| 00:00.12 | *** join/#asterisk _dusty (n=dusty@12-219-148-217.client.mchsi.com) |
| 00:00.21 | Qwell[] | So, how about them Bears? |
| 00:00.25 | Nugget | The initials "DC" are an abbreviation for Detective Comics, after one of the company's flagship titles. |
| 00:00.29 | mog_work | Da' Bears |
| 00:00.31 | Nugget | good 'ol internet. |
| 00:00.34 | mog_work | ahh Nugget |
| 00:00.38 | mog_work | thats interesting |
| 00:00.42 | Shaun222 | jesus christ you guys are still going off on DC... |
| 00:00.46 | *** join/#asterisk alexis101 (n=alexis@70.54.204.92) |
| 00:00.48 | Qwell[] | Shaun222: It's your fault |
| 00:00.51 | mog_work | isnt detective comics where batman came from? |
| 00:00.55 | justinu | so is it "Detective Comics Comics"? |
| 00:00.55 | mog_work | like issue 17 |
| 00:01.11 | Qwell[] | justinu: nobody calls it DC Comics |
| 00:01.12 | Nugget | However, the fame of Detective Comics was assured by issue #27 (May 1939), which featured the first appearance of Batman (as "The Bat-Man"). He would eventually become the star of the title. |
| 00:01.24 | mog_work | actually alot of peoplce call it dc commics |
| 00:01.24 | Nugget | http://en.wikipedia.org/wiki/Detective_Comics |
| 00:01.30 | Qwell[] | oh |
| 00:01.36 | mog_work | just like you go to the atm machine |
| 00:01.36 | justinu | i've always heard people refer to it as DC comics |
| 00:01.40 | mog_work | or read the bible |
| 00:01.44 | Qwell[] | mog_work: huh? |
| 00:01.45 | mog_work | or lcd display |
| 00:01.49 | Vco | stfu about the bible |
| 00:01.51 | Nugget | DC Comics call themselves "DC Comics" |
| 00:01.51 | justinu | i'll put some NIC cards in my pc |
| 00:01.52 | Qwell[] | I don't get that last one |
| 00:01.54 | mog_work | bible means the book |
| 00:01.55 | alexis101 | hello everyone ... I was wondering if anyone know how to tell asterisk to not enter a queue if there is no agent loged in this queue ? |
| 00:01.57 | Vco | heard enough of that shit yesterday |
| 00:02.01 | mog_work | so the the book |
| 00:02.01 | Nugget | http://www.dccomics.com/ |
| 00:02.02 | bkw__ | direct current (abbr.: DC) noun an electric current flowing in one direction only. Compare with alternating current . |
| 00:02.04 | Qwell[] | ahh, redundant the |
| 00:02.04 | bkw__ | NEXT!!! |
| 00:02.10 | bkw__ | move on |
| 00:02.24 | Qwell[] | in context, DC actually did mean data center though :P |
| 00:02.31 | mog_work | yes it did |
| 00:02.33 | mog_work | in the beginning |
| 00:03.04 | mog_work | alexis101, show queues |
| 00:03.10 | mog_work | or something like that |
| 00:03.29 | alexis101 | well i mean in the extentions.conf or something like that |
| 00:03.30 | Qwell[] | joinempty=no |
| 00:03.43 | alexis101 | this is in the queues.conf ? |
| 00:03.46 | Qwell[] | yes |
| 00:03.54 | alexis101 | thanks |
| 00:04.12 | Shaun222 | any recommendations on a good reliable sip provider.... |
| 00:04.48 | shmaltz | anybody here like snapping wrapping bubbles that come with those nice VoIP phones? |
| 00:05.00 | mog_work | amen shmaltz |
| 00:05.53 | *** join/#asterisk St1ckm4n (n=shortes9@68.178.74.166) |
| 00:06.06 | Abydos313 | its hard to resist popping those bubbles |
| 00:06.32 | mog_work | i hate those bags of air though |
| 00:06.37 | mog_work | i like the bubble wrap |
| 00:06.38 | shmaltz | http://www.snapbubbles.com/ |
| 00:06.51 | mog_work | aww flash |
| 00:08.53 | St1ckm4n | I'm sure this question get's asked a million times and I feel bad posting it here but I haven't found a solution despite all of my googling |
| 00:09.13 | St1ckm4n | what's the easiest way to show actual # of calls in combined queues? |
| 00:09.40 | mog_work | show queues and a caluclator? |
| 00:09.45 | _Sam-- | show queues? |
| 00:09.59 | mog_work | cli command |
| 00:10.02 | St1ckm4n | ok |
| 00:10.04 | mog_work | also works over manager |
| 00:10.09 | mog_work | i believe |
| 00:10.14 | _Sam-- | i was answering the question..not asking :) |
| 00:10.26 | _Sam-- | i think there is software that does queue status now |
| 00:10.29 | _Sam-- | i think zoa wrote some |
| 00:10.32 | St1ckm4n | I looked at that and was hoping there might have been another command so I wouldn't have to parse all those lines and add the totals |
| 00:10.37 | _Sam-- | it sits in the tray of your windows |
| 00:10.43 | _Sam-- | ~seen zoa |
| 00:10.47 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 12d 7h 27m 30s ago, saying: 'it looks kinda suspicious :p'. |
| 00:11.47 | Qwell[] | ohhhh file... |
| 00:11.49 | _Sam-- | http://www.asteriskguru.com/tutorials/asterisk_queue_statistics.html |
| 00:11.51 | Qwell[] | where are you? |
| 00:11.57 | _Sam-- | er |
| 00:12.00 | _Sam-- | i guess he took it down |
| 00:12.12 | _Sam-- | http://www.asteriskguru.com/tutorials/queue_stats_product_overview.html |
| 00:12.24 | _Sam-- | hmm that doesnt show you real time status, sorry. |
| 00:12.27 | alexis101 | well i know that joinempty=no does'nt work |
| 00:12.53 | St1ckm4n | yeah I'm running queue stats having logs dumped every hour |
| 00:13.39 | _Sam-- | you could probably (ugh) use something like flash operator panel |
| 00:13.46 | _Sam-- | i know it shows each queue and how many calls are in it |
| 00:13.53 | St1ckm4n | the version of fop we have now seems kind of buggy |
| 00:14.05 | St1ckm4n | some ghost calls seem to hang even though they aren't in a queue |
| 00:14.08 | _Sam-- | i think thats probably most version |
| 00:14.09 | _Sam-- | s |
| 00:14.12 | _Sam-- | but i wouldnt really know |
| 00:14.50 | St1ckm4n | I was hoping someone would've already done the legwork of pulling queue stats on a php page |
| 00:14.50 | _Sam-- | how many queues do you have? |
| 00:15.04 | St1ckm4n | it wouldn't be such a big deal if my stupid !feof was working |
| 00:15.10 | St1ckm4n | 18 |
| 00:15.24 | _Sam-- | it should only take a few hours to do in php |
| 00:15.34 | _Sam-- | depending on how nice you format it :) |
| 00:15.55 | St1ckm4n | yeah I did one to show agent status real time, my problem is after I socket and send my command is getting the results |
| 00:16.17 | St1ckm4n | for some reasion my !feof keeps hanging and I have to do a while loop for my fgets |
| 00:16.32 | St1ckm4n | meaning I have to count the # of lines it's returning |
| 00:16.37 | mog_work | ~thebook |
| 00:16.40 | jbot | hmm... thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 00:16.40 | _Sam-- | how are you sending it? through the manager interface or from php to run |
| 00:16.47 | St1ckm4n | from php |
| 00:17.07 | _Sam-- | does your php connect to manager or run a shell type script |
| 00:17.15 | St1ckm4n | connects to manager |
| 00:17.22 | _Sam-- | because you could run 'asterisk -rx show queue blah' |
| 00:17.26 | _Sam-- | from a shell type |
| 00:17.50 | _Sam-- | we connect php to manager fine here...but not for doing what you want |
| 00:17.55 | _Sam-- | i would love to though :) |
| 00:18.15 | St1ckm4n | I'll try using a shell script it'll probably be cleaner too |
| 00:18.24 | St1ckm4n | thx |
| 00:18.35 | _Sam-- | sure thing, let me check it out when you get it :) |
| 00:18.45 | St1ckm4n | will do |
| 00:19.07 | St1ckm4n | I'm still a noob when it comes to this so it probably won't be pretty but hopefully functional |
| 00:19.19 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 00:19.47 | _Sam-- | i can have someone clean it up if it works well |
| 00:20.01 | _Sam-- | the command would be more like asterisk -rx 'show queue blah' |
| 00:20.16 | _Sam-- | or show queues |
| 00:20.35 | St1ckm4n | yeah, I'm going to do the show queues |
| 00:20.53 | St1ckm4n | the one I did for agents works but just feels hoakey |
| 00:21.22 | St1ckm4n | I have it refresh every 5 seconds and change the agent's color depending on their status |
| 00:21.36 | _Sam-- | how many agents do you have? |
| 00:21.52 | St1ckm4n | the most we ever have logged in at a time is around 10 |
| 00:22.06 | terrapen | dammit voipsupply |
| 00:22.09 | _Sam-- | what phones do you use? |
| 00:22.10 | terrapen | answer your damned phones! |
| 00:22.17 | terrapen | i want to spend money! |
| 00:22.21 | _Sam-- | terrapen...as in station, or UMD? |
| 00:22.22 | St1ckm4n | polycom P301 |
| 00:22.23 | *** join/#asterisk duplex- (n=simplex@72.242.34.141) |
| 00:22.26 | St1ckm4n | for the agents |
| 00:22.45 | terrapen | sam, eh? |
| 00:22.46 | St1ckm4n | they were just alot cheaper than the ciscos and a little bit cheaper than the linksys |
| 00:22.51 | duplex- | I'm having these problems when I try to compile mpg123 in the subversion repositories for asterisk. http://pastebin.ca/47225 |
| 00:23.02 | _Sam-- | sorry....terrapin station = grateful dead, terrapin station = univ. of maryland. |
| 00:23.08 | terrapen | oh, station |
| 00:23.14 | terrapen | terrapin was taken |
| 00:23.16 | _Sam-- | nice |
| 00:23.28 | terrapen | well, historically it was. so i have always been terrapen |
| 00:23.35 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 00:23.49 | _Sam-- | you see many shows? |
| 00:23.58 | terrapen | i've been to maybe 8 |
| 00:24.16 | terrapen | so not a ton of them |
| 00:24.20 | terrapen | listened to many :) |
| 00:24.57 | _Sam-- | i still listen to many myself |
| 00:25.12 | _Sam-- | St1ckm4n : who uses the agent status page? |
| 00:25.28 | St1ckm4n | we put them up on some LCD's we bought and some managers pull the web page locally |
| 00:25.46 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 00:25.52 | _Sam-- | you could probably just get some type of BLF thing |
| 00:25.57 | terrapen | anybody used a Redfone foneBridge yet? |
| 00:26.09 | _Sam-- | like a polycom with a sidecar thingie |
| 00:26.16 | St1ckm4n | BLF? |
| 00:26.17 | _Sam-- | and they could see which agents are doing whatever |
| 00:26.28 | _Sam-- | busy lamp field |
| 00:26.52 | *** join/#asterisk newmember (n=username@ptr-66-11-81-65.ptr.terago.ca) |
| 00:26.56 | _Sam-- | like my cheap granstreams, i can see who is on the phone, up to 7 |
| 00:27.00 | St1ckm4n | the page I have now is working decent, it shows the agent's extension their last name and the queue they're on |
| 00:27.18 | St1ckm4n | i have them yellow if they're on a call and blue if they're available |
| 00:27.19 | newmember | what are people using for PoE switches these days |
| 00:27.19 | terrapen | or has anybody seen a thingee that can take a PRI and send out a PRI to an Asterisk server, failing over to a second server should the primary fail? |
| 00:27.42 | terrapen | like, it would have three PRI interfaces. one from my provider and one to each of my asterisk boxes |
| 00:28.03 | terrapen | i need some way to do failover |
| 00:28.10 | _Sam-- | or you could have a quad pri card and plug all the pris into one * |
| 00:28.20 | _Sam-- | and write your dialplan well |
| 00:28.54 | newmember | terrapen: PRI -> thingee ->* 1 and ->*2 |
| 00:29.26 | shmaltz | somehting wrong with yahoo.com |
| 00:29.31 | shmaltz | anybody can confirm? |
| 00:29.36 | _Sam-- | if you have good hardware, i think the there is maybe a better chance of your PRI going down than the hardware anymore |
| 00:29.41 | _Sam-- | at least that is my opinion about MY hardware |
| 00:30.10 | _Sam-- | no yahoo for me |
| 00:30.12 | Grizzy | shmaltz - yahoo chat drops messages often |
| 00:30.28 | shmaltz | Grizzy, this is not from chat, but direct DNS lookup |
| 00:31.00 | _Sam-- | wonder if its just certain backbones |
| 00:31.17 | _Sam-- | but then again im on a two or three on this connection |
| 00:31.25 | Grizzy | I've been getting congestion on my music captures... |
| 00:31.30 | newmember | terrapen: I use cisco access servers to improve pstn connectivity |
| 00:31.38 | newmember | terrapen: http://www.voip-info.org/wiki/view/Failover+switches |
| 00:31.47 | *** join/#asterisk _Simon (n=IRC@i216-58-40-193.cybersurf.com) |
| 00:31.57 | _Sam-- | nah you're right, i got no yahoo dns on like 4 or more different backbones |
| 00:32.29 | shmaltz | looks like ns5.yahoo.com is down |
| 00:32.57 | shmaltz | http://www.dnsstuff.com/tools/lookup.ch?name=www.yahoo.com&type=A |
| 00:33.01 | shmaltz | that gives the answer |
| 00:33.22 | Grizzy | ok, I'm getting "non-authoritative" results for yahoo.com |
| 00:33.28 | harlequin516 | Okay I have FWD working now but when I try to dial # it attempts to transfer the call to local extension instead of sending DTMF #. Where do I set this not to happen? |
| 00:33.56 | _Sam-- | its a google takeover |
| 00:34.21 | harlequin516 | Aliens or did someone dig up some tree roots and knock out the e-world? |
| 00:35.19 | shmaltz | bottom line is Yahoo is off the map |
| 00:35.47 | shmaltz | but Digium.com and www.2600.com still work |
| 00:35.56 | Grizzy | Aliens lasered a north-south line through north america. |
| 00:36.11 | harlequin516 | Wow |
| 00:36.23 | shmaltz | harlequin516, wow what? |
| 00:36.36 | Pegger | has anyone set up billing with asterisk before in real time? |
| 00:36.39 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:36.41 | harlequin516 | Its exciting somehow when things go really really wrong. Somewhere outthere there are some very panicked people.... |
| 00:36.56 | gandhijee | how do i bind my 4 zap lines in to a group? |
| 00:37.00 | _Sam-- | with many others breathing down their necks |
| 00:37.11 | [av]bani | whee. polycom nat sux |
| 00:37.16 | gandhijee | i;ve been lookin the asteriskTFOT book and i see nada on it |
| 00:37.21 | _Sam-- | i like working under that kind of stresee myself |
| 00:37.31 | _Simon | could someone help me with getting a iax trunk working in my extensions.conf? I don't understand what to do but I have the provider in iax.conf |
| 00:37.36 | shmaltz | harlequin516, not really, its just an outage ;) |
| 00:37.37 | Abydos313 | yahoo works here |
| 00:37.44 | harlequin516 | I like when people are breathing down my neck, but its not at all important. |
| 00:37.46 | Abydos313 | just kidding, heh |
| 00:37.52 | Pegger | Abydos313, what part of yahoo |
| 00:38.02 | shmaltz | [av]bani, took you so long to figure that out? |
| 00:38.20 | Abydos313 | won't come up here either |
| 00:38.24 | shmaltz | Abydos313, where you located? |
| 00:38.29 | Abydos313 | california |
| 00:38.35 | St1ckm4n | sam in order for my asterisk php to execute a shell do I need to add the web server user asterisk to some file for permission? |
| 00:38.49 | Abydos313 | yahoo is down |
| 00:38.53 | Abydos313 | i was kidding |
| 00:39.31 | _Sam-- | St1ckm4n : shouldnt have to, as long as whatever runs your php can run the script |
| 00:39.36 | shmaltz | Abydos313, I was about to tell you that your provider is worth shit, because if Yahoo works for you then your provider has been caching the DNS records for far toooooooooo long |
| 00:40.01 | _Sam-- | like if your web server runs as "apache" you might want to make sure user apache can run "asterisk -rx..." |
| 00:41.27 | St1ckm4n | yeah I made sure the permissions was there |
| 00:41.36 | St1ckm4n | I keep getting a "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
| 00:42.23 | *** join/#asterisk tophat (n=umlaut@203.29.62.189) |
| 00:42.33 | _Sam-- | you're sure you can run the command fine from your shell? |
| 00:42.54 | St1ckm4n | yep |
| 00:43.33 | cytrak | just out of curiosity when they might be a release of * for MAC servers ? |
| 00:45.02 | St1ckm4n | sorry Sam i lied, I can execute the script from shell when I was logged in but not when I su'd to asterisk |
| 00:45.33 | _Sam-- | you run all your asterisk as root? |
| 00:45.57 | St1ckm4n | yeah the company that configured this switch for us left it all up as root |
| 00:46.34 | _Sam-- | im sure you could add user asterisk to a group and make it work |
| 00:50.07 | *** join/#asterisk Splatty47 (n=splatski@host217-34-149-45.in-addr.btopenworld.com) |
| 00:50.48 | Splatty47 | hello, any one know why I cant telnet my asterisklive box on port 5060 ? I should be able to do this right ? |
| 00:50.56 | orlock | no |
| 00:50.59 | orlock | asterisk uses udp |
| 00:51.01 | orlock | telnet is tcp |
| 00:51.02 | orlock | :) |
| 00:51.05 | Splatty47 | heh |
| 00:51.07 | orlock | well, SIP is UDP rather |
| 00:51.20 | Splatty47 | its driving me crazy |
| 00:51.29 | orlock | tcpdump to watch the packets |
| 00:51.31 | orlock | :) |
| 00:51.37 | shmaltz | orlock, sip over UDP is UDP |
| 00:52.18 | Shaun222 | is SIP the most commen/best to use? |
| 00:52.19 | orlock | dos sip over tcp exist? |
| 00:52.46 | duplex- | Is mpg123 necessary to have music play while someone's on hold? |
| 00:52.47 | shmaltz | orlock, yep, but Asterisk does't support, although I have seen some work on the bug tracker to add TCP |
| 00:52.53 | Splatty47 | OK, this is awful. I'm not used to asking for help! I can't get my Snom 360's to connect to my asterisk live box |
| 00:52.55 | tophat | can anybody offer more extensive advice to get incoming callerid id than the info on this page http://www.voip-info.org/wiki/view/Asterisk+and+Australian+Caller+ID |
| 00:52.55 | _Sam-- | duplex- : no. |
| 00:53.13 | shmaltz | duplex, if you don't install format_mp3 from asterisk-addons then yes |
| 00:53.26 | duplex- | I see. |
| 00:53.41 | duplex- | Thanks guys. I'm having compile errors with make mpg123, so I'll just install the addons I guess. |
| 00:53.54 | shmaltz | duplex, what distro you using? |
| 00:54.16 | duplex- | debian, but I'm compiling the subversion repositories, not the debian package. |
| 00:54.45 | shmaltz | duplex, I don't want to know |
| 00:54.55 | shmaltz | duplex, I think debian is over complicated |
| 00:55.12 | Splatty47 | is there a GUI for setup ? |
| 00:55.13 | tophat | hmm... |
| 00:55.19 | Splatty47 | that can do the conf files for me ? |
| 00:55.26 | ManxPower | um, "make mpg123" in the asterisk source directory |
| 00:55.46 | duplex- | ManxPower: Did that. Compile errors :-/ |
| 00:55.56 | ManxPower | duplex-, never heard of that before |
| 00:55.58 | duplex- | I'll just install the asterisk-addons instead. |
| 00:56.06 | ManxPower | unless, of course, you don't have any of the -dev packages installed |
| 00:56.21 | duplex- | ManxPower: http://pastebin.ca/47225 |
| 00:56.34 | ManxPower | duplex-, sorry, I don't fix that sort of stuff. |
| 00:56.49 | shmaltz | Splatty47, yes its called vi |
| 00:56.49 | duplex- | That's the errors, in case you were wondering. |
| 00:57.51 | ManxPower | duplex-, no idea |
| 00:57.58 | shmaltz | duplex, why don't you try downloadin mpg123 from the original site? |
| 00:58.12 | tophat | would you guys know how to get callerid happening? |
| 00:58.14 | ManxPower | shmaltz, I thought make mpg123 did that. |
| 00:58.25 | duplex- | shmaltz: Well make mpg123 wgets it from the original site. |
| 00:58.26 | Splatty47 | shmaltz: LOL, , using nano! |
| 00:58.28 | ManxPower | tophat, not without knowing what COUNTRY you are in. |
| 00:58.43 | tophat | ah, yes, sorry, in australia |
| 00:58.45 | duplex- | Thanks anyways, I'll just use the asterisk addons, that's probably better anyways. |
| 00:58.49 | ManxPower | no idea then |
| 00:58.56 | Shaun222 | i'm reading the install docs from http://www.digium.com/en/docs/asterisk_handbook/downloading_compiling.html , it talks about downloading zaptel, capata, libpri, and asterisk. |
| 00:58.56 | tophat | fair enough |
| 00:58.57 | shmaltz | ManxPower, but looking at the errors it looks like asterisk has some real problems when passing the make command ot the source of mpg123 |
| 00:59.14 | Splatty47 | if I just buy the business edition, will digium support talk me through setup ? |
| 00:59.16 | ManxPower | shmaltz, looks like someone broke it in Asterisk |
| 00:59.20 | shmaltz | duplex, try changing into the mpg directory and do make linux from there |
| 00:59.23 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 00:59.25 | Shaun222 | do i need all of those for just a simple asterisk server using sip to makes calls in and out? |
| 00:59.30 | duplex- | shmaltz: already did =) |
| 00:59.33 | Qwell[] | Splatty47: You could call them and ask what they'll support |
| 00:59.34 | ManxPower | Splatty47, you'll have to ask Digium. |
| 00:59.36 | shmaltz | duplex, and?????????/ |
| 00:59.44 | duplex- | shmaltz: Exact same errors =) |
| 01:00.01 | shmaltz | duplex, did you take a peek in the Makefile what it does want? |
| 01:00.03 | Splatty47 | 447 Extension phone system! |
| 01:00.23 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 01:00.25 | Splatty47 | Meridian -> VOIP. |
| 01:00.27 | duplex- | shmaltz: It looks like a typo error to me. make linux is one of the valid options. |
| 01:00.28 | Splatty47 | will not be easy. |
| 01:00.31 | ManxPower | OH GOD NO! |
| 01:00.50 | shmaltz | ManxPower, what? |
| 01:00.51 | ManxPower | Splatty47, we do LIMITED interface between our nortel and the asterisk box. |
| 01:01.13 | Splatty47 | ManxPower: My intention is to completely replace the Meridian. |
| 01:01.19 | Splatty47 | within 2 months |
| 01:01.20 | duplex- | shmaltz: It's okay, don't worry about it. Thanks though. |
| 01:01.38 | Splatty47 | phone systems working side by side, not together, then I'll switch off the meridian |
| 01:01.38 | ManxPower | PRI cards for those things are EXPESNIVE and then you have to buy the software to support PRI, then if you want to actually REALLY interface between them you need to buy the PRI TANDEM software or something like that. |
| 01:01.45 | Splatty47 | I just bought 400 Snom 360 phones |
| 01:01.52 | ManxPower | prolly end up costing about $10,000 just for the hardware and software |
| 01:02.16 | Splatty47 | heh |
| 01:02.17 | ManxPower | Splatty47, I hope you bought ONE SNOM first to test with. |
| 01:02.31 | shmaltz | Splatty47, why did you buy 400 phones when you got no clue how to configure them? |
| 01:03.08 | Splatty47 | ManxPower: Not exactly. Sister company has just implemented Snom 360s with asterisk and works great for them - bigger office than us. |
| 01:03.23 | shmaltz | Splatty47, where you located? |
| 01:03.46 | Splatty47 | shmaltz: London |
| 01:03.55 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 01:03.58 | Splatty47 | sister co in NY, NY and also Tel-Aviv, IL |
| 01:04.21 | shmaltz | Splatty47, what do you guys do? |
| 01:04.36 | shmaltz | Splatty47, I would fly to London to do it for you |
| 01:04.45 | Splatty47 | I figured I run 200 servers here , and 400 workstations - on *nix and M$ platforms |
| 01:04.52 | shmaltz | I'm going to Israel anyhow in May, I might as well help you there as well |
| 01:04.55 | Splatty47 | shmaltz: Medical R&D |
| 01:04.56 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 01:05.00 | ManxPower | Splatty47, how many IT staff? |
| 01:05.04 | *** join/#asterisk file (n=jcolp@mctnnbsa24w-142167058031.pppoe-dynamic.nb.aliant.net) |
| 01:05.10 | Splatty47 | Manx: 15 |
| 01:05.19 | *** join/#asterisk sjobeck (n=sjobeck@london.sjobeck.com) |
| 01:05.35 | ManxPower | My largest customer has about 400 people, 18 locations and has 2 IT people and 1 consultant |
| 01:05.38 | Abydos313 | less |
| 01:05.40 | shmaltz | Splatty47, you interested in this offer? |
| 01:05.52 | Splatty47 | shmaltz: possibly. |
| 01:06.01 | shmaltz | ManxPower, I'm betting it's not a public company |
| 01:06.17 | ManxPower | shmaltz, no, if it was all the directors would be in jail. |
| 01:06.23 | shmaltz | lol |
| 01:06.27 | Splatty47 | shmaltz: Neither are we - although that I might change soon. |
| 01:06.41 | Shaun222 | what is zaptel for? |
| 01:06.50 | ManxPower | Shaun222, PSTN interface |
| 01:06.54 | shmaltz | ManxPower, it's usualy the big companies that are pbulic that spend money like that it's not theirs |
| 01:06.57 | Splatty47 | whoops minus the I |
| 01:06.57 | Shaun222 | PSTN? |
| 01:07.10 | shmaltz | Shaun222, to tie your shoes |
| 01:07.24 | Splatty47 | I'd be happy getting 2 extensions ringing each other! |
| 01:07.24 | ManxPower | ~thebook |
| 01:07.26 | jbot | well, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 01:07.27 | Strom_C | Shaun222, public switched telephone network |
| 01:07.32 | ManxPower | Shaun222, read that and then come back |
| 01:07.52 | Splatty47 | ManxPower read it all, and the getting started guide. |
| 01:07.54 | lzhang | hey guys, what do I have to do to get buddies working on a 601 + expansion module |
| 01:07.57 | Splatty47 | still can't get the thing to work. |
| 01:08.02 | shmaltz | Shaun222, next time you try to come back to this channel jbot will first give you a test on thebook |
| 01:08.10 | ManxPower | lzhang, polycom only supports having 8 buddies |
| 01:08.14 | shmaltz | otherwise you wont be able to join this channle |
| 01:08.17 | Qwell[] | ManxPower: 7? |
| 01:08.27 | ManxPower | you knew that, right, after your research? |
| 01:08.30 | terrapen | can anybody recommend a non-sucky competitor to voipsupply? |
| 01:08.32 | shmaltz | izhang, it's on the user list at leaset twice a day |
| 01:08.45 | shmaltz | ManxPower, 7 not 8 |
| 01:08.56 | ManxPower | shmaltz, ah, I sit corrected. |
| 01:09.06 | shmaltz | terrapen, whats wrong with voipsupply? |
| 01:09.22 | ManxPower | Qwell, subscribe to the asterisk-users list. I'll give you my procmail filter if you want. It sends a lot of the crap to .Trash |
| 01:09.29 | Splatty47 | any chance someone could send me a sample sip.conf and extensions.conf ? |
| 01:09.29 | Qwell[] | no thanks |
| 01:09.41 | shmaltz | ManxPower, do my messages got to trash? |
| 01:09.51 | shmaltz | ManxPower, what about Dougs? |
| 01:10.00 | lzhang | is there any way to put the speed dial directory on the sidecar? is that on the lists often as well |
| 01:10.00 | Qwell[] | Doug's go into Humour/ |
| 01:10.20 | shmaltz | Splatty47, yes its in /usr/src/asterisk/configs/ |
| 01:10.21 | ManxPower | shmaltz, well if they have words like these in the subject: gpl license g729 g723.1 sql |
| 01:10.49 | shmaltz | lzhang, you should be able to apply what's on the list aobut the buddy to that as well, jsut RTFM |
| 01:11.13 | shmaltz | ManxPower, I thought that it went by name :) |
| 01:11.28 | shmaltz | Brian said he trahes Dougs thats why I asked |
| 01:11.33 | terrapen | shmaltz, i need a competing bid |
| 01:11.35 | shmaltz | Brian from astlinux |
| 01:11.43 | terrapen | i never get bids from just one vendor |
| 01:11.44 | lzhang | shmaltz: hmm can you point me to that info? do I just need to search the lists? |
| 01:11.57 | shmaltz | terrrapen, for what? |
| 01:12.07 | terrapen | some phones, plus a redfone fonebridge |
| 01:12.19 | ManxPower | shmaltz, the complete list: http://pastebin.ca/47230 |
| 01:12.31 | shmaltz | lzhang, search google like this: |
| 01:12.33 | shmaltz | polycom hint buddy site:lists.digium.com |
| 01:12.57 | shmaltz | terrrapen, you mean for voipsupply? |
| 01:13.02 | shmaltz | how many of those? terrapen? |
| 01:13.04 | Qwell[] | I should start giving my services away for free, for people in certain geographical locations |
| 01:13.22 | shmaltz | Qwell, does that include north eastern US? |
| 01:13.28 | Qwell[] | "You pay for airfare, hotel, and food, and sure, I'm come to Hawaii to install your * box." |
| 01:13.52 | terrapen | to start, i need one (1) of four different phones, to test. once i know which model i like, i will be getting about 40 for the initial test and when that goes well, i'm going to buy an additional 270 phones |
| 01:13.59 | Qwell[] | shmaltz: Not unless it's like NY or something :p |
| 01:14.00 | terrapen | plus a few redfone fonebridges |
| 01:14.07 | ManxPower | As you can see, my procmailrc filters out a lot of the -users crap |
| 01:14.19 | fuzzbawl | I'm trying to have Asterisk dial a 1 before a 10 digit number if someone forgot (I have quite a few users who dial starting with the area code or 800 without hitting 1 first) |
| 01:14.28 | fuzzbawl | training the users is not working :/ |
| 01:14.34 | shmaltz | terrapen, then you might want to try direct |
| 01:14.56 | terrapen | shmaltz, too much hassle. certainly there must be another reliable VoIP vendor out there besides voipsupply |
| 01:14.57 | ManxPower | fuzzbawl, exten => _NXXNXXXXXX,1,Playback(YOUMUSTDIALONE) |
| 01:14.58 | shmaltz | Qwell, to NY you would work for free if I pay lodging food, fare, and some pocket money? |
| 01:15.11 | shmaltz | ManxPower, I dont blame you |
| 01:15.14 | Qwell[] | shmaltz: pretty much :P |
| 01:15.19 | ManxPower | terrapen, I've used VoIPSupply and Voxzilla |
| 01:15.22 | fuzzbawl | 1+NXXXXXXXXX isn't working as an outbound trunk rule |
| 01:15.35 | shmaltz | terrapen, not that I know of |
| 01:16.01 | ManxPower | and of course my own vendor |
| 01:16.05 | shmaltz | Qwell, thats so generous, I have never seen anybody do that before |
| 01:16.17 | Qwell[] | shmaltz: I don't get to travel much :P |
| 01:16.24 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 01:16.25 | shmaltz | lol |
| 01:16.33 | terrapen | no redfone at voxilla...but a good start nonetheless |
| 01:16.45 | Qwell[] | the stipulation of course, is that I bring my wife, and her airfare and such are included |
| 01:17.01 | shmaltz | ManxPower, I actualy like that list for trash, those are usualy the ones that go to trash anyhow |
| 01:17.12 | ManxPower | fuzzbawl, it sounds like your users are even more stupid than my users and I did know that was possible. |
| 01:17.30 | Qwell[] | So, if anybody in say...Jamaica, or Paris would like to take me up on my offer... |
| 01:17.31 | shmaltz | Qwell, so you married, Mazel Tov |
| 01:17.37 | shmaltz | lol |
| 01:17.51 | shmaltz | Qwell, any kids? |
| 01:17.53 | [av]bani | http://bani.anime.net/linux_is_bad.gif |
| 01:17.54 | *** join/#asterisk rj66 (n=rjrae@67.95.13.46) |
| 01:18.02 | Qwell[] | shmaltz: yeah, a newblet |
| 01:18.07 | ManxPower | Qwell[], I'm finally starting to think about accepting more non-local contracting jobs |
| 01:18.08 | fuzzbawl | ManxPower, I had to get emailing of voicemails working since they didn't understand why they had to "dial into some stupid system to check voicemail. Our OLD phone system always emailed" |
| 01:18.13 | Qwell[] | ManxPower: fun |
| 01:18.27 | Qwell[] | fuzzbawl: attach=yes |
| 01:18.28 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:18.28 | ManxPower | Qwell[], Flying sucks. |
| 01:18.29 | shmaltz | Qwell, just one? |
| 01:18.33 | Qwell[] | shmaltz: indeed |
| 01:18.38 | Qwell[] | free on planes though :P |
| 01:18.38 | ManxPower | Must less so in Europe, as I experienced |
| 01:18.54 | Qwell[] | ManxPower: What ever happened to moving there anyhow? |
| 01:19.15 | ManxPower | Qwell[], Katrina |
| 01:19.17 | fuzzbawl | Qwell[]: yea, got that working. But the users keep comparing Asterisk to our older Interactive Intelligence box. The next step is to find a nice GUI for asterisk so they can see who is logged in/status |
| 01:19.20 | Qwell[] | ahh... |
| 01:19.30 | Qwell[] | fuzzbawl: write one? |
| 01:19.35 | shmaltz | Qwell, how long you married? |
| 01:19.37 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:19.44 | shmaltz | Qwell, you in Austin right? |
| 01:19.47 | Qwell[] | ManxPower: I can see how that could put a damper on things |
| 01:19.53 | Qwell[] | shmaltz: 2 yrs, southern CA |
| 01:20.05 | ManxPower | Qwell, Where I live now is rather nice. |
| 01:20.07 | shmaltz | damn dnsstuff, gave me the wrong city for you |
| 01:20.21 | shmaltz | Qwell, any plans on another kid? |
| 01:20.22 | ManxPower | Will be nicer when it warms up a little. |
| 01:20.25 | fuzzbawl | Qwell[]: heh. I'm no coder |
| 01:20.27 | Qwell[] | shmaltz: "unaffiliated/qwell" actually resolves? |
| 01:20.32 | shmaltz | how old is your child? |
| 01:20.33 | Qwell[] | fuzzbawl: See my above offer :P |
| 01:20.38 | Qwell[] | shmaltz: a year |
| 01:20.44 | ManxPower | shmaltz, you would hope he learned after the first one. |
| 01:20.48 | shmaltz | ===Qwell[] <i=north@unaffiliated/qwell> North Antara |
| 01:20.50 | shmaltz | ===Qwell[]: member of #asterisk |
| 01:20.51 | shmaltz | ===Qwell[]: attached to irc.freenode.net http://freenode.net/ |
| 01:20.52 | shmaltz | ===Qwell[] is identified to services |
| 01:20.54 | shmaltz | ---End of WHOIS information for qwell[] |
| 01:20.55 | file | Daddy Qwell! |
| 01:21.00 | Qwell[] | file: indeed |
| 01:21.07 | ManxPower | North Antara? |
| 01:21.17 | Qwell[] | ManxPower: yeah...something like that |
| 01:21.21 | file | I know Qwell's true name, so I have power over him |
| 01:21.26 | shmaltz | qwell, you right I used the wrong IP |
| 01:21.29 | Qwell[] | file is under NDA though :D |
| 01:21.29 | fuzzbawl | Qwell[]: would you really want to visit South Bend, IN? :P |
| 01:21.38 | Qwell[] | fuzzbawl: yeah...not so much :P |
| 01:21.47 | shmaltz | Qwell, same here, my baby is 12 months old |
| 01:21.58 | shmaltz | QWell, but my wife is due in July for the second one |
| 01:22.13 | shmaltz | lol |
| 01:22.31 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 01:22.32 | Qwell[] | ManxPower: You should adopt. :p |
| 01:22.44 | ManxPower | Qwell[], no need to get vulgar |
| 01:22.51 | shmaltz | Qwell, I like that tip of using an RDNS that doesn resolv |
| 01:23.00 | ManxPower | children are little evil disease carrying monsters |
| 01:23.20 | ManxPower | They can't even really be thought of as "human" until they have been socialized. |
| 01:23.44 | ManxPower | shmaltz, I do not claim to be any different when I was a child. |
| 01:23.54 | shmaltz | ManxPower, to your parents you are still that nohuman child |
| 01:24.26 | ManxPower | shmaltz, it took a while, but I became (more or less) socialized. |
| 01:24.38 | RoyK | ~seen zoa |
| 01:24.49 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 12d 8h 41m 32s ago, saying: 'it looks kinda suspicious :p'. |
| 01:24.49 | shmaltz | ManxPower, do you just look at me like I look at those breeding dogs, cats etc.? |
| 01:25.12 | ManxPower | shmaltz, only if you have a eugenics program. |
| 01:25.39 | ManxPower | Otherwise I just assume you could not control your preprogrammed biological need to continue the species. |
| 01:26.32 | ManxPower | don't feel bad about it, it's a common enough problem. 8-) |
| 01:26.40 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net) |
| 01:26.41 | shmaltz | lol |
| 01:26.48 | Qwell[] | affects some 90% of the population :P |
| 01:26.50 | shmaltz | ManxPower, you are toasted |
| 01:27.48 | ManxPower | shmaltz, much like communism, it can sound like a good idea, but can never work in practice. |
| 01:28.30 | *** join/#asterisk forao (n=dfasdfs@pool-141-150-77-17.mad.east.verizon.net) |
| 01:28.46 | ManxPower | RAH had a good solution to it. Pay people to have socially desireable traits to have children. |
| 01:28.55 | shmaltz | ManxPower, I think its sadistic |
| 01:29.13 | shmaltz | who is/was RAH? |
| 01:29.15 | ManxPower | people THAT have desirable |
| 01:29.16 | shmaltz | Hitler? |
| 01:29.25 | *** part/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 01:29.25 | ManxPower | Robert A Heinelin |
| 01:29.27 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 01:29.55 | lzhang | hmm I can't seem to get the expansion module to show anything at all right now, is there some sort of default behavior this thing is supposed to do? |
| 01:29.56 | ManxPower | Si Fi writer |
| 01:30.06 | ManxPower | shmaltz, anything that forces anyone to do anything is horrible. |
| 01:30.13 | lzhang | right now the display is blank and all the lights are flickering red/green |
| 01:30.31 | RoyK | ~seen zoa |
| 01:30.35 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 12d 8h 47m 18s ago, saying: 'it looks kinda suspicious :p'. |
| 01:30.41 | shmaltz | ManxPower, does that mean if I force a gunman to put down the gun I'm horrible? |
| 01:31.04 | ManxPower | shmaltz, only if your life is not in danger |
| 01:31.15 | shmaltz | RoyK, why do you think that after 12+ days he will show up now? |
| 01:31.38 | shmaltz | ManxPower, why not when my life is in danger? |
| 01:32.02 | ManxPower | shmaltz, if your life is in danger, then you should be able to protect yourself. |
| 01:32.10 | shmaltz | ManxPower, is forcing a kid to go to school horrible? |
| 01:32.13 | RoyK | Shaun222: i only hope |
| 01:32.21 | shmaltz | lol |
| 01:32.26 | Qwell[] | anyone implies person |
| 01:32.26 | shmaltz | RoyK, I guess you meant me |
| 01:32.29 | Qwell[] | child != person |
| 01:32.30 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 01:32.31 | Qwell[] | right? :P |
| 01:32.36 | shmaltz | :( |
| 01:32.37 | ManxPower | shmaltz, children are not considered human until they turn 18 (at least in the USA) |
| 01:32.38 | shmaltz | I can cry |
| 01:32.39 | RoyK | Shaun222: i've paid the guy for writing a jitterbuffer that doesn't work |
| 01:32.48 | justinu | uh-oh |
| 01:32.57 | justinu | you have no other way to get ahold of him than irc? |
| 01:33.00 | shmaltz | RoyK, how did you pay him? |
| 01:33.07 | shmaltz | dispute it |
| 01:33.12 | RoyK | how? |
| 01:33.16 | RoyK | money?? |
| 01:33.22 | shmaltz | how did you pay? |
| 01:33.36 | RoyK | does it matter? |
| 01:34.11 | ManxPower | RoyK, I think he means "you can file a dispute with your credit card company" |
| 01:34.27 | *** join/#asterisk juice (i=1000@209.33.109.213) |
| 01:34.37 | ManxPower | shmaltz, look it up in the USA legal code. |
| 01:34.49 | shmaltz | http://en.wikipedia.org/wiki/Human |
| 01:34.57 | shmaltz | there is a dispute on the definition |
| 01:35.02 | justinu | if they're not human until they're 18, how is killing a child murder? |
| 01:35.10 | ManxPower | The right to life, liberty, and the pursuit of happiness. Well, children do not have liberty. |
| 01:35.13 | shmaltz | but according the current one, children would be part of it |
| 01:35.23 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com) |
| 01:35.38 | ManxPower | justinu, not sure. |
| 01:35.41 | shmaltz | ManxPower, thats why I would much rather prefer a monorchy |
| 01:35.46 | RoyK | ManxPower: the deal was pay one third up front, get a patch, then pay the second, and pay the third when it's stable. |
| 01:36.26 | RoyK | ManxPower: they refuse to do any more about it now, after the second payment. just bullocs |
| 01:37.16 | RoyK | and the amount being EUR 2500 devided by three, it's quite annoying |
| 01:37.21 | ManxPower | One does have to ask what is so different between a puppy and a baby. |
| 01:38.01 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 01:38.11 | RoyK | we've paid for a product. not a bug |
| 01:38.15 | MikeJ[Laptop] | ManxPower, puppies are furrier, and more capable |
| 01:38.17 | shmaltz | ManxPower, the ability to be happy to follow a leaders will is something we all lack and existed under certain monarchys |
| 01:38.36 | ManxPower | MikeJ[Laptop], Well that would be my answer, but I think most people dill disagree. |
| 01:38.56 | MikeJ[Laptop] | babies are very cute and all.. |
| 01:39.05 | MikeJ[Laptop] | but they are basically eating pooping machines |
| 01:39.07 | shmaltz | ManxPower, the ability to mature is the difference |
| 01:39.31 | MikeJ[Laptop] | shmaltz, yeah.. but as babies themselves.. they are basically just poop machines |
| 01:39.34 | MikeJ[Laptop] | cute |
| 01:39.39 | MikeJ[Laptop] | but poop machines |
| 01:39.45 | ManxPower | shmaltz, perhaps you mean the ability to become selfaware. |
| 01:39.47 | MikeJ[Laptop] | in the long run, they differentiate |
| 01:39.48 | RoyK | ManxPower: do you mean it's right to take pay for something that doesn't work? |
| 01:39.54 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 01:39.56 | shmaltz | MikeJ, nope, they are the parents life and joy |
| 01:40.12 | ManxPower | RoyK, none of my comments other than the oune about credi cards had anything to do with your situation. |
| 01:40.33 | ManxPower | shmaltz, and my cats are my pride and joy. |
| 01:41.00 | ManxPower | MikeJ[Laptop], basically it comes down to biology. Humans are preprogrammed by their DNA to love babies and to want to have them. |
| 01:41.02 | shmaltz | ManxPower, but not your life |
| 01:41.37 | ManxPower | shmaltz, of course not. technology is my live, with a healthy dose of getting laid in there too. |
| 01:42.00 | shmaltz | lol |
| 01:42.16 | shmaltz | gtg, My wife and baby are waiting for me |
| 01:42.44 | fuzzbawl | ok, something weird is going on. all of a sudden I have no outbound trunk |
| 01:43.15 | *** part/#asterisk terrapen (n=cjs@166.70.183.108) |
| 01:43.40 | MikeJ[Laptop] | hehe |
| 01:44.02 | fuzzbawl | omg n/m, I'm a moron, forgot to hit 9 first |
| 01:44.18 | Strom_C | pebcap |
| 01:44.30 | Strom_C | problem exists between chair and phone |
| 01:44.47 | Qwell[] | Strom_C: in the rj12 handset wire? |
| 01:45.02 | Strom_C | handset wire isn't rj12 |
| 01:45.11 | Qwell[] | can be :p |
| 01:45.22 | Strom_C | rj12 is the mounting cord IIRC |
| 01:45.26 | Qwell[] | Which one am I thinking? |
| 01:46.19 | justinu | rj12 is rj11 w/ 3 pairs |
| 01:46.36 | Qwell[] | I know what rj12 is now...what's the smaller one? |
| 01:46.43 | justinu | rj11 |
| 01:46.45 | Strom_C | no |
| 01:46.47 | Qwell[] | no |
| 01:46.52 | Strom_C | RJ11 is 6p2c |
| 01:46.57 | Strom_C | RJ9 is 4p4c IIRC |
| 01:47.03 | Qwell[] | 9, that's the one |
| 01:47.06 | justinu | oh |
| 01:47.30 | Strom_C | RJ12 is 6p4c |
| 01:47.35 | Strom_C | RJ14 is 6p6c |
| 01:47.55 | Qwell[] | RJ13 is unlucky |
| 01:49.10 | Qwell[] | bbl |
| 01:52.08 | *** join/#asterisk t1n (n=tin-st@213-152-33-178.dsl.eclipse.net.uk) |
| 01:52.11 | _Simon | I was wondering if someone could help me with my iax provider, I'm getting an error message, but I can't find out what I'm doing wrong. I think its in my extensions.conf |
| 01:52.38 | Strom_C | _Simon, what's the error message? |
| 01:53.49 | _Simon | when I call the number I'm trying to dial (PSTN) I get a voice message saying there was a problem |
| 01:53.52 | _Simon | but in my CLI theres no issue |
| 01:53.59 | _Simon | so I must not be passing something properly to the provider |
| 01:54.05 | Strom_C | what does the voice message say? |
| 01:54.18 | _Simon | contact tech support |
| 01:54.19 | _Simon | lol |
| 01:54.31 | Strom_C | I want the exact text of the message |
| 01:54.44 | _Simon | thats what the voice says |
| 01:54.47 | _Simon | "contact tech support" |
| 01:54.53 | _Simon | theres no CLI error |
| 01:54.57 | Strom_C | ok |
| 01:55.05 | _Simon | I'm using globotech via IAX2 |
| 01:55.06 | Strom_C | what are you passing to the itsp? |
| 01:56.22 | _Simon | my dialplan in extensions.conf says |
| 01:56.24 | _Simon | exten => 6132222222,1,Dial(IAX2/Globotech:/${EXTEN}) |
| 01:56.31 | _Simon | (the 222etc being a real number) |
| 01:56.40 | Strom_C | why in gods name are you not matching wildcards? |
| 01:56.52 | _Simon | I'm just trying to test 1 line first |
| 01:56.58 | _Simon | before I start playing with wildcards |
| 01:57.08 | Strom_C | ...yeah |
| 01:57.10 | Strom_C | anyway |
| 01:57.37 | Strom_C | point me at globotech's config pages and then pastebin the relevant section of iax.conf |
| 01:57.39 | _Simon | sorry ignore the : thats in there |
| 01:58.10 | _Simon | ok well I don't wanna paste all my globotech info lol |
| 01:58.10 | _Simon | but basically in iax.conf I have: |
| 01:58.15 | Strom_C | PASTEBIN |
| 01:58.20 | Strom_C | ~pastebin |
| 01:58.21 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 02:00.06 | *** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 02:00.34 | PMantis | I just upgraded to 1.2.6 on a production server... and it won't start. |
| 02:00.35 | PMantis | <PROTECTED> |
| 02:00.35 | PMantis | <PROTECTED> |
| 02:00.44 | PMantis | drops back to prompt |
| 02:01.16 | MikeJ[Laptop] | PMantis, it's choking on somthing with config |
| 02:01.23 | MikeJ[Laptop] | start with a bunch of v's |
| 02:01.27 | MikeJ[Laptop] | what does it say then? |
| 02:01.40 | PMantis | Here's how I'm starting it: asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 02:01.47 | PMantis | That enough? |
| 02:01.48 | PMantis | :-) |
| 02:01.51 | _Simon | haha |
| 02:02.08 | PMantis | The 2 lines I pasted is how it ends.. |
| 02:02.09 | *** join/#asterisk talljon84 (n=talljon8@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 02:02.21 | PMantis | <PROTECTED> |
| 02:02.30 | sjobeck | pmantis: dont know, but, do you need to upgrade asterisk-addons ? |
| 02:02.46 | PMantis | sjobeck, I'm at 1.2.2... which is latest, right? |
| 02:03.09 | sjobeck | y |
| 02:03.12 | PMantis | AFAIK, I have the latest version of all |
| 02:03.49 | PMantis | A few days ago, I was trying to get cdr working... but never started |
| 02:04.20 | talljon84 | After a yum update on a fresh CentOS install, zaptel no longer starts. I checked the udev.rules and udev.permissions and both appear to contain the needed lines but when I look a /dev, no zap exists. lspci shows the TDM400. How do I get zap to repopulate? |
| 02:04.56 | PMantis | MikeJ[Laptop], Anything to add, or would you like me to past more? |
| 02:05.15 | *** join/#asterisk dextro (n=dextro@cpe-70-116-10-201.austin.res.rr.com) |
| 02:05.27 | sjobeck | i'll got to hawaii for all that minus the food :) |
| 02:05.51 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 02:05.54 | SpaceBass | evening |
| 02:05.59 | SpaceBass | Qwell you around? |
| 02:06.29 | sjobeck | SpaceBass: you on Verizon fiber ? how is it? |
| 02:06.46 | SpaceBass | sjobeck its ok |
| 02:06.57 | SpaceBass | 45mbs down / 15 up (yeah right) |
| 02:07.04 | SpaceBass | but latency seems to be an issue |
| 02:07.05 | sjobeck | SpaceBass: not great!? |
| 02:07.27 | sjobeck | SpaceBass: ahhhh. Dont they give you TV, 12 voice channels, with all the bw ? |
| 02:07.43 | SpaceBass | sjobeck honestly, the up pipe is notically better than my old comcast 8mbs/752k...but not always sure the down is |
| 02:07.56 | SpaceBass | sjobeck i just have the internet |
| 02:08.27 | SpaceBass | for $45 for the residental, it would be a good deal, but they block servers (ports 80, 25, etc)...i have the business class for like $300/month |
| 02:08.41 | SpaceBass | and I have problems with broadvoice breaking up on me |
| 02:08.58 | sjobeck | SpaceBass: what your QoS device ? |
| 02:09.07 | SpaceBass | not sure if its BV or my connection, but I have noticed that if I quit bittorrent or my VPN session, it gets a lot better |
| 02:10.12 | SpaceBass | sjobeck good question...and a source of fustration... i use IPcop which uses "traffic shaping" and they (the developers) claim its for prioritization of packets...but others tell me thats not how it works |
| 02:12.06 | SpaceBass | probably more info than you wanted about FiOS and my setup :) |
| 02:13.16 | sjobeck | SpaceBass: nah, good to know. I wonder what pixiedust you need to make IPcop do its thing (or put in Packeteer for a gazillion dollars will do it) |
| 02:13.22 | SpaceBass | PMantis what did you say? was it about this week's sopranos...if so I dont want to hear it |
| 02:13.53 | SpaceBass | sjobeck there was a time I could have gotten a packeteer at cost...now thats EXACTLY what my home network needs! |
| 02:14.40 | PMantis | SpaceBass, Heh, no... was aking for help. got an initial response, but nothing after that. I care little about tv shows. lol |
| 02:15.11 | SpaceBass | ok...just don't spoil this week's sopranos...someone at work tried to tell me what happened and I havent been home to see anything on my tivo in a while |
| 02:15.12 | PMantis | It's ok though.. I commented out the loading of the cdr_addons_mysql.ko... working now |
| 02:16.57 | SpaceBass | sjobeck did your isp set up your reverse DNS ? |
| 02:16.58 | Qwell | SpaceBass: nope |
| 02:17.05 | SpaceBass | hummmmmm |
| 02:18.18 | SpaceBass | Qwell yesterday you were telling me what you thought of the linksys wip-330 and i had to run |
| 02:18.39 | Qwell | SpaceBass: I only had it in my hand for all of 90 seconds |
| 02:18.48 | SpaceBass | ahhh |
| 02:19.01 | SpaceBass | just wish i could actually find a place to buy one...i might breakdown and do it |
| 02:21.34 | SpaceBass | next question...anyone use telasip? |
| 02:22.40 | PMantis | SpaceBass, I tried to sign up... |
| 02:22.54 | PMantis | Well, let me start back b4 that... |
| 02:23.01 | SpaceBass | not sounding good |
| 02:23.02 | PMantis | asked questions.. no response. |
| 02:23.22 | PMantis | signed up, then replied, saying there... I signed up, NOW answer |
| 02:23.34 | PMantis | 2 days later, I got a response that they dont' offer service in my area |
| 02:23.42 | PMantis | Seems like a 1 man show |
| 02:23.56 | SpaceBass | ah |
| 02:24.02 | PMantis | almost 2 weeks later, I got a refund |
| 02:24.03 | SpaceBass | a la voxbone |
| 02:24.07 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-150-58-20.msy.bellsouth.net) |
| 02:24.27 | SpaceBass | although I had good success with voxbone...but seemed one man |
| 02:25.45 | *** join/#asterisk alexns (n=ibtek04@static-acs-24-154-114-15.zoominternet.net) |
| 02:26.10 | alexns | anyone configuring snom 360 via tftp ? |
| 02:27.15 | alexns | .. seeing some strange results |
| 02:27.41 | SpaceBass | PMantis yeah...im thinking the same thing |
| 02:27.58 | PMantis | SpaceBass, problems? |
| 02:27.58 | SpaceBass | just ported my POTS number of 7 years to them...hopeing it wasnt a mistake |
| 02:28.09 | SpaceBass | but I couldn't live with the zaptel problems i was having |
| 02:28.13 | alexns | sometimes it is |
| 02:28.21 | SpaceBass | PMantis breakup and dropouts sometimes |
| 02:28.27 | PMantis | SpaceBass, Heh... they have a good deal |
| 02:28.30 | alexns | i am using teliax , phone usually works but somtimes drops calls |
| 02:28.43 | SpaceBass | BV has GREAT prices |
| 02:28.44 | alexns | but at least you get the calls :) |
| 02:28.53 | SpaceBass | and works fairly well...most of the time |
| 02:29.01 | PMantis | SpaceBass, but closest DID to me is LD to everyone I know... we use it for OB only |
| 02:29.10 | SpaceBass | and I can send faxes via BV too |
| 02:29.34 | SpaceBass | PMantis BV didn't have local DIDs but they could port mine....might be true for other providers |
| 02:29.47 | alexns | ? what is the state of T.38 in asterisk |
| 02:30.03 | PMantis | SpaceBass, But you're right.. my wife could be on the phone talking, and I suddenly hear, "Hello?.... Hello???. yooohooo? can you hear me??.... STEEEEEEEEEEEVVVVVVVVVVVVEEEE!!! (that' my name) |
| 02:30.08 | SpaceBass | PMantis until my port was complete, I was using BV only for OB LD and nufone for locals (since I could set caller ID) |
| 02:30.37 | PMantis | nufone? isn't that metered? |
| 02:30.39 | SpaceBass | PMantis my wife HATES our asterisk setup...she almost refuses to answer the phone...i think that was mostly b/c of our zap problems |
| 02:30.48 | SpaceBass | PMantis yeah...which I don't care for (metered) |
| 02:31.04 | SpaceBass | but the whole setcallerid() is GREAT for pranks |
| 02:31.12 | PMantis | Our inbound works 95% |
| 02:31.17 | PMantis | SpaceBass, LOL, Yeah! |
| 02:31.27 | SpaceBass | i'd say ours is 90% |
| 02:31.33 | PMantis | SpaceBass, I called a friend once as "Emergency, 911" lol |
| 02:31.41 | SpaceBass | wow! that takes guts! |
| 02:31.53 | SpaceBass | i usually call people at work from their houses....then just breath and hang up |
| 02:31.55 | SpaceBass | :) |
| 02:32.03 | PMantis | Then I said, "Who called from this number??" |
| 02:32.47 | SpaceBass | i do worry about 911 since I travel a lot... so I took an old cell phone with not service and put it on a charger near the nightstand |
| 02:33.02 | PMantis | LOL, what we do here. |
| 02:33.03 | SpaceBass | at least my wife could use that if it got bad |
| 02:33.10 | PMantis | I wrote in marker, "911 phone" |
| 02:33.16 | mogorman | lol |
| 02:33.21 | mogorman | i have that at my apt |
| 02:33.24 | SpaceBass | and I have 911 mapped to the non emergency number...but haven't tested that yet |
| 02:33.26 | mogorman | its plugged into the land line |
| 02:33.28 | mogorman | thats turned off |
| 02:33.32 | mogorman | but the copper is live |
| 02:33.37 | mogorman | and will allow me to dial 911 |
| 02:33.47 | SpaceBass | mogorman does coper have to dial 911 no matter what? |
| 02:33.57 | mogorman | if you get dial tone |
| 02:33.59 | mogorman | it has to |
| 02:34.09 | mogorman | they often disconnect copper in areas where its short |
| 02:34.17 | mogorman | but where i live ive never found a dead jack |
| 02:34.21 | mogorman | its like cell service |
| 02:34.27 | mogorman | if a tower gets a 911 call |
| 02:34.29 | mogorman | it has to route it |
| 02:34.42 | mogorman | thus dead cell phones are great 911 device |
| 02:34.42 | mogorman | s |
| 02:34.44 | PMantis | Then hope the pickup. |
| 02:34.49 | SpaceBass | hummmm I'll keep that in mind when I get rid of my POTS |
| 02:34.51 | PMantis | s/the/they/ |
| 02:35.01 | PMantis | WOW! |
| 02:35.08 | PMantis | cool bot feature |
| 02:35.09 | SpaceBass | oooh who put re into jbot? |
| 02:35.10 | Qwell | Just test for dialtone every 5 minutes, on a cron |
| 02:35.21 | mogorman | it will always pickup |
| 02:35.31 | mogorman | they arent gonna disco copper in an apt complex |
| 02:35.36 | mogorman | just to redo it next month |
| 02:35.43 | Qwell | mogorman: yeah |
| 02:35.43 | mogorman | same thing with electric water and cable |
| 02:35.48 | SpaceBass | Qwell in a crazy way that reminded me of a question.... is there a way to do SMS with asterisk? |
| 02:35.51 | Qwell | electric, they do shut off here... |
| 02:36.03 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 02:36.12 | mogorman | yes you can do sms a couple of ways |
| 02:36.14 | SpaceBass | electeric will shut you off and not think twice! |
| 02:36.15 | mogorman | they dont here |
| 02:36.16 | Qwell | mogorman: How else are they gonna justify the $90 setup fee? |
| 02:36.23 | mogorman | they do it anyways |
| 02:36.25 | Qwell | heh |
| 02:36.28 | mogorman | its just a change of billing fee |
| 02:36.30 | SpaceBass | i need to look into SMS... would love to do that |
| 02:36.33 | mogorman | i got it for all my services |
| 02:36.35 | mogorman | was lame as hell |
| 02:36.44 | Qwell | I actually had to pay a fee, when they put the electricty to the wrong apt when I first moved in |
| 02:36.54 | Qwell | "uhh...yeah...dude...I live here, not there." |
| 02:37.18 | Qwell | "Ahh, we're sorry about that sir. We're going to have to charge you a $15 fee for that." |
| 02:37.29 | Nugget | that's nuts |
| 02:37.33 | SpaceBass | your shitting me? you didn't pay it did you?!?! |
| 02:37.37 | Qwell | well, it was kinda my fault, heh |
| 02:37.42 | Qwell | well...no, not my fault either |
| 02:37.51 | Qwell | carpet installer fucks fault |
| 02:38.07 | mogorman | heh |
| 02:38.12 | mogorman | suck |
| 02:38.13 | Qwell | I told them like a week early that I was gonna move in on xyz date, and to have it turned on before I get there |
| 02:38.26 | Qwell | carpet dipshits weren't done, so I had to get a diff apt |
| 02:38.31 | SpaceBass | ahhhhhhhhhh |
| 02:38.35 | Qwell | cable company was worse... |
| 02:38.39 | SpaceBass | sounds like carpetfucks' fault |
| 02:38.43 | mogorman | cable company pisses me off |
| 02:38.44 | Qwell | They had to cancel my appointment, and setup a new one |
| 02:38.46 | Qwell | 2 weeks later |
| 02:39.02 | mogorman | but those dumb people are giving me free extended cable for over a month now |
| 02:39.02 | SpaceBass | that's why if I can get everything over FiOS (and directv) I'll be happy...no more cable, no more telco |
| 02:39.36 | mogorman | fios? |
| 02:39.40 | Qwell | mogorman: verizon |
| 02:39.42 | Qwell | fiber |
| 02:39.44 | Qwell | ...telco |
| 02:39.46 | mogorman | ooh |
| 02:39.56 | SpaceBass | I had cable internet (but not TV) for a long time...that equalled free cable (non digital) which was nice...but already had directv |
| 02:40.01 | mogorman | i think somehow im gonna bounce internet from digium |
| 02:40.14 | mogorman | might cost me 500 to bounce connection around |
| 02:40.21 | mogorman | but fun project |
| 02:40.32 | SpaceBass | the fiber is nice |
| 02:41.42 | SpaceBass | Qwell i guess I have to have an IP phone that supports sms first huh? |
| 02:44.43 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 02:47.31 | SpaceBass | and it got quiet |
| 02:48.23 | PMantis | heh |
| 02:50.07 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 02:50.22 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 02:51.05 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 02:55.41 | *** join/#asterisk jmacz (n=jmacz@201.244.240.87) |
| 02:56.09 | *** join/#asterisk lyl (n=lyl@222.188.133.148) |
| 03:05.27 | St1ckm4n | does anyone know what the following stands for when you do a show queues or queue |
| 03:05.27 | St1ckm4n | <PROTECTED> |
| 03:05.37 | St1ckm4n | I'm assuming SL is our service level |
| 03:05.59 | St1ckm4n | is W: # of calls Waiting, C: total # of calls and A: abandoned? |
| 03:06.00 | Qwell | w is weight |
| 03:06.50 | Qwell | c is calls completed, a is calls abandoned |
| 03:07.17 | St1ckm4n | if I'm looking for # of calls in queue waiting is that the (0s holdtime) |
| 03:07.30 | St1ckm4n | I'm sorry not the # of calls in queue but how long calls in queue are waiting |
| 03:07.45 | Qwell | yeah, holdtime |
| 03:07.53 | St1ckm4n | ok, thanks |
| 03:13.15 | *** part/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 03:14.06 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-150-160.dsl.chcgil.sbcglobal.net) |
| 03:14.27 | Flauto | hi all |
| 03:14.31 | Flauto | 1.2.6 now |
| 03:14.34 | Flauto | what is new then |
| 03:14.39 | *** join/#asterisk forao (n=dfasdfs@pool-141-150-41-204.mad.east.verizon.net) |
| 03:14.48 | mogorman | its just more bug fixes |
| 03:14.55 | mogorman | like all 1.2.X releases |
| 03:15.20 | Qwell | and all 1.4.X releases will be |
| 03:15.20 | Flauto | i dont' think it will make a big difference to my usage |
| 03:16.33 | Flauto | i am running into a problem here |
| 03:17.18 | Flauto | i was not using any landline untill a couple of weeks ago, when i got a landline installed and connected to asterisk via x100p |
| 03:17.37 | SpaceBass | Flauto echo and jitter problems? |
| 03:17.41 | Flauto | sometimes when there is a call coming in |
| 03:17.59 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 03:18.06 | Flauto | it is not a problem, it goes into my ivr |
| 03:18.17 | Flauto | but the problem is |
| 03:18.27 | Flauto | when the call is hang up |
| 03:18.46 | Flauto | it shows as the call is dissconnected |
| 03:18.56 | Flauto | but a few seconds later |
| 03:19.26 | Flauto | it is showing that zap is coming in again without any real call |
| 03:19.36 | Flauto | it keeps looping in my system |
| 03:19.49 | SpaceBass | Flauto do you have call forwarding or anything? |
| 03:20.02 | Flauto | SpaceBass, echo is another problem |
| 03:20.15 | Flauto | especially the first few seconds of the call |
| 03:20.15 | SpaceBass | i had the problem when I tried to use my telco's call forward feature...it caused a "ghost" call on my * box |
| 03:20.31 | Flauto | then, it seems echo conceling kicks in |
| 03:20.39 | SpaceBass | Flauto thats odd...echo usually starts ok and gets worse |
| 03:20.59 | Flauto | i have a ivr menu to give choices to dial ext or 0 for operator |
| 03:21.07 | Flauto | really |
| 03:21.14 | Flauto | mine is not that bad indeed |
| 03:21.30 | Flauto | though, it is there at the begining of every call |
| 03:21.36 | Flauto | outbound is worse |
| 03:22.10 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 03:22.50 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 03:28.44 | St1ckm4n | has anyone enabled the RealTime Queues by adding the line queues=>mysql,asterisk,queue_table? |
| 03:29.16 | St1ckm4n | I need to be able to get some stats on our switch that only seem to be possible if I'm dumping realtime |
| 03:29.33 | St1ckm4n | calls in queue/hold time/agents available etc... |
| 03:38.09 | Flauto | http://pastebin.com/626305 |
| 03:38.21 | Flauto | would anyone help |
| 03:38.39 | Flauto | the problem only happens from zap |
| 03:39.19 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 03:39.56 | lzhang | is there a way to have a polycom 601 have line presence down the left side softkeys... and if you press the key it will dial out over that line? |
| 03:41.28 | Flauto | it keeps looping |
| 03:41.33 | Flauto | ghost call |
| 03:43.33 | *** part/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 03:46.46 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 03:47.32 | chris_ast | Hi People, Can anyone help me on Asterisk Directory? |
| 03:49.38 | chris_ast | In voicemail.conf what will happen if two users have same name? Asterisk is going to one user after other accoriding to customer_id but how do the dialer know who is the rite person? |
| 03:50.12 | Qwell | chris_ast: I guess they don't |
| 03:50.56 | chris_ast | Qwell: I have tested the case, it goes to one by one accoring to their cust id |
| 03:51.03 | *** join/#asterisk bmg505 (n=leon@dsl-146-6-207.telkomadsl.co.za) |
| 03:59.40 | *** join/#asterisk Mourning (i=sa@w167.z208037076.nyc-ny.dsl.cnc.net) |
| 04:00.12 | Mourning | Hi all |
| 04:01.00 | Mourning | Is it possible to get Call waiting to work on a PSTN line coming in to * via a Sipura 3000? |
| 04:01.36 | Flauto | mourning, how do you make a phone call to come in from pstn through spa 3000 |
| 04:02.16 | Mourning | Flauto: I'm not sure what you mean |
| 04:02.54 | Flauto | how do you let a call coming in to asterisk through spa 300 |
| 04:02.57 | Flauto | 0 |
| 04:04.10 | Mourning | Flauto: no idea what you mean |
| 04:04.13 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 04:04.24 | *** join/#asterisk pengyong (n=lala@222.188.133.148) |
| 04:05.20 | Flauto | i am asking you how to connect pstn line to asterisk via spa 3000 |
| 04:06.10 | Mourning | Flauto: are you asking me how to set it up? |
| 04:06.16 | Flauto | yes |
| 04:06.55 | Mourning | Flauto there is an automatic configuration utility on Voxilla |
| 04:07.27 | Mourning | http://voxilla.com/PNphpBB2-printview-t-1283-start-15.html |
| 04:07.57 | Mourning | Anyone have an answer to my question about Call Waiting? |
| 04:13.36 | *** join/#asterisk |||sLaSh||| (i=lockpad@203.215.100.96) |
| 04:13.39 | Mourning | Or if its possible to use Callwaiting on a PSTN line coming in to asterisk via a digium card |
| 04:14.03 | *** join/#asterisk fugitivo (n=user@201.255.183.220) |
| 04:14.37 | fugitivo | 1.2.6? |
| 04:14.49 | Mourning | released today |
| 04:15.43 | fugitivo | any important bug with 1.2.5? |
| 04:15.58 | VeNoMouS_ | all ure base are belong to me! |
| 04:16.17 | VeNoMouS_ | fugitivo : voicemail bugs |
| 04:16.20 | VeNoMouS_ | mixmonitor bug |
| 04:16.24 | VeNoMouS_ | zap bug |
| 04:16.30 | VeNoMouS_ | hrmm that were the majors i think |
| 04:17.04 | fugitivo | I'll check the changelog |
| 04:25.21 | *** part/#asterisk Mourning (i=sa@w167.z208037076.nyc-ny.dsl.cnc.net) |
| 04:26.54 | St1ckm4n | has anyone set up the dynamic realtime with Mysql? |
| 04:27.47 | |Vulture| | wonder what happens if Mysql goes down |
| 04:27.54 | |Vulture| | does it use cached... or are you fucked? |
| 04:28.07 | Qwell | |Vulture|: You're fucked |
| 04:28.08 | St1ckm4n | I think you're fucked from what i read |
| 04:28.35 | |Vulture| | see that sucks.. cause I would love to run a central mysql server and have all my servers connected |
| 04:28.42 | St1ckm4n | I need to come up with a solution to show calls in queue with hold time |
| 04:28.44 | |Vulture| | thats the only reason I stayed away |
| 04:29.02 | St1ckm4n | I don't think the holdtime in show queues will suffice |
| 04:29.14 | St1ckm4n | some of my queues have a hold time even though there's no calls in there |
| 04:29.21 | |Vulture| | I love calling a company and hearing the * menus |
| 04:29.29 | fugitivo | Realtime sucks |
| 04:29.43 | |Vulture| | by far * has the best queue system for the client very informative |
| 04:30.09 | St1ckm4n | yeah I've heard that realtime sucks from several people |
| 04:30.21 | St1ckm4n | I'ld like to stay away from it if I could come up with another way |
| 04:30.32 | *** part/#asterisk talljon84 (n=talljon8@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 04:30.58 | |Vulture| | Im still using multi-.confs |
| 04:31.22 | |Vulture| | was gunna do AEL but I was told horror stories from using the current state |
| 04:36.59 | *** join/#asterisk kgeffert (n=reptyle@52.107.207.68.cfl.res.rr.com) |
| 04:37.43 | *** part/#asterisk kgeffert (n=reptyle@52.107.207.68.cfl.res.rr.com) |
| 04:40.23 | rajiv | hmm linksys seems to have upgraded the spa-942 so that it has 100meg ports. or at least their new pdf datasheet says that |
| 04:40.40 | justinu | that's great news |
| 04:40.53 | rajiv | and the display is backlit |
| 04:41.43 | rajiv | oh wiat. the 921 and 922 have backlighting but not the 941 or 942 |
| 04:41.45 | rajiv | odd |
| 04:42.06 | rajiv | i really need new phones. the ones i have do not support call parking |
| 04:44.05 | *** join/#asterisk testshifter (n=testshif@203.172.17.212) |
| 04:44.16 | testshifter | hello there! im a newbie in asterisk |
| 04:44.22 | testshifter | can someone help me install this |
| 04:45.50 | rajiv | ~docs |
| 04:45.53 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 04:47.09 | testshifter | I need someone to answer my questions regarding this |
| 04:47.20 | rajiv | just ask. if someone can answer, we will |
| 04:47.26 | justinu | money would get answers faster |
| 04:50.30 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:50.43 | rajiv | wacky prob on my sip phones: on a call, press flash, dial 700, no sound, asterisk debug shows Playing 7 0 1, then the sound comes back on the phone and it is still connected to the original party. in 45 seconds, * tried to send the parked call back but get a BUSY and drops all calls. |
| 04:52.02 | VeNoMouS_ | hrm with mixmonitor() what bitrate does the wav encode @ , cause im trying to convert to mp3 using toolame and its comming out way to fast |
| 04:52.54 | justinu | 8000hz |
| 04:52.56 | justinu | 8bit |
| 04:53.05 | VeNoMouS_ | hrm shit |
| 04:53.13 | VeNoMouS_ | cause choose -s 8 |
| 04:53.15 | VeNoMouS_ | but it dont accept 8 |
| 04:53.33 | VeNoMouS_ | even tho its listed |
| 04:53.54 | Qwell | 8 or 8000? |
| 04:54.04 | VeNoMouS_ | it does it khz |
| 04:54.08 | VeNoMouS_ | 8,16,24,32... |
| 04:54.15 | VeNoMouS_ | 8000hz == 8khz |
| 04:54.29 | Qwell | ok |
| 04:56.37 | VeNoMouS_ | damn it cant do native mpe |
| 04:56.38 | VeNoMouS_ | damn it cant do native mp3 |
| 04:56.51 | VeNoMouS_ | app_mixmonitor.c:173 mixmonitor_thread: Cannot open /var/lib/asterisk/monitor/28-03-2006-16:55:57-021772937-099705560.mp3 |
| 04:57.29 | Qwell | format_mp3 from asterisk-addons |
| 04:57.42 | Qwell | it can read them |
| 04:58.06 | VeNoMouS_ | i have asterisk-addons installed |
| 04:58.27 | VeNoMouS_ | ok or maybe i dont ave format_mpe |
| 04:58.28 | VeNoMouS_ | ok or maybe i dont ave format_mp3 |
| 04:58.43 | VeNoMouS_ | shit i dont ave addons installed on this box |
| 05:00.08 | VeNoMouS_ | heh that could be why |
| 05:00.41 | VeNoMouS_ | Mar 28 17:00:15 WARNING[9221]: file.c:981 ast_writefile: No such format 'mp3' |
| 05:00.42 | Idle | on my wildcard, do I need to have the FXO on 1 and 2, and FXS on 3-4? |
| 05:01.20 | Qwell | Idle: most people put the FXS lower, but it doesn't matter |
| 05:01.35 | Idle | or... is there something more then just plugging the modules in? its not really loading anything other the channel 1 |
| 05:01.46 | Qwell | Idle: zaptel.conf |
| 05:02.04 | Idle | fxoks=1-2 fxols=3-4 |
| 05:02.15 | Idle | oh lol |
| 05:02.18 | Qwell | yeah... |
| 05:02.24 | Qwell | fxsks |
| 05:02.47 | VeNoMouS_ | lol Qwell |
| 05:02.50 | Idle | whats ks? dont I want loop start? |
| 05:02.54 | Qwell | either way |
| 05:02.55 | VeNoMouS_ | the readme states |
| 05:02.56 | VeNoMouS_ | This is a module for asterisk to play mp3 natively. |
| 05:02.56 | Qwell | k is kerlwe |
| 05:02.57 | VeNoMouS_ | They *SHOULD* be already at 8khz and *SHOULD* be mono. |
| 05:02.57 | VeNoMouS_ | otherwise they will be consuming CPU alot more than need be. |
| 05:02.59 | Qwell | kewler |
| 05:03.00 | VeNoMouS_ | its not for encoding |
| 05:03.12 | Qwell | Idle: kewlstart.. |
| 05:03.40 | Qwell | VeNoMouS_: yeah, just reading |
| 05:04.32 | Idle | wtf... invalid arguement |
| 05:04.40 | Qwell | ? |
| 05:04.44 | Idle | the green are the fxs correct |
| 05:04.48 | Qwell | right |
| 05:04.52 | Qwell | but...it's backwards |
| 05:04.57 | Qwell | fxs modules use fxs signalling |
| 05:04.58 | Idle | of lame |
| 05:05.04 | Qwell | erm |
| 05:05.06 | Qwell | fxs modules use fxo signalling |
| 05:05.14 | Qwell | so, for fxs, you'd do fxoks |
| 05:05.21 | Idle | ok, its working, I guess |
| 05:06.10 | VeNoMouS_ | lets try lame instead of toolame |
| 05:06.12 | Idle | ya know, I'm just gonna accept ks... |
| 05:07.16 | Idle | ok, its working |
| 05:07.26 | Idle | I need some solid docs on zaptel |
| 05:09.11 | Idle | how can I tell which channel the call came in on for my dialplan? |
| 05:15.12 | chris_ast | Is there any app_conference installation compatible with asterisk 1.2.4? Please tell me. |
| 05:16.32 | chris_ast | Qwell, Idle, VeNoMouS_: Please help me. |
| 05:16.40 | Idle | I have no idea |
| 05:17.21 | chris_ast | ok Idle, can someone please tell me this? |
| 05:19.30 | *** join/#asterisk }btorch{ (n=btorch@c-69-180-105-139.hsd1.fl.comcast.net) |
| 05:21.24 | Idle | ah, just different contexts... lame, but OK |
| 05:23.07 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 05:23.21 | clyrrad | Hey all, I used to have an extension phones could dial and hear MusicOnHold, now its not working any longer, maybe i have changed something accidentaly. How can I have an extension for a person to dial and hear MusicOnHOld from their phone? |
| 05:25.17 | clyrrad | How can i just play back MOH music when a given extension is dialed? |
| 05:27.11 | x86 | exten => 666,1,Answer exten => 666,2,MusicOnHold(default) |
| 05:28.20 | VeNoMouS_ | sweet |
| 05:28.31 | VeNoMouS_ | got my mixmonitor -> wav -> mp3 going |
| 05:28.50 | VeNoMouS_ | so rather then bout 1meg a min its bout 500k a min |
| 05:30.29 | VeNoMouS_ | heh i should write a mp3 to base64 to mysql script |
| 05:30.33 | VeNoMouS_ | and store the mp3s in mysql |
| 05:31.53 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.165.216.telnor.net) |
| 05:32.43 | brc_ | yo yo yo |
| 05:32.53 | brc_ | look who's back |
| 05:33.03 | brc_ | it's only been what, 8 months? |
| 05:33.27 | brc_ | does anybody use dell servers for asterisk, and if so could you suggest a specific model? |
| 05:34.12 | chris_ast | Is asterisk app_conference scalabe? |
| 05:34.33 | brc_ | define scaleable |
| 05:35.10 | chris_ast | when more people are using conference it should not crash Asterisk |
| 05:35.24 | De_Mon | scalable = stable ? |
| 05:35.41 | brc_ | chris_ast, define more people |
| 05:35.41 | De_Mon | people say it runs better than app_meetme with large conferences |
| 05:35.51 | chris_ast | I think both are different terms though cloasely related |
| 05:36.02 | De_Mon | brc_ how many people does it take before app_conference has "problems" |
| 05:37.04 | De_Mon | chris_ast it could be said the app is stable up-to-xnumber-of-members |
| 05:37.34 | VeNoMouS_ | De_Mon it depends on the spec of the box |
| 05:37.36 | chris_ast | De_Mon,brc_: Is there a version compatible with asterisk 1.2.4 |
| 05:37.42 | VeNoMouS_ | ure talking bout linking x number of channels |
| 05:37.43 | De_Mon | I've not heard of any problems with app_conference and any volume of coferences... |
| 05:37.57 | VeNoMouS_ | so it depends on really how many x number of channels the box can handle |
| 05:37.58 | De_Mon | that's not app_conferences fault though |
| 05:38.11 | De_Mon | it'll handle whatever the box can handle |
| 05:38.13 | VeNoMouS_ | app_conferences just echos to the other channels |
| 05:38.27 | chris_ast | Is there a version compatible with asterisk 1.2.4? |
| 05:38.30 | VeNoMouS_ | picture it as a sip call, with more then 2 channels listening |
| 05:38.30 | VeNoMouS_ | heh |
| 05:38.37 | De_Mon | chris_ast I've been told it comples against 1.2.5, but I've yet to get it to compile, period |
| 05:38.51 | De_Mon | it's on my *to-do-list |
| 05:39.00 | chris_ast | ok :) |
| 05:39.06 | clyrrad | my MOH just stopped working :( |
| 05:39.11 | clyrrad | cant see why |
| 05:39.18 | VeNoMouS_ | clyrrad is mpg123 running? |
| 05:39.24 | clyrrad | i used to have *6613 to hear music |
| 05:39.31 | clyrrad | from my phone, and now i just hear silence |
| 05:39.38 | chris_ast | I just found that we have to make some changes to makefile and someother file for asterisk 1.0.7 |
| 05:39.40 | clyrrad | any idea why it does not work now? |
| 05:39.57 | clyrrad | VeNoMous, yes i have it installed |
| 05:40.01 | clyrrad | is that what you mean? |
| 05:40.06 | chris_ast | So just curious about the changes needed for asterisk 1.2.4? |
| 05:40.34 | De_Mon | the person I spoke to downloaded app_conference svn and compiled without any work afaik |
| 05:41.45 | chris_ast | any links for that? |
| 05:43.45 | De_Mon | :pserver:anonymous@cvs.sourceforge.net:/cvsroot/iaxclient app_conference |
| 05:43.53 | *** join/#asterisk dextro (n=dextro@cpe-70-116-10-201.austin.res.rr.com) |
| 05:44.07 | De_Mon | guess it's cvs not svn |
| 05:44.26 | chris_ast | ok got it |
| 05:48.04 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 05:48.37 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 05:49.12 | websae | does anyone here push high call volume, as like a call center, or retailer? I am curious how your quality is, and if you you're using IAX? |
| 05:50.30 | websae | certainly a quiet channel tonight here... |
| 05:51.24 | Abydos313 | everyone is on the phone :P |
| 05:53.30 | shido6 | heh |
| 05:53.35 | *** part/#asterisk }btorch{ (n=btorch@c-69-180-105-139.hsd1.fl.comcast.net) |
| 05:54.35 | wasim | poor websae |
| 06:00.16 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:07.14 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-36.claranet.co.uk) |
| 06:10.02 | *** join/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:10.24 | *** part/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:10.31 | *** join/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:12.06 | glazzier | is there a url prefix for dialing like pstn://18005551212 |
| 06:12.51 | *** part/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:14.03 | *** join/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:14.18 | *** part/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:16.23 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 06:17.03 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com) |
| 06:17.05 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 06:17.21 | *** join/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:17.35 | *** part/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:17.38 | *** join/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:17.48 | *** part/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:18.47 | *** join/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 06:19.47 | glazzier | hello am i unregeister or not? |
| 06:20.05 | *** join/#asterisk Souvent22 (n=chatzill@c-69-143-189-36.hsd1.va.comcast.net) |
| 06:20.11 | Souvent22 | hello. |
| 06:20.18 | glazzier | hello. |
| 06:20.24 | Souvent22 | I was wondering...what exaclty is a "channel" in Asterisk? |
| 06:21.06 | glazzier | its a plug for sound to come in and out of. |
| 06:21.21 | glazzier | "sound" |
| 06:21.38 | Souvent22 | ah, ok. |
| 06:21.47 | Souvent22 | are you familiar with Wildfire or Asterisk-IM? |
| 06:21.55 | glazzier | nope |
| 06:22.30 | glazzier | anyone else there? |
| 06:23.04 | glazzier | I think this is the wrong channel. I registered and all. did I get banned or something |
| 06:23.19 | Souvent22 | where are you from? |
| 06:23.27 | glazzier | california |
| 06:23.38 | Souvent22 | could be the time. |
| 06:23.40 | dpryo | heh |
| 06:23.47 | dpryo | This is #asterisk :) |
| 06:23.53 | Souvent22 | my other IRC channels are going fine. |
| 06:23.59 | dpryo | I just woke up, here in Norway :) |
| 06:25.12 | Souvent22 | ha |
| 06:25.50 | glazzier | anyone know if there is a standard for urls and autodialer/ |
| 06:26.14 | glazzier | something like pstn://18885551234 |
| 06:28.05 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:36.04 | *** join/#asterisk medusaXX (n=medusaxx@p54A98207.dip0.t-ipconnect.de) |
| 06:40.12 | *** join/#asterisk kimosabe (n=kimosabe@201.153.15.149) |
| 06:40.53 | kimosabe | is there a way to flash cisco router with asterisk |
| 06:42.10 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-141.west.biz.rr.com) |
| 06:43.27 | Qwell | kimosabe: That would be somewhat silly, IMO |
| 06:43.43 | Qwell | cisco routers are quite good at what they do...heavily tuned for it |
| 06:44.04 | kimosabe | qwell i have several cisco routers laying around that are no longer good since asterisk |
| 06:44.19 | adelas | anyone fimilar with Belkin VoIP boxes?I want to know how the dial plan setting goes, the original is "|1[2-9]xxxxxxxxx|011x.T|" - i want to set it so i don't have to enter the 1+area code eveytime, anyone have a clue? |
| 06:50.37 | *** join/#asterisk markdd3 (n=markdd@203-59-210-134.dyn.iinet.net.au) |
| 06:50.50 | markdd3 | hi everyone... |
| 06:51.26 | markdd3 | quick question - I've got an * box, and incoming calls are working over the PRI, but outgoing isn't. |
| 06:51.40 | markdd3 | I get : Zap/5-1 is proceeding passing it to SIP/400-810e |
| 06:51.42 | markdd3 | <PROTECTED> |
| 06:51.57 | markdd3 | any ideas? |
| 06:52.12 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:54.10 | kimosabe | can some one guide me 2 a config where e-1 card is being used for data trafic on lease line please |
| 06:56.06 | markdd3 | heh - looks like we're out of gurus here... |
| 06:57.23 | kimosabe | every where i go i replace servers and routers with linux and unix i want 2 replace a pair of motorolla routers with asterisk via e-1/t-1 card is it possible ? |
| 06:58.51 | SwK | if they are just point to point or frame relay data T1/E1s then you dont need asterisk |
| 06:59.08 | SwK | you just need linux/bsd/whatever and a T1/E1 card |
| 07:00.53 | SwK | not that what you are doing is nessecarily bad, but why not use a real router... they are built specifically for that sort of thing and have lower TCOs when you factor in maint and admin cost over the life of the thing, not to mention peecee hardware just doesnt give you "5 9s" of uptime like a solidstate hardware router |
| 07:00.55 | kimosabe | swk can you lead me 2 the info on this please a digium card will do ? and with what aplication for the dlci and all normal config for lease lines |
| 07:01.26 | orlock | aha |
| 07:01.36 | orlock | 5 nines from just using a router |
| 07:01.37 | orlock | sif |
| 07:01.51 | SwK | sif? |
| 07:01.59 | orlock | how else can it fail, let me count the ways.. |
| 07:02.03 | orlock | line cuts, power outages |
| 07:02.17 | SwK | info's on digiums website for for setting up point to point stuff |
| 07:02.26 | orlock | not all routers are perfect, they have OS's and are susceptible to bugs, flaws, heat issues and memory leaks as wel |
| 07:02.26 | orlock | :) |
| 07:02.29 | kimosabe | swk nixs running 5 yrs no reboot can i rellay be afected on lease line with nix ? |
| 07:03.00 | SwK | unixes running 5 years w/ no reboot is asking to get hacked |
| 07:03.02 | orlock | one you get above 4 9's, you NEED to have a redundant backup for every single link in the chain |
| 07:03.17 | orlock | cos it will fail |
| 07:03.26 | SwK | orlock: thats why real routers have things like redundant power supplies etc |
| 07:03.30 | orlock | but thats past a specific device and into the whole site layout/engineering |
| 07:03.36 | SwK | not to mention almost no moving parts |
| 07:03.38 | orlock | they dont |
| 07:03.41 | orlock | and they have shite fans |
| 07:04.06 | SwK | if you want true highuptime cisco isnt want you are looking for anyway |
| 07:04.14 | SwK | you want Junipers |
| 07:04.16 | harlequin516 | When I use the console driver can I send DTMF using the dial cammand (dial #2345) after a call is already connected? |
| 07:04.25 | kimosabe | swk where can i find a sample config on somthing like that |
| 07:04.42 | kimosabe | swk more familiar with motorolla routers on lease lines |
| 07:04.46 | SwK | kimosabe: digiums site has all the info |
| 07:04.51 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 07:04.56 | kimosabe | thanks man |
| 07:04.58 | SwK | its usually just ppp or the like |
| 07:05.15 | kimosabe | swk i thought in order to pick up a lease line that i would need asterisk 2 link it |
| 07:05.17 | harlequin516 | Seems not to work for me... |
| 07:05.34 | SwK | asterisk does telephony |
| 07:06.12 | SwK | and when you use like a TE405 for voice and data, the data never makes it to asterisk, even when using a single T1 or E1 split part voice, part data |
| 07:06.13 | kimosabe | ok what does lease lines just the card but i want to give my server the coper directlly multiplex from e-1 2 t-1 for international crosings |
| 07:06.49 | kimosabe | swk let me look at digium site for mor info then |
| 07:07.58 | SwK | kimosabe: there is software that goes with it, but your specific configuration can vary... is it just PPP on the link, is it frame relay etc... however, moving data is beyond the scope of asterisk in what you have asked for, there are other utilities for it in linux |
| 07:08.07 | kimosabe | u see here they give me a hdsl pair gain then from there it goes to a cdu/dsu and then from there a get a V.35 |
| 07:08.16 | kimosabe | i want 2 recive the coper directllly |
| 07:08.41 | kimosabe | can u just give me a clue where 2 look |
| 07:08.43 | SwK | so the pairgain drops a 4wire E1 or T1... |
| 07:08.53 | SwK | look at digiums support pages |
| 07:08.53 | kimosabe | 4 wire true |
| 07:09.11 | x86 | run it into a smartjack then to a T1 card with built-in CSU/DSU |
| 07:09.44 | SwK | if its just a leased line, then you just run pppd on the *nix box and be done with it, theres setup instructions on digiums support page |
| 07:09.58 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 07:10.25 | kimosabe | swk you make it really sound easy |
| 07:10.29 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 07:11.52 | markdd3 | I've got a sangoma card here and the wanrouter stuff seems to allow you to use part of the span for data.. |
| 07:12.09 | markdd3 | but I haven't tried to set it up. |
| 07:12.29 | markdd3 | Anyone able to help with my outgoing calls over an E1 problem? |
| 07:14.02 | vgster | i cant help you |
| 07:14.08 | *** join/#asterisk gfox (n=vince@calvix/staff/gfox) |
| 07:14.42 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:15.19 | vgster | when picking up a call is it possible to view the callers number on the phone doing the pickup? or is this for the future? |
| 07:15.44 | shido6 | t100p has been doing that for yrs, markdd3 :) |
| 07:16.00 | x86 | vgster: how did you get call pickup working? i cant seem to get it to work... |
| 07:16.40 | vgster | well if the exts are in pickup groups a *8# will do it |
| 07:17.07 | vgster | and i have directed pickup using *8XXXX# but obviously i dont get the callers info just the number i dialed on the display |
| 07:17.17 | x86 | ah |
| 07:17.24 | x86 | you have to use a # at the end |
| 07:17.29 | vgster | or send |
| 07:17.32 | x86 | i only care about directed pickup |
| 07:17.35 | x86 | hmm |
| 07:17.38 | vgster | i have that too |
| 07:17.56 | vgster | but id luike the call info as i have a lot of internal calls and they get the whole welcome to XXXXXX |
| 07:17.58 | x86 | i have a BT101 (extension 103) and a X-Lite soft phone (extension 100) |
| 07:18.14 | x86 | if 100 is ringing, and i dial *8100# (or send), i get a 485 error from asterisk |
| 07:18.30 | vgster | you need to add it to your dialplan |
| 07:18.38 | x86 | ah |
| 07:18.43 | x86 | can you give me an example? |
| 07:19.06 | vgster | yes 1 mo |
| 07:19.20 | vgster | http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup |
| 07:19.30 | vgster | look at the example on that page |
| 07:20.08 | x86 | w00t :) |
| 07:20.20 | x86 | also my BT101 does not support conference calling (3 way) |
| 07:20.44 | x86 | i followed the instructions in the user manual but when i'm on an active call and hit conference, it does nothing... |
| 07:21.19 | x86 | it's supposed to put party B on hold and give me dialtone so i can call party C, then hit conference again to bridge them |
| 07:21.39 | vgster | never bothered with 3 way calling on them, i did it with the gxp-2000's though |
| 07:22.21 | x86 | i'm assuming there is no other way to handle 3 way besides on the phone itself/ |
| 07:22.21 | x86 | ? |
| 07:22.54 | markdd3 | conference? |
| 07:23.07 | x86 | but then people have to call in right? |
| 07:23.11 | x86 | like with MeetMe, etc |
| 07:23.19 | markdd3 | yep, but you could transfer them to it. |
| 07:23.35 | x86 | my BT101's transfer is busted too :( |
| 07:23.48 | vgster | what button? |
| 07:23.54 | x86 | "Transfer" |
| 07:24.26 | x86 | it puts party B on hold and gives me dialtone, but when i dial a number and hang the phone up, it rings the call back to me |
| 07:25.22 | x86 | wait, now it works ;) |
| 07:25.27 | x86 | blind anyway... |
| 07:25.35 | x86 | i cant figure out assisted :( |
| 07:25.49 | x86 | or attended... whatever ;) |
| 07:27.06 | vgster | i did, use the flash button i think. ill have to have a look cos i documented it for work |
| 07:28.56 | vgster | you could always config it via features.conf |
| 07:29.11 | x86 | call pickup still isnt working |
| 07:29.16 | x86 | still giving me a 484 error |
| 07:29.38 | x86 | these are SIP extensions and a SIP call, if that matters |
| 07:33.47 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:36.03 | vgster | so anyone know if its possible to have the caller number etc on the ext that does the pickup() |
| 07:37.59 | *** join/#asterisk Strom_M (n=strom@66.159.243.59) |
| 07:39.03 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 07:40.03 | *** part/#asterisk robin_sz (n=nospam@adsl.redpoint.org.uk) |
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| 07:45.46 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
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| 08:00.43 | *** part/#asterisk X-Gen (n=x-gen@dsl-145-231-103.telkomadsl.co.za) |
| 08:03.09 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 08:11.37 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 08:15.26 | Shaun2222 | if i dont have any hardware and only plan on using a SIP provider right now do i still need zaptel? |
| 08:15.40 | Shaun2222 | can i build asterisk by it self? |
| 08:16.14 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:16.18 | x86 | you'll need zaptel (ztdummy) for timing for many things like MOH and MeetME |
| 08:16.34 | x86 | so i'd say yes, unless you dont care about that ;) |
| 08:17.03 | Shaun2222 | i'm not sure what those things are :) |
| 08:22.19 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:23.49 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 08:23.51 | *** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
| 08:30.38 | Shaun2222 | anybody know why i'm having build issues.. http://pastebin.com/626545 |
| 08:32.51 | dlynes | It might prove useful to know which version of the zaptel drivers you're compiling, too |
| 08:33.40 | Shaun2222 | latest from csv |
| 08:39.00 | dlynes | You mean svn? |
| 08:39.08 | *** join/#asterisk salviadud (n=salviadu@dsl-201-129-86-188.prod-infinitum.com.mx) |
| 08:39.20 | x86 | probaby he means cvs ;) |
| 08:40.26 | dlynes | Yeah, but asterisk isn't on cvs anymore |
| 08:40.30 | dlynes | It's on svn now |
| 08:40.42 | dlynes | So I'm guessing he must be installing some old buggy version of 1.0 |
| 08:41.59 | Shaun2222 | i'm following... http://www.digium.com/en/docs/asterisk_handbook/downloading_compiling.html |
| 08:42.11 | *** join/#asterisk mko-025 (n=korpim@p54989868.dip0.t-ipconnect.de) |
| 08:43.07 | Shaun2222 | i dont see a way to find out what version i have |
| 08:43.14 | salviadud | i just called a rehab clinic |
| 08:43.20 | salviadud | got it on mixmonitor |
| 08:43.35 | x86 | GIVE ME! |
| 08:43.36 | x86 | lol |
| 08:43.46 | x86 | salviadud: /dcc send me ;) |
| 08:43.50 | salviadud | alright! |
| 08:43.51 | salviadud | hehe |
| 08:44.11 | salviadud | let me find it first... |
| 08:44.19 | salviadud | its time stamped |
| 08:44.51 | x86 | lol |
| 08:45.26 | Shaun2222 | where is the current documentation? |
| 08:46.25 | x86 | salviadud: maybe email is better bryce@shellshark.net |
| 08:46.30 | Shaun2222 | all of these docs i'm seeing are using cvs... |
| 08:46.33 | x86 | my DCC seems busted |
| 08:46.40 | salviadud | alright |
| 08:48.38 | dlynes | Shaun222, http://www.asterisk.org/download |
| 08:48.47 | dlynes | Shaun222, then scroll down until you see the info about svn |
| 08:48.47 | *** part/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 08:49.22 | dlynes | Shaun222, but i don't understand why you're trying to cvs/svn the latest zaptel code...why not use the latest stable code? |
| 08:49.36 | dlynes | Shaun222, It's available as a tarball, and you don't have to muck with svn |
| 08:50.54 | Shaun2222 | i dont know, i read that you could only get it with cvs from some doc |
| 08:51.01 | Shaun2222 | apparently alot of this shit is out-dated |
| 08:51.07 | dlynes | Yeah...must be a really oooooooooooooooold doc |
| 08:51.56 | dlynes | The latest version of zaptel driver is 1.2.5 (it just came out yesterday) |
| 08:51.58 | tecnico | is there a variable similar to ${EXTEN} to know the calling party's extension ? |
| 08:52.09 | dlynes | The latest version of asterisk is 1.2.6 (it just came out yesterday as well) |
| 08:52.20 | Shaun2222 | whats libpri used for? |
| 08:52.24 | dlynes | ${CALLERIDNUM} |
| 08:52.26 | salviadud | i'm using 1.2.6 |
| 08:52.30 | tecnico | dlynes: tnx |
| 08:52.30 | salviadud | works pretty good |
| 08:52.32 | dlynes | Shaun222, for the pri cards |
| 08:53.10 | Shaun2222 | ok, guess i dont need that |
| 08:53.21 | tecnico | how about a variable for the extension's context ? |
| 08:53.29 | dlynes | Shaun222, probably not, unless you're setting up a lot of phone lines |
| 08:53.34 | tecnico | the calling party's extension I mean |
| 08:53.36 | Shaun2222 | http://www.asterisk.org/support that site has a link to docs that are outdated... |
| 08:54.04 | Shaun2222 | dlynes: do you have a link to any good documentation thats current.. |
| 08:54.04 | dlynes | Shaun222, no idea....i haven't gone there for a while, but you might want to mail someone at digium to let them know |
| 08:54.29 | dlynes | Shaun222, http://www.voip-info.org/wiki/index.php?page=Asterisk |
| 08:54.43 | dlynes | Shaun222, that's the most current there is |
| 08:55.05 | dlynes | Shaun222, it's pretty much the official asterisk wiki |
| 08:56.40 | x86 | salviadud: bahahahaha :) |
| 08:57.20 | fourcheeze | anyone know of a symbian SIP client? |
| 08:57.36 | fourcheeze | (nokia n70) |
| 08:57.54 | fourcheeze | apparently it has a sip stack |
| 08:58.08 | fourcheeze | seen lots of promises of clients, but no actual software |
| 08:58.24 | dlynes | tecnico, http://www.voip-info.org/wiki/index.php?page=Asterisk%20variables |
| 08:58.41 | dlynes | tecnico, that should give you a full list of all predefined global variables...you'll need to scroll down a bit to see them |
| 08:59.13 | dlynes | fourcheeze, I seem to recall seeing something somewhere for it |
| 08:59.24 | tecnico | dlynes: tnx. that's what I need. By the way, the ${CALLERIDNUM} doesn't do it for what I need 'cause the caller id of the client is set to a pstn number, but it's extension on the PBX is different.. |
| 08:59.29 | dlynes | fourcheeze, I put it out of mind though cause Internet on cell phones are so expensive |
| 08:59.45 | dlynes | tecnico, If that's what you're wanting, no...there's no variable for that |
| 08:59.47 | fourcheeze | yes, this is true, however I just want to try it |
| 08:59.57 | salviadud | x86 :) |
| 09:00.03 | x86 | i'm trying to setup MeetMe. I've setup an extension to create a dynamic, pin-less room and what not and that works fine... |
| 09:00.04 | fourcheeze | it's already possible to get umetered 3g bandwidth |
| 09:00.11 | dlynes | fourcheeze, I think I seen it on the Symbian OS or the free pda software list or somethign |
| 09:00.17 | x86 | i'm trying to setup an extension to get into an existing conference |
| 09:00.35 | x86 | exten => _9986X.,1,Macro(app-join-conf,1,${EXTEN:4}) |
| 09:00.58 | x86 | where that macro is: s,1,MeetMe(${ARG1}) |
| 09:02.09 | x86 | anyone know what i'm doing wrong? |
| 09:04.38 | fourcheeze | x86: what happens that you don't want? |
| 09:04.38 | dlynes | fourcheeze, congrats :) |
| 09:05.53 | x86 | fourcheeze: well, i dial 9986# and it creates the conference as expected, says you are the only person in the conference blah blah, starts hold music... |
| 09:06.22 | x86 | fourcheeze: i dial 99860# from another device (where 0 is the conference number it auto-created), and i get a 484 incomplete address response |
| 09:07.17 | x86 | fourcheeze: in the CLI with verbose set as high as it will go, i see nothing when 99860# is dialed, unless i turn on SIP debugging also... |
| 09:07.27 | x86 | fourcheeze: so it looks like it's not even hitting the dialplan? |
| 09:08.59 | x86 | odd, when the conference is created, i see this in CLI "Created MeetMe conference 1023 for conference '0'" |
| 09:09.06 | x86 | so what's the conference number, 0 or 1023? |
| 09:10.12 | x86 | w00t, got it ;) |
| 09:10.30 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 09:11.53 | x86 | had to make it be exten => _9986. instead of _9986X. |
| 09:12.07 | x86 | makes sense ;) |
| 09:13.06 | fourcheeze | :-) |
| 09:14.43 | x86 | now to setup automon |
| 09:15.09 | exten123 | Who has selling the services of establish asterisk for others company? can share some experience with me? |
| 09:18.00 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:18.04 | Dandre | Hello all |
| 09:18.20 | exten123 | hello |
| 09:18.41 | Dandre | does anyone knows where I could find some free mp3 files for MOH? |
| 09:19.27 | dlynes | btw...anyone experience problems with sipura 3000's where you'll have a conversation on the incoming line with a sip phone, and the sipura 3000 forgets it's having a conversation, and accepts another call, and then the original two people are talking, and the new person comes in as a 3-way call, that person realizes he's in a conversation he's not supposed to be in, drops the call, and consequently all parties are disconnected? |
| 09:20.35 | dlynes | Dandre, just ask google the question, 'Where do i get free classical mp3s?' |
| 09:20.46 | dlynes | Dandre, you'll find a plethora of links |
| 09:25.48 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 09:28.47 | x86 | is there a way i can record a call, and shove it in ${CALLERIDNUM} 's voicemail box? |
| 09:30.15 | x86 | i know i could just copy the wave file to their inbox, but not sure how to create the text files needed to make it act like a real message |
| 09:31.36 | *** join/#asterisk yuta-vcnet (i=yuta-vcn@212.118.246.50) |
| 09:33.30 | yuta-vcnet | hi, I wonder if someone can help with a codecs-voicemail issue |
| 09:33.50 | yuta-vcnet | I have my SIP.conf setup so g729 is the preferred codec, then GSM |
| 09:34.07 | yuta-vcnet | the problem is that I don't have g729 licenses and I want to use the voicemail |
| 09:34.30 | yuta-vcnet | is there anyway to tell the voicemail application not to use g729 at all? |
| 09:34.43 | yuta-vcnet | in the SDP I can see: |
| 09:34.44 | yuta-vcnet | Capabilities: us - 0x70e (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) |
| 09:36.37 | x86 | dont use g729 ;) |
| 09:36.43 | tecnico | I keep getting this, any hints on how to fix it ? "chan_iax2.c:7551 socket_read: Received mini frame before first full voice frame" |
| 09:36.47 | x86 | 729 == bad ;) |
| 09:36.48 | x86 | mmkay |
| 09:38.39 | yuta-vcnet | I probably can use iLBC for the SIP phones, but I would like to know where are defined the voicemail codec capabilities |
| 09:38.59 | yuta-vcnet | I was going to go through the source file app_voicemail.c |
| 09:43.03 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:47.53 | *** join/#asterisk jwitte (n=jwitte@port-212-202-101-206.static.qsc.de) |
| 09:49.08 | exten123 | how to run a program with extension of .tcl |
| 09:49.26 | jwitte | Hello, one question: What do I need to do, to activate CLIR on PRI? I set usecallingpres=yes and SetCallerPres(prohib) but this doesn't seemto work |
| 09:50.51 | x86 | yuta-vcnet: what's wrong with GSM? |
| 09:51.09 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 09:52.45 | *** join/#asterisk Master_PE (n=masterpe@cl-35.ams-05.nl.sixxs.net) |
| 09:57.48 | *** join/#asterisk smeevil (n=smeevil@gateway.office.sod.nl) |
| 09:57.54 | smeevil | hello |
| 09:58.26 | smeevil | i am trying to compile asterisk with bristuff, but i keep getting errors like : /usr/include/linux/kernel.h:105: error: syntax error before 'size_t' |
| 09:58.49 | yuta-vcnet | x86, GSM is fine, however I am fully testing Asterisk and I want to ensure the most functionality as possible |
| 09:59.16 | *** part/#asterisk jwitte (n=jwitte@port-212-202-101-206.static.qsc.de) |
| 10:00.28 | yuta-vcnet | if I can find a workaround for the g729 options in the SDP, I don't mind using GSM for voicemail and g.729 for IP-phone to IP-phone |
| 10:00.28 | yuta-vcnet | g.729 is less bandwidth at the end of the day |
| 10:01.35 | *** join/#asterisk Ansonmus (n=ahaeser@a213-84-26-148.adsl.xs4all.nl) |
| 10:01.43 | *** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 10:03.08 | *** join/#asterisk Lino` (i=Lino@i577BCC71.versanet.de) |
| 10:04.17 | *** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net) |
| 10:08.16 | x86 | yuta-vcnet: true, but GSM isnt that much more... |
| 10:08.27 | Ansonmus | Hi, what is the best option for 2 or 3 x ISDN2 (BRI) to asterisk ? |
| 10:13.49 | n0cturnal_ | I'm using asterisk with a Linksys PAP2 ATA.. is there an easy way to place someone on hold? or how do i configure this? |
| 10:15.47 | x86 | hit flash on the phone, then hang up? |
| 10:15.53 | x86 | it should hold the call |
| 10:16.03 | x86 | might ring the phone back though when you hang up ;) |
| 10:16.19 | n0cturnal_ | doesn't work... =\ |
| 10:16.23 | n0cturnal_ | just hangs up |
| 10:16.38 | x86 | i dunno then |
| 10:16.43 | x86 | never messed with a PAP2 |
| 10:16.52 | *** join/#asterisk apardo (n=apardo@87.218.44.228) |
| 10:24.03 | x86 | whoa cool.... this ENUM / E164 stuff is totally rad |
| 10:25.00 | x86 | i can make outbound calls without involving my PSTN trunk, if the destination number is in the E164 database... |
| 10:25.11 | x86 | or, one of them, i should say :) |
| 10:25.11 | {zombie} | n0cturnal_: hookflash to put them on hold, then you can either dial another number to transfer them, or hookflash again to get them back |
| 10:25.25 | n0cturnal_ | hookflash? |
| 10:25.32 | x86 | n0cturnal_: same thing as flash ;) |
| 10:25.34 | {zombie} | yeah, the flash button on your phone |
| 10:25.41 | {zombie} | or hold down the "hook" for a very short amount of time |
| 10:26.06 | {zombie} | if the flash button isn't working then it's probably holding the hook down for either too short or too long |
| 10:26.14 | {zombie} | you can adjust those parameters in the sipura |
| 10:26.19 | {zombie} | I mean the linksys pap2 |
| 10:26.35 | {zombie} | and you can often adjust it on the phone too |
| 10:26.52 | kmilitzer | Is it really right, that the ChanIsAvail-Application in the test-this-branch jumps to +101 even if priority jumping is not set with the j option? |
| 10:29.09 | n0cturnal_ | cheers guys... just had to find the stupid flash button :P |
| 10:29.17 | n0cturnal_ | ruddy fancy cordless phones |
| 10:38.31 | *** join/#asterisk mover (n=dlu@213.9.46.7) |
| 10:41.01 | *** join/#asterisk Modcuts (n=bob@proporta.gotadsl.co.uk) |
| 10:42.20 | Modcuts | what is the best way to look at how ip packets are leaving the asterisk box? |
| 10:44.56 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 10:47.15 | mover | hi all |
| 10:48.10 | *** join/#asterisk danzig (n=chatzill@ruc-kj-013.ruc.dk) |
| 10:49.55 | mover | Modcuts: ngrep 5060 |
| 10:50.16 | mover | for sip pakets |
| 10:50.29 | mover | or just ngrep for all :-) |
| 10:50.49 | *** join/#asterisk lehel (n=mey@86.125.98.100) |
| 10:51.04 | *** join/#asterisk NoRemorse (n=fred@202.161.68.2) |
| 10:51.06 | NoRemorse | hi all. |
| 10:51.14 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-231-103.telkomadsl.co.za) |
| 10:51.50 | NoRemorse | is there any way to change what dtmf setting the voicemail prompt listens for? I have rfc2833 set, but when i enter my pin it is ignored |
| 10:52.02 | mover | can i limit iax2 channels per peer like in sip? i read tons of pages but no hint at all |
| 10:52.22 | NoRemorse | hey mover how do you limit that in sip?! |
| 10:52.26 | kmilitzer | Modcuts: tcpdump or (t)ethereal |
| 10:52.40 | mover | call-limit=x |
| 10:52.49 | NoRemorse | ah ok. |
| 10:53.08 | NoRemorse | like line 1 to 4 in x-lite? |
| 10:53.38 | mover | NoRemorse: you can set it in peers setup with dtmf= |
| 10:54.11 | NoRemorse | in sip.conf? yeah its set to rfc2833 in sip.conf and on my client |
| 10:54.37 | mover | NoRemorse: call-limit=1 mean one outgoing and one incoming call at the same time is maximum |
| 10:54.58 | NoRemorse | I actually suspect it's a bug in the client hardware ands it's using info instead which asterisk seems to ignore for voicemail prompts. |
| 10:55.21 | mover | NoRemorse: try dtmf=info its much more stable |
| 10:55.28 | mover | and relaxdtmf=yes |
| 10:55.55 | Modcuts | kmilitzer: cheers |
| 10:56.08 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 10:56.11 | NoRemorse | ah ok. |
| 10:56.50 | mover | NoRemorse: dtmf mode on UA should be INFO . On asterisk side rfc2833 is ok |
| 10:56.59 | kmilitzer | Modcuts: or ngrep |
| 10:57.30 | NoRemorse | stupid client only has rfc or inband |
| 10:59.18 | NoRemorse | whats the point of setting it in the conf file anyway? what happens if conf is different to what the client is set to do? |
| 10:59.48 | mover | NoRemorse: try inband. what code you use? |
| 10:59.54 | mover | NoRemorse: try inband. what codec you use? |
| 11:00.12 | Modcuts | kmilitzer: ngrep goes nuts would need to dump it |
| 11:00.37 | kmilitzer | Modcuts: Do you want to capture SIP or also IAX? For SIP simply to ngrep port 5060 |
| 11:00.38 | NoRemorse | I am using g729 |
| 11:01.21 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:01.22 | mover | on the asterisk side you have g729 licensed? |
| 11:01.38 | NoRemorse | yes 10. this aint a codec prob |
| 11:01.45 | NoRemorse | all calls work fine |
| 11:02.08 | kmilitzer | NoRemorse: DTMF-Handling is a pain in the ass ... maybe this command can help you: SetDTMFMode |
| 11:02.18 | mover | ok but inband with compression shouldnt will work stable |
| 11:03.06 | kmilitzer | Modcuts: If you want it in a file to tcpdump -w <filename> port 5060 or tethereal -w <filenemae> ... |
| 11:04.54 | mover | kmilitzer: setdtmfmode is the same like dtmf= the difference is only the switching at dialplan |
| 11:05.53 | NoRemorse | yeah seems unstable over g729, have to enter thye number multiple times sometimes, nhowever thats ok, the prob is that asterisk voicemail ignores inband?! |
| 11:06.37 | mover | NoRemorse: no it dont ignre. it dont recognise it. try alaw as codec an all will be fine |
| 11:06.57 | NoRemorse | is there *any* way to see what a call is using for dtmf at call establishment time using asterisk console? |
| 11:07.18 | NoRemorse | why does my upstream trunk recognise inband then?! |
| 11:08.48 | mover | NoRemorse: because your asterisk send dtmf oder info |
| 11:09.00 | mover | oder=over |
| 11:09.17 | mover | i dunno your complete setup |
| 11:09.37 | danzig | >noremorse are your running IP all the way? relaxdtmf only works with zap |
| 11:10.12 | NoRemorse | ok client is set to inband, so bottom line is the numbers get sent via inband tones, so either the dtmf reaches the destination through my provider as inband or it gets converted to info by *, either way * hould recognise it for the voicemail pin! |
| 11:10.28 | NoRemorse | yes ip all the way, upstream is sip |
| 11:10.49 | fnordian | huh |
| 11:11.06 | danzig | norem>> * voicemail does not accept inband. known limitation |
| 11:11.17 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 11:11.28 | fnordian | i checked out the subscribemwi-branch, compiled and launched with -fg which led to segfault :-( |
| 11:11.29 | kmilitzer | danzig: since when should that be? |
| 11:11.33 | NoRemorse | lol thought as much. thats ok, I just cant seem to get my dumb client to go back to rfc |
| 11:12.54 | danzig | kmiltz>> "dtmfmode=info does not work with Asterisks voicemail system" http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode (if it is correct) it certainly was about 6 months ago... |
| 11:13.12 | mover | rfc2833 is also unstable on compressed codes i guess |
| 11:13.47 | NoRemorse | forget info, my client doesnt even have it, it has rfc or inband, but * vm doesnt seem to accept inband |
| 11:14.10 | danzig | norem>> that is also my experience. Use rfc2833 |
| 11:14.23 | mover | NoRemorse: you have tried a uncompressed codec for testing inband??? |
| 11:14.36 | kmilitzer | danzig: OK ... you mean info not inband does not work with voicemail |
| 11:15.03 | mover | SIP_INFO still wont work in 1.2.5 |
| 11:15.16 | danzig | We ue rfc2833 and ALAW - works fine with voicemail - BUT on incoming truncs from provider (SIP/ALAW), there are problems maybe 20% of the time - mainly 0 getting recognised as 00 etc. |
| 11:15.40 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 11:16.54 | mover | we are usinf alaw and inband. in all directions no problems sip and pstn |
| 11:17.11 | danzig | norem>>from same page: "If the codec is not ulaw or alaw then the DTMF tones will be distorted by the audio compression and will not be recognised" I would have thought that g729 would be ok, but maybe you should try with alaw/ulaw |
| 11:17.19 | NoRemorse | I have trouble with alaw on some client they only have crappy 64k upstreams |
| 11:17.44 | *** join/#asterisk Op3r (n=op3r@202.71.189.90) |
| 11:17.54 | mover | danzig: thats what i wrote lines before |
| 11:17.58 | NoRemorse | danzig: yes this is the case for inband only, rfc and info both use data codex muxed into the rtp stream |
| 11:18.12 | NoRemorse | codex=codes |
| 11:18.25 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 11:18.56 | danzig | mover>> yes, yours was just a bit shorter :-) |
| 11:19.15 | *** join/#asterisk bjweeks (n=bjweeks@24.137.180.138) |
| 11:19.16 | NoRemorse | its a catch 22, my sip provider only accepts inband (ie they dont have ANYTHING confogured for dtmf) and asterisk voicemail doesnt accept inband, I have to decide if the custs get voicemail or they get telephone banking lol |
| 11:19.32 | mover | NoRemorse: what compression (ALL CODECS <> a/ulaw) mean that the rtp will be compressed. so you cant expect that the demuxed informations will be sorrect transported.. sorry |
| 11:19.55 | NoRemorse | yes. |
| 11:20.03 | NoRemorse | ok easiest solution disable voicemail lol |
| 11:20.11 | mover | no |
| 11:20.38 | danzig | norem>> depends on what your * is doing - maybe you can use rfc2833 between the phones and Asterisk, and use inband for your trunks - is asterisk 'proxying' these connections? |
| 11:21.30 | NoRemorse | yes, no invites |
| 11:21.46 | mover | easiest is to use alaw for voicemail. if this is impossible your journey is to bring it to alaw :-) |
| 11:22.56 | x86 | why alaw? |
| 11:23.00 | bjweeks | does anybody know if there is a 'cheat sheet' for all the commands you can run from a phone like time, weather, voicemail, etc... ? |
| 11:23.01 | x86 | i always use ulaw ;) |
| 11:23.07 | NoRemorse | anyways... what should I put after the client Dial command , Hangup or Congestion? |
| 11:23.17 | mover | hehe |
| 11:23.21 | danzig | our asterisk 'proxys' all connections between the phones and the trunks, so we can mix rfc and inband. We use rfc2833 between the phones and asterisk, rfc2833 for our main trunks, but inband for trunks voipdiscount (cause voipdiscount don't honor rfc, so one could not use telbanking, answering machines otherwise ) |
| 11:24.33 | danzig | norem>>you just define dtmftype=rfc in the phones peer definition, and dtmftype=inband in the trunk lines peer def. You may have problems with people trying to listen to their voicemail via the trunks. |
| 11:26.21 | danzig | And if anyone can tell me how to stop dtmf 8001 being interpreted by Asterisk as 800011 (over SIP/alaw trunk on a 4% used 100 Mbit fibre, no QoS), I am very interested! |
| 11:27.03 | x86 | is there a way i can record a call, and shove it into a voicemail box? |
| 11:27.24 | x86 | like a user hits *1 before dialing a number, it records the call, saves it to thier voicemail |
| 11:28.00 | *** join/#asterisk Ansonmus (n=ahaeser@a213-84-26-148.adsl.xs4all.nl) |
| 11:28.53 | Ansonmus | Hello, what is the best option for BRI 4 channels? |
| 11:29.20 | danzig | x86>yes, use http://www.voip-info.org/wiki-Asterisk+cmd+monitor , and then a System call to put the file + a text description file in their voicemail folder - but you will probably have to write a short script yourself to make the text file. |
| 11:29.36 | NoRemorse | thanks guys |
| 11:30.04 | x86 | danzig: yeah i know i could do it that way, but i didnt want to make the text file myself ;) |
| 11:31.57 | danzig | x86>maybe you could make life easier for yourself by not putting it in voicemail, but always in the same place, and they get it by email/web/dial an extension. Then when they record the next call the file gets overwritten... |
| 11:33.03 | *** join/#asterisk devilpim (n=Pim@195.135.145.195) |
| 11:33.06 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 11:35.02 | devilpim | hi....anyone's working with spandsp? |
| 11:35.46 | NoRemorse | danzig: I have set up my client to rfc, it can access vm no probs. I set dtmf=inband in the peer conf for my provider, and telebanking ignores dtmf.... |
| 11:36.57 | NoRemorse | shouldnt that 1 entry in the peer section turn the dtmf into inband tones? |
| 11:37.43 | danzig | norem>> thats strange - works for me - rfc from phone to *, inband from * to trunk provider, I can call the tax office and press 9 to hold to bad music... You reloaded etc.? |
| 11:38.32 | NoRemorse | yep :( |
| 11:39.47 | danzig | norem>>Maybe your inband DTMF is getting too mangled for bank to like it - what are your trunks? also g729? Or maybe something changed. I am on Asterisk 1.0.9. |
| 11:40.07 | NoRemorse | no, remember it recognises it when client is set to inband.... |
| 11:40.28 | NoRemorse | trunks alaw |
| 11:40.28 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 11:41.26 | *** join/#asterisk TinoW (n=tino@living-examples.com) |
| 11:41.29 | TinoW | hiho |
| 11:41.48 | NoRemorse | hang on I will call my mobile and see if I hear ANY tones, even mangkled I'll hear something |
| 11:41.51 | danzig | norem>>Possibly my phones also send inband when they send rfc, and yours don't - mine are Grandstream GXP2000... |
| 11:42.10 | *** part/#asterisk devilpim (n=Pim@195.135.145.195) |
| 11:42.31 | NoRemorse | no tones :( |
| 11:43.15 | NoRemorse | can I set dtmf in dialplan? I have a feeling my sip.conf peer ssetting is being bypassed |
| 11:46.09 | danzig | norem>> I still would have thought that Asterisk would change the rfc to inband when bridging the call - it should, IMHO. |
| 11:47.17 | NoRemorse | not if it the default setting is rfc |
| 11:47.30 | NoRemorse | did you mention a dialplan command to set dtmf earlier? |
| 11:50.23 | danzig | norem>>no, but if you are telling the trunk to use inband, it should. I have. dtmfmode=inband in the trunk peer def. Before, I had dtmfmode=rfc2833, and telebanking did not work. I dont know a dialplan cmd offhand... |
| 11:52.07 | NoRemorse | yes but as I said, I think my sip.conf section for my provider is being ignored |
| 11:52.20 | *** join/#asterisk coppice (n=chatzill@168.197.17.210.dyn.pacific.net.hk) |
| 11:52.44 | NoRemorse | it was mover who mentioned setdtmfmode |
| 11:53.08 | NoRemorse | no hits on voip-info.org wiki tho |
| 11:54.19 | NoRemorse | kmilitzer I mean |
| 11:55.28 | *** join/#asterisk Aurs (i=aurs@hallo.aurs.info) |
| 11:57.20 | NoRemorse | ok danzig I just confirmed my dialplan is using the peer setting my using @provider rather than @ipaddress, and it still doesnt pickup the conversion. |
| 11:57.28 | NoRemorse | I am giving up :( |
| 12:02.33 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 12:04.50 | *** join/#asterisk Hermis (n=guitarug@85.21.204.146) |
| 12:07.11 | Hermis | Why, when I Dialing number from asterisk it's generate warning4702 Unable to reopen DSP Device |
| 12:07.13 | Hermis | ? |
| 12:09.23 | stone | is it possible to call external sip (like ekiga.net) from a sip soft-phone connected to asterisk? |
| 12:11.48 | RoyK | stone: yes, rtfm :) |
| 12:11.50 | RoyK | ~docs |
| 12:11.52 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:12.39 | stone | a rtfm... bah :) |
| 12:13.27 | *** join/#asterisk Hermis (n=guitarug@85.21.204.146) |
| 12:16.10 | stone | any pointers what to look for? The info/docs are massive :) |
| 12:17.09 | *** join/#asterisk _andre (n=andre@fosforo.k8.com.br) |
| 12:17.29 | _andre | good morning |
| 12:17.32 | *** join/#asterisk ToTo (n=ToTo@81-174-33-2.f5.ngi.it) |
| 12:17.45 | _andre | has anyone seen this warning message: "Mar 27 17:23:42 WARNING[7850]: chan_sip.c:2530 sip_write: Asked to transmit frame type 2, while native formats is 256 (read/write = 256/256)" ? |
| 12:17.52 | Hermis | good evening:) |
| 12:17.56 | _andre | :) |
| 12:17.59 | danzig | stone>> u can certainly do it by setting up a peer ekiga.net in sip.conf, an extension for that peer in extensions.conf and then dialling the number for that extension, but it depends on what you want to do. Look at docs for sip.conf and extensions.conf |
| 12:18.05 | RoyK | stone: you just dial into asterisk using SIP, and then dial out again using SIP, aka Dial(SIP/ekiga/${EXTEN}) |
| 12:18.31 | TinoW | hm. looking for a small howto/hint/guide to lookup callerid in a database and setting the result |
| 12:19.00 | RoyK | TinoW: use agi :) |
| 12:20.06 | Hermis | 2TinoW you can simply read/edit sip.conf etc to understand callerid settings |
| 12:20.06 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 12:21.02 | stone | ah I got that setup already for digisip so I just do that for SIP also... ofcourse.. |
| 12:21.02 | stone | thanks |
| 12:22.07 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 12:22.44 | RoyK | ~seen zoa |
| 12:22.48 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 12d 19h 39m 31s ago, saying: 'it looks kinda suspicious :p'. |
| 12:22.54 | Dr-Linux | question, how can i send DTMF to other end? |
| 12:22.54 | RoyK | fuck |
| 12:23.31 | Dr-Linux | RoyK: why? |
| 12:24.08 | michael-i | does anyone have any experience setting up CCBS with SIP devices? |
| 12:24.10 | NoRemorse | is there an easy way to Playback "Extension blah is unavailable" in the dialplan? |
| 12:24.41 | Dr-Linux | RoyK: i want as i connect to remote IVR, after 2 sconds i can sent them some 8 digits DTMF ? |
| 12:24.43 | Dr-Linux | how can i? |
| 12:25.02 | michael-i | It's not in Asterisk per default but has anyone written an application / channel hack to support this? |
| 12:25.19 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 12:25.20 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 12:25.22 | kippi | hi |
| 12:25.46 | kippi | I am getting this error when I try and complie asterisk, can anyone help? |
| 12:25.46 | kippi | /usr/bin/ld: cannot find -lssl |
| 12:25.46 | kippi | collect2: ld returned 1 exit status |
| 12:25.46 | kippi | make: *** [asterisk] Error 1 |
| 12:25.55 | Dr-Linux | anybody answer my question? |
| 12:26.12 | danzig | dr-linux>exten dial, wait(2), sendDTMF(1), sendDTMF(2) etc. |
| 12:26.13 | RoyK | Dr-Linux: erm. try again.... |
| 12:26.24 | RoyK | kippi: apt-get install openssl? |
| 12:26.36 | kippi | is that what it is? |
| 12:26.38 | TinoW | RoyK: libssl-dev |
| 12:26.41 | RoyK | ah |
| 12:26.42 | RoyK | yes |
| 12:26.59 | Aurs | kippi: have you installed all the dependencies?= |
| 12:27.11 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 12:27.24 | kippi | just checking now |
| 12:27.36 | TinoW | kippi: which OS? |
| 12:27.39 | Aurs | there is a list on the download page on www.asterisk.org |
| 12:27.55 | kippi | TinoW: FC4 |
| 12:28.21 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 12:28.29 | Aurs | kippi: wild guess (as I said yesterday when you asked about this): you have to install openssl and openssl-devel |
| 12:28.30 | TinoW | kippi: never mind! ;) |
| 12:30.50 | x86 | what's a 603 error represent? |
| 12:30.59 | *** join/#asterisk CKGC (i=CKGC@202.8.86.162) |
| 12:32.38 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 12:33.33 | Dr-Linux | danalien: i wanna send dtmf after 2 seconds while contecting to remote end |
| 12:38.36 | *** join/#asterisk RGi_ (i=RGi@62.97.247.44) |
| 12:39.52 | Dr-Linux | <danzig> dr-linux>exten dial, wait(2), sendDTMF(1), sendDTMF(2) etc. >> but i want to do it in same line |
| 12:41.03 | RGi_ | I have a problem with my queue setup.. when I get a caller and dont take the phone the caller waiting secons should count upwards.. but get reset every 5 sec or so... |
| 12:41.24 | Dr-Linux | i want like this type |
| 12:41.25 | Dr-Linux | exten => 3939,Dial(Zap/gi/918008449087/Wait,2/Senddtfm,somedtfm) |
| 12:41.56 | TinoW | *grr* pyastre seems a bit old... |
| 12:42.18 | kippi | thanks for the help!! just about to try it again |
| 12:42.35 | TinoW | this non extensible C calling schema sux :( |
| 12:42.54 | *** part/#asterisk Hermis (n=guitarug@85.21.204.146) |
| 12:43.25 | danzig | dr-l>>"but i want to do it in same line"? Do not understand what you mean. |
| 12:44.33 | x86 | anyone got ENUM / E.164 lookups working? |
| 12:44.40 | x86 | http://www.voip-info.org/wiki/view/RFC+Compliant+ENUM+Macro |
| 12:44.47 | x86 | i went off that, but it doesnt seem to want to work |
| 12:46.05 | danzig | dr-linux>> why? That is not the syntax for extensions. Dial on the first line, wait on the second line, senddtmf on the tird line. If you want to do it many places, make a macro. |
| 12:47.05 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 12:47.44 | *** join/#asterisk |cleric| (n=dacleric@p5482988D.dip0.t-ipconnect.de) |
| 12:48.26 | *** join/#asterisk inv_Arp (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 12:48.56 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 12:51.45 | Modcuts | other then tcpdump,ethereal what is the best way to look at the ip packet content, to be able to see which tos is being used? |
| 12:51.48 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 12:53.41 | RGi_ | damn this queue stuff... :( |
| 12:55.05 | *** join/#asterisk Splatty47 (n=splatski@host217-34-149-45.in-addr.btopenworld.com) |
| 12:55.42 | RGi_ | exten => 400,5,Queue(400|t|||0) what is this line doing ? |
| 12:55.58 | x86 | Modcuts: you cut out the best ways, then ask what the best way is lol |
| 12:56.15 | Aurs | RGi_: show application Queue |
| 12:56.20 | Modcuts | x86:so what ethereal? |
| 12:56.25 | kippi | starting asterisk with -vvvvvc but its not starting, its getthin here but stoping == Manager registered action AgentCallbackLogin |
| 12:56.25 | kippi | <PROTECTED> |
| 12:56.25 | kippi | [root@localhost ~]# |
| 12:56.27 | *** join/#asterisk pengyong (n=lala@222.185.17.29) |
| 12:56.32 | x86 | Modcuts: either or |
| 12:56.58 | RGi_ | Aurs ah.. thanx :) |
| 12:57.26 | Modcuts | x86: what would be the best way to look at the tos with tcpdump? |
| 12:57.27 | Aurs | RGi_: np ;) |
| 12:58.03 | RGi_ | Aurs : btw.. do you have a url where I can read more about it? :) |
| 12:58.25 | Aurs | RGi_: I bet there is lots of info on www.voip-info.org |
| 12:58.26 | x86 | Modcuts: -X |
| 12:59.20 | x86 | anyone got ENUM / E.164 lookups working? |
| 13:04.18 | *** join/#asterisk Skarmeth (n=Skarmeth@201008193102.user.veloxzone.com.br) |
| 13:07.59 | Skarmeth | hi all |
| 13:08.49 | Skarmeth | someone can suggest a easy tool for provisioning polycom's soundpoint IP 301/501/601? |
| 13:09.30 | TinoW | SkalTura: asterisk? ;) |
| 13:11.03 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:12.57 | Damin | Skarmeth: FTP... |
| 13:13.04 | *** join/#asterisk Tili (i=Tili@219.136.98.101) |
| 13:14.00 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:15.24 | *** join/#asterisk x3me (n=x3me@201.11.226.147) |
| 13:15.27 | x3me | hi |
| 13:15.35 | *** join/#asterisk tamp4x (n=Lab@64.201.13.170) |
| 13:15.35 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-217.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 13:15.36 | x3me | with asterisk i can made calls for what kind of people ? |
| 13:15.40 | x3me | only for another asterisk ? |
| 13:16.06 | TinoW | x3me: usually for people with at least one working ear |
| 13:16.16 | x3me | i can made calls for an cell phone for example? |
| 13:16.20 | Damin | x3me: Have you read any of the documentation available at asterisk.org? |
| 13:16.28 | x3me | Damin, just a little... |
| 13:16.48 | TinoW | x3me: even the most introductory page tells you about it ;) |
| 13:16.51 | Damin | x3me: And what does that say? |
| 13:17.00 | Ansonmus | Hello, what is the best option for BRI 4 channels? |
| 13:17.20 | x3me | TinoW, ok |
| 13:18.41 | x3me | TinoW, i only want to know if i can made calls from outworld with the softphones behind my asterisk server heheh |
| 13:19.00 | x3me | sure, without adictional hardware.. |
| 13:19.07 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 13:19.15 | TinoW | x3me: sure you can, provided you have the hardware to connect your asterisk box with the outworld |
| 13:19.27 | x3me | hmm |
| 13:19.39 | TinoW | x3me: which is clearly stated at the very front page - not to cound all the examples ;) |
| 13:19.44 | TinoW | count |
| 13:19.51 | Damin | x3me: The first two sentences on the about page should answer your question. |
| 13:19.55 | x3me | ok, let me see |
| 13:20.04 | Damin | x3me: "Asterisk is a complete PBX in software. It runs on Linux, BSD and MacOSX and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. |
| 13:20.27 | TinoW | Damin: maybe diginum should run the demo node with the demo text for people like x3me to call ;) |
| 13:20.41 | Damin | x3me: Sounds pretty clear to me that it can operate with almost all "standards-based" telphony hardware. |
| 13:20.46 | x3me | i read it now.. ;) |
| 13:21.01 | Skarmeth | Damin, that's not a easy tool for end-admin-users |
| 13:21.11 | Damin | x3me: Good. Now go buy the book "Asterisk, the Future of Telephony" from O'Reilly and read that too.. |
| 13:21.33 | Damin | Skarmeth: Ohh.. you want an EASY tool. Well, then no. Your screwed. |
| 13:21.33 | Aurs | ...or download the book from asteriskdocs |
| 13:21.35 | Skarmeth | :) for us it's a good tool, but not for most users |
| 13:21.41 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 13:21.43 | x3me | and if only buy an ADSL link and configure an Asterisk server without the adictional hardware... |
| 13:21.47 | x3me | i can made calls for... ? ;) |
| 13:21.59 | x3me | only another node with asterisk ? |
| 13:22.19 | x3me | or only to the contacts i've created in the asterisk configuration? |
| 13:22.56 | TinoW | x3me: there are SIP provider which connect you to POTS if you dont have/want to buy the hardware. |
| 13:23.15 | Aurs | x3me: you can make calls to everybody from a asterisk box with no telephony hardware. if you use a voip provider |
| 13:23.26 | x3me | hmm |
| 13:23.28 | x3me | cool ;) |
| 13:23.40 | TinoW | x3me: actually you dont even need asterisk for that ;) |
| 13:24.22 | x3me | my questions are because the existence of skype and another... |
| 13:25.19 | TinoW | x3me: you cant call skype with a SIP client. No matter if its asterisk or a soft or hardphone. You would at least need a gateway |
| 13:26.24 | x3me | and the gateway is the voip provider.. |
| 13:27.04 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 13:28.00 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:39.29 | *** join/#asterisk Hermis (n=guitarug@85.21.204.146) |
| 13:45.52 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F144B.dip0.t-ipconnect.de) |
| 13:46.18 | *** join/#asterisk Strom_M (n=strom@66.159.243.59) |
| 13:47.20 | [ProB]CrazyMan | hello is it possible to make an extension who matchen several numbers like I want to match number 0800X,0173X,0900X ... and so on? or do I have to make for each an extension and forward it with goto ? |
| 13:47.38 | *** join/#asterisk ReD-MaN (i=redman@207.210.38.45) |
| 13:47.54 | *** part/#asterisk Hermis (n=guitarug@85.21.204.146) |
| 13:48.31 | RoyK | [ProB]CrazyMan: do it with goto or a macro.. |
| 13:49.48 | [ProB]CrazyMan | thx RoyK |
| 13:53.27 | Strom_M | no no, obviously the goal is to get the entire dialplan down to a single line :) |
| 13:59.23 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 14:00.05 | Darwin35 | http://pastebin.ca/47296 here you go have fun . use and add features to it...... |
| 14:02.43 | Darwin35 | let me know if you like it or if it needs work |
| 14:02.51 | Darwin35 | but its there for all |
| 14:03.14 | Strom_M | Darkhalf, you will never have 7-digit and 10-digit dialing in the same location |
| 14:03.16 | Strom_M | er |
| 14:03.20 | Strom_M | Darwin35, |
| 14:03.55 | Darwin35 | it workds |
| 14:04.06 | Darwin35 | I use it all the time |
| 14:04.40 | Strom_M | yes, it works because the 7-digit times out, but any dialplan that's even half-assedly thrown together will never ever combine 7-digit with 10-digit |
| 14:05.10 | Strom_M | 7 and 1+, or 10 and 1+, but never 7 and 10 in the same location |
| 14:07.08 | Strom_M | also, don't just assume N11 codes are free for your taking |
| 14:07.10 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 14:07.31 | Darwin35 | brb |
| 14:08.25 | Strom_M | see, this is why I always say that anyone who intends to do this correctly really does need to take a course in traditional telephony :P |
| 14:10.07 | tzanger | yueah but then they'll get tripped up thinking about circuits |
| 14:10.34 | Strom_M | haha |
| 14:12.11 | Strom_M | yeah, looking at this dialplan, it's quite obvious that Darwin35's network contains only telephones where the complete number to be dialed is sent as part of the call setup message, because otherwise he'd never do something as monumentally stupid as assigning single-digit vertical service codes |
| 14:13.08 | Strom_M | this is, of course, assuming he's actually /read/ the vertical service code assignments, which he doesn't appear to have done |
| 14:13.23 | Hmm-work | answer the phone, I know that you're home, I wanna get you alone and do it again, do it again |
| 14:13.32 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 14:13.53 | frenzy | where do I define the default musiconhold catergory for SIP? |
| 14:14.27 | Hmm-work | Hello to you too |
| 14:15.25 | frenzy | ? |
| 14:15.39 | wasim | frenzy: musiconhold.conf and sip.conf and extensions.conf |
| 14:16.03 | frenzy | what do I have to define in sip.conf? |
| 14:16.10 | frenzy | whats the string |
| 14:16.34 | Strom_M | I'm just going to bite my tongue on the fact that Darwin35 can't seem to spell his way out of a wet paper bag either. :) |
| 14:16.56 | frenzy | wasim : ? |
| 14:16.58 | wasim | frenzy: will you kick yourself if i told you musiconhold= |
| 14:17.12 | Hmm-work | he should, he seems a little demanding |
| 14:17.12 | frenzy | damn |
| 14:17.16 | frenzy | wait one sec |
| 14:17.22 | frenzy | am kicking the wall :) |
| 14:17.24 | Hmm-work | and unable to read |
| 14:17.29 | wasim | or google |
| 14:17.43 | frenzy | i'm not using the sample files |
| 14:17.46 | frenzy | :) |
| 14:17.52 | Hmm-work | all of the sample files are on the wiki |
| 14:17.59 | Hmm-work | which are heavily commented |
| 14:18.02 | frenzy | not for SIP |
| 14:18.03 | frenzy | :) |
| 14:18.28 | Hmm-work | sure it is |
| 14:18.45 | *** join/#asterisk hanchi (n=telliott@68-112-44-203.static.sprn.tx.charter.com) |
| 14:19.23 | Hmm-work | http://www.voip-info.org/wiki-Asterisk+config+sip.conf sfw |
| 14:23.37 | x86 | anyone got ENUM / E.164 lookups working? |
| 14:25.38 | frenzy | Hmm-work: hmm interesting |
| 14:26.10 | Katty | mew. |
| 14:26.16 | x86 | no one? hehe |
| 14:26.16 | mocker | Woo. |
| 14:26.23 | mocker | Got my home asterisk setup talking to Vonage. |
| 14:26.25 | frenzy | woof woof |
| 14:26.31 | Strom_M | mocker, eeeeeeeewwwwwwwwwwwwwwww |
| 14:26.37 | x86 | mocker: that's sick :( |
| 14:26.45 | mocker | hah. |
| 14:26.49 | x86 | mocker: you are paying way too much using Vonage... |
| 14:26.50 | Strom_M | let me guess - you have the terminal adapter hooked into an fxo port |
| 14:26.52 | Katty | i got my asterisk setup working (= |
| 14:27.06 | Katty | it's nice when it works. |
| 14:27.07 | iDunno | \o/ |
| 14:27.08 | mocker | Strom_M: No, I have the softphone connected via a SIP account. |
| 14:27.13 | Katty | iDunno: :> |
| 14:27.23 | Katty | iDunno: are you gonna go to cluecon? |
| 14:27.24 | Strom_M | mocker, ok, wow, thats actually impressive |
| 14:27.39 | mocker | :) |
| 14:27.48 | Hmm-work | frenzy quite |
| 14:27.56 | mocker | I'm doing the asterisk boot camp and stayed late the first day to get it done. |
| 14:28.01 | iDunno | Katty: when/where is cluecon? ;) |
| 14:28.02 | Hmm-work | wow this dialplan I wrote 6 months ago is intersting |
| 14:28.06 | iDunno | (probably not, though) |
| 14:28.22 | Katty | iDunno: august, me thinks.....in chicago |
| 14:28.41 | iDunno | that's a bit of a trip :) |
| 14:28.59 | Katty | a smidgen. |
| 14:29.22 | Katty | my company is....but then again i'm a lot closer. |
| 14:29.38 | docelm0 | Katty, COME TO ASTRICON! |
| 14:29.42 | Katty | docelm0: no |
| 14:29.45 | docelm0 | meanie |
| 14:29.48 | Katty | docelm0: yup |
| 14:29.54 | docelm0 | Typical woman |
| 14:29.59 | Katty | docelm0: obviously. |
| 14:30.12 | docelm0 | cruel and unusual... |
| 14:30.14 | Katty | docelm0: perhaps a smidgen smarter though ;) |
| 14:30.19 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl8p.dialup.mindspring.com) |
| 14:30.30 | iDunno | :) |
| 14:30.33 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfl8p.dialup.mindspring.com) |
| 14:30.45 | docelm0 | women == evil creatures... |
| 14:30.48 | Katty | mhmm |
| 14:30.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:31.13 | docelm0 | Trust me she looks nothing like Yogi... |
| 14:32.19 | iDunno | it's possibly because I read smarter and got "smarter than the average bear" in my head ;) |
| 14:32.23 | iDunno | and yes, I know :) |
| 14:33.02 | Katty | docelm0: you've never really seen me ya know |
| 14:33.13 | Katty | docelm0: for all you know, i pay off everyone i meet |
| 14:33.22 | Hmm-work | haha |
| 14:33.53 | x86 | SWEET! I got E.164 lookups working with this uber-macro :) :) :) |
| 14:34.15 | x86 | had the next macro in line mis-labeled and it was causing it to fuss |
| 14:34.42 | *** join/#asterisk Morak (n=my@217-18-88-204.bunting.cust.nseuk.net) |
| 14:34.43 | docelm0 | Katty, naa.. Damin is pretty grounded.. |
| 14:35.34 | Katty | who's damin? |
| 14:35.52 | docelm0 | Greg |
| 14:35.58 | wasim | curiosity killed the ... |
| 14:35.59 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:36.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:36.05 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:36.11 | frenzy | LOL |
| 14:36.44 | *** join/#asterisk apardo (n=apardo@87.218.44.228) |
| 14:36.45 | Katty | anthm: mew. |
| 14:37.15 | anthm | hi |
| 14:37.55 | x86 | whoa! e164.org lets you call toll-free numbers :) |
| 14:38.01 | x86 | via ENUM :) |
| 14:38.49 | Splat | can anyone point me to some good dialplan documentation.. I'm setting up a click-to-dial application.. but I want it to give a message when it calls you before it dials the other parties number.. if I just tell it to use my from-internal context it'll make the call.. but if I have a custom context that plays 'pls-wait-connect-call' it won't actually make the call.. so I need to work out how to fix it.. hehe |
| 14:39.36 | TinoW | iDunno: southpark-like? ;) |
| 14:39.57 | iDunno | :) |
| 14:40.12 | *** join/#asterisk cfh (n=luca@82.193.23.6) |
| 14:41.51 | cfh | hi all, i have a hfc card with 4 port on nt mode and i use bristuff / zaptel driver for asterisk |
| 14:42.45 | TinoW | cfh: *noted* |
| 14:43.01 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr2001187710.cable.net.co) |
| 14:44.12 | cfh | but the card dont see the isdn phone attached |
| 14:44.20 | *** join/#asterisk angler_ (n=johnb@199.227.185.58) |
| 14:45.36 | *** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-84-152.d-ip.magma.ca) |
| 14:45.41 | *** join/#asterisk dwmw2 (n=dwmw2@baythorne.infradead.org) |
| 14:47.58 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 14:49.27 | *** join/#asterisk diego_br (n=diego@200.208.241.178) |
| 14:51.38 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 14:51.43 | Morak | hi there... does anyone happen to know if you can put an image below the clock on a Polycom IP600 phone? |
| 14:53.04 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 14:53.40 | Katty | Morak: i've never tried. |
| 14:53.42 | *** join/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84) |
| 14:53.43 | SibRw0rk | hey |
| 14:54.18 | SibRw0rk | anyone know how to do rollover calls with asterisk? |
| 14:54.33 | wasim | SibRw0rk: yep, use a treat, like a piece of liver or something |
| 14:55.12 | SibRw0rk | waah waaah |
| 14:55.26 | [TK]D-Fender | Morak : Nope. |
| 14:56.15 | Morak | ok, thanks. ill try the idle image then, see if that will do. |
| 14:56.38 | SibRw0rk | [TK]D-Fender: sup dude |
| 14:57.42 | Morak | just a logo, sitting underneath (not behind) the clock. I know you can put a logo on the main screen of a Cisco 7960, just trying to do the same thing on the polycom. |
| 14:58.13 | [TK]D-Fender | SibRw0rk : Blargh |
| 14:58.18 | SibRw0rk | [TK]D-Fender: nice nice |
| 14:58.26 | SibRw0rk | do you know how to do call rollover? |
| 14:58.31 | SibRw0rk | if line 1 is busy, ring line 2? |
| 14:58.44 | rpm | you use dialplan Hints |
| 14:58.51 | SibRw0rk | ?? |
| 14:59.02 | [TK]D-Fender | SibRw0rk : Not sure exactly what you mean... there are 2 very different meanings for that |
| 14:59.10 | SibRw0rk | ok |
| 14:59.16 | [TK]D-Fender | SibRw0rk : on the PHONE level, or incoing line level? |
| 14:59.33 | SibRw0rk | dial exten 646850XXX1 and it's busy, so i want 646850XXX2 to ring |
| 14:59.50 | [TK]D-Fender | Those are EXTENSION #'s? |
| 14:59.56 | SibRw0rk | yeah |
| 15:00.00 | TinoW | bbl |
| 15:00.08 | [TK]D-Fender | EEK. why?! God-aweful long... |
| 15:00.17 | SibRw0rk | what they want |
| 15:00.22 | SibRw0rk | i only do what i'm told |
| 15:00.24 | wunderkin | the horror! |
| 15:00.36 | SibRw0rk | it's like a hunt group |
| 15:01.11 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 15:03.22 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 15:03.28 | SibRw0rk | [TK]D-Fender: any thoughts? |
| 15:03.52 | wasim | mlor |
| 15:04.27 | Katty | my favorite song is on mlor. |
| 15:04.40 | Katty | eric_hill: run along, silly rabbit. |
| 15:04.59 | eric_hill | You like my fluffy tail, don't you... :) |
| 15:05.14 | Katty | not when you have your head stuck in it. |
| 15:05.28 | Katty | ;) |
| 15:05.36 | eric_hill | Did you hear about the Bear and the Rabbit in the woods? |
| 15:05.51 | Katty | apparently i was too busy burning floyd albums. |
| 15:05.57 | eric_hill | The Bear says to the Rabbit, "do you ever have problems with shit sticking to your fur?" |
| 15:06.07 | eric_hill | The Rabbit replies, "No. Why?" |
| 15:06.26 | eric_hill | The Bear says "great!", picks the Rabbit up, and wipes his ass with the Rabbit. |
| 15:06.41 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 15:06.57 | Katty | you're definately a male. |
| 15:07.13 | qseek | hey everyone |
| 15:07.31 | eric_hill | I sure hope so. Otherwise I should have stolen a female identity. |
| 15:07.52 | Katty | iDunno: it's a floyd album, hun. |
| 15:07.55 | [TK]D-Fender | SibRw0rk : Thats very basic dial-plan stuff... look at the STDEXTEN macro sample in the WIKI for some inspiration. |
| 15:08.24 | [TK]D-Fender | Katty: mew. |
| 15:08.28 | SibRw0rk | i found something |
| 15:08.36 | SibRw0rk | exten => 1,1,Dial(SIP/001, 10) |
| 15:08.36 | SibRw0rk | <PROTECTED> |
| 15:08.36 | iDunno | Katty: *ah* :) |
| 15:09.08 | eric_hill | It's even a pink CD no less |
| 15:09.18 | [TK]D-Fender | Morak : While you can't put an image behind the clock, when you use the "idle" screen the clock goes into 1-line mode on the bottom. Good for displaying your logo / etc. |
| 15:09.31 | [TK]D-Fender | SibRw0rk : basically that works. |
| 15:09.47 | Katty | [TK]D-Fender: mew. |
| 15:10.18 | Katty | i still say the cover of ummagumma is the /hottttest/ |
| 15:10.31 | Katty | 1970 was a good look. |
| 15:10.32 | *** join/#asterisk livesNbox (n=livesNbo@user-12l2mrd.cable.mindspring.com) |
| 15:10.35 | livesNbox | Hey guys -- I'm having a few dropped calls today to some of my remote callers -- Should I do something to lower the amount of bandwidth is being used for their calls? Change the codec or? |
| 15:11.13 | [TK]D-Fender | Katty: Move is over, just a few odds and ends to pick up now. |
| 15:12.09 | iDunno | livesNbox: that's not entirely descriptive of the fault. |
| 15:12.18 | Katty | [TK]D-Fender: woo! |
| 15:12.45 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 15:12.49 | SibRw0rk | later |
| 15:13.13 | iDunno | livesNbox: remote callers could be SIP, IAX2, ISDN, etc etc... by the mention of "bandwidth" I assume that it's SIP or IAX2, so maybe changing the codec would help, but that's not garanteed... maybe there's a wibble of routing between you and $remote |
| 15:15.51 | livesNbox | it's sip -- but I more ment the vocoder codec |
| 15:15.54 | livesNbox | meant* |
| 15:16.01 | livesNbox | to reduce the bandwidth required.. |
| 15:16.05 | livesNbox | I'm not sure it's a bandwidth issue though |
| 15:17.28 | iDunno | if it's not a bandwidth issue, then changing the codec isn't going to help much :) |
| 15:17.29 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 15:17.31 | wasim | livesNbox: what codec are you using now? |
| 15:17.32 | [TK]D-Fender | livesNbox : What are they using now? |
| 15:17.43 | Morak | fender: Thanks... have made the changes, phone is rebooting now... just going to see if it works :) |
| 15:18.35 | livesNbox | PCMU I think.. |
| 15:18.46 | wasim | livesNbox: confirm it |
| 15:18.57 | [TK]D-Fender | livesNbox : and what kind of phone on the other side? |
| 15:19.03 | livesNbox | it's a grandstream phone |
| 15:19.11 | livesNbox | how do you confirm the codec in use? |
| 15:19.15 | livesNbox | my logs say "format for call is gsm" |
| 15:19.33 | x86 | err? |
| 15:19.40 | x86 | Katty: ummagumma? |
| 15:19.46 | [TK]D-Fender | livesNbox : GSM is already pretty light.... |
| 15:19.59 | x86 | Katty: Pink Floyd is my most favorite band ever :) |
| 15:20.09 | x86 | Katty: i have all of their albums minus maybe 2 |
| 15:20.14 | livesNbox | [TK]D-Fender ok -- maybe that's not really the problem then |
| 15:22.06 | Katty | x86: yes, the cover of ummagumma |
| 15:22.11 | Katty | x86: dreamy hot. |
| 15:22.15 | x86 | heh |
| 15:24.51 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:25.16 | Katty | everyone should grow their hair out. |
| 15:25.50 | *** join/#asterisk Immosky (n=chatzill@62-2-138-202.business.cablecom.ch) |
| 15:26.22 | eric_hill | Katty: Just what the world needs -- me with a mullett. Yikes! |
| 15:26.25 | Immosky | hello everybody |
| 15:27.39 | Katty | eric_hill: longer than a mullet, kthx. |
| 15:27.46 | Katty | eric_hill: we're talking /at least/ chin length. |
| 15:27.57 | Katty | eric_hill: shoulder would be infinately better. |
| 15:28.22 | Hmm-work | god I hate it when this company pawns me off on support |
| 15:29.10 | Immosky | i have a small problem with the german soundfiles. i know that there is this problem with the file 1F.gsm. i renamed the file "eine.gsm" into 1F.gsm - not working, i made a link - not working, i copied the renamed file into ervery folder of asterisk - not working. is there another file wich is missing? |
| 15:32.38 | Immosky | btw using asterisk 1.2 |
| 15:32.49 | iDunno | Katty: hmm - nah - long hair suits a subset of people - most females, f'rinstance... not convinced that most blokes can pull it off ;) |
| 15:33.07 | Katty | hmm, true. |
| 15:33.14 | Hmm-work | ahhh my episode of prison break is turning out nicely |
| 15:33.20 | Katty | but i've yet to date a guy with short hair. |
| 15:33.23 | [TK]D-Fender | Katty : I'm working on that. the top is getting better now, but eh sides are annoying me. |
| 15:33.24 | Hmm-work | Katty: so how'd stairway go? |
| 15:33.25 | Katty | that's fursure. |
| 15:33.31 | Katty | Hmm-work: i'm still picking it out. |
| 15:33.45 | Katty | Hmm-work: it gets all nuts when the flute line becomes the chord (right hand) and the base becomes the melody (left hand) |
| 15:34.06 | Katty | Hmm-work: it's like playing piano backwards >.< |
| 15:34.16 | Hmm-work | yeah I didn't look any farther into into the first few measures of the bass line |
| 15:34.19 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 15:34.31 | iDunno | Katty: hmmm. is that through deliberate choice, though? :) |
| 15:34.36 | Katty | it's pretty repetative, but i'm just trying to get my left hand to work :P |
| 15:34.44 | Katty | iDunno: what's that? |
| 15:35.19 | Hmm-work | yeah getting that left hand to move really sucks some days, makes me want to throw my guitar out the window |
| 15:35.28 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:35.30 | iDunno | 16:33 < Katty> but i've yet to date a guy with short hair. |
| 15:36.10 | Katty | on purpose |
| 15:36.27 | Katty | it's long hair, or i'm dating a girl. |
| 15:37.15 | blitzrage | I like dating girls too -- we have something in common! :) |
| 15:37.23 | Immosky | please!!! anybody can't help me? |
| 15:37.32 | Katty | blitzrage: ;) |
| 15:37.34 | iDunno | Katty: hmmmm :) |
| 15:37.36 | blitzrage | sure! I'll do my best to not help :) |
| 15:37.48 | Immosky | thanks |
| 15:38.07 | Katty | speaking of dating. |
| 15:38.10 | Katty | Hmm-work: how's the chica? |
| 15:38.22 | Katty | Hmm-work: being miss drama queen of the world? |
| 15:38.28 | blitzrage | yah-- I haven't dated for what seems like forever... and I'm not a scary looking guy either :) |
| 15:39.04 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 15:39.23 | SwK[Work] | i havent had a date in like 6 years |
| 15:39.29 | blitzrage | jeezus! :) |
| 15:39.42 | blitzrage | SwK[Work]: we gotta get out more :) |
| 15:40.48 | SwK[Work] | blitzrage: still trying to convince my wife to let me date |
| 15:40.50 | SwK[Work] | heh |
| 15:41.48 | blitzrage | lol |
| 15:41.52 | blitzrage | oh I see :) |
| 15:42.08 | blitzrage | yah -- I don't have a good excuse like that :) |
| 15:42.23 | blitzrage | my excuse is that I work from home, and I just haven't been able to meet the right woman in my bedroom, lol |
| 15:42.29 | Katty | SwK[Work]: yeah..but i know why you don't have any dates |
| 15:42.31 | blitzrage | or even the wrong eoman :D |
| 15:42.40 | Katty | SwK[Work]: you're not a pretty sight at 8am ;) |
| 15:42.44 | hanchi | Is there a softphone, other than iaxRpt, that is compatible with app_rpt |
| 15:42.45 | *** join/#asterisk rene- (n=rene-@201.127.10.16) |
| 15:42.54 | SwK[Work] | katty: you weren't complaining ;) |
| 15:43.02 | rene- | hello all |
| 15:43.11 | Katty | SwK[Work]: ;_ |
| 15:43.13 | Katty | SwK[Work]: ;) |
| 15:43.19 | Katty | SwK[Work]: i just wanted my visa back, heh |
| 15:43.24 | Immosky | hanchi: so you are a grandmaster? |
| 15:43.48 | rene- | i wanted to say that the latest sip image (8.2) is super easy to install and doesnt requires a cco. |
| 15:44.00 | SwK[Work] | katty: thats what you say now... I'm sure you had other motives |
| 15:44.03 | rene- | i meant the latest cisco image |
| 15:44.33 | *** join/#asterisk bweschke (n=bweschke@198.sub-70-192-244.myvzw.com) |
| 15:44.42 | Katty | SwK[Work]: i might have had other motives if you were wearing armani |
| 15:44.49 | Katty | SwK[Work]: but that definately /wasn't/ armani |
| 15:44.51 | hanchi | ???....No |
| 15:45.23 | iDunno | errr. |
| 15:45.25 | SwK[Work] | katty: what was I wearing? spongbob boxers? |
| 15:45.29 | Katty | SwK[Work]: yes. |
| 15:45.29 | rene- | i also have a question regarding g729, i am using the free binaries of g729 available in the net, i have already ordered codecs from digium, should i expect a quality increase in my voice calls when i switch the free codecs for digium codecs, all other things equal? |
| 15:45.45 | *** part/#asterisk cfh (n=luca@82.193.23.6) |
| 15:45.45 | bkw__ | no |
| 15:45.52 | bkw__ | you'll gain nothing but a license |
| 15:45.53 | Katty | bkw__: no? |
| 15:45.57 | Immosky | hanchi: well in karate the grandmaster of a style is called "hanchi".... |
| 15:46.03 | Katty | oh. |
| 15:46.03 | hanchi | I need to link 24 two way radios to a single dispatch console with patching capabilites, and tie it to * |
| 15:46.07 | Katty | i like hugs. |
| 15:46.24 | Katty | ... |
| 15:46.27 | hanchi | that is usually hanshi, hanchi is a title, but also a last name |
| 15:46.28 | Katty | bkw__: what do you want? |
| 15:46.41 | bkw__ | Katty, waiting on mr. mike to finish up contracts on hotel for Cluecon |
| 15:46.42 | bkw__ | :P |
| 15:46.47 | bkw__ | then we can start registration |
| 15:47.27 | rene- | i am experiencing small cuts in the calls, with one call in an otherwise unused 1300/512 link, i could of course blame the itsp but what can i do to improve the call quality? |
| 15:47.41 | Immosky | sorry my fault! |
| 15:48.24 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 15:48.27 | Katty | bkw__: excellent. |
| 15:48.29 | hanchi | np |
| 15:48.30 | [TK]D-Fender | rene- : you using SIP or IAX? IAX trunking can save on bandwidth |
| 15:48.42 | Katty | bkw__: you going to have someone meet me at the amtrak station? |
| 15:49.02 | docelm0 | Nope.. They are gonna leave you hanging |
| 15:49.06 | hanchi | any ideas on a softphone with more features than iaxRpt, that is app_rpt compatible??? |
| 15:49.12 | Katty | docelm0: horrors. |
| 15:49.18 | rene- | [TK]D-Fender: right now SIP i intend to sit an IAX box at the ITSP premises |
| 15:49.19 | Katty | the streets of chicago are no place for a Kat. |
| 15:49.34 | docelm0 | MEW MEW MEW! MEW MF-R MEW! |
| 15:49.46 | Katty | ... |
| 15:49.55 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:49.56 | rene- | but right now it is plain SIP+g729 |
| 15:50.23 | Immosky | does anybody know something about the error with the german 1F.gsm file? |
| 15:50.46 | hanchi | Also, does anyone know of a 911 psap using *??? |
| 15:50.56 | bkw__ | hanchi, no if they were I'm sure they would get sued |
| 15:51.07 | bkw__ | I wouldn't even trust asterisk with my life.. |
| 15:51.14 | Katty | mishehu: you'll come get me. |
| 15:51.31 | bkw__ | Katty, where is the train station? |
| 15:51.34 | hanchi | not for the whole 911 gear for the psap, just the CTI |
| 15:51.42 | bkw__ | hanchi, still no trust there |
| 15:52.15 | Katty | bkw__: it's downtown, not far from the sear's tower. |
| 15:52.22 | bkw__ | Katty, kewl |
| 15:52.35 | Katty | bkw__: i just don't wanna be there alone. |
| 15:52.44 | Katty | bkw__: some guy walk past me and i'll flip out. |
| 15:52.46 | Immosky | well the problem is that asterisk is not able to find the file, and there are solutions (making link or rename anohter file) but its not working on my asterisk |
| 15:54.05 | hanchi | we have a stand alone 911 system with the CML Patriot VOIP package, the system forwards all ALI/ANI data from the calls to the CAD(computer aided dispatch), one of the vendors at APCO told us one user had linked this to the * pbx in the building, to pick up the 911 call on * |
| 15:54.24 | bkw__ | Immosky, show us how you're trying to play the file |
| 15:54.24 | mishehu | Katty: just got here, what plans are we discussing here? |
| 15:54.26 | bkw__ | chances are you're doing it wrong |
| 15:54.34 | bkw__ | hanchi, what country are you in? |
| 15:54.39 | hanchi | USA Texas |
| 15:54.54 | bkw__ | I hope I never have to call 911 in your town |
| 15:54.54 | mishehu | Texas is definitely another country. yee haw. |
| 15:55.09 | x86 | everyone should sign up for e164.org, it's free |
| 15:55.10 | Katty | mishehu: i need an escort from amtrak to cluecon so i don't freak out. |
| 15:55.45 | Immosky | well easy, its part of the voicemail function i do not do anything, it occours when you dial the voicemail and only if there is only one message in the folder.... |
| 15:55.51 | mishehu | bkw__: with a limo and a placard? |
| 15:55.58 | Katty | bkw__: yes'm, i shall need fetching. |
| 15:56.07 | hanchi | the 911 system is bullet proof here, can be picked up at the CTI terminal, or on the key pbx, currently in place, meets all NEBA 5 requirement. I'm only looking for a link to *, when we move to new facility under construction |
| 15:56.26 | mishehu | Katty: depends on where cluecon is. ;-) |
| 15:57.08 | Katty | mishehu: that one place. |
| 15:57.12 | Katty | mishehu: ya know, with the curtains. |
| 15:57.17 | rene- | can an 80ms ping time between communicating sites be a factor for experiencing bad voice quality? |
| 15:57.34 | mishehu | Katty: the curtain store? |
| 15:57.37 | x86 | Katty: go sign up for E.164 ;) |
| 15:57.41 | Katty | mishehu: they have those? |
| 15:57.45 | mishehu | it's coitains for you, rocky |
| 15:57.46 | Katty | mishehu: i wanna go! |
| 15:57.48 | mishehu | coitains |
| 15:57.53 | Immosky | bkw_:did that help? |
| 15:57.55 | *** join/#asterisk bweschke (n=bweschke@124.sub-70-195-244.myvzw.com) |
| 15:58.00 | Hmm-work | Katty dear can you do me a favor? |
| 15:58.06 | Katty | Hmm-work: probably. |
| 15:58.20 | bkw__ | x86 shut up about e164 |
| 16:00.46 | Morak | hoping someone can help here.... trying to set up idle screen on polycom ip600 and having no joy. Created 208x110 4BBP image and put it in TFTP root and modified the sip.cfg file as per the voip-info instructions, but no image appears when i reboot phone :( |
| 16:00.58 | Morak | 4BBP = 4BPP |
| 16:01.14 | Splat | can anyone tell me how I can have a context that will play a message and then follow all the normal dialplan routing? or point me to some documentation that will tell me? |
| 16:01.25 | mishehu | *sqwak* Polly Com *whistle* |
| 16:01.33 | x86 | bkw_: sign up for it and i will ;) |
| 16:01.46 | *** join/#asterisk salviadud (n=ralfalfa@201.138.132.150) |
| 16:01.48 | ManxPower | Splat, "show application playback" |
| 16:01.55 | bkw__ | x86, um no |
| 16:02.00 | `Sauron | splat: have the sip.conf drop it into a context that plays the announcement, then drops it into the "normal" dialplan start... |
| 16:02.10 | bkw__ | rene-, SIP? |
| 16:02.11 | mishehu | and it's a smelly one, I assure you. |
| 16:02.28 | rene- | bkw_: yes |
| 16:02.31 | Katty | iDunno: i hear muffinery works well. |
| 16:02.38 | Splat | I have the playing the announcement fine.. but not continuing on as normal afterwards.. |
| 16:02.54 | bkw__ | rene-, it could be jitter... since asterisk has NO jitter buffer on RTP |
| 16:03.27 | rene- | it does sounds like jitter, now that you say it |
| 16:03.28 | ManxPower | Splat, That is impossible. |
| 16:04.09 | rene- | i will try with the asterisk box sitting at the ITSP first |
| 16:04.18 | *** join/#asterisk GuruDom (n=domiplus@66-202-165-66.rev.knet.ca) |
| 16:04.31 | rene- | i first had a problem with one of the phones using vad and that gave me terrible audio quality |
| 16:04.35 | jbalcomb | What does everyone have thier port range set to in rtp.conf? |
| 16:04.38 | mishehu | I'm sooo exhausted. :-/ |
| 16:04.44 | Katty | mishehu: go nap. |
| 16:04.53 | mishehu | Katty: class in 15 minutes :-/ |
| 16:04.54 | ManxPower | Splat, http://pastebin.ca/47308 |
| 16:05.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:05.20 | ManxPower | jbalcomb, 16384 - 16393 |
| 16:05.39 | ManxPower | that should allow me about 5 calls |
| 16:06.02 | Katty | mishehu: :<< |
| 16:06.19 | mishehu | Katty: last week I was on spring break, it felt SOOO good to not have to get up at the crack of dawn. |
| 16:06.19 | Katty | mishehu: nap in class! |
| 16:06.24 | blitzrage | Hmm-work: air drumming rocks |
| 16:07.06 | iDunno | Katty: really? I was thinking that a snake could help with this problem ;) |
| 16:07.17 | Splat | ManxPower: the problem there is specifying what it's to dial out through.. and I use pstn for some calls (1800, 1300, 13, 000) and voip for other calls.. specifying the outgoing trunk means that it can't follow my call routing.. |
| 16:07.18 | Katty | iDunno: snake? |
| 16:07.22 | rene- | i have heard that air drumming is one of the hardest test to get into american idol, but it wasnt a very reliable source so dont bet your life on that |
| 16:07.37 | ManxPower | Splat, Huh? |
| 16:07.53 | salviadud | wasim, i'm still in school |
| 16:08.12 | salviadud | and i just wanna graduate... |
| 16:08.18 | ManxPower | My example is NOT technology specific. |
| 16:08.34 | ManxPower | my example simply shows you how I do it. |
| 16:08.46 | x86 | ManxPower: have you setup E164 yet? |
| 16:08.50 | Splatty47 | After I install asterisk, is their some type of GUI that I can get running so that I can configure everything easily ? |
| 16:09.04 | ManxPower | x86, no, and I doubt I ever will. |
| 16:09.04 | wasim | Splatty47: vi in an xterm |
| 16:09.09 | x86 | ManxPower: why? |
| 16:09.11 | Splatty47 | without having to understand all the confs ? |
| 16:09.21 | Splatty47 | wasim: not quite what I meant. |
| 16:09.32 | x86 | ManxPower: save a bunch on outbound calls with it, and people who call you can save lots too |
| 16:09.37 | ManxPower | x86, why should I. |
| 16:09.49 | ManxPower | x86, I spend $10/month on calls. |
| 16:09.50 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 16:09.50 | wasim | Splatty47: the sooner you get to understanding the .confs the less you'll need to badger folks here |
| 16:10.06 | x86 | ManxPower: what about people that call you? |
| 16:10.16 | ManxPower | x86, what about them? |
| 16:10.17 | Splatty47 | wasim: have you used asterisklive (cd) b4 ? |
| 16:10.26 | wasim | Splatty47: no sir |
| 16:10.51 | iDunno | Katty: python :) |
| 16:11.05 | Katty | iDunno: python's dreamy. |
| 16:11.10 | x86 | ManxPower: they could save money! :P |
| 16:11.11 | blitzrage | boooo python |
| 16:11.12 | blitzrage | PHP! |
| 16:11.16 | Katty | blitzrage: you shush. |
| 16:11.16 | iDunno | Katty: indeed - it's a lovely language :) |
| 16:11.20 | blitzrage | Katty: no way! :) |
| 16:11.21 | ManxPower | perhaps you don't realize just how BAD VoInternet can be. |
| 16:11.25 | iDunno | blitzrage: you're broken ;) |
| 16:11.25 | Katty | blitzrage: eat that, hush :P |
| 16:11.26 | blitzrage | Katty: thanks! :) |
| 16:11.29 | Hmm-work | word up , everybody say, when you hear the call you got to get it underway |
| 16:11.29 | x86 | PHP == Pile of Hyped up Poop |
| 16:11.36 | jbalcomb | I've just finished setting up a new asterisk 1.2.5 server and when i try to make a call i'm getting no audio. |
| 16:11.37 | x86 | Perl++ |
| 16:11.41 | Katty | perl's also nice. |
| 16:11.46 | blitzrage | PHP does everything i Need it to, and its easy, and doesn't look as crappy as python :) |
| 16:11.51 | Hmm-work | I detect a holy war coming on |
| 16:11.55 | x86 | Katty: you've setup E164? :) |
| 16:11.55 | blitzrage | perl is ok... its pretty "hacky" though :) |
| 16:11.57 | ManxPower | x86, Why do I care if they save money? There is not a single one of them that can use e164 |
| 16:12.07 | blitzrage | I'm instantiating a programming jihad |
| 16:12.30 | blitzrage | but hey -- use whatever works, thats really all it comes down to -- PHP is EASY for a non-programmer like me :D |
| 16:12.42 | x86 | blitzrage: you do realize php was originally written in perl, right? :P |
| 16:12.52 | x86 | way back in the day |
| 16:12.54 | blitzrage | plus I can use it in a web environment, AGI, or CLI to generate configs -- and its got good DB support |
| 16:13.03 | blitzrage | x86: hrmmm... wonder why its not anymore :D |
| 16:13.08 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 16:13.08 | Hmm-work | really big hair used to be really cool too |
| 16:13.13 | blitzrage | Hmm-work: LOL |
| 16:13.15 | x86 | blitzrage: PHP doesnt even have real data structures ;) |
| 16:13.17 | GuruDom | Anyone have any Zptel probs with freebsd? |
| 16:13.29 | blitzrage | x86: we don't need no stinkin' data structures! :) |
| 16:13.36 | x86 | :P |
| 16:13.38 | blitzrage | GuruDom: uhh.. yah -- it doesn't work on freebsd |
| 16:13.40 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 16:13.42 | blitzrage | afaik |
| 16:13.54 | x86 | i'm all about data organization... something i can't do in PHP... one of the major turnoffs for me |
| 16:14.09 | ManxPower | x86, for anyone without a massive volume of calls e164 is more trouble than it's worth |
| 16:14.14 | ManxPower | calls are 2 cents/min. |
| 16:14.25 | ManxPower | I charge $120/hr. you do the math |
| 16:14.27 | Katty | x86: ummmm, no., |
| 16:14.30 | x86 | blitzrage: i will give PHP the speed bonus though, but it's aimed more for web stuff... I make a lot of console applications with Perl |
| 16:14.33 | pigpen2 | Hi all...when checking voicemail, one of the options is to hit # to exit....it does a ringback to the extension I am on ...how can I fix this?? (* ver 1.2.4) |
| 16:14.38 | brookshire | x86: that's what pear is for! |
| 16:14.46 | Katty | mm, pear. |
| 16:14.50 | Katty | with /chocolate/ |
| 16:14.54 | ManxPower | pigpen, did you read "show application dial" |
| 16:14.57 | brookshire | or adodb :) |
| 16:15.05 | x86 | ManxPower: it took me a little under an hour to set it up :) |
| 16:15.20 | ManxPower | x86, so I would need to save a little over $100. |
| 16:15.23 | rene- | ruby is cute http://www.poignantguide.net/ruby/ |
| 16:15.26 | pigpen2 | ManxPower, I thought so..but I will do it again.... |
| 16:16.12 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 16:16.23 | blitzrage | x86: yah -- I built my first big program in Perl -- its not terrible, just really easy to write unmaintainable code if you're not a good programmer (although thats really possible in ALL languages -- just find it a bit easier to do it with Perl) |
| 16:16.40 | bkw__ | blitzrage, thats true for any language |
| 16:16.42 | bkw__ | not just perl |
| 16:16.44 | pigpen2 | ManxPower, no..I didn't ...doing it now. |
| 16:16.49 | blitzrage | didn't I JUST say that? |
| 16:16.57 | brookshire | he did just say that |
| 16:16.57 | bkw__ | sorry doing 10 things at once :P |
| 16:17.05 | Aurs | so.. how does this realtime queues thing work? |
| 16:17.08 | rene- | i find easier to forget stuff in perl than in other languages, |
| 16:17.19 | blitzrage | bkw__: be careful -- you're screwing things up :) |
| 16:17.41 | bkw__ | perl, c, php and javascript all are about 90% the same |
| 16:17.42 | Katty | Hmm-work: i've ripped through almost 50 albums |
| 16:18.01 | blitzrage | I like PHP because its easy and I know it :) |
| 16:18.22 | blitzrage | doesn't mean other languages aren't good, but PHP does everything i need it to do |
| 16:18.42 | bkw__ | I'm going to say one thing... php needs to stay on the web where it belongs :P |
| 16:18.44 | rene- | bkw_ but that is a realization you have after having worked with all of them, for me C started to make sense after i learned a bit of ruby, when i did perl C seemed very foreign to me |
| 16:19.03 | rene- | it is a very good thing to switch languages |
| 16:19.34 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 16:19.52 | MikeJ[Laptop] | bkw__, you swing both ways? |
| 16:20.00 | rene- | heh |
| 16:20.01 | MikeJ[Laptop] | perl and C? |
| 16:20.10 | jbalcomb | my 'sip show settings' has Codecs: none. Is this a problem and how can I fix it? |
| 16:20.48 | bkw__ | jbalcomb, allow some codecs :P |
| 16:20.59 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 16:21.29 | jbalcomb | bkw_ I have 'allow=ulaw' in my sip.cong |
| 16:21.41 | MikeJ[Laptop] | jbalcomb, in general? |
| 16:21.45 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 16:22.56 | GuruDom | T1/E1 digium/sagnoma cards use Zaptel or Libpri? |
| 16:23.03 | jbalcomb | MikeJ[Laptop] it seems not, just in the user definition. i didn't imagine the sample config being so dysfunctional. I'm unremarking and reloading. |
| 16:23.15 | Immosky | bye |
| 16:23.26 | [TK]D-Fender | GuruDom : Both |
| 16:23.26 | GuruDom | ah |
| 16:23.32 | MikeJ[Laptop] | GuruDom, huh? |
| 16:24.00 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 16:24.02 | GuruDom | not good, i want to use freebsd with my * servers that use PRI's |
| 16:24.12 | [TK]D-Fender | bkw_ : I am a bilingual illiterate.... I can't read in 2 languages ;) |
| 16:24.19 | GuruDom | and if the Zaptel drivers dont work then not good |
| 16:24.31 | ManxPower | GuruDom, I'm glad to see you have a rich fantasy life. |
| 16:24.35 | jbalcomb | ok, no the ulaw codec is listed in sip settings but i still don't have any audio. |
| 16:24.47 | [TK]D-Fender | GuruDom : No you don't.... Zaptel on BSD's is inviting pain. It can be done, but I'd suggest just using Linux. |
| 16:24.50 | ManxPower | GuruDom, Zaptel drivers only officially run on Linux. |
| 16:25.14 | GuruDom | Specifically why doesnt it run with bsd? |
| 16:25.30 | jbalcomb | because its not made to? |
| 16:25.40 | GuruDom | im a bsd fan cause its uber stable |
| 16:26.08 | Dr-Linux | please, look at my urgent question |
| 16:26.09 | Dr-Linux | http://pastebin.com/627066 |
| 16:26.26 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 16:26.53 | MikeJ[Laptop] | GuruDom, sangoma has a seperate API that doesn't use zaptel. |
| 16:26.57 | [TK]D-Fender | Dr-Linux : Try the "M()" option and have it do a wait then playback DTMF |
| 16:27.00 | Hmm-work | man this riff that starts of modern day cowboy is kicking my @$$ |
| 16:27.02 | GuruDom | Now when we are talking Zaptel on BSD are you considering the actual Zaptel drivers made for BSD on freshports.org? |
| 16:27.13 | MikeJ[Laptop] | would take some dev work to get it up to speed for asterisk |
| 16:27.37 | MikeJ[Laptop] | the other option would be to fix the zaptel issues on bsd |
| 16:27.45 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 16:28.42 | MikeJ[Laptop] | I am generally a cross platform kind of guy... so I lean towards making drivers that are not linux specific. |
| 16:30.35 | jbalcomb | [TK]D-Fender i added a user '2001' to sip.conf and voicemail.conf which registers fine. shouldn't i be able to call 8500 on a fresh install and get the VM system atleast? |
| 16:30.55 | pigpen2 | ManxPower, ok...I read it over...but it doesn't state anything regarding the # to disconnect in voicemail...I don't think this has anything to do with the dial command. |
| 16:30.55 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 16:30.56 | GuruDom | Has anyone atually tried the Zaptel patch for bsd availble on freshports.org? |
| 16:31.20 | jbalcomb | [TK]D-Fender or even just 500 for the demo? |
| 16:31.36 | [TK]D-Fender | jbalcomb : The default sample file= poo. |
| 16:31.45 | [TK]D-Fender | jbalcomb : I don't trust that is does ANYTHING. |
| 16:32.03 | jbalcomb | [TK]D-Fender i'm seeing this. i don't know that i know enough to wipe them and write my own from scratch |
| 16:32.09 | [TK]D-Fender | jbalcomb : For details as to what could/should work, you'd have to pastebin.... |
| 16:32.20 | pigpen2 | So, I will post my issue again: when checking voicemail, one of the options is to hit # to exit....it does a ringback to the extension I am on ...how can I fix this?? (* ver 1.2.4) |
| 16:33.08 | jbalcomb | [TK]D-Fender ok, the samples are too long and crazy so i'm going to try scratching them then and if it still doesn't work i'll pastebin |
| 16:34.34 | [TK]D-Fender | jbalcomb : Here, start with this : http://pastebin.com/627082 |
| 16:35.21 | jbalcomb | [TK]D-Fender ok, thanks. |
| 16:36.48 | *** join/#asterisk b66mer (i=fwuser@blackhole.c5i.com) |
| 16:37.43 | b66mer | anybody tell me what macro-stdPrivacyexten is? why use it versus macro-stdexten? |
| 16:38.32 | ManxPower | pigpen2, sorry, I I had not finished my first cup of coffee. "show application voicemailmain" |
| 16:38.39 | ManxPower | or "show application voicemail" |
| 16:39.26 | pigpen2 | ManxPower, ok..I was trying to figure that one out...but from what you have done for me in the past...I gave you the benefit of the doubt. |
| 16:39.31 | pigpen2 | Thanks. |
| 16:40.59 | *** part/#asterisk yuta-vcnet (i=yuta-vcn@212.118.246.50) |
| 16:43.33 | [TK]D-Fender | b66mer : read what it does. things explain themselves. The sample extensions.conf file is a worthless pile of junk. |
| 16:43.37 | pigpen2 | ManxPower, ok...sorry, but still no luck (or I am really dense this morning and needing more coffee myself) |
| 16:44.14 | ManxPower | pigpen2, # exits you from voicemail and the call will then continue on the next priority of the dialplan |
| 16:45.35 | b66mer | member:identifier:[tk]d-fender: I am writing my own... using that one as a guide... but couldn't figure out why that Privacy one was there... is there a web resource someone might guide me to for helping construct a professional dialplan? |
| 16:45.40 | pigpen2 | ah.....hmm.. |
| 16:45.41 | pigpen2 | thanks. |
| 16:45.50 | pigpen2 | sounds like I need a "Hangup" |
| 16:47.14 | [TK]D-Fender | Hey, I've got a rogue value in the ASTDB I'm trying to kill off from the CLI but its a 3-level one whose format for removal I'm not sure of. can someone lend a hand? "/belanger/Agent/8945 : NOTFOUND |
| 16:47.46 | *** join/#asterisk Chaosmonkey (n=jon@c-71-199-252-53.hsd1.fl.comcast.net) |
| 16:47.54 | jbalcomb | [TK]D-Fender the section in there [mainmenu] should be in place of [default] from the samples right? |
| 16:48.00 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 16:48.15 | Chaosmonkey | hey guys |
| 16:48.23 | [TK]D-Fender | jbalcomb : you should never have a context so genrically named as [default]. |
| 16:48.52 | Chaosmonkey | have my asterisk server all setup but need to know how to forward to a voip line if my fxs line is busy |
| 16:48.53 | jbalcomb | [TK]D-Fender understandable. so in my sip.conf and voiemail.conf for the extensions it should be 'mainmenu' |
| 16:49.13 | b66mer | buhler? any good resources for professional dialplan construction? |
| 16:49.13 | [TK]D-Fender | jbalcomb : [mainmenu] is the starting context of a quick menu and should never receive calls directly. look at the [incoming] context above it. |
| 16:49.38 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:49.46 | [TK]D-Fender | jbalcomb : no, your phones should be in context [myphones] becasue that one defines what a phone can DO. |
| 16:50.18 | jbalcomb | [TK]D-Fender ok, so in sip.conf i should put context=myphones? |
| 16:50.20 | [TK]D-Fender | jbalcomb : Keep in mine that that context INHERITS the access of 3 others. |
| 16:50.25 | [TK]D-Fender | jbalcomb : correct. |
| 16:50.31 | *** join/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 16:50.35 | jbalcomb | [TK]D-Fender ok, got it. thanks. |
| 16:50.53 | [TK]D-Fender | jbalcomb : Then you add more phones to [internal] and away you go. |
| 16:51.42 | Chaosmonkey | anyone have any ideas on how to do that? |
| 16:51.59 | jbalcomb | [TK]D-Fender is it a coincidence that we both chose 2001 as our sample/test extension? |
| 16:52.23 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
| 16:52.31 | *** join/#asterisk ghotiboy1 (n=ghotiboy@24-176-0-219.dhcp.klmz.mi.charter.com) |
| 16:52.39 | ghotiboy1 | good day |
| 16:52.43 | [TK]D-Fender | jbalcomb : I modded it that way expressly for you :) |
| 16:53.00 | ghotiboy1 | i have a question about using an spa3000 and fax detection |
| 16:53.07 | jbalcomb | [TK]D-Fender you are a sweetheart ;) |
| 16:53.24 | Darwin35 | ok who was I talking to this am |
| 16:53.25 | ghotiboy1 | actually it is more of just how to direct calls to the correct extension |
| 16:53.33 | [TK]D-Fender | jbalcomb : I am indeed the best thing since sliced bread |
| 16:53.53 | mut | yesssssss |
| 16:53.54 | mut | http://www.impactlab.com/modules.php?name=News&file=article&sid=7761 |
| 16:54.21 | ghotiboy1 | i have asterisk all setup correctly and my spa3000 sends calls to asterisk, but it doesn't come into the right context (seemingly) |
| 16:55.18 | jbalcomb | [TK]D-Fender custom/yourcalleridis does not exist. is there another sounds package for asterisk i'm missing? i only have 196 files so it seems like it |
| 16:55.38 | [TK]D-Fender | jbalcomb : No, you need to make that :) use the *40 feature. |
| 16:55.58 | jbalcomb | [TK]D-Fender ah |
| 16:56.13 | [TK]D-Fender | jbalcomb : and make a folder called "custom" in the sounds folder to put your stuff... |
| 16:56.46 | *** join/#asterisk mikeyb_work (n=michael@66-193-82-211.gen.twtelecom.net) |
| 16:57.20 | [TK]D-Fender | jbalcomb : And you'll need to make all the stuff in "custom" as referred to by my sample. |
| 16:57.57 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 16:58.35 | kimosabe | is it difficult 2 get a t-1 going with a carier for did purposes ? |
| 16:58.42 | [TK]D-Fender | jbalcomb : this sample I've given you covers the widest range of essential things to learn for dialplans : context inheritance, pattern matches, IVR, macros, and phones "features" |
| 16:59.06 | [TK]D-Fender | kimosabe : Depends where you are, but the actual process isn't hard. |
| 16:59.20 | [TK]D-Fender | kimosabe : We suggest getting a T1 PRI |
| 17:00.30 | kimosabe | is that what i shouyld request getting from sbc ? |
| 17:00.36 | Chaosmonkey | should i redirect to voip line in extensions.conf if busy |
| 17:02.11 | [TK]D-Fender | kimosabe : Yes. |
| 17:02.38 | [TK]D-Fender | Chaosmonkey : Try rephrasing your whole question...... |
| 17:02.49 | jbalcomb | [TK]D-Fender guess i need to make 'pleaserecordafterbeep' first. ;) |
| 17:04.20 | [TK]D-Fender | jbalcomb : Would be ironic if you left that for last :) |
| 17:05.20 | eric_hill | I got a call from one of our sites just now that said voicemail wasn't working. They dial into the system, hear a pre-recorded greeting, then press 1 for a mailbox... |
| 17:05.45 | [TK]D-Fender | lunch time! |
| 17:05.54 | Chaosmonkey | ok... I have a Asterisk server with a Digium TDM400P[1 FXO / 1FXS]... I have everything setup properly... Using SIP for VOIP IN/OUT and Digium Card for In/Out (pending Dialing prefix). The only problem Im having is when some dials the POTS line that the Digium Card picks up while someone else is already speaking on it. it will just ring and ring and continue to ring. I need to know how to fix this issu (by redirecting it to a voip l |
| 17:05.55 | eric_hill | ...I asked what they were expecting besides the "beep" and they said "it didn't tell us to record anything, so we didn't. It's broken". |
| 17:06.06 | eric_hill | Sigh. |
| 17:06.18 | *** join/#asterisk Assid (n=assid@203.115.64.8) |
| 17:06.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 17:07.16 | Darwin35 | NANPA sucks |
| 17:08.12 | jbalcomb | [TK]D-Fender ok, according to what i'm seeing reported in the CLI the calls are working perfectly and i got my recordings, however... |
| 17:08.14 | Darwin35 | you should be able to set he Vertical service codes to what ever you want |
| 17:08.25 | jbalcomb | [TK]D-Fender I still don't have audio. :(.. |
| 17:09.04 | MikeJ[Laptop] | Chaosmonkey, don't have somthing else use the same line you have plugged in to your fxo |
| 17:10.06 | Chaosmonkey | people call in using the fxo line |
| 17:10.17 | Chaosmonkey | but its only alowing one call |
| 17:10.46 | salviadud | how are you recording? |
| 17:10.46 | jbalcomb | [TK]D-Fender i'm not seeing any trouble reported anywhere on the cli or in the logs and i am uncertain how to troubleshoot this problem. |
| 17:10.49 | salviadud | mixmonitor |
| 17:10.51 | salviadud | monitor? |
| 17:11.04 | jbalcomb | salviadud me? |
| 17:11.12 | salviadud | yeah jbalcomb |
| 17:11.27 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 17:11.34 | Chaosmonkey | Im sorry, local people call in using the FXO line and Distant people call in using the Voip |
| 17:12.06 | Chaosmonkey | my only problem is i need i want the FXO line to redirect if its already currently in use |
| 17:12.12 | jbalcomb | salviadud well, its not the recordings i'm having problems with actually it's that my calls dont have audio |
| 17:12.41 | salviadud | what type of channels are we talking about here? |
| 17:12.52 | jbalcomb | salviadud i'm thinking this would be some issue with RTP settings or traffic but i dunno |
| 17:13.15 | ManxPower | jbalcomb, then you have either a firewall or a nat problem |
| 17:13.15 | jbalcomb | salviadud just phone to server, internal. no outgoing lines and actually only one phone |
| 17:13.49 | jbalcomb | ManxPower i must politely decline to receive any further assistance from you, but thank you. |
| 17:13.57 | Chaosmonkey | same with out going calls if its local the system redirect to POTS and if its distant it will use VOIP |
| 17:14.25 | jbalcomb | salviadud its on the same network as our main * server and 120 phones which are all functioning fine |
| 17:14.40 | salviadud | odd |
| 17:14.49 | jbalcomb | salviadud indeed |
| 17:14.56 | salviadud | what version of *? |
| 17:15.09 | jbalcomb | salviadud 1.2.5 |
| 17:15.29 | salviadud | well, no mayor changes with 1.2.6 |
| 17:15.39 | *** join/#asterisk iGotNoTime (n=joshua@cpe-65-189-240-199.woh.res.rr.com) |
| 17:15.47 | salviadud | i've never had audio problems before |
| 17:16.01 | salviadud | so, i'm kinda in the dark over here |
| 17:16.02 | jbalcomb | salviadud the weirdest part for my newb mind to comprehend is that asterisk seems to think everything is working just dandy |
| 17:16.17 | salviadud | oh, i get that |
| 17:16.23 | salviadud | the CLI reports no errors |
| 17:16.47 | salviadud | i guess the data packets are not going through... |
| 17:16.48 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 17:16.56 | salviadud | still, i don't get it, is it pri |
| 17:16.58 | salviadud | is it sip |
| 17:16.59 | salviadud | iax2? |
| 17:17.00 | jbalcomb | salviadud exactly. its shows the call, dialplan steps, and 0 entries in the log file. 100% A-OK. |
| 17:17.20 | jbalcomb | salviadud no PIR or iax, just SIP/RTP |
| 17:17.42 | salviadud | definetely a nat problem |
| 17:17.46 | x86 | i've got a question about how the voicemail system works... |
| 17:17.52 | jbalcomb | salviadud 50' of ethernet and two switches between the phone and server |
| 17:18.06 | x86 | when a new voicemail comes in, it creates 4 files in the user's voicemail box on the asterisk server... |
| 17:18.22 | x3me | hi |
| 17:18.25 | ManxPower | x86, four? |
| 17:18.28 | x3me | ppl, take a look... |
| 17:18.31 | x86 | like "msg0000.WAV", "msg0000.wav", "msg0000.gsm", and "msg0000.txt" |
| 17:18.35 | salviadud | if you could fix sip.conf to work with that config you got there, it would probably fix itselft |
| 17:18.41 | x86 | ManxPower: yes, four |
| 17:18.49 | x3me | only for test.. if i configure an linux box with asterisk, and include 2 users on the config files... |
| 17:18.52 | ManxPower | x86, that's only because you are allowing three file formats |
| 17:18.57 | x3me | install 2 softphones in 2 pc's in my office |
| 17:19.00 | ManxPower | it would be only two if you only allowed 1 file format |
| 17:19.06 | x3me | i can make calls between this users? |
| 17:19.12 | Darwin35 | grrr |
| 17:19.15 | x3me | without voip prodiver? |
| 17:19.17 | x3me | provider? |
| 17:19.18 | x86 | ManxPower: the ".WAV" and ".gsm" files are relatively small, while the ".wav" file seems to be rather large |
| 17:19.21 | Darwin35 | rewriting to meet nanpa sucks |
| 17:19.27 | ManxPower | x86, correct. |
| 17:19.34 | jbalcomb | salviadud [TK]D-Fender my sip.conf http://pastebin.com/627165 |
| 17:19.35 | Darwin35 | nanpa needs to suck donkey balls |
| 17:19.35 | x86 | ManxPower: ah.... so i need to trim that down to just one format :) |
| 17:19.36 | ManxPower | x86, what is the actual problem? |
| 17:19.45 | x86 | ManxPower: that's all ;) |
| 17:19.55 | x86 | ManxPower: i said "question" not "problem" :P |
| 17:20.07 | ManxPower | ah. |
| 17:20.12 | ManxPower | well you have your answer. |
| 17:20.28 | Chaosmonkey | :( |
| 17:20.28 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 17:20.38 | x86 | so what's the difference between wav49 and wav? |
| 17:20.47 | Qwell[] | x86: about 49 |
| 17:20.50 | salviadud | i wonder... change nat to yes |
| 17:20.52 | iGotNoTime | can someone give me a cheatsheet URL to print for all the * CLI commands? |
| 17:21.00 | x86 | Qwell[]: nice ;) |
| 17:21.02 | Qwell[] | iGotNoTime: type "help" |
| 17:21.03 | iGotNoTime | I am searching but only finding wiki's |
| 17:21.04 | tsume | iGotNoTime: no :) |
| 17:21.07 | iGotNoTime | :P |
| 17:21.15 | iGotNoTime | never thought of something so simple |
| 17:21.18 | Qwell[] | I'm not kidding. Type "help" :P |
| 17:21.27 | iGotNoTime | hi tsume been a while :D |
| 17:21.28 | Qwell[] | and show applications, and show functions, etc |
| 17:21.39 | iGotNoTime | ok will do :) |
| 17:22.35 | Corydon-w | Or just hit <tab> |
| 17:22.44 | iGotNoTime | HA! |
| 17:22.53 | Qwell[] | Corydon-w: That only gives first level commands |
| 17:23.01 | iGotNoTime | tab is a bit more llimited but I had never hit tab before |
| 17:23.01 | Corydon-w | Qwell[]: true |
| 17:23.11 | iGotNoTime | nice simple stuff :) |
| 17:23.23 | Corydon-w | Tab command line completion has been there the entire time |
| 17:23.25 | salviadud | well if it's not behind a nat, then there should be no reason |
| 17:23.26 | jbalcomb | salviadud tried that, no change. bleh on this situation. |
| 17:23.40 | salviadud | sorry jbalcomb |
| 17:23.42 | Qwell[] | and of course, "help <blah>" |
| 17:23.59 | jbalcomb | salviadud its cool. i appreciate the assist all the same. |
| 17:23.59 | iGotNoTime | I have only used Linux full time for a few months now :P |
| 17:24.06 | iGotNoTime | I am very new to CLI still |
| 17:24.10 | [TK]D-Fender | jbalcomb : So you made the recordings, moved & renamed the, put them in the proper folder and you still don't hear ANYTHING? |
| 17:24.24 | ManxPower | x86, 1 is raw audio, one is gsm wrapped into a microsoft package |
| 17:24.27 | jbalcomb | [TK]D-Fender yeah, i dont even hear the beep |
| 17:24.41 | jbalcomb | [TK]D-Fender volume on the phone is all the way up |
| 17:24.45 | x86 | ManxPower: ah, gotcha |
| 17:24.48 | salviadud | but, its only 1 softphone? |
| 17:24.52 | x86 | ManxPower: thanks |
| 17:24.56 | iGotNoTime | When I change these config files do I need to restart * ? |
| 17:24.57 | salviadud | i mean |
| 17:24.59 | salviadud | just 1 sip |
| 17:25.04 | iGotNoTime | I mean every little change? |
| 17:25.06 | salviadud | 2001 |
| 17:25.14 | ManxPower | iGotNoTime, what config file? |
| 17:25.15 | jbalcomb | salviadud yeah, 1 gxp-2000 set to ext. 2001 |
| 17:25.25 | iGotNoTime | SIP.conf ManxPower |
| 17:25.27 | [TK]D-Fender | jbalcomb, what phone? |
| 17:25.31 | ManxPower | iGotNoTime, a reload works for most changes |
| 17:25.34 | iGotNoTime | ok |
| 17:25.35 | Assid | err.. is this normal? http://pastebin.com/627175 |
| 17:25.45 | jbalcomb | [TK]D-Fender 1 gxp-2000 set to ext. 2001 |
| 17:25.47 | salviadud | and if you go 'sip show peers' in the cli |
| 17:25.48 | Qwell[] | iGotNoTime: and you can do specific reloads, like "sip reload" |
| 17:25.59 | salviadud | no problems |
| 17:26.17 | ManxPower | Assid, I can't tell you, but if you have 6 users logged in, you have other problems |
| 17:26.21 | [TK]D-Fender | jbalcomb : And you checked your RTP settings on the phone and set them for * default and leeft * at 10000-20000? |
| 17:26.24 | jbalcomb | salviadud seems fine. '2001/2001 10.0.101.151 D N 5060 Unmonitored' |
| 17:26.31 | iGotNoTime | ok Qwell still just installed so no big deal running at home not business. I just didn't know if the files were 'live' :) |
| 17:26.38 | ManxPower | Assid, this might be related to the similar report I saw yesterday |
| 17:26.38 | iGotNoTime | thx for the help guys :) |
| 17:26.43 | Assid | ManxPower: well.. getting different outputs |
| 17:26.44 | [TK]D-Fender | jbalcomb : Add qualify=yes to that phone's entry |
| 17:27.15 | Assid | err. qualify is required for pap2 apparently |
| 17:27.18 | Assid | else. it dies out |
| 17:27.21 | [TK]D-Fender | jbalcomb, then place a call and do "sip show channels" in CLI, then "sip show channel sip/2001-whatever " the channel happens to be while in call. |
| 17:28.52 | Assid | ManxPower? |
| 17:29.21 | jbalcomb | [TK]D-Fender i only see one setting on the gxp-2000 for rtp 'Local RTP Port' which i changed from default of 5004 to 15004. asterisk is still on default 10,000-20,000 |
| 17:30.01 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 17:30.32 | ManxPower | Assid? |
| 17:30.37 | CrashHD | my asterisk processes seem to be crashing every morning |
| 17:30.42 | CrashHD | I'm using safe asterisk |
| 17:30.43 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 17:30.51 | Assid | you said something about similar report. so i thought you were gonna lemme know what it was |
| 17:31.00 | CrashHD | any thoughts on how to make the asterisk more reliable, or atleast restart if there is a problem? |
| 17:31.08 | ManxPower | Assid, nope. you would have seen it yourself if you were on the mailinglists. |
| 17:31.16 | [TK]D-Fender | jbalcomb : Take another phone model to test with.... |
| 17:31.23 | ManxPower | CrashHD, find out what the problem actually is. |
| 17:31.34 | CrashHD | nothing in the logs |
| 17:31.41 | ManxPower | [TK]D-Fender, I'll bet his asterisk server has 2 interfaces on it. |
| 17:31.59 | ManxPower | CrashHD, start asterisk as "asterisk -cvvv" then LEAVE THE WINDOW OPEN |
| 17:32.10 | jbalcomb | [TK]D-Fender http://pastebin.com/627198 |
| 17:32.24 | ManxPower | you will see something on the console, or perhaps you could read backtrace.txt in the asterisk docs directory for information on generating a backtrace to report a bug |
| 17:32.28 | CrashHD | these are production systems, any other way? |
| 17:32.40 | ManxPower | CrashHD, um, why can't you do this on a production system? |
| 17:32.47 | Qwell[] | You can get a backtrace on a prod system |
| 17:32.55 | CrashHD | if it dies it would take me restarting it manually |
| 17:33.02 | CrashHD | if I'm not there |
| 17:33.06 | CrashHD | it could be down for awhile |
| 17:33.16 | ManxPower | CrashHD, oh then don't run asterisk -cvvv but get a backtrace |
| 17:33.40 | CrashHD | ~backtrace |
| 17:33.42 | jbot | hmm... backtrace is a debugging tool that is invaluable when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read README.backtrace) |
| 17:33.48 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:33.54 | PakiPenguin | evening |
| 17:33.54 | CrashHD | ok |
| 17:33.57 | CrashHD | thanks ManxPower |
| 17:34.13 | ManxPower | looks like bell totally screwed up our provider switch today |
| 17:34.22 | CrashHD | are there any better init scripts out there? or should I just write my own? |
| 17:34.43 | Chaosmonkey | still no takers on the redirect issue? |
| 17:34.49 | ManxPower | CrashHD, write your own. safe_asterisk is fine of everyone else. |
| 17:35.13 | CrashHD | it's just funny that it doesn't restart asterisk as it should |
| 17:35.30 | ManxPower | CrashHD, can you tell why when you look at the script? |
| 17:36.21 | [TK]D-Fender | jbalcomb : Seriously, grab another phone model.... |
| 17:36.54 | jbalcomb | [TK]D-Fender setting up my new polycom ip 501 for ext 2002 right now.. |
| 17:37.53 | jbalcomb | [TK]D-Fender 'Saved useragent "PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067" for peer 2002' |
| 17:38.12 | CrashHD | ooking |
| 17:38.54 | CrashHD | if the exit status was 0 it wouldn't restart |
| 17:39.39 | *** join/#asterisk opc0de (i=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com) |
| 17:40.09 | opc0de | hey can anyone tell me, is it possible to assign more than one context to a channel in zapata.conf ? |
| 17:40.10 | jbalcomb | [TK]D-Fender ok, calling the other phone actually makes it ring which didn't work yesterday but i think thats cause i had the polycom misconfigured. |
| 17:40.44 | jbalcomb | [TK]D-Fender and the damn call works fine. it didn't work yesterday and i thought focusing on just being able to hear asterisk prompts would make sense. |
| 17:41.03 | [TK]D-Fender | :/ |
| 17:41.07 | jbalcomb | [TK]D-Fender so calls work except for the prompts from asterisk! |
| 17:41.22 | Hmm-work | its one of those fun SER days |
| 17:41.31 | Hmm-work | that make me want to tear my eyes out |
| 17:41.37 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 17:41.48 | [TK]D-Fender | jbalcomb : something else is wrong.... wish I could pop-infor a look... |
| 17:42.39 | jbalcomb | [TK]D-Fender me too. the /other/ phone admin is gone for 24 days so i am soley responsible for the whole system now and i don't feel safe. :/ |
| 17:42.49 | opc0de | anyone? can you define a channel to have more than one context? |
| 17:42.53 | Hmm-work | you can pay me to feel safe |
| 17:43.00 | Hmm-work | otherwise I'll bust your legs |
| 17:43.08 | CrashHD | ManxPower thanks for the info, ttyl |
| 17:43.14 | ghotiboy1 | anyone here have some experience with the spa3000? i have mine working but it doesn't seem to register with asterisk, which means all incoming calls go to [from-internal] instead of [from-pstn] (the context the spa3000 user is directed to) |
| 17:43.27 | ghotiboy1 | all passwords quadruple checked |
| 17:43.30 | Hmm-work | someone is using amp |
| 17:43.34 | ghotiboy1 | true |
| 17:43.35 | salviadud | i got a spa3000 |
| 17:43.38 | jbalcomb | Hmm-work money doesn't work like that here i'm afraid. i've been paying for t-1 for three weeks now but haven't gotten a PO approved for the router to plug it into. |
| 17:43.39 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 17:43.47 | salviadud | what exactly is your setup? |
| 17:44.02 | Hmm-work | jbalcomb: <chuckle> ok, i was joking anyway |
| 17:44.30 | jbalcomb | Hmm-work i figured. ;) but i'm not. *sniff* *sob* SOB |
| 17:44.42 | ghotiboy1 | i am directing incoming pstn calls to (<S0:1>) |
| 17:45.28 | salviadud | ghotiboyl, pastebin your sip.conf |
| 17:45.31 | ghotiboy1 | it sends calls to asterisk just fine...just doesn't log in for some reason, so [from-internal] is the context instead of the one defined for the user |
| 17:45.49 | Hmm-work | ghotiboy1=firoze? |
| 17:46.04 | ghotiboy1 | Hmm-work: ??? |
| 17:46.07 | Hmm-work | nm |
| 17:46.42 | Hmm-work | so i wish this endpoint wasn't retarded |
| 17:48.08 | ghotiboy1 | salviadud: http://pastebin.com/627231 |
| 17:49.00 | *** join/#asterisk somegeek_ (i=levin@unaffiliated/somegeek) |
| 17:49.07 | ghotiboy1 | i pust my sip_additional.conf there too |
| 17:49.24 | ManxPower | ghotiboy1, look at the topic. |
| 17:49.26 | salviadud | well yeah |
| 17:49.29 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 17:49.31 | salviadud | i only see 19 lines |
| 17:49.38 | salviadud | not much help |
| 17:49.40 | Hmm-work | ghotiboy1: because default for all extensions in amp is from-internal |
| 17:50.16 | ghotiboy1 | Hmm-work: that is why i think it is a registration issue with the spa3000 |
| 17:50.30 | salviadud | wait |
| 17:50.33 | salviadud | you are using amp? |
| 17:50.35 | *** join/#asterisk lilo_ (i=levin@freenode/staff/pdpc.levin) |
| 17:50.36 | ghotiboy1 | yes |
| 17:50.43 | salviadud | duuuuuuuuuude. i can't help you out that much |
| 17:50.49 | salviadud | i use plain asterisk |
| 17:50.52 | Hmm-work | first clue-> context from-internal |
| 17:51.05 | ManxPower | salviadud, anytime you see sip_additional.conf it means "i'm running AMP" |
| 17:51.06 | ghotiboy1 | i can edit the files by hand if need be |
| 17:51.22 | octothorpe | ghotiboy1: look at this http://members.optusnet.com.au/~bsharif/asterisk/AsteriskDumbMeGuide.htm#_Toc131220368 |
| 17:51.37 | ManxPower | ghotiboy1, the config files for Asterisk@Home / AMP are 10 times more complicated than the normal asterisk config files. |
| 17:51.52 | ghotiboy1 | i agree |
| 17:52.00 | ManxPower | which is why most people here won't help with it. |
| 17:52.00 | salviadud | damn, no wonder |
| 17:52.16 | ManxPower | ghotiboy1, #freepbx was not helpful. |
| 17:52.26 | Hmm-work | heh, i've got some more complicated configs than amp makes |
| 17:52.31 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 17:52.33 | l-fy | hello |
| 17:52.46 | salviadud | yeah, but you did them |
| 17:52.48 | l-fy | is there any document describing iax protocol |
| 17:52.49 | l-fy | ? |
| 17:52.50 | salviadud | nobody did them for you |
| 17:53.09 | ManxPower | l-fy, yes. |
| 17:53.23 | salviadud | www.voip-info.org |
| 17:53.26 | l-fy | ManxPower: any link please? |
| 17:54.08 | ManxPower | l-fy, no idea. I would have to search google for it. |
| 17:54.19 | l-fy | i'm looking now also |
| 17:54.51 | l-fy | i've found something |
| 17:55.03 | l-fy | i will use that as the background |
| 17:55.50 | iGotNoTime | my * IP would be my linux box's internal IP, my internet ISP IP or my router IP ? |
| 17:56.15 | ManxPower | iGotNoTime, stop being lazy and tell us things like WHERE ARE YOU SPECIFYING THE IP |
| 17:56.28 | l-fy | thanks ManxPower |
| 17:56.30 | l-fy | bye |
| 17:56.32 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 17:56.49 | iGotNoTime | ManxPower: I am not trying to be lazy I am reading the help files. I am specifying the IP on my Wifi sip phone config |
| 17:57.38 | iGotNoTime | ManxPower: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Zyxel+P2000W that page say Asterisk IP is 10.10.3.5 |
| 17:57.50 | iGotNoTime | I don't know what the 10.10.3.5 is supposed to signify |
| 17:58.00 | ManxPower | iGotNoTime, if the phone is on the local lan then you specify the ip of the asterisk server, if it's not onthe local lan and it's behind nat then you specify the external ip of your NAT router, unless you are using services, in which case you specify the dns name of the sip services dns entry |
| 17:58.16 | iGotNoTime | ok that was my question :) |
| 17:58.18 | iGotNoTime | thank you |
| 17:58.34 | ManxPower | unless asterisk is on a public ip, in which case you specify the ip of the asterisk server |
| 18:01.52 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 18:03.32 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 18:03.59 | ManxPower | I need to install a wind powered generator. |
| 18:04.04 | ManxPower | The wind just never tops here. |
| 18:04.07 | ManxPower | stops |
| 18:04.27 | salviadud | where do you live man? |
| 18:04.36 | Assid | hey ManxPower: you good with agents and stuff right ? |
| 18:04.46 | ManxPower | Assid, no |
| 18:05.00 | ManxPower | salviadud, on the top of chandler mountian in Steele, AL |
| 18:05.46 | mut | OH NO! YOU'RE A REDNECK! |
| 18:06.46 | salviadud | manxpower you got FWD? |
| 18:07.00 | ManxPower | salviadud, hell no. |
| 18:07.05 | salviadud | iaxtel? |
| 18:07.13 | ManxPower | for me, FWD is a totally useless thing, as is iaxtel. |
| 18:07.28 | starlein | is it possible to set the channel status manually by some application or agi/manager command? |
| 18:07.33 | salviadud | well, if i wanted to call you |
| 18:07.41 | salviadud | via iax2 |
| 18:07.51 | ManxPower | salviadud, you can't call me via iax2 |
| 18:07.54 | ManxPower | salviadud, ping me |
| 18:08.01 | starlein | because my phpagi always return hangup (channel-status: down) |
| 18:08.35 | salviadud | <PROTECTED> |
| 18:08.42 | ManxPower | salviadud, unless you are offering large piles of cash, I don't talk to people. |
| 18:08.55 | salviadud | i just want to talk for fun man |
| 18:08.59 | justinu | haha |
| 18:09.10 | ManxPower | But I'll be happy to insult you for free. 8-) |
| 18:09.23 | salviadud | yeah, that's what i'm talking about |
| 18:10.40 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 18:11.18 | justinu | haha |
| 18:13.19 | salviadud | maddox sent out his newsletter |
| 18:13.26 | salviadud | that book is gonna be funky |
| 18:13.46 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 18:13.56 | mut | what kinda moonshine ya got? |
| 18:14.21 | *** join/#asterisk stoffell (n=stoffell@d51A4D49E.access.telenet.be) |
| 18:14.56 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 18:15.03 | qseek | hello all |
| 18:15.31 | qseek | does anyone know how to get an application from the asterisk repository which is not distributed with the general software release |
| 18:15.49 | qseek | i am specifically trying to get the source for app_amd.c |
| 18:17.41 | ManxPower | I need to find a vandal resistant outdoor phone. |
| 18:18.45 | justinu | payphone |
| 18:19.34 | ManxPower | justinu, that's a good idea. |
| 18:19.48 | iGotNoTime | ok I got my wifi sip to register with the server (thanks to ManxPower) now I have an error and was hoping for some help decoding it... I am sooo close to be able to dial I can feel it :P |
| 18:19.53 | iGotNoTime | Mar 28 18:30:42 NOTICE[32389]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'sip' |
| 18:20.12 | iGotNoTime | Does anyone know from that what config file I need to edit? |
| 18:20.20 | ManxPower | the rednecks sometimes drive past and shoot up everyone's mailboxes |
| 18:20.22 | iGotNoTime | is it sip.conf again? |
| 18:20.43 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 18:20.51 | iGotNoTime | ManxPower: you have a public phone?? |
| 18:21.16 | ManxPower | iGotNoTime, uhu? |
| 18:21.22 | iGotNoTime | nevermind :P |
| 18:21.37 | MRH2 | u pulled manx? |
| 18:22.56 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 18:23.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 18:23.18 | justinu | manx: http://cgi.ebay.com/PayPhone-Western-Electric-Pay-Phone-1970-1980-era_W0QQitemZ6267124686QQcategoryZ985QQrdZ1QQcmdZViewItem |
| 18:23.57 | iGotNoTime | justinu: leave it to ebay LOL |
| 18:24.08 | jbalcomb | [TK]D-Fender so if the calls work from phone to phone and reinvites are off then RTP through the server is working right? |
| 18:24.13 | x86 | hmm |
| 18:24.20 | x86 | any way to do post-hangup processing of a call? |
| 18:24.53 | tzanger | x86: catch the 'h' exten |
| 18:24.59 | iGotNoTime | I have googled NOTICE[32389] and there are no asterisk related results, does anyone know what that error is? |
| 18:25.07 | tzanger | x86: be aware though, that it buggers your CDR |
| 18:25.14 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 18:25.31 | *** join/#asterisk power1 (i=daemon@dsl-165-135-10.telkomadsl.co.za) |
| 18:26.25 | ManxPower | iGotNoTime, that is because 32389 is a PID and is different every time you start asterisk |
| 18:26.29 | jbalcomb | iGotNoTime the 32389 is not an error number, its just an ID for that entry in the log file |
| 18:26.39 | iGotNoTime | LOL |
| 18:26.40 | iGotNoTime | ok |
| 18:26.45 | iGotNoTime | sorry :) |
| 18:27.16 | power1 | Hey all, could you offer me some advice......I have a small home network asterisk setup, Currently using "ulaw" I was wondering if you reccomend using the paid version of "9729" in this environment.....mainly calls coming from pst via digium tdm400p and I will be setting up a sip trunk with a provider........??? |
| 18:27.31 | x86 | for example, i've made a script that will take a MixMonitor'ed call and convert it to gsm, and stuff it into a user's voicemail box |
| 18:27.35 | power1 | Sorry that G729 |
| 18:27.37 | x86 | tzanger: tried that... |
| 18:27.42 | justinu | i can't believe a real western electric payphone is only 120 bucks |
| 18:27.43 | jbalcomb | qseek you should be able to just ftp it from ftp.asterisk.org |
| 18:27.47 | justinu | i might have to get one myself |
| 18:27.53 | tzanger | x86: what are you trying to do, exactly |
| 18:27.57 | x86 | tzanger: s,1,MixMonitor(${UNIQUEID}|bW2) s,2,Dial(IAX2/trunk/${ARG2}|100|tr) h,1,System( ... ) |
| 18:28.01 | x86 | tzanger: not doing anything after the call dies :( |
| 18:28.04 | b66mer | Anybody know why when I dial out it seems to take a longer time to connect than if I was using a traditional PBX? |
| 18:28.17 | tzanger | x86: use 'g' in the Dial and then do you Systen in s,3 |
| 18:28.19 | x86 | power1: surely you mean G.729 |
| 18:28.23 | x86 | righto ;) |
| 18:28.23 | ManxPower | b66mer, yes. |
| 18:28.30 | Qwell[] | b66mer: Using a cheesy $15 fxo? |
| 18:28.33 | b66mer | yea |
| 18:28.36 | b66mer | :( |
| 18:28.37 | x86 | tzanger: i already said... record a call and shove it in a user's voicemail box when the call is over, so they can listen to it just like any other VM message |
| 18:28.40 | Qwell[] | That's why |
| 18:28.47 | ManxPower | Asterisk collects the DTMF digits, then once you are done dialing it dials out the phone line |
| 18:28.56 | b66mer | ok... so I need to pickup a digium? |
| 18:29.02 | b66mer | x100p |
| 18:29.05 | x86 | tzanger: Dial(IAX2/trunk/${EXTEN}|100|trg) ? |
| 18:29.08 | Qwell[] | digium doesn't sell x100p anymore |
| 18:29.17 | b66mer | what do you recommend? |
| 18:29.18 | tzanger | sounds good, why are you using 'r' though |
| 18:29.20 | ManxPower | b66mer, how are you dialing out? |
| 18:29.34 | b66mer | cheesy fxo |
| 18:29.41 | power1 | x86, yes sorry G.729 |
| 18:29.43 | b66mer | $12 I think if I remember |
| 18:29.45 | ManxPower | tzanger, be must be a newbie if he's using "r" |
| 18:29.55 | x86 | ManxPower: *nods* |
| 18:29.59 | x86 | what's 'r' do ? |
| 18:30.07 | tzanger | x86: if you don't know what it's for don't use it |
| 18:30.08 | ManxPower | b66mer, how do you know it takes longer to dial out if you can't dialout? |
| 18:30.11 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 18:30.13 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 18:30.15 | tzanger | not trying to be rude, but that's a big mistake most people make |
| 18:30.18 | b66mer | I can dialout |
| 18:30.21 | x86 | tzanger: i got it off some howto ;) |
| 18:30.22 | ManxPower | x86, r overrides any sound you should be hearing and plays a fake ringing tone |
| 18:30.26 | b66mer | it just takes 2-3 seconds |
| 18:30.28 | tzanger | x86: the howto's wrong |
| 18:30.31 | x86 | ManxPower: ahhh |
| 18:30.33 | jbalcomb | Got RTP packet from 10.0.100.147:2234 (type 0, seq 1502, ts 240400, len 160) |
| 18:30.34 | terrapen | still trying to find another VoIP equipment supplier |
| 18:30.34 | ManxPower | b66mer, HOW ARE YOU DIALING OUT? |
| 18:30.40 | jbalcomb | is my RTP the 2234? |
| 18:30.40 | b66mer | longer than if I was dialing normally |
| 18:30.42 | terrapen | besides VoIPSupply |
| 18:30.42 | b66mer | POTS |
| 18:30.53 | x86 | tzanger: what's "t" and "g" do? |
| 18:30.54 | tzanger | x86: at any rate though, yes that should help with what you're trying to do. |
| 18:30.57 | ManxPower | b66mer, how are you interfacing POTS with Asterisk? |
| 18:30.58 | tzanger | x86: show application dial |
| 18:31.00 | power1 | x86, any reccomendations? |
| 18:31.07 | b66mer | ManxPower: cheap FXO card from ebay |
| 18:31.18 | b66mer | pci card |
| 18:31.21 | ManxPower | b66mer, well you have your answer. |
| 18:31.23 | Qwell[] | ManxPower: You're slow :p |
| 18:31.30 | b66mer | so what do you recommend |
| 18:31.31 | Qwell[] | We determined that 5 minutes ago. ;) |
| 18:31.31 | b66mer | ??? |
| 18:31.45 | justinu | some people like the SPA-3000 |
| 18:31.47 | ManxPower | b66mer, changing the card is not going to fix the problem. |
| 18:31.55 | justinu | some people like TDM400 |
| 18:32.01 | b66mer | ok? what should I use? |
| 18:32.05 | terrapen | is there a mailing list for admins of large Asterisk installations? |
| 18:32.12 | ManxPower | b66mer, no equipment will fix your problem. |
| 18:32.18 | ManxPower | it's JUST THE WAY IT WORKS |
| 18:32.24 | ManxPower | unless, of course you go with a PRI. |
| 18:32.30 | ManxPower | but those are expensive |
| 18:32.36 | stoffell | terrapen, the regular user list ? |
| 18:32.47 | stoffell | terrapen, and define large ;) |
| 18:32.49 | b66mer | I have a PRI... but don't use it for outbound |
| 18:33.06 | ManxPower | b66mer, how many channels on the PRI? |
| 18:33.10 | b66mer | 24 |
| 18:33.18 | ManxPower | actually 23 voice. |
| 18:33.20 | terrapen | stoffell, I want a list where I don't have to read about somebody trying to get their FXO card working |
| 18:33.33 | b66mer | what is this a quiz? |
| 18:33.47 | power1 | could you offer me some advice......I have a small home network asterisk setup, Currently using "ulaw" I was wondering if you reccomend using the paid version of "G.729" in this environment.....mainly calls coming from pst via digium tdm400p and I will be setting up a sip trunk with a provider........??? |
| 18:34.01 | terrapen | stoffell, I want to talk about big-ass channel banks, T3s, lots of PRIs, call centers, massive voicemail servers, etc. |
| 18:34.04 | b66mer | ok... so no recommendations on a replacement for Digiums x100p? |
| 18:34.08 | stoffell | terrapen, i understand your "problem", but there's no list like that.. |
| 18:34.15 | terrapen | I'm going to start one. |
| 18:34.19 | ManxPower | b66mer, the issue, of course is that it takes TIME for asterisk to dial. I AM assuming that you are not doing something REALLY stupid like using "." in your extension patterns and do not have overlaping non-unique extensions |
| 18:34.34 | stoffell | terrapen, but i'm also "interested" in those subjects.. |
| 18:34.43 | ManxPower | b66mer, yes. Anyone that has a X100p clone and a PRI but is not using the PRI, I suspect they are confused. |
| 18:34.46 | terrapen | in FXO cards? :) |
| 18:34.48 | x86 | tzanger: ok, now the System executes before the MixMonitor stops... |
| 18:34.54 | x86 | tzanger: which wont work ;) |
| 18:35.05 | stoffell | terrapen, lol, no the big-ass and especially, redundant things :p |
| 18:35.07 | jbalcomb | [TK]D-Fender http://pastebin.com/627316 how come its playing digit/2 but not 0,0,1? |
| 18:35.17 | terrapen | stoffell: sweet. |
| 18:35.25 | x86 | tzanger: any other ideas? |
| 18:35.58 | tzanger | x86: so mixmonitor runs until the end of the call. You may just have to queue them up and run a batch outside the dialplan, or try to get 'h' to work |
| 18:36.05 | ManxPower | b66mer, assume you are dialing an 11 digit number and that each DTMF tone is 250ms long with a 50ms delay between digits. |
| 18:36.08 | ManxPower | now do the math |
| 18:36.25 | x86 | tzanger: any ideas what was preventing 'h' from working? |
| 18:36.28 | justinu | grandma dials faster than that :P |
| 18:36.30 | ManxPower | on a PRI, of course the digits are sent as data and so you don't have the delay. |
| 18:36.46 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 18:36.53 | ManxPower | of course if you have "." or over lapping extens in your dialplan, then asterisk will wait for DigitTimeout before processing the call |
| 18:36.54 | SpaceBass | howdy |
| 18:37.09 | ManxPower | x86, h works when one side hangs up, g works when the other side hangs up. |
| 18:37.15 | tzanger | x86: do you have an exten => h,1,NoOp(hit the hangup extension for ${UNIQUEID}) or soemthing? |
| 18:37.23 | ManxPower | I don't recall which side (caller or callee) is associated with each one |
| 18:37.28 | x86 | tzanger: no |
| 18:37.28 | terrapen | would anybody here sign up for asterisk-enterprise if I set one up? |
| 18:37.36 | tzanger | not me :-) |
| 18:37.41 | ManxPower | not me |
| 18:37.43 | terrapen | heh |
| 18:37.51 | terrapen | you guys do not run large installations? |
| 18:37.57 | tzanger | terrapen: not really, no. |
| 18:38.01 | terrapen | ah, ok |
| 18:38.01 | qseek | ok i figured it out..thanks anyways |
| 18:38.03 | ManxPower | me neither. |
| 18:38.06 | tzanger | biggest I could see setting up in the near future is maybe 30 sets |
| 18:38.15 | ManxPower | I manage 6 - 8 asterisk servers, all of them small. |
| 18:38.16 | stoffell | terrapen, I prefer "redundant" installations, but that qualifies as large also? :p |
| 18:38.32 | terrapen | sure. |
| 18:38.37 | Qwell[] | terrapen: I have a 1 user install |
| 18:38.46 | Qwell[] | Is that large? |
| 18:38.53 | tzanger | besides, asterisk-enterprise sounds like the kind of place doug would hang out in and demand that we solve all of his problems because asterisk isn't enterprise ready and doesn't work exactly as he expects it to |
| 18:38.55 | stoffell | Qwell[], depends on the user? :p |
| 18:39.06 | terrapen | HA Asterisk, lots of users, multiple PRIs, etc. |
| 18:39.09 | Qwell[] | power user? |
| 18:39.18 | stoffell | Qwell[], it counts :p |
| 18:39.20 | ManxPower | tzanger, RoyK too |
| 18:39.26 | ManxPower | (or is that the same person) |
| 18:39.31 | justinu | heh |
| 18:39.38 | *** join/#asterisk nvicf (n=nvicf@201.250.169.63) |
| 18:39.45 | nvicf | hello there |
| 18:39.47 | terrapen | basically, the idea is to make a list where enterprise admins can talk about Asterisk without all the "How do I make my X100P work?" crap |
| 18:40.09 | zoa | anybody seen royk? |
| 18:40.11 | ManxPower | terrapen, Here, let me introduce you to procmail |
| 18:40.21 | Qwell[] | zoa: He was looking for you yesterday. :P |
| 18:40.27 | ManxPower | zoa, he's been looking for you for days. |
| 18:40.30 | terrapen | manx, no thanks, procmail is not going to solve this. |
| 18:40.34 | justinu | lol |
| 18:40.44 | terrapen | there is no filter for "n00b idiot" |
| 18:41.00 | *** join/#asterisk Axel69 (i=Axel@200.62.38.91) |
| 18:41.05 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 18:41.08 | ManxPower | terrapen, sure there is. two filters, actually, one on "help" and the other on "possible" |
| 18:41.10 | Axel69 | hi guys |
| 18:41.19 | terrapen | heh |
| 18:41.32 | stoffell | terrapen, it's easy enough to set up a list, but without a text on the digium site it doesn't 'catch' a lot of people i'm afraid |
| 18:41.42 | Axel69 | i'm installing the asterisk, which one you recomend i used the AAH but i need the best |
| 18:41.45 | ManxPower | oh, and the asshole filters are "license" "gpl" and "theft" |
| 18:41.51 | terrapen | Well, I'm going to talk to digium about it. |
| 18:42.00 | Nugget | heh |
| 18:42.06 | stoffell | okay terrapen, great idea |
| 18:42.15 | terrapen | I could care less if they run the list. |
| 18:42.39 | *** join/#asterisk point (i=1000@213.27.44.55) |
| 18:42.49 | bkw__ | terrapen, hrm enterprise admin? Asterisk? Do they exist? |
| 18:42.53 | bkw__ | :P |
| 18:42.54 | terrapen | oh wait |
| 18:42.55 | *** join/#asterisk clive- (n=pirch@dsl-145-4-09.telkomadsl.co.za) |
| 18:42.59 | terrapen | there already is a list |
| 18:43.02 | terrapen | asterisk-biz |
| 18:43.04 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:43.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:43.12 | ManxPower | that's for BUINESS stuff |
| 18:43.25 | bkw__ | the -biz list is just as bad |
| 18:43.26 | terrapen | exactly |
| 18:43.30 | ManxPower | like "i'm selling 300 polycoms at $200 each", or "I need A-Z routes" |
| 18:43.33 | terrapen | hrmmm |
| 18:43.33 | bkw__ | its just as bad as users |
| 18:43.37 | terrapen | fuck. |
| 18:43.42 | bkw__ | all the mailing lists are worthless |
| 18:43.43 | ManxPower | terracon, read the archives |
| 18:43.44 | clive- | how does one do a svn update , as opposed to a svn checkout ? |
| 18:43.49 | terrapen | yeah, looking @ them |
| 18:43.51 | bkw__ | clive-, you do svn update |
| 18:43.58 | bkw__ | just like cvs update |
| 18:44.05 | clive- | Thanks Brian, |
| 18:44.28 | mog_work | bkw__, whats got you riled up? |
| 18:44.42 | bkw__ | mog_work, as if you don't know |
| 18:44.45 | ManxPower | bkw_, After SIX MONTHS of working with bell, they still could not do a provider change. |
| 18:45.06 | mog_work | i know of many things, and probably what you are refering to |
| 18:45.08 | ManxPower | apparently they handed the switch programming guy a 5 month old port/number sheet |
| 18:45.23 | mog_work | but i dont know what i have to do with it or the mailing lists etc |
| 18:45.30 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 18:45.48 | zoa | pervert! |
| 18:45.49 | zoa | :p |
| 18:45.52 | zoa | how are you brian ? |
| 18:45.56 | mog_work | lol zoa |
| 18:46.03 | zoa | stop looking at me like that |
| 18:46.04 | jarrod | anyone used Linux-HA with two asterisk servers? |
| 18:46.10 | zoa | no conversions here :) |
| 18:46.24 | stoffell | jarrod, not personally but i read about some success stories on it |
| 18:46.31 | jarrod | ive got static configs loaded into a sql server which is loaded at runtime/reload and wanna run linux-ha in case one fails |
| 18:46.38 | bkw__ | zoa, i'm good |
| 18:46.47 | jarrod | the only problem ive seen is that it seems asterisk itself fails before the hardware would fail |
| 18:46.53 | ManxPower | jarrod, be careful! you could cause a tear in space-time |
| 18:47.13 | justinu | chroniton particles |
| 18:47.44 | jarrod | is there a load balancer (i guess the equivalent of an SBC that isnt 35k+) that can manage the sessions from external sources between the two softswitches |
| 18:47.56 | jarrod | id rather use two in tandem then have a failover |
| 18:48.12 | justinu | like SER? |
| 18:48.20 | jarrod | SER is amazing as a SIP proxy |
| 18:48.28 | justinu | you can do some basic load balancing with it |
| 18:48.29 | jarrod | but ive only seen deployment of failover routing |
| 18:48.44 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 18:48.48 | justinu | load balancing with DNS SRV records w/ equal priorities works |
| 18:49.01 | jarrod | hmmm |
| 18:49.12 | justinu | however, failover using only DNS SRV doesn't :( |
| 18:49.27 | jarrod | but UDP is sessionless... and if it registers and then sends to the different SER box that it is not registered with could cause probs |
| 18:49.35 | stoffell | jarrod, round robin dns and dundi is what everybody shouts.. |
| 18:50.57 | justinu | i'm not sure what your specific config is, but SER has a transaction module to track the state of dialogs, etc. |
| 18:51.00 | mut | ya know... |
| 18:51.00 | mut | .there is no real good unified definition for up is there |
| 18:52.36 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:52.38 | *** join/#asterisk powerchip (n=powerchi@197.80-202-229.nextgentel.com) |
| 18:55.27 | *** join/#asterisk ricko73 (n=dhartman@206.40.109.147) |
| 18:55.35 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 18:56.15 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:57.00 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 18:57.30 | *** join/#asterisk gavi1 (n=gaving@grabes2.enter.net) |
| 18:58.14 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 18:59.25 | lullabud | anybody have a preferred softphone for os x? i'm using SJphone right now but looking for something more polished and with features like forwarding. |
| 19:00.03 | gavi1 | Hi guys, quick question. When a sip peer is setup fro call waiting, and they are logged into a queue. If they are engaged in a call, is there any way for asterisk not to pass queue calls to that member? |
| 19:00.13 | *** part/#asterisk ricko73 (n=dhartman@206.40.109.147) |
| 19:00.24 | Corydon-w | call-limit=1 |
| 19:00.43 | gavi1 | But, in that case call waiting will never work correct? |
| 19:00.47 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 19:00.54 | wasim | gavi1: pausequeuemember |
| 19:01.27 | wasim | lullabud: moziax |
| 19:01.29 | SpaceBass | lullabud I like idefsk or what ever |
| 19:01.37 | SpaceBass | lullabud its from the astguru folks I think |
| 19:02.41 | justinu | lullabud: eyebeam is the best softphone, imo |
| 19:02.44 | justinu | but it costs money |
| 19:02.49 | gavi1 | wasim: That seems tedious to do everytime someone wants to pick up the phone and call someone? |
| 19:03.39 | wasim | gavi1: clicking on a url is tedious? |
| 19:04.35 | lullabud | awesome, thanks guys |
| 19:04.38 | gavi1 | everytime I would like to make a call? Yes that does. Explaining that to 40 sales people... well thats the hard part |
| 19:05.07 | wasim | sales people should be tortured with a predictive dialer any way |
| 19:05.21 | Corydon-w | gavi1: why don't you do that automatically in the dialplan? |
| 19:05.25 | clive- | does anyone know why zttest gives such a bad score with ztdummy on centos 4.3 ? |
| 19:05.27 | justinu | tell them if they don't like it, they can bring in their own phone system from home |
| 19:06.09 | gavi1 | Can I have it pause an agent inside the dialplan? |
| 19:06.34 | jbalcomb | gavi1 i dont think so cause i have had to turn off call waiting for all employees in our call center |
| 19:06.38 | SpaceBass | lullabud also check out toms phone tools (think thats what its called) |
| 19:06.54 | SpaceBass | lullabud its a tool that will let you dial from the OS X address boox |
| 19:07.17 | konfuzed | hey does the vega devices support IAX |
| 19:08.17 | gavi1 | Thanks guys, its a shame I need to get rid of call waiting because of queueing! |
| 19:09.50 | [TK]D-Fender | gavi1 : its what I had to do.... |
| 19:09.53 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 19:11.11 | _andre | is there any way to add the extension of the person who answered a call from a queue to the name of the file? |
| 19:11.25 | _andre | i mean, the recorded file with Monitor |
| 19:11.49 | _andre | i know i can set MONITOR_FILENAME, but in extensions.conf i still don't know who'll answer the call |
| 19:11.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:12.10 | jbalcomb | _andre maybe you could use a variable in the filename? |
| 19:13.00 | jbalcomb | what do i need to do to get asterisk to see changes i've made to voicemail.conf? |
| 19:13.08 | _andre | but when i set MONITOR_FILENAME (in extensions.conf), the call wasn't answered yet |
| 19:13.16 | _andre | so i don't know what the extension will be |
| 19:13.37 | _andre | jbalcomb: reload app_voicemail.so |
| 19:13.54 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 19:13.56 | Darwin35 | honey I'm home |
| 19:14.03 | Darwin35 | whats for dinner |
| 19:14.08 | Assid | anyone have a good link for agents/queue |
| 19:14.12 | websae | hey if anyone is looking for quality* termination and origination at a great price...we are a carrier on the level 3 network...just let me know, and we can get you setup :)...we run asterisk ! |
| 19:14.22 | jbalcomb | _andre ah, hrm.. how about a system call to move the file after its recorded? |
| 19:14.23 | Darwin35 | read the queues.conf |
| 19:14.28 | Darwin35 | its really easy |
| 19:14.31 | justinu | what price web? |
| 19:14.37 | jpablo | hey people, I'm getting response 503 "Service Unavailable" when calling the pstn extension of a sipura 3000 any ideas ? |
| 19:14.54 | websae | us and canada= 1.4cents |
| 19:15.00 | lullabud | SpaceBass: oh, sweet. i tried an app called... JackenIAX that had address book support, but I couldn't get it to work with my asterisk system. |
| 19:15.06 | justinu | prepaid, or what? |
| 19:15.46 | websae | prepaid normally |
| 19:16.35 | clive- | what do you guys get as a test score on zttest ? |
| 19:16.46 | justinu | clive: |
| 19:16.47 | justinu | --- Results after 96 passes --- |
| 19:16.47 | justinu | Best: 99.987793 -- Worst: 94.750977 -- Average: 99.914805 |
| 19:17.30 | octothorpe | jpablo: check this out, it may help: http://members.optusnet.com.au/~bsharif/asterisk/AsteriskDumbMeGuide.htm#_Toc131220368 |
| 19:17.59 | clive- | justinu, hi thanks, I am not getting more that 98% |
| 19:20.06 | clive- | wondering why, since I used to get 99.75% on my other, almost identical box |
| 19:20.21 | justinu | i'm having my own issues with app_meetme... |
| 19:20.32 | justinu | put 2 sip channels into a meetme conf, and over time, the latency gets worse and worse |
| 19:20.40 | justinu | running ztdummy |
| 19:20.48 | justinu | applied the latest async RTP patches to 1.2.6 |
| 19:21.00 | justinu | (which supposedly solved this) |
| 19:22.21 | clive- | sounds strange, maybe that RTP patch is not all its cracked up to be |
| 19:22.51 | konfuzed | ok so vega stream does not support iax, is there any voip gateways that compare to vega stream 50 features but also support IAX |
| 19:22.56 | justinu | i dunno, it's frustrating |
| 19:22.59 | justinu | i need a solution |
| 19:23.05 | *** join/#asterisk ChrisN (n=ChrisN@zonebbs.com) |
| 19:23.21 | _andre | Darwin35: sorry, i don't get it... queue.conf only tells me to use Set() |
| 19:23.37 | clive- | I am wondering if my 98% score on zttest will be good enough for iax2 trunking |
| 19:24.17 | stoffell | clive-, what is your "best" and "average" ? |
| 19:24.35 | clive- | Best: 98.730469 -- Worst: 98.425293 -- Average: 98.469849 |
| 19:25.35 | iGotNoTime | is the ; a requirement on each line of the config files or is that a way to comment the line out? |
| 19:25.42 | clive- | stoffel, its not a great score, and I can't figure out why |
| 19:25.44 | stoffell | clive-, and you're running any zaptel channels? (digium/other cards..) |
| 19:25.57 | octothorpe | use the ; to comment out |
| 19:26.02 | clive- | stoffel I have 2 sirrix isdn bri cards |
| 19:26.09 | iGotNoTime | ok |
| 19:26.15 | x86 | ManxPower: care to have a look at my dialplan? |
| 19:26.29 | stoffell | clive- , and what does "cat /proc/interrupts" say about irq sharing? |
| 19:26.31 | ManxPower | x86, only if you are offering large piles of cash. |
| 19:26.42 | ManxPower | and even then, I can't until this evening at least. |
| 19:27.14 | octothorpe | x86: I could take a look now, but again, it will cost |
| 19:27.15 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 19:27.30 | clive- | everything is on its own IO-APIC-level |
| 19:27.45 | x86 | ManxPower: bahahaha ;) |
| 19:27.57 | eric_hill | Anybody know what the dialing digit pattern in Mexico is? (Like the USA xxx-xxx-xxxx) |
| 19:27.59 | x86 | :P |
| 19:28.24 | justinu | ask salviadud |
| 19:28.38 | SpaceBass | eric_hill google "dialing mexico" |
| 19:28.42 | stoffell | clive-, what kernel and distro you're running? |
| 19:28.48 | Katty | hi lovables. |
| 19:28.50 | SpaceBass | not being smart...i know there is a site out there |
| 19:28.52 | iGotNoTime | is it possible to have only 3 lines in the extensions.conf file? I keep getting an error about the context even though I have defined the context in extensions.conf. |
| 19:28.58 | jbalcomb | [TK]D-Fender ok, back on that, the IP501 (ext.2002) can call the gxp-2000 (ext.2001) but the gxp-2000 (ext.2001) can not call the IP501 (ext.2002) |
| 19:29.03 | clive- | stoffel, centos 4.3 |
| 19:29.10 | iGotNoTime | the default lines I was given by teliax is only three lines |
| 19:29.13 | jbalcomb | [TK]D-Fender the gxp-2000 (ext.2001) gets a 404 |
| 19:29.35 | eric_hill | SpaceBass: thanks - I found this site, http://www.westel.net/mexico_dialing.htm, but it didn't really tell me what I needed. Just "dialing mexico" worked great! |
| 19:30.07 | justinu | 2 or 3 digit area codes, 7 or 8 digit local numbers... w00t! |
| 19:30.43 | GerbilNut | iGotNoTime, post the extensions.conf (minus passwords), sip.conf (minus passwords), and iax.conf (minus passwords) to pastebin.ca and i'll look at it for ya |
| 19:30.51 | GerbilNut | the exact error would also be nice |
| 19:31.22 | [TK]D-Fender | jbalcomb : Did you add the other phone to [internal] ? |
| 19:31.47 | Katty | are the pixies any good? |
| 19:31.48 | jbalcomb | [TK]D-Fender just realized that and just added it, its working. |
| 19:31.48 | octothorpe | Hi Katty |
| 19:31.54 | Katty | octothorpe: (= |
| 19:32.10 | iGotNoTime | thank you GerbilNut will do now |
| 19:32.11 | clive- | stoffel I wonder, maybe I could install a spare x100p card I have to see if that improves things |
| 19:32.13 | jbalcomb | [TK]D-Fender can i change it to 2XXX and then they both work? |
| 19:32.26 | stoffell | clive-, i remember readin on centos and a bug, don't know if it's on that version... |
| 19:32.37 | stoffell | clive-, yes, or remove the current cards, and only install the x100p, just to see... |
| 19:32.51 | justinu | the pixies? i liked their music |
| 19:32.54 | [TK]D-Fender | jbalcomb : BAD practice.. its one of the things I'd UNDO on your work config... |
| 19:32.58 | justinu | it's educational!! |
| 19:33.00 | stoffell | ~centos |
| 19:33.02 | jbot | [centos] better than Fedora Core except for that silly bug, see ~centosbug for details |
| 19:33.10 | stoffell | ~centosbug |
| 19:33.11 | jbot | methinks centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 19:33.17 | clive- | stoffel, thanks, the centos bug has to do with a typo |
| 19:33.23 | jbalcomb | [TK]D-Fender ah, ok, i'll hold off on that then. |
| 19:33.29 | stoffell | clive-, okay, thanks :d |
| 19:33.36 | [TK]D-Fender | jbalcomb : hardcode |
| 19:33.58 | jbalcomb | [TK]D-Fender i've been debugging and tcpdumping to see if I could figure out why I'm not able to hear prompts from the server. |
| 19:34.19 | jbalcomb | [TK]D-Fender it sure seems like the traffic is being generated and sent but just not accomplishing anything. |
| 19:34.23 | jbalcomb | :/ |
| 19:35.00 | jbalcomb | [TK]D-Fender hardcode all 120+ extensions? |
| 19:35.05 | Luke-Jr | Should I be able to specify 'qualify=yes' in iax.conf [general]? |
| 19:35.06 | [TK]D-Fender | jbalcomb : YUP |
| 19:35.11 | iGotNoTime | GerbilNut: here it is, please tell me if you need more :) |
| 19:35.13 | iGotNoTime | http://pastebin.ca/47325 |
| 19:35.26 | jbalcomb | [TK]D-Fender gotcha |
| 19:36.03 | [TK]D-Fender | jbalcomb : where did you put the recordings? |
| 19:36.26 | GerbilNut | iGotNoTime, i don't need the instructions from Teliax, i need your actuall files pasted there |
| 19:37.15 | iGotNoTime | sorry... will do again |
| 19:37.37 | jbalcomb | [TK]D-Fender /var/lib/asterisk/sounds/custom |
| 19:37.52 | GerbilNut | ok, in the drop down box there is an "Asterisk Configuration" option, select that too while you're at it |
| 19:38.08 | jbalcomb | [TK]D-Fender i dont hear anything from asterisk though, not the beeps, queue msgs, or voicemail prompts |
| 19:39.20 | opc0de | hey can anyone tell me how to forward a call from a SIP phone, yet have the callerid from the original call come through? |
| 19:40.29 | iGotNoTime | GerbilNut: http://pastebin.ca/47327 |
| 19:41.50 | *** join/#asterisk juanmanuel (n=jmacz@201.244.240.87) |
| 19:41.55 | Splatty47 | is their a graphical interface that can add extensions etc to the sip.conf ? |
| 19:42.01 | *** part/#asterisk gavi1 (n=gaving@grabes2.enter.net) |
| 19:42.20 | jbalcomb | So I see this in my Asterisk CLI (http://pastebin.com/627434) but hear nothing. |
| 19:42.32 | *** join/#asterisk apardo (n=apardo@87.218.44.228) |
| 19:42.52 | Splatty47 | and extensions.conf etc etc... |
| 19:43.13 | GerbilNut | iGotNoTime, ok, the Teliax instructions say to modify three seperate files |
| 19:43.24 | iGotNoTime | I only seen two |
| 19:43.54 | GerbilNut | actually, let me re-read the ones you posted, i'm used to iax connections with Teliax |
| 19:44.22 | iGotNoTime | should I use IAX? I thought just SIP would be fine |
| 19:45.18 | GerbilNut | SIP is fine |
| 19:45.30 | GerbilNut | ok, one problem is at the very bottom of Extensions.conf |
| 19:45.44 | iGotNoTime | That was pasted :P |
| 19:46.08 | iGotNoTime | you mean the first line? |
| 19:46.11 | iGotNoTime | the number? |
| 19:46.33 | GerbilNut | the first, 1 priority should be removed and replaced with, exten => _1XXXXXXXXXX,1,DIAL(SIP/teliax/${EXTEN},30,tr) |
| 19:47.11 | GerbilNut | and in sip.conf, delete the info after context=default |
| 19:47.15 | iGotNoTime | I put in the 898 because that was the phone "ID", so it should be exten => _14192994337,1,DIAL(SIP/teliax/${EXTEN},30,tr) for example? |
| 19:47.16 | GerbilNut | delete the parts in ()'s |
| 19:47.36 | GerbilNut | are you using this for incoming, or outgoing calls? |
| 19:47.38 | ManxPower | NEVER USE r |
| 19:47.48 | iGotNoTime | ummm both |
| 19:48.02 | [TK]D-Fender | jbalcomb : Something is seriously whacked. Firewall issue I'm betting. |
| 19:48.02 | ManxPower | and don't use t or T unless you KNOW what they do. They can allow someone to make calls and bill them to you. |
| 19:48.06 | clive- | stoffel, hi, ...I removed those cards and zttezt still gives me 98% |
| 19:48.23 | Darwin35 | Teliax rocks |
| 19:48.25 | GerbilNut | tell you what, let me modify your paste bin, and you compare the two |
| 19:48.32 | iGotNoTime | ok |
| 19:48.34 | GerbilNut | no, Teliax sucks, but hey, what are you going to do |
| 19:48.36 | Darwin35 | welcome all you teliax users to the real world |
| 19:48.43 | iGotNoTime | ManxPower: are you talking to me? |
| 19:48.55 | ManxPower | iGotNoTime, yes |
| 19:48.58 | Darwin35 | we have done alot to correct our network issues |
| 19:49.04 | Darwin35 | we dropped cogent |
| 19:49.18 | Darwin35 | now our service is much better |
| 19:49.32 | *** join/#asterisk imcdona (n=imcdonal@38.100.225.67) |
| 19:49.35 | iGotNoTime | that is the default sample from teliax, do you have a link to info on those commands and their respective other options? |
| 19:49.51 | ManxPower | iGotNoTime, "show application dial" in the asterisk CLI. |
| 19:49.54 | ManxPower | also the Wiki |
| 19:49.56 | imcdona | Anyone know a good provider for DID's in Australia? |
| 19:50.12 | iGotNoTime | would another option mean my phone won't dial out? |
| 19:50.17 | opc0de | can anyone tell me how to transfer a call from one voip phone to another, yet have the callerid reflect the information from the original incoming call? |
| 19:50.20 | ManxPower | if tr were in the Teliax sample then someone at Teliax needs to be spanked. |
| 19:50.27 | iGotNoTime | hehe |
| 19:50.41 | ManxPower | opc0de, that is the default for BLIND transfers |
| 19:50.47 | GerbilNut | iGotNoTime http://pastebin.ca/47329 |
| 19:51.31 | Darwin35 | Teliax does not support alaw |
| 19:51.51 | opc0de | ManxPower: when someone calls and hits an extension, I see the caller id on that extension.. if that person on that extension then transfers the call to me, I see the caller id from the SIP phone, rather than the original caller.. is that the default? |
| 19:51.55 | Darwin35 | only ulaw/gsm/ilbc/g726/g729 |
| 19:52.03 | *** join/#asterisk juice (n=juice@mo-71-0-60-40.dyn.sprint-hsd.net) |
| 19:53.50 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 19:54.49 | iGotNoTime | GerbilNut: sorry for the delay copying to try :) |
| 19:57.09 | brad_mssw | Darwin35: any word on an east-coast server ? |
| 19:57.21 | [TK]D-Fender | Hey I could really use a hand here with something. I'm getting a /29 at home and need a quick guid on how to set up IPTABLES to dole out the IP's to my various devices... |
| 19:57.28 | ManxPower | opc0de, what kind of transfer, blind or consultative? |
| 19:57.51 | justinu | DHCP is what you want, fender |
| 19:57.56 | justinu | iptables is just a firewall |
| 19:58.52 | opc0de | ManxPower: it's alright, I've figured it out.. the phone has an option for blind transfer, that does it automatically.. I thought this was an asterisk setting, cause I notice there's a "useincomingcalleridonzaptransfer=yes" in zapata.conf.. thought I needed a similar setting for a sip phone transfer |
| 19:59.08 | iGotNoTime | GerbilNut: ok that got rid of that error, thank you |
| 19:59.22 | iGotNoTime | I am off to read again to move on to the ATA stuff |
| 19:59.22 | ManxPower | opc0de, a consultative transfer is really more of a threeway call. |
| 19:59.37 | GerbilNut | enjoy |
| 19:59.42 | iGotNoTime | hehe |
| 19:59.49 | iGotNoTime | i appreciate it alot |
| 19:59.59 | [TK]D-Fender | justinu : I'm running PPPoE with RP-PPPOE, and I used to do DHCP for my internal LAN. I need to change the IP's I'll be giving out internally and route accordingly. |
| 20:00.20 | justinu | uhh |
| 20:00.27 | justinu | wht's RP-PPPoE? |
| 20:00.35 | stoffell | [TK]D-Fender, or use proxyarp? :) |
| 20:00.40 | ManxPower | justinu, a PPoE client |
| 20:00.44 | file | roaring penguin pppoe client |
| 20:00.45 | justinu | ah, for linux? |
| 20:00.47 | file | it roars! |
| 20:01.06 | [TK]D-Fender | justinu :Roaring Penguin PPPoE package for Linux |
| 20:01.43 | X-Gen | hey clive- & stoffell freaks |
| 20:01.54 | justinu | i haven't actually worked with PPPoE... just heard horrors about it |
| 20:02.35 | clive- | hi X-gen |
| 20:03.07 | stoffell | lol, hi X-Gen |
| 20:03.17 | iGotNoTime | when dialing out I have gotten an error reading: Timeout, but no rule 't' in context 'default' |
| 20:03.19 | clive- | X-gen, I am going to find my old x100p to see if I get better zttest times |
| 20:03.26 | iGotNoTime | is that due to the rule ManxPower mentioned? |
| 20:03.33 | Nugget | I like that I don't have to deal with pppoe, but in reality it's not that awful. |
| 20:04.02 | justinu | so how does it work? they send traffic for an entire /29 over PPP, and you route it out on your LAN with that client? |
| 20:04.33 | [TK]D-Fender | I'm wondering if my server will pick up the 1st IP of my /29 (8 total, 6 usable) and that I just need to enable forwarding and assign DHCP to give out IP's 3-7 to my other devices. |
| 20:04.34 | Nugget | yep. heck, any modern dsl router appliance can do the pppoe internally and it's invisible to your systems. |
| 20:04.38 | X-Gen | clive-: enable ACPI or APCI or somthing like that |
| 20:04.43 | [TK]D-Fender | justinu : I *think* so. |
| 20:04.45 | justinu | i think that's why my DSL router does |
| 20:04.55 | justinu | DSL bridge, whatever the fuck it is |
| 20:05.06 | Nugget | and it's pretty painless to do ahe os level if you prefer that. |
| 20:05.16 | Nugget | s/ahe/at the/ |
| 20:05.17 | [TK]D-Fender | Nugget : I don't have a DSL router, I'm running a Sangoma S518. |
| 20:05.18 | Katty | ewwo, Nugget |
| 20:05.22 | Nugget | dang gprs lag. |
| 20:05.27 | *** part/#asterisk ChrisN (n=ChrisN@zonebbs.com) |
| 20:05.30 | justinu | TK, describe that? |
| 20:05.46 | [TK]D-Fender | justinu : What my S518? |
| 20:05.46 | justinu | pci ADSL modem |
| 20:05.57 | *** join/#asterisk kposmyk (n=kposmyk@195-128-242-5.akk.net.pl) |
| 20:05.57 | justinu | so what does their software give you? a PPP interface? |
| 20:06.00 | kposmyk | :) |
| 20:06.02 | justinu | in linux |
| 20:06.13 | *** join/#asterisk Denmark (n=fake@62.242.24.182) |
| 20:06.19 | [TK]D-Fender | justinu : Yeah, it just provides a raw interface over which I run PPPoE. |
| 20:06.26 | *** join/#asterisk chr|s_ (n=chris@217.171.51.191) |
| 20:06.37 | justinu | ok, can you show my ifconfig -a? |
| 20:06.41 | justinu | for learning purposes? |
| 20:06.48 | Denmark | kposmyk : Ask your question here. :) |
| 20:07.03 | justinu | s/show my/show me/ |
| 20:07.26 | [TK]D-Fender | justinu : not now, its at home..... and I'm not sure if it will work any time soon.... |
| 20:07.38 | justinu | it doesn't have to work, i was just curious what kind of interfaces you had |
| 20:07.45 | kposmyk | I have a hardware PBX and I use only ISDN |
| 20:08.09 | stoffell | justinu, a ppp0 interface is just like eth0, or tun, or anything else.. |
| 20:08.14 | kposmyk | I need to record all cals... is it possible by using asterisk.... |
| 20:08.15 | justinu | yeah, that I know |
| 20:08.22 | justinu | but is that what his software gives him? |
| 20:08.31 | kposmyk | ? |
| 20:09.02 | stoffell | justinu, the rp-pppoe you mean? |
| 20:09.14 | justinu | yeah |
| 20:09.36 | justinu | or he is planning on running PPPoE on his own LAN? |
| 20:09.41 | stoffell | justinu, that software enables you to use pppoe on a plain standard ppp interface |
| 20:09.44 | justinu | or terminating it at theSangoma Card |
| 20:09.53 | imcdona | Anyone know of a good provider of DID's in Australia? |
| 20:10.06 | stoffell | justinu, pppoe runs through that 'sangoma' box (which acts as a |
| 20:10.08 | stoffell | modem) |
| 20:10.11 | justinu | ah |
| 20:10.26 | kposmyk | brb |
| 20:10.28 | X-Gen | kposmyk: |
| 20:10.34 | chr|s_ | guys - question about the sample extensions.conf |
| 20:10.36 | justinu | so the RP-PPPoE client provides the ppp0 interface |
| 20:10.36 | X-Gen | basic rate or primary rate ISDN ? |
| 20:10.46 | *** join/#asterisk Z0m81e (n=pault@85-210-143-122.dsl.pipex.com) |
| 20:10.49 | kposmyk | 23B+D |
| 20:10.59 | chr|s_ | there is a lot of stuff like iaxtel700 exten => etc |
| 20:11.05 | justinu | rip it out |
| 20:11.06 | chr|s_ | can I keep this in there, or best to remove it? |
| 20:11.07 | *** join/#asterisk eipi (n=eipi@OL17-54.fibertel.com.ar) |
| 20:11.08 | justinu | all useless |
| 20:11.09 | X-Gen | 23 ? not e1 or t1 ? |
| 20:11.18 | *** join/#asterisk adamsih300u (n=adamsih3@m-h32.rh.sunyit.edu) |
| 20:11.23 | chr|s_ | aah good, justinu , was that the answer? |
| 20:11.28 | kposmyk | X-Gen, E1 |
| 20:11.39 | justinu | chr|s_: rip it out... it's basically useless |
| 20:12.08 | kposmyk | X-Gen, I just need to insert something between my telecom and my hardware PBX |
| 20:12.11 | X-Gen | e1 = 30B+D just btw. i hear it can be done, but afaik u will need 2 e1 cards |
| 20:12.15 | eipi | i have no audio in this scheme... anyone can help me? wip300 (VOIP wireless phone) <-> internet <-> wrt54gs <-> asterisk 1.2.6 linuxbox |
| 20:12.26 | *** join/#asterisk web_ustaad (n=__web_us@202.61.51.115) |
| 20:12.38 | chr|s_ | justinu, what about the 'macro-stdexten' section, is the whole file useless? |
| 20:13.07 | kposmyk | X-Gen, can this solution be transparent to my PBX ? |
| 20:13.20 | web_ustaad | where can I find different configurations |
| 20:13.21 | Z0m81e | Does anyone have experience of using the spa3k fxo with asterisk? |
| 20:13.28 | web_ustaad | related to asterisk |
| 20:13.34 | shmaltz | Z0m81e |
| 20:13.38 | shmaltz | yes I do |
| 20:14.00 | chr|s_ | justinu, I am keeping everything upto and including [globals] as it is required conf by the looks |
| 20:14.01 | *** join/#asterisk Ansonmus (n=ahaeser@dsl97-13-100.fastxdsl.nl) |
| 20:14.11 | X-Gen | kposmyk: i'm not sure at all how it would work :( |
| 20:14.18 | Z0m81e | shmaltz do you know any good resources for how to configure? I've looked at voxilla but the setup isn't working at the mo |
| 20:14.39 | shmaltz | Z0m81e, what's not working? |
| 20:14.43 | justinu | no, macro-stdexten is ok |
| 20:15.16 | chr|s_ | heh, I removed that, oh well :p |
| 20:15.16 | x86 | any way to record a call (both parties like MixMonitor does), but stop recording when the call is hung up by either party? |
| 20:15.27 | Z0m81e | shmaltz, good question the line just drops after I dial, i need to do some debugging will you be here for 10 mins? |
| 20:15.46 | justinu | chr|s_: no worry, you can get it off the wiki |
| 20:15.50 | justinu | probably a better one, at that |
| 20:15.50 | x86 | MixMonitor seems to want to stall until after everything is done, including all 'h' extension handlers... |
| 20:15.52 | shmaltz | what do you mean it drops? incoming works? |
| 20:16.24 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 20:16.34 | Z0m81e | shmaltz, I haven't checked, incoming was the first thing I tried, i'll just give it a call and see what happens |
| 20:16.42 | anthm | unless it's hacked to pieces when they committed it i used to have an option in there to only record while a call is bridged |
| 20:16.55 | shmaltz | Z0mb81e, could be a dialplan problem |
| 20:17.07 | chr|s_ | justinu, ok, so these built in demos work out of the box? |
| 20:17.26 | chr|s_ | I am impressed, for now, I am just configuring one extension, a sip phone, to dial out and answer incoming... |
| 20:17.28 | justinu | mostly |
| 20:17.43 | chr|s_ | *soft phone |
| 20:17.52 | justinu | none of that stuff is by no means necessary to have |
| 20:17.59 | Z0m81e | shmaltz, i'm in the UK so i've hard coded in 1471 which reads back the last called on your line, my money is on sip.conf I was a little hazy about what voxilla was asking me to do it wasn't explained why the settings were as they are |
| 20:18.01 | *** join/#asterisk ederaam (n=ederaam@200.30.102.50) |
| 20:18.04 | justinu | probably better to start simple and build on that, when you're a beginner |
| 20:18.08 | *** join/#asterisk firekid (n=chatzill@pix013-155.pix.wmich.edu) |
| 20:18.16 | ederaam | Can you help me with fxs fxo????? |
| 20:18.34 | firekid | Hi everyone.. |
| 20:18.43 | firekid | great to be here |
| 20:18.59 | Katty | anthm: you have website up for cluecon yet? |
| 20:19.05 | chr|s_ | ederaam, what about them? |
| 20:19.21 | anthm | finishing touches |
| 20:19.28 | firekid | Hi can someone plz help me with openvxi?? |
| 20:19.36 | ederaam | I need how to configure a fxs fxo target |
| 20:20.03 | ederaam | I have a target but I don know hot configure there. |
| 20:20.49 | justinu | [TK]D-Fender: i see your sangoma card uses the same wanpipe drivers I use for my T1 card |
| 20:20.59 | [TK]D-Fender | justinu : Yup./.. |
| 20:21.40 | [TK]D-Fender | justinu : Works great and doesn't have the throttling problems of using an ethernet card + external modem. Allows for much better traffic shaping. |
| 20:21.49 | edobe | when a call is ended and the other party hung ups, should the sip phone end automatically the call or must the Hungup button be pressed? |
| 20:25.35 | *** join/#asterisk Muecke77 (n=muecke77@p54A9EAD0.dip.t-dialin.net) |
| 20:26.27 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 20:26.42 | gaspiz | i'm keep getting -- Remote UNIX connection disconnected |
| 20:26.56 | *** join/#asterisk Hmmhesays (n=Neg@72.24.105.126) |
| 20:27.07 | gaspiz | <PROTECTED> |
| 20:27.14 | stoffell | just wondering, kernel config: timer frequency.. is it best to set it to 100 or 1000hz on an * server? |
| 20:27.17 | Hmmhesays | well asterisks retarded use of srv recordes just threw a big ol' wrench in my plans |
| 20:27.19 | gaspiz | can someone help? |
| 20:27.25 | imcdona | Gaspiz.....it means that something is conencting to the manager interface..... |
| 20:27.42 | clive- | stoffel, x-gen that old x100p makes my zttest score 100%..yay |
| 20:27.51 | imcdona | AAH has some programs that access the Manager interface.... |
| 20:27.59 | imcdona | Flash operator panel.... |
| 20:27.59 | stoffell | clive-, awesome, 100 all the way ? |
| 20:28.09 | X-Gen | clive-: u get centos sorted out with the compiling..etc ? |
| 20:28.10 | imcdona | and asterisk status button in AAH |
| 20:28.14 | clive- | yup, I cant belive it |
| 20:28.20 | gaspiz | imcdona: might by thanks |
| 20:28.39 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 20:28.42 | clive- | x-gen, the compiling is no problem, the zttest was giving too low results |
| 20:29.01 | clive- | stoffel I need a good score for my iax2 trunking |
| 20:29.42 | eipi | i have no audio in this scheme... anyone can help me? wip300 (VOIP wireless phone) <-> someone router <-> internet <-> my router wrt54gs <-> asterisk 1.2.6 linuxbox |
| 20:29.53 | stoffell | eipi, check firewall+NAT issues |
| 20:29.59 | *** join/#asterisk Eggplant (i=No@dsl-745.cascadeaccess.com) |
| 20:30.20 | clive- | stoffel are you in south africa also ? |
| 20:30.22 | SpaceBass | eipi where did you get the WIP300? |
| 20:30.32 | eipi | voipsupply |
| 20:30.34 | stoffell | clive-, no, up north.. (europe, belgium) |
| 20:30.38 | eipi | dont but it |
| 20:30.43 | eipi | dont buy it |
| 20:30.45 | SpaceBass | Eitch thought they were back ordered |
| 20:30.50 | SpaceBass | eipi dont like it? |
| 20:30.57 | eipi | yes, too many bugs |
| 20:30.58 | SpaceBass | gotta be better than the other wifi phones out there |
| 20:31.05 | konfuzed | oh globetel.net makes IAX2 gateways |
| 20:31.06 | eipi | battery life its short |
| 20:31.14 | SpaceBass | does it at least support wpa? |
| 20:31.18 | eipi | yes |
| 20:31.21 | eipi | wap/wap2 |
| 20:31.25 | eipi | tkip/aes |
| 20:31.40 | SpaceBass | NICE |
| 20:31.52 | eipi | linksys has no support for wip300 |
| 20:32.08 | eipi | not real support |
| 20:32.52 | *** join/#asterisk Chaosmonkey (n=jon@c-71-199-252-53.hsd1.fl.comcast.net) |
| 20:33.19 | X-Gen | clive-: with a nick like that i also thought he was from africa. /whois whois stoffell |
| 20:34.10 | stoffell | hehe :D |
| 20:39.04 | konfuzed | !gsm |
| 20:40.11 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 20:40.22 | *** join/#asterisk x-mark (n=brg@c-68-43-129-244.hsd1.mi.comcast.net) |
| 20:40.28 | x-mark | I can't seem to get my callwaiting to work. I have a POTs line into |
| 20:40.28 | x-mark | <PROTECTED> |
| 20:40.28 | x-mark | <PROTECTED> |
| 20:41.06 | x86 | nice paste |
| 20:41.13 | x-mark | sorry |
| 20:42.25 | *** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) |
| 20:43.52 | *** join/#asterisk apardo (n=apardo@87.218.44.228) |
| 20:44.11 | Chaosmonkey | is it possible to make a specific extenion use a specific trunk |
| 20:45.06 | eric_hill | Chaosmonkey: Maybe Goto a context based on CALLERID? |
| 20:45.07 | jpablo | Chaosmonkey, yes, just send it to another context |
| 20:46.00 | Chaosmonkey | i thought so just making sure |
| 20:46.00 | SpaceBass | Chaosmonkey using A@H? |
| 20:46.02 | hfb | Hey Strom_M |
| 20:46.03 | Chaosmonkey | yes |
| 20:46.21 | SpaceBass | ahhh... check out #freepbx by the way...but thats a pretty requested feature |
| 20:46.29 | SpaceBass | user based outbound routing |
| 20:46.40 | SpaceBass | its quite possable, but its one of the things that AMP complicates a lot |
| 20:46.52 | SpaceBass | I've been meaning to set it up on my work vs home lines |
| 20:47.03 | SpaceBass | but been lazy, so I just dial *8 for my work trunk |
| 20:47.12 | *** join/#asterisk angler_ (n=johnb@199.227.185.58) |
| 20:47.24 | Chaosmonkey | yeah well its the only solution i can come up with for sending faxes |
| 20:47.29 | Chaosmonkey | i recieve them just fine |
| 20:47.33 | Chaosmonkey | but sending is a hassle |
| 20:47.42 | jbalcomb | [TK]D-Fender well, this traffic is all layer-2 and on the same network as all the other phones working off the current production server so no firewall anywhere. |
| 20:48.00 | Chaosmonkey | so im just going to plug my fax machine into FXS port and set that extension to only use the FXO Line |
| 20:48.05 | jbalcomb | [TK]D-Fender i'm wondering if its something to do with the app that plays the .gsm files maybe.. |
| 20:48.53 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 20:49.01 | SpaceBass | Chaosmonkey you can easily write that into the dial plan in extensions_custom.conf ... our just put something like *9|. in your trunk's plan and dial *9 before each fax |
| 20:49.14 | SpaceBass | Chaosmonkey lucky you...i can send but cannot recieve faxes |
| 20:49.40 | iGotNoTime | I have a 12 inch touchscreen for the Asterix box, can anyone suggest a GUI to use with it? CLI is not so easy with the touchscreen |
| 20:49.57 | SpaceBass | iGotNoTime FOP (flash operators pannel) |
| 20:49.59 | *** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu) |
| 20:50.00 | Chaosmonkey | were would i configure the *9| at |
| 20:50.01 | iGotNoTime | ok |
| 20:50.10 | iGotNoTime | SpaceBass: it is fairly stable? |
| 20:50.11 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 20:50.14 | SpaceBass | Chaosmonkey in the trunk in your AMP gui |
| 20:50.21 | SpaceBass | iGotNoTime think so...depends on what you are trying to do |
| 20:50.41 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 20:50.43 | iGotNoTime | simple stuff :) basically logging |
| 20:50.58 | x-mark | I can't seem to get my callwaiting to work. I have a POTs line into and FXo port and a phone connected to the FXS port when a call is coming through via callwaiting I can't seem to answer it |
| 20:51.19 | *** part/#asterisk kposmyk (n=kposmyk@195-128-242-5.akk.net.pl) |
| 20:51.20 | SpaceBass | Chaosmonkey oops...spoke too soon...its in dial paters under OUTBOUND ROUTES |
| 20:51.57 | SpaceBass | x-mark thats a little tricky, did you write a dial plan for flashing the zap line or check out the one on voip info? |
| 20:52.33 | x-mark | No i did not write a dial plan for flashing the zap channel |
| 20:52.47 | justinu | [TK]D-Fender: this is kinda informtive: http://scottstuff.net/blog/articles/2005/02/11/sangoma-s518-pci-adsl-modem-review |
| 20:52.49 | x-mark | Ok I'll look at voip-info.org |
| 20:52.58 | SpaceBass | x-mark I have a suggeston for you....you aren't going to like it at first, but then you are going to pull your hair out trying everything else then come back to it and realize it was elegant |
| 20:53.14 | x-mark | lay it on me.... |
| 20:53.50 | SpaceBass | x-mark flashing a zaptel line is pretty much a pain...you have to basically transfer your call to an extension that puts you on hold, flash the line and then it calls you back with the 2nd call...meanwhile you have none of your asterisk features (like music) available |
| 20:54.31 | SpaceBass | x-mark soooooo get a BYOD lite account at www.broadvoice.com for $5/month...its unlimited incoming minutes and calls (IE moer than one at once)...then have your telephoen company set up "call forward on busy" to your broadvoice number |
| 20:54.59 | justinu | i think broadvoice is charging 40 bucks activation for BYOD now |
| 20:55.21 | SpaceBass | that way, rather than call waiting, it forwards to a SIP account right in to your asterisk box...you have all the features of asterisk, including more than one incoming call...not just 2 like traditional call waiting |
| 20:55.25 | SpaceBass | justinu bastards! |
| 20:55.30 | [TK]D-Fender | justinu : No help, but thanks. the card "just works", what I need is linux networking tips for the /29. not sure how to set that up. |
| 20:55.36 | Hmmhesays | is there anywhere that tells asterisk how long to wait before receiving an ack to a sip invite? |
| 20:55.39 | jbalcomb | [TK]D-Fender is it possible that i have failed to setup or install proper support for the gsm codec? |
| 20:55.55 | SpaceBass | Hmmhesays!!! hey there |
| 20:55.56 | [TK]D-Fender | jbalcomb : I wouldn't imagine so.... |
| 20:56.04 | Hmmhesays | hey SpaceBass |
| 20:56.10 | x-mark | I see. I have a DID with nufone. I suppose it should work just the same. |
| 20:56.37 | jbalcomb | [TK]D-Fender the only difference i see between the phone call traffic and the voice prompts traffic is the phone calls are ulaw and the prompts are gsm |
| 20:57.38 | [TK]D-Fender | jbalcomb : I see your point, but its too freakish.... |
| 20:58.01 | Hmmhesays | because right now dial takes way to long to timeout on no response |
| 20:58.39 | jbalcomb | [TK]D-Fender agreed, but it seems like anything else would have to interfere with the actuall SIP or RTP traffic for both situations. my tcpdump, sip debug, and rtp debug suggest there is no traffic interference |
| 20:59.08 | [TK]D-Fender | jbalcomb : Sure nothing is being blocked? |
| 21:00.01 | jbalcomb | [TK]D-Fender as best i can tell. the fact that the other server and all its phones work fine does support that notion. |
| 21:00.37 | jbalcomb | [TK]D-Fender the only option i see is that somehow the other phone server is interfering but i seriously don't see an option for that |
| 21:01.29 | *** part/#asterisk Utah_Dave (n=boucha@0-2pool130-217.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 21:01.36 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 21:03.03 | *** join/#asterisk JSabines (n=alancast@201.138.136.215) |
| 21:04.04 | SpaceBass | eipi how long did it take to get your wip300? |
| 21:04.11 | SpaceBass | eipi what kind of bugs are you getting? |
| 21:06.24 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
| 21:06.56 | MattH | Hi.. I just installed asterisk on a new server no firewall (yet)... and the phone is not behind a firewall (they are on the same subnet)... tcpdump (and audio I hear) indicates the phone is sending data to asterisk, but asterisk is just sending nothing out... any thoughts on this? |
| 21:09.16 | konfuzed | ok i was looking into channel banks a bit, but it does not really seem like what I actually want. Perhaps with my description of what I need/want , somebody can tell me what that is called. for context, a 40 unit apartment complex has asked to make the move entirely to voip (pros and cons of course, would also provide some failover) so, I want to take the existing rj11 & cat3 wiring (40 to 80 outlets) and remove them from the incoming Bell se |
| 21:09.16 | konfuzed | rvice and plug them into (probably individually) an external device or direct into an asterisk-pbx server. What is that device? |
| 21:09.27 | *** join/#asterisk scrambray8927 (n=scrambra@12.104.121.147) |
| 21:09.46 | konfuzed | or how do I plug 40 internal phone outlets into asterisk? |
| 21:10.10 | jbalcomb | konfuzed IP phones or analog phones? |
| 21:10.15 | MattH | konfuzed, yeah channel bank for analog phones :) |
| 21:10.28 | konfuzed | analog channel bank |
| 21:10.35 | jbalcomb | konfuzed yeps |
| 21:10.49 | konfuzed | well there touchtone phones in their existing rj11 jack |
| 21:10.53 | eric_hill | konfuzed: The Rhino channel banks look like a good choice for that - they'll support many (all?) of the CO features such as caller ID, message waiting stutter, etc. |
| 21:11.06 | konfuzed | ah |
| 21:11.21 | jbalcomb | MattH you mean like if you dial voicemail you see the lines on the CLI indicating its playing the prompts but you hear nothing on the phone? |
| 21:11.27 | konfuzed | i didnt see rhino listed on voip-info channel banks page |
| 21:11.35 | konfuzed | ill have to find them |
| 21:12.05 | shido6 | they are all over google |
| 21:12.07 | eric_hill | http://www.voip-info.org/wiki/view/Asterisk+Channel+Bank |
| 21:12.42 | shido6 | they are around $1300 |
| 21:12.47 | SpaceBass | ouch! |
| 21:13.01 | *** join/#asterisk juanmanuel (n=jmacz@201.244.240.87) |
| 21:13.02 | SpaceBass | bulk order some linksys PAP2s and an switch off ebay :) |
| 21:13.04 | shido6 | for 24 |
| 21:13.27 | konfuzed | ah I was somewhat confused with http://www.voip-info.org/wiki/view/What+is+a+GSM+Channel+Bank%3F |
| 21:13.42 | shido6 | 930 for 15 paps |
| 21:13.45 | eric_hill | Have a look at their site: http://www.rhinoequipment.com/cb.html |
| 21:14.34 | SpaceBass | i guess PAPs are like $40...thats not a great difference in price |
| 21:14.44 | scrambray8927 | I'm looking for recommendations for VoIP service providers that allow me to spoof caller id for outbound calls. Can anyone recommend any companies? I've looked at Nufone and VoicePulse. |
| 21:15.17 | SpaceBass | nufone |
| 21:15.26 | SpaceBass | for pranks just charge it with like $5...that enough for months of fun |
| 21:15.26 | shido6 | dont do it. |
| 21:15.42 | X-Gen | do it do it. |
| 21:15.50 | scrambray8927 | not for pranks, for attack and penetration engagements for clients. |
| 21:15.53 | MattH | it's illegal depending on what you are doing for it |
| 21:15.58 | MattH | yeah that's even worse |
| 21:16.07 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 21:16.07 | scrambray8927 | we're covered with our Statement of Work |
| 21:16.11 | shido6 | think in terms of court costs |
| 21:16.21 | tmccrary | Is there a way to get Asterisk force calls to go peer to peer? Like MGCP? |
| 21:16.34 | shido6 | yesh |
| 21:16.37 | SpaceBass | illegal you say? |
| 21:16.56 | tmccrary | Is that some sort of option somewhere? |
| 21:17.13 | tmccrary | Sorry, I should clarify |
| 21:17.26 | tmccrary | To make SIP calls go phone to phone, to not act as a bridge |
| 21:17.49 | SpaceBass | tmccrary its a dial plan option that keeps asterisk in the loop...cannot remember...think its 't' |
| 21:18.15 | konfuzed | 1300 / 40 is only 32.50 so that is reasonable |
| 21:18.19 | tmccrary | Also, by SIP, I mean make the RTP data go phone to phone |
| 21:18.36 | SpaceBass | I used setcallerID() and annoyed a friend by calling him as his own phone over an entire weekend...he eventually called verizon and got really worked up...so i quit and came clean |
| 21:18.42 | [TK]D-Fender | tmccrary : canreinvite=yes |
| 21:18.45 | SpaceBass | verizon is probably tracking me down now...great |
| 21:18.49 | harlequin516 | Is Dundi only used with ASterisk, or is it something more abstract for VOIP than specifically for asterisk? |
| 21:18.51 | SpaceBass | canreinvite...thats it! |
| 21:19.08 | Chaosmonkey | is there a reason that everywhere i call out it the recipients cid says unknown |
| 21:19.16 | Chaosmonkey | it is also the same with all calls i recieve |
| 21:19.22 | SpaceBass | Chaosmonkey who is your provider |
| 21:19.26 | Chaosmonkey | teliax |
| 21:19.45 | SpaceBass | sounds like a teliax issue...do they allow setcallerid()? |
| 21:19.54 | Chaosmonkey | yes |
| 21:19.58 | Chaosmonkey | it was working until today |
| 21:20.05 | *** join/#asterisk lrizzo (n=luigi@81-174-38-222.f5.ngi.it) |
| 21:20.50 | lrizzo | q |
| 21:20.55 | *** join/#asterisk sirukin (n=sirk@h64-42-196-1.gtconnect.net) |
| 21:20.58 | sirukin | hey hey |
| 21:21.53 | scrambray8927 | MattH or SpaceBase, do you have a nufone account? |
| 21:22.35 | sirukin | I have a kx-tda30 panasonic hybrid ip pbx |
| 21:22.55 | [av]bani | \o> |
| 21:22.56 | [av]bani | <o/ |
| 21:23.44 | SpaceBass | scrambray8927 I do |
| 21:24.26 | fnordian | scrambray8927: what's your experience with nufone, do they allow spoofing? |
| 21:24.30 | VeNoMouS_ | hrm is there way u can jump to a context and return with a value? |
| 21:24.43 | VeNoMouS_ | to the context u called the goto from? |
| 21:25.17 | SpaceBass | scrambray8927 lets stick here...I'm in and out today... |
| 21:25.27 | riksta | anyone using LDAP to authenticate SIP users? |
| 21:25.43 | SpaceBass | scrambray8927 i don't use nufone as my primary at all...although for the past 2 weeks i've used it as my primary for local calls...seems to work well |
| 21:25.51 | GerbilNut | anyone gotten two Asterisk servers communicating via IAX2, using switch, sharing dial-plans? |
| 21:26.00 | SpaceBass | fnordian yes they allower setcallerid() |
| 21:26.18 | scrambray8927 | k |
| 21:26.24 | SpaceBass | GerbilNut i had it working for a while with older versions of A@H...then A@H broke it in 2.something |
| 21:26.27 | fnordian | to isdn? |
| 21:26.54 | konfuzed | ok so these channel banks equipement not mentioning anything about SIP or IAX leads me to presume that it is irrelevant. So, I should probably verify, im under the impression that FXS and FXO dont actually use sip/iax/h323 cause its a direct hardware connect vs a SOFTWARE Conenct and so the fxs/fxo part is not actually VoIP cause there is no ethernet or ip involved (well on the local side that is). And so incoming to the pbx is only VoIP if i |
| 21:26.54 | konfuzed | t comes in over the internet WAN and not VoIP if the call come in via copper from phone co. |
| 21:27.06 | SpaceBass | scrambray8927 haven't had any audio problems...but bet i've only made 100 short calls compaired to 5,000 calls with broadvoice where I have audio issues about %20 of the time |
| 21:27.09 | GerbilNut | well the boxes have Asterisk 1.2.24 installed on them, and I get a nifty error when I try to place the call |
| 21:27.44 | websae | SpaceBass: if you need a quality termination/origination provider, let me know... |
| 21:27.59 | riksta | konfuzed: FXO is for picking up calls that come in on your old normal phone line, fxs is for allowing normal phone handsets to work with asterisk |
| 21:28.03 | GerbilNut | websae, what company? |
| 21:28.05 | SpaceBass | konfuzed that is correct...but I think some banks connect via fiber...just as a means to connect...not sure |
| 21:28.17 | SpaceBass | websae thanks...think we are good now... what does the SAE stand for? |
| 21:28.39 | tmccrary | riksta: Yes, I am working on an asterisk-based product that uses LDAP for authentication |
| 21:28.55 | SpaceBass | konfuzed you could also look at ATAs for each apartment and run cat 5 to them...or keep it all in a basement or somethign and run regular POTS to the ATAs |
| 21:28.59 | Axel69 | any gui manager that anyone recomend |
| 21:28.59 | Axel69 | ? |
| 21:29.00 | justinu | society of automotive engineers? |
| 21:29.04 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 21:29.06 | riksta | tmccrary: I have an interesting problem that i'd be interested to see if you have any solution would you mind a PM for 5 minutes? |
| 21:29.07 | SpaceBass | Axel69 for what? |
| 21:29.11 | Axel69 | for asterisk |
| 21:29.13 | tmccrary | sure |
| 21:29.17 | SpaceBass | Sleep And Eat ? |
| 21:29.21 | *** part/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 21:29.27 | konfuzed | so over these long months it is actually gelling (in my head that is) to some extent and im not quite as konfuzeD as when I started. |
| 21:29.37 | SpaceBass | :) |
| 21:30.07 | SpaceBass | konfuzed sounds like a neat project....just make sure everything is on a good UPS...people HATe to lose phones...personally, I'd hate to be responsible for a tennits phone system |
| 21:30.15 | SpaceBass | but I can see major advantages as well |
| 21:30.28 | SpaceBass | is websae a bot? |
| 21:30.39 | konfuzed | SpaceBass, yeah the initial idea was to just run ethernet cable to provide them all quality internet connectivity |
| 21:30.58 | SpaceBass | konfuzed worthwhile! |
| 21:31.12 | konfuzed | and then some jack ass said sure we can throw in voip for $3/mth/unit |
| 21:31.23 | SplasPood | CVS 28-10-03, am I reading that correctly as Oct 28 2003 ? |
| 21:31.31 | SpaceBass | i can see reselling phone and internet and making a little money and providing neat services like dood bell intercom and unit-to-unit calling....cheap LD... |
| 21:32.02 | SpaceBass | $3/month? thats not going to bring in the big bucks :) |
| 21:32.08 | *** join/#asterisk Deciphan (n=icechat5@c-67-174-56-25.hsd1.ca.comcast.net) |
| 21:32.20 | konfuzed | i figure I might be able to get them up to $3000 of hardware *PBX and channel bank or similar |
| 21:32.44 | SpaceBass | I've talked to developers who do developments for densly populated units about voip and net services |
| 21:32.53 | konfuzed | only cause they must sign a 5year agreement which allows us to finance the equipement |
| 21:33.15 | konfuzed | I really wanted to kick that guy in the head - $3/mth/unit |
| 21:33.35 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 21:33.41 | SpaceBass | well...its great for the landlord (and lesee) ...great sales pitch |
| 21:33.41 | konfuzed | did i mention I hate sales |
| 21:33.44 | konfuzed | im just a geek |
| 21:33.50 | SpaceBass | unlimited phone for $3? I'd take that! |
| 21:33.55 | scrambray8927 | Does anyone know how I can I figure out (according to my asterisk box's specs) how many calls I can initiate outbound at once? |
| 21:34.04 | bkw__ | scrambray8927, check the wiki |
| 21:34.07 | bkw__ | it all depends on many factors |
| 21:34.09 | *** join/#asterisk lrizzo (n=luigi@81-174-38-222.f5.ngi.it) |
| 21:34.13 | SpaceBass | wiki has a calculator type page |
| 21:34.15 | bkw__ | you'll need to load test to ensure your appliction will scale |
| 21:34.19 | bkw__ | still that will not work right |
| 21:34.25 | bkw__ | the best way is to do that yourself |
| 21:35.12 | scrambray8927 | bkw_ is the voip-info.org wiki the one you're referring to? |
| 21:35.19 | scrambray8927 | thanks |
| 21:35.20 | bkw__ | yes |
| 21:35.51 | harlequin516 | Hmm, I still don'tquite understand why Zaptel/FXO dialout can't detect when the dest phone is answered.. Can't it just monitor the ringing sound and change status when it is on-hook but the ringing has stopped? |
| 21:35.51 | SpaceBass | sprint has this new $5/month for unlimited calling from your cell to your house...how long will it take before they drop me for making ONLY calls to my house? |
| 21:36.12 | SpaceBass | harlequin516 think its actually based on voltage...but I'm not expert |
| 21:36.13 | harlequin516 | I'm sorry Imeant off hook |
| 21:36.34 | *** join/#asterisk x-mark (n=brg@c-68-43-129-244.hsd1.mi.comcast.net) |
| 21:36.46 | SpaceBass | x-mark get your telco to do the call forward on busy? |
| 21:37.20 | konfuzed | SpaceBass, i figure to play it safe give them all a voicemail box on the pbx for the $3/mth and they can do IP to IP only calls. They can take on extra sevices to activate DID and calls to non ip destinations. |
| 21:37.22 | x-mark | spacebass -- I just wanted to say your right.... Thas was an excellent idea -- works like a charm. I just have to wait for the telco to impliment |
| 21:37.38 | SpaceBass | konfuzed thats a good plan! |
| 21:37.51 | x-mark | spacebass - thank you very much for a great suggestion |
| 21:37.56 | *** part/#asterisk sirukin (n=sirk@h64-42-196-1.gtconnect.net) |
| 21:38.01 | SpaceBass | x-mark wish I could take 100% credit...roots of it came from this # and Hmmhesays (I think) |
| 21:38.02 | harlequin516 | SpaceBass: You're right kewlstart loopstart groundstart, but they don't work for dialout I think.. I would need to order something called answer supervision or something |
| 21:38.18 | SpaceBass | harlequin516 over my head :) |
| 21:38.37 | harlequin516 | Yeah seems like mine too |
| 21:38.45 | konfuzed | so plug their incoming Bell line into pbx and outgoing rj11 to analog channel bank maybe a hardwired fail over for their rj11 line-2 to still be hardwired to Bell copper |
| 21:39.00 | konfuzed | and then the may hem can start |
| 21:39.03 | konfuzed | 8^( |
| 21:39.09 | justinu | you can order answer sup on an analog line? |
| 21:39.17 | justinu | how does that work? |
| 21:39.48 | [av]bani | nope |
| 21:39.55 | [av]bani | your ata has to decode the indications |
| 21:39.56 | *** join/#asterisk sjaak538 (n=sjaaknab@d5c53145.dsl.concepts.nl) |
| 21:40.16 | jbalcomb | if i'm not using anything usb on my server can i get rid of the 'usbcore' module shown by running 'lsmod'? |
| 21:40.17 | [av]bani | analog sux :< |
| 21:40.46 | SpaceBass | konfuzed 911 is important too...people aren't going to be comfortable with out that |
| 21:41.20 | SpaceBass | [av]bani tell me about it...almost done with POTS for ever.... |
| 21:41.37 | SpaceBass | just need to find a sip/iax provider for my business line that can bill my company (and not my credit card) |
| 21:41.39 | [av]bani | \o/ |
| 21:41.53 | Darwin35 | herpies |
| 21:42.01 | b66mer | nice |
| 21:42.09 | Darwin35 | nanpa sucks donkeyballs |
| 21:42.23 | SpaceBass | websae what company? |
| 21:42.34 | websae | Websae |
| 21:42.39 | websae | we are on the level 3 network |
| 21:42.41 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:42.50 | websae | have servers in LA and Michigan |
| 21:42.51 | konfuzed | SpaceBass, I figure to route any 911 direct to a bell line and bypass the internet |
| 21:42.54 | Darwin35 | why should asterisk have to fallow nanpa vertical dialplans |
| 21:43.15 | jbalcomb | hrm.. i guess i'm not the first one to have this 'no audio' problem. http://pastebin.com/627641 |
| 21:43.15 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 21:43.17 | Darwin35 | it sucks thier layout |
| 21:44.14 | jbalcomb | Asterisk reports it's playing the sound, but no audio occurs and no matter how long I wait, it doesn't get to the next line to hang up the call. |
| 21:44.22 | konfuzed | see they cant simply disconnect all their bell services cause then there wont be any internet access at their crappy last mile location |
| 21:44.28 | Darwin35 | update your zaptel and libpri |
| 21:44.44 | Darwin35 | and asterisk |
| 21:45.28 | tmccrary | When I use canreinvite=yes, if the asterisk server loses a connection, both phones go dead. |
| 21:45.34 | Deciphan | can someone give me a quick clue on doing a blind transfer through the dialplan? or does this need to be configured in the phone? |
| 21:45.52 | Darwin35 | look in features.conf |
| 21:45.55 | Darwin35 | its there |
| 21:46.03 | Deciphan | ah.. cool, thanks |
| 21:48.40 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 21:49.33 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 21:50.55 | Darwin35 | WEBsae dod you offer did's and pstn termination |
| 21:51.19 | Darwin35 | or do you get everything threw l3 |
| 21:51.21 | websae | yes |
| 21:51.41 | websae | everything goes through the level 3 network |
| 21:51.47 | websae | since we are on it |
| 21:51.56 | Deciphan | will the blind transfer option work with 1.0.9 cvs head? kinda looks like that's new in 1.2? |
| 21:52.34 | Darwin35 | its for 1.2 branch |
| 21:52.45 | Darwin35 | dont think it was back ported to 1.0.9 |
| 21:52.45 | Deciphan | k.. time to upgrade |
| 21:52.46 | Deciphan | :) |
| 21:52.54 | websae | special deal for asterisk users :) 1.5cents/min US and CANADA with CID....Inbound 1cent/min ($1.50/DID per month).........LNP=FREE :)!!!!! |
| 21:54.01 | loko | websae url? |
| 21:54.22 | GerbilNut | what's the difference between the 1.2 branch and the 1.0 branch? |
| 21:54.42 | brad_mssw | 1.2 == supported, 1.0 == unsupported |
| 21:54.48 | shmaltz | GerbiNut, read the realease nots |
| 21:55.16 | brad_mssw | websae: company? url? |
| 21:55.37 | brad_mssw | websae: iax or sip proxy to traceroute? |
| 21:56.40 | jbalcomb | *RESOLVED* ok, so i did modprobe -r wct4xxp and now my playbacks work. wtf. |
| 21:57.07 | VeNoMouS_ | has anyone here used spandsp and rxfax? |
| 21:57.40 | websae | sip.websae.com |
| 21:57.53 | websae | sip2.websae.com |
| 21:57.54 | VeNoMouS_ | if so have you had corrupted tiffs with it not setting "StripOffsets" |
| 21:58.03 | VeNoMouS_ | im getting that on a few |
| 21:58.05 | *** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1) |
| 21:58.14 | VeNoMouS_ | and it seems libtiff is just aving a fit trying to touch them |
| 21:59.58 | *** part/#asterisk lrizzo (n=luigi@81-174-38-222.f5.ngi.it) |
| 22:00.05 | Darwin35 | LOL just found a user using *@H to provide service to 200 users |
| 22:00.11 | *** join/#asterisk Whisk (n=whisk@whisk.gotadsl.co.uk) |
| 22:03.34 | GerbilNut | anyone gotten two Asterisk servers communicating via IAX2, using switch, sharing dial-plans? I'm getting an interesting "rejected connection attempt from xxx.xxx.xxx.xxx trying to reag 'TBD@default' error |
| 22:05.44 | iGotNoTime | by default my caller ID shows as : The Hyatt Regancy ? |
| 22:05.50 | iGotNoTime | how do I modify that? LOL |
| 22:06.00 | *** part/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 22:06.10 | GerbilNut | why would you want too |
| 22:06.15 | iGotNoTime | LOL |
| 22:06.18 | iGotNoTime | I am not a hotel |
| 22:06.23 | Hmmhesays | ok what is the point of srvlookup=yes, when asterisk doesn't look up the freaking srv records when you specify a host in the dialplan |
| 22:06.24 | Hmmhesays | ARG |
| 22:06.31 | iGotNoTime | kinda funny though :D |
| 22:06.49 | iGotNoTime | my brother refused to answer till I called from my cellphone |
| 22:08.03 | justinu | asterisk's SRV support sucks |
| 22:08.09 | justinu | :( |
| 22:11.35 | iGotNoTime | is there a way to change it? |
| 22:11.56 | iGotNoTime | Even the wiki has people asking with no replies |
| 22:14.25 | SpaceBass | iGotNoTime using asterisk@home |
| 22:14.29 | x86 | any way to record a call (both parties like MixMonitor does), but stop recording when the call is hung up by either party, that way i can do post-call processing with the 'h' extension after one or both parties have hung up? |
| 22:18.23 | [av]bani | hmm |
| 22:18.28 | [av]bani | asterisk[8316] general protection rip:2aaaae2771d9 rsp:400c2080 error:0 |
| 22:18.31 | [av]bani | not good :< |
| 22:20.24 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 22:22.18 | *** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-84-152.d-ip.magma.ca) |
| 22:27.02 | SplasPood | Asterisk CVS-10/28/03-07:16:52 thats from 2003?? |
| 22:29.08 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:29.25 | Hmmhesays | removing sip retransmissions and retrydial works to my advantage now |
| 22:29.26 | Hmmhesays | weeee |
| 22:29.50 | Katty | Hmmhesays: make greyhound run between me and bloomington, in |
| 22:30.07 | Hmmhesays | take the train |
| 22:30.19 | iGotNoTime | do I simply put the following anywhere in extensions.conf? >>> callerid = "Mark Spencer" <256 428-6000> |
| 22:30.26 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 22:30.48 | Katty | Hmmhesays: it doesn't go there. |
| 22:30.58 | Hmmhesays | take it to somewhere the bus goes then |
| 22:33.34 | Assid | Mar 29 04:02:41 WARNING[6849]: chan_agent.c:1842 __login_exec: Extension '3001@default' is not valid for automatic login of agent '1001' |
| 22:33.39 | Assid | can someone help me on this? |
| 22:34.02 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 22:34.34 | VeNoMouS_ | blah @ agent groups! |
| 22:38.44 | Assid | why the hell is it @default |
| 22:42.14 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 22:43.59 | iGotNoTime | Assid I dunno I am new |
| 22:49.10 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 22:50.27 | *** join/#asterisk JSabines (i=JSabines@201.138.136.215) |
| 22:50.52 | *** join/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 22:51.44 | Assid | <PROTECTED> |
| 22:51.51 | Assid | but it doesnt set htta as the name |
| 22:51.54 | *** join/#asterisk oej (n=oej@gateway.digium.com) |
| 22:54.09 | Shaun2222 | can anybody recommend a good sip provider? looking for one with 24/7 phone support that actually has people capable of fix'ing problems on off hours. |
| 22:54.28 | justinu | talk to websae |
| 22:56.50 | *** join/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 22:56.56 | Shaun2222 | justinu: their domain is registered to a hotmail address and it's only been registered sinec auguest 2005 |
| 22:57.02 | Shaun2222 | dont think i'll be using them... |
| 22:57.10 | justinu | fine by me |
| 22:58.33 | iGotNoTime | LOL |
| 22:58.51 | iGotNoTime | off topic... justinu how long have you run *? |
| 22:58.59 | iGotNoTime | you seem to know alot about it |
| 22:59.10 | iGotNoTime | not asking for help, just curious :) |
| 22:59.11 | Shaun2222 | justinu: is websae your company? |
| 22:59.14 | justinu | nope |
| 22:59.22 | justinu | been running asterisk since october 05 |
| 22:59.36 | justinu | level3 backend |
| 22:59.42 | justinu | i have a few PBX installs out there also |
| 23:00.06 | iGotNoTime | Shaun222 you should probably know that you can update the details of your domain registration at any given time |
| 23:00.23 | iGotNoTime | I have changed my contact email 3x due to spam |
| 23:00.41 | iGotNoTime | you get spammed by hosting companies when you have several domains :) |
| 23:01.29 | *** join/#asterisk pixolex (n=chatzill@87-196-250-204.net.novis.pt) |
| 23:02.03 | iGotNoTime | justinu: with me being so new, all the lingo if very tough to comprehend, then many in here seem elitists and unwilling to help the guy who just installed... Besides the wiki, do you have any suggested links to help me learn faster? |
| 23:02.15 | justinu | no, just hang around and ask questions |
| 23:02.18 | justinu | absorb what you can |
| 23:02.20 | iGotNoTime | hehe |
| 23:02.31 | iGotNoTime | yes I have been logging and searching logs before asking :) |
| 23:02.35 | justinu | good |
| 23:02.39 | iGotNoTime | thanks for the tip :) |
| 23:02.44 | justinu | np |
| 23:07.06 | rollergrrl | Is there a way, in the dialplan, that I can check the return condition of an app I run? |
| 23:07.26 | tzanger | nope |
| 23:07.44 | tzanger | I think there should be but have been unable to convince the powers that be of that |
| 23:07.55 | rollergrrl | ugh |
| 23:08.14 | tzanger | currently you must check an app-specific ${VAR}, screw around with priority jumping or just wing it |
| 23:08.33 | justinu | heh |
| 23:08.42 | rollergrrl | so if there is an app that doesn't have a var, I'll have to go in and add one |
| 23:08.43 | rollergrrl | great |
| 23:08.43 | justinu | the powers that be seem pretty close minded |
| 23:08.46 | *** join/#asterisk willcampos123 (n=willcamp@198.87.100.3) |
| 23:08.50 | *** part/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 23:08.56 | tzanger | rollergrrl: pretty much |
| 23:09.16 | willcampos123 | Anyone knows how to write the IP address of the origin caller in the cdrs? |
| 23:09.22 | tzanger | justinu: no, I understand their unwillingness to change such a core concept |
| 23:09.29 | tzanger | willcampos123: use the user field |
| 23:10.19 | willcampos123 | but how to read the IP ? |
| 23:10.30 | tzanger | check out the channel variables and see what's there |
| 23:11.56 | Shaun2222 | iGotNoTime: whats that have to do with the date the domain was reged or the shaddy hotmail address... |
| 23:11.58 | willcampos123 | Thanks, I am going to read then a little bit about that... cause i dont know whata variable use |
| 23:15.26 | *** join/#asterisk kend (n=chatzill@host-64-65-199-187.man.choiceone.net) |
| 23:16.27 | *** join/#asterisk angler_ (n=johnb@199.227.185.58) |
| 23:16.55 | kend | Hmmm. My PRI's down (Sangoma A104d) -- any ideas? "The H100 slave has lost its framing on the bus!" and "The CT_C8_A clock behavior does not conform to the H.100 spec!" As far as I can tell, nothing's changed -- and I've rebooted about a zillion times. |
| 23:19.49 | ManxPower | kend, no idea. not many people here run sangoma |
| 23:20.02 | kend | True 'nuff. |
| 23:20.57 | mog_work | sounds like your losing sync |
| 23:21.04 | mog_work | you getting all your interuppts? |
| 23:21.36 | kend | mog_work: Not quite sure what you mean. Not seeing any lost interrupt messages in syslog. |
| 23:21.56 | mog_work | you wouldnt |
| 23:22.01 | mog_work | what kind of machine this in? |
| 23:22.07 | *** join/#asterisk xtr-II (i=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 23:22.09 | Qwell[] | mog_work: You support Sandoma now? :p |
| 23:22.13 | Qwell[] | Sangoma rather* |
| 23:22.21 | Qwell[] | right-o |
| 23:22.26 | mog_work | should have bout a digium card ^_^ |
| 23:22.33 | Qwell[] | indeed |
| 23:22.34 | mog_work | but meh we will get him next time |
| 23:22.34 | russellb | mog_work: that'a boy |
| 23:22.40 | kend | Generic AMD w/3100. |
| 23:22.57 | mog_work | try a different slot? |
| 23:23.03 | mog_work | and does this happen under heavy load |
| 23:23.05 | mog_work | or always |
| 23:23.15 | kend | [Note: bought Sangoma after several bad experiences with TDM400's.] |
| 23:23.24 | mog_work | : ( |
| 23:23.51 | kend | Hmmm. Different slot -- interesting idea. Load's a non-issue, and always happening. It's DOWN. |
| 23:24.23 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.16.10.telnor.net) |
| 23:24.58 | kend | So, since we're talking brand 'n stuff: are the Digium T1 cards a) reliable, and b) truly good at echo cancellation? *ponders buying one and having it shipped overnight* |
| 23:25.29 | mog_work | or send you to people who could |
| 23:25.38 | mog_work | where are you kend? |
| 23:25.47 | kend | mog_work: New Hampshire. |
| 23:26.01 | kend | mog_work: working on this when my brand-spanking-new baby is in hospital. *unhappy* |
| 23:26.03 | ManxPower | kend, I would say "no" for both questions under some situations. |
| 23:26.20 | ManxPower | Digium cards are VERY VERY sensitive to jitter in interrupt latency |
| 23:26.27 | mog_work | : ( |
| 23:26.45 | kend | ManxPower: Well, damn. I mean, honestly, I don't care *whose* I buy -- I just want it to work. :( |
| 23:27.05 | ManxPower | kend, people say sangoma make great cards. |
| 23:27.10 | CrashHD | ~backtrace |
| 23:27.12 | jbot | [backtrace] a debugging tool that is invaluable when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read README.backtrace) |
| 23:27.12 | file | too many variables in the world to have it work under all situations |
| 23:27.17 | ManxPower | it's just that THIS channel is not a good place to get support for them |
| 23:27.32 | file | Digium work great for some, Sangoma work great for some |
| 23:27.36 | ManxPower | kend, sangoma always claims to have great tech support, use it and see. |
| 23:27.39 | kend | ManxPower: as far as I can tell, they do. True 'nuff -- look elsewhere. *ponders what time it is in Sangoma's area code* |
| 23:27.51 | ManxPower | kend, they are in eastern timezone |
| 23:27.53 | file | Sangoma is in Ontario so it's 6:27PM there |
| 23:27.57 | ManxPower | 6:28pm |
| 23:27.59 | kend | ManxPower: Well, damn. |
| 23:28.12 | *** join/#asterisk mrbnet (n=sureal@CPE-24-94-219-49.mn.res.rr.com) |
| 23:28.36 | mog_work | digium still has light on ^_^ |
| 23:29.25 | kend | Wow -- still in office. |
| 23:29.42 | mog_work | we dont close till 7 eastern officially |
| 23:29.52 | mog_work | err central |
| 23:30.10 | mog_work | but my side of building stays longer |
| 23:30.15 | mog_work | comes in later though |
| 23:31.19 | mrbnet | if I signup for broadvoice.com service will that allow simultaneous calls or do I need to signup for multiple accounts? |
| 23:32.06 | mog_work | i think all of broadvoices plans are like "virtual lines" |
| 23:32.23 | mog_work | you can get termination from people like nufone etc that are just the minutes |
| 23:33.15 | mrbnet | thanks |
| 23:38.21 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 23:38.40 | *** join/#asterisk JSabines (i=JSabines@201.138.136.215) |
| 23:40.30 | CrashHD | what are the base modules I need for sip/iax/pri trunking connections? |
| 23:40.53 | Qwell[] | chan_sip, chan_iax, and chan_zap might help |
| 23:41.04 | mog_work | lol |
| 23:41.12 | mog_work | and chan_mgcp |
| 23:41.21 | Qwell[] | and?! |
| 23:41.22 | CrashHD | any other pretty basic modules I should make sure are loaded (I figured the three you said) |
| 23:41.32 | mog_work | why not just load al? |
| 23:41.34 | mog_work | er all |
| 23:41.40 | Qwell[] | chan_al? |
| 23:41.41 | mog_work | its only memory |
| 23:41.45 | mog_work | and not much |
| 23:42.05 | CrashHD | I'm trying to reduce and optimize the problems that could occur with the server |
| 23:42.14 | CrashHD | damn thing is crashing randomly |
| 23:42.17 | mog_work | if you dont use them they cant cause them |
| 23:42.20 | CrashHD | just some troubleshooting tactics |
| 23:42.24 | mog_work | other than res |
| 23:42.28 | mog_work | why not just run asterisk -g |
| 23:42.35 | mog_work | and look at the core as to why it crashed |
| 23:42.45 | CrashHD | how do I look at the core I guess would be the question |
| 23:42.47 | CrashHD | -g is being run |
| 23:43.05 | mog_work | gdb |
| 23:43.14 | mog_work | there should be a core.NUMBER |
| 23:43.18 | mog_work | on your machine |
| 23:43.19 | CrashHD | ya I have those |
| 23:43.21 | *** join/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 23:43.23 | mog_work | probably several |
| 23:43.23 | CrashHD | binary files |
| 23:43.26 | mog_work | yeah |
| 23:43.31 | mog_work | run gdb asterisk |
| 23:43.34 | CrashHD | ~gdb |
| 23:43.35 | jbot | from memory, gdb is The GNU Debugger. URL: http://www.gnu.org/software/gdb/ or http://sources.redhat.com/gdb/ |
| 23:43.38 | mog_work | then core-file /path/to/core |
| 23:43.48 | mog_work | there is a guide in asterisk docs |
| 23:43.50 | orlock | Hmm... |
| 23:43.54 | CrashHD | ok |
| 23:43.56 | mog_work | asterisk-source/docs |
| 23:44.01 | CrashHD | I don't have the gdb installed |
| 23:44.03 | orlock | i get 404 on outbound calls, i think my dialplans/call routes are broken |
| 23:44.06 | CrashHD | so I'll work on that |
| 23:44.10 | CrashHD | thank you mog |
| 23:44.14 | mog_work | no prob |
| 23:44.20 | *** join/#asterisk finchy (n=finchy@65.83.56.131) |
| 23:44.28 | mog_work | and taking out modules isnt a bad thing |
| 23:44.34 | mog_work | just want to solve the problem |
| 23:44.35 | mog_work | not mask it |
| 23:44.58 | CrashHD | *nods* thank you, good advice |
| 23:46.00 | mog_work | best of luck, feel free to come back and bug me or others when you have more info |
| 23:46.31 | CrashHD | thanks mog |
| 23:46.35 | CrashHD | I just read the core file |
| 23:46.42 | CrashHD | what would I be looking for specifically? |
| 23:46.44 | CrashHD | the last line? |
| 23:47.06 | Qwell[] | CrashHD: check out backtrace.txt or README.backtrace |
| 23:47.13 | CrashHD | ok Qwell |
| 23:47.17 | mog_work | well you want to find which module of asterisk is acting up |
| 23:47.17 | CrashHD | thank you |
| 23:47.20 | mog_work | also |
| 23:47.25 | mog_work | if you want to have clean debug files |
| 23:47.30 | mog_work | you will have to rebuild asterisk |
| 23:47.34 | mog_work | as per those readmes |
| 23:47.35 | CrashHD | it's the chan_iax2 I think |
| 23:47.41 | CrashHD | I'll read those readmes |
| 23:47.59 | CrashHD | see where I get on my own |
| 23:48.02 | CrashHD | thanks for the pointers |
| 23:48.09 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-160.apple.com) |
| 23:48.44 | Qwell[] | That quit msg was so fitting |
| 23:48.54 | Qwell[] | especially since Digium has Digium/Asterisk labelled screwdrivers.. |
| 23:49.11 | mog_work | heh |
| 23:49.24 | mog_work | and a pocket knife |
| 23:49.33 | Qwell[] | yes, but are they COMBINED?! |
| 23:49.40 | mog_work | nope |
| 23:49.43 | mog_work | seperate |
| 23:49.43 | Qwell[] | well then |
| 23:49.51 | mog_work | screwdriver in the knife sucks |
| 23:49.55 | *** join/#asterisk _Soul_ (n=Soul@87-196-11-170.net.novis.pt) |
| 23:49.56 | Qwell[] | heh, they usually do |
| 23:50.22 | VeNoMouS_ | qwell[] give me a hand |
| 23:50.32 | Qwell[] | VeNoMouS_: ? |
| 23:50.34 | _MartinCabrera_ | Someone has solved random lockups on GrandStream phones (GXP and BT100) ? |
| 23:50.39 | VeNoMouS_ | to kill all the ppl who made libtiff |
| 23:50.58 | Qwell[] | _MartinCabrera_: yes. by switching to a different brand |
| 23:51.08 | _MartinCabrera_ | :) |
| 23:51.16 | Qwell[] | Have you upgraded the firmware to the latest? |
| 23:51.39 | _MartinCabrera_ | i'm running latest firmware and * 1.2.5 |
| 23:51.49 | VeNoMouS_ | try 1.2.6 |
| 23:51.50 | VeNoMouS_ | >:P |
| 23:52.00 | CrashHD | hmm fun |
| 23:52.07 | CrashHD | so it is iax2 module that is giving me the problems |
| 23:52.23 | Qwell[] | CrashHD: what version of *? |
| 23:52.27 | CrashHD | 1.2.5 |
| 23:52.36 | Qwell[] | VeNoMouS_: ^ |
| 23:52.48 | CrashHD | *nods* |
| 23:53.35 | _MartinCabrera_ | which brand do you suggest me instead of GrandStream? Polycom maybe? |
| 23:53.40 | *** join/#asterisk willcampos123 (n=willcamp@198.87.100.3) |
| 23:53.41 | VeNoMouS_ | cisco! |
| 23:53.42 | Qwell[] | polycom, cisco.. |
| 23:53.43 | willcampos123 | Does any one know how to record the IP address of the call that is coming in |
| 23:53.44 | CrashHD | I only see an ani problem fixed for chan_iax2 with 1.2.6 |
| 23:53.44 | willcampos123 | on the asterisk. |
| 23:53.46 | justinu | polycom is good |
| 23:53.47 | Qwell[] | I <3 cisco, personally |
| 23:53.47 | VeNoMouS_ | 7940 |
| 23:54.00 | VeNoMouS_ | i have 2 7912 & a 7940 on my desk |
| 23:54.01 | VeNoMouS_ | heh |
| 23:54.28 | VeNoMouS_ | man 7912's are fulgy |
| 23:54.32 | VeNoMouS_ | fugly |
| 23:55.30 | justinu | _MartinCabrera_: are you still having trouble there with those BT101s? |
| 23:55.45 | Qwell[] | VeNoMouS_: well, if you don't want it |
| 23:58.01 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 23:58.07 | willcampos123 | Does anyone know hot to get the IP addres of a SIP client and put it in a cdr.userfield? |
| 23:58.56 | VeNoMouS_ | lol |
| 23:59.00 | VeNoMouS_ | man if the photo copier breaks |
| 23:59.03 | VeNoMouS_ | do not offer to help |
| 23:59.09 | VeNoMouS_ | well do not remove ink thingie |
| 23:59.49 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com) |
