem | can't even request a soft hangup on the MGCP channel.. doesn't think it's a valid channel. |
| 20:13.16 | Juggie | not sure what causes them |
| 20:13.25 | brettnem | and requesting a soft hangup on the IAX side never actually hangs up |
| 20:13.39 | brettnem | Juggie: it is service affecting.. not jsut a CLI annoance |
| 20:13.53 | brettnem | looks like the MGCP stack froze up |
| 20:14.01 | brettnem | not sure how what to do with that |
| 20:14.04 | brettnem | ideas anyone? |
| 20:15.10 | km- | qwell: shit, I like being special |
| 20:15.17 | brettnem | doh |
| 20:15.20 | km- | but not THAT kind of "special" |
| 20:15.35 | km- | I like VIP special as opposed to Special Olympics Special, if ya know what I mean. |
| 20:15.44 | km- | Although, I make my wife dial 9 |
| 20:15.47 | heison | does anyone know if how to execute an AGI in the background? |
| 20:15.55 | km- | so maybe I should compete in the bonehead special olympics :) |
| 20:15.57 | Qwell | yeah...thats why they sent you to tier2 :p |
| 20:16.00 | km- | hahaha |
| 20:16.20 | km- | there was no hold music on the queue, I think I got disconnected |
| 20:16.33 | km- | I just called em back and left the guy a VM |
| 20:19.35 | *** join/#asterisk yaaar (~chatzilla@deliberant-ap-1.tranquility.net) |
| 20:19.39 | yaaar | word |
| 20:19.45 | Essobi | What version of gcc is requied for * |
| 20:20.01 | yaaar | anybody here have gnophone working on gentoo? i'm havin the emerge die on my lappy |
| 20:20.08 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 20:20.12 | yaaar | some weird parse error |
| 20:21.15 | jontow | i have it installed.. but not working really |
| 20:21.25 | *** join/#asterisk brettnem (~mive29@208.54.232.29) |
| 20:21.27 | brettnem | argh |
| 20:22.50 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 20:24.05 | *** join/#asterisk ClayReiche123 (~creiche@199-112.35-65.tampabay.res.rr.com) |
| 20:24.06 | yaaar | jontow: so, no problems with the build? |
| 20:24.12 | jontow | nope, not that i saw |
| 20:24.14 | yaaar | hrm |
| 20:24.22 | jontow | using 2004.3, newly "emerge --sync"'d |
| 20:24.25 | ClayReiche123 | Anyone know what this means? Mar 16 15:18:16 DEBUG[10157]: chan_sip.c:7753 sipsock_read: Failed to grab lock, trying again... |
| 20:26.59 | yaaar | jontow: yeah, me too....it gives me an error like this: |
| 20:27.13 | *** join/#asterisk lutor (~lutor@icable-docsis2-108.ibw.com.ni) |
| 20:27.14 | yaaar | In file included from audio-phone.c:29: |
| 20:27.16 | yaaar | /usr/include/linux/ixjuser.h:353: error: parse error before '*' token |
| 20:27.50 | lutor | hi, |
| 20:27.55 | heison | looking for recommendation... i want to use Jabber to display callerID info to Gaim, using AGI to call jabber.php seems to be a sequential operation, it blocks until jabber.php finishes executing. If I use System(), would things be happening in the background? |
| 20:28.51 | bkw_ | asdf |
| 20:28.55 | yaaar | jontow: the line in that file I don't understand at all really....it's the second to last line here: http://pastebin.com/258373 |
| 20:28.58 | Qwell | eh? |
| 20:29.44 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 20:29.45 | Sedorox | brb |
| 20:30.24 | afe | ctcp bkw_ |
| 20:30.28 | afe | oops |
| 20:30.34 | afe | and lol |
| 20:30.37 | lutor | i'm trying to work with pickupgroup, but i can't get my analog phones to detect *8 extension.. |
| 20:30.44 | lutor | can somebody help me.. |
| 20:30.47 | *** join/#asterisk SpaceBass (~sp@24.125.33.214) |
| 20:31.30 | SpaceBass | installed asterisk@home and the flash operator pannel just blinks... doesnt seem to respond at all |
| 20:32.41 | ariel_ | SpaceBass, you need to put your web address in the settings. web_hostname=kasipbx.homedns.org something like that. |
| 20:33.09 | SpaceBass | ariel_ thanks! the host name of the server I assume- not the client |
| 20:33.24 | ariel_ | SpaceBass, yes ther server in op_server.cfg |
| 20:33.55 | ariel_ | you can use the maintance section and ediconfig |
| 20:34.09 | ariel_ | edit config i mean |
| 20:34.34 | yaaar | jontow: hm. neat. bugzilla strikes again: http://bugs.gentoo.org/show_bug.cgi?id=67747 |
| 20:35.06 | SpaceBass | thats what Im using now- thanks |
| 20:35.20 | SpaceBass | ariel_ does it have to match the hostname in /etc/hosts or the dns hostname? |
| 20:35.26 | SpaceBass | (which currently are different) |
| 20:36.10 | SpaceBass | ariel_ nevermind |
| 20:36.12 | ariel_ | it needs to match what you access it by. If you access it by the ip address then that is what you need if you access via the www.XXXX.com then that is what you need. |
| 20:36.22 | SpaceBass | realized that right after i asked |
| 20:38.38 | SpaceBass | another generic question- I am trying to learn the dialpath / confi file language and structure. but I cannot tell where things go |
| 20:39.10 | brettnem | so can anyone help me out with the MGCP channel hanging up my system?? :/ |
| 20:39.16 | SpaceBass | for instance, I want to configure my box to check incoming caller id from my PSTN and if it matched a number then it should play a different IVR message than normal |
| 20:39.21 | SpaceBass | but I cannot determine where that would go |
| 20:39.25 | ariel_ | SpaceBass, Amp sets most things up for you. You can then use the edit config to see what it changes in the files. |
| 20:40.09 | SpaceBass | ariel_ been doing that for things like sip extensions, and I see where they go and how they are setup, but I dont understand did and trunks, etc... been looking for good doc and cannot find it... i have a lot of doc on specifics, but nothing on basics |
| 20:40.27 | *** join/#asterisk darby_t (~tom@doh200.neoplus.adsl.tpnet.pl) |
| 20:41.25 | ariel_ | SpaceBass, There are no real doc's on Amp. But incoming calls is where you set the did to go an ivr. Also there is a location here called #amportal that can also help with your questions. |
| 20:41.45 | SpaceBass | cool, thanks |
| 20:41.55 | Qwell | Whats the acronym I'm looking for? RB.. Regional Bell..something? |
| 20:41.56 | SpaceBass | last ?.. is did basically incoming caller id? |
| 20:42.18 | *** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) |
| 20:42.18 | ariel_ | I edit most of my setup files my self. And just use amp for sip and iax users and allow it to supplie my flash operator panel. |
| 20:42.23 | Qwell | rboc? |
| 20:42.43 | ariel_ | SpaceBass, no it's the number that they called to get to your system. |
| 20:42.49 | jtodd | Here's an interesting photo: |
| 20:42.50 | jtodd | http://pulver.com/reports/images/spring05von/08mar05-1.jpg |
| 20:42.53 | SpaceBass | i'd prefer to learn to edit the files myself, but want to let amp do it at first to make sure syntax is correct |
| 20:43.16 | brettnem | anyone know why I can't unload -f chan_mgcp.so?? |
| 20:43.21 | SpaceBass | ariel_ so if I was going to have it check for a specific caller id on an incoming call, where would that be? |
| 20:43.22 | Qwell | jtodd: Who are they? |
| 20:43.23 | SpaceBass | which file? |
| 20:43.31 | ChulJin | I think that's kram...who's the other person? |
| 20:43.34 | jtodd | Qwell: Mark Spencer and John Sculley. :-) |
| 20:43.37 | ChulJin | (forgive my ignorance :P) |
| 20:43.50 | brettnem | help :( |
| 20:44.08 | jtodd | Nothing implied by this picture - this is from Jeff Pulver's blog site... |
| 20:44.24 | ariel_ | SpaceBass, this you will have to setup your own custom setup. |
| 20:44.31 | jtodd | Asterisk is getting headline attention, I think, is my point. |
| 20:44.36 | jtodd | But no surprise there. |
| 20:45.04 | SpaceBass | ariel_ added the server to that config file for the pannel, and rebooted, now I cannot even connect to amp via the web |
| 20:45.07 | SpaceBass | box pings fine |
| 20:45.51 | ChulJin | jeez, now I have to find out who John Sculley is. |
| 20:46.31 | brettnem | mgcp anyone? |
| 20:46.35 | ariel_ | SpaceBass, this is how I get to mine: http://kasipbx.homedns.org/panel/ |
| 20:47.35 | SpaceBass | think i broke it in my /etc/hosts file |
| 20:48.05 | *** join/#asterisk marc32344 (~marc32344@69-90-241-15.dsl.teksavvy.com) |
| 20:48.10 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 20:51.25 | ManxPower | ~docs |
| 20:51.26 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:51.29 | ManxPower | ~mailinglist |
| 20:51.30 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 20:51.47 | bannerman | yay, built and installed the mpg123 that came with asterisk and moh is back in business |
| 20:52.14 | bannerman | I think I screwed it up when I had this thing on my desk and installed a different mpg123 for playing mp3s on my desktop :) |
| 20:52.27 | bannerman | <-- bad administrator |
| 20:53.45 | brettnem | everything was looking good until I tried to transfer.. blah |
| 20:54.04 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 20:56.18 | *** join/#asterisk bah (048830696@ACA15888.ipt.aol.com) |
| 20:56.25 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 20:58.09 | Qwell | In order to unprovision a line on a 7960, do you have to use tftp, or can you just edit it somehow from the phone? |
| 20:58.18 | *** join/#asterisk goldenolden (~goldenold@65.171.196.23) |
| 20:58.21 | goldenolden | hi |
| 20:58.46 | goldenolden | anyone had any luck creating the zap directories under /dev in freebsd for the zaptel/meetme drivers? |
| 20:59.06 | jontow | qwell; should be able to do it from the phone? .. i seem to remember doing that.. |
| 20:59.15 | jontow | but it doesn't update the config file so when the phone reboots it may re-provision it |
| 20:59.22 | Qwell | ahh |
| 20:59.41 | Qwell | boss is asking me how...I've never even picked one up, let alone used one |
| 20:59.50 | jontow | best just edit the config file and reboot it.. |
| 20:59.59 | Qwell | so, tftp then? |
| 21:00.01 | jontow | yep |
| 21:00.26 | Qwell | Any ideas on a good tftp server for Windows? |
| 21:00.34 | brettnem | grr stubborn channels |
| 21:00.38 | Qwell | ...not that I run windows or anything |
| 21:00.48 | brettnem | for 'wine' I'm sure.. |
| 21:00.53 | *** join/#asterisk n4y (~Tom@fire2.orasoft.net.pl) |
| 21:01.01 | SpaceBass | will cuteFTP do it? |
| 21:01.06 | brettnem | don't think so |
| 21:01.06 | Qwell | Thats a client. |
| 21:01.07 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 21:01.16 | brettnem | try google |
| 21:01.29 | brettnem | ~google windows tftp server |
| 21:01.40 | *** join/#asterisk afe ([iQ7YyQZ+w@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 21:02.12 | Qwell | yeah, found the cisco one |
| 21:02.30 | SpaceBass | Qwell cuteFTP has a server too |
| 21:02.34 | Qwell | oh |
| 21:02.34 | SpaceBass | but not sure it does tftp |
| 21:02.40 | ManxPower | urpmi tftp-server |
| 21:02.43 | ManxPower | That's all you need. |
| 21:02.45 | Qwell | ManxPower: Windows. :p |
| 21:02.56 | Qwell | I'd just emerge it, if I needed it |
| 21:03.02 | ManxPower | Qwell, Well then he should install linux first, then run urpmi |
| 21:03.06 | Qwell | heh |
| 21:03.31 | brettnem | really..don't do tftp with windows. it's bound to f-up whatever it is you are trying to do |
| 21:03.36 | Qwell | He's going to, after he gets hardware... |
| 21:03.44 | Qwell | I've already convinced him |
| 21:03.54 | ManxPower | The ATA-186 firmware even has a TFTP server as part of the package. I would assume the SIP for other Cisco phones would have that too! |
| 21:05.34 | ManxPower | Joy! CVS-HEAD crashed again. |
| 21:06.01 | brettnem | ManxPower: I'm running a older CVS HEAD and I have a locked up MGCP stack.. blah |
| 21:07.51 | ManxPower | brettnem, I'm (accidently) running CVS-HEAD-03/03/05 |
| 21:08.56 | brettnem | ManxPower: I'm (accidently) running CVS-HEAD-02/09/05 |
| 21:09.25 | brettnem | I'm just waiting to remove asterisk from my network.. |
| 21:09.44 | MuppetMaster | brettnem: Why? |
| 21:10.25 | afe | anyone tried the atxfer patch to be able to get the call back? |
| 21:10.54 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 21:10.55 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:11.05 | brettnem | MuppetMaster: mistake.. need to update |
| 21:11.25 | brettnem | I wonder if that might be contributing to my MGCP problems.. just don't see much MGCP activity in the code tho.. |
| 21:12.06 | ManxPower | brettnem, I forgot the -r v1-0 on my cvs co command and didn't realise it until I had rsync'd the code to three servers. |
| 21:12.38 | brettnem | ek |
| 21:12.53 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 21:12.54 | brettnem | ManxPower: you do any MGCP? |
| 21:13.11 | ManxPower | brettnem, I tried it once, realized it sucks with NAT and never tried it again. |
| 21:13.18 | brettnem | blah |
| 21:13.30 | ManxPower | I won't try it again until there's a MGCP maintainer |
| 21:13.53 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 21:13.54 | brettnem | yeah..I'm trying to setup a bunch of ADIT 600s with 300 lines |
| 21:14.25 | ManxPower | I like the whole "phone is dumb as a rock" idea of MGCP. |
| 21:14.32 | brettnem | right |
| 21:14.49 | brettnem | I think it's kinda the right way to do it. besides, some trunking technologies require it |
| 21:14.52 | ManxPower | i.e. no silly things like a dialplan in the phone or crap like that. |
| 21:15.34 | brettnem | well I'm just trying to figure out how to release a MGCP channel without restartng my system now.. |
| 21:15.40 | brettnem | looks like I hit a bug |
| 21:17.45 | heison | bkw_: are u around? |
| 21:18.16 | Qwell | Is there much difference between the 7960 and 7970 configs? |
| 21:18.46 | *** join/#asterisk jedaustin (~chatzilla@austin-j.its.dist.maricopa.edu) |
| 21:18.54 | *** join/#asterisk Inv_arp (junya@adsl-8-232-25.mia.bellsouth.net) |
| 21:20.46 | jedaustin | Nice article about Mark Spencer in the Topic |
| 21:22.37 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 21:22.40 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 21:22.54 | MattH | Hi... what do I need to do (or which config file do I edit) to enable and disable codecs? |
| 21:23.38 | sudhir492 | May not be appropriate question for this forum, but how do we disable devices (e.e USBs) which are not used in PC. |
| 21:23.48 | brettnem | sudhir492: bios |
| 21:23.56 | brettnem | sudhir492: or disable the modules |
| 21:24.28 | sudhir492 | modules, will that be through udev |
| 21:25.30 | jedaustin | Where is the ipaddress set that shows up in emails for voice messages? |
| 21:26.41 | jedaustin | found it.. vm_email.inc |
| 21:26.57 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 21:27.00 | mutilator | hi all |
| 21:27.36 | jedaustin | mutilator: Hello |
| 21:27.51 | ManxPower | MatsK, Which protocol? |
| 21:28.23 | MattH | ManxPower: I'm using G711 on most of my phones |
| 21:28.37 | MattH | ManxPower: but I have one phone that supports everything BUT that.. but asterisk doesn't seem to want to talk to it |
| 21:28.46 | ManxPower | MatsK, What PROTOCOL? SIP, MGCP, IAX, SCCP? |
| 21:28.57 | ManxPower | H323? |
| 21:28.59 | ManxPower | RPT? |
| 21:29.03 | ManxPower | PRI? |
| 21:29.34 | MattH | sip |
| 21:29.35 | MattH | sorry |
| 21:29.57 | ManxPower | MatsK, sip.conf |
| 21:30.29 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 21:30.49 | MattH | so I put an allow line in for the codec? do I need to install any codec drivers for like say iLBC, or G729 ? |
| 21:31.03 | *** join/#asterisk rious (~rious@24.231.227.77.gha.mi.chartermi.net) |
| 21:31.07 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 21:31.09 | Juggie | get the phone working with ulaw first |
| 21:31.23 | PBXtech | what providers of local DID allow passing ot CID? |
| 21:31.28 | MattH | that's the issue I don't see a ulaw codec.. unless I'm mis reading something |
| 21:31.38 | Juggie | g711? |
| 21:31.47 | MattH | not an option on this phone (yeah eh?!?!) |
| 21:31.55 | *** join/#asterisk ACiDV (~joel@122-68-181.dr.cgocable.ca) |
| 21:32.01 | Juggie | gsm? |
| 21:32.03 | ACiDV | Reason for Return: Incompatible with E75xx chipset |
| 21:32.03 | ACiDV | Replacement Request: Super TE410P |
| 21:32.06 | MattH | I have: PCMA, G723, G728, G729, G726-32, iLBC |
| 21:32.09 | MattH | nope.. no gsm either |
| 21:32.15 | ACiDV | doh... a "special" model ? |
| 21:32.23 | Juggie | PCMA? |
| 21:32.34 | MattH | yeah I gave it that look too when I read PCMA |
| 21:32.41 | CoaxD | Any of you lamers have a windows XP cd I can snag? |
| 21:32.54 | Juggie | MattH, did u let the phone register atleast |
| 21:32.59 | Juggie | to see what it requests |
| 21:32.59 | johnnyb | MattH - does it have PCMU? That's Ulaw. |
| 21:33.11 | MattH | yes it does have pcmu |
| 21:33.12 | Juggie | ahhhh |
| 21:33.20 | Juggie | PCMA = alaw PCMU = ulaw |
| 21:33.22 | Juggie | :) |
| 21:33.38 | johnnyb | MattH: Is this a Budgetone? I'm guessing from the codec list. |
| 21:33.51 | MattH | yes.. |
| 21:33.51 | MattH | hehe |
| 21:33.55 | MattH | I'm not liking it at all |
| 21:34.03 | johnnyb | MattH: I actually like ours a lot. |
| 21:34.17 | MattH | I have used sipuras and they seem to have more config options |
| 21:34.19 | johnnyb | MattH: By default, for some reason unknown, sip.conf disables ulaw by default. |
| 21:34.32 | MattH | that would do it |
| 21:34.40 | johnnyb | MattH: I like using iLBC with the phones -- MUCH less bandwidth. |
| 21:34.41 | MattH | but I do have: disallow=all |
| 21:34.41 | MattH | allow=ulaw |
| 21:34.41 | MattH | allow=alaw |
| 21:34.52 | ManxPower | MatsK, why are you allowing BOTH ulaw and alaw. |
| 21:34.54 | johnnyb | Sorry, I meant disables iLBC. It enables ULAW. |
| 21:35.16 | ManxPower | MatsK, in [general] allow=all then in each [phonesection] disallow=all and allow=theonecodecyouwant |
| 21:35.23 | MattH | ManxPower: because I'm fairly new to voip.... |
| 21:35.34 | MattH | ManxPower: however, what if I want to allow a 30/50/90kbps selection per phone? |
| 21:35.40 | johnnyb | MattH: G711 IS U-Law. |
| 21:35.42 | MattH | ManxPower: to allow the end user to select their bandwidth? |
| 21:35.48 | ManxPower | MatsK, Why would you want to do that? |
| 21:36.08 | MattH | ManxPower: some users have little bandwidth to spair (like when offering a voip service to a customer) |
| 21:36.12 | ManxPower | MatsK, If you want to be terribly nice you can have allow= lines for mor than one codec. |
| 21:36.19 | *** join/#asterisk miguel3239 (~myoung@ip66-105-121-66.z121-105-66.customer.algx.net) |
| 21:36.24 | ManxPower | MatsK, each customer has an entry in sip.conf. |
| 21:36.28 | johnnyb | MattH: I would just use iLBC -- it actually improves voice quality from what I can tell. It seems to remove non-voice noise fairly well. |
| 21:36.52 | MattH | johnnyb: really.. hrmm I was under the impression it REALLY compressed |
| 21:36.59 | ManxPower | johnnyb, You DO realize that almost NO phones support iLBC, right? |
| 21:37.07 | johnnyb | ManxPower: Budgetones do. |
| 21:37.15 | MattH | yeah this one definately does |
| 21:37.22 | ManxPower | johnnyb, That's 1 phone. |
| 21:37.29 | johnnyb | MattH: Just finished setting our office up with them last week. |
| 21:37.43 | johnnyb | ManxPower: Just put it first in the list, followed by u-law. |
| 21:37.51 | johnnyb | disallow=all |
| 21:37.55 | johnnyb | allow=ilbc |
| 21:37.57 | mutilator | what does defaultexpirey do in the sip.conf |
| 21:37.58 | MattH | ok.. while I'm here got a question.. how do I do like *77 or *asteriskkeycode when my phone rips the *code off? |
| 21:38.01 | johnnyb | allow=whatever. |
| 21:38.08 | ManxPower | johnnyb, You do know that codec order is totally broken in all 1.0.x releases of Asterisk, right? |
| 21:38.13 | mutilator | if it's set to 0, does it never expire? |
| 21:38.24 | PBXtech | what providers of local DID allow passing of my custom CID number? |
| 21:38.28 | ManxPower | MatsK, make the phone stop doing that. |
| 21:38.48 | johnnyb | ManxPower - I thought it was supported globally. |
| 21:39.11 | johnnyb | MattH: why is the phone ripping off *? It shouldn't. |
| 21:39.24 | ManxPower | MatsK, Here is what I do: For phones that are on the same LAN as the Asterisk server I force them to use ulaw. For phones NOT on the same lan as the Asterisk server are forced to G729. Inter-Server calls are forced to SpeeX |
| 21:39.24 | jedaustin | * = 11 |
| 21:39.26 | jedaustin | Don't know if it works in this situation |
| 21:39.28 | MattH | johnnyb: I think because *77 is a ACD feature of the phone |
| 21:39.38 | johnnyb | MattH: You could try "Disable Advanced Call Features" in the admin menu. |
| 21:39.43 | MattH | yeah |
| 21:40.34 | johnnyb | MattH: You could also use a different key code in your dial plan. |
| 21:40.52 | johnnyb | What firmware version are your phones on? |
| 21:41.11 | MattH | .10 |
| 21:41.12 | *** join/#asterisk r3d5un (~r3d5un@83-64-21-154.dynamic.home.xdsl-line.inode.at) |
| 21:41.18 | MattH | I need to upgrade to .16 |
| 21:41.27 | ManxPower | MatsK, SIP phones seem to do everything in the phone that you don't want them do to (like handleing *XX codes in the phone) and do nothing that you WANT them to do (like dialplan pattterns) |
| 21:41.36 | *** join/#asterisk foogster (~rob@rrcs-24-73-84-62.se.biz.rr.com) |
| 21:41.43 | foogster | Hey Folks |
| 21:41.46 | johnnyb | MattH: I'm on .23. |
| 21:41.46 | MattH | lol |
| 21:41.52 | MattH | er .23 yeah not .16 |
| 21:42.03 | johnnyb | It's available at http://www.grandstream.com/BETATEST/ |
| 21:42.05 | ManxPower | MatsK, We got ONE grandstream and will never get another one. |
| 21:42.19 | johnnyb | ManxPower: What didn't you like about it? |
| 21:42.31 | ManxPower | We use Polycom IP 300, Polycom IP 500, and will also be using SIPura SPA-841 |
| 21:42.37 | MattH | It doesn't seem too bad.. associated right up now that I got the codec right |
| 21:42.47 | MattH | I do kinda like the sipura better.. but the grandstream isn't too bad |
| 21:42.59 | ManxPower | johnnyb, No support for dialplan stuff. |
| 21:43.06 | johnnyb | ManxPower: like what? |
| 21:43.27 | ManxPower | Users do not want to have to press DIAL on any phones except cell phones. |
| 21:43.27 | brettnem | grandstreams are pieces of JUNK |
| 21:43.44 | brettnem | had registrations problems too |
| 21:43.48 | ManxPower | johnnyb, And honestly, I've never heard of a SIP phone company that releases such buggy firmware. |
| 21:43.53 | brettnem | lines were noisy |
| 21:44.01 | johnnyb | ManxPower: Hmmm. Mine don't mind just waiting the 4 seconds. |
| 21:44.01 | MattH | I don't have to press dial on my grandstream |
| 21:44.19 | *** join/#asterisk mentat_ (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 21:44.26 | brettnem | 4 seconds of silence in the telecom world is a long time.. |
| 21:44.30 | johnnyb | brettnem: I found that using iLBC actually cut a lot of the noise. |
| 21:44.45 | ManxPower | Oh, and I want callerid name |
| 21:44.50 | ManxPower | and I want dialplan |
| 21:44.57 | ManxPower | and I want non-buggy firmware |
| 21:45.06 | brettnem | johnnyb: if I plug in a fxo line, i get a nice hum on the lien |
| 21:45.08 | brettnem | er line |
| 21:45.20 | brettnem | and it never played with nat well |
| 21:45.41 | DyOS | Anyone know what the synatx is to setup an iax connection to dial the same # with multiple connections from one line? |
| 21:45.46 | ManxPower | The SIPura SPA-841 has been out for, what, 2 months, 3 months, already their firmware is a galaxy ahead of grandstream's 3 year old product |
| 21:45.51 | johnnyb | Is there another phone for a similar price that works? |
| 21:45.53 | DyOS | like denial of service for a phone is that possible? |
| 21:45.53 | MattH | shouldn't the dial-plan be done on the PBX though nto the phone? |
| 21:46.02 | ManxPower | MatsK, In theory yes. |
| 21:46.21 | ManxPower | In SIP that's called "early dial" and grandstream has had it for a year. It still doesn't work from what I hear. |
| 21:46.22 | brettnem | johnnyb: get a sipura ata |
| 21:46.43 | spackle | I don't see any software here: http://www.grandstream.com/BETATEST/ |
| 21:46.45 | MattH | do the sipuras support dial without having to press "dial" |
| 21:46.46 | brettnem | I've never seen early dial supported real well |
| 21:47.03 | brettnem | MattH: They support a pretty configurable dial string |
| 21:47.06 | ManxPower | MatsK, When correctly configured, yes. They even support continueing dialtone after dialing 9 for an outside line. |
| 21:47.18 | brettnem | I really like the sipuras |
| 21:47.18 | johnnyb | They must have taken it down |
| 21:47.20 | johnnyb | Strange.. |
| 21:47.33 | brettnem | I'd definately recommend a sipura + analog phone over any grandstream product |
| 21:47.39 | spackle | johnnyb: probably buggy--- ;-) |
| 21:47.48 | ManxPower | We would use the SIPura phones as the primary phone, but they lack two important things and two less important things. |
| 21:47.48 | MattH | hrmm where do you set the early dial on the supera? in the dial-plan? |
| 21:48.01 | brettnem | MattH: Sipura isn't using early dial |
| 21:48.08 | MattH | yeah but is there a way to make it? |
| 21:48.10 | ManxPower | MatsK, SIPura supports dialplan on the phone, not early dial |
| 21:48.13 | spackle | Manxpower: yes? |
| 21:48.23 | brettnem | MattH: don't think so.. but it isn't relly needed |
| 21:48.34 | MattH | ok sorry I must be mixing things up |
| 21:48.37 | Darwin35 | ook bv is a killer |
| 21:48.44 | ManxPower | SPA-841 major issue: No second ethernet port, not PoE. minor: no back light display, no tilt display |
| 21:48.44 | MattH | early dial isn't dialing 323-5555 and waiting the 4 seconds to have it dial? |
| 21:48.46 | Beirdo | Oh crap, I should email that guy back about the SPA3000 |
| 21:49.14 | brettnem | matth: early dial is the phone sending each digit as it is dialed to the server to see if it matches a dialable number |
| 21:49.15 | ManxPower | MatsK, correct. Early dial is the phone sends each digit to the server AS IT'S DIALED |
| 21:49.24 | MattH | ahh ok |
| 21:49.28 | spackle | Manxpower: agreed. and the polish of a few more firmware releases & damn if I can get the headset to stay on the phone. |
| 21:49.43 | spackle | sucker slides off too easily. |
| 21:49.49 | ManxPower | spackle, Oh! Yes the headset port has some issues. |
| 21:50.04 | ManxPower | spackle, The latest firmware resolved all issues for me. |
| 21:50.17 | foogster | Question: I just got Asterisk installed and want to begin creating the PBX. I want to be completely VOIP to a provider's PSTN. I have researched a lot of companies and hear +/- about each one. Any recommendations? |
| 21:50.19 | MattH | so in the sipura's how do you make it dial after waiting the 4 seconds like the grandstreams do (so you don't have to hit the dial button)? |
| 21:50.36 | MattH | foogster: have you checked out level3? |
| 21:50.54 | SwedMiro | foogster..might help if you told us which country your in? |
| 21:50.56 | foogster | I have been told about them and SIPstorm |
| 21:50.57 | foogster | US |
| 21:51.07 | MattH | foogster: we use level3 here |
| 21:51.12 | foogster | from another Asterisk user - and Voicepulse and Vonage, etc |
| 21:51.20 | foogster | I will check it out |
| 21:51.26 | foogster | Thanks Matt |
| 21:51.36 | jhiver | guys, is there any free (as in freedom, not free beer), skinnable, non-sucky windows IAX softphone I can use? |
| 21:51.39 | MattH | yeah if you only need a couple lines.. go with voicepulse... |
| 21:52.07 | foogster | Isn't there a way to run more than one conversation down the same "phone number"? |
| 21:52.09 | johnnyb | MattH: You're missing the point. On dialplan-supported phones, the phones don't need to wait four seconds. They know the formats of valid numbers and send them immediately once a certain number format is given. |
| 21:52.44 | MattH | johnnyb: so what if I have 570-323-xxxx for long distance but want to offer 323-xxxx for local dialing? |
| 21:53.05 | johnnyb | MattH: You set up several different patterns. It's the same way asterisk handles it. |
| 21:53.52 | MattH | johnnyb: makes sence... |
| 21:54.34 | *** join/#asterisk ClayReiche123 (~creiche@199-112.35-65.tampabay.res.rr.com) |
| 21:54.40 | ClayReiche123 | I'm having some trouble, I'm getting 1 way audio. PSTN to asterisk SIP, PSTN phone can hear everything, SIP phone can not hear... http://pastebin.ca/7588 |
| 21:54.53 | Juggie | ClayReiche123, nat? |
| 21:54.54 | johnnyb | ManxPower: I had never heard of the SIPura's. I might check those out. |
| 21:54.58 | ManxPower | My Phone dialplan: ([2-7]xxx|9,1[2-9]xx[2-9]xxxxxx|9,[2-9]xxxxxx) |
| 21:54.59 | ClayReiche123 | This line looks pertinent.... Mar 16 16:45:50 DEBUG[11502]: channel.c:2663 ast_channel_bridge: Didn't get a frame from channel: SIP/8138644419-a985 |
| 21:55.22 | johnnyb | ClayReiche123: What kind of phone, and what protocol? |
| 21:55.35 | MattH | johnnyb: yeah they are very nice |
| 21:55.38 | johnnyb | ClayReiche123: also, what's the output of sip show channel |
| 21:55.48 | foogster | We had a SIPura in the office yesterday. Call it from a phone and it will tell you its IP address...pretty neat |
| 21:56.02 | ClayReiche123 | Juggie:yes the phone is behind nat. It's a WorldAccxx TA200 analogue telephone adapter. |
| 21:56.31 | Qwell | ManxPower: What would you say the most expensive country to call is? |
| 21:56.31 | Juggie | ahh nat... |
| 21:56.31 | Juggie | did u set nat=yes |
| 21:56.51 | johnnyb | ClayReiche123: I'm guessing it's a NAT problem. |
| 21:57.32 | johnnyb | ClayReiche123: I had heard about SIP problems w/ NAT's, and decided never to even try. From what I hear, though, IAX works great over NATs. |
| 21:57.33 | ClayReiche123 | Yes nat=yes in sip.conf. |
| 21:57.44 | Juggie | ClayReiche123, how many network cards in your * box |
| 21:57.56 | jedaustin | foogster: conference calling? |
| 21:58.31 | Juggie | as well try enabeling sip debug |
| 21:58.39 | Juggie | so you can see what ips are being passed back and forth |
| 21:59.44 | SexyKen | Does anyone here know how to get the Polycom buddies function to work? |
| 21:59.49 | *** join/#asterisk MattB2 (~mattb@pcp01068561pcs.andrsn01.tn.comcast.net) |
| 22:00.21 | ClayReiche123 | Juggie: 2 why???! |
| 22:00.37 | Juggie | clay, heres the problem |
| 22:00.52 | johnnyb | ClayReiche123: Because it may be sending data to the wrong IP address. |
| 22:00.52 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 22:00.52 | Juggie | your sip packets are going out with your sip phone ip as the return |
| 22:01.07 | Juggie | so the other end has no idea how to send them back |
| 22:01.07 | ClayReiche123 | Well.... only 1 is enabled. |
| 22:01.17 | ACiDV | hmmm what is the difference between a TE410P and a "Super TE410P" card ? I just receive a mail from digium for RMA my card and they will send me a "Super TE410P" that is compatible with Intel SE75xx chipsets |
| 22:01.18 | Juggie | well ok... just fuck up my theory then :) |
| 22:01.26 | Juggie | dont tell me 2 if its just one :P |
| 22:01.27 | ClayReiche123 | :) |
| 22:01.32 | *** join/#asterisk file[laptop] (~file@mctn1-7303.nb.aliant.net) |
| 22:01.42 | ClayReiche123 | Well... there ARE 2 in the machine! |
| 22:02.00 | Juggie | does the machine have just its lan ip |
| 22:02.02 | Juggie | and local host? |
| 22:02.05 | ClayReiche123 | yes |
| 22:02.13 | Shido6 | does inbound from iconnecthere work wit * ? |
| 22:02.16 | Darwin35 | Verbosity was 0 and is now 8 |
| 22:02.16 | Darwin35 | <PROTECTED> |
| 22:02.16 | Darwin35 | <PROTECTED> |
| 22:02.16 | Darwin35 | <PROTECTED> |
| 22:02.16 | Darwin35 | <PROTECTED> |
| 22:02.17 | Darwin35 | <PROTECTED> |
| 22:02.21 | ClayReiche123 | The 2nd nic has never been configured in Linux |
| 22:02.21 | Darwin35 | <PROTECTED> |
| 22:02.23 | Darwin35 | <PROTECTED> |
| 22:02.23 | jedaustin | is it nat=yes or nat=1 |
| 22:02.25 | Darwin35 | <PROTECTED> |
| 22:02.27 | Darwin35 | <PROTECTED> |
| 22:02.29 | Darwin35 | sorry |
| 22:02.31 | Beirdo | ~pastebin |
| 22:02.32 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 22:02.32 | Darwin35 | was ment for pastebin |
| 22:02.33 | ClayReiche123 | checkin |
| 22:02.41 | Beirdo | :) |
| 22:02.43 | Juggie | are the phones and sip gateways all routeable between each other? |
| 22:02.53 | Beirdo | just in case |
| 22:03.12 | ClayReiche123 | It's nat=yes. |
| 22:03.20 | Juggie | clay try canreinvite=no |
| 22:03.44 | Pinhole | nat=yes often needs qualify=yes to work. |
| 22:04.02 | Juggie | yah, canreinvite,nat,qualify |
| 22:04.05 | Juggie | set those all to yes |
| 22:04.06 | ClayReiche123 | Just tried a call and it is working fine..... fucking inconsistancies are killingme.... I'm gonna go shoot myself now.... |
| 22:04.09 | Juggie | and give it a whirl |
| 22:04.32 | Juggie | ClayReiche123, that could be related to qualify=yes not being set |
| 22:04.35 | ClayReiche123 | ohh.. I don't have qualify=yes |
| 22:04.41 | ClayReiche123 | heheheh |
| 22:04.53 | Darwin35 | ok this is a kikiler |
| 22:04.58 | MattB2 | hi all.. got a CallerID question (don't run away just yet!) I'm from UK where calelrid is just the caller's phone number digits. However I moved over to the USA and now callerid has company names, people's names, geographical areas in it and all sorts! Our Crappy Panasonic exchange supports it but can Asterisk? |
| 22:05.04 | Darwin35 | i can dial out but no inbound |
| 22:06.13 | Pinhole | It can work without qualify=yes for a short time after the phone registers and then stop working later. qualify keeps it working. |
| 22:07.54 | modulus_ | wow |
| 22:08.01 | modulus_ | cuba is the most expensive to deliver voice there |
| 22:08.04 | modulus_ | surprise surprise |
| 22:08.21 | ClayReiche123 | back to broken again |
| 22:08.36 | *** join/#asterisk SpaceBass (~sp@24.125.33.214) |
| 22:08.40 | SpaceBass | hey folks |
| 22:08.48 | SpaceBass | having trouble understanding DIDs |
| 22:08.58 | SpaceBass | can anyone clairify? |
| 22:09.19 | niZon | Direct inward dial |
| 22:09.29 | niZon | a number someone can dial and reach you |
| 22:09.46 | SpaceBass | niZon so if my incoming line is a PSTN, then that is my number from the telco? |
| 22:10.07 | niZon | yeah :P |
| 22:10.13 | SpaceBass | gotcha |
| 22:10.22 | niZon | or if you were connected to a voip provider they could give you one too |
| 22:10.55 | SpaceBass | and what is a trunk then? |
| 22:11.03 | *** join/#asterisk iceyp (~icepick@202.150.105.150) |
| 22:11.24 | *** join/#asterisk L|NUX (~linux@202.5.145.55) |
| 22:11.36 | Juggie | ClayReiche123, sip show peers and see what the status of the phone is |
| 22:14.17 | SpaceBass | niZon how is a trunk different from a dial list? |
| 22:14.50 | ariel_ | ok is there any thing better for testing a t1 pri line other then zttool? |
| 22:14.50 | SpaceBass | s/'dial list'/'call group' |
| 22:17.43 | *** join/#asterisk Frantic__ (~ab@TechnologicPartners35.dsl.concentric.net) |
| 22:23.11 | brc_ | ~seen Maveric |
| 22:23.14 | jbot | maveric <maveric@ip68-3-248-136.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 98d 1h 58m 5s ago, saying: 'can be found at the wiki'. |
| 22:23.44 | SwedMiro | ~seen a better life? |
| 22:23.44 | jbot | i haven't seen 'a better life', SwedMiro |
| 22:23.50 | SwedMiro | shit! |
| 22:23.56 | *** join/#asterisk mishehu (mishehu@cshells.shavedgoats.net) |
| 22:24.22 | ClayReiche123 | Juggie:8138644419/8138 65.77.37.242 D N 255.255.255.255 5060 OK (116 ms) |
| 22:24.52 | ClayReiche123 | I just put it on an external IP and I'm getting no Inbound audio... |
| 22:24.57 | CosmicRay | wow, livevoip has 800 numbers for $1/mo at 1.27 cents per minute. |
| 22:25.01 | CosmicRay | i might just have to get one. |
| 22:25.01 | ClayReiche123 | Can't be NAT... |
| 22:25.27 | ClayReiche123 | I'm gonna try an Xlite phone and see if it's my crappy ta..... |
| 22:27.20 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 22:27.42 | Juggie | ClayReiche123, sip debug |
| 22:28.01 | RoyK | has the 1.2 feature freeze been yet? |
| 22:28.40 | *** join/#asterisk madounet (~mad|net@juvenal-3-82-226-155-19.fbx.proxad.net) |
| 22:29.09 | *** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 22:29.09 | Grooby | aight peeps |
| 22:29.11 | Grooby | i am outtie |
| 22:30.10 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 22:30.14 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 22:30.15 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 22:30.16 | RoyK | there |
| 22:30.39 | RoyK | anyone that knows if the feature-freeze is done? |
| 22:34.19 | Tall-guy | speaking of feature freeze.....is there a decent document location where I can kinda figure out how the development, and the stable releases related to each other vis-a-vis features? |
| 22:35.36 | SpaceBass | anyone know about adding a custome goto in AMP ? |
| 22:35.38 | RoyK | Tall-guy: the 'stable' releases have the same features, more or less. "nothing" is being added to stable, but of course some stuff slips in.. |
| 22:35.46 | *** join/#asterisk sezuan (sezuan@port-212-202-55-249.dynamic.qsc.de) |
| 22:35.52 | SpaceBass | ie if I want an IVR option to go to voice mail or the directory |
| 22:35.54 | *** join/#asterisk paulc (paulc@S010600062586a0b4.vc.shawcable.net) |
| 22:36.05 | RoyK | SpaceBass: costume? going to a party? |
| 22:36.06 | sezuan | Hi! |
| 22:36.13 | RoyK | hi |
| 22:36.18 | SpaceBass | <PROTECTED> |
| 22:36.19 | SpaceBass | thanks |
| 22:36.50 | RoyK | what is amp? |
| 22:37.10 | sezuan | How does asterisk detect the sample size of an ilbc frame? I think it could be 20 or 30ms? |
| 22:38.05 | RoyK | it doesn't detect anything |
| 22:38.24 | RoyK | asterisk uses a fixed 20ms frame size in rtp |
| 22:38.29 | RoyK | if using iax, i don't know |
| 22:39.06 | RoyK | asterisk sucks big time about sample rates |
| 22:39.40 | sezuan | hmm.. ok that explains why it doesn't work :-) |
| 22:40.26 | Tall-guy | royk: I dunno if that cleared me up at all :) |
| 22:41.08 | RoyK | Tall-guy: see the cvs mailing list. I don't think there's a good changelog |
| 22:41.30 | RoyK | sezuan: see rtp.c and the codec code |
| 22:41.50 | RoyK | sezuan: there really isn't too much docs on the subject |
| 22:42.11 | Tall-guy | royk: okie dokie, thanx for your input....I appreciate it. |
| 22:42.33 | bkw_ | Ten Out of Ten For Style, But Minus Several Million For Good Thinking!!! |
| 22:42.38 | bkw_ | NEXT!!! |
| 22:43.05 | RoyK | what? |
| 22:43.25 | bkw_ | NO |
| 22:44.22 | bkw_ | You guys apparently don't get it |
| 22:44.39 | RoyK | get what? |
| 22:46.33 | sezuan | RoyK: ok. I didn't found any client that support the "mode=??" stuff. And I thought you have to be rfc compliant to be allowed to use ilbc. |
| 22:46.34 | bkw_ | where that is from |
| 22:47.15 | *** join/#asterisk madounet (~mad|net@juvenal-3-82-226-155-19.fbx.proxad.net) |
| 22:47.19 | RoyK | bkw_: what are you talking about? please tell |
| 22:47.28 | ChulJin | hell, I dunno...HGTTG? |
| 22:47.55 | file[laptop] | nope. |
| 22:48.38 | ChulJin | are you quite sure? I think it is...scene 62 |
| 22:48.58 | ChulJin | oops |
| 22:52.00 | ChulJin | good god, it's an RFC? |
| 22:52.01 | ChulJin | haha |
| 22:52.18 | RoyK | rfc2829 |
| 22:52.19 | RoyK | :) |
| 22:52.23 | RoyK | iirc |
| 22:53.11 | ChulJin | 2324 |
| 22:53.17 | RoyK | yep |
| 22:53.36 | ChulJin | 2829 is 'Authentication Methods for LDAP'...rather dull. |
| 22:54.10 | ChulJin | I don;t quote by number any more, ever since the King Lear incident. |
| 22:54.20 | Tall-guy | ok, straight up question and I'm ready to be flamed: Am I freekin insane running CVS-HEAD on prod boxes? |
| 22:54.56 | RoyK | better then read rfc 1149 and 2549 |
| 22:55.12 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 22:55.36 | johnnyb | Tall-guy, Yes, but I do it, too. |
| 22:55.45 | Tall-guy | johnnyb: well THAT helps :) |
| 22:56.41 | MattB2 | hi all.. got a CallerID question (don't run away just yet!) I'm from UK where calelrid is just the caller's phone number digits. However I moved over to the USA and now callerid has company names, people's names, geographical areas in it and all sorts! Our Panasonic exchange supports it but can Asterisk? |
| 22:58.43 | *** join/#asterisk crich1999 (~crich@217.9.52.18) |
| 22:58.45 | file[laptop] | MattB2: yes. |
| 22:58.58 | MattB2 | :) yay |
| 22:59.01 | crich1999 | hello all |
| 22:59.09 | MattB2 | can X100P support it oo? |
| 22:59.22 | RoyK | MattB2: iirc, asterisk is mainly USAish, so why not? |
| 22:59.54 | MattB2 | ok cool.. was just checking :) from asterisk CLI how can I get it to show me the callerid of the calling person? |
| 22:59.59 | MattB2 | there's no zap debug? |
| 23:00.00 | ACiDV | hmmm what is the difference between a TE410P and a "Super TE410P" card ? I just receive a mail from digium for RMA my card and they will send me a "Super TE410P" that is compatible with Intel SE75xx chipsets ( |
| 23:00.13 | *** join/#asterisk ClayReiche123 (~creiche@65.35.112.199) |
| 23:00.17 | PTG123 | its super :) |
| 23:00.26 | ManxPower | I hate conference calls. |
| 23:00.37 | paulc | Anyone around using FC3 with MySQL upgraded to 4.x and Myth 0.17? I'm doing a fresh FC3 install, would ideally like MySQL 4 (for other stuff I'm working on) and am wondering if I should upgrade to that before touching MythTV? |
| 23:00.42 | paulc | uh |
| 23:00.42 | paulc | fuck |
| 23:00.44 | paulc | wrong window |
| 23:00.45 | *** join/#asterisk christo (~chris@212.18.226.160) |
| 23:00.48 | christo | evening all |
| 23:01.19 | ManxPower | ACiDV, I would assume that "Super" means "compatable with intel chipset. |
| 23:01.35 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 23:01.39 | crich1999 | what do you say about bug #3764 (not the topic ;) |
| 23:01.43 | christo | How to I tell * to listen for incoming calls on a particular number routed to our equipment? |
| 23:01.50 | *** join/#asterisk riksta (~rick@81-178-173-194.dsl.pipex.com) |
| 23:02.02 | ManxPower | christo, That interface? Analog, T-1, or PRI? |
| 23:02.02 | paulc | christo: exten => 12345,1,SomeApp |
| 23:02.13 | ManxPower | ..er.. What interface.... |
| 23:02.23 | christo | ManxPower = E1 |
| 23:02.26 | ACiDV | Yes, I think but never heard about different type of TE410P (super/normal) :P |
| 23:02.31 | christo | s/=// |
| 23:02.36 | ManxPower | christo, Do With DID / DDI service? |
| 23:02.46 | ManxPower | arch! |
| 23:02.50 | ManxPower | christo, With DID / DDI service? |
| 23:03.37 | ManxPower | If asterisk knows the number dialed then it will try to match the dialed number to an exten => line/. |
| 23:03.39 | christo | ManxPower - sry, I have no idea :/ |
| 23:03.46 | christo | aaah I see |
| 23:04.25 | ManxPower | But if the telco doesn't send Asterisk the dialed number than asterisk will look for an exten => s. "s" means "I don't know what number was dialed". |
| 23:04.28 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 23:04.28 | christo | well I'm told the number I'm using is routed to us, but I see nothing in the asterisk console when I dial in, which is odd |
| 23:04.35 | *** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl) |
| 23:04.38 | ManxPower | I would have called it exten => unknown but I didn't write Asterisk |
| 23:04.38 | christo | right |
| 23:04.48 | christo | which is which between DID/DDI? |
| 23:05.05 | ManxPower | christo, They DID is a USA term, DDI is a Euro term. Same thing. |
| 23:05.28 | christo | aah I see.. I'll have to confirm that in the morning |
| 23:06.05 | ManxPower | Ugh! My apartment is getting as messy as my boyfriend's apartment |
| 23:06.43 | christo | I bet mine's worse ;) |
| 23:06.52 | ChulJin | RoyK: http://home.swipnet.se/cfmd/rfc/dir/fye.html. 'nuff said. |
| 23:07.11 | Tall-guy | christo: your boyfriend is worse? |
| 23:07.46 | christo | heh - my apartment :P |
| 23:07.55 | Tall-guy | sorry... :) |
| 23:07.59 | Tall-guy | had to. |
| 23:08.10 | christo | aye - I'll let you off :) |
| 23:08.14 | crich1999 | Is there someone here which has got experiences with writing asterisk channel drivers ? |
| 23:08.43 | *** join/#asterisk kfuq-lap (~somedude@c-24-17-224-78.client.comcast.net) |
| 23:09.44 | jedaustin | What is the Cisco Config section for in Asterisk@home? Do you mess with it if you don't have a Cisco router? |
| 23:11.37 | riksta | jedaustin: i think that is for use with cisco ip phones, rather than routers |
| 23:11.42 | riksta | to configure the xml services |
| 23:11.48 | jedaustin | Ah |
| 23:12.24 | jedaustin | Speaking of phones.. I've heard the Sipra phones are good, any to stay away from? |
| 23:12.43 | riksta | well they are better than the budgetone thats for sure |
| 23:12.54 | riksta | it's the shoddiest piece of kit ever |
| 23:13.10 | jedaustin | Budgetone.. who makes that? |
| 23:13.15 | riksta | grandstream |
| 23:13.17 | JerJer[mobile] | Barbietone |
| 23:13.22 | riksta | lol yeah |
| 23:13.25 | jedaustin | Ah.. I've heard of them |
| 23:13.26 | riksta | it feels like a toy, thats for sure |
| 23:13.38 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.6) |
| 23:13.43 | riksta | :) |
| 23:14.02 | jedaustin | For the money what phones are good? |
| 23:14.23 | riksta | depends how much you wanna spend |
| 23:14.52 | jedaustin | riksta: less than $150 |
| 23:14.53 | riksta | SPA-841 is ok |
| 23:14.58 | riksta | for low end |
| 23:15.00 | paulc | although the Sipura 841's aren't bad for the buck.. just quirky.. |
| 23:15.03 | crich1999 | i'm author of chan_misdn, which enables asterisk to use mISDN supported HW, mISDN gots a Kernel DSP Module which has got a very good DTMF Detector. I'd like to use that detector instead of the ast_dsp's dtmf detector, so at the moment I'm queueing for every detected DTMF Tone an AST_CONTROL_DTMF Frame, but additionally the DTMF Sound is is queued also. I'm not sure wether this is correct.. has anyone a clue ? |
| 23:15.22 | riksta | SPA-841 looks like budgetone, but build qual is much better i hear |
| 23:15.48 | *** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) |
| 23:15.50 | paulc | It's not a bad phone.. I finally got my hands on one last week and have been playing.. |
| 23:16.28 | ariel_ | riksta, it's an ok phone. I wish it had backlit screen. but it's heavy and feels like a normal phone. Keys you got to get used to them. |
| 23:16.55 | riksta | well, my 7940 doesn't have a backlight either, kinda sucks |
| 23:17.08 | riksta | are the keys rubbery |
| 23:18.29 | jedaustin | What about stuff like this: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61840&item=5760963016&rd=1 |
| 23:18.29 | ariel_ | riksta, at first the keys feel sticky. but once you use them they become normal. |
| 23:19.06 | riksta | once you covered them in shit |
| 23:19.11 | riksta | and dust |
| 23:19.12 | riksta | :) |
| 23:19.34 | ariel_ | The cost of shipping is $74.99 within the continental United States. |
| 23:19.48 | *** join/#asterisk Guybrush|Numb (~franz@host78-184.pool62211.interbusiness.it) |
| 23:19.53 | ariel_ | jedaustin, so you get it for 9.99 then pay 75 to ship it. |
| 23:20.03 | Qwell | Thats so dumb |
| 23:20.08 | jedaustin | ariel_:Ew.. that falls in the too good to be true column :) |
| 23:20.30 | ariel_ | how do people get away with that? |
| 23:20.31 | jedaustin | Just a way for them to limit their ebay fees |
| 23:20.38 | jedaustin | I should complain |
| 23:20.48 | Qwell | "Shipping to France is $7 extra in addition to the posted $74.99 s/h." |
| 23:20.56 | ariel_ | jedaustin, no if they have to refund it says refund does not include shipping and handling. |
| 23:21.12 | jedaustin | Plain old fraud |
| 23:21.20 | ManxPower | I need suggestions on where I can get reasonaly priced DDR RAM and CPU's. Newgg? |
| 23:21.23 | Guybrush|Numb | anyone that uses spandsp, i can send faxes over a TDM400P without any trouble, but receives give me the "trainin failed" error |
| 23:21.38 | Qwell | ManxPower: newegg for sure |
| 23:21.41 | jedaustin | Anyone tried phones like this one http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=41374&item=6377597771&rd=1 ? |
| 23:22.04 | SexyKen | Hey guys I have a polycom but I want to be able to use my xml screen to show queue info |
| 23:22.14 | jedaustin | Just an interface to a soft phone right? |
| 23:22.33 | ManxPower | SexyKen, Whoch Polycom model? |
| 23:22.44 | AgiNamu | wow, I'm so psyched about Asterisk + SOAP |
| 23:23.02 | jedaustin | Hmm.. This one looks promising: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61840&item=5761015260&rd=1 |
| 23:23.35 | SexyKen | IP 600 |
| 23:23.55 | crich1999 | Guybrush|Numb: Have you disable Echo Cancel and added Timing on the PSTN side port ? |
| 23:24.04 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 23:24.25 | Guybrush|Numb | crich1999 : I disabled echo cancel, but can you clarify the "timing" thing ? |
| 23:24.33 | crich1999 | kram: I need some Info about pvt_tech->send_digit |
| 23:24.33 | *** join/#asterisk herag (herag@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 23:24.39 | crich1999 | anthm: I need some Info about pvt_tech->send_digit |
| 23:24.51 | jedaustin | Polycom SoundPoint IP500 SIP Phone Good? |
| 23:25.28 | crich1999 | Guybrush|Numb: in zaptel.conf you need somethin like span=1,[0«|1],.. where the second argument means wether you get the timing from this span |
| 23:25.46 | SexyKen | ManxPower -- Yea it's a IP 600 |
| 23:25.48 | SexyKen | You got any ideas |
| 23:26.12 | Guybrush|Numb | crich1999: if i only have fxs and fxo (no tdm), i can't define spans |
| 23:26.16 | Guybrush|Numb | am i right ? |
| 23:26.18 | ManxPower | SexyKen, No, but I'm sure you've seen the mailing list messages about the subject. |
| 23:26.24 | G0shen | AgiNamu: SOAP? |
| 23:26.29 | G0shen | ~soap |
| 23:26.30 | jbot | ACTION bends over to pick up the soap, winking suggestively. |
| 23:26.37 | G0shen | doh' |
| 23:26.52 | Dr-Linux | tzanger |
| 23:27.00 | crich1999 | Guybrush|Numb: uhum .. sorry i thought this was 405P .. |
| 23:27.05 | SexyKen | ManxPower -- I haven't. |
| 23:27.37 | crich1999 | Guybrush|Numb: theres also an option: echocancel_whenbridged |
| 23:28.14 | gr8nash | anyone using LiveVOIP here? |
| 23:28.36 | crich1999 | whats LiveVOIP ? |
| 23:28.37 | AgiNamu | SOAP, yea, XML remote calls |
| 23:28.40 | SexyKen | How many calls can asterisk support on a DUAL XEON going out at one time |
| 23:28.44 | AgiNamu | So, for instance, when a call comes in |
| 23:28.59 | AgiNamu | I am setting up asterisk to query a SOAP webservice (Before the call is negotiated) |
| 23:29.17 | jedaustin | SexyKen: I would think your bandwidth would be the limiter before your processor was |
| 23:29.17 | *** join/#asterisk dalabera (~dalabera@203sdl30m10.codetel.net.do) |
| 23:29.18 | AgiNamu | So, it'll come in, and I'll do soap_call__getCodec(theDid) |
| 23:29.45 | AgiNamu | or say, I have to deliver a local number, I can query a webservice to find out how to do so (call forwarding, voicemail, etc.) |
| 23:30.01 | dalabera | Hello guys, I'm in a hurry, please help me configuring a handytone where my udp port 5060 is block |
| 23:30.03 | AgiNamu | I did some perf tests this morning, just using my desktop Windows machine as the server :) |
| 23:30.08 | dalabera | don't know where |
| 23:30.22 | AgiNamu | average response under 0.05 seconds |
| 23:30.29 | AgiNamu | 470+ requests/sec. |
| 23:30.39 | AgiNamu | limited by my crap CPU :) |
| 23:30.47 | AgiNamu | And that includes going out to SQL Server to do a lookup |
| 23:31.12 | AgiNamu | 0.05 seconds is far faster than using Asterisk to do an in-memory lookup :) |
| 23:31.30 | AgiNamu | and under real average conditions, it's more like 0.01 seconds |
| 23:31.38 | jedaustin | Hmm.. Voip payphones! |
| 23:32.09 | AgiNamu | SexyKen, depends on what you are doing |
| 23:32.18 | AgiNamu | bandwidth shouldn't be an issue if you have a real hosting centre |
| 23:32.33 | AgiNamu | If you're transcoding to G729, not many. Maybe, I dunno, 120 channels? |
| 23:32.46 | AgiNamu | 90 channels? something like that. Digium has some stats on their G729 page. |
| 23:33.03 | AgiNamu | If you're just handing the SIP traffic and not touching the media.... wow, a ton |
| 23:33.11 | jedaustin | AgiNamu: true |
| 23:33.13 | AgiNamu | And with a few hacks to the source, a LOT more. |
| 23:33.22 | AgiNamu | someone here has put in hashtables instead of linked lists |
| 23:33.26 | AgiNamu | and obviously that's a shitload faster. |
| 23:33.47 | AgiNamu | Like, sheesh, loading iax.conf and sip.conf... SUCKS |
| 23:34.01 | AgiNamu | takes 1 minute to load 25,000 friends! |
| 23:34.11 | Qwell | Takes me 20 seconds to load 2. heh |
| 23:34.20 | AgiNamu | But, strip out some of their code that traverses the linked lists, and it takes 3 seconds to load 25,000. |
| 23:36.06 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 23:36.06 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:36.14 | AgiNamu | As far as IAX registrations, I know a single server can handle at least 1000 IAX2 registrations |
| 23:37.00 | AgiNamu | But again, IIRC, they use mutexes on everything |
| 23:37.06 | AgiNamu | there's no readerwriter/spinlocks |
| 23:37.13 | AgiNamu | So that's gotta be terribly inefficient |
| 23:37.42 | pigpen | Question: is there an open source implemention of the G.729 codec? |
| 23:37.48 | AgiNamu | pigpen yes |
| 23:38.01 | pigpen | works with asterisk? |
| 23:38.06 | AgiNamu | Go google the asterisk lists. there's intel's G729 code + some asterisk wrapper |
| 23:38.07 | AgiNamu | yea |
| 23:38.16 | pigpen | Great..Thanks! |
| 23:38.22 | AgiNamu | but you can't legally license it |
| 23:38.22 | AgiNamu | so |
| 23:38.34 | AgiNamu | Well, I should say, you CAN |
| 23:38.42 | AgiNamu | but i dont know how or who |
| 23:38.49 | jedaustin | I found a review of ipphones on the net.. http://www.commweb.com/pbx/showArticle.jhtml?articleId=54800721 They only reviewed 5 vendors though. Anyone have the Shoretel 560 phone they give top marks to? |
| 23:38.53 | AgiNamu | it's unlike the [expensive] Digium G729 codec. |
| 23:39.09 | denon | $10 per channel is expensive? |
| 23:39.12 | AgiNamu | yea |
| 23:39.13 | denon | doesnt seem to terrible |
| 23:39.18 | AgiNamu | not horrible |
| 23:39.25 | AgiNamu | Digium only pays like $2 or so |
| 23:39.29 | denon | I mean, one-time fee |
| 23:39.33 | AgiNamu | actually, probably closer to $1 or so |
| 23:39.37 | denon | yeah .. |
| 23:39.43 | denon | shrugs .. seems like a fair amount of hassle .. |
| 23:39.46 | AgiNamu | so they're marking it up 10x |
| 23:39.51 | denon | specially since they probably sell it 1-off |
| 23:39.53 | AgiNamu | yea, they have to prepay a ton |
| 23:39.56 | denon | 1 or 2 channels to home lusers |
| 23:39.58 | AgiNamu | like $50,000 worth |
| 23:40.02 | jhiver | well, you know, they give away asterisk for like $0 |
| 23:40.12 | AgiNamu | nah, that's all automated. |
| 23:40.16 | denon | they probably get cheap if you want tons of em |
| 23:40.25 | AgiNamu | Although I *really* hate the stupid activation system |
| 23:40.26 | denon | few hundred channels or such |
| 23:40.27 | AgiNamu | annoying as hell |
| 23:40.32 | AgiNamu | I'm so tempted to crack it :P |
| 23:40.40 | denon | hehe |
| 23:40.53 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 23:40.55 | jhiver | so I don't mind paying some money to digium for g729 |
| 23:40.56 | AgiNamu | Hey, I'm more than willing to pay for asterisk |
| 23:41.01 | AgiNamu | I practically begged them |
| 23:41.12 | AgiNamu | But they are not business people , so... |
| 23:41.13 | denon | buy bumper stickers ;) |
| 23:41.46 | AgiNamu | Digium needs to hire a program manager. |
| 23:41.53 | jhiver | k lads, it's 4am for me, need to go to bed :-/ cya! |
| 23:41.57 | AgiNamu | cya |
| 23:42.11 | pigpen | Ok...so with the Open Source G.729 codec...can I use it commercially...free? |
| 23:42.16 | AgiNamu | no pigpen |
| 23:42.18 | AgiNamu | you can't license it |
| 23:42.27 | mogorman | NO |
| 23:42.29 | pigpen | ok..so essentially...pay up |
| 23:42.29 | `Sauron | nope |
| 23:42.30 | mogorman | is illegal... |
| 23:42.36 | mogorman | in most of the world |
| 23:42.42 | `Sauron | you know |
| 23:42.44 | mogorman | but if you live in elbonia |
| 23:42.46 | mogorman | your good |
| 23:42.47 | AgiNamu | it's for testing, development, etc. |
| 23:42.49 | `Sauron | $10 is like, a drop in the bucket |
| 23:42.51 | pigpen | Best bet to just buy the G.729 from digium? |
| 23:42.52 | AgiNamu | Intel's reference code |
| 23:42.55 | mogorman | still illegal |
| 23:43.03 | AgiNamu | how is it illegal mogor? |
| 23:43.13 | AgiNamu | ACTUALLY, you might be able to buy a G729 license from someone else |
| 23:43.17 | AgiNamu | and use that with the open source one |
| 23:43.18 | mogorman | it is illegal to compile and run anywhere |
| 23:43.19 | AgiNamu | and be fine |
| 23:43.23 | mogorman | cant |
| 23:43.26 | mogorman | still illegal |
| 23:43.29 | AgiNamu | I dont believe that. |
| 23:43.31 | jedaustin | Will a phone that supports iAX natively perform better than a sip phone (or is a iax phone a sip phone?)_ |
| 23:43.31 | mogorman | its in the license |
| 23:43.31 | bkw_ | Moc |
| 23:43.32 | bkw_ | damn it |
| 23:43.33 | bkw_ | wake up |
| 23:43.35 | AgiNamu | "here's some source, but it's illegal to compile" |
| 23:43.36 | bkw_ | mogorman, hey boi |
| 23:43.37 | mogorman | it doesnt matter hat you believe |
| 23:43.39 | mogorman | its the law |
| 23:43.45 | `Sauron | is the g729 an integer codec, or a FP |
| 23:43.45 | mogorman | hey bkw |
| 23:43.50 | AgiNamu | Yea, care to point to any backing material? |
| 23:43.52 | mogorman | integer |
| 23:43.57 | mogorman | still illegal though... |
| 23:43.57 | `Sauron | or, anyone know if there's a integer version of it |
| 23:44.01 | bkw_ | mogorman, should we close all the bugs associated with meetme and make them one bug.. since we have like 5 from Moc all with tiny changes |
| 23:44.12 | bkw_ | illegal eh |
| 23:44.16 | mogorman | that and, we should close those feature ones |
| 23:44.17 | bkw_ | what are the codec cops gonna show up? |
| 23:44.23 | mogorman | i agree |
| 23:44.36 | mogorman | but when the bsa knocks on your door |
| 23:44.38 | AgiNamu | OK, here's something you could do. just go license G729 and write your own implementation. |
| 23:44.41 | mogorman | are you ready to foot half a mil |
| 23:44.48 | mogorman | yes |
| 23:44.50 | mogorman | that is legal |
| 23:44.53 | AgiNamu | mogorman, so you're saying that Intel has a contract with the BSA for their sample G729 code? |
| 23:44.54 | bkw_ | bsa? |
| 23:44.55 | mogorman | but costs several million |
| 23:44.57 | bkw_ | at my HOUSE? |
| 23:44.59 | Guybrush|Numb | mogorman: elbonia is no good, very easy misunderstandings there :) |
| 23:45.00 | AgiNamu | I highly, highly , doubt that |
| 23:45.00 | bkw_ | riiigiht |
| 23:45.05 | mogorman | lol |
| 23:45.07 | AgiNamu | mogorman, wtf? its like $50K minimum purchase. |
| 23:45.11 | mogorman | yeah |
| 23:45.16 | mogorman | well i dont have 50k |
| 23:45.18 | bkw_ | ya know I don't mind paying for software.. or usage of patents |
| 23:45.18 | mogorman | maybe bkw |
| 23:45.20 | bkw_ | but come on |
| 23:45.22 | AgiNamu | not several million. |
| 23:45.22 | mogorman | you have 50k? |
| 23:45.23 | bkw_ | lets get seious |
| 23:45.25 | bkw_ | er serious |
| 23:45.28 | AgiNamu | At any rate, you can find someone who already has licensed G729 |
| 23:45.29 | AgiNamu | and buy from them |
| 23:45.33 | AgiNamu | i.e., digium isn't the only ones |
| 23:45.36 | mogorman | but they have to pay rates |
| 23:45.37 | mogorman | i agree |
| 23:45.41 | mogorman | digium maybe shouldnt be |
| 23:45.44 | mogorman | but currently |
| 23:45.49 | AgiNamu | And I dont think you're correct about intels code being "illegal to compile" |
| 23:46.03 | bkw_ | ya know they can all bite me |
| 23:46.05 | mogorman | the only way to legally use is digi729 |
| 23:46.07 | AgiNamu | There wouldnt be much fucking point to making "illegal to compile code" now would there? |
| 23:46.08 | bkw_ | B I T E M E |
| 23:46.21 | mogorman | it is illegal to have binary |
| 23:46.26 | mogorman | it is just the way it is |
| 23:46.29 | AgiNamu | Mogorman, so what's the point of the code? |
| 23:46.33 | mogorman | bkw_ can you speak up |
| 23:46.35 | mogorman | for me |
| 23:46.38 | AgiNamu | This is is such a ridiculous argument. |
| 23:46.49 | mogorman | i know its stupid |
| 23:46.50 | AgiNamu | "Hi, this is Intel. We wrote some code that shows off something. BUT, you can't compile it." |
| 23:46.52 | mogorman | go read gpl though |
| 23:46.53 | ManxPower | buy your G729 license from Digium. THEY payed the 50K already. |
| 23:46.55 | mogorman | it is for ref |
| 23:47.03 | AgiNamu | Intel's code isn't GPL'd |
| 23:47.06 | bkw_ | haha |
| 23:47.07 | AgiNamu | and this has nothign to do with GPL |
| 23:47.08 | ManxPower | I think mark said that it would take them 5 years to sell enough G729 licenes to break even on it. |
| 23:47.19 | ManxPower | Intel's G729 isn't even licensed. |
| 23:47.25 | AgiNamu | There. |
| 23:47.39 | AgiNamu | So the GPL doesn't fit into any of this. |
| 23:47.51 | bkw_ | its odd I have some code here that says GPL |
| 23:48.20 | AgiNamu | Regardless, even if it WAS GPL'd, the GPL doesn't say "illegal to compile GPL code", so wtf. |
| 23:48.24 | mogorman | a large majority is gpl |
| 23:48.31 | mogorman | if someone had read code |
| 23:48.36 | mogorman | instead of just ripped it off... |
| 23:48.45 | mogorman | its not illegal in all places |
| 23:48.47 | mogorman | just most |
| 23:48.55 | mogorman | go write your congress man |
| 23:48.57 | bkw_ | ok MOVE ALONG |
| 23:48.57 | AgiNamu | again, wtf are you talking about? |
| 23:48.58 | bkw_ | ok MOVE ALONG |
| 23:49.01 | bkw_ | ok MOVE ALONG |
| 23:49.04 | mogorman | heh |
| 23:49.05 | bkw_ | ok MOVE ALONG |
| 23:49.06 | mogorman | thanks bkw |
| 23:49.08 | bkw_ | Lets smash these bugs |
| 23:49.13 | mogorman | word |
| 23:49.16 | bkw_ | is it safe to smash all 4 of the meetme bugs from Moc? |
| 23:49.17 | `Sauron | dum di dum |
| 23:49.20 | bkw_ | and say MAKE ONE HUGE BUG |
| 23:49.23 | bkw_ | or start MeetMe2 |
| 23:49.25 | `Sauron | what happened to voip-info? |
| 23:49.34 | `Sauron | just sloooow |
| 23:49.42 | mogorman | no meetme2... |
| 23:49.57 | ManxPower | You want a G729 codec? Download this http://www.fnords.org/~eric/asterisk/wffs.tar.gz There's even the G723.1 codec in there |
| 23:50.06 | bkw_ | not a codec |
| 23:50.10 | bkw_ | an implementation |
| 23:50.12 | bkw_ | from the ITU |
| 23:50.13 | *** join/#asterisk ClayReiche123 (~creiche@65.35.112.199) |
| 23:50.15 | bkw_ | thats it |
| 23:50.19 | ManxPower | It's not LICENSED, and won't even build for Asterisk, and it's from the IRU. |
| 23:50.19 | bkw_ | I have a g728 codec too |
| 23:50.32 | ManxPower | You want it, go get it. |
| 23:50.37 | mogorman | you too man... |
| 23:50.47 | mogorman | bkw mosh bugs for meetme and features |
| 23:51.03 | AgiNamu | Manx, don't you get it? Theres this magical thing for code that makes it ILLEGAL to compile! |
| 23:51.07 | ManxPower | You also get the entire World Famous Fnords (Asterisk) Site. |
| 23:51.18 | ManxPower | AgiNamu, no, it's the license. |
| 23:51.25 | mogorman | thank you |
| 23:51.33 | ManxPower | Talk to the hand...er...patent holder. |
| 23:51.40 | mogorman | aginamu laws dont have to make since |
| 23:51.43 | mogorman | sense* |
| 23:51.51 | AgiNamu | mogorman, im not talking about laws |
| 23:52.03 | mogorman | if we are talking about illegality |
| 23:52.03 | AgiNamu | Why would intel write code, debug it, make it work, etc. and then say it's illegal to compile. |
| 23:52.04 | ManxPower | the patent holder is the only people with the LEGAL authority to tell you what you can and cannot do with G729 code. |
| 23:52.08 | mogorman | we are taling about laws |
| 23:52.11 | AgiNamu | thank you manx |
| 23:52.15 | mogorman | thank you manxpower |
| 23:52.24 | ManxPower | AgiNamu, Because you can get a license from the patent holders and once you have that.... |
| 23:52.25 | mogorman | and they have stated it was illegal |
| 23:52.27 | *** join/#asterisk coolacid-p (~jason@216.99.98.39) |
| 23:52.58 | AgiNamu | OK, well, they should go after Intel for INDUCING people to compile and make illegal binaries |
| 23:53.11 | ManxPower | I don't know if the code is illegal to COMPILE, but I do know it's illegal to use, even in a non-commercial/educational way without a license. |
| 23:53.25 | AgiNamu | at any rate, I'm not worried about that, and I'll gladly buy a digium license and run the open source one if there was any point |
| 23:53.36 | AgiNamu | and they can go work things out in court if they ever magically find out |
| 23:53.36 | *** join/#asterisk cjk (~cjk@80.92.75.143) |
| 23:53.44 | bkw_ | mogorman, now go close jabber pops.. and we'll be at 99 |
| 23:53.45 | bkw_ | haha |
| 23:53.48 | AgiNamu | because quite frankly, it wont make one flying fuck of difference. |
| 23:53.49 | mogorman | heh |
| 23:53.53 | mogorman | i probably should |
| 23:53.59 | mogorman | res_jabber is comming along nicely |
| 23:54.09 | mogorman | but at same time i wanted more people to see it |
| 23:54.13 | mogorman | and try it.... |
| 23:54.29 | mogorman | maybe i will reimplement it tonight with res_jabber, and release both |
| 23:54.40 | mogorman | well good for you aginamu |
| 23:54.46 | mogorman | but it doesnt change the facts |
| 23:54.51 | AgiNamu | nope, not at all |
| 23:54.57 | bkw_ | mogorman, res_jabber baby |
| 23:54.58 | bkw_ | ya |
| 23:54.58 | mogorman | but i do agree most courts would throw such a case out |
| 23:54.58 | AgiNamu | But to someone interested in doing that, it's fine |
| 23:55.19 | mogorman | yeah its cool, but i have nothing to implement it with : ( |
| 23:55.31 | mogorman | i mean other than using manager to control and get events |
| 23:55.32 | AgiNamu | and actually, the patent holders DONT have full control over everything |
| 23:55.33 | `Sauron | anyone here have experience with iaxclient? |
| 23:55.36 | mogorman | which is disgusting |
| 23:55.42 | AgiNamu | if they license something, then they're bound to their licensing |
| 23:55.49 | `Sauron | or iaxlib |
| 23:55.50 | AgiNamu | So it's all about going over how strong they license, etc. |
| 23:56.16 | mogorman | omg bkw_ how did we get to 100 |
| 23:56.21 | mogorman | im tempted to add a bug |
| 23:57.11 | cjk | hi, i do not know how to get tftpd up and running. maybe anyone here who can paste me his config. tftp exists since X years and it isn as advanced as * |
| 23:57.37 | mogorman | can we kill the oh323 bug bkw |
| 23:57.43 | mogorman | its not related to asterisk... |
| 23:58.06 | bkw_ | mogorman, ok kill it |
| 23:59.04 | `Sauron | cjk: the man page isn't rocket science |
| 23:59.24 | *** join/#asterisk carrar (tim@osburn.com) |
| 23:59.27 | cjk | `Sauron, i do not know, but really i tried everything, but it is not working as it should |
| 23:59.36 | jedaustin | Any major retailer carry IP phones? |
| 23:59.40 | carrar | holy massive more people in here these days |
