| 00:00.06 | paryl | furiousgeorge: yeah... five or so |
| 00:00.08 | delmar | FuriousGeorge, but when the call is placed... SIP to remote SIP.. echo |
| 00:00.22 | delmar | FuriousGeorge, meh. ill figure it. |
| 00:00.28 | FuriousGeorge | paryl: do you wait(3) when you answer a call to give the voip time to "settle" |
| 00:01.07 | paryl | furious: actually no. now, i'm talking about an incoming call from pstn here |
| 00:01.17 | paryl | over a tdm400p |
| 00:02.23 | paryl | wait(3)... that isn't 3 seconds... right? |
| 00:02.27 | FuriousGeorge | delmar: im no expert: but i was saying, for echo there must be delay, but i dont see how the digitized sound could "leak" so have you tried other clients, or other handsets or curly cords to the handsets (thats a long shot). |
| 00:02.35 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 00:02.45 | linagee | you say potawto, i say tatertots |
| 00:03.00 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:03.48 | FuriousGeorge | paryl: if you answer() and wait() ive heard that gives asterisk a moment to settle things. use it with echo training, and the calling pary will not really know the difference |
| 00:03.51 | delmar | FuriousGeorge, well the phones are not that great and im suspecting them actually... they are PA168X based phoens at each end.... i want to take one of my 2 grandstreams to the remote site and test that way...at some point. |
| 00:04.15 | FuriousGeorge | or even just dl a free client like xten and give that a whirl |
| 00:04.32 | delmar | FuriousGeorge, now that i have this PSTN echo fixed, im gonna have some lunch then take a look at it. bbs. |
| 00:05.36 | FuriousGeorge | lemme know how that goes. i find that with any problem you just gotta swap till you find the lowest common denominator, you know |
| 00:06.15 | paryl | ok... i ended up recompiling and tried mark2 with agressive cancellation, and as far as i can tell, that worked |
| 00:06.24 | paryl | but everythign i've read recommended against it |
| 00:06.44 | FuriousGeorge | paryl: problem with aggressive cacellation, i think, is that if both people talk at the same time you get a walkie talkie effect |
| 00:07.00 | paryl | walkie talkie? |
| 00:07.00 | FuriousGeorge | try answer (2) wait (2) |
| 00:07.38 | FuriousGeorge | no one hears anyone when both talk at the same time, but thats just what ive /heard/ |
| 00:08.05 | shido6 | thats what memory is for |
| 00:08.28 | FuriousGeorge | shido6: whats that? making me recollect incorrectly :) |
| 00:08.43 | shido6 | hah |
| 00:09.20 | FuriousGeorge | i should have a dedicated chip in my left brain dedicated to storing factoids |
| 00:09.23 | FuriousGeorge | and phone numbers |
| 00:09.33 | FuriousGeorge | and i need a usb port to be able to move that data |
| 00:09.40 | FuriousGeorge | wireless is too unreliable |
| 00:11.09 | paryl | ok... i'm going to let that go tomorrow and see if anyone notices it |
| 00:11.15 | paryl | thanks for the help furious |
| 00:11.49 | perd | is there a way to make an AGI script sit there and execute all the time.. like polling a queue, then once something hits the queue it starts dialing extensions and playing messages and crap? |
| 00:12.11 | perd | from what i see agi scripts have to be activated by someone hitting an extension and you loading them with application or res_perl or something |
| 00:12.19 | test34 | asterisk is not very helpful with its error messages when you make a mistake in your diaplan |
| 00:13.19 | FuriousGeorge | test34: insert some noops when you debug it |
| 00:13.44 | test34 | ok I will try that thanks FuriousGeorge |
| 00:14.00 | FuriousGeorge | *for when you debug it |
| 00:14.08 | FuriousGeorge | noop(im over here now) |
| 00:14.18 | test34 | ok |
| 00:14.21 | FuriousGeorge | noop(buggaluggaloo) |
| 00:15.40 | test34 | the error message almost sounds like asterisk can't use the hardware, ie: Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN' |
| 00:16.13 | FuriousGeorge | test34: what does ztcfg -vv say about the channel |
| 00:16.49 | FuriousGeorge | vv looks alot like w in this font :) |
| 00:17.08 | test34 | its working now since I fixed a mistake in extensions.conf |
| 00:17.11 | test34 | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 00:17.26 | sjobeck99 | hi, all, what kind of you you having taking faxes in over ulaw, and putting to POTS line from TDM card FXO? |
| 00:17.29 | ManxPower | BTW, can someone try using the forward message function of voicemail in CVS-HEAD and tell me if it's broken for you too? |
| 00:17.42 | sjobeck99 | sorry .... "what kind of luck" |
| 00:17.43 | FuriousGeorge | test34: did you forget that fxs use fxo signalling :) |
| 00:18.24 | FuriousGeorge | oh wait i misunderstood, you said extensions.conf |
| 00:18.38 | luite | hm |
| 00:18.45 | luite | oops, wrong channel :) |
| 00:21.00 | FuriousGeorge | sjobeck99: im scared to try in a professional env, and i dont take/send faxes from home |
| 00:21.10 | *** join/#asterisk Inv_arp (i=junya@adsl-144-17-25.mia.bellsouth.net) |
| 00:21.40 | sjobeck99 | FuriousGeorge: i am very scared too. i really dont like where we are right now, but ......... |
| 00:25.48 | FuriousGeorge | sjobeck99: i tried getting faxdetection goiong in 1.0.9 and even that didnt work for me |
| 00:27.54 | *** join/#asterisk remibreval (i=Remek@pro75-3-82-234-175-208.fbx.proxad.net) |
| 00:28.03 | remibreval | hi everybody !!! |
| 00:28.18 | FuriousGeorge | hello dr. nick |
| 00:28.49 | remibreval | I have a little trouble on my * and I'm searching if the pb is from my provider of my configuration |
| 00:28.56 | remibreval | can someone help please ? |
| 00:29.12 | remibreval | the pb is post here (to avoid cut/paste) http://forums.digium.com/viewtopic.php?t=2354&sid=d14b86479ee4415e28fb6119062f168a |
| 00:29.42 | *** join/#asterisk orlok (n=jwr@202-44-174-4.nexnet.net.au) |
| 00:29.50 | remibreval | I'm going sick and crazy with this pb as I don't know were the problem is from :-( |
| 00:30.11 | ManxPower | try posting the pb on the pb |
| 00:30.13 | ManxPower | ~pastebin |
| 00:30.16 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 00:31.34 | remibreval | ok this is the pastebin http://pastebin.ca/28059 |
| 00:33.03 | *** join/#asterisk remmo (n=rem@smack.isp.net.au) |
| 00:33.05 | FuriousGeorge | remibreval: i dont use a@h because i didnt want to limit myself, so im not sure how config goes. it seems that when you pick up that extension something is being dialed (a macro is being called) |
| 00:33.10 | FuriousGeorge | pastebin extensions.conf |
| 00:33.12 | remmo | hey all |
| 00:33.56 | remibreval | extension.conf is pastbinded at the end of the post |
| 00:34.02 | *** join/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi) |
| 00:34.11 | Ariel_ | ManxPower, the project I was talking about can be done remotely. And does not have to be done this week. |
| 00:34.34 | *** join/#asterisk marc324 (n=marc3234@206-248-128-180.dsl.teksavvy.com) |
| 00:34.49 | ManxPower | Ariel_, I'm way overworked as it is. |
| 00:34.52 | remibreval | it is line 34 |
| 00:34.55 | remibreval | extention.conf |
| 00:35.31 | ManxPower | Ariel_, But if they are willing to pay $120/hr, I'll consider it if it's at least 4 hours worth of work |
| 00:35.31 | remibreval | I just can't understand what's happening when I pick up !! |
| 00:35.33 | Ariel_ | ManxPower, ok then do you have any cheat sheet connecting to a Meridian opt 81 system? via a T1 not pri. |
| 00:36.08 | Ariel_ | ManxPower, actually I think this is going to be at least that mush. |
| 00:36.35 | remibreval | I think, first Goto(ext-local,7364,1) is called |
| 00:36.35 | ManxPower | Ariel_, should not be hard to do. span=1,0,0,esf,b8zs |
| 00:36.37 | remibreval | then |
| 00:36.47 | remibreval | Macro(exten-vm,7364@default,7364) |
| 00:36.55 | ManxPower | You prolly need a T-1 crossover cable. |
| 00:37.01 | remibreval | this works I think. The problem is after... |
| 00:37.03 | ManxPower | chances are the signaling is FXS_KS |
| 00:37.09 | remibreval | but I don't know were |
| 00:37.15 | ManxPower | But I CANNOT help with the Nortel side, only the Asterisk side. |
| 00:37.30 | Ariel_ | ManxPower, ok. I will give it a shot. I know they have there own nortel tech on site |
| 00:37.46 | ManxPower | Ariel_, It should be dead simple. |
| 00:38.05 | ManxPower | Ariel_, Would the nortel assume it's connected to the telco? |
| 00:38.29 | remibreval | FuriousGeorge, do you have any idea of what can happend ? |
| 00:38.39 | Ariel_ | I guess so. I have not gotten a normal t1 to connect from a nortel other then a pri. |
| 00:38.49 | FuriousGeorge | what context is your voip-provider in now? |
| 00:38.49 | ManxPower | Ariel_, if the nortel tech is any good, I might hire him for a couple of other jobs |
| 00:38.55 | Ariel_ | But the main thing they need the asterisk is for wireless hitachi phones |
| 00:38.56 | TedC | Has anybody gotten the M(<macro) argument working with Dial? |
| 00:39.05 | remibreval | it is ext-did |
| 00:39.06 | justinu | yes |
| 00:39.25 | TedC | is there a way to debug it? I can't tell if the macro is even being called |
| 00:39.50 | TedC | I see nothing on console about it, even in verbose 9 |
| 00:40.14 | justinu | tedc: put some noops in there |
| 00:40.22 | justinu | if they're still not printing... you've got other problems |
| 00:40.25 | remibreval | TedC, you can use the Asterisk console |
| 00:40.55 | remibreval | TedC, use asterisk -vvvvvvvrc |
| 00:40.56 | FuriousGeorge | remibreval: i dont have time to follow the tangled mess of logic you have there. i recommend changing that goto() line with a simple dial and seeing if that works first |
| 00:41.07 | FuriousGeorge | i gotta run atm |
| 00:41.08 | ManxPower | NEVER USE -r and -c AT THE SAME TIME |
| 00:41.10 | FuriousGeorge | ill be back later |
| 00:41.22 | *** join/#asterisk punk0 (n=edward@dsl-200-95-15-204.prod-infinitum.com.mx) |
| 00:41.24 | Ariel_ | did not know you could use the r and c together |
| 00:41.27 | FuriousGeorge | ManxPower: ive never tried but now im tempted to |
| 00:41.27 | punk0 | hi all |
| 00:41.38 | FuriousGeorge | ManxPower: wanna just tell me what happens |
| 00:41.38 | remibreval | FuriousGeorge, You mean, directly [ext-did] |
| 00:41.39 | remibreval | include => ext-did-custom |
| 00:41.39 | remibreval | exten => |
| 00:41.40 | justinu | lol |
| 00:41.54 | FuriousGeorge | remibreval: yeah and then you call macros |
| 00:41.56 | punk0 | i have a problem whit my new asterisk and the zap channel |
| 00:41.59 | ManxPower | FuriousGeorge, I don't recall exactly. I think it depends on if Asterisk is already running. |
| 00:42.03 | FuriousGeorge | i got that far and relaized i gotta go out to dinner |
| 00:42.05 | remibreval | I mean, I put in my ext-did, dial(MyExtention) ? |
| 00:42.20 | FuriousGeorge | ManxPower: ill make a mental note not to try it |
| 00:42.23 | ManxPower | -r = connect to an already running instance of asterisk..... -c = start a new instance of Asterisk in the corefround |
| 00:42.40 | FuriousGeorge | remibreval: ill give you the exact line but then i gotta run |
| 00:42.41 | ManxPower | corefound = foreground |
| 00:43.01 | remibreval | FuriousGeorge, ok really thanks a lot !! |
| 00:43.02 | TedC | justinu: Nothing from the NoOp |
| 00:43.13 | TedC | I've got plenty of other Application output, just not from that Macro |
| 00:43.48 | ManxPower | telling someone to use -r and -c togather is sort of like telling someone to use the "r" option of dial or the "tT" option to dial. It's pretty much the same as telling someone that your computer has 1.4Ghz of RAM. |
| 00:44.01 | TedC | the dial command goes through without executing the Macro at all. |
| 00:44.07 | ManxPower | Everyone chuckles to themselves and mutters "moron" under their breath. |
| 00:44.08 | Ariel_ | ROFL |
| 00:45.45 | remibreval | ManxPower, why not use r and -v -c at same time ? |
| 00:45.51 | FuriousGeorge | remibreval: the dial is easy, you should really learn at least this dialplan command to configure it http://pastebin.ca/28061 |
| 00:46.16 | TedC | it looks like if you -r and -c, the -c is ignored. |
| 00:46.22 | ManxPower | remibreval, ManxPower -r = connect to an already running instance of Asterisk and -c = start a new instance of Asterisk in the foreground |
| 00:46.26 | FuriousGeorge | remibreval: 7634 is your sip extension from sip.conf right? |
| 00:46.35 | remibreval | ok. I promise I will. I'm also writing a white paper about it. I'm still learnign It. Thanks a lot again !!! |
| 00:46.40 | remibreval | right |
| 00:46.44 | ManxPower | You can't run Asterisk in the foreground AND connect to an Asterisk running in the background at the same time. |
| 00:46.45 | justinu | LOL |
| 00:46.52 | justinu | why not?? |
| 00:47.01 | FuriousGeorge | remibreval: lemme know how it goes, bbl |
| 00:47.21 | remibreval | ok manxpower, sorry for my confusion, I picked it up from wiki :-( |
| 00:47.22 | ManxPower | I believe the -r will override the -c |
| 00:47.34 | FuriousGeorge | ManxPower: i thought you were talking about asterisk -rvc before |
| 00:47.45 | remibreval | ok FGeorges. I'm doing it right now |
| 00:47.56 | ManxPower | FuriousGeorge, the -v is harmless |
| 00:48.09 | FuriousGeorge | you can asterisk -cv in one shell then asterisk -rv in another, no? |
| 00:48.32 | test34 | how can I extract the area code from ${CALLERIDNUM} ? |
| 00:49.10 | ManxPower | FuriousGeorge, of course. |
| 00:49.44 | FuriousGeorge | callerid number is always the same length right? cant you ${CALLERIDNUM:-7} or something if its always 10 digits |
| 00:49.54 | FuriousGeorge | ManxPower: just making sure we are on the same page |
| 00:50.05 | FuriousGeorge | remibreval: dont forget to reload the dialplan |
| 00:50.13 | remibreval | yep sure |
| 00:50.14 | remibreval | :) |
| 00:50.22 | ManxPower | FuriousGeorge, using r and c should not blow anything up, it's just not logical to use them togather. one will override the other. |
| 00:50.22 | FuriousGeorge | so im still here did it work? |
| 00:50.37 | FuriousGeorge | ManxPower: makes sense |
| 00:50.41 | FuriousGeorge | definatly ilogical |
| 00:50.46 | ManxPower | It's like trying to start a fire with water. |
| 00:50.53 | test34 | FuriousGeorge, ":" is for regex right ? |
| 00:51.22 | justinu | no, it's a substring |
| 00:51.33 | FuriousGeorge | : tells it strip digits, im always stripping prefices but i think :-X strips X digits from the "right" side of the number |
| 00:51.45 | test34 | ahh ok |
| 00:51.48 | test34 | thanks |
| 00:52.21 | FuriousGeorge | gotta run, lemme know how it goes remibreval |
| 00:53.09 | remibreval | I'm just calling... |
| 00:53.38 | remibreval | humm.. this way nothing happend |
| 00:53.43 | remibreval | nothing on console |
| 00:53.46 | ManxPower | test34, This EXACT issue has been discussed on the mailing list over the past couple of days |
| 00:53.56 | remibreval | i tries to call and it stops to call after a while |
| 00:54.46 | remibreval | strange isn't it ? |
| 00:54.47 | ManxPower | remibreval, you're not doing something stupid like using Asterisk@Home or AMP or something like that, are you? |
| 00:55.01 | remibreval | no, I'm editing .conf files |
| 00:55.12 | remibreval | but yes, using asterisk at home |
| 00:55.16 | ManxPower | Asking for help on those two things is like a Mac user coming into a MS Windows forum and asking for help. |
| 00:55.24 | ManxPower | remibreval, look at your logger.conf |
| 00:55.29 | remibreval | AMP + manual conf = pb because of database, but I know this |
| 00:55.33 | ManxPower | /etc/asterisk/logger.conf |
| 00:55.51 | remibreval | lol manxPower !! |
| 00:56.02 | Ariel_ | remibreval, there is a #amportal section here. And the ones you need to edit are the extensions_custom.conf files |
| 00:56.35 | remibreval | I'm in "full" in the loger |
| 01:01.49 | remibreval | OK, I may reboot all my servers and connections for test. I come back to tell you if it change something |
| 01:04.11 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 01:15.44 | *** join/#asterisk MajestiK (n=MajestiK@S0106000024c058cc.ed.shawcable.net) |
| 01:17.21 | *** join/#asterisk gman55s1f (n=joe@128.100.33.138) |
| 01:18.46 | *** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 01:20.18 | *** part/#asterisk bumblefsck (n=bumblefs@69-160-145-156.ontrca.adelphia.net) |
| 01:24.16 | *** join/#asterisk pyrosteve (n=steve@wsip-70-183-114-254.no.no.cox.net) |
| 01:24.24 | *** part/#asterisk pyrosteve (n=steve@wsip-70-183-114-254.no.no.cox.net) |
| 01:24.27 | *** join/#asterisk pyrosteve (n=steve@wsip-70-183-114-254.no.no.cox.net) |
| 01:25.37 | *** join/#asterisk kn0x (n=nunya@tor/session/x-54aadc9b838ea05d) |
| 01:25.40 | kn0x | hello |
| 01:25.54 | kn0x | problem |
| 01:26.55 | kn0x | http://pastebin.ca/28065 |
| 01:27.08 | kn0x | what are all those reload errors about... |
| 01:27.14 | kn0x | and then i cant do any playback |
| 01:27.26 | kn0x | its looking for a ulaw file format??? wtf? |
| 01:27.30 | kn0x | im very confused |
| 01:27.32 | robbie2 | hi all |
| 01:27.37 | robbie2 | i have asterisk on my home gateway |
| 01:27.48 | robbie2 | from behind it i have sipura phone |
| 01:27.54 | robbie2 | my friend is using x-lite from outside |
| 01:28.00 | robbie2 | we can call each other |
| 01:28.04 | robbie2 | but i cant hear him |
| 01:28.08 | robbie2 | he cant hear me |
| 01:28.13 | robbie2 | he cna hear me |
| 01:28.15 | robbie2 | can even |
| 01:28.21 | robbie2 | and he can leave voicemail for me |
| 01:28.30 | robbie2 | but i cant hear him when we call each other |
| 01:28.39 | justinu | you need to set externip in your sip.conf |
| 01:28.41 | robbie2 | which is rather fun, i can abuse him and he cant answer back :) |
| 01:28.44 | pyrosteve | robbie2: nat is not your friend |
| 01:29.00 | robbie2 | his nat ? |
| 01:29.31 | kn0x | justinu- note that it is not availible in 1.0.9 |
| 01:29.31 | kn0x | externip |
| 01:29.39 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
| 01:29.47 | justinu | oh, then upgrade : |
| 01:29.54 | justinu | because it can't work right without that |
| 01:29.57 | robbie2 | im running 1.2 |
| 01:30.05 | kn0x | then ur fine |
| 01:30.16 | kn0x | just wanted to point it out.. |
| 01:32.19 | kn0x | phew.... mysql finally finished compiling lol |
| 01:32.58 | justinu | robbie2: also, make sure canreinvite=no in sip.conf |
| 01:35.09 | *** join/#asterisk apardo (n=apardo@212.Red-83-46-199.dynamicIP.rima-tde.net) |
| 01:36.06 | *** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au) |
| 01:36.38 | spootnick | anybody out there with some exp using siproxd? i can't figure how do i pass my sip phone registrations to asterisk from it |
| 01:37.11 | robbie2 | sweet |
| 01:37.14 | justinu | you need to add some kind of static route |
| 01:37.17 | robbie2 | changed protocol |
| 01:37.20 | robbie2 | and worked |
| 01:37.31 | justinu | or setup DNS SRV records |
| 01:38.43 | spootnick | justinu: was that for me? |
| 01:38.56 | justinu | yes |
| 01:39.21 | spootnick | well, as for the static route, i followed the "recipe" they left on the online docs |
| 01:39.25 | spootnick | http://siproxd.sourceforge.net/?op=odoc |
| 01:39.49 | justinu | ah |
| 01:40.01 | spootnick | there's an example on how to redirect outgoing sip traffic to siproxd. that's allright, but i want to have incoming traffic bouncing at siproxd and then registering in asterisk |
| 01:40.04 | justinu | i've used siproxd |
| 01:40.08 | justinu | and it works |
| 01:40.16 | justinu | but I never configured it myself |
| 01:40.38 | spootnick | ah, ok... |
| 01:41.56 | justinu | so you want to point your phones at the siproxd, and have the proxy foward to a * server inside your NAT? |
| 01:42.10 | spootnick | correct |
| 01:42.21 | spootnick | hoping siproxd will magically solve all my sip nat problems |
| 01:42.25 | spootnick | =) |
| 01:42.26 | justinu | it can |
| 01:42.51 | InfraRed | look at iptables reverse nat example |
| 01:42.53 | spootnick | i thought so. but it's weird that in the siproxd.conf file there's no entry for someone to specify an external registrar |
| 01:42.54 | InfraRed | online |
| 01:43.05 | justinu | but you need some kind of route that'll tell it : "when phones register for this sip URI, forward the request to this IP" |
| 01:43.50 | spootnick | justinu: well, that's more like it. i just don't know where to do it in siproxd. and i've read the docs a couple hundred times already |
| 01:44.09 | *** join/#asterisk TheCops (n=xdz@got.securebinary.com) |
| 01:44.15 | justinu | spootnick: ser+rtpproxy can also help |
| 01:44.19 | spootnick | InfraRed: i've set up my router to point sip/rtp traffic to my * box inside my lan. i managed to register some phones from home (internet) |
| 01:44.22 | justinu | but it might even be harder to set it up! :) |
| 01:44.49 | spootnick | justinu: yeah, i read that as well. but you're right, it's kind of hard and i don't have the time right now to figure. i thought siproxd would be a shortcut |
| 01:44.51 | TheCops | What's the best VoIP Gateway devices for more than 200 PSTN Lines (FXO) interface ? |
| 01:44.55 | InfraRed | spootnick: what I do is to vlan into work |
| 01:45.05 | justinu | hey, the cops are back |
| 01:45.17 | TheCops | hi justinu |
| 01:45.25 | InfraRed | spootnick: the issue is not in siproxd |
| 01:45.35 | InfraRed | it's in iptables |
| 01:45.39 | spootnick | InfraRed: well, that can do, but i'm deploying it to a customer, and some extensions will probably be shifting around their locations |
| 01:45.48 | InfraRed | look online for reverse nat exmaple on iptables |
| 01:46.09 | justinu | you might consider ser+rtpproxy for them also |
| 01:46.11 | spootnick | InfraRed: i was saying, i managed to register some phones from the internet in my * box inside my lan. trouble is, there's some inconsistency happening. sometimes they miss calls for no reason |
| 01:46.17 | justinu | more flexible |
| 01:46.28 | InfraRed | it's a lost reinvite |
| 01:46.36 | justinu | nat bindings closing, probably |
| 01:46.47 | justinu | qualify set? |
| 01:46.50 | spootnick | nope |
| 01:46.53 | spootnick | and i can't set it |
| 01:46.58 | justinu | why not? |
| 01:46.59 | spootnick | cos i'm using realtime extensions |
| 01:47.09 | justinu | ok, how about turning on nat keepalive on the phones? |
| 01:47.19 | spootnick | ahm... let me see if it's possible |
| 01:47.35 | spootnick | it's an spa-841 |
| 01:47.39 | spootnick | a spa |
| 01:47.46 | justinu | it's definitely possible |
| 01:47.49 | justinu | i have one here |
| 01:48.12 | justinu | spootnick: you should be getting errors on the console when * sends out reinvites that aren't acknowledged |
| 01:48.35 | spootnick | justinu: not really. i just get a "calling 5000" (for example), and i never get the "ringing" in the console |
| 01:48.48 | kn0x | help!?! |
| 01:48.51 | kn0x | http://pastebin.ca/28065 |
| 01:48.54 | justinu | maybe it only happens when sip debug is on |
| 01:48.56 | kn0x | what going on in here? |
| 01:49.11 | justinu | kn0x: i looked... no idea |
| 01:49.29 | robbie2 | hmm |
| 01:50.10 | spootnick | justinu: dammit... yeah, probably i would see it if i had sip debug on |
| 01:50.33 | spootnick | justinu: any ideas on what tab i can find NAT keepalive? i'm going through them now without success so far |
| 01:50.40 | spootnick | (spa-841) |
| 01:50.41 | justinu | make sure you turn on advanced |
| 01:51.06 | justinu | it's on the ext 1 and ext 2 tab |
| 01:51.09 | kn0x | justinu= oh |
| 01:51.55 | spootnick | justinu: so i guess "NAT keep alive dest" has to be set to something else than $PROXY |
| 01:52.10 | spootnick | no, i'm wrong |
| 01:52.15 | spootnick | i'm confusing it with outbound proxy |
| 01:52.16 | justinu | is nat keep alive enabled? |
| 01:52.58 | spootnick | enabled it now |
| 01:53.06 | justinu | might help |
| 01:53.37 | justinu | if the phone works at first then dies, it's probably nat bindings getting closed |
| 01:55.42 | spootnick | no, it never rings. that's the issue really. that's what lead me to believe it was a nat issue |
| 01:55.49 | justinu | oh ok |
| 01:56.22 | justinu | did you turn on NAT mapping enable? |
| 01:56.35 | spootnick | nope |
| 01:56.40 | justinu | do that too |
| 01:56.42 | spootnick | ok |
| 01:57.50 | TheCops | What's the best VoIP Gateway devices for more than 200 PSTN Lines (FXO) interface ? |
| 01:58.15 | spootnick | how do i check if a SIP invite failed via sip debug? i can see the invites being sent |
| 01:58.29 | spootnick | and lots of retransmissions too |
| 01:58.35 | justinu | yeah, then they're failing |
| 01:59.39 | delmar | hrm. it's a tricky little bit of a balancing act .. this txgain thing... i've just been getting incoming fax reception working... and had to increase txgain to 4.5 to get fax reception to work right... so then i retest the echo on incoming calls.. and there is a little bit of echo at the start but it soon gets deal too... so.. this is tolerable vs having incoming faxes not work right.... and even then... i still dont think incoming faxes are reliable usi |
| 01:59.46 | delmar | still seems to drop more faxes than it receives. |
| 01:59.48 | spootnick | does i make sense that the "To: " field in SIP points to the phone lan address? (when it's connected to my * box over the internet) |
| 02:00.44 | justinu | um, it shouldn't if you turned on nat mapping |
| 02:00.49 | justinu | but that explains the problem |
| 02:00.54 | justinu | check the Contact: headers also |
| 02:01.22 | justinu | oh, try this |
| 02:01.26 | justinu | sipura, sip tab |
| 02:01.32 | justinu | nat support params |
| 02:01.36 | justinu | stun enable: yes |
| 02:01.40 | justinu | stun server: stun.xten.net |
| 02:02.10 | spootnick | stun enable yes? i thought it took some setup on the * side to work |
| 02:02.21 | justinu | well, that's an alternative way to do it |
| 02:02.37 | marc324 | is radius or mysql better for authentication/autorization? |
| 02:02.40 | justinu | either the phone is smart enough to tell asterisk about it's external IP (STUN), or asterisk sets nat=yes |
| 02:02.59 | spootnick | justinu: i'm using nat=yes and it won't work. nat=no has the same effect |
| 02:03.17 | justinu | ok, so turn on stun, and set nat=no |
| 02:03.25 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 02:03.26 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:03.29 | justinu | also, if * is behind NAT, you need externip set in sip.conf |
| 02:03.33 | spootnick | justinu: but in order to use stun, all i need to do is enable it at the phone? no asterisk setup involved? |
| 02:03.41 | justinu | spootnick: correct |
| 02:03.45 | spootnick | justinu: yeah, i have the externip set up |
| 02:03.49 | justinu | ok |
| 02:03.49 | spootnick | justinu: and it is behind nat |
| 02:04.05 | justinu | so turn on stun, and when your phone registers, pay attention to the Contact header |
| 02:04.12 | justinu | it should show the phones public ip |
| 02:05.26 | delmar | hrm |
| 02:05.40 | delmar | im not so sure this incoming fax thing is handling multiple pages correctly..... |
| 02:05.48 | justinu | hey twisted |
| 02:06.16 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
| 02:06.45 | justinu | spootnick: you're lucky you have sipura, their software rocks |
| 02:06.51 | justinu | you can definitely make it work right with that phone |
| 02:09.38 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
| 02:10.20 | spootnick | justinu: you know, for some reason, a phone that was configured as nat=yes (rt extension) suddenly appeared as "no nat" in sip debug |
| 02:10.36 | spootnick | so it seems it's losing it "NAT status" after a while |
| 02:10.39 | justinu | i don't think that's a problem |
| 02:10.49 | perd | anyone know off the top of their head what would cause asterisk not to hear or acknowledge extensions punched into it? |
| 02:10.52 | spootnick | well, that's when i was unable to make it ring again |
| 02:11.01 | spootnick | perd: wrong dtmfmode |
| 02:11.12 | perd | it isnt the config files because i put all the sample ones in /etc/asterisk and tried |
| 02:11.36 | brettnem | hey all |
| 02:11.45 | perd | in what config file? |
| 02:11.54 | perd | i'm making an outbound connection to a regular telephone via IAX |
| 02:12.12 | perd | i see in sip.conf dtmfmode settings.. and in modem.conf, not in iax tho |
| 02:12.13 | brettnem | hey all, I'm having a weird dtmf mode problem. anyone want to help me out? |
| 02:12.18 | spootnick | justinu: you're right. but now i know what that is. after a while, the phone won't ring anymore. if i reboot it, it works |
| 02:12.41 | spootnick | so it has to be related with the keepalive |
| 02:13.04 | brettnem | I'm using dtmfmode=inband.. and sending a call from a pstn gateway (that only supports inband) using ulaw of course.. the directory app isn't recognizing my digits.. any thoughts? |
| 02:13.37 | spootnick | brettnem: i had my extensions using rfc2833. and it sorted it out |
| 02:13.47 | spootnick | brettnem: gsm was disabled |
| 02:13.58 | linagee | with RAID 4, don't you have the advantage of no slowdown if the parity disk itself goes out? :-) |
| 02:14.07 | brettnem | well I'm using ulaw and dtmfmode=inband and having problems.. |
| 02:14.49 | spootnick | brettnem: try using rfc2833 |
| 02:15.03 | brettnem | I can't.. the gateway only supports inband |
| 02:15.45 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:15.57 | brettnem | actually, it'll send rfc2833 to me, but expects inband from me.. so not sure what to do about that.. seems like I can't ever get dtmf to work quite right |
| 02:16.01 | brettnem | any thoughts? |
| 02:16.15 | blitzrage | what is expecting inband? |
| 02:16.20 | brettnem | I can't figure out how to make seperate inbound and outbound dtmfmode configurations. |
| 02:16.29 | blitzrage | brettnem: I don't think you can... |
| 02:16.41 | brettnem | blitzrage: the pstn gateway |
| 02:16.53 | blitzrage | seems like something you would be negotiating with in SDP |
| 02:17.09 | blitzrage | but I'm not positive about that |
| 02:17.26 | brettnem | yeah dtmfmode isn't negotiated in asterisk |
| 02:17.34 | brettnem | at least not in <1.2 |
| 02:18.05 | blitzrage | ahhh... I was having problems with DTMF in 1.0.9 with a couple of service providers -- seems theres some sort of timing bug or something that was resolved in 1.2 I believe... |
| 02:18.16 | blitzrage | didn't have problems with Leve(3), but other people I did. |
| 02:18.57 | brettnem | hmm.. |
| 02:19.17 | blitzrage | sorry -- off to catch a taxi |
| 02:19.23 | brettnem | thanks |
| 02:21.17 | *** join/#asterisk marc32344 (n=marc3234@206-248-128-180.dsl.teksavvy.com) |
| 02:24.03 | brettnem | anyone know about dtmfmode=auto ?? |
| 02:27.11 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
| 02:27.58 | spootnick | justinu: great stuff man. worked sweet |
| 02:28.04 | spootnick | tks a lot |
| 02:29.32 | delmar | ok... i just realised that it was running like.. .sip-hardphone(ulaw) =====> Asterisk =====(IAX2/G729)=====Asterisk ========> sip-hardphone(ulaw)... and the codec translation on both boxes is quite lopsided.. ulaw-g729... ie. from g729-ulaw=4 ulaw-729=15, and the remote box is worse... 9 and 29 respectivly. SO... I thought .. that must be the source of some problems.. maybe the echo... so now its ulaw all the way.. and the echo is WORSE... lol |
| 02:29.50 | delmar | (sigh) |
| 02:30.14 | *** join/#asterisk indigirl1 (n=thd@cpe-066-056-142-077.triad.res.rr.com) |
| 02:31.51 | delmar | do i need to use jitterbuffer over the IAX leg? |
| 02:32.06 | delmar | i mean. .do i need to set jitterbuffers etc at each end in the configs? |
| 02:34.46 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-60-46.cybersurf.com) |
| 02:34.49 | Qwell | kram: y0 |
| 02:34.55 | *** join/#asterisk Alystair (i=Alystair@CPE000d88a7a3b5-CM00407b8794db.cpe.net.cable.rogers.com) |
| 02:35.27 | delmar | hey dude |
| 02:35.33 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:35.53 | delmar | my * to * via IAX echo issue just got worse :( |
| 02:36.01 | delmar | going backwards. lol |
| 02:36.35 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 02:41.05 | delmar | Ok so starting with one of the SIP phones.. the Audio Settings that can be played with are like... AudioType =ulaw, Audioframes=2, G.723.1 High Rate, VAD, AFC, AEC all=no, Handset IN=4 Handset Out =27, Speaker Out =12. anyone know what these should be or are they all ok? |
| 02:46.34 | *** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net) |
| 02:48.56 | delmar | ok what should Voice Frames per TX be when using ulaw.... and what should it be for g729 .. and what factors come into play when deciding these values? |
| 02:50.27 | *** join/#asterisk aRJAy (n=aRJAy@adsl-130-112.swiftdsl.com.au) |
| 02:53.43 | *** join/#asterisk bmillares (n=bmillare@208-45-12-218.chyn.qwest.net) |
| 02:54.07 | bmillares | is anyone runing v 1.2.0 beta2 on RedHad ES 4 |
| 02:54.12 | TheCops | What's the best VoIP Gateway devices for more than 200 PSTN Lines (FXO) interface ? |
| 02:55.01 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 02:55.28 | bmillares | is anyone runing v 1.2.0 beta2 on RedHad ES 4 |
| 02:57.53 | Vco | ahh..thats funny..i never knew xten was from vancouver.. |
| 02:58.00 | Math` | ah it is? |
| 02:58.06 | Qwell | makes a lot of sense now |
| 02:58.27 | Vco | down on w.pender |
| 02:59.05 | Vco | what makes sense now? |
| 02:59.15 | Qwell | nothing |
| 02:59.18 | Vco | the fact their documentation comes on rolling papers? |
| 03:00.23 | *** join/#asterisk devonst17 (n=devonst1@pool-71-106-83-161.lsanca.dsl-w.verizon.net) |
| 03:00.48 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 03:01.14 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 03:02.30 | devonst17 | what would cause a call to not go through |
| 03:02.32 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
| 03:02.44 | devonst17 | it keeps timing out when i try to connect to the box |
| 03:02.48 | Qwell | devonst17: be about 50 times more specific |
| 03:02.49 | Math` | devonst17: nat? |
| 03:03.05 | devonst17 | i'm within the network |
| 03:03.16 | Vco | the internetwork? |
| 03:03.26 | devonst17 | Sorry, that was my friend. |
| 03:03.33 | devonst17 | Heres what I got, |
| 03:03.49 | perd | hmmm what the shit... why can't i dial extensions on my system anymore |
| 03:03.56 | perd | it just doesnt hear them! |
| 03:04.12 | bmillares | are any of you guys runing v 1.2.0 beta2 on RedHad ES 4? |
| 03:04.18 | devonst17 | I have X-Lite on a iBook, I have the Asterisk Binary on another iBook, on a WAP in Gayle's Perks. |
| 03:04.18 | *** part/#asterisk TheCops (n=xdz@got.securebinary.com) |
| 03:04.20 | bmillares | I'm having some problems. |
| 03:04.26 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 03:04.57 | devonst17 | I try to dial into extention 100, but it times out every time. |
| 03:05.21 | devonst17 | I know this is really broad, but does anyone know any really good tutorials for Asterisk on the mac? |
| 03:05.31 | devonst17 | all the ones I have found so far have sucked |
| 03:05.34 | Qwell | ~docs |
| 03:05.35 | jbot | somebody said docs was 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 |
| 03:05.51 | devonst17 | All of this applys to MacOSx? |
| 03:06.08 | Math` | they apply to asterisk, once you've installed it, it doesnt matter on which os you are |
| 03:06.13 | Math` | the config is the same |
| 03:06.14 | mog_home | not all of it |
| 03:06.15 | devonst17 | Excelent |
| 03:06.16 | devonst17 | Thanks |
| 03:06.19 | mog_home | but asterisk is the same on any posix |
| 03:07.50 | *** join/#asterisk _T3_ (n=test@157.100.18.33) |
| 03:08.23 | bmillares | I can't seem to be able to start asterist on a redhat es4 box.. I have it running on FC4 without any problems.. Can someone point me in the right direction please? |
| 03:09.19 | CoaxD | bmillares: Whats the problem |
| 03:10.15 | bmillares | I run: asterisk -v and it fails with: Loading module app_chanspy.so failed! |
| 03:10.34 | CoaxD | hang |
| 03:11.19 | CoaxD | okay |
| 03:11.20 | bmillares | I tryied to do disable the module on modules.conf but it then fails on the next module. |
| 03:11.22 | CoaxD | in /etc/asterisk/modules.conf |
| 03:11.44 | CoaxD | okay. it probably doesnt have any modules in /usr/lib/asterisk/modules |
| 03:12.05 | bmillares | they are there ... let me who owns the files ... |
| 03:12.16 | CoaxD | are the modules YOU need in there? |
| 03:12.30 | CoaxD | type 'id' and hit enter. |
| 03:12.32 | CoaxD | paste output. |
| 03:12.55 | bmillares | ok .. give me a sec |
| 03:14.42 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 23 of modules.conf |
| 03:14.43 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 24 of modules.conf |
| 03:14.43 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 25 of modules.conf |
| 03:14.43 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 26 of modules.conf |
| 03:14.45 | bmillares | <PROTECTED> |
| 03:14.47 | bmillares | <PROTECTED> |
| 03:14.50 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 19 of modules.conf |
| 03:14.55 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 20 of modules.conf |
| 03:14.58 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 21 of modules.conf |
| 03:15.01 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 22 of modules.conf |
| 03:15.04 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 23 of modules.conf |
| 03:15.13 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 24 of modules.conf |
| 03:15.16 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 25 of modules.conf |
| 03:15.19 | bmillares | Nov 8 21:14:20 WARNING[22322]: config.c:575 cfg_process: Unknown directive 'noload' at line 26 of modules.conf |
| 03:15.23 | delmar | bmillares, dude can u not do that |
| 03:15.24 | bmillares | Nov 8 21:14:20 WARNING[22322]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_chanspy.so: undefined symbol: ast_config_AST_MONITOR_DIR |
| 03:15.27 | bmillares | Nov 8 21:14:20 WARNING[22322]: loader.c:440 load_modules: Loading module app_chanspy.so failed! |
| 03:15.28 | delmar | bmillares, use pastebin ffs. |
| 03:15.34 | CoaxD | jesus christ |
| 03:15.40 | bmillares | sorry |
| 03:15.47 | CoaxD | and you didnt paste what i asked for |
| 03:16.05 | bmillares | ok, what do you need? |
| 03:16.10 | CoaxD | what i asked for |
| 03:17.07 | bmillares | ok, I got it now... here we go: uid=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wheel) |
| 03:17.30 | CoaxD | okay. now we can go somewhere. |
| 03:17.45 | CoaxD | this asterisk installation is horribly busted, man |
| 03:17.51 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 03:17.58 | CoaxD | where did you get the installation files? Right outta RHEL? |
| 03:18.16 | delmar | meh. must be someone who has an idea about this echo problem.... why would there be echo in a scenario like sip-hardphone(ulaw)<===>Asterisk<===IAX2(ulaw)===>Asterisk<===>sip-hardphone(ulaw) ... and when it was IAX2/g729 in the middle, there was less echo, despite the translation times... I just can't figure it :( |
| 03:18.27 | bmillares | I checkout out the beta version from asterisk.org |
| 03:18.40 | CoaxD | bmillares: Did you concurrently have asterisk rpms installed? |
| 03:18.51 | bmillares | not that I know.. let me check .. |
| 03:19.05 | CoaxD | bmillares: rpm -qa |grep asterisk |
| 03:19.14 | bmillares | I just did that and no rpm. |
| 03:19.22 | CoaxD | bmillares: Okay. Very good |
| 03:19.36 | CoaxD | bmillares: did you compile everything from scratch? or? |
| 03:19.44 | bmillares | yes |
| 03:20.03 | CoaxD | sigh. bastards |
| 03:20.04 | bmillares | I checkout out the software and then built and installed all the components.. |
| 03:20.27 | bmillares | I did not have any errors during build. |
| 03:20.29 | CoaxD | bmillares: its like it doesnt understand the format of modules.conf |
| 03:20.39 | CoaxD | bmillares: Sounds like you did eveyrthing the way you were supposed to |
| 03:20.51 | Qwell | except probably put stuff outside of the context |
| 03:20.53 | CoaxD | bmillares: (Tho obviously, you didnt - or, rhel is busted. one of the two) |
| 03:20.58 | CoaxD | qwell: in modules.conf ?? |
| 03:21.00 | Qwell | yes |
| 03:21.03 | Qwell | [modules] |
| 03:21.11 | CoaxD | Could be. maybe he deleted the wrong line |
| 03:21.21 | bmillares | it works fine on FC4 I did not do anything different.. |
| 03:21.30 | Qwell | bmillares: pastebin your modules.conf |
| 03:21.32 | CoaxD | bmillares: I bet qwell's right. |
| 03:21.44 | CoaxD | bmillares: Lets see the modules.conf file in pastebin |
| 03:22.12 | bmillares | how do I pastebin? |
| 03:22.15 | Qwell | ~pb |
| 03:22.16 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 03:23.57 | bmillares | Ok I got it: http://pastebin.ca/28082 |
| 03:24.11 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 03:24.26 | Qwell | ... |
| 03:24.33 | Qwell | or...alternatively...you know what you might try? |
| 03:24.38 | Qwell | doing what I asked for. thanks |
| 03:24.40 | CoaxD | Qwell: He's not listening |
| 03:24.50 | CoaxD | qwell: Pointless to try and help these folks when they dont even do what you ask |
| 03:24.55 | bmillares | what do you mean? |
| 03:25.06 | bmillares | I'm sorry, I'm trying to follow.. |
| 03:25.06 | Math` | and when they don't understand they don't do what you ask |
| 03:25.13 | CoaxD | Math: Exactly |
| 03:25.15 | bmillares | I'm not familiar with x-chat or irc... |
| 03:25.20 | Math` | somebody should write "Asterisk for dummies" lol |
| 03:25.30 | CoaxD | bmillares: What part of 'paste modules.conf' do you not comprehend? |
| 03:25.40 | bmillares | ok, sorry |
| 03:25.48 | CoaxD | bmillares: Thats twice |
| 03:26.02 | bmillares | maybe we are no all just as intelligent ... I'm a little slower than most ... let me try again please.. |
| 03:26.19 | CoaxD | bmillares: Slow isnt a problem. Not listening is a problem. Please continue. |
| 03:26.42 | CoaxD | http://pastebin.ca/28082 |
| 03:26.45 | CoaxD | err |
| 03:26.50 | CoaxD | ast_load_resource: /usr/lib/asterisk/modules/app_chanspy.so: undefined symbol: ast_config_AST_MONITOR_DIR |
| 03:27.01 | CoaxD | this is obviously an issue. it didnt compile something. |
| 03:27.18 | bmillares | http://pastebin.ca/28085 |
| 03:27.21 | CoaxD | but its loading other stuff |
| 03:28.03 | CoaxD | i can see zilch wrong with thi |
| 03:28.03 | CoaxD | s |
| 03:28.10 | *** part/#asterisk indigirl1 (n=thd@cpe-066-056-142-077.triad.res.rr.com) |
| 03:29.07 | CoaxD | kill out chanspy |
| 03:29.10 | CoaxD | then what module fails |
| 03:29.36 | bmillares | app_directed_pickup.so |
| 03:29.41 | bmillares | then func_uri.so |
| 03:29.43 | bmillares | etc |
| 03:30.01 | CoaxD | kill it |
| 03:30.03 | CoaxD | in fact |
| 03:30.04 | CoaxD | here |
| 03:30.17 | bmillares | if you notice in the modules.conf file I had noload directives for the modules in their order of failure.. |
| 03:30.27 | CoaxD | yes i did note that |
| 03:30.33 | CoaxD | and you dont need even 1/10th of that shit loaded |
| 03:31.00 | CoaxD | i sent you a file via DCC. its a modules.conf replacement. Its what I use. (older version of *, but it might work.) |
| 03:31.04 | CoaxD | Try 'er out |
| 03:31.09 | bmillares | ok |
| 03:31.13 | bmillares | give me a sec |
| 03:31.25 | CoaxD | there's a bunch of stuff in your modules.conf that just really doesnt need loading |
| 03:31.44 | CoaxD | my asterisk was CVS current as of like april. *lol* |
| 03:31.45 | bmillares | do you have any idea where x-chat puts the file you sent? |
| 03:32.13 | CoaxD | bmillres: In your home directory |
| 03:32.25 | bmillares | go it .. .xchat2/downloads |
| 03:32.36 | CoaxD | yep |
| 03:32.38 | CoaxD | sounds right |
| 03:33.02 | Vco | show channels |
| 03:33.04 | CoaxD | find . -name modules.conf |
| 03:33.05 | Vco | errrr... |
| 03:33.13 | bmillares | go it . |
| 03:33.34 | Vco | arigatogozaimasu |
| 03:34.09 | bmillares | ok, I replaced it and tryied again... same output. |
| 03:34.20 | CoaxD | bmillares: where does it fail |
| 03:34.30 | fg-away | how does that pause command go: dial(w,tech/exten) |
| 03:34.32 | CoaxD | just pate the last couple lines |
| 03:34.45 | CoaxD | fg-away: pause command? |
| 03:34.55 | bmillares | same spot as before ... I'll paste the outpout to that site .. |
| 03:35.02 | CoaxD | bmillares: k |
| 03:35.04 | FuriousGeorge | CoaxD: yeah |
| 03:35.15 | FuriousGeorge | like to pause a fraction of a second before dialing |
| 03:35.22 | FuriousGeorge | i think it only works with zap |
| 03:35.28 | bmillares | http://pastebin.ca/28087 |
| 03:35.32 | FuriousGeorge | dail,www(tech/exten)? |
| 03:35.45 | Vco | yo-yo-yo i gots my data on the go.... |
| 03:35.48 | CoaxD | I dunno, i'd just wait() |
| 03:36.04 | tzanger | FuriousGeorge: no |
| 03:36.09 | tzanger | 'w' is part of the exten |
| 03:36.14 | tzanger | at least with zap |
| 03:36.19 | tzanger | you don't pause with other channel techs |
| 03:36.20 | FuriousGeorge | CoaxD: im specifically having a problem where the damn line isnt "ready" for the first digit the fxo throws at it |
| 03:36.30 | tzanger | Dial(Zap/g1/www${EXTEN}) |
| 03:36.30 | CoaxD | FuriousGeorge: That is LAME |
| 03:36.46 | tzanger | I want to get dialtone detection into zaptel |
| 03:36.48 | CoaxD | tzanger: Hmmmm. never seen that hack |
| 03:36.51 | tzanger | dsp.c is already running |
| 03:36.53 | tzanger | CoaxD: you're kidding |
| 03:36.58 | FuriousGeorge | CoaxD: tell me about it |
| 03:37.00 | tzanger | that's exactly how you do it with slow telco switches |
| 03:37.03 | FuriousGeorge | tzanger: thanks |
| 03:37.12 | CoaxD | tzanger: Oh, of this i have no doubt |
| 03:37.21 | CoaxD | tzanger: Picking up the damn phone should make a dialtone. |
| 03:37.27 | tzanger | ? |
| 03:37.30 | CoaxD | tzanger: if ya cant rely on that, shit, what CAN you rely on |
| 03:38.26 | bmillares | CoaxD: could this be an issue with the linker during build? |
| 03:38.29 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 03:39.22 | devonst17 | Another question... Is there a good monitor application for the asterisk daemon? |
| 03:40.06 | bmillares | CoaxD: is obvious that the issue is RedHat because I have the same code working on other distros. In fact I've tryed with version all thre major versions of asterisk and they all faild the same way. |
| 03:41.22 | CoaxD | bmillares: its just a distro |
| 03:41.50 | CoaxD | bmillares: And yes, it most certainly could be an issue w/ the linker during build. its not failing within module loading stuff |
| 03:41.58 | CoaxD | bmillares: (i.e. modules.conf) |
| 03:42.12 | *** join/#asterisk Zilasb (n=info@c-24-92-153-118.hsd1.ga.comcast.net) |
| 03:42.39 | CoaxD | bmillares; now that i look, its not even getting to the modules.conf stuff. its loading the internal crap it HAS to load |
| 03:42.48 | CoaxD | bmillares: Its loading SOME stuff, but not all |
| 03:42.54 | bmillares | right ... |
| 03:43.16 | Zilasb | I would buy lots of ()] if somebody would tell me why my asterisk needs a full paths to the sound file for it to play... asterisk.conf is good... |
| 03:43.20 | CoaxD | bmillares: there's definitely something fishy going on.. I suggest rm /usr/lib/asterisk/modules/* and 'make install' again from the asterisk dir |
| 03:43.32 | CoaxD | bmillares: As long as you dont 'make samples' everything should still be ok |
| 03:43.48 | CoaxD | bmillares: but back up /etc/asterisk to a safe spot just in case |
| 03:43.50 | bmillares | the actual error is that ast_config_AST_MONITOR_DIR is not defined ... |
| 03:44.03 | CoaxD | bmillares; yeah. because the symbol isnt found |
| 03:44.29 | CoaxD | bmillares: Which probably indicates that asterisk is internally depending upon a condition to be true - when in fact, it exists everywhere but in RHEL |
| 03:44.34 | bmillares | It should had been defined by a macro during build.. |
| 03:44.35 | CoaxD | (So in fact, is a * bug) |
| 03:44.43 | CoaxD | bmillares: EXACTLY |
| 03:45.14 | CoaxD | the probelm is probably in ast_config.c or somesuch |
| 03:45.55 | CoaxD | i would try latest CVS. if that dont fix it, life sux for you and you should write up a bug report |
| 03:46.02 | *** join/#asterisk Zenofex (n=Zenofex@cpe-66-68-71-114.austin.res.rr.com) |
| 03:47.23 | bmillares | the funny thing is that, I have the same problem using r 1.0 to the latest version... you would think that someone else would have experience this issue since RH Enterprise is a popular OS.. |
| 03:47.44 | Zenofex | hey, im having some issues with my nortel ip phone and my linux firewall, can anyone tell me what ports i need to forward, the phone will get connected but i cannot get sound/voice. i read somewhere that its a rtp issue but i cant seem to find the rtp ports to forward for my firewall |
| 03:47.57 | Qwell | nortel ip phone? eww |
| 03:48.00 | CoaxD | bmillares: it IS? :) |
| 03:48.04 | CoaxD | bmillares: You coulda fooled me |
| 03:48.14 | Zenofex | yeah, its what the company i work for uses :-( |
| 03:48.16 | CoaxD | bmillares: at $750 a license, you can bet your ass it isnt that popular |
| 03:48.41 | bmillares | :) .. I see .. |
| 03:48.48 | Zenofex | im just trying to figure out what ports rtp uses for our bcm system |
| 03:49.04 | CoaxD | bmillares: at least not for asterisk users |
| 03:49.13 | bmillares | Zenofex: Are you running asterisk on RHES4? |
| 03:49.18 | bmillares | I get it .. |
| 03:49.20 | CoaxD | bmillares: (asterisk users want cheap or free. not RHEL) |
| 03:49.50 | bmillares | I normaly use Fedora my self .. but since they split I think they are looking more and more diffierent .. |
| 03:50.11 | CoaxD | bmillares: They *are*. Mostly becasue people actually USE FC4. |
| 03:50.34 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 03:51.40 | *** join/#asterisk bmg505 (n=leon@rndf-146-57-141.telkomadsl.co.za) |
| 03:51.51 | bmillares | Well listen, thanks for the help.. I guess I'll have to look for another hosting company that offers a different distro. |
| 03:52.54 | CoaxD | you're welcome man. Its not that you couldnt run it there. you just have to fix the problem |
| 03:53.10 | CoaxD | you running colo on one of those vmware hosting co's or something? |
| 03:53.34 | bmillares | rackspace ... dedicated server. |
| 03:54.04 | bmillares | the tech support is nice .. but the client choose the OS a while back, I just came into the picture to setup the pbx |
| 03:54.15 | CoaxD | bmillares: So why cant you just ask them to put up fc4? |
| 03:54.21 | CoaxD | ahhhh |
| 03:54.51 | CoaxD | bmillares: You obviously dont know much about debugging at the source level.. that sucks.. I have a few more tricks up my sleeve than you do, i suppose, but then again, i've probably been doing this a hell of a lot longer than you |
| 03:55.01 | bmillares | they already have multiple applications running .... That will not be an option ... however I can find another home for the voip stuff |
| 03:55.02 | CoaxD | bmillares: if you knew more about that side of things, you could probably fix this |
| 03:55.15 | Math` | whats the problem? |
| 03:55.16 | CoaxD | bmillares: Ah. sounds like money isnt so much of an object - as time |
| 03:55.29 | CoaxD | bmillares: TRY getting the latest cvs code for asterisk |
| 03:55.30 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 03:55.33 | CoaxD | bmillares: Follow the instructions for that |
| 03:55.41 | CoaxD | bmillares: If it still doesnt work, go your route |
| 03:55.47 | bmillares | ok |
| 03:55.49 | CoaxD | bmillares: Or for god sakes, file a bug |
| 03:55.55 | bmillares | sounds like a plan ... |
| 03:56.05 | CoaxD | bmillares: digium is there for a reason, and i guarantee you, they'd want to know if * didnt run on RHEL |
| 03:56.18 | bmillares | you have a point there.. |
| 03:56.35 | bmillares | I'll go ahead and submit a bug report... |
| 03:56.41 | bmillares | it might be a simple thing ... |
| 03:56.45 | Qwell | it very likely is |
| 03:56.46 | mog_home | what do people need from digium |
| 03:56.56 | CoaxD | bmillares: Exactly |
| 03:57.17 | CoaxD | bmillares: If nobody looks at it in a week, man, yeah, i'd go the other route - especially if yer not much of a debugger |
| 03:57.56 | CoaxD | (I'd find out what defines that symbol - and find out where in the compile its breaking) |
| 03:58.03 | bmillares | I was checking astconf.h and the constant gets defined as an integer of 255, it seems like a fool proof assigment but .. who knows.. |
| 03:58.12 | CoaxD | ...and if its easy, #define the damn thing the way its supposed to be - or the way you want it to be, one of the two |
| 03:58.29 | CoaxD | bmillares: look for places it gets #undef'ed |
| 03:58.29 | Math` | bmillares: if you're comfortable with giving me access to that box, I'd check it out |
| 03:59.00 | FuriousGeorge | can i do this: _40[#*],1,dial(blah/exten) |
| 03:59.15 | FuriousGeorge | 40# or 40* |
| 03:59.31 | Math` | you should |
| 03:59.42 | bmillares | you can login into my box at home and we run a screen session, the other box is my client's ... |
| 03:59.48 | CoaxD | wow. postgres 8.1.0 just released |
| 03:59.52 | Math` | :o |
| 03:59.55 | FuriousGeorge | Math`: i should do that? |
| 03:59.59 | Math` | bmillares: ok, lets screen -x |
| 04:00.09 | Math` | FuriousGeorge: I mean, you should be able to do it |
| 04:00.18 | FuriousGeorge | cool thanks |
| 04:01.53 | mog_home | new hotness |
| 04:01.56 | mog_home | available |
| 04:02.11 | CoaxD | wonderful |
| 04:02.29 | CoaxD | gosub/return. heh |
| 04:08.38 | *** join/#asterisk jayceeLFT (n=jaycee@219.95.45.104) |
| 04:09.31 | jayceeLFT | hi |
| 04:09.43 | jayceeLFT | can i post questions ere? |
| 04:10.01 | Qwell | jayceeLFT: seems to me you just did |
| 04:10.44 | jayceeLFT | hehe, that's right |
| 04:11.10 | jayceeLFT | I just joined this chat, so i'm not realli sure what ppl in this chatroom do |
| 04:12.00 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 04:15.41 | hardwire | blah |
| 04:18.29 | Pegger | jayceeLFT we talk abotu asterisk! |
| 04:19.01 | Pegger | jayceeLFT what where you guessing was discussed in here |
| 04:19.33 | robbie2 | can i build asterisk with g726 support ? |
| 04:20.55 | asteriskmonkey | MancPower: ever had an issue with no echo on local numbers yet echo on ld numbers on a pri? |
| 04:21.21 | asteriskmonkey | or anyone else for that matter |
| 04:21.34 | mog_home | asterisk by default has 726 |
| 04:21.38 | mog_home | just not 723 |
| 04:21.44 | mog_home | and 729 you need a lic. for |
| 04:23.48 | hardwire | mog mog mog |
| 04:23.50 | hardwire | you have a fun name |
| 04:24.17 | mog_home | yeah |
| 04:24.21 | mog_home | thats why i go buy it |
| 04:24.24 | mog_home | my real name |
| 04:24.41 | mog_home | mathias matthew maximillian kolbe of the house of blue gormain the first |
| 04:24.43 | mog_home | not so fun |
| 04:24.48 | mog_home | so i go by mog |
| 04:25.12 | syle | u play final fantasy? |
| 04:25.21 | mog_home | not a whole lot |
| 04:25.26 | mog_home | comes from mogorman |
| 04:25.28 | mog_home | but i am into the mog |
| 04:25.47 | robbie2 | a mog is a truck used by the .au army |
| 04:25.50 | robbie2 | :) |
| 04:25.52 | robbie2 | army mog |
| 04:26.00 | mog_home | heh i was not aware of such mog |
| 04:26.13 | syle | no its your house to store your weapons and money in final fantasy 11 |
| 04:26.18 | robbie2 | should my folke be able to hook through on their 256/64 adsl ? |
| 04:26.28 | mog_home | bump that its all about 3/6 where you get to be mog |
| 04:26.29 | robbie2 | should i specify a particualr codec, its choopy doo |
| 04:27.03 | syle | i got banned from that game for killing people lol |
| 04:27.10 | syle | so now i just play dead or alive |
| 04:27.11 | mog_home | lol |
| 04:27.13 | mog_home | bad syle |
| 04:27.25 | syle | where i am allowed to kill people lol |
| 04:28.08 | syle | gonna be a big lineup on the 22nd for the xbox 360 |
| 04:28.33 | Lostfrog | Midnight 21st |
| 04:30.28 | syle | whats open at midnight |
| 04:30.58 | Lostfrog | Most of the US stores. |
| 04:31.04 | Lostfrog | On that night. |
| 04:31.11 | syle | interesting |
| 04:31.32 | syle | i live in canada, unfortunate for that night lol |
| 04:32.29 | Math` | syle: where in ca? |
| 04:32.36 | syle | winnipeg |
| 04:32.38 | Math` | ok |
| 04:32.40 | Math` | montreal here |
| 04:33.20 | syle | my parents live there, although i don't visit very often, loosing alot of my french lol |
| 04:33.31 | Math` | lol |
| 04:34.19 | syle | which is almost irony considering i didn;t speak a word of english till i was like 7-10 |
| 04:34.46 | Math` | ah you grew up in montreal |
| 04:35.11 | syle | jonquiere, chicoutimi |
| 04:35.15 | Math` | heh |
| 04:35.30 | Math` | what bringed you to winnipeg? |
| 04:35.52 | syle | i been all over canada, work mostly |
| 04:36.00 | Math` | what do u work in? |
| 04:36.04 | syle | moved back from BC 2 years ago, before that i was in toronto |
| 04:36.13 | *** join/#asterisk robbins (n=robbins@adsl-068-209-107-007.sip.mia.bellsouth.net) |
| 04:36.50 | syle | oww i was just starting out as a programmer back then, moving between different cities all the time to better paying jobs |
| 04:36.57 | syle | now i;m married, just stay here lol |
| 04:37.22 | Math` | lol |
| 04:37.56 | syle | someone doesn;t like remote work, screw them :) |
| 04:38.18 | Math` | hehe |
| 04:38.31 | Math` | your still coding? |
| 04:38.51 | syle | yeah, that and trying to get a company off the ground now |
| 04:39.11 | Math` | nice, Im doing that too |
| 04:39.19 | syle | i;ve made many companies rich over the years, its my turn |
| 04:40.09 | Math` | what are you offering? |
| 04:40.25 | syle | right now i;m coding billing software for asterisk |
| 04:40.55 | syle | been working on the module forever lol |
| 04:41.48 | syle | all in c embedded into asterisk of course, no overhead of agi or thousand lines of procedures in a trigger :) |
| 04:42.15 | Math` | nice, Im parting of a starting company in the point of sale industry |
| 04:42.28 | Math` | I plan to integrate asterisk support eventually |
| 04:42.45 | syle | what you selling? |
| 04:42.50 | Math` | email me your contact info at mathieu.rene@gmail.com |
| 04:42.52 | robbins | aanyone seen any issues with ztdummy on 2.6 kernel? |
| 04:42.52 | Lostfrog | POS |
| 04:42.53 | Math` | point of sales |
| 04:42.55 | robbins | is udev necessary? |
| 04:42.58 | syle | k |
| 04:43.20 | Math` | syle: will more turn out as a fully-featured management software (such as SAP, but for small-businesses) |
| 04:43.31 | morale | syle, where in BC you at? |
| 04:43.43 | syle | i moved back from BC |
| 04:43.45 | syle | in winnipeg |
| 04:43.48 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-206.hsd1.ut.comcast.net) |
| 04:43.54 | morale | winnipeg? sorry to hear that :) |
| 04:43.58 | syle | lol |
| 04:44.04 | Math` | lol |
| 04:44.07 | syle | hell whats christmas without snow lol |
| 04:44.17 | Math` | we've snow in montreal too :P |
| 04:44.39 | morale | christmas without block heaters, shoveling your driveway everyday, white outs.. |
| 04:44.42 | morale | i love alberta. |
| 04:44.46 | syle | yeah north quebec got 7 inches i hear alread |
| 04:44.47 | syle | y |
| 04:45.05 | syle | north of quebec city i think |
| 04:45.29 | Utah_Dave | hey I have a newbie question. I have access to a dual processor server (1.1 mhz, 4 gig ram) What gives Asterisk the most boost, dual processors or more ram? |
| 04:45.30 | Math` | 7 inches :| |
| 04:45.46 | syle | in BC i lived in nanaimo and parksville |
| 04:45.53 | syle | was never found of big cities |
| 04:45.59 | morale | i grew up in port alberni with the loggers. |
| 04:46.12 | Math` | syle: you don't like big cities? |
| 04:46.35 | Math` | Utah_Dave: depends.... for a lot of transcoding, cpu helps |
| 04:46.38 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 04:46.54 | syle | i lived smack downtown in toronto once, hated it, was faster to get places with subway than a car |
| 04:47.16 | Math` | you dont have to live downtown |
| 04:47.34 | Utah_Dave | ok, thanks Math` |
| 04:47.42 | syle | hehe at that time my work was 2 blocks away so it made sense |
| 04:47.49 | Math` | hehe |
| 04:47.56 | Lostfrog | There is a probably a happy medium, Math`. |
| 04:48.03 | justinu | it's 60 degrees at night here |
| 04:48.10 | syle | for the single like anyways |
| 04:48.13 | syle | life |
| 04:48.22 | syle | course that don;t exist anymore |
| 04:48.41 | syle | you married math? |
| 04:48.50 | Math` | no |
| 04:48.52 | justinu | i'm engaged now |
| 04:48.53 | Math` | Im 18 :P |
| 04:49.04 | Lostfrog | You can't be 18 and married? |
| 04:49.09 | Math` | yeah you can |
| 04:49.16 | Math` | but I wouldnt do it |
| 04:49.22 | Igbothom_III | 1.2 RC1 has been released, it seems |
| 04:49.26 | Math` | it is |
| 04:49.42 | syle | email sent |
| 04:49.47 | Igbothom_III | methinks the channel title needs updating :) |
| 04:49.48 | MikeJ[Laptop] | yep... |
| 04:49.50 | Lostfrog | I wonder what are the changes from Beta2 to RC1 |
| 04:50.02 | MikeJ[Laptop] | Lostfrog, lots o bug fixes... |
| 04:50.06 | MikeJ[Laptop] | a couple apps |
| 04:50.58 | MikeJ[Laptop] | a lot of the changes for the no jump +101 stuff |
| 04:50.58 | morale | is rc1 in cvs head or do i have to checkout from a different tag? |
| 04:50.58 | Math` | I dunno but I was trying to make a patch and I couldn't get the same tree between 2 cvs checkouts because there were too many commits |
| 04:50.58 | Math` | syle: got it, thanks |
| 04:50.58 | MikeJ[Laptop] | app changes to use arg parsing avi |
| 04:50.58 | Igbothom_III | I fecking installed B2 2 days ago! :) |
| 04:50.58 | MikeJ[Laptop] | changes to some of the include file structure |
| 04:51.04 | Math` | syle: will that be open-source? |
| 04:51.09 | Math` | or commercial |
| 04:51.21 | syle | who knows, whatever feeds my family |
| 04:51.25 | Math` | lol |
| 04:51.37 | MikeJ[Laptop] | grrrr |
| 04:51.42 | *** join/#asterisk oogle_ (n=jart@ool-4578cc36.dyn.optonline.net) |
| 04:51.45 | MikeJ[Laptop] | this stuff is still broke... |
| 04:52.57 | syle | well when i started this project i thought hell...just 100 lines of c , whatever why not |
| 04:53.02 | oogle_ | i'm having a problem changing the caller id... I'm calling my PBX on my cell phone and the call is coming in through IAX, then i change the caller id and bridge the call using Dial to my other cell and the caller id doesn't change to what i want |
| 04:53.07 | syle | 1000 lines later and still not done |
| 04:53.11 | oogle_ | any ideas ? |
| 04:53.13 | Math` | syle: hehe |
| 04:53.17 | *** join/#asterisk zamsler (n=zamsler@c-67-184-232-149.hsd1.il.comcast.net) |
| 04:53.22 | Math` | syle: its usually what happens, starting small... grows fast |
| 04:53.27 | zamsler | hmm.. |
| 04:53.31 | Lostfrog | Like having a baby? |
| 04:53.45 | morale | why does asterisk not use subversion? its so much easier to comprehend :) |
| 04:53.45 | zamsler | Is a macro supposed to be private until it exits? |
| 04:53.48 | Lostfrog | Pretty soon it's a teenager and says it hates you. |
| 04:54.26 | *** join/#asterisk idefix (n=idefix@c-67-160-193-161.hsd1.ca.comcast.net) |
| 04:54.49 | oogle_ | so it's PSTN -> IAX -> * -> SIP -> PSTN but the caller id is getting preserved despite me changing it |
| 04:55.12 | Math` | oogle_: how do you change it |
| 04:55.36 | oogle_ | i tried SetCIDNum() |
| 04:55.45 | oogle_ | Math`: I'm not sure how to use the "new" syntax |
| 04:55.54 | Math` | Set(CALLERID(number)=name) |
| 04:56.23 | oogle_ | Math`: thanks, let me try that again.. i might have been doing it wrong before |
| 04:56.36 | Math` | SetCIDNum should work |
| 04:56.45 | Math` | make sure you have no callerid= in your sip.conf |
| 04:57.06 | *** join/#asterisk Barmal (n=info@c-24-92-153-118.hsd1.ga.comcast.net) |
| 04:57.41 | syle | thats where it should be to begin with unless yoru spoofing calls on purpose hehe |
| 04:58.21 | zamsler | H3LP Is a macro supposed to be private until it exits? |
| 04:58.40 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 04:58.58 | *** join/#asterisk tainted_ (n=identd@ppp-71-134-197-25.dsl.irvnca.pacbell.net) |
| 04:59.43 | zamsler | :( |
| 04:59.51 | zamsler | where is everyone? |
| 05:00.29 | Math` | syle: yeah but Set(CALLERID(num)=name) is useless if there's an entry in sip.conf |
| 05:01.00 | Math` | zamsler: what do u mean |
| 05:01.32 | zamsler | I have a macro that I get digits, and right after the digits are entered, it goes to dial. |
| 05:01.42 | zamsler | and I don't have dial included. |
| 05:02.01 | zamsler | I thought that Macros' vars were private until exit |
| 05:02.20 | Math` | vars are private to the channel |
| 05:02.32 | oogle_ | i found the problem, i had to write CALLERID(all)=12318567489... we need better docs on that because CALLERID(number)=123456468 didn't work |
| 05:02.49 | zamsler | ok, I am trying to write a privacy macro. |
| 05:02.49 | oogle_ | there could even be a bug involved |
| 05:02.56 | zamsler | and right after i get the phone# it dials it. |
| 05:03.02 | Math` | oogle_: er... lets say its Set(CALLERID(15145551212)=myname) |
| 05:03.41 | oogle_ | Math`: i tried putting the number there and it generated errors, it seems to look for a string like "number" or "all" inside CALLERID(xxx) |
| 05:04.19 | oogle_ | oh, i also wrote a cool voice changer app, as soon as my work signs off copyright interest, asterisk hackers can be spies too |
| 05:05.18 | zamsler | :( |
| 05:06.00 | Lostfrog | Luke.... I am your father. |
| 05:06.26 | oogle_ | heh, the possibilities will be endless, combine the voice with a fake caller id and you're golden |
| 05:06.40 | Lostfrog | Still traceable. |
| 05:06.50 | Math` | what else.... a DSP processor for chaining voice effects? |
| 05:06.56 | oogle_ | yes but they'll only trace you if you make death threats |
| 05:07.16 | Lostfrog | Or mention the president and bomb in the same call. |
| 05:07.20 | Math` | lol |
| 05:07.53 | justinu | bounce it thru 5 different PSTN gateways |
| 05:07.56 | oogle_ | Lostfrog: ah now you got this channel red flagged, the FBI will be here any minute |
| 05:08.03 | Lostfrog | "Did you watch the presidential debate? It was the bomb!" |
| 05:08.11 | *** join/#asterisk dsasad (i=dsasad@c-24-10-240-101.hsd1.ut.comcast.net) |
| 05:08.12 | Lostfrog | NSA, oogle. |
| 05:08.17 | Lostfrog | Get your acronyms right. |
| 05:08.25 | oogle_ | yea yea |
| 05:08.26 | dsasad | Hello Everyone ? |
| 05:08.29 | Math` | hi |
| 05:10.03 | dsasad | Does anybody know how many phones can one computer with one network card can handle ? |
| 05:10.28 | dsasad | ip phones, on asterisk. |
| 05:11.11 | mishehu | probably more than you can afford |
| 05:11.47 | mishehu | think about it, 80 kbps for g711 ulaw per phone. |
| 05:11.56 | dsasad | aha |
| 05:12.06 | dsasad | well i have to install it for a big company |
| 05:12.11 | dsasad | around 120 local extentions |
| 05:12.13 | mishehu | if you have no transcoding going on, there is low cpu utilizaton most of the time |
| 05:12.45 | mishehu | it's transcoding and possibly fax that would suck cpu. |
| 05:13.09 | dsasad | aha |
| 05:13.12 | *** join/#asterisk Assid (n=assid@203.115.64.59) |
| 05:13.34 | dsasad | would you say 120 ip phones should be fine ? |
| 05:14.21 | zamsler | I think I found a bug.... |
| 05:14.27 | Math` | :o |
| 05:14.30 | Math` | post it |
| 05:14.34 | zamsler | Can I get some input?? |
| 05:15.12 | zamsler | in AEL. if I do a background followed by a read, it is passed to the parent context. ie dials the # |
| 05:15.24 | zamsler | but If I foa Playback it works as expected. |
| 05:15.46 | zamsler | foa = do |
| 05:17.36 | *** join/#asterisk huslage (n=huslage@c-67-169-200-122.hsd1.or.comcast.net) |
| 05:18.02 | huslage | does anyone have Cisco SIP firmware 2.1 for the 7940? |
| 05:18.44 | zamsler | hmm |
| 05:19.06 | huslage | i've got a 7940 with old SCCP code on it that won't upgrade past 2.1 first |
| 05:20.08 | mishehu | dsasad: what is the hardware of the machine, and what type of network? |
| 05:20.09 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
| 05:22.15 | *** join/#asterisk JunK-Y (n=junky@MTL-HSE-ppp197065.qc.sympatico.ca) |
| 05:22.50 | *** join/#asterisk file[laptop] (n=jcolp@MTL-HSE-ppp197065.qc.sympatico.ca) |
| 05:23.29 | Lostfrog | Quad P5 9Ghz w/ 4TB memory, DS3 network. |
| 05:23.38 | *** join/#asterisk _santiago_ (n=santiago@208.195.215.124) |
| 05:23.56 | h3x | what the hell is a p5 |
| 05:24.08 | robbie2 | an imagination |
| 05:24.11 | Lostfrog | Wait DS3 is less than 100B-T. |
| 05:24.23 | Math` | DS3 is ~ 50mbps |
| 05:24.24 | h3x | and even via epias have 1000base on them now |
| 05:24.28 | h3x | its 45 |
| 05:24.31 | h3x | minus overhead |
| 05:24.43 | h3x | ds3s are shitty |
| 05:24.44 | h3x | hehe |
| 05:24.46 | h3x | i have two of them |
| 05:24.49 | zamsler | lol |
| 05:24.54 | zamsler | I have 1 for now. |
| 05:24.57 | Math` | still, better than adsl lol |
| 05:24.59 | zamsler | ATT is ok. |
| 05:25.12 | zamsler | never been down |
| 05:25.17 | h3x | my data one costs what most people pay for a t1 |
| 05:25.18 | zamsler | been a year so far |
| 05:25.19 | h3x | coz im on net |
| 05:25.19 | h3x | heh |
| 05:25.30 | zamsler | I am kinda on ne. |
| 05:25.35 | zamsler | $42.50/meg |
| 05:25.39 | zamsler | and 1450 for the loop |
| 05:25.42 | jontow | woo, datacenter's down for UPS maintenance.. that sucks |
| 05:25.48 | h3x | my loop is $0 |
| 05:25.48 | h3x | well |
| 05:25.57 | orlok | jontow: wtf |
| 05:25.59 | orlok | jontow: thats shit |
| 05:26.00 | h3x | i have to pay 300 a month to colo 24 strands of fiber and 12 strands of coax with wiltel |
| 05:26.00 | zamsler | I can get a 0 loop. |
| 05:26.01 | dsasad | mishehu: its a P4 2.8 |
| 05:26.03 | h3x | but that aint shit |
| 05:26.03 | jontow | yeah it is |
| 05:26.16 | orlok | jontow: they shouldent need to go down! |
| 05:26.20 | orlok | i think that means |
| 05:26.21 | jontow | we're workin in our maintenance window -- had to replace stepdown transformers in our main UPS |
| 05:26.32 | jontow | they failed and we lost a leg in the last power outage (yesterday) |
| 05:26.34 | orlok | "we were testing it, and didnt have a full backup system and found a fault" |
| 05:26.37 | zamsler | I have a 22 mile loop for $1450, I thin kis agood. SBC wanted ~6000/mo\ |
| 05:26.53 | jontow | shittay |
| 05:26.56 | jontow | but its being fixed :) |
| 05:27.18 | jontow | i made sure only internal shit went down, all customers are routed through working power and have full connectivity |
| 05:29.29 | Qwell | hmm |
| 05:29.43 | zamsler | oi? |
| 05:29.58 | mishehu | dsasad: I'd imagine it would be fine. you'd have to actually talk to someone else who has deployed such a config. also, the other part of the problem is what type of phone. for example, if they are polycoms you'll have additional load due to ftp... |
| 05:29.58 | *** part/#asterisk huslage (n=huslage@c-67-169-200-122.hsd1.or.comcast.net) |
| 05:42.23 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 05:48.17 | Pegger | h3x what are you doing with 24 strands of fiber? |
| 05:55.21 | alephcom | Well, I wonde what I screwed up. When I try connecting a user from -realtime in rc1 asterisk gives a segmentation fault. :-( Any thoughts? |
| 05:55.51 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 05:56.32 | *** join/#asterisk convey (n=test@68-70-106-142.clspco.adelphia.net) |
| 05:57.22 | convey | anyone wanna help diagnose why I am unable to recieve calls with *? |
| 05:57.36 | file[laptop] | how nicely vague |
| 05:57.53 | file[laptop] | so much that I'm passing out |
| 05:58.16 | convey | I was trying to be polite |
| 05:59.49 | convey | But I appreciate your sarcasm |
| 06:00.07 | Qwell | he was being polite |
| 06:00.17 | file[laptop] | I'm literally passing out, I'm soooooo tired |
| 06:00.22 | Qwell | ironically, you'd know it if he were being sarcastic. |
| 06:00.39 | convey | lol |
| 06:00.59 | Qwell | and I cannot believe I spelled that right |
| 06:00.59 | file[laptop] | but seriously, it's very vague |
| 06:01.16 | convey | I wasn't really taking it seriously.. :) |
| 06:01.26 | convey | but I was trying to be polite |
| 06:02.01 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 06:02.38 | convey | here is the story. When I dial my DID the call never even hits my dialplan, and it is a simple two line dialplan. I see the channel created when I type show sip shannels.. but that is it. |
| 06:02.48 | file[laptop] | sip debug |
| 06:03.03 | alephcom | amen. I bet that you don't have a line that matches it. |
| 06:03.18 | convey | a line? |
| 06:03.23 | convey | please elaborate |
| 06:03.32 | alephcom | sorry, an extension in extensions.conf |
| 06:03.52 | alephcom | sip debug is your friend. |
| 06:04.12 | Corydon76-home | Does the DID come in via a SIP provider or via a Zap channel? |
| 06:04.27 | convey | SIP provider |
| 06:04.29 | brettnem | argh.. anyone know what would cause asterisk to simply not play any audio to sip channels? I see it playing it.. but I can't hear anything.. Going right to milliwatt works jsut fine.. |
| 06:04.53 | Vco | earwax |
| 06:05.09 | Corydon76-home | Do you have a matching extension in your dialplan for that DID, in the context you specified for your provider (in sip.conf) ? |
| 06:05.15 | brettnem | I'm trying to go into the Directory application.. I see 'Playing 'dir-intro'' but don't hear anything SDP looks good |
| 06:05.44 | Corydon76-home | brettnem: did you Answer the line? |
| 06:06.06 | brettnem | yeah, I don't usually, but I am now.. still not working |
| 06:06.15 | brettnem | I answer and Wait 1 |
| 06:06.25 | brettnem | I get the same thing going to VoicemailMain |
| 06:06.47 | brettnem | no error about the audio file missing.. |
| 06:06.53 | convey | Corydon76-home: I have context in sip.conf that matches context in extensions.conf that sends the caller to my IP phone. |
| 06:07.27 | Corydon76-home | convey: but it doesn't ring your IP phone? |
| 06:07.54 | convey | Corydon76-home: I don't even see the call answered on the CLI. |
| 06:07.58 | Qwell | gah...freaking cisco |
| 06:08.06 | Qwell | wrap their firmware in windows executables |
| 06:08.12 | brettnem | Corydon-w: another weird thing.. So I don't hear the greeting to the Directory, but if I type a name, It plays back the first letter of the user.. then for each DTMF I send to *, it plays back (well it attempts to play) back an additional character out of the name. |
| 06:08.13 | Corydon76-home | convey: do you have verbose set to 3 or above? |
| 06:08.26 | convey | Corydon76-home: set to 10 |
| 06:09.00 | convey | Corydon76-home: I can make OB calls with no problems |
| 06:09.05 | Corydon76-home | convey: yet a 'show channels' finds the channel coming in? |
| 06:09.26 | convey | Corydon76-home: Yes I see a channel with my providers IP attached |
| 06:09.31 | brettnem | ugh, this is weird |
| 06:09.41 | dsasad | does anybody know if Zhone channel bank is compatible with asterisk ? |
| 06:09.52 | Corydon76-home | convey: pastebin your extensions.conf and sip.conf |
| 06:09.56 | Corydon76-home | ~pastebin |
| 06:09.57 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 06:10.00 | Qwell | dsasad: mog_work was telling me yesterday that he likes them |
| 06:10.05 | convey | Corydon76-home: ok |
| 06:10.09 | *** topic/#asterisk by drumkilla -> Asterisk 1.2.0 RC1 has been released! Please try it out! || http://www.asterisk.org |
| 06:10.24 | dsasad | oh that's great .. because they're pretty cheep T1 channel bank for $150 |
| 06:10.51 | Math` | $ cat /etc/asterisk/extensions.conf | pastebin |
| 06:10.51 | Math` | http://pastebin.ca/12345 |
| 06:10.52 | Math` | $ |
| 06:11.09 | brettnem | Math`: that'd be neat |
| 06:11.15 | Qwell | Math`: friend of mine wrote one for uuoc.com (which he also wrote) |
| 06:11.30 | Math` | its kinda simple to do |
| 06:11.33 | Qwell | yeah |
| 06:14.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:14.06 | convey | Corydon76-home: http://pastebin.ca/28093 |
| 06:14.55 | convey | Corydon76-home: Running CVS Head BTW. |
| 06:15.23 | brettnem | Corydon-w: any ideas for my weird no audio problem? |
| 06:15.47 | Corydon76-home | convey: it's highly unlikely that teliax is sending the "s" extension |
| 06:16.22 | Corydon76-home | convey: besides, you have two lines, both at priority 1, so only the first is going to get executed |
| 06:16.23 | convey | Corydon76-home: Ahh, I see that... |
| 06:17.01 | Corydon76-home | once again, we chalk up the problem to a typo... |
| 06:17.16 | Math` | lol |
| 06:17.49 | convey | Corydon76-home: Yeah too many changes in one day :) |
| 06:17.50 | Corydon76-home | brettnem: no idea |
| 06:17.59 | convey | Corydon76-home: Thanks!!! |
| 06:18.05 | brettnem | blah |
| 06:18.59 | Corydon76-home | dsasad: Yeah, the zhone will work fine... as long as you don't want callerid |
| 06:22.02 | brettnem | so anyone know how * holds up with extension files several thousand lines long? |
| 06:23.38 | jontow | brett; i've heard of no length issues with the dialplan |
| 06:24.01 | brettnem | so if I made one.. 15,000 lines long, you think it'd work?? |
| 06:24.09 | jontow | i think you'd be alright |
| 06:24.16 | jontow | best bet.. generate a fake one with a perl script or something |
| 06:24.35 | brettnem | neat.. I'll give it a whirl |
| 06:24.42 | JonR800 | do you really need an extension.conf 15,000 lines long? |
| 06:24.54 | jontow | say, extension 00000 through 15000 with maybe 3-5 actions a piece, maybe a couple macros, whatever |
| 06:24.57 | jontow | see if it flies |
| 06:25.02 | brettnem | well it's either that or I do some advanced routing in SER |
| 06:25.07 | brettnem | which I'm doing now.. |
| 06:25.12 | JonR800 | wouldn't there be a simpler way to do it, i can only imagine the maintenance. |
| 06:25.14 | jontow | JonR800: if he's got an awful lot of clients, maybe its necessary |
| 06:25.27 | jontow | i can't comment on 15k lines, hehe |
| 06:25.28 | brettnem | nah, it's not clients.. it's definitions for local calling scopes.. |
| 06:25.41 | jontow | i know IAX tends to perform horribly with that many registrations ;) |
| 06:26.00 | jontow | but that was tested quite some time ago, in exactly the manner i described above, but with iax.conf, not the dialplan |
| 06:26.34 | brettnem | like I said, it's not clients.. it's definitions for local calling scopes |
| 06:26.37 | JonR800 | wouldn't there be some way to script that? maybe that'd give worse performance.. i dunno. |
| 06:27.05 | jontow | sucks, but its probably necessary at some level or another. |
| 06:27.14 | brettnem | like agi?: yeah, I think that'd kill performance.. |
| 06:27.22 | brettnem | yeah I do it with SER now.. but it's hard to manage |
| 06:29.56 | jontow | AGI is slow |
| 06:30.03 | jontow | not scalable at all, in fact |
| 06:30.09 | jontow | ask anyone who has tried ;) |
| 06:30.48 | brettnem | yeah I know.. |
| 06:31.12 | brettnem | in fact, I haven't bothered testing or learning it to avoid those issues.. was afraid I'd love the flexibility and then get hammered on it |
| 06:31.55 | jontow | indeed |
| 06:32.08 | jontow | have heard that one a few times |
| 06:32.13 | brettnem | just trying to find an elegant want to define my calling scopes |
| 06:42.37 | alephcom | Is fastagi no more scalable than the other methods? |
| 06:45.47 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 06:45.53 | Uberbot | Hi all. |
| 06:46.23 | Uberbot | Does anyone know if get_variable() works under 1.0.9? (AGI) |
| 06:50.40 | *** join/#asterisk ve6tcg (n=denis@S01060040f433f249.ed.shawcable.net) |
| 06:50.52 | *** join/#asterisk Msalim1 (n=msalim_5@210.211.246.47) |
| 06:51.06 | Msalim1 | hello there |
| 06:51.08 | *** join/#asterisk WillySilly (n=WillySil@c-24-23-145-194.hsd1.ca.comcast.net) |
| 06:51.34 | Msalim1 | I am unable to log the CDR records from the asterisk.... |
| 06:51.47 | Msalim1 | i have configured the MYSQL |
| 06:52.32 | Qwell | Msalim1: what happens? |
| 06:52.43 | Msalim1 | ok |
| 06:53.29 | *** part/#asterisk WillySilly (n=WillySil@c-24-23-145-194.hsd1.ca.comcast.net) |
| 06:53.31 | Msalim1 | I have configured the MySQL in asterisk in the cdr_mysql.conf |
| 06:54.09 | Msalim1 | but when I am realoading the asterisk , I am gettting the message "Unregistered 'mysql' CDR backend". |
| 06:54.23 | Qwell | That isn't a bad thing, or anything... |
| 06:54.34 | Msalim1 | I shall paste the log on the pastebin |
| 06:54.41 | Qwell | sure |
| 06:54.44 | Msalim1 | will you check it out |
| 06:54.48 | Msalim1 | ok wiat a sec |
| 06:54.49 | Msalim1 | please |
| 06:55.45 | Msalim1 | here is the link: http://pastebin.com/422715 |
| 06:55.50 | Msalim1 | please check it out |
| 06:56.38 | Qwell | okay? So what's the problem? |
| 06:57.05 | Msalim1 | but the thing is I'm not able to log the CDR records for the call |
| 06:57.13 | Qwell | what happens? |
| 06:57.15 | Msalim1 | is there some the reason for it |
| 06:57.20 | Msalim1 | notthig |
| 06:57.24 | Qwell | Do you get errors? Does your computer catch fire? |
| 06:57.27 | Msalim1 | I am not getting anything in the DB |
| 06:57.34 | Msalim1 | no |
| 06:57.50 | Msalim1 | the call is placeed properly but I am not getting the CDR in the DB.. |
| 06:58.16 | Qwell | and you don't get anything at all on the CLI? |
| 06:58.27 | Qwell | I highly doubt that... |
| 06:58.55 | Math` | set verbose 10? |
| 06:59.08 | Msalim1 | no I'm not getting any notification of the logging of the CDR records |
| 06:59.12 | Msalim1 | on the CLI |
| 06:59.28 | Qwell | turn up verbosity, and try again |
| 06:59.47 | Msalim1 | how do I set the verbosity? |
| 07:00.11 | Qwell | set verbose 10 |
| 07:00.16 | Msalim1 | ok |
| 07:00.21 | Msalim1 | on the CLI? |
| 07:00.51 | Math` | ya |
| 07:00.55 | Msalim1 | yup I did |
| 07:02.34 | Msalim1 | nop, i am still not able to get thru... :( |
| 07:02.47 | Qwell | you don't get anything at all on the CLI? |
| 07:02.59 | Math` | not even the call showing up? |
| 07:03.10 | Msalim1 | only the details of the call setup |
| 07:03.24 | Msalim1 | nothing related to logging of the CDr |
| 07:03.39 | Qwell | still doubting it |
| 07:03.42 | Qwell | pastebin your cli output |
| 07:03.47 | Msalim1 | ok |
| 07:04.56 | Msalim1 | here is the CLI output: http://pastebin.com/422719 |
| 07:05.00 | Msalim1 | please check it |
| 07:05.30 | Qwell | there should be more |
| 07:06.45 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 07:06.51 | lehel | mornin |
| 07:07.17 | Msalim1 | no this is it that I am getting |
| 07:07.30 | Msalim1 | once I place a call |
| 07:13.10 | Msalim1 | Qwell are you there |
| 07:13.30 | Qwell | You get CDRs when you hangup... |
| 07:14.43 | Msalim1 | no... |
| 07:16.08 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:25.01 | mazzanet | uh huh... |
| 07:25.07 | mazzanet | Nov 9 18:24:30 NOTICE[6127]: chan_iax2.c:6741 socket_read: Rejected connect attempt from 192.246.69.187, who was trying to reach '697482@' |
| 07:25.27 | mazzanet | <PROTECTED> |
| 07:25.27 | mazzanet | d |
| 07:25.31 | mazzanet | what does that mean? |
| 07:28.48 | *** part/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 07:30.22 | morale | is CVS HEAD asterisk-rc1? or do i have to checkout a different cvs branch? |
| 07:31.21 | denon | anyone know where the voicemail gain setting is? cant remember .. |
| 07:34.04 | mazzanet | hmmm so the trunk name actually matters for iax |
| 07:37.12 | konfuzed | wow is zhone.com the same as the zhone xhannel bank mentioned earlier |
| 07:39.13 | konfuzed | how do i stop drooling |
| 07:39.17 | konfuzed | ;^) |
| 07:39.32 | konfuzed | ;^),,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,, |
| 07:39.49 | sjobeck99 | stop that |
| 07:40.39 | konfuzed | cause its late |
| 07:40.43 | konfuzed | wait this is the internet |
| 07:41.04 | morale | the internet is only microsoft internet explorer |
| 07:41.16 | konfuzed | I got this building of 40 units |
| 07:41.26 | konfuzed | morale: dont be so silly |
| 07:42.18 | konfuzed | the guy on the board is practically begging me to put IPTV on the network going to be installed there next month |
| 07:42.43 | mazzanet | shoutcast nsv :P |
| 07:42.52 | konfuzed | i just wanted to give em a PBX with optional DIDs |
| 07:43.47 | konfuzed | seriously there cable co TV package for 40 is up |
| 07:43.52 | konfuzed | and they wanna jump |
| 07:44.00 | *** join/#asterisk Jakes (n=jakes@196.34.241.242) |
| 07:44.22 | konfuzed | looks like zhone has got the hardware to really make a fat pipe happen |
| 07:44.31 | konfuzed | and really deliver on ipTV |
| 07:49.09 | konfuzed | wow ive not looked at any iptv or high volume gear like that before |
| 07:50.19 | konfuzed | im so pleased they did not spoil it by posting the prices ;^) |
| 07:51.51 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 07:57.31 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 07:59.04 | konfuzed | oh man i gotta call em when the sun is up |
| 07:59.14 | konfuzed | just to hear the bad news on prices |
| 08:11.46 | IronHelix | i bet they are used to hearing peoples hearts sink when they talk pricing |
| 08:12.04 | konfuzed | yeah |
| 08:12.40 | konfuzed | but I got a property of 40 apartments that can finance something |
| 08:12.48 | IronHelix | caller: HEY YOUR SHIT ROCKS, HOW MUCH IS IT? CSR: its $xxxxxxxxxxx.xx caller: Oh. nevermind. thanks :(. (click) |
| 08:12.53 | konfuzed | if its viable to replace the existing cable contract |
| 08:13.14 | konfuzed | yaya |
| 08:13.17 | konfuzed | I cant wait |
| 08:14.03 | konfuzed | they guy already wants me to put in gigabit lan |
| 08:15.27 | konfuzed | im like too expensive |
| 08:15.35 | konfuzed | hes like please |
| 08:15.46 | IronHelix | tell him this |
| 08:15.49 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:15.52 | IronHelix | you will run cat5e cable everywhere |
| 08:15.53 | IronHelix | or cat6 |
| 08:16.02 | IronHelix | you will try to get him one of the few gbit ports |
| 08:16.04 | IronHelix | but no guarantees |
| 08:16.33 | konfuzed | yeah Im gonna have to quote that |
| 08:16.38 | konfuzed | I suck shit at sale |
| 08:16.49 | konfuzed | thanks man |
| 08:17.32 | konfuzed | i did tell him they got a relatively small project cause were target 100 to 300 unit MUDs |
| 08:17.38 | IronHelix | np |
| 08:17.58 | IronHelix | eh |
| 08:18.00 | IronHelix | that can backfire |
| 08:18.04 | IronHelix | makes them feel not important |
| 08:18.10 | IronHelix | i assume you are wiring the condo? |
| 08:18.25 | konfuzed | and all this expensive fantasy (well didnt really say fantasy) he'll have to pay extra for |
| 08:18.43 | IronHelix | or do you own / work for the condo and the guy lives there |
| 08:18.45 | konfuzed | well ya my gig is the lan |
| 08:19.00 | konfuzed | my buddy will do the * pbx |
| 08:19.02 | IronHelix | im saying is the guy/client the dude that owns the building or a tenant? |
| 08:19.29 | konfuzed | and check this out http://100bain.com/~videovic |
| 08:19.42 | konfuzed | the buy is on the board at the building |
| 08:19.47 | konfuzed | 8 people |
| 08:20.45 | konfuzed | he's relatively new into it but already knows of asterisk |
| 08:21.23 | IronHelix | sounds like a good guy |
| 08:21.25 | konfuzed | so hes all copesetic with my idea of complete open source solution |
| 08:21.25 | IronHelix | early adopter |
| 08:21.43 | konfuzed | videovic is my buddy not the client |
| 08:21.53 | konfuzed | but he could implement the iptv |
| 08:22.23 | konfuzed | and nearly implemented satelite cable replacement for the local rogers cable |
| 08:22.38 | konfuzed | for that coop he is at any way |
| 08:22.51 | IronHelix | cool |
| 08:23.03 | konfuzed | where vic is the coop is 260 units |
| 08:23.21 | konfuzed | but the place is so fucked up politcally its embarassing |
| 08:23.39 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 08:23.41 | *** join/#asterisk ComputerWarm (n=dan@rddrpx27-port-47.dial.telus.net) |
| 08:23.46 | zobia | Hello everyone |
| 08:23.53 | konfuzed | they got a free computer lab given to them by a few members and a grant from the government |
| 08:24.01 | zobia | i make the realtime of extension of asterisk with mysql |
| 08:24.06 | IronHelix | lol wow |
| 08:24.11 | IronHelix | thats wierd kon |
| 08:24.24 | zobia | now i want to add the context to the dialplan. anybody knows how to do? |
| 08:24.38 | konfuzed | then a few on the board were so pissed off about how it got done that they actually treat it like and believe it is a seperate legal entity |
| 08:24.55 | konfuzed | it gets worse |
| 08:24.59 | konfuzed | but I wont go into it |
| 08:25.04 | konfuzed | tooo stupid |
| 08:26.00 | IronHelix | yeah its always dumb shit like that |
| 08:26.19 | konfuzed | for me its the stuff thats even worse ;^) |
| 08:26.51 | konfuzed | id rather talk about zhone at 40 unit coop that wants to put iptv and gig lan with asterisk |
| 08:27.00 | konfuzed | this guy is sharp |
| 08:27.17 | konfuzed | see im smiling again ;^) |
| 08:28.21 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 08:30.25 | IronHelix | hehe |
| 08:30.28 | IronHelix | yeah so would i any day |
| 08:30.29 | konfuzed | i guesstimate the only parts of the IPTV puzzle we dont have is the zhone customer NID and hopefull some form of iptv content proxy that sits at the existing DSL providers datacenter. |
| 08:30.56 | konfuzed | happens to be the same data center that can pipe me DID via IAX |
| 08:31.34 | konfuzed | and of course the dsl connections to the building |
| 08:31.51 | konfuzed | oh this looks to good |
| 08:32.00 | konfuzed | that gear must be expensive |
| 08:33.00 | *** join/#asterisk ozzzz (n=dsasad@213.186.187.84) |
| 08:33.19 | konfuzed | hmmm if we extend the service agreement to include seperate monthly billing for Internet Access and VoIP on Asterisk and IPTV cable package |
| 08:33.22 | ozzzz | Hello everyone, Does anybodoy know how to add a VOIP gateway to asterisk ? |
| 08:33.27 | ozzzz | h323 gateway |
| 08:33.55 | ozzzz | i have downloaded and installed the plugin. |
| 08:34.10 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c938qu.cable.mindspring.com) |
| 08:34.20 | konfuzed | then it could be a viable monthly agreement worthy of some financing |
| 08:34.25 | konfuzed | presto |
| 08:34.43 | konfuzed | the nerd collective is in the Cable TV business too |
| 08:34.49 | konfuzed | ;^) |
| 08:35.33 | konfuzed | uhm TriplePlay Voice Video Data for MUD |
| 08:35.34 | *** join/#asterisk grimse (n=grimse@p5481E413.dip.t-dialin.net) |
| 08:35.41 | konfuzed | hm |
| 08:36.29 | konfuzed | IronHelix: ever done anything like triple play before |
| 08:37.21 | IronHelix | nah, not my market, i dont wanna deal with real customers :) |
| 08:37.30 | IronHelix | i mostly do generic technical consulting |
| 08:37.49 | IronHelix | * until now has been a hobby but i'm trying to get into installing * pbx's and servicing them |
| 08:37.51 | IronHelix | lots of $$ there |
| 08:37.56 | IronHelix | people hate their phone vendors |
| 08:40.10 | ozzzz | anybody knows to add a h323 voip gateway to be managed by asterisk ? |
| 08:43.34 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 08:45.55 | mazzanet | IronHelix: heh i have a nice an ancient commander pbx at work |
| 08:46.03 | *** join/#asterisk infinity1 (n=brendon@solara.netcal.com) |
| 08:46.04 | IronHelix | hehehe |
| 08:46.25 | mazzanet | commander d series to be exact |
| 08:46.30 | mazzanet | little over 10 years old |
| 08:46.35 | IronHelix | wow |
| 08:46.42 | IronHelix | thats ancient in computer terms |
| 08:46.51 | mazzanet | has ~60 internal exts running off it |
| 08:47.34 | *** join/#asterisk DYOGI_E (i=DYOGI_E@dsl-202-173-191-109.qld.westnet.com.au) |
| 08:47.43 | DYOGI_E | hi from the gold coast |
| 08:47.53 | DYOGI_E | has anyone used Xoom-RAPID to set up asterisk?? |
| 08:49.04 | mazzanet | i'm really not a fan of prepackaged + mostly configured apps |
| 08:49.41 | mazzanet | takes the 'fun' out of doing it all manually |
| 08:49.59 | IronHelix | agreed |
| 08:50.19 | DYOGI_E | hmm |
| 08:50.36 | mazzanet | plus by doing it myself i know *exactly* where everything is/what it's doing |
| 08:51.01 | DYOGI_E | i will 110% agree with that, stuff is all over the place |
| 08:51.34 | DYOGI_E | having a bit of trouble with the voicemail will not accept passwords even when they are stipulated in the conf? |
| 08:51.50 | DYOGI_E | any ideas on how I could fix that? or if you have that prob before |
| 08:52.15 | konfuzed | wow thats fun window shopping |
| 08:52.27 | konfuzed | i know what I want for christmas ;^) |
| 08:52.52 | IronHelix | dyo- are the users in the right context? |
| 08:54.44 | DYOGI_E | possibly not the Xoom-Rapid has a really weird dial plan that I just can't get my head around could that be why? |
| 08:54.49 | delmar | Can someone please confirm that doing a "cvs checkout zaptel" as opposed to "cvs checkout -r HEAD zaptel" would or would not fetch/update different branches.. or are they the same.. ie dont specify any revision.. = HEAD by default.. whats the story? |
| 08:55.52 | Qwell | HEAD is the very latest |
| 08:55.57 | delmar | i know this |
| 08:55.59 | Qwell | (and is the default) |
| 08:56.12 | delmar | doesn't seem to be the case.... |
| 08:56.24 | syle | anyone have PRI line or VOIP site |
| 08:57.58 | delmar | Qwell, i have the latest head branch and WAS having huge echo problems... I pulled down zaptel doing "cvs checkout zaptel" and compiled it and tried... echo gone... to re-confirm the head version.. i pulled down a fresh HEAD of zaptel.. compiled and installed the modules... echo returns... now I can swap between the modules and have echo .. or not. |
| 08:58.02 | *** part/#asterisk oej (n=Olle@apollo.webway.se) |
| 08:58.34 | kb1_kanobe | delmar: interesting... go into ztconfig.h and see what echo canceller is defined in each plz. |
| 08:58.53 | Qwell | delmar: try to cvs up -A |
| 08:59.00 | kb1_kanobe | sorry, make that zconfig.h |
| 08:59.07 | syle | ummm |
| 08:59.10 | syle | cd zaptel |
| 08:59.12 | delmar | kb1_kanobe, kb1 in both |
| 08:59.12 | syle | make update |
| 08:59.27 | Qwell | you probably have sticky flags set |
| 08:59.36 | kb1_kanobe | delmar: that's very wrong... you're saying that the latest zaptel code is malfunctioning? |
| 08:59.39 | kb1_kanobe | (wrt echo?) |
| 08:59.41 | Qwell | but... -r is sticky, supposedly |
| 08:59.45 | heroine | matin \o/ |
| 09:00.39 | Qwell | hell, checkout both, and do a diff between them |
| 09:00.44 | delmar | kb1_kanobe, if i use the latest zaptel HEAD with the same asterisk zapata settings and with the same /etc/fxotune.conf settings etc... i get echo. |
| 09:00.59 | kb1_kanobe | what card is this plz? |
| 09:01.12 | delmar | kb1_kanobe, how can I run a DIFF on the two zaptel CVS's i have? maybe that would be the go? |
| 09:01.22 | delmar | kb1_kanobe, TDM400 |
| 09:01.25 | kb1_kanobe | errr... there will be a lot of changes. |
| 09:01.27 | delmar | with 1FXO |
| 09:01.46 | Qwell | cvs checkout -r HEAD zaptel && mv zaptel zaptel-head && cvs checkout zaptel && diff -ur zaptel zaptel-head |
| 09:01.47 | delmar | ok my point is.... |
| 09:01.49 | kb1_kanobe | that's quite interesting... I only have ISDN trunks myself. |
| 09:02.40 | kb1_kanobe | kb1.h and kb1_const.h is (supposed to be) identical in both branches. |
| 09:02.54 | delmar | ok, gonna check out some stuff. brb |
| 09:03.16 | kb1_kanobe | So something else is messing with your performance. Possibly data transfer issues to/from card. Possibly there is a problem w/transcoding somewhere... |
| 09:03.38 | Qwell | all the files between -r HEAD and without a revision are exactly the same |
| 09:03.40 | *** join/#asterisk h3x (n=h3xor@64.192.116.16) |
| 09:03.47 | Qwell | (as expected) |
| 09:04.07 | kb1_kanobe | only within zaptel? |
| 09:05.06 | Qwell | user error |
| 09:05.16 | delmar | mine seem to have differences.... |
| 09:05.23 | delmar | (sigh) |
| 09:05.29 | Qwell | <Qwell> delmar: try to cvs up -A |
| 09:05.42 | delmar | Qwell, what does that do? |
| 09:05.44 | kb1_kanobe | There are a fair amount of changes to zaptel, but mostly card level issues - new cards etc. |
| 09:05.51 | Qwell | delmar: cvs --help up |
| 09:06.00 | Qwell | kb1_kanobe: not between head and...head |
| 09:06.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:06.10 | delmar | ah ok |
| 09:06.27 | kb1_kanobe | Qwell: ah, yes, of course... :-) |
| 09:06.27 | delmar | bah |
| 09:07.05 | *** join/#asterisk jebba (n=jebba@ip-216-17-154-58.rev.frii.com) |
| 09:07.10 | delmar | fact is, somehow, i have a zaptel that is not the latest HEAD or differs somehow, and using it instead of what comes with HEAD right now, is better |
| 09:07.20 | *** join/#asterisk MajestiK (n=MajestiK@S0106000024c058cc.ed.shawcable.net) |
| 09:07.27 | Qwell | so figure out what changed |
| 09:07.29 | delmar | so it aint broke.. so ill stop stressing on that issue.... |
| 09:07.33 | kb1_kanobe | delmar: do please diff the kb1.h and kb1_const.h files. |
| 09:07.42 | delmar | kb1_kanobe sure. sec |
| 09:09.00 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 09:09.46 | Qwell | bed |
| 09:09.47 | kb1_kanobe | delmar: this is quite important. If you're absoloutly certain that the _newer_ zaptel malfunctions wrt echo, yet the echo canceller code diff's as identical then this deserves reporting in the bugtracker and inspection by others more familiar than I. |
| 09:10.15 | kb1_kanobe | Be sure to save exactly all of the two different zaptel directories for later inspection and comparison. |
| 09:10.16 | kb1_kanobe | brb |
| 09:10.40 | delmar | OK, there are no diffs between those two files at all |
| 09:10.54 | delmar | ill just confirm both Makefiles have same stuff too |
| 09:11.47 | zobia | Hello everyone |
| 09:11.52 | delmar | yep, same echo_can set too |
| 09:11.59 | zobia | does any body has experience for realtime extensions? |
| 09:13.14 | syle | just ask |
| 09:13.29 | zobia | okay. i have make the realtime extensions work. |
| 09:13.40 | zobia | but i have problem with goto of realtime extensions |
| 09:13.56 | zobia | i have two context in the realtime database. |
| 09:14.17 | zobia | but i can not use goto to jump from first context to the second context. |
| 09:14.29 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 09:14.35 | syle | you using it correctly? |
| 09:14.45 | zobia | but can use goto to jump to to static context dialplan |
| 09:15.10 | zobia | yes. if goto some context which is static in the extensions.conf then it will go |
| 09:15.20 | zobia | if goto the context which is a realtime context it won't go. |
| 09:16.02 | zobia | syle , any idea? |
| 09:16.05 | syle | i have 1 |
| 09:16.17 | zobia | have 1 context? |
| 09:16.20 | syle | rtcache? |
| 09:16.48 | zobia | what's rtcache? |
| 09:16.49 | kb1_kanobe | delmar: I would recommend reporting to bugs.digium.com, but expect some fairly close scrutiny. |
| 09:17.07 | zobia | u r talking to me syle? |
| 09:18.01 | delmar | kb1_kanobe, ill take a closer look tomorrow.. if what people say is correct.. that the default is HEAD... then all i have in that case.. is a newer Zaptel working better than the one I had only a day or two ago. |
| 09:18.08 | syle | yes |
| 09:18.14 | *** join/#asterisk demetrio (n=demetrio@host13-49.pool870.interbusiness.it) |
| 09:18.15 | syle | see if you can put rtcachefriends=no |
| 09:18.40 | zobia | okay. let me see |
| 09:18.44 | delmar | kb1_kanobe, at a glance i think this might be the case, so perhaps there was some fast haxing to fix the shitty echo problems on the HEAD version i was having issues with.. |
| 09:19.17 | demetrio | I'm not able to receive calls via IAX. tcpdump shows that packets are exchanged, but iax2 debug will say absolutely nothing. Why? |
| 09:19.24 | kb1_kanobe | unlikely - the only bit that does the actual echo cancellation is in kb1.h and kb1_const.h |
| 09:19.50 | kb1_kanobe | if there is something else that has negatively affected it then we need to know what that was because it might still be affecting other ppl. |
| 09:20.03 | delmar | kb1_kanobe, anyway.. that asside... can u or anyone help me with an echo problem i have in a different scenario... .siphardphone(ulaw) to Asterisk ===(iax/ulaw)===Asterisk to siphardphone(ulaw).. oh and the thing is.. with IAX/g729 between the * boxen, there was less echo.. still a pain in the ass but less than with ulaw... |
| 09:20.27 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 09:20.38 | delmar | kb1_kanobe, i will spend time on it tomorrow to confirm what the hell I have done LOL :P |
| 09:20.50 | delmar | kb1_kanobe, and if it's worthwhile i might make a post |
| 09:21.19 | cjk | hi, whats the way to withheld a number? for example i send my outbound traffic via sip to some carriers.... does anyone has experiences with this? |
| 09:21.29 | kb1_kanobe | delmar: please do, thanks! :-) |
| 09:22.09 | zobia | syle i put rtcachefriends=no in the general of sip.conf but it still won't go between context. |
| 09:23.54 | syle | whats sip.conf got to do with extensions.conf |
| 09:24.16 | delmar | anyone? help with sip to * IAX to * to sip echo problems please !!. |
| 09:25.48 | kb1_kanobe | delmar: it's probably acoustic echo inside one of the sip phones. |
| 09:26.04 | zobia | i only see rtcachefriends in my sip.conf so i changed this one. so .. now i put it in the general of extensions.conf . reload the asterisk .won't work either |
| 09:26.41 | delmar | kb1_kanobe, i decided to try a grandstream phone at this end.. same problem.. both ends hear echo of them selves. |
| 09:27.10 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:27.23 | delmar | kb1_kanobe there is about 170ms between here and there plus * translation if any, and latency between * and SIP phone at each end. |
| 09:27.47 | delmar | kb1_kanobe, but still, the phone at the far end shouldnt be "leaking" audio back to the other end... |
| 09:27.50 | kb1_kanobe | ah, so if there is acoustic echo it'll be painfully obvious. Either that or there is echo cancellation in one or both ends that's feeling ill. |
| 09:28.54 | delmar | kb1_kanobe, well if there is an echo_can that can be turned on/off on the grandstream i can't see it... however on the PA168X phones I have this and .. and remote.. they have echo can, vad, afc and all that crap... well thats ALL turned off.... |
| 09:29.09 | kb1_kanobe | err... not good. |
| 09:29.16 | delmar | reason being.. when I use those phones here between each other, or PSTN to * to PA168X.. they work MINT.... |
| 09:29.23 | delmar | with AEC turned on.. they do wierd things. |
| 09:29.47 | delmar | that asside.. the Grandstream this end.. should have an echo_can for sure....but the far end still hears themselves... |
| 09:29.56 | zobia | hello syle , any other idea? |
| 09:30.14 | *** join/#asterisk kks (n=kks@202.73.8.130) |
| 09:30.38 | delmar | kb1_kanobe, i dont see why there needs to be echo-can, the damn phone at the other end shouldnt be leaking the received audio back out again... |
| 09:30.54 | kb1_kanobe | it happens quite often within the handset. |
| 09:31.17 | kb1_kanobe | if the far end turns off their mike (ie. mutes) and you stop hearing your own voice, voila!, acoustic echo. |
| 09:32.13 | delmar | kb1_kanobe, right.. so ... the same way the handset deliberatly leaks audio from the mic back to the ear piece (because our brain expects to hear our voice when we speak) for comfort... the audio can leak back the other way.. form earpiece to the Mic.. ? |
| 09:32.28 | delmar | ah ok |
| 09:32.29 | delmar | i see |
| 09:32.42 | delmar | kb1_kanobe ah.. i have a test... |
| 09:33.02 | delmar | kb1_kanobe, i will setup a record/playback function within Asterisk at the other end... I bet there is no echo :P |
| 09:33.16 | kb1_kanobe | delmar: you're describing sidetone, which is by design (locally leaked signal). the delayed product would have to enter the microphone somewhere on a non-pstn, non-analog connection. |
| 09:33.29 | syle | you tried using switch? |
| 09:33.34 | *** join/#asterisk kks (n=kks@202.73.8.130) |
| 09:34.38 | delmar | kb1_kanobe, ok, test done from here to the remote * box... recording and playback .. no echo. not one bit. |
| 09:36.08 | zobia | yes |
| 09:36.13 | zobia | i used two swich |
| 09:36.14 | delmar | kb1_kanobe, so its the siphardphones.... i bet even .. if i used perhaps.. xlite and a headset tomorrow.. and tested to the office.. the person at the other end wont hear themselves.. but perhaps I will... since im sure the soft-phone wont have this.. acoustic echo. |
| 09:36.19 | zobia | witch |
| 09:36.27 | zobia | switch |
| 09:36.50 | zobia | i use two switch for this 2 context in the extensions.conf |
| 09:37.26 | *** join/#asterisk |stefan| (n=stefan@142cm74.cable.soderhamn-net.com) |
| 09:37.38 | kb1_kanobe | delmar: so long as you use a headset w/a boom mike, yes. For sure. |
| 09:37.41 | kb1_kanobe | good luck. |
| 09:37.58 | delmar | kb1_kanobe, yeah it's gotta be the phones doing this huh. |
| 09:37.59 | |stefan| | ey =) quick question. i've been checking the manpages etc. but i can't find any option for server reconnect timeout and so on. anyone know about this ? |
| 09:39.34 | delmar | kb1_kanobe, of course, these phones are all the same as a PC running a softphone really though arent they... all they are is an embedded PC running some embedded OS (probably Linux) and the handset is a Mic and Speaker jacked into the on-board sound..... |
| 09:40.06 | delmar | kb1_kanobe, hrm... so it still makes no sense why a softphone on a PC would work any better, but I will be trying that out tomorrow |
| 09:40.12 | kb1_kanobe | as a rule, correct. ie. the linksys pap2 is just an embedded linux device. |
| 09:40.35 | kb1_kanobe | softphone w/headset has no path for the sound to travel from earpiece to mike. |
| 09:40.50 | delmar | yeah, thats what i figure |
| 09:44.19 | delmar | so we are stuck between a rock and a hardplace.. the phones work nicer on the local lan with NO echo_can, and thats because there IS echo but its under 50ms and just heard like.. sidetone is on a regular phone... but if we enable the AEC, the echo will improve for VoIP calling.. and i say improve.. its still there... but then in doing that... there are other issues that are hard to explain .. to do with the audio of the phone when using it for pstn or si |
| 09:45.28 | syle | well |
| 09:45.33 | syle | i;m reading the source code |
| 09:45.51 | delmar | heh i know how to describe it.. it's kinda like.. the sidetone .. if you like.. being removed or messed with... but a little worse... hrm.. thats still not quite right but close as i can describe |
| 09:46.00 | syle | it would appear that a goto might not work as it does a select * on your context to begin with |
| 09:46.42 | zobia | what's this mean? |
| 09:47.00 | syle | snprintf(sql, sizeof(sql), "SELECT * FROM %s WHERE %s%s '%s'", table, newparam, op, newval); |
| 09:47.52 | syle | it means do something like |
| 09:48.03 | syle | exten => 701,1,Goto(testrealtime,s,1) |
| 09:48.19 | syle | [testrealtime] |
| 09:48.26 | syle | switch => Realtime/tester@extusers |
| 09:48.26 | syle | exten => s,3,Hangup |
| 09:48.28 | syle | or whatever |
| 09:49.35 | zobia | u mean i need to put the static context on the extensions.conf and put the switch inside for the targed context which i want to jump to? |
| 09:49.51 | syle | yes |
| 09:50.28 | syle | so lets say i want to jump to context blah |
| 09:50.39 | syle | switch => Realtime/blah@extusers |
| 09:50.55 | syle | then the code will read the blah context |
| 09:50.59 | *** join/#asterisk jazor (n=jazor@212.168.164.8) |
| 09:51.05 | jazor | hi all |
| 09:51.05 | zobia | i tried. it work like this way |
| 09:51.17 | morale | anyone use mysql realtime voicemail in asterisk? |
| 09:51.37 | syle | or try something like |
| 09:51.44 | syle | switch => Realtime/test1@extusers |
| 09:51.46 | syle | switch => Realtime/test2@extusers |
| 09:52.28 | syle | he really should have coded this so that switch could be called in mysql |
| 09:52.33 | zobia | yes. before i put these two context in the same context. now create another context for the second context and put a switch inside |
| 09:53.32 | zobia | yes. i think this is not full realtime. what i want is both extension and context both are realtime updated. |
| 09:54.20 | emdub | <PROTECTED> |
| 09:54.24 | syle | well here you go |
| 09:54.25 | zobia | the problem is whenever i add one new context i must change the extensions.conf manually also. this is what the user can not do. |
| 09:54.28 | syle | <PROTECTED> |
| 09:54.32 | syle | email him |
| 09:54.44 | zobia | Oh. okay, thank you . |
| 09:54.53 | zobia | i will email him. |
| 09:55.18 | zobia | syle. really thanks. |
| 09:55.24 | syle | np |
| 09:55.28 | pooh_ | did RC1 gain 1 extra SIP accounts field called 'fullcontact' ? |
| 09:56.25 | pooh_ | MySQL RealTime: Query Failed because: Unknown column 'fullcontact' in 'field list' |
| 09:56.47 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 09:57.24 | emdub | sup shido |
| 09:57.33 | shido6 | :) |
| 09:59.27 | syle | i don;t see fullcontact anywhere in the source |
| 10:00.09 | DYOGI_E | anyone know of any voip phones with the ability to connect a blue tooth headset?? |
| 10:00.20 | pooh_ | syle: strange... |
| 10:00.21 | joelsolanki | i have g729 license install in asterisk. i m using linksys pap2 and set " use the prefer codec only " prefer codec is g729. but when i make calls from both lines of pap2 1 works and 2 gives me error on console " no compatible codecs " |
| 10:00.31 | joelsolanki | what could be the problem ? |
| 10:00.46 | shido6 | how many licenses? |
| 10:01.01 | shido6 | and how many modules do you have loaded that have "g729" in them |
| 10:01.01 | joelsolanki | 4 |
| 10:01.02 | shido6 | :) |
| 10:01.10 | shido6 | show modules |
| 10:01.25 | joelsolanki | yes show modules show me the g729 license module installed. |
| 10:02.26 | pooh_ | syle: MySQL RealTime: Query: UPDATE sip_accounts SET ipaddr = '192.168.5.118', port = '5060', regseconds = '1131529866', username = '9011', fullcontact = 'sip:9011@192.168.5.118:5060' WHERE name = '9011' |
| 10:02.38 | pooh_ | syle: MySQL RealTime: Query Failed because: Unknown column 'fullcontact' in 'field list' |
| 10:02.47 | joelsolanki | any ideas ? |
| 10:03.05 | syle | well add the column fullcontact |
| 10:03.17 | pooh_ | syle: yeah... |
| 10:03.58 | pooh_ | joelsolanki: try 'show translation' and see if g729 is really active |
| 10:04.52 | joelsolanki | hmm ok let me check |
| 10:05.03 | syle | you are probably using cvs head |
| 10:05.12 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 10:05.44 | pooh_ | syle: me, no RC1 |
| 10:05.52 | shido6 | but how many MODULES do you have installed that say "g729" |
| 10:06.02 | pooh_ | syle: added the field 'fullcontact' => MySQL RealTime: Updated 1 rows on table: sip_accounts |
| 10:06.40 | pooh_ | syle: fullcontact contains :sip:9011@192.168.5.118:5060 |
| 10:06.53 | shido6 | tehn do you have disallow=all allow=g729 in sip/iax/ or whatever protocol.conf you're using? :) |
| 10:07.03 | syle | pooh |
| 10:07.09 | syle | is that sip connection natted |
| 10:07.19 | pooh_ | syle: no, LAN |
| 10:08.37 | syle | did you update asterisk-addons? |
| 10:09.12 | pooh_ | syle: yes, all of it |
| 10:09.14 | *** join/#asterisk |stefan| (n=stefan@142cm74.cable.soderhamn-net.com) |
| 10:09.39 | pooh_ | syle: it seems RC1 contains 1 additional SIP accounts field, simple as that |
| 10:09.45 | *** join/#asterisk ful|work (n=fulgas@209.8.233.163) |
| 10:10.08 | syle | yep |
| 10:10.19 | ful|work | hey |
| 10:11.05 | joelsolanki | pooh: what should be the output in show translation ? |
| 10:11.22 | joelsolanki | it shows me g729 line |
| 10:11.50 | pooh_ | joelsolanki: does it contain numbers or just - |
| 10:12.47 | joelsolanki | it is only - |
| 10:12.50 | joelsolanki | no number |
| 10:13.54 | pooh_ | joelsolanki: so g729 is NOT active/loaded |
| 10:14.11 | joelsolanki | oh ..show modules shows me loaded. |
| 10:14.15 | pooh_ | syle: added info to the wiki RealTime SIP |
| 10:14.18 | joelsolanki | how to activated |
| 10:14.46 | pooh_ | joelsolanki: it is active automagically when loaded, licence correct? |
| 10:15.20 | joelsolanki | yes i downloaded from digium and they send me some keys. |
| 10:15.45 | joelsolanki | does this license works on 1 machine only ? |
| 10:16.49 | pooh_ | joelsolanki: dunno |
| 10:17.05 | joelsolanki | something is messed up. i will install asterisk again. |
| 10:17.17 | *** join/#asterisk Assid (n=assid@203.115.64.59) |
| 10:28.48 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 10:39.35 | nesys | hi folks |
| 10:39.58 | nesys | someone using g729 pass-through on freebsd platform? |
| 10:48.29 | kb1_kanobe | Dear world... I am so pissed off. FYI, never decomission a 39Gb harward raid array drive that contains a live 29Gb Exchange Server primary mailstore, particularly while the store is up and running and you haven't backed it up yet today, thinking incorrectly that it was on the other array... :-( |
| 10:49.32 | h3x | hehe if you cant handle exchange |
| 10:49.38 | h3x | just wait til you make everybodys phones quit working |
| 10:49.48 | kb1_kanobe | been there, done that. |
| 10:49.49 | syle | windows based server |
| 10:49.51 | syle | ohh god |
| 10:49.53 | syle | just quit |
| 10:50.04 | kb1_kanobe | physical pain. |
| 10:50.12 | kb1_kanobe | what a spectacular fuckup. |
| 10:51.20 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:51.23 | kb1_kanobe | thanks for sharing my pain all - now I shall go get 4hrs sleep before I have to explain to the CAO what just happened... there is still a glimmer, I can possibly rebuild the array metadata, however... |
| 10:51.48 | h3x | chief asshole officer? |
| 10:52.17 | morale | ls |
| 10:52.19 | kb1_kanobe | No, not really. Just the man with the big stick - I expect this issue will side step my boss, the General Manager. |
| 10:52.21 | h3x | rm |
| 10:52.54 | h3x | Just tell them the whole internet went down. ... "haven't you been watching the news?" |
| 10:53.02 | kb1_kanobe | lol! thanks for the grin. :-) |
| 10:53.26 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 10:53.35 | KriS83 | Hi @ all |
| 10:56.10 | inspired | hi, after I upgraded my kernel from 2.6.8-smp to 2.6.12-smp (debian kernel) yesterday I have been getting a strange problem. suddenly after a SIP call I get the following error and SIP stops working: Failed to grab lock, trying again |
| 10:56.27 | RoyK | http://karlsbakk.net/fun/printer-problem-mouse-is-stuck.png |
| 10:57.19 | mutilator | mornin all |
| 10:57.38 | h3x | aw poor mouse |
| 10:58.16 | *** join/#asterisk Phantom-X (n=there@212.31.186.47) |
| 10:59.17 | *** join/#asterisk Phantom-X (n=there@212.31.186.47) |
| 11:01.50 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
| 11:01.51 | christo | hi all |
| 11:02.09 | KriS83 | I need some assistance with zaptel/asterisk configuration. I get the following error: http://pastebin.ca/28109 |
| 11:03.46 | pooh_ | KriS83: what is your setup, what did you do? |
| 11:04.31 | pooh_ | RoyK: is there a way to force a sip phone to register from * next to the register timeout on the phone itself ? |
| 11:04.59 | KriS83 | Basicly I downloaded the CVS from http://www.junghanns.net/en/download.html |
| 11:05.24 | KriS83 | And just executed the isntall.sh |
| 11:05.36 | KriS83 | This downloaded some files from cvs from digium etc. |
| 11:05.41 | KriS83 | compiled it and installed it |
| 11:05.47 | KriS83 | this was all successfull |
| 11:05.55 | KriS83 | then I did a modprobe zaptel |
| 11:06.11 | KriS83 | and a modins qozap.ko |
| 11:06.24 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 11:07.07 | RoyK | pooh_: i beleiv eyou should be able to just set the register timeout in sip.conf |
| 11:07.15 | RoyK | that should tell the phone to follow that |
| 11:07.33 | pooh_ | KriS83: you need to read the readme's that come with junghanns package and adjust the config files for zap*thing* |
| 11:07.53 | pooh_ | RoyK: so only a global parameter I guess |
| 11:08.12 | RoyK | pooh_: prolly quite easy to change, though |
| 11:08.16 | RoyK | use the source, luke |
| 11:08.26 | pooh_ | RoyK: no way to make a 'call' to the phone and say 'he register with me again' ... ;-) |
| 11:08.51 | RoyK | not that i know, no |
| 11:09.21 | pooh_ | RoyK: ok, thx |
| 11:11.32 | pooh_ | RoyK: I am fiddling with the MWI indication on the phone and dynamically changing the SIP account settings |
| 11:12.07 | pooh_ | RoyK: it works, but only after the phone registers itself again |
| 11:12.24 | RoyK | pooh_: yes. i know... |
| 11:13.34 | joelsolanki | pooh_: see i have provided some real outputs. can u plz look what could be the problem ? both channels of linksys pap2 doesnt work at a time. |
| 11:13.41 | inspired | how do I set gcc-3.3 to be used instead of gcc-4? |
| 11:13.42 | joelsolanki | pooh_: here it is http://pastebin.ca/28111 |
| 11:13.56 | inspired | is there some variable I can set? |
| 11:14.41 | pooh_ | joelsolanki: yup, now it should work if you allow g729 |
| 11:14.59 | KriS83 | pooh_, basicly I took the zapata.conf from an example they deliver with the package from their page. As the example is exacly what we'd like to do. |
| 11:15.10 | KriS83 | or do you mean the zaptel.conf? |
| 11:15.56 | joelsolanki | pooh_:yes my provider only supports g729. |
| 11:16.00 | pooh_ | KriS83: yes, that too. Do yu have a BRI installed ? |
| 11:16.40 | joelsolanki | pooh_: i have only allowed g729 in sip.conf but still it doesnt work |
| 11:16.52 | KriS83 | pooh_, 2x quadBRI |
| 11:17.07 | pooh_ | joelsolanki: make another call and pastebin it |
| 11:17.15 | KriS83 | 1x in TE Mode, 1x in NT mode |
| 11:17.17 | joelsolanki | ok let me do that. |
| 11:18.08 | pooh_ | KriS83: start with /etc/zaptel.conf, pastebin it |
| 11:18.18 | *** join/#asterisk guzzler (n=chatzill@213.63.190.66) |
| 11:19.12 | joelsolanki | pooh_: here it is |
| 11:19.14 | joelsolanki | http://pastebin.ca/28112 |
| 11:19.36 | joelsolanki | one call only works both calls doesnt work at same time :( |
| 11:21.23 | KriS83 | pooh_, basiclly this seems the problem I think, as I have no idea how to configure zaptel.conf. But here it is: http://pastebin.ca/28113 |
| 11:22.59 | pooh_ | joelsolanki: make sure YOU are using g729 and reloaded asterisk. Check the codec by ' show channels' when in call |
| 11:24.05 | pooh_ | KriS83: 3 spans ? |
| 11:24.36 | joelsolanki | yes its g729 linksys is also using g729 ...do u think any problem with linksys pap2 configuration ? |
| 11:24.51 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:24.56 | pooh_ | joelsolanki: nah, codec related, not config |
| 11:25.32 | pooh_ | joelsolanki: dial echo test locally and see what codec is being used |
| 11:25.48 | KriS83 | pooh_, I have only worked with * with a B1 Card before.. so no idea what is in the zaptel.conf nor what it means. |
| 11:26.01 | KriS83 | Basicly I'd like to have a simple example |
| 11:26.08 | KriS83 | and learn from there |
| 11:26.10 | pooh_ | KriS83: low level config for the zaptel hardware |
| 11:26.13 | morale | is it much cheaper implementing a pstn phone line pbx or asterisk pbx? its going to cost pretty much the same amount to run a t1 loop isn't it? the only difference it is probably easier to mange.. |
| 11:26.49 | pooh_ | KriS83: pastebin lspci |
| 11:27.28 | pooh_ | KriS83: and visit voip-info for the basics of zaptel |
| 11:27.39 | KriS83 | http://pastebin.ca/28114 |
| 11:27.43 | KriS83 | Thats lspci |
| 11:28.12 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@host.161.115.68.195.rev.coltfrance.com) |
| 11:29.05 | pooh_ | KriS83: cat /proc/interrupts |
| 11:30.31 | KriS83 | http://pastebin.ca/28115 |
| 11:30.32 | pooh_ | RoyK: what is wrong with this syntax pls? : realtime update sip_accounts name 9011 mailbox 201 |
| 11:30.49 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 11:30.50 | RoyK | nfi |
| 11:31.15 | pooh_ | RoyK: :-) |
| 11:31.35 | joelsolanki | pooh_: how do i test echo test :( |
| 11:31.42 | pooh_ | joelsolanki: dial 999 |
| 11:32.02 | pooh_ | KriS83: so hardware looks ok right ? |
| 11:32.38 | joelsolanki | ok |
| 11:33.14 | joelsolanki | pooh_: but i have ser in between linksys and asterisk. |
| 11:33.25 | KriS83 | pooh_, yepp |
| 11:33.26 | pooh_ | joelsolanki: ewwww |
| 11:33.34 | pooh_ | KriS83: are those junghanns cards ? |
| 11:33.43 | KriS83 | pooh_, yepp 2 of them |
| 11:33.51 | pooh_ | KriS83: contacted them already ? |
| 11:33.56 | *** join/#asterisk phpboy (i=flipside@196.26.21.108) |
| 11:34.41 | KriS83 | pooh_, no... will do if you (or someone else can't give me a hint ;) ) |
| 11:34.47 | pooh_ | KriS83: what dos ztcfg -vv tell you |
| 11:35.05 | joelsolanki | pooh_: should i directly create user in asterisk and check then ? |
| 11:35.44 | phpboy | hey all... this is what my extentions look like http://pastebin.ca/28116 ... what I wanna know is.. can I setup special extentions in that range... e.g. 6980 and point it to a zap number? |
| 11:35.48 | pooh_ | joelsolanki: asterisk <--> SER <--> Phone or what ? |
| 11:36.34 | KriS83 | http://pastebin.ca/28117 <- pooh_ that what it looks like.. (not very nice ehh?) I tried to load the wcfxo and now I got these errors |
| 11:36.39 | joelsolanki | right now its asterisk <--> Ser <--> linksys phone |
| 11:37.02 | pooh_ | phpboy: just add that exact extension, it is longer then the others so it will make a difference |
| 11:37.06 | *** join/#asterisk jpmcallister (n=jpmcalli@kapla.escelsa.com.br) |
| 11:37.46 | pooh_ | KriS83: time to call Junghanns |
| 11:37.55 | *** join/#asterisk Corndawg_ (i=whoisit@c-66-176-66-83.hsd1.fl.comcast.net) |
| 11:38.02 | KriS83 | pooh_, ok thanks for your help ;) |
| 11:38.09 | pooh_ | KriS83: np, good luck |
| 11:38.41 | pooh_ | joelsolanki: that looks ugly to test things |
| 11:38.46 | joelsolanki | pooh_: should i make linksys pap2 to connect to asterisk directly and then try ? |
| 11:38.54 | pooh_ | joelsolanki: yes |
| 11:39.22 | InfraRed | nOOb! |
| 11:39.48 | joelsolanki | ok i m changing wait. |
| 11:41.00 | pooh_ | InfraRed: YaheyYaheyYahey |
| 11:43.36 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 11:50.09 | guzzler | Hi all! I have an asterisk box that is also doing NAT. The calls between phones on the same lan behind the NAT or even the calls between phones beyond the NAT have no problems but the calls between a phone that is behind the NAT and phone that isn't, the audio just work in one way. Is there any Asterisk configuration to deal with this? Thanks! |
| 11:50.15 | joelsolanki | pooh_: i did the config. but i m facing the same error. |
| 11:50.27 | joelsolanki | pooh_: one line is working and one is not working. |
| 11:50.32 | myke420247 | guzzler: reinvite=no |
| 11:50.37 | myke420247 | and notransfer=yes |
| 11:50.42 | pooh_ | joelsolanki: pastebin the *complete* call flow |
| 11:50.46 | bon | what can be the reason |
| 11:50.50 | bon | that i don't hear the other side |
| 11:50.52 | bon | while making a call? |
| 11:50.56 | joelsolanki | k |
| 11:50.57 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 11:50.57 | bon | using ulaw/alaw/gsm |
| 11:51.00 | bon | didn't change a thing |
| 11:51.03 | pooh_ | bon: ports |
| 11:51.10 | bon | that is? |
| 11:51.22 | pooh_ | bon: RTP ports, NAT ? |
| 11:51.33 | bon | the machine is not behind a nat |
| 11:51.38 | bon | and i am accessing via public ip |
| 11:51.42 | pooh_ | any realtime guru around pls ? |
| 11:52.00 | pooh_ | bon: did it work before ? |
| 11:52.06 | bon | think it did |
| 11:52.13 | bon | just switched to realtime now |
| 11:52.14 | pooh_ | bon: *think* ? |
| 11:52.19 | bon | and somehow it doesn't anymore :) |
| 11:53.14 | KriS83 | pooh_, I've done it, without Junghanns. I've got 24 Channels configured |
| 11:53.30 | pooh_ | KriS83: cool, what was the issue |
| 11:53.44 | KriS83 | I googled for "zaptel.conf quadBRI" |
| 11:53.54 | KriS83 | It was misconfiguration of the zaptel.conf |
| 11:54.04 | pooh_ | KriS83: ok |
| 11:54.10 | guzzler | myke420247: Thanks for the tip. I'm going to try that! |
| 11:54.16 | KriS83 | I found an example and edited it to our need I think :) |
| 11:54.24 | pooh_ | KriS83: :-) |
| 11:54.43 | KriS83 | but I will go thru the zaptel.conf configuration Documenation now.. |
| 11:55.01 | pooh_ | KriS83: good idea ;-) |
| 11:55.05 | KriS83 | now I have it working and * running, I can start to understand what I have configured |
| 11:55.32 | pooh_ | KriS83: where are you located ? |
| 11:58.06 | KriS83 | pooh_, germany |
| 11:58.12 | KriS83 | Near Munich |
| 11:58.16 | pooh_ | KriS83: thought so :-) |
| 11:58.30 | KriS83 | Oi... what do you mean by that? :) |
| 11:59.03 | pooh_ | KriS83: nothing special, just adding up junghanns/time of day |
| 11:59.17 | KriS83 | ahh ok |
| 12:02.31 | zoa | pooh, whats your problem ? |
| 12:03.21 | pooh_ | zoa: pbx*CLI> realtime update sipfriends name 9011 mailbox 201 |
| 12:03.21 | pooh_ | zoa: Failed to update. Check the debug log for possible SQL related entries. |
| 12:03.51 | pooh_ | zoa: all parameters are correct |
| 12:04.30 | pooh_ | zoa: debug on the CLI says nothing |
| 12:05.22 | zoa | pgsql or mysql ? |
| 12:05.44 | pooh_ | zoa: mysql |
| 12:05.55 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@host.190.115.68.195.rev.coltfrance.com) |
| 12:05.58 | pooh_ | zoa: sip iax and voicemail are all mysql |
| 12:06.10 | pooh_ | zoa: and working ;-) |
| 12:06.40 | *** join/#asterisk demetrio (n=demetrio@host13-49.pool870.interbusiness.it) |
| 12:06.55 | bon | Connected to asterisk@, port 3306 with username sip for 12 hours, 42 minutes, 9 seconds. |
| 12:06.58 | bon | sorry |
| 12:07.20 | zoa | hmm dunno then |
| 12:07.22 | zoa | sorry |
| 12:07.22 | zoa | :/ |
| 12:07.47 | pooh_ | zoa: np, thx for thinking |
| 12:08.39 | pooh_ | bon, can you update a SIP account from the CLI ? |
| 12:08.48 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 12:08.52 | *** join/#asterisk coppice (n=chatzill@36.192.17.210.dyn.pacific.net.hk) |
| 12:09.22 | *** join/#asterisk oej (n=Olle@host-38.homerun.telia.com) |
| 12:09.57 | pooh_ | oej: time to take a shot at my small problem with RealTime pls? |
| 12:10.34 | zoa | hey olle |
| 12:10.48 | pooh_ | coppice: evening, do you have RealTime experience? |
| 12:11.58 | coppice | Yes. I have extensive realtime experience. I've never used that thing in * with the bogus name, though |
| 12:12.31 | luite | what's currently the best channel type for hfc-s pci cards (I have two of those in my * server), for * 1.2.0 beta2? |
| 12:13.28 | pooh_ | coppice: :-) |
| 12:13.40 | joelsolanki | pooh_: ??? what shud i do ? |
| 12:14.03 | joelsolanki | pooh_: i m getting same error. which i provided :( |
| 12:14.14 | pooh_ | joelsolanki: (12:50:42) pooh_: joelsolanki: pastebin the *complete* call flow |
| 12:14.40 | RoyK | hm... |
| 12:14.51 | joelsolanki | ahh sorry. complete means the logs on asterisk command line? |
| 12:15.07 | RoyK | that qualify= login in chan_sip seems quite stupid |
| 12:15.09 | pooh_ | yes, from start to end, pastebin it |
| 12:16.24 | joelsolanki | pooh_: this came on the screen http://pastebin.ca/28120 |
| 12:17.27 | af_ | it looks like outbound dtmf detection does not work... (reading an email from the ml) |
| 12:17.51 | pooh_ | joelsolanki: http://lists.digium.com/pipermail/asterisk-users/2005-January/080756.html |
| 12:18.09 | joelsolanki | ok i m checking |
| 12:21.02 | demetrio | to get IAX to work when asterisk is behind a NAT, is it enough to forward the iax port? |
| 12:21.53 | pooh_ | demetrio: should not be nessesary |
| 12:22.13 | pooh_ | demetrio: but if so, UDP 4569 |
| 12:22.19 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:22.31 | puzzled | morning all |
| 12:23.43 | demetrio | asterisk doesn't see IAX incoming calls, iax2 debug will say nothing at all but I can see from tcpdump that packets arrive. what have I messed up? |
| 12:25.00 | pooh_ | demetrio: IAX vs IAX2 ? |
| 12:25.16 | joelsolanki | pooh_: silence suppression is not enabled in the linksys. |
| 12:25.21 | pooh_ | puzzled: middag luie donder |
| 12:26.01 | zoa | ela pooh, stinker |
| 12:26.12 | zoa | braaf zijn eh |
| 12:26.20 | pooh_ | zoa: :-) |
| 12:27.24 | zoa | they are probably burning cars there now as we speak |
| 12:27.35 | pooh_ | zoa: so I noticed |
| 12:27.49 | coppice | isn't it a bit late in the year for that kind of thing? |
| 12:28.04 | zoa | its all a preparation for christmas |
| 12:28.06 | demetrio | pooh_: you mean incoming call is ignored because the protocol is different? how sould I enable iax calls? |
| 12:28.08 | zoa | a rite of passage |
| 12:28.12 | zoa | burn a car |
| 12:28.34 | pooh_ | demetrio: make sure you use IAX2 everywhere not IAX |
| 12:28.46 | coppice | this kind of violence normally happens in august or september. same with violence in sports crowds |
| 12:28.46 | h3x | zoa dont you belong in #pike :) |
| 12:29.06 | h3x | oh you are there |
| 12:29.06 | demetrio | pooh_: well I cannot make decisions for the calling pary |
| 12:29.12 | h3x | im just in the wrong pike on caudiumnet |
| 12:29.12 | pooh_ | coppice: yeah, but the 2 deaths just happened |
| 12:29.12 | h3x | heh |
| 12:29.13 | zoa | im in pike |
| 12:29.13 | zoa | you are not |
| 12:29.17 | h3x | hahah |
| 12:29.29 | pooh_ | demetrio: then the other party is using IAX instead of IAX2 ? |
| 12:29.40 | coppice | deaths mean nothing without a little hot weather to make people irritable |
| 12:29.56 | mutilator | ooo |
| 12:29.58 | mutilator | maggots |
| 12:30.04 | zoa | NI NI NI NI NI NI NI NI NI NI |
| 12:30.12 | pooh_ | coppice: dissatisfaction found itself a good enough reason to start riotting |
| 12:30.37 | coppice | rubbish. riots are never about dissatisfaction. |
| 12:30.38 | zoa | nah, those people just need a hobby |
| 12:31.07 | pooh_ | coppice: come over to Paris, Germany and Belgium and ask to bous ;-) |
| 12:31.11 | pooh_ | boys |
| 12:31.49 | pooh_ | and.... gangs making a competition out of it now |
| 12:31.52 | RoyK | is it only me, or does the qualify= in chan_sip seem a little touchy?. even on a LAN i get TOO LAGGED (>500ms) from time to time |
| 12:32.39 | pooh_ | RoyK: if I know the peers whereabouts I always disable qualify due to that reason |
| 12:32.48 | *** join/#asterisk remibreval (i=Remek@pro75-3-82-234-175-208.fbx.proxad.net) |
| 12:33.00 | pooh_ | any M$ guru around ? |
| 12:33.16 | remibreval | Microsoft guru you mean ? |
| 12:33.21 | pooh_ | yup |
| 12:33.37 | remibreval | let's try... ;-) but I'm not so good... |
| 12:33.40 | coppice | M$ guru? you mean someone from marketing? |
| 12:34.02 | remibreval | I'm searching for an asterisk semi-guru :-) simplify my problem http://pastebin.ca/28122 |
| 12:34.13 | pooh_ | ok, I need to catch the username and IP number of the client logging into a domain from within login.bat and send it to a * server |
| 12:34.25 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
| 12:35.09 | remibreval | Ho, I know it is possible, someone has done this before, let me search in my hard drive kb... |
| 12:35.33 | mutilator | damnit |
| 12:35.35 | pooh_ | remibreval: would be nice |
| 12:35.41 | mutilator | i just reinstalled new ver of winamp |
| 12:35.53 | demetrio | pooh_: I don't know, does freeworlddialup use IAX instead of IAX2? I don't think so, but are there any other reasons for this odd behaviour? |
| 12:35.53 | mutilator | and it overwrote my temp playlist that had all my shoutcast stations |
| 12:37.02 | remibreval | arff sorry pooh_, it was using api |
| 12:37.09 | pooh_ | remibreval: ok |
| 12:37.34 | remibreval | maybe you can use a bat command using NET and use a redirection to a file ? |
| 12:37.56 | pooh_ | remibreval: nope, using proxy so must be realtime |
| 12:38.29 | demetrio | ok, I somehow managed to change the problem |
| 12:38.48 | demetrio | now I get " Rejected connect attempt from xxx.xxx.xxx.xxx" |
| 12:39.17 | DYOGI_E | anyone know of any voip phones with the ability to connect a blue tooth headset?? |
| 12:40.02 | remibreval | someone has an idea why I get redirected to my IAX voicemail provider when my phone rings and I answer ?? I'm just doing dial(sip/extention) |
| 12:40.18 | remibreval | and I redirected to IAX voicemail instead of the call is answered |
| 12:43.46 | remibreval | what does this line means exactly ? : == Spawn extension (ext-did, 33870441450, 2) exited non-zero on 'IAX2/rbreval@voipiax/2' |
| 12:43.46 | remibreval | -- Hungup 'IAX2/rbreval@voipiax/2' |
| 12:44.13 | *** join/#asterisk Guggemand (i=Guggeman@80.198.131.42) |
| 12:48.57 | *** join/#asterisk without (n=dean_dav@CPE-60-226-176-110.qld.bigpond.net.au) |
| 12:49.10 | *** join/#asterisk netrunner (n=andreas@anvame.net) |
| 12:49.10 | tzanger | coppice: in asterisk, each 'tap' on the echo canceller covers 8ms, no? |
| 12:49.17 | netrunner | good day. |
| 12:49.49 | RoyK | tzafrir_laptop: yes |
| 12:49.49 | tzanger | I thought 128 taps gave about 16ms of tail |
| 12:49.49 | coppice | 1/8th of a ms |
| 12:49.49 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 12:49.49 | tzanger | 1/8 that was it |
| 12:49.50 | mutilator | what would a person use a fxo on an ata for |
| 12:49.50 | mutilator | ? |
| 12:50.09 | RoyK | mutilator: loop it back to the fxs for fun :) |
| 12:50.23 | mutilator | heh |
| 12:50.33 | netrunner | how can I recognize a module offering a translator path? |
| 12:51.00 | mutilator | well i was thinkin it might be useful for like.. remote location, hook it to a channel bank on a port to do debugging, make calls on that channel bank and stuff |
| 12:51.08 | mutilator | i dunno how you'de place a call on it |
| 12:56.08 | Inkubot | how can i define the flash key in the extensions.conf file ? |
| 12:59.13 | demetrio | I need to get SIP to work behind a NAT and i have a dynamic public IP, so I need to occasionally change the value of externip. A script that edits sip.conf can do the trick, but is there any cleaner way to do this? |
| 12:59.21 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:00.24 | myke420247 | demetrio, i think there's another variable that periodically checks a dns hostname and gives you the ip |
| 13:00.38 | myke420247 | to do exactly what you're trying to do |
| 13:01.48 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 13:02.32 | netrunner | solved (it's the codecs) |
| 13:02.35 | netrunner | thx, bye |
| 13:02.36 | *** part/#asterisk netrunner (n=andreas@anvame.net) |
| 13:07.52 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 13:11.00 | *** join/#asterisk Mother__ (n=m@53.Red-217-126-93.staticIP.rima-tde.net) |
| 13:11.05 | Mother__ | multiple greetings |
| 13:11.19 | Mother__ | any experiences linking VoIP over satellite? |
| 13:11.30 | Mother__ | i.e. how can one cope with the long delays |
| 13:11.39 | mutilator | yes |
| 13:11.46 | mutilator | had a guy over in iraq with an ata |
| 13:11.52 | mutilator | too much latency |
| 13:11.58 | mutilator | couldn't get it to work at all |
| 13:12.04 | mutilator | crackled voice is all it;de do |
| 13:12.14 | wasim | Mother__: get used to saying "over" |
| 13:12.41 | mutilator | it's too bad it didn't work |
| 13:12.52 | mutilator | cause i know that woulda went over awesome |
| 13:13.04 | Mother__ | wasim: LOL indeed |
| 13:13.26 | Mother__ | I'm having problems with UMTS too...too much latency |
| 13:13.42 | mutilator | umts? |
| 13:14.20 | Mother__ | 3G |
| 13:14.57 | mutilator | i see |
| 13:15.06 | Mother__ | I believe it stands for universal mobile telephone service |
| 13:15.24 | mutilator | yeh |
| 13:15.27 | *** join/#asterisk kks (n=kks@202.73.8.130) |
| 13:16.33 | mutilator | only voip & wireless i run is a cpl thousand square mile WAN with 802.11b and 802.11a |
| 13:17.20 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 13:18.38 | joelsolanki | hello all: http://pastebin.ca/28127 i m only able to make 1 call at a time. i have 2 licenses from digium. can any body have idea whats wrong inthis ? |
| 13:20.00 | joelsolanki | i have been trying from morning but still no sucess. if any body could look then its good. |
| 13:21.13 | joelsolanki | any idea ??? |
| 13:21.17 | joelsolanki | :( |
| 13:21.50 | kshumard | joelsolanki, try turning off VAD or silence suppression on the client |
| 13:22.17 | kshumard | this isn't a problem with the number of licenses, it's a problem with the actual configuration of the codec that the client is using |
| 13:23.28 | joelsolanki | kshumard: i m using linksys pap2 and i have disabled silence suppression on both lines :( |
| 13:23.46 | kshumard | are you trying to use g729 on both lines out of the pap2? |
| 13:24.01 | kshumard | it's quite possible that the device only has 1 license, so it won't let you use g729 for 2 calls |
| 13:24.02 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
| 13:24.03 | joelsolanki | yes g729 on both lines of pap2 |
| 13:24.16 | kshumard | you might research a bit and make sure it actually has 2 licenses |
| 13:24.27 | joelsolanki | ahh ..ok let me find out that. |
| 13:24.32 | kshumard | I've heard of 2 line atas that only have 1 g729 license so you can't use g729 on both lines at the same time |
| 13:25.59 | joelsolanki | oh ok let me find out. |
| 13:27.44 | *** join/#asterisk Lathos42 (n=Lathos42@adsl-68-255-78-218.dsl.lgtpmi.ameritech.net) |
| 13:31.02 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 13:34.36 | *** join/#asterisk sapo_original (n=Fenix@200.138.76.58) |
| 13:34.58 | sapo_original | hi.. how i convert .mp3 em .gsm ? |
| 13:35.30 | Delvar | sox ? |
| 13:35.37 | Delvar | mpg123 and sox |
| 13:35.44 | sapo_original | Delvar how? |
| 13:35.51 | Delvar | i duno never odne it |
| 13:35.58 | Delvar | id use mpg123 to convert to wav |
| 13:36.04 | Delvar | then sox to convert to gsm |
| 13:36.09 | Delvar | unles sox supports mp3 |
| 13:36.15 | christo | I have some scripts to do that |
| 13:36.23 | christo | nice ones :) |
| 13:37.30 | christo | I'm using the manager API to originate calls. I can see the manager login and then logout when my script runs, but nothing else happens.. weird |
| 13:38.01 | christo | I wonder if I can get IO::Socket::INET to capture whatever comes back over the socket |
| 13:39.00 | yxa | any uses debian 2.4 kernel and compiled zaptel-source? |
| 13:40.11 | Delvar | christo: use telnet and pretend u are the szcript |
| 13:40.55 | christo | Delvar - aye. good idea |
| 13:41.18 | christo | however it would be good to be able to script that so that I can build error checking into the dial process. |
| 13:41.32 | *** join/#asterisk file[laptop] (n=jcolp@Toronto-HSE-ppp3742612.sympatico.ca) |
| 13:41.32 | yxa | i can modprobe zaptel but not wcfxo and wcfxs |
| 13:42.57 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 13:51.41 | *** join/#asterisk gvag11 (n=g@ipa75.7.tellas.gr) |
| 13:51.44 | gvag11 | hi all |
| 13:52.05 | *** part/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 13:52.17 | gvag11 | can somebody help with asterisk 1.0.9 with TE210 and spandsp ????? |
| 13:53.59 | corruptor | hi all. plz help me. I installed * on Fedora Core 4 and can't get it to start at boot time as non-root. I use script for rh9 from contrib dir. Made some changes, set all permissions and added it to init.d. I can even run it by typing /etc/init.d/asterisk start from my asterisk account and it works. It also works if I type 'service asterisk start' as root. Then i see the proc called asterisk for user asterisk. That's ok but at boot time safe_asterisk doe |
| 13:54.19 | PoWeRKiLL | coppice are you there ? |
| 13:55.33 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 13:56.25 | eldu | hello |
| 13:56.31 | Katty | hi lads. |
| 13:56.46 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 13:56.47 | eldu | is there some bad issues with queues and customs moh ? |
| 13:57.06 | eldu | i never heard moh when i set custom moh in my queues |
| 13:57.17 | eldu | even using default moh files |
| 13:59.24 | gvag11 | can somebody help with asterisk 1.0.9 with TE210 and spandsp ????? |
| 14:06.26 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 14:06.35 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:07.43 | *** part/#asterisk RoyK[Laptop] (n=roy@55.80-202-161.nextgentel.com) |
| 14:07.45 | *** join/#asterisk RoyK[Laptop] (n=roy@55.80-202-161.nextgentel.com) |
| 14:10.11 | coppice | PoWeRKiLL: no |
| 14:10.15 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 14:12.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:13.33 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 14:16.45 | *** join/#asterisk enemy^x (i=lkqw@212.62.250.98) |
| 14:17.07 | enemy^x | could anyone help me in how to debug PRI Got reject for frame 2 ..... ? |
| 14:20.26 | morale | Anyone know who gives good deals on PRI trunks in Western Canada? I am looking for either 256/512Kbps (about 6 or 8 64Kbit channels) |
| 14:20.38 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:25.40 | inspired | is it possible to use music on hold without using mp3 files? wav or gsm or similar instead? |
| 14:28.24 | *** join/#asterisk seong (n=seong@60.49.68.160) |
| 14:29.23 | RoyK | inspired: ja |
| 14:34.38 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:34.44 | Ariel_ | hello everyone |
| 14:35.11 | *** join/#asterisk E|nyPRI_ (n=les@206-45-135-175.static.mts.net) |
| 14:36.52 | remibreval | how can I manage to call a standard sip number like 12345@domain.com ?? |
| 14:37.01 | *** part/#asterisk E|nyPRI_ (n=les@206-45-135-175.static.mts.net) |
| 14:37.25 | RoyK | remibreval: show application dial, and it shows where you can give the url |
| 14:37.25 | RoyK | iirc |
| 14:37.50 | ManxPower | ~docs |
| 14:37.51 | jbot | 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 |
| 14:38.04 | *** join/#asterisk ful|work (n=fulgas@209.8.233.111) |
| 14:38.07 | ManxPower | morale, PRIs are not sold by bandwidth, they are sold based on channels. |
| 14:38.16 | remibreval | RoyK, you mean, I just have to add a rule, and the domain can be resolved ? |
| 14:38.28 | remibreval | It seems that X-Lite do not accept char |
| 14:38.35 | remibreval | like @ and letters... |
| 14:38.37 | RoyK | no idea then |
| 14:38.40 | yxa | i'm getting lotsa of zaptel.c:6161: error: dereferencing pointer to incomplete type when i try to compile zaptel drivers on debian. am i missing some dependencies? |
| 14:38.56 | ManxPower | yxa, sounds like it. |
| 14:39.13 | ManxPower | yxa, make sure linux kernel and kernel-headers are installed. |
| 14:40.24 | yxa | ManxPower i already installed kernel-source. isnt kernel-headers a subset? |
| 14:40.34 | ManxPower | yxa, depends on the distro. |
| 14:40.58 | ManxPower | yxa, in /usr/src/linux can you do a "make menuconfig" and then exit the menuconfig program? |
| 14:41.04 | remibreval | ok, sorry, it was a problem of Domain (explanation are there if someone need it) http://www.astmasters.net/howtos.html |
| 14:41.07 | yxa | manx did it |
| 14:42.13 | ManxPower | yxa, Hmm? |
| 14:42.26 | yxa | ManxPower holyshit, kernel-headers did it |
| 14:42.43 | ManxPower | yxa, 8-) I'm frequently right 8-) |
| 14:43.10 | wasim | 'cept when he's left |
| 14:43.30 | ManxPower | 'morning wasim |
| 14:43.37 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 14:43.37 | yxa | ManxPower is it ok if i use the stock asterisk packages from debian while i compile zaptel drivers from cvs? |
| 14:43.45 | ManxPower | <-- owns a car now, no longer a 2nd class citizen in the USA |
| 14:43.53 | ManxPower | yxa, no. |
| 14:44.15 | ManxPower | yxa, Asterisk and Zaptel are the ONLY things I always build from source. |
| 14:44.38 | ManxPower | Asterisk is not a mature project yet and it changes often. |
| 14:44.58 | file[laptop] | fear my blueberry muffin |
| 14:46.10 | coppice | ManxPower: I typically build the stuff I am developing from source |
| 14:46.25 | ManxPower | coppice, I'm not a developer 8-) |
| 14:47.17 | yxa | ManxPower modprobes zaptel went fine. but wctdm errors |
| 14:47.51 | ManxPower | yxa, what error? |
| 14:48.09 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 14:48.09 | ManxPower | yxa, loading zaptel never has an error. |
| 14:48.16 | yxa | <PROTECTED> |
| 14:48.16 | yxa | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
| 14:48.30 | ManxPower | yxa, what Digium card do you have? |
| 14:49.12 | yxa | tdm400p |
| 14:49.18 | remibreval | Arff, I'm confusing about callig SIP URI. In article they say I should add all the domain I want to call in dns ??!!!! Strange, it should resolve it, no ? |
| 14:49.29 | remibreval | I thought it was just o problem of dialplan... |
| 14:49.38 | remibreval | I find nothing in wiki |
| 14:49.53 | remibreval | It should works without any trunks isn't it ? |
| 14:51.15 | ManxPower | yxa, You remembered to plug the power into the card? |
| 14:51.33 | ManxPower | remibreval, Why do you want to call by SIP uri? |
| 14:51.34 | yxa | ManxPower yeah pretty damn sure. but let me check again |
| 14:52.00 | ManxPower | yxa, On some motherboards you have to try the card in a different slot for it to work. |
| 14:52.13 | luite | are the misdn channels usable with asterisk 1.2.0 rc1? I vaguely remember reading something about it being not ready yet, but I'm not sure if that's correct, or what alternatives I have |
| 14:52.49 | luite | (for use with hfc-s pci) |
| 14:54.16 | remibreval | ManxPower, I know people around the world, they have sip uri number only (FWD, Sipphone, etc...). Here I have Asterisk and pstn services, I can call them on their regular phone but I have to pay... so I wonder know how to call their sip uri direclyt !! |
| 14:54.23 | *** join/#asterisk Phantom-X (n=there@212.31.186.47) |
| 14:54.41 | ManxPower | remibreval, they have to have setup their DNS to support that. |
| 14:54.58 | remibreval | ManxPower, and don't you think in some years, pstn will no longer exists and everybody will use sipuri or something like that ???!... |
| 14:55.21 | remibreval | ManxPower, so if I want to call FWD number I have to register FWS trunk |
| 14:55.23 | remibreval | ? |
| 14:55.24 | ManxPower | remibreval, No I don't think that |
| 14:56.21 | remibreval | ManxPower, what the point of let the pstn telephony alive ? I mean, if everybody use VoIP, it will no longer be usefull, isn't it ? |
| 14:56.39 | ManxPower | remibreval, VoIP is not reliable. |
| 14:56.57 | ManxPower | Specifically Voice over INTERNET is not reliable. |
| 14:57.02 | file[laptop] | anything that traverses the public internet is not reliable |
| 14:57.08 | file[laptop] | it can't be, too many variables |
| 14:57.11 | Phantom-X | so you know if there is a prob for asterist to set the TOS bit in user mode for gentoo also ? |
| 14:57.17 | mog_home | voice over ip is reliable, voice over public internet is not |
| 14:57.28 | ManxPower | mog_home, that's what I meant |
| 14:57.32 | mog_home | yeah i know |
| 14:57.35 | remibreval | ManxPower, you really think this will never be reliable other public IP ? Even in future ? |
| 14:57.57 | mog_home | maybe, as pipes get fatter and latency get lower |
| 14:58.04 | mog_home | jitterbuffers get better |
| 14:58.10 | mog_home | but the internet is not as stable as you think |
| 14:58.14 | mog_home | i mean just the other day |
| 14:58.21 | remibreval | ok I see |
| 14:58.27 | mog_home | my service provider blocked all packets from europe |
| 14:58.33 | mog_home | no one in europe could go to my site |
| 14:58.40 | *** part/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com) |
| 14:58.42 | mog_home | they all got lost on some hop after newyork |
| 14:58.46 | Phantom-X | what do you mean wih voice over internet isn't reliable ? reliable as nonsniffable or reliable as in not always able to communicate ? |
| 14:58.48 | remibreval | ARff !! I'm from france !! |
| 14:58.49 | mog_home | and there was squat i could do about it |
| 14:58.53 | remibreval | SO I can't contact you ! |
| 14:58.55 | mog_home | for 2 days it was gone |
| 14:58.59 | mog_home | you can now |
| 14:59.06 | mog_home | but 2 days a biz is too much |
| 14:59.13 | mog_home | thankfully im not a biz |
| 14:59.15 | MikeJ[Laptop] | mog_home, go to mog_work |
| 14:59.17 | MikeJ[Laptop] | :P |
| 14:59.19 | mog_home | lol |
| 14:59.28 | mog_home | i go to work at 3 on wed. |
| 14:59.28 | remibreval | lol mog |
| 14:59.39 | mog_home | as i have mog_class comming up soon |
| 14:59.56 | yxa | ManxPower reboot and it worked. cbfm |
| 15:00.36 | *** join/#asterisk epablo (n=epablo@200.75.139.188) |
| 15:00.51 | ManxPower | yxa, that can happen. COLD boot and I'll bet it won't work, warmboot and I bet it will work. |
| 15:01.29 | yxa | ManxPower really? |
| 15:01.35 | yxa | ManxPower any workarounds? |
| 15:01.59 | ManxPower | yxa, move it to a different slot and/or use a different motherboard |
| 15:02.58 | yxa | ManxPower anything to do with IRQs? |
| 15:03.06 | ManxPower | yxa, Usually |
| 15:03.21 | ManxPower | Make SURE the Digium card does not share IRQs with anything |
| 15:03.25 | Phantom-X | yxa is swedish for axe :) |
| 15:04.19 | epablo | Hi guys and gals |
| 15:05.25 | Phantom-X | yxa: you see you are from senegal |
| 15:05.35 | Phantom-X | but the word "yxa" IS swedish |
| 15:06.00 | Phantom-X | sorry singapore |
| 15:06.01 | Phantom-X | :) |
| 15:06.20 | Phantom-X | yxa was inspires from the ericssson AXE system |
| 15:06.55 | yxa | ericsson is also famous for the ERLANG programming language |
| 15:07.00 | Phantom-X | yeah |
| 15:07.14 | Phantom-X | uppsala university helped eriksson with developping such |
| 15:07.25 | Phantom-X | I recall since I was studying there when that happened |
| 15:08.49 | *** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl) |
| 15:08.52 | testmachine | hello |
| 15:09.08 | testmachine | i have the asterisk server up and running |
| 15:09.12 | testmachine | the clients can connect |
| 15:09.17 | testmachine | but now i have only 1 problem |
| 15:09.22 | remibreval | ok I putted FWD trunk and I cand call fws uri :) |
| 15:09.32 | testmachine | my clients cant call too eatch others |
| 15:09.43 | epablo | I've got a problem with IAX: No such context/extension |
| 15:09.57 | testmachine | -t |
| 15:10.21 | epablo | I have friend entries on both servers, and have a context defined on both of them. Any ideas |
| 15:10.36 | remibreval | I still have my incomming problem, but I suspect that it is a problem from IAX provider... I mean, I hope it is !!! (if someone has 2 minutes just to tell me if pb is from config or IAX provider please !!! |
| 15:10.37 | remibreval | http://pastebin.ca/28122 |
| 15:13.46 | Rowter | if I add a dinamic member to queue with Local/ it calls it but does not return back to queue if its busy!. thats normal? |
| 15:14.53 | Rowter | if I add them as SIP/ it does return to queue to try another dynamic member if last does not answer.. |
| 15:15.08 | christo | is it okay to include a dialplan into the [default] context using #include patterns.inc where the patterns.inc includes all the rules for different number types |
| 15:15.10 | KriS83 | CAn Anybody tell me what: "PRI got event: HDLC Abort (6) on Primary D-channel of span 3" means? |
| 15:15.43 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com) |
| 15:16.30 | yxa | should i configure my grandstream to send DTMF in-audio, via RTP or SIP INFO? |
| 15:16.40 | azzie | yxa: rfc2833 |
| 15:17.14 | yxa | azzie thats RTP then. whats the diff? |
| 15:17.43 | azzie | yxa: lots of. rfc2833 is most common |
| 15:18.34 | *** join/#asterisk shog (n=rene@62.8.240.129) |
| 15:18.59 | shog | hello! where can i find documentation for the master.csv file? |
| 15:22.19 | Inkubot | if i compile rxfax and txfax in beta2 and then copy to the modulos for rc1... should works ? |
| 15:23.04 | bweschke | inkubot: it may work, but that's not advisable |
| 15:23.12 | epablo | http://www.voip-info.org/tiki-index.php?page=Asterisk |
| 15:23.28 | epablo | shog: There is the main source for asterisk info |
| 15:23.55 | bweschke | Rowter: unfotunately, yes. The Local channel has no good way to currently pass back the DIALSTATUS it received from the channel that proxied the call for it |
| 15:26.40 | CoffeeIV_ | I have been trying for a while to make my Asterisk server that is behind an iptables gateway available to the internet, and I can't get the right iptables rules. I can't find examples on google or voip-info. Can anyone tell me why this doesn't work: http://pastebin.ca/28139 |
| 15:26.42 | Rowter | bweschke, ahh.. and Agents retuns to queue but does not work that well, it sometimes has deadlocks, I thought you could return to queue after calling a Local.. |
| 15:27.28 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 15:27.40 | *** join/#asterisk mog_class (n=mogorman@user-24-236-84-48.knology.net) |
| 15:27.46 | bweschke | Rowter: you can, if Local returns a busy status. but it doesn't, because it presently has no real good way to know whether or not the channel that made the call for Local was actually succesful in reaching the member |
| 15:28.24 | epablo | I'm calling intra-servers with IAX. I have friend defined on both servers each with a context. But I'm still getting a No such context/extension error. Any ideas? |
| 15:28.33 | Rowter | bweschke, ${DIALSTATUS} ? |
| 15:28.38 | bweschke | Rowter: I've got several systems in production with clients using Agents and queues that haven't been deadlocking. What version are you having a problem with? |
| 15:29.38 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 15:29.39 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 15:29.45 | mog_class | it happens with older code bweschke, when digium rolled into agents we had problems for a bit and then code got fixed |
| 15:29.48 | bweschke | Rowter: correct. the DIALSTATUS returned by a call to SIP/device directly won't be the same as the DIALSTATUS returned by a call to Local that makes that same call via the DP |
| 15:29.49 | Rowter | bweschke, I downgrade to 1.09, and am suffering of channel blocking, I was using beta 1 with deadlocks.. and I think beta 2 also has them. |
| 15:30.05 | mog_class | beta 2 def. should not |
| 15:30.38 | Rowter | http://bugs.digium.com/view.php?id=4244 |
| 15:31.00 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 15:31.41 | Phantom-X | hmm I did get a segmentation fault when trying to run my newly compiled asterisk... |
| 15:32.17 | mog_class | whats the error phantom |
| 15:33.39 | Phantom-X | Segmentation fault (core dumped) |
| 15:33.39 | Phantom-X | galaxian asterisk # Ouch ... error while writing audio data: : Broken pipe |
| 15:33.59 | mog_class | can you run with -vvvv or in gdb |
| 15:34.03 | mog_class | so we can get more info |
| 15:34.11 | mog_class | what module is loading when it crashes |
| 15:34.21 | Phantom-X | can Iforce asterisk using alsa swmizer ? |
| 15:34.25 | Phantom-X | can Iforce asterisk using alsa swmixr ? |
| 15:34.45 | oogle | i can't wait for Asterisk 1.2 to drop |
| 15:34.48 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:34.49 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 15:34.49 | Phantom-X | /usr/sbin/asterisk -vvvvgc |
| 15:35.22 | Phantom-X | <PROTECTED> |
| 15:35.22 | Phantom-X | Segmentation fault (core dumped) |
| 15:35.22 | Phantom-X | galaxian asterisk # Ouch ... error while writing audio data: : Broken pipe |
| 15:35.37 | Phantom-X | I suspect it doent like that other processes uses the coundcard |
| 15:35.39 | mog_class | run in gdb |
| 15:35.45 | mog_class | nah |
| 15:35.51 | mog_class | it should handle that more cleanly |
| 15:35.57 | Phantom-X | hmm |
| 15:36.00 | mog_class | oogle 1.2 rc 1 dropped go play |
| 15:36.06 | mog_class | type gdb asterisk |
| 15:36.08 | mog_class | then run -c |
| 15:36.16 | mog_class | it should print out more info as to why it died |
| 15:36.43 | shog | epablo: i looked through the site, but couldn't find anything that explains the messages in master.csv |
| 15:37.03 | CoffeeIV_ | Is anyone here making an asterisk server availble to SIP phones on the internet through an iptables NAT ? Is there any chance you could pastebin the rules you used ? or look at my non-working one: http://pastebin.ca/28139 |
| 15:38.10 | without | sip uses UDP not TCP :) |
| 15:38.16 | mog_class | shog http://www.voip-info.org/wiki/view/Asterisk+cdr+csv |
| 15:38.18 | Phantom-X | ok I got info but I am nmot sure what it refers to |
| 15:38.24 | mog_class | that is explanation |
| 15:38.30 | mog_class | can you pastebin it phantom |
| 15:38.34 | *** join/#asterisk dos000 (n=dos000@CPE0006256d190c-CM0011aeff5db6.cpe.net.cable.rogers.com) |
| 15:38.36 | CoffeeIV_ | without: thanks, let me try that change |
| 15:39.00 | bweschke | Phatrom-X: rm /usr/lib/asterisk/modules* |
| 15:39.07 | bweschke | Phantom-X: then re-do make install |
| 15:39.09 | shog | mog_class: thank you |
| 15:39.17 | bweschke | Phantom-X: then try and reload |
| 15:39.22 | CoffeeIV_ | does sip use tcp AND udp, or just udp ? |
| 15:39.26 | Phantom-X | Program received signal SIGSEGV, Segmentation fault. |
| 15:39.26 | Phantom-X | [Switching to Thread 16384 (LWP 17383)] |
| 15:39.26 | Phantom-X | 0xb692f808 in noise_codebook_unquant () from /usr/lib/libspeex.so.1 |
| 15:39.33 | RoyK | hm |
| 15:39.33 | Phantom-X | bweschke: I run gentoo |
| 15:39.40 | epablo | CoffeeIV_: UDP |
| 15:39.43 | RoyK | how can i have one asterisk server do a sip native transfer to another? |
| 15:39.46 | Nugget | Linux is poo. |
| 15:39.57 | bweschke | royk: you can't, yet |
| 15:40.00 | mog_class | yeah phantom bweschke is right |
| 15:40.09 | *** part/#asterisk santiago (n=santiago@208.195.215.124) |
| 15:40.11 | RoyK | bweschke: hm. no? |
| 15:40.13 | without | you dont need that many RTP ports open |
| 15:40.15 | mog_class | canreinvite will try to do a direct reinvite |
| 15:40.16 | bweschke | royk: it's in the bug tracker I think to be fixed |
| 15:40.16 | RoyK | bweschke: i thought that was possible |
| 15:40.23 | RoyK | bweschke: what bug? |
| 15:40.26 | mog_class | if its possible |
| 15:40.30 | bweschke | royk: it works under certain circumstances, but not all |
| 15:40.37 | Phantom-X | hnmm |
| 15:40.40 | RoyK | bweschke: what bug? |
| 15:40.46 | bweschke | one sec pls |
| 15:40.47 | mog_class | exactly |
| 15:40.50 | bweschke | will find it for you |
| 15:41.13 | Phantom-X | ok ill try that |
| 15:41.26 | epablo | Anyone know a way of debuging iax so you can see the exten/context it is trying to dial when receving a call. I did iax debug with set verbose 99 and can't see it |
| 15:41.51 | mog_class | in the initial section of the call epablo it will come in |
| 15:41.55 | mog_class | but you need iax debug |
| 15:42.00 | mog_class | not verbosity |
| 15:42.15 | mog_class | but if you dont set anything anywhere it will just go to default |
| 15:42.54 | epablo | mog_class: I have a context set for that friend. it should go there right? |
| 15:43.06 | mog_class | it will default there |
| 15:43.16 | mog_class | but the other end can request a diff context |
| 15:43.16 | bweschke | royk: 3710 |
| 15:43.18 | mog_class | i believe |
| 15:44.23 | epablo | This is what I got: http://pastebin.ca/28144 |
| 15:44.28 | RoyK | bweschke: but, er, that's like 8 months old.... |
| 15:44.34 | epablo | And the incomming context is defined.. |
| 15:44.57 | yxa | how do i edit dialplan so that when I dial 9, it will give me and external line on Zap/1 |
| 15:45.18 | Phantom-X | I get same core dump still even if I recompiled it |
| 15:45.25 | Phantom-X | I think speex perhaps is the prob |
| 15:45.30 | mog_class | context is incomming |
| 15:45.43 | mog_class | yeah phantom, thought it might have been incompatible |
| 15:45.52 | bweschke | royk: correct. but oej has a patch .. he hasn't posted it yet, but he says he's working on it from a NOTE made in it yesterday |
| 15:45.57 | mog_class | does incomming exist epablo |
| 15:46.14 | Phantom-X | I though partage should deal with this automagically |
| 15:46.17 | epablo | [ Context 'incomming' created by 'pbx_config' ] |
| 15:46.17 | epablo | <PROTECTED> |
| 15:46.17 | epablo | <PROTECTED> |
| 15:46.21 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 15:46.24 | file[laptop] | yuck, class! |
| 15:46.26 | epablo | sorry about that |
| 15:46.32 | *** join/#asterisk _T3_ (n=rposada@200.63.231.210) |
| 15:46.35 | mog_class | epablo that would not work |
| 15:47.00 | bweschke | Phantom-X: do you use speex in your setup ? if not, just noload the mod |
| 15:47.08 | epablo | mog_class: why? How do I make it work? |
| 15:47.33 | Phantom-X | bweschke: the point was to use speex to guide the callers |
| 15:47.35 | mog_class | you need [incomming] |
| 15:47.45 | mog_class | non of that other garbage in the tag |
| 15:47.51 | mog_class | what is pbx_config |
| 15:48.05 | epablo | That was what came of a show dialplan incomming |
| 15:48.28 | mog_class | ohhh |
| 15:48.35 | epablo | [incomming] |
| 15:48.35 | epablo | include => DID |
| 15:48.35 | epablo | exten => 551XXXXXXX,1,Goto(TellFree,${EXTEN},1) |
| 15:48.37 | mog_class | i thought that was from flat file |
| 15:48.39 | mog_class | my bad |
| 15:48.40 | tdonahue | hi all |
| 15:48.43 | mog_class | i never look at said info |
| 15:48.53 | epablo | This is the flat file version |
| 15:49.16 | RoyK | the US calls Iran extreme: http://msnbc.msn.com/id/9967813/ |
| 15:49.19 | tdonahue | i'm trying to use spandsp with a te110p card that has a PRI attached to it. so I need to have "faxdetect" enabled in my zapata.conf? |
| 15:49.28 | mog_class | the numbers dont match epablo |
| 15:49.30 | Phantom-X | does speex need something special to work ? |
| 15:49.32 | bweschke | phantom-x: sorry. not sure what you mean "guide the callers" ... do you need to use the speex codec in your setup? is that what you're saying? |
| 15:49.32 | pooh_ | hmm, mysql cmd can not put something INTO mysql...?? |
| 15:49.45 | yxa | how do i edit dialplan so that when I dial 9xxxxxxxx, it will give me and external line on Zap/1 and dial that xxxxxxxxx |
| 15:49.51 | *** join/#asterisk kiwnix (n=egarcia@82.158.153.4) |
| 15:49.51 | epablo | mog_class:??? where? |
| 15:49.58 | mog_class | nevermind |
| 15:50.01 | mog_class | i cant count |
| 15:50.10 | Phantom-X | I nneed/want speex to be able to so syhtesis feediack to poeple calling when they are on hold etc |
| 15:50.17 | pooh_ | bweschke: any RealTime experience ? |
| 15:50.22 | Phantom-X | I thought speex is used for that |
| 15:50.33 | bweschke | pooh_: no sir. not yet |
| 15:50.49 | dos000 | is there a simple tutorial that shows how to get * routing calls to sip termination ? |
| 15:50.51 | pooh_ | bweschke: ok |
| 15:51.07 | *** join/#asterisk fugitivo (n=ajf@209.13.246.124) |
| 15:51.26 | tdonahue | RoyK: there are different types of extremes... the whole debate in kansas is extreme stupidity |
| 15:51.40 | epablo | mog_class: any other ideas? |
| 15:51.50 | mog_class | id need to look at it more closely |
| 15:51.52 | mog_class | epablo |
| 15:52.05 | mog_class | and right now im mog_class when i become mog_work id be happy to ^_^ |
| 15:52.18 | RoyK | tdonahue: it still compares to the religious extremity you find in other parts of the world, only "it's ok as long as people beleive in jesus" |
| 15:52.40 | bweschke | phantom-x: speex is a codec. you must have clients or peers that use speex to interact with them and use that codec. if you're looking for a text to speech client, you'll want to look at festival integration |
| 15:52.46 | epablo | mog_class: LOL..Ok |
| 15:53.13 | Phantom-X | okay |
| 15:53.23 | Phantom-X | I am familiar with festival tts :) |
| 15:53.30 | CoffeeIV_ | I (finally) got my iptables NAT'ing to allow access to my asterisk server thanks to help from here -- my rules in case someone wants them for future reference: http://pastebin.ca/28145 Thanks! |
| 15:53.33 | Phantom-X | so I can remove speex thebn |
| 15:53.37 | bweschke | correct |
| 15:54.15 | bweschke | asterisk probably built the codec because it saw you had some include files in the "right place" on your system and it assumed you had compatible libraries too |
| 15:54.27 | bweschke | but that was probably a bad assumption in this case |
| 15:54.36 | mog_class | it checks for headers i thought |
| 15:54.47 | mog_class | its odd youd have the headers and not libspeex.so |
| 15:54.49 | tdonahue | RoyK: all I will say about it is if they want to be teaching people about a religious belief it should be in a theology class, not in a science class where people should be learning about scientific theories... but what do I know, i'm not a republican from kansas |
| 15:54.49 | mog_class | or whatever |
| 15:55.00 | Phantom-X | well I externally included speex |
| 15:55.05 | bweschke | yep |
| 15:55.17 | Phantom-X | it was me assuming compiiling speex and asterisk would make it work |
| 15:55.21 | bweschke | speex.h in /usr/include or /usr/local/include |
| 15:55.24 | Phantom-X | which it didnt |
| 15:55.27 | bweschke | if it's there, it'll attempt to build it |
| 15:55.38 | Phantom-X | hmm |
| 15:56.01 | Phantom-X | that would be a mistake in the ebuilds then for the gentoo |
| 15:56.34 | bweschke | so - rm -f /usr/lib/asterisk/modules/*_speex.so and you should be good to go |
| 15:56.35 | *** join/#asterisk Weezey (i=Weezey@vaio2.tsnetworks.ca) |
| 15:56.51 | Phantom-X | I am about to set up a sip proxy with voicemailboxes etc |
| 15:56.52 | RoyK | tdonahue: agreed |
| 15:56.55 | jpmcallister | Hello. I need help with cdr records. |
| 15:56.55 | Phantom-X | ok |
| 15:57.13 | tdonahue | jpmcallister: what exactly are you trying to do? |
| 15:57.30 | bweschke | phantomx: a sip server you mean? |
| 15:57.38 | jpmcallister | I'm making calls using sipphone. The problem is that to make a call I need to use the SetCallerID command |
| 15:57.59 | Phantom-X | I have an cisco ata 186/188 which I want to boost up with help of asterisk :) |
| 15:58.00 | jpmcallister | to the username of the sipphone |
| 15:58.06 | *** join/#asterisk kiwnix (n=egarcia@82.158.153.4) |
| 15:58.40 | jpmcallister | then the cdr records src, and clid is registered as the username of the sippohone istead of the actual extension making the call |
| 15:58.47 | Phantom-X | ok not things works fine .. thank you for your assistence about speex prob :) |
| 15:59.12 | Weezey | jpmcallister: you should be using the src not the clid for call routing |
| 15:59.17 | Weezey | the clid means shit. |
| 15:59.27 | pattieja | wow |
| 15:59.42 | pattieja | FWD's IAX2 server's been unavailable for some time now |
| 15:59.42 | Phantom-X | do I have to set TOS bits with iptables for sip when running as non root user or does it still work anyways ? |
| 16:00.03 | jpmcallister | Weezey: booth src and clid are set to the username of the siphone |
| 16:00.21 | Weezey | isn't that who is making the call? |
| 16:00.43 | jpmcallister | I'm making the call troung a sip extension |
| 16:01.04 | jpmcallister | if I use a iax extension the record is registered fine |
| 16:01.42 | Phantom-X | Nov 9 16:58:25 WARNING[2154]: chan_oss.c:257 sound_thread: Read error on sound device: Resource temporarily unavailable |
| 16:01.50 | Phantom-X | how do I diable oss stuff ? |
| 16:01.51 | jpmcallister | sipphone is a peer defined which is my voip provider |
| 16:01.57 | mutilator | anyone know if there is a way to check voicemail of a verizon cellphone from a landline/other phone? |
| 16:01.57 | Phantom-X | o run alsa not oss |
| 16:02.04 | bweschke | phantomx: it should still work |
| 16:02.08 | Phantom-X | okay |
| 16:02.24 | bweschke | rm /usr/lib/asterisk/modules/chan_oss.so |
| 16:02.33 | bweschke | and reload * |
| 16:02.33 | Phantom-X | thank you |
| 16:04.22 | Phantom-X | Nov 9 17:03:51 WARNING[2171]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled <--- ?? what's this ? |
| 16:05.33 | *** join/#asterisk adam225 (n=adam@rephlexon.rit.edu) |
| 16:05.37 | Phantom-X | I think ill read the fine manual more . its too many config files there that I don't use .. I am about to use SIP only |
| 16:05.58 | adam225 | hey i'm have some pri questions, anyone alive that could take a crack at it? |
| 16:06.13 | tdonahue | mutilator: for verizion wireless you can dial # to get the password prompt |
| 16:06.23 | tdonahue | adam225: sure why not |
| 16:06.31 | adam225 | cool |
| 16:06.53 | adam225 | so i have a T110P connected to a WIC-CSU/DSU in a cisco 2600 |
| 16:06.56 | adam225 | here in a lab |
| 16:07.01 | adam225 | doesn't get more basic |
| 16:07.07 | tdonahue | ok |
| 16:07.22 | adam225 | have my 2600 doig the timing |
| 16:07.31 | Weezey | how can I dial from the manager |
| 16:07.49 | yxa | when did this exten => _9XXXXXXXX,1,Dial(Zap/4/${EXTEN:1}), it only accepts 8 normal digit numbers. but special 4 digits dont work. pls help |
| 16:07.58 | Weezey | I want the manager to call me, then dial a number and then combine our calls. |
| 16:08.03 | adam225 | etc/zaptel.conf --> span=1,0,0,esf,b8zs |
| 16:08.41 | adam225 | when i show span |
| 16:08.44 | adam225 | in asterisk cli |
| 16:08.52 | adam225 | i get Status: Provisioned, Down, Active |
| 16:09.09 | *** part/#asterisk epablo (n=epablo@200.75.139.188) |
| 16:09.16 | adam225 | not sure why it says Down |
| 16:09.45 | adam225 | and some more FYI, this thing is going to show that we can configure asterisk for voice and data |
| 16:09.56 | adam225 | i'm just trying to get it talking with the T1 gracefully to start off |
| 16:10.10 | tdonahue | adam225, you have pri signalling configured on both sides? |
| 16:10.48 | adam225 | i think so |
| 16:10.51 | adam225 | tell me how to check |
| 16:11.06 | tdonahue | i don't know on the cisco side |
| 16:11.20 | bweschke | phatomx: that's just the old sccp protocol. you can disregard that msg |
| 16:11.28 | adam225 | oh |
| 16:11.36 | adam225 | well the cisco is setup to be master signalling |
| 16:12.00 | adam225 | actually it is setup for internal clock |
| 16:12.13 | adam225 | interface Serial0/2 |
| 16:12.13 | adam225 | <PROTECTED> |
| 16:12.13 | adam225 | <PROTECTED> |
| 16:12.13 | adam225 | <PROTECTED> |
| 16:12.14 | adam225 | <PROTECTED> |
| 16:12.37 | tdonahue | are you trying to do an "Integrated T-1" type of setup where you have voice and data over the same T-1? |
| 16:12.39 | *** join/#asterisk shanky (n=shanky@82.159.214.9) |
| 16:12.44 | shanky | hi, good afternoon |
| 16:13.00 | mog_class | morning |
| 16:13.01 | adam225 | tdon: correct |
| 16:13.30 | *** join/#asterisk SuidBit (n=LinuxSec@darwin.fundanet.br) |
| 16:14.06 | tdonahue | adam225: is this a t-1 coming from your telco, or is it a point to point T-1 you are trying to multiplex the voice and data over? |
| 16:14.19 | adam225 | yea point to point |
| 16:14.21 | pooh_ | hmmm, how do I get a CLI command line in my dialplan? |
| 16:14.28 | adam225 | i just have a T1 in a router here in the lab |
| 16:14.35 | adam225 | connected to the asterisk box |
| 16:14.54 | adam225 | it's litterally a router connected to a PC via T1 line |
| 16:15.06 | Math` | can a TDM card be used to pass data in addition of phone traffic via T-carriers? |
| 16:15.06 | bweschke | pooh_: what do you mean? what are you trying to do? |
| 16:15.23 | bweschke | Math`: though I haven't done it, yes. to my knowledge, it can |
| 16:15.32 | *** join/#asterisk pcm (n=pcm@user-69-73-0-22.knology.net) |
| 16:15.34 | Math` | ok |
| 16:15.41 | pooh_ | bweschke: this is the command on the CLI: realtime update sippeers name 9011 mailbox 201 |
| 16:15.58 | tdonahue | adam225: please excuse my ignorance, but what are you trying to make happen? |
| 16:16.27 | pooh_ | bweschke: how can I let my dialplan do this? |
| 16:16.36 | adam225 | tdon: that's fine, i'm trying to show that "if we were to have a T1", we could use asterisk for voice and data |
| 16:16.39 | bweschke | pooh_: lemme check the src a bit... will get back to you |
| 16:16.45 | adam225 | so i'm in a lab |
| 16:16.58 | adam225 | and i have a router with a T1 card in it, which in "real world" would be the provider |
| 16:17.24 | adam225 | and i have my asterisk system connected with a T110P to this router (think provider) |
| 16:17.45 | adam225 | totally isolated situation now |
| 16:18.21 | bweschke | pooh_: you could probably use an ODBC function to issue the specific SQL query to make the update you're requesting if your realtime source is odbc compatible |
| 16:18.29 | pooh_ | bweschke: thx |
| 16:18.42 | adam225 | does that make it clearer tdonahue:? |
| 16:18.59 | pooh_ | bweschke: I am using Mysql |
| 16:19.42 | *** join/#asterisk oej (n=Olle@ti111210a080-7686.bb.online.no) |
| 16:20.03 | bweschke | oej: I'm going to fire this one over to you... :) |
| 16:20.43 | bweschke | oej: pooh_ is asking how he can get the equivalent or "realtime update sippeers name 9011 mailbox 201" that he can issue on the CLI to happen via a predefined extension in the DP |
| 16:20.45 | bweschke | ? |
| 16:20.45 | *** join/#asterisk adtech (n=asdf@199.185.139.89) |
| 16:21.22 | tdonahue | adam225: so you are trying to simulate a "Integrated T-1" coming from MCI (or sbc, or verizion...) with the data routing and the voice going to asterisk? |
| 16:21.23 | bweschke | i don't know of any way to do it aside of an ODBC function to issue the SQL query to make the update he's requesting |
| 16:21.25 | oej | There must be some way to do that via a realtime app or a system call |
| 16:21.34 | oej | bweschke: It's not really a SIP issue :-) |
| 16:21.37 | adam225 | tdonahue: bingo |
| 16:21.45 | bweschke | oej: yes, I know. you're right |
| 16:21.52 | christo | okay, I'm totally stumped by this : No translator path exists for channel type IAX2 (native 63500) to 0 |
| 16:21.56 | christo | anybody recognise that? |
| 16:22.13 | christo | the end results is "Unable to create channel of type 'IAX2'" |
| 16:22.21 | pooh_ | oej: realtime app only does queries, no saves |
| 16:22.54 | Corydon-w | bweschke: can we make StripLSD and Substring go away before the release of 1.2? |
| 16:23.04 | oej | pooh_: Then write a perl script and call it via the system() call |
| 16:23.05 | Corydon-w | StripMSD, rather |
| 16:23.50 | bweschke | corydon: no idea. it certainly is better suited for a DP function instead of an app, no? |
| 16:23.57 | pooh_ | oej: yeah I was afraid of that, too bad there isn't a CLI command like the system command for system calls |
| 16:23.58 | mog_class | corydon they should be gone |
| 16:23.59 | oej | Corydon-w: Do you know a good example of using the ignorepattern |
| 16:24.09 | mog_class | but they will be kept for back compatibility |
| 16:24.15 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 16:24.15 | mog_class | i think they will be gone in 1.3 |
| 16:24.15 | oej | pooh_: Have you tried ! |
| 16:24.21 | mog_class | and they should be gone for 1.4 |
| 16:24.30 | mog_class | or at least thats my understanding |
| 16:24.35 | pooh_ | oej: tried what? |
| 16:24.37 | oej | mog_class: Are they marked as deprecated? |
| 16:24.47 | mog_class | yeah |
| 16:24.48 | oej | Tried ! exclamation mark from CLI |
| 16:24.56 | tdonahue | adam225: the cisco router, AFAIK, can not correctly emulate that environment. in my experience with integrated t-1's we have used an extneral CSU/DSU (like the Adtran Total Access 600R) to split the voice portion off to a separate asterisk server. |
| 16:24.57 | mog_class | they are ubber depricated |
| 16:25.05 | mog_class | like depricated before i started at digium |
| 16:25.27 | mog_class | by the : thing |
| 16:25.35 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 16:25.56 | oej | Anyone that can explain the ignore patterns? Give an example? |
| 16:25.59 | pooh_ | oej: ah ok, but that is executing a shell command from the CLI, not executing a CLI command from the dialplan |
| 16:26.13 | adam225 | tdonahue: hmmm i'm pretty sure this fractional T1 card can get it done. do you have any documentation that you know of that i could take a look at? |
| 16:26.29 | oej | pooh: Sorry, then run system(asterisk -rx "stop now") from the dialplan |
| 16:27.03 | pooh_ | oej: yeah.. AGI script or whatever |
| 16:27.22 | adam225 | tdonahue: would you be able to tell me some steps at least i could take to test out if my configuration is working now? |
| 16:27.57 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 16:28.01 | adam225 | tdonahue: The link is up and i have no warnings on the line but I don't know if communication is happening from the debug messages |
| 16:28.31 | tdonahue | adam225: you can take a look at www.voip-info.org for the asterisk side, but I'm not seeing anything on integrated t-1's |
| 16:29.04 | bweschke | pooh_: ya, actually that would be for sure an indirect way of solving the problem |
| 16:29.20 | tdonahue | in that setup you wouldn't be able to use PRI signalling though |
| 16:29.32 | adam225 | ok |
| 16:29.39 | adam225 | pri would be only for all voice? |
| 16:29.49 | pooh_ | bweschke: yeah, the systemcall will not work for I can not channel variables to the system command |
| 16:30.00 | tdonahue | yes, integrated t-1's work like channel banks |
| 16:30.13 | adam225 | so what would i use then? |
| 16:30.40 | tdonahue | i have no idea what signalling cisco routers support, you would have to find that on their TAC |
| 16:31.11 | tdonahue | and you will need to make sure that your IOS revision supports voice stuff, not all of them do |
| 16:32.25 | *** join/#asterisk XTR (n=xtr@staff-nat.netnation.com) |
| 16:32.51 | adam225 | http://72.14.207.104/custom?q=cache:RTJgBe8fu1gJ:www.voip-info.org/wiki-Asterisk%2Bconfig%2Bzapata.conf+voice+and+data+asterisk&hl=en&ie=UTF-8 |
| 16:33.02 | bweschke | pooh_: why not? |
| 16:33.04 | adam225 | look at "multi-link PPP" options |
| 16:33.12 | adam225 | can i do something like that |
| 16:34.09 | bweschke | pooh_: build a string with your chan variables you need and then pass that string variable into the system call |
| 16:35.07 | pooh_ | bweschke: got, exten => 402,n,System(asterisk -r -x "realtime update sippeers name ${user} mailbox ${EXTEN}" |
| 16:35.13 | pooh_ | bweschke: workd |
| 16:35.16 | pooh_ | works |
| 16:35.29 | tdonahue | adam225: if your telco provides it, but simulating it will be "interesting" |
| 16:36.04 | tdonahue | i have no idea that it would even be possible to simulate on a cisco router |
| 16:39.33 | christo | I keep getting an error which says "No translator path exists for channel type IAX2 (native 63500) to 0". I looked in the modules.conf and there's no load => chan_iax2.so. However my modules directory doesn't exist under /var/lib/asterisk - I can only find it under /usr/src/Asterisk..bla What could be going on here? |
| 16:40.09 | christo | I've tried doing another 'make install' in the source dir.. just in case, but no differences |
| 16:41.13 | dos000 | how does one dial a sip provider that requires a port ? |
| 16:41.16 | shog | what does "congestion" as 8.item in the Master.csv logfile mean? |
| 16:43.22 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 16:43.22 | christo | :) |
| 16:43.26 | *** join/#asterisk manelideo (n=mdsilva@194.117.41.7) |
| 16:43.39 | manelideo | hey! does someone knows how to put my AMP to authenticate? |
| 16:43.51 | manelideo | when someone opens the browser, must put a username and password? |
| 16:44.19 | Katty | probably you |
| 16:44.38 | manelideo | yes lol I know |
| 16:45.11 | manelideo | what I want is a way to prompt for user and pass |
| 16:46.03 | brad_mssw | not that I use AMP, but a .htaccess file not sufficient ? |
| 16:47.15 | manelideo | I thought that file would be enough |
| 16:47.21 | manelideo | and maybe it is |
| 16:47.32 | manelideo | i can past my code |
| 16:47.58 | manelideo | AuthType Basic |
| 16:47.58 | manelideo | AuthName "Asterisk Management Portal" |
| 16:47.58 | manelideo | AuthUserFile /etc/httpd/conf/amppasswd |
| 16:47.58 | manelideo | Require user amp |
| 16:48.09 | manelideo | just this... |
| 16:48.14 | brad_mssw | you need to make sure your webserver has AllowOverride all or similar to be set on that directory, otherwise the .htaccess won't be read |
| 16:48.28 | brad_mssw | that should be in your apache configuration |
| 16:48.40 | manelideo | yes...I´ll go check it out |
| 16:50.54 | *** join/#asterisk robtro (i=rob@unaffiliated/robtro) |
| 16:51.27 | adam225 | tdonahue: have you setup a voice & data asterisk before? |
| 16:51.42 | adam225 | tdonahue: what hardware did you use |
| 16:51.51 | adam225 | tdonahue: where did you get the info to config it? |
| 16:52.23 | *** join/#asterisk file[laptop] (n=jcolp@Toronto-HSE-ppp3742612.sympatico.ca) |
| 16:53.30 | manelideo | brad_mssw ok, I´ve changed it, and it worked! |
| 16:58.52 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:59.17 | *** join/#asterisk S4w (n=asas@adsl-3-67-211.mia.bellsouth.net) |
| 17:00.04 | S4w | hey guys, I dont know if you remember me from yesterday but I got my problem working. If any one of you have a sipura SPA-1001 upgrade the firmware |
| 17:00.34 | S4w | 3.1.8 |
| 17:00.35 | InfraRed | tried the sipura website? |
| 17:00.52 | shanky | I'll be back in a minut |
| 17:00.54 | S4w | I am not asking ;-) I just said that it solved my problem |
| 17:02.39 | S4w | but i do have a question. Why is it that sometimes asterisk cant bridge betwen AVAILABLE codecs? |
| 17:04.19 | Lostfrog | Maybe the bridge was out. |
| 17:04.36 | mutilator | ya, it is hurricane season after all |
| 17:04.39 | S4w | hmm? ^o) |
| 17:05.00 | pooh_ | bweschke: works like a charm, thx, oej too :-) |
| 17:05.07 | Lostfrog | What codecs where involved, S4w? |
| 17:06.30 | tdonahue | adam225: i have done asterisk over an integrated t-1 before. The T-1 comes in from MCI to an Adtran IAD (Total Access 600, IRRC) and the IAD hands off a T-1 that presents itself like a channel bank using E&M Wink to the asterisk server. The data is handled by the router in the IAD or with other models could be handed off to a router as a serial or fractional T-1 connection |
| 17:06.54 | S4w | g726 and gsm another thing is that I am making outbound calls using X provider in contect default. and my ata is located in the ATA context. the entensions to make the outbound calls are defined in both contexts. Sometimes it works and some others it does not |
| 17:07.05 | pattieja | mutilator, did anyone help you with the voicemail? |
| 17:07.36 | Phantom-X | I seem to fail to make asterisk register at my sip proxy provider... |
| 17:07.39 | *** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300757.sympatico.ca) |
| 17:07.57 | asteriskmonkey | can anyone help me debug a sip issue? |
| 17:08.12 | justinu | what kind of sip issue? |
| 17:08.18 | bweschke | pooh_: np. it would seem though that it would be real nice to be able to do that via a DP function - will have to take a look at that |
| 17:08.28 | Lostfrog | When I take a sip, sometimes I spill my drink all over me. |
| 17:08.47 | asteriskmonkey | lol |
| 17:09.06 | Lostfrog | I'm in an antagonistic mood today. |
| 17:09.06 | asteriskmonkey | my problem is in my extensions.conf when i tell it to ring the sip client it never does |
| 17:09.08 | RoyK | asteriskmonkey: pastebin it, and then ask |
| 17:09.10 | Phantom-X | Nov 9 18:07:31 NOTICE[2500]: chan_sip.c:6854 handle_response: Failed to authenticate on REGISTER to '<sip:0906990040@proxy.digisip.net>;tag=as4b6f3b91' |
| 17:09.12 | RoyK | ~pb |
| 17:09.15 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 17:09.17 | Phantom-X | I am not sure what goes wrong |
| 17:09.36 | RoyK | Phantom-X: pastebin the whole debug |
| 17:09.47 | justinu | you failed to authenticate on register :P |
| 17:09.56 | oogle | announcing the Asterisk Realtime Voice Pitch Changer! http://www.lobstertech.com/voicechanger/ |
| 17:10.10 | Phantom-X | the think is that I don't know the diff what is the user and userID |
| 17:10.10 | oogle | please help out with a few things mentioned on the page if you can |
| 17:10.24 | *** join/#asterisk Spudz0rz (n=matthew@CPE-144-136-203-188.sa.bigpond.net.au) |
| 17:10.27 | Lostfrog | Please note that changing the pitch of your voice will not foil the feds. |
| 17:10.30 | asteriskmonkey | justinu: if i failed to authenticate how come i can dial out from the phone? |
| 17:10.37 | *** join/#asterisk shanky (n=shanky@82.159.214.9) |
| 17:10.42 | shanky | I'm here again |
| 17:10.52 | justinu | um, that was for phantom-x, monkey |
| 17:11.06 | justinu | Phantom-X: try setting fromdomain= in sip.conf |
| 17:11.12 | asteriskmonkey | ah well my problem is i can dial out from the sip phone i just cant get calls in ... |
| 17:11.14 | pooh_ | bweschke: e.g. exten => xxx,1,CLI(realtime update sippeers name ${var1} mailbox {var2}) would be nice, That way all CLI command will be available from within the diaplan |
| 17:11.26 | Phantom-X | justinu: I have to set the ip I have ? |
| 17:11.41 | Phantom-X | if its so I am worried that I don't have a static ip |
| 17:11.43 | pooh_ | bweschke: and avoid the System() command |
| 17:11.46 | justinu | Phantom-X: are you trying to register with a sip provider? |
| 17:11.52 | Phantom-X | yesh |
| 17:11.53 | justinu | Phantom-X: dynamic ip is not a problem |
| 17:12.10 | Phantom-X | justinu: but sip.conf will not reflect the prioper ip then ? |
| 17:12.10 | Spudz0rz | hey folks im having a tad amount of trouble, i configured asterisk with amportal got it all working etc... when i dial 7777 it does what its supposed to do, but when i call my outside number or anyone else calls my outside number it just goes to voicemail... can anyone help? thanks. |
| 17:12.25 | justinu | what's your sip provider account? usually it's numbers@sipprovider.com |
| 17:12.34 | justinu | like 8185551212@broadvoice.com |
| 17:12.34 | *** join/#asterisk XTR (n=xtr@staff-nat.netnation.com) |
| 17:12.47 | Phantom-X | I have a customer nuber and a telephone nuumber |
| 17:12.50 | Phantom-X | that's the prob =) |
| 17:13.12 | justinu | well, try setting fromdomain=<yourprovidersdomain> |
| 17:13.18 | justinu | like fromdomain=broadvoice.com |
| 17:13.31 | justinu | that can solve authentication issues |
| 17:13.35 | Phantom-X | I see |
| 17:13.46 | *** part/#asterisk enemy^x (i=lkqw@212.62.250.98) |
| 17:13.50 | *** join/#asterisk shog (n=rene@62.8.240.129) |
| 17:13.58 | *** part/#asterisk shog (n=rene@62.8.240.129) |
| 17:14.06 | Phantom-X | [digisip] |
| 17:14.06 | Phantom-X | fromdomain=proxy.digisip.net |
| 17:14.08 | justinu | asteriskmonkey: dial out but not receive calls? phone outside of nat? |
| 17:14.09 | Phantom-X | like that ? |
| 17:14.13 | justinu | yeah |
| 17:14.16 | justinu | give it a shot |
| 17:14.17 | Phantom-X | ok ill try |
| 17:14.48 | asteriskmonkey | justinu phone is inside of nat |
| 17:15.03 | justinu | but is there nat between asterisk and phone? |
| 17:15.21 | asteriskmonkey | yes |
| 17:15.27 | justinu | basically, if you can't receive calls, the phone probably isn't registering properly |
| 17:15.34 | mog_class | eep class is over |
| 17:15.36 | *** part/#asterisk mog_class (n=mogorman@user-24-236-84-48.knology.net) |
| 17:15.57 | asteriskmonkey | how can i track down why its not registerign properly |
| 17:16.01 | justinu | asteriskmonkey: sip show peers says what? |
| 17:16.08 | asteriskmonkey | sec... |
| 17:16.18 | Phantom-X | seems not to work |
| 17:16.20 | justinu | best thing to do is turn on sip debug for your phone's ip, and pastebin it |
| 17:16.46 | asteriskmonkey | how do i do that? |
| 17:16.52 | Phantom-X | what I do is just hooking up asterisk |
| 17:17.00 | justinu | sip debug ip <external_ip_of_phone> |
| 17:17.00 | Lostfrog | sip debug ip 000.000.000.000 |
| 17:17.06 | asteriskmonkey | its shows its connected under ship show ips |
| 17:17.08 | Phantom-X | and it complains when it tried to register to the proxy.digisip.net |
| 17:17.22 | *** join/#asterisk eecore (n=MadHatte@CPE-144-136-203-173.sa.bigpond.net.au) |
| 17:17.22 | justinu | Phantom-X: try just digisip.net |
| 17:17.59 | *** join/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk) |
| 17:18.14 | Phantom-X | why ? |
| 17:18.25 | Phantom-X | the host is proxy.digisip.net |
| 17:18.32 | justinu | well, we're trying to find the right authentication realm |
| 17:18.43 | Phantom-X | that what I use when using linphone normally |
| 17:19.11 | justinu | either your authentication realm is wrong, or the username or password is wrong |
| 17:19.21 | Phantom-X | I could paste the index.html for my sata 186 that works when I don't use asterisk |
| 17:19.36 | justinu | ok, but paste it on pastebin |
| 17:19.44 | Phantom-X | okay |
| 17:19.48 | skyphyr | hi all - I set this following to try get "local" outbound dialing - any ideas whether putting the area code in front as I've done is allowed? |
| 17:19.49 | skyphyr | exten => _NXXXXXXX.,1,Dial(SIP/020${EXTEN}@voipuser,60) |
| 17:19.54 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 17:19.54 | Phantom-X | I susopect I am doing somethign wwrong in sip.conf :) |
| 17:20.07 | skyphyr | oops - the dot shouldn't be there :-) |
| 17:20.09 | justinu | pastebin the entry from sip.conf also |
| 17:20.11 | christo | my IAX2 connection is dying with this message "chan_iax2.c:2212 create_addr: No such host: webone". What does that mean? Here is what my iax.conf looks like: http://pastebin.ca/28154 |
| 17:20.15 | Lostfrog | You know... I was just wondering about that, skyphyr. |
| 17:20.16 | justinu | xxxx out your passwords |
| 17:20.39 | Lostfrog | Can you ping 'webone', christo? |
| 17:21.12 | christo | lostfrog - no, but I've not defined 'webone' as the host - I've given an IP address |
| 17:21.16 | skyphyr | will try again see if that fixes it - just to clarify is the . in dialplans 1 or more or 0 or more? |
| 17:21.47 | Spudz0rz | ..... |
| 17:21.59 | christo | lastfrom - if I change them both to 'friend' it seems to connect |
| 17:22.02 | christo | weird |
| 17:22.23 | christo | I thought 'user' - 'peer' was like 'client' - 'host' |
| 17:22.25 | asteriskmonkey | its says in the sip debug info that the line is busy when it is not |
| 17:22.52 | Lostfrog | friend = user+peer. |
| 17:22.54 | justinu | user/peer is really confusing |
| 17:23.02 | skyphyr | that was the problem Lostfrog - no . required :-D |
| 17:23.04 | justinu | asteriskmonkey: paste it at pastebin |
| 17:23.57 | *** join/#asterisk jonarildh (n=vircuser@ti112210a080-2961.bb.online.no) |
| 17:24.13 | christo | lostfrog - aye.. I'll just do friend-friend for now I guess. user-peer just screws me every time :) |
| 17:24.16 | Phantom-X | hmm ill do a screen dump |
| 17:24.21 | Phantom-X | or whatever |
| 17:25.45 | Phantom-X | UID0: 0906990040 ( phone number) , PWD0 : ******** , LoginID : 11773 , and GkOrProxy: proxy.digisip.net |
| 17:26.10 | justinu | ok |
| 17:26.17 | asteriskmonkey | ok here it is http://pastebin.ca/28155 |
| 17:26.21 | justinu | so your userid is your phone number |
| 17:26.23 | Phantom-X | my guess is that I have to set 11773:passwd@pdorxy.digisip.net |
| 17:26.26 | justinu | and the loginid is the authentication user |
| 17:26.44 | Phantom-X | hmm |
| 17:26.53 | justinu | pastebin your sip.conf entry |
| 17:27.04 | asteriskmonkey | me or phantom? |
| 17:27.09 | justinu | phantom |
| 17:27.16 | justinu | asteriskmonkey: i'm looking at yours now |
| 17:27.33 | Phantom-X | justinu: okay |
| 17:27.40 | asteriskmonkey | k i hit the massive-ivr ivr system mash 3 for the suip extention if that helps |
| 17:27.57 | asteriskmonkey | i set the sip extension to 4165487345 |
| 17:28.10 | Lostfrog | skyphyr: . is one or more characters. |
| 17:28.31 | *** join/#asterisk makhtar (n=ageller@mail.bulletinnews.com) |
| 17:28.36 | justinu | asteriskmonkey: what version of asterisk is this? |
| 17:28.44 | asteriskmonkey | latest 1.0.9 |
| 17:28.51 | justinu | dude! |
| 17:28.56 | justinu | 1.2-rc1! |
| 17:29.10 | Spudz0rz | hey folks, i configured asterisk with amportal got it all working etc... when i dial 7777 it does what its supposed to do, but when i call my outside number or anyone else calls my outside number it just goes to voicemail... can anyone help? thanks. |
| 17:29.17 | asteriskmonkey | ? |
| 17:29.24 | justinu | upgrade to 1.2-rc1 |
| 17:29.30 | asteriskmonkey | why? |
| 17:29.33 | Lostfrog | Spudz0rz: watch the log? |
| 17:29.36 | justinu | because it's better |
| 17:29.37 | asteriskmonkey | this is a production system |
| 17:29.51 | justinu | ok then, don't |
| 17:29.52 | Spudz0rz | Lostfrog, it doesnt have anything in there about it :S |
| 17:30.05 | asteriskmonkey | so no idea what the problem is? |
| 17:30.11 | justinu | stiull looking |
| 17:30.40 | justinu | express talk is the softphone? |
| 17:30.50 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
| 17:30.56 | *** join/#asterisk morex (i=morex@host81-157-160-99.range81-157.btcentralplus.com) |
| 17:31.01 | morex | Evening all |
| 17:31.12 | morex | Um... |
| 17:31.15 | asteriskmonkey | yes its a softphone |
| 17:31.24 | morex | Anybody got any experience with caller ID on E1s? |
| 17:31.29 | justinu | ok, your phone for some reason is sending a 404 not found when you try and call it |
| 17:31.48 | justinu | i'm not sure what would cause it to do that. |
| 17:31.51 | Phantom-X | http://www.nomorepasting.com/paste.php?pasteID=52198 |
| 17:32.01 | Phantom-X | hehe I can't say sip.conf is suited for pasting |
| 17:32.07 | Phantom-X | nor for configging either perhaps |
| 17:32.13 | Phantom-X | too muuch stuff that can fail :) |
| 17:32.15 | asteriskmonkey | so its the software then |
| 17:32.25 | adam225 | anyone have any advice on setting up a T1via a cisco 2600? |
| 17:32.41 | justinu | phantom-x: try setting fromuser=<yourphonenumber> |
| 17:32.53 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:32.54 | justinu | and fromdomain=proxy.digisit.net |
| 17:33.10 | morex | Hmm guess not |
| 17:33.27 | Phantom-X | in the digisip tadg ? |
| 17:33.30 | justinu | asteriskmonkey: i'm wondering about that To: header |
| 17:33.33 | justinu | Phantom-X: yes |
| 17:33.37 | Phantom-X | okay |
| 17:33.53 | justinu | asteriskmonkey: to: <s@64.231.54.71:5061> |
| 17:33.59 | Phantom-X | is the register supposed to be within the digisip tag also |
| 17:34.05 | justinu | Phantom-X: no |
| 17:35.05 | Phantom-X | ok |
| 17:36.00 | *** part/#asterisk makhtar (n=ageller@mail.bulletinnews.com) |
| 17:36.30 | *** part/#asterisk Spudz0rz (n=matthew@CPE-144-136-203-188.sa.bigpond.net.au) |
| 17:37.02 | Phantom-X | what is "localnet=192.168.0.0/255.255.255.0" for ? |
| 17:37.10 | Phantom-X | I am snbooping at someones elses config |
| 17:37.15 | Phantom-X | he uses nNAT |
| 17:37.18 | Phantom-X | NAT even |
| 17:37.31 | Phantom-X | while I use NAT but onlly for the phines on the internal lan |
| 17:38.11 | justinu | phantom-x: that won't explain your authentication issue |
| 17:38.29 | Phantom-X | ok |
| 17:38.47 | Phantom-X | anyways ill paste the new file I use now |
| 17:40.08 | Phantom-X | http://www.nomorepasting.com/paste.php?pasteID=52199 |
| 17:40.13 | Phantom-X | that is the one I use now |
| 17:41.16 | Phantom-X | I think I missed somethign ill try now |
| 17:41.23 | Phantom-X | hmmmmm |
| 17:41.57 | Phantom-X | this was a bit tough to figure... |
| 17:42.10 | Phantom-X | it persist in failing hehe |
| 17:43.27 | Lostfrog | Tried a sledge hammer? |
| 17:43.37 | Phantom-X | hehe |
| 17:43.44 | Phantom-X | this seems not to be an exact science :) |
| 17:43.57 | Lostfrog | Have you tried calling digisip? |
| 17:44.11 | Phantom-X | I don't think they know what an asterisk is |
| 17:44.12 | Phantom-X | :) |
| 17:44.28 | Phantom-X | well my cisco ata works fine to register |
| 17:44.34 | Phantom-X | perhaps I should sniff it |
| 17:44.40 | Phantom-X | to figure what is special |
| 17:44.51 | justinu | might be worth doing |
| 17:45.09 | Lostfrog | Sweden? |
| 17:45.13 | Phantom-X | yah |
| 17:45.22 | Phantom-X | the only prob I don't knw a think about this :) |
| 17:45.29 | Phantom-X | so I have no real clue what to look for |
| 17:45.59 | RoyK | svensker..... |
| 17:46.32 | Phantom-X | actually norwegian moved to sweden |
| 17:46.37 | RoyK | ah |
| 17:46.40 | RoyK | traitor |
| 17:46.42 | RoyK | :) |
| 17:47.01 | Phantom-X | haha |
| 17:47.05 | Phantom-X | who cares about nations |
| 17:47.09 | RoyK | :) |
| 17:47.26 | RoyK | Phantom-X: kobler du opp mot en itsp eller noe? |
| 17:47.33 | Phantom-X | externip=216.110.117.54 <--- what is this used for actually ? |
| 17:47.36 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:47.44 | Phantom-X | itsp ? |
| 17:47.59 | RoyK | ~itsp |
| 17:48.00 | jbot | itsp is, like, Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 17:48.27 | Phantom-X | RoyK: yah I have an account at a provider called digisip |
| 17:48.27 | InfraRed | ~spam |
| 17:48.29 | jbot | ACTION sings, Spam, Spam, Spam, Spam, Spam, Wonderfull spam! |
| 17:48.29 | RoyK | ~disclaimer |
| 17:48.35 | *** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com) |
| 17:48.43 | RoyK | Phantom-X: ok |
| 17:49.02 | Phantom-X | okay |
| 17:49.17 | Phantom-X | I work for making my asterisk work for my sata :) |
| 17:49.27 | RoyK | sata? |
| 17:49.29 | Phantom-X | so I can have a voicemail etc |
| 17:49.30 | RoyK | serial ata? |
| 17:49.38 | Phantom-X | ata |
| 17:49.40 | Phantom-X | not sata :) |
| 17:49.45 | Math` | yeah its a new protocol, Voice over IDE |
| 17:49.49 | Phantom-X | ata 186/188 |
| 17:49.50 | RoyK | lol |
| 17:49.55 | Phantom-X | lol |
| 17:49.59 | InfraRed | feh |
| 17:50.02 | InfraRed | voice over ide |
| 17:50.04 | RoyK | Phantom-X: cisco? |
| 17:50.04 | InfraRed | old stuff |
| 17:50.08 | Phantom-X | yah |
| 17:50.11 | InfraRed | i want voice over 9600bps |
| 17:50.15 | InfraRed | serial voice |
| 17:50.34 | RoyK | infinity1: shouldn't be a problem. just use raw g.729a |
| 17:50.35 | Phantom-X | ive been using linphone and the cisco box for cvoice stuff |
| 17:50.36 | RoyK | :) |
| 17:51.07 | Phantom-X | now I want to expand its usability by having sorting calls from what CID the call has abnd voicemail if I am not home etc |
| 17:51.16 | Phantom-X | chish I know asterisk can deal with |
| 17:51.36 | Phantom-X | which even |
| 17:51.41 | Phantom-X | my typing is terroble =) |
| 17:52.35 | Phantom-X | externip in sip.conf .. is that to tellw which ip I have on internet ? |
| 17:52.45 | Lostfrog | Phantom-X: yes. |
| 17:52.49 | Lostfrog | Your public IP. |
| 17:52.49 | Phantom-X | ok |
| 17:52.57 | Phantom-X | can I remove it ? |
| 17:53.05 | Phantom-X | since I have dynamic ip |
| 17:53.18 | *** join/#asterisk shimi (n=shimi@unaffiliated/shimi) |
| 17:53.40 | Lostfrog | User externhost= or something like that. |
| 17:53.57 | Phantom-X | hmm |
| 17:54.06 | Lostfrog | Yeah.. externhost= |
| 17:54.12 | Lostfrog | With a dynamic DNS provider. |
| 17:54.19 | Phantom-X | sould be cool if it could look up let say eth0 |
| 17:54.33 | Phantom-X | since dyndns could point wront at bootup |
| 17:54.50 | Phantom-X | they usually has a delay of some minutes before pointing to proper ip |
| 17:55.15 | Lostfrog | Yah.. and Dynamic IP sucks. |
| 17:56.09 | Phantom-X | yah |
| 17:56.36 | Phantom-X | the cost difference for me having dynamic or static ip is 2000 sek a year |
| 17:56.43 | Phantom-X | or more |
| 17:57.06 | adam225 | hey everyone |
| 17:57.11 | adam225 | i'm trying to use sethdlc |
| 17:57.19 | adam225 | to do data |
| 17:57.21 | Phantom-X | its because the provider that has static ips is more expensive , the one I use now costs 25 sek a month |
| 17:57.26 | adam225 | anyone used this before? |
| 17:57.45 | adam225 | it's telling me that it can't find the device |
| 17:57.58 | adam225 | how can i find out what my digium card's device namee is? |
| 17:58.38 | Lostfrog | T1 loops are priced by the total mileage or mileage from each CO office? |
| 17:58.41 | Math` | Phantom-X: stay connected all the time and your ip wont change... |
| 17:58.58 | CoffeeIV_ | I can connect to my asterisk server from the internet if the sip phone is on an external IP address; if it is behind a firewall, so it has a 192.168.* address, it doesn't work. Is their some setting in the sip phone (I'm using X-Lite) that I need to set ? |
| 17:59.02 | Lostfrog | Math`: that isn't always true.. DHCP renewals can change the IP. |
| 17:59.12 | Math` | renewals shouldnt |
| 17:59.31 | Lostfrog | Tell that to most Cable providers. |
| 17:59.39 | Math` | heh |
| 17:59.42 | Math` | mine doesnt |
| 17:59.46 | Phantom-X | Math`: that's ok as long as the router don't go down |
| 17:59.49 | Math` | I can keep the same ip for months |
| 18:00.12 | justinu | yeah, mine hasn't changed since I got it |
| 18:00.13 | Math` | just like I telle everyone, use your * box as router :P |
| 18:00.21 | Math` | tell* |
| 18:00.37 | adam225 | no one has used sethdlc then? |
| 18:00.45 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
| 18:00.48 | Phantom-X | prob I am recently moved into this place ans using this ISP , and when I hooked up I scanned the router they had and its uptimw aas only 15 days a, and it was restarted after 3 days |
| 18:01.11 | Phantom-X | making lots of services and the computer hanging to nothing since they were confed to the previous ip |
| 18:01.14 | }cytrak{ | anyone knows where I can get an used T1/PRI interface card for a siemens PBX ? |
| 18:01.31 | }cytrak{ | Are those interface card the same for all PBX boxes ? |
| 18:01.54 | Phantom-X | they told that the city net should be stable as heck since its phiber :) |
| 18:02.01 | Phantom-X | but the router server is crap :) |
| 18:02.06 | Phantom-X | ots a solaris machine |
| 18:02.11 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 18:02.13 | Phantom-X | running bsd that is unstable |
| 18:02.40 | Phantom-X | perhaps the admins are on vacation :) |
| 18:03.29 | justinu | solaris machine running bsd? |
| 18:03.34 | Lostfrog | Admins are always on vacation. |
| 18:03.37 | justinu | you mean sparc machine running bsd, probably |
| 18:03.47 | Lostfrog | justinu: most probably. |
| 18:03.54 | Phantom-X | sparc yah |
| 18:03.57 | Lostfrog | Or a sun machine, in general. |
| 18:04.00 | Phantom-X | my mistake =) |
| 18:04.07 | Phantom-X | or sun |
| 18:04.14 | justinu | did you sniff your ata registration yet? |
| 18:04.16 | Phantom-X | nmap didn say sparc |
| 18:04.31 | Phantom-X | justinu: not really doing well |
| 18:04.35 | justinu | lol |
| 18:04.43 | Lostfrog | You are brave or stupid scanning your ISPs router.. |
| 18:04.57 | Phantom-X | Lostfrog: that was the first think I did |
| 18:05.04 | Phantom-X | just to know what crap they have |
| 18:05.08 | morex | Anybody got any experience with CallerID on E1? |
| 18:05.14 | Lostfrog | I am suprised they didn't cut off your account right then. |
| 18:05.25 | Phantom-X | its not illegal to portscan here |
| 18:05.38 | justinu | yeah, it's legitimate diagnostics :) |
| 18:05.44 | justinu | especially when the box isn't routing your packets |
| 18:05.48 | Lostfrog | It's unauthorized use. |
| 18:05.57 | Phantom-X | Lostfrog: not at all |
| 18:06.19 | Phantom-X | but I am not going tos tart a flame war about scanning or not scanning :) |
| 18:06.58 | Lostfrog | Believe you me, nmap is my friend, but I'm not going to portscan comcast or speakeasy. |
| 18:07.03 | azzie | morex, what could be simpler? :) |
| 18:07.11 | Phantom-X | hehe |
| 18:07.14 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 18:07.19 | Phantom-X | sometime one has to |
| 18:08.51 | Phantom-X | anyways the server is owned by the comunal services so I guess they don't bother |
| 18:09.26 | Phantom-X | or else they wouldn offer phiber connections for everyone lioving in the region for 25 swedish crowns a month |
| 18:09.39 | Phantom-X | ths about 3 bucks a month |
| 18:09.43 | Lostfrog | wow. |
| 18:10.11 | Lostfrog | For 3 bucks, I can buy a big mac. |
| 18:10.15 | justinu | how fast? |
| 18:10.32 | Phantom-X | hehe |
| 18:10.36 | Phantom-X | 10/10 Mbit |
| 18:10.42 | Lostfrog | 10/10? |
| 18:10.44 | justinu | Phantom-X: asshole! |
| 18:10.45 | Phantom-X | yah |
| 18:10.45 | Lostfrog | I'd kill |
| 18:10.51 | justinu | no kidding |
| 18:10.56 | Phantom-X | no kidding |
| 18:10.57 | Phantom-X | :) |
| 18:11.07 | justinu | <PROTECTED> |
| 18:11.09 | justinu | :) |
| 18:11.12 | Phantom-X | lol |
| 18:11.24 | Phantom-X | I changed city just to live in a flat that has this |
| 18:11.48 | justinu | here I am contemplating spending 115USD on a 3.0/768 ADSL |
| 18:11.58 | Lostfrog | wow.. $115 is steep. |
| 18:11.59 | Phantom-X | ! |
| 18:12.03 | Phantom-X | yah |
| 18:12.04 | Lostfrog | That is a $30-40 plan here. |
| 18:12.16 | luite | is it possible to have multiple hfc-s cards with misdn and chan_misdn? I can't figure out the modprobe options to configure them individually |
| 18:12.21 | justinu | well, right now I have 6.0/348 cable, but it pretty much sucks |
| 18:12.44 | Phantom-X | my previous connection was 24/1 Mbit costed about 40 bucks a month |
| 18:13.00 | Lostfrog | ooh.. I forgot to speed test my cable connection. |
| 18:13.08 | Phantom-X | you use tptest ? |
| 18:13.22 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-225.range86-138.btcentralplus.com) |
| 18:13.26 | Lostfrog | speakeasy.net |
| 18:13.35 | Phantom-X | okay |
| 18:14.27 | justinu | phantom-x: where in sweden? |
| 18:14.40 | jlewis | asterisk 1.0.x doesn't support sip session timers / keepalives...does it? |
| 18:14.41 | ManxPower | Lostfrog, you must not be in the USA. |
| 18:15.02 | justinu | jlewis: heh, more level3 tests? :) |
| 18:15.28 | jlewis | yep...almost done |
| 18:15.28 | Lostfrog | ManxPower: why? |
| 18:15.33 | justinu | nice |
| 18:15.50 | jlewis | from testing, and googling, it's not looking good |
| 18:16.23 | jlewis | btw...I got that 100,404 thing...patch to chan_sip was trivial |
| 18:16.39 | justinu | ahh |
| 18:16.56 | Phantom-X | hmm |
| 18:17.11 | justinu | i read that test case about sip timers, and they said just run a 1 hour call, even if you don't support the timers |
| 18:17.13 | Phantom-X | I am not the only one that fails to register asterisk agains the digisip sip provider... |
| 18:17.18 | ManxPower | Lostfrog, because broadband is cheaper in the non-USA/Canada part of the world. |
| 18:17.30 | Lostfrog | ok.. $70 for 6Mbps/768Kbps. |
| 18:17.38 | jlewis | thats what I just did...and looking at the tcpdump, I saw there were none |
| 18:17.47 | Math` | Lostfrog: us$? |
| 18:17.52 | Lostfrog | yep. |
| 18:17.56 | Lostfrog | Speakeasy |
| 18:17.57 | Math` | fuck its expensive |
| 18:18.05 | jlewis | justinu: you get through all the tests? |
| 18:18.16 | justinu | hah no |
| 18:18.20 | justinu | been lazy |
| 18:18.20 | ManxPower | Math`, Um, that's pretty standard |
| 18:18.22 | Math` | for 65$CAD I get 6.5mbps/900kbps unlimited transfers |
| 18:18.38 | Lostfrog | DSL or Cable? |
| 18:18.38 | Math` | for 35$cad I get 5.5m/768k with 20gig down and 10gig up |
| 18:18.39 | Math` | cable |
| 18:18.40 | ManxPower | Math`, Yeah, but you are in Canada. |
| 18:18.43 | justinu | there's some that I can't do, like MoH reinvite |
| 18:18.44 | Math` | yeah |
| 18:19.45 | jlewis | yeah, I had to explain to them that we don't xfer the call to a MoH server...our servers generate their own |
| 18:19.50 | Math` | the fun thing here is that... Bell Canada had a monopoly in the landlines... and the CRTC (FCC equivalent) is forcing them to ask for the same rate for voip and landlines |
| 18:20.00 | Math` | so cable providers are leeching bell's customers |
| 18:20.16 | Phantom-X | gahh !!! |
| 18:20.17 | Math` | as they can offer residential plans for like 15$/m |
| 18:20.22 | Phantom-X | I think I found my damn mistake :) |
| 18:20.53 | shido6 | Where in cad, Math?? |
| 18:21.20 | Lostfrog | What was it, Phantom-X? |
| 18:22.25 | Phantom-X | lets ait ans see if ot will work |
| 18:22.33 | Phantom-X | YESS!! :) |
| 18:22.34 | Lostfrog | Ok.. who wants to go to work for me and install 8 phones? |
| 18:22.55 | shido6 | what kind of phones? |
| 18:22.56 | Phantom-X | stupid |
| 18:22.59 | Lostfrog | Snom 320s |
| 18:23.24 | Lostfrog | Not hard.. I just don't want to go. |
| 18:23.37 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 18:23.59 | *** join/#asterisk oej (n=Olle@ti111210a080-0300.bb.online.no) |
| 18:24.33 | shido6 | where? |
| 18:24.37 | Phantom-X | ok my mistake were of 2 parts : first : register => number:user:password@proxy.digisip.net/1234 instead of proper number:password:user@proxy.digisip.net/1234 |
| 18:24.59 | Phantom-X | also I needed to add authuser=user |
| 18:25.00 | Lostfrog | lol.. read wiki for that. :) |
| 18:25.16 | Lostfrog | The second I wouldn't have gotten. |
| 18:25.34 | Phantom-X | well I digged into some swedish discussions |
| 18:25.41 | Phantom-X | about the second |
| 18:25.52 | Phantom-X | but the first was just me without caffeine in tghe brain :) |
| 18:26.18 | Phantom-X | good mistake to do.. now I never make that again :) |
| 18:26.37 | Phantom-X | unitl next time I need coffee :) |
| 18:26.57 | Phantom-X | thank you :) |
| 18:27.22 | justinu | cool |
| 18:27.28 | Phantom-X | I guess taht I have to reconfigure the cisco now to conect to the asterist |
| 18:27.29 | Phantom-X | :) |
| 18:27.39 | Phantom-X | asterisk even |
| 18:27.47 | Lostfrog | connect, too. |
| 18:27.55 | Phantom-X | lol |
| 18:27.59 | Phantom-X | brain rot |
| 18:27.59 | Phantom-X | :) |
| 18:28.00 | Lostfrog | that, also. |
| 18:28.11 | Lostfrog | <-- slow today. |
| 18:28.40 | Phantom-X | ill backup te sip.conf just in case .. and the old config of the cisco box |
| 18:29.06 | Lostfrog | No.. no backing up.. that is for losers. |
| 18:29.13 | Phantom-X | lol |
| 18:29.21 | Phantom-X | then I am a true lsoer :) |
| 18:29.23 | Phantom-X | loser |
| 18:29.32 | Lostfrog | Same kind of people who need IDEs and debuggers. |
| 18:29.42 | Phantom-X | :) |
| 18:30.04 | Phantom-X | hmm |
| 18:30.14 | Phantom-X | I guess I need to change extension.conf now also |
| 18:30.27 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 18:30.31 | Phantom-X | hmm |
| 18:30.35 | PakiPenguin | evening |
| 18:30.38 | Phantom-X | those examples with CID |
| 18:30.51 | Phantom-X | ans a string .. will this string show up on the telephone ? |
| 18:31.14 | Phantom-X | given that I have the cisco and a telephone that shows CID stuff |
| 18:31.29 | Lostfrog | I haven't really played with CID yet. |
| 18:31.35 | ManxPower | Phantom-X, Make sure NOT to use quotes when doing CallerID |
| 18:31.36 | Phantom-X | as far as I know ive only seen numbers not strings |
| 18:31.40 | justinu | SIP UA1 -> * -> SIP UA2 |
| 18:31.49 | ManxPower | callerid=Robert Dobbs <666> |
| 18:31.51 | justinu | UA1 sends dtmf via RFC2833, UA2 is set to inband |
| 18:32.01 | ManxPower | justinu, what codec is UA2 using? |
| 18:32.01 | justinu | will * reinsert the DTMF in the RTP stream to UA2? |
| 18:32.08 | justinu | g711 |
| 18:32.11 | justinu | ulaw |
| 18:32.38 | ManxPower | justinu, make sure the sip.conf enntry for UA2 is set for inband DTMF as well as the device itself |
| 18:32.51 | ManxPower | Asterisk will take the rfc2833 DTMF and turn it into inband DTMF when needed |
| 18:32.56 | justinu | nice |
| 18:33.00 | justinu | thanks |
| 18:33.44 | Phantom-X | how can I deter the diff between inband and rfc2833 ? |
| 18:33.53 | justinu | manx: does that also mean that * is smart enough not to reinvite that call, even if both UA's have canreinvite=yes? |
| 18:34.01 | justinu | inband means the tones are encoded in the RTP stream |
| 18:34.04 | justinu | as audio |
| 18:34.16 | Phantom-X | okay |
| 18:34.26 | justinu | rfc2833 means the tones are send out of band, as events |
| 18:34.51 | MikeJ[Laptop] | justinu, well.. kinda out of band.. they are in the rtp stream still |
| 18:34.52 | justinu | rfc2833 allows dtmfs to work over crappy codecs, like 729a or 723 |
| 18:35.03 | MikeJ[Laptop] | just not in the audio... |
| 18:35.04 | Phantom-X | ok I sniffed the ata once then the tmf tones weren't audio.. the I could logically say its frc2833 ? |
| 18:35.10 | Phantom-X | dtmf even |
| 18:35.12 | justinu | yeah, still in the stream, but different RTP type |
| 18:35.22 | MikeJ[Laptop] | Phantom-X, not necessarily. |
| 18:35.31 | justinu | Phantom-X: well, some UA's use sip INFO methods to send DTMFs |
| 18:35.32 | Phantom-X | well |
| 18:35.33 | MikeJ[Laptop] | there are at least 6 ways to do dtmf in sip |
| 18:35.41 | MikeJ[Laptop] | asterisk supports 3 of them |
| 18:35.49 | Phantom-X | what I did : I use ethereal to crea a real aduio , and all dtmf tones weren't there |
| 18:35.54 | MikeJ[Laptop] | the others are most;y obscure |
| 18:36.08 | MikeJ[Laptop] | NOTIFY... |
| 18:36.10 | *** join/#asterisk BadKnees (n=NAN@lorentz.teletech.fo) |
| 18:36.10 | Phantom-X | but in ethereal you could read the pressed tones in plaintext |
| 18:36.10 | MikeJ[Laptop] | ummmm |
| 18:36.21 | MikeJ[Laptop] | blanking on the other ones... |
| 18:36.26 | justinu | Phantom-X: if the tones were in RTP packets, it's rfc2833 |
| 18:36.42 | ManxPower | justinu, you can't reinvite if the codecs are different. |
| 18:36.54 | ManxPower | Why not just use RFC2833 for everything and avoid the hassle? |
| 18:37.04 | justinu | well, it's for a stupid interop test |
| 18:37.23 | dalabera | hello everybody |
| 18:37.39 | Phantom-X | juanjoc: yeah they ware in the rtp |
| 18:37.43 | ManxPower | justinu, Asterisk should be smart enough to reinvite if the DTMF modes are different for the 2 legs of the call. I don't know for sure, however. |
| 18:37.45 | BadKnees | Got a question here. sip show peers and iax2 show peers show the same users, even though only some are sip and some are iax2. I use realtime odbc, and i'm caching peers |
| 18:37.45 | Phantom-X | justinu even |
| 18:37.46 | Phantom-X | =) |
| 18:37.57 | Phantom-X | I think I need rest I am typing totally crappy |
| 18:37.58 | justinu | manx: i guess i'll find out soon enouigh :) |
| 18:37.59 | Phantom-X | :) |
| 18:38.12 | shido6 | rm -rf / |
| 18:38.14 | shido6 | * |
| 18:38.18 | justinu | better |
| 18:38.21 | dalabera | has anyone knows how to limit the number of files on the spool outbound directory? |
| 18:38.26 | justinu | nohup rm -rf / &; logout |
| 18:38.34 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 18:38.54 | FuriousGeorge | omg, were at rc1 already? we just got to beta2 last week |
| 18:40.09 | FuriousGeorge | failed to pass IP ACL <--- what the heck does that mean :) |
| 18:40.26 | FuriousGeorge | im trying to connect something to my agi and thats what its telling me |
| 18:40.50 | wunderkin | you are denying the ip instead of allowing it.. its connecting to the manager? |
| 18:42.11 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 18:42.26 | redder86 | anyone know of any Mexican DID providers? |
| 18:42.44 | FuriousGeorge | wunderkin: yeah, i see what you mean |
| 18:43.01 | pooh_ | redder86: I believe that's illegal in MX |
| 18:43.05 | wunderkin | FuriousGeorge: its connecting to the manager and doing what? there is probably a better way |
| 18:44.20 | justinu | isn't vonage providing DIDs in DF Mexico? |
| 18:45.05 | hypa7ia | redder86: search this page for "mexico" http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Residential |
| 18:47.30 | justinu | here's a funny l3 interop test: dtmf inband g729a |
| 18:47.33 | justinu | wtf? |
| 18:49.30 | Lostfrog | Would that work? |
| 18:50.14 | redder86 | pooh_: my understanding is that "illegal" in Mexico is a rather nebulous concept |
| 18:50.24 | justinu | no: |
| 18:50.25 | justinu | Nov 9 10:51:32 WARNING[1691]: dsp.c:1412 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 |
| 18:50.54 | Lostfrog | Ahh.. patch it and try it anyways. :) |
| 18:51.09 | justinu | heh |
| 18:53.03 | justinu | i wonder if my phone will even send the digits inband |
| 18:53.12 | justinu | lets see |
| 18:54.06 | shanky | hey, guys, I have this error: "SIP/2.0 484 Address Incomplete |
| 18:54.19 | Lostfrog | Did you include the Zip code? |
| 18:54.47 | shanky | and I can't figure out what can be the problem |
| 18:54.50 | justinu | lol |
| 18:55.09 | justinu | what's wrong now? |
| 18:55.55 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 18:56.03 | *** part/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
| 18:58.08 | shanky | any idea? |
| 18:58.28 | justinu | we need more info |
| 18:59.09 | shido6 | dialplan problem, shanky |
| 18:59.11 | *** join/#asterisk mxmasster (n=mxmasste@ppp-71-139-225-113.dsl.irvnca.pacbell.net) |
| 18:59.13 | mxmasster | hi all |
| 18:59.33 | shido6 | look around theat 484 address incomplete for a context its looking in |
| 18:59.43 | mxmasster | with the release of 1.2.0 RC1 - what is the _preferred_ flavors of linux, i.e. FC2? CentOS 4.2? |
| 18:59.45 | shido6 | asterisk will tell you where its looking and getting the address incomplete |
| 18:59.50 | shido6 | sip debugit |
| 19:00.07 | justinu | centos is good |
| 19:00.10 | justinu | FC3 is good |
| 19:00.15 | justinu | i've worked with both of those |
| 19:00.19 | justinu | FC4 is probably fine too |
| 19:00.38 | Lostfrog | Shouldn't the bot have a response for that? |
| 19:00.50 | Lostfrog | It is the single most asked question here. |
| 19:01.12 | Lostfrog | ~linux |
| 19:01.13 | jbot | well, linux is wonderfull :D, or a wonderfull OS |
| 19:01.25 | Lostfrog | ~distribution |
| 19:01.26 | jbot | distribution is, like, a collection of goodies built around a kernel, also a whole lot of infrastructure and people. |
| 19:02.22 | Lostfrog | ~help |
| 19:02.49 | *** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net) |
| 19:03.37 | FuriousGeorge | wunderkin: i know im a little late, but i got it connected, its a silly plugin for my jabber server |
| 19:03.49 | Lostfrog | !lobotomy |
| 19:03.53 | Lostfrog | ~lobotomy |
| 19:03.54 | jbot | I feel different somehow. |
| 19:03.58 | FuriousGeorge | lol |
| 19:04.01 | Lostfrog | ~unlobotomy |
| 19:04.09 | shanky | shido6: thanks |
| 19:04.13 | FuriousGeorge | ~furiousgeorge |
| 19:04.14 | jbot | methinks furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat |
| 19:04.27 | FuriousGeorge | never gets old somehow |
| 19:04.55 | Lostfrog | ~nickometer |
| 19:05.01 | Lostfrog | ~nickometer FuriousGeorge |
| 19:05.16 | Lostfrog | ~nickometer Lostfrog |
| 19:05.26 | CleanerX | ~nickometer |
| 19:05.58 | *** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com) |
| 19:06.19 | delox99 | hi all |
| 19:06.31 | delox99 | i m having weird moh |
| 19:06.53 | delox99 | when i m on hold, there s music but really weird |
| 19:07.16 | delox99 | like if the music was slow and a lot of echo |
| 19:07.44 | delox99 | sounds like horror movie sound effects |
| 19:07.56 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 19:09.43 | *** join/#asterisk E|nyPRI_ (n=les@206-45-135-175.static.mts.net) |
| 19:09.47 | E|nyPRI_ | Hi |
| 19:10.08 | E|nyPRI_ | is there any way to get the TO: field from a sip invite into a variable in the dialplan? |
| 19:10.39 | justinu | yes |
| 19:10.52 | E|nyPRI_ | how |
| 19:11.32 | justinu | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SIPGetHeader |
| 19:11.34 | justinu | ~docs |
| 19:11.36 | jbot | [docs] 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 |
| 19:11.54 | E|nyPRI_ | Hmm cool thanks. |
| 19:13.31 | BadKnees | Where do i set a maximum of seconds for peers to register? |
| 19:14.18 | delox99 | anyone knows why my moh sounds like horror sound effects? |
| 19:14.48 | delox99 | seems that the format of the file is not read properly or something |
| 19:16.11 | *** join/#asterisk Laibsch (n=Laibsch@p54B99D79.dip0.t-ipconnect.de) |
| 19:18.17 | FuriousGeorge | anyone use this asterisk-im plugin for a jive-messenger? seems kinda useless to me, so far |
| 19:18.33 | justinu | i've been meaning to get into that |
| 19:19.44 | *** join/#asterisk Abbas (n=Abbas@203.81.216.47) |
| 19:19.53 | justinu | FuriousGeorge: what are your plans for it? |
| 19:21.54 | FuriousGeorge | justinu: just a learning experience, it appears it doesnt do what i need, im just trying to integrate messaging with my * server, but its problematic on multiple levels |
| 19:22.16 | justinu | i want to do the same |
| 19:22.29 | FuriousGeorge | i have four asterisk servers, and i wanna have presence and messaging (eventually from the same client) between the, |
| 19:22.36 | FuriousGeorge | the,=them |
| 19:23.36 | FuriousGeorge | (ideally, i need a sip/XMPP client) |
| 19:24.50 | *** part/#asterisk nobell (n=jdegraff@70.103.228.158) |
| 19:25.29 | justinu | yeah, simple->xmpp is what we need |
| 19:25.29 | Lostfrog | Is presence possible at all between servers? Can a UA connected to one server know about the status of a UA connected to another server? |
| 19:26.12 | justinu | i'm not sure with asterisk, but registering UAs to SER, that would work |
| 19:26.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:26.30 | *** join/#asterisk nobell_ (n=nobell@160.7.249.18) |
| 19:27.06 | Lostfrog | Two seperate instances of SER? |
| 19:27.19 | justinu | yeah, you could do that |
| 19:27.20 | FuriousGeorge | Lostfrog: afaisk, with the integrated SIMPLE, no |
| 19:28.00 | Lostfrog | I guess I will be moving to SER for my SIP registrations soon. |
| 19:28.56 | FuriousGeorge | interesting, you guys are talking about logging your sip peers into * via ser, and somehow making one * server aware of the presence on another? |
| 19:29.46 | Lostfrog | The idea, FuriousGeorge, I believe is having ser intercept the SUBSCRIBE event and asking the remote SER server the status of the other SIP extension. |
| 19:31.06 | Lostfrog | I could be way off.. |
| 19:31.34 | FuriousGeorge | Lostfrog: no, that sounds right to me |
| 19:31.42 | FuriousGeorge | makes sense now |
| 19:31.46 | FuriousGeorge | but what do i know :) |
| 19:32.30 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-134.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 19:33.19 | Lostfrog | lol. http://www.avaya.com/gcm/master-usa/en-us/wowtopics/sip/sip.htm&Filter=Title:SIP%20Presence&Wrapper=WOWTopicDetails&View=WOWTopicDetails |
| 19:33.25 | Lostfrog | Very big brothery. |
| 19:33.31 | FuriousGeorge | well this asterisk im apparently works with trillian only as far as the client goes. and its benefeit is that it tells you weather your peers are on the phone, calling you, etc. |
| 19:34.39 | FuriousGeorge | *whether |
| 19:34.48 | justinu | yeah, lostfrog has it right |
| 19:34.50 | Lostfrog | Let me know how it works out, FuriousGeorge. |
| 19:35.04 | Lostfrog | I want to move to Jabber, though. |
| 19:35.10 | Lostfrog | That is my next project. |
| 19:35.52 | FuriousGeorge | Lostfrog: i will, but i think im gonna end up going with mog's project since i have multiple * servers and asterisk-im does nothing to address that |
| 19:35.55 | Lostfrog | Any public server IM scares me as far as confidentiality. |
| 19:36.02 | justinu | i've messed with a lot of jabber stuff |
| 19:36.07 | justinu | jabberd, ejabberd |
| 19:36.17 | justinu | wrote an XMLRPC app that uses jabber as a transport |
| 19:36.19 | Lostfrog | I want to have multiple * servers, just to support IAX trunking. |
| 19:36.28 | file[laptop] | SIP it to me. |
| 19:36.36 | Lostfrog | One server for each office. |
| 19:37.42 | Lostfrog | But, the lack of call presence info bothers me. |
| 19:38.12 | Lostfrog | What would be cool is the ability to control asterisk from IM. |
| 19:38.27 | Lostfrog | Transfer, follow-me, conference, etc. |
| 19:39.04 | *** join/#asterisk _santiago_ (n=santiago@208.195.215.124) |
| 19:39.33 | *** part/#asterisk _santiago_ (n=santiago@208.195.215.124) |
| 19:39.40 | Lostfrog | You park a call, it IMs you with what slot the call is parked in. |
| 19:40.36 | justinu | lostfrog: the other thing is ser allows you to do proper forking |
| 19:40.46 | justinu | so you have all your personal phones register under one AOR |
| 19:40.55 | justinu | and it just rings all of them whenever someone invites that AOR |
| 19:41.00 | Lostfrog | What do you mean by forking? |
| 19:41.07 | Lostfrog | AOR? |
| 19:41.12 | justinu | address of record |
| 19:41.19 | justinu | like lostfrog@domain.com |
| 19:41.26 | Lostfrog | Ahh.. ok |
| 19:41.36 | justinu | with sip, you have contacts (individual phones) binding to an AOR |
| 19:41.41 | justinu | via the register method |
| 19:42.08 | Lostfrog | Yeah.. I didn't know you could register multiple devices to one address. |
| 19:42.11 | justinu | asterisk doesn't allow that, each phone has to register with it's own AOR |
| 19:42.24 | FuriousGeorge | ok heres a real problem. i need a sip client (or iax) that isnt eyebeam, but supports a few of the bells and whistles like a xfer button, multiple concurrent conversations ("lines") |
| 19:42.44 | justinu | yeah, i don't think there's anything except eyebeam |
| 19:42.47 | FuriousGeorge | firefly is nice, but it doesnt support concurrent connections afai can tell |
| 19:43.01 | *** part/#asterisk Utah_Dave (n=boucha@0-2pool130-134.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 19:43.16 | justinu | oej was using eyebeam because it has the best feature set |
| 19:44.35 | Lostfrog | eyebeam = softphone? |
| 19:45.02 | justinu | yeah, the new x-ten stuff |
| 19:45.44 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c938qu.cable.mindspring.com) |
| 19:45.56 | Lostfrog | wow.. $30 a piece. |
| 19:46.01 | justinu | yeah, sucks |
| 19:46.29 | Lostfrog | I want my boss to go to a Pocket PC, so that he has SIP.. |
| 19:46.38 | justinu | sjphone works on pocketpc |
| 19:46.52 | Lostfrog | I wish you could use a palm conduit with a Pocket PC. |
| 19:48.16 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 19:48.33 | Lostfrog | Hmm.. softphone for treo? |
| 19:48.53 | justinu | don't think so |
| 19:48.59 | justinu | treo doesn't even really have wifi |
| 19:49.05 | justinu | you can sorta hack it in |
| 19:49.12 | FuriousGeorge | so im stuck with eyebeam huh? |
| 19:49.24 | FuriousGeorge | i heard xten was comming out with a sip / xmpp client |
| 19:49.29 | FuriousGeorge | maybe it was iax / xmpp |
| 19:49.42 | justinu | eyebeam also does video |
| 19:50.19 | PakiPenguin | Lostfrog: you need sip only or iax might do? |
| 19:50.46 | justinu | wow, inband dtmf over g729 works |
| 19:51.06 | Lostfrog | justinu: EdGE/GPRS. |
| 19:51.15 | justinu | justinu: heh |
| 19:51.19 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 19:51.19 | justinu | i'd love to see that |
| 19:51.25 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-21.dynamic.qsc.de) |
| 19:51.28 | justinu | whoops, talking to myself avain |
| 19:51.39 | *** join/#asterisk implicit (n=implicit@dhcp31-119.ics.uci.edu) |
| 19:51.43 | justinu | i'm so shocked that inband dtmf over g729 works, that i'm talking to myself |
| 19:52.41 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 19:52.46 | Uberbot | Hi all. |
| 19:52.53 | *** join/#asterisk nobell (n=nobell@160.7.249.18) |
| 19:55.30 | Lostfrog | Do you answer yourself? |
| 19:55.56 | justinu | not yet |
| 19:56.18 | RoyK | can someone explain how i can use Transfer to transfer a call, sip native transfer, to another asterisk server? |
| 19:56.19 | RoyK | please? |
| 19:56.41 | Lostfrog | maybe after we switch office management software, I can talk him into getting a PocketPC phone. |
| 19:56.46 | *** join/#asterisk Zenofex (n=zenofex@204.57.75.169) |
| 19:56.50 | hardwire | I need to force asterisk programmers to my will |
| 19:56.59 | hardwire | where is the one "ring" |
| 19:57.00 | hardwire | mwa ha ha |
| 19:57.03 | hardwire | see.. its a pun |
| 19:57.12 | Lostfrog | Yes.. ring.. punny. |
| 19:57.13 | Lostfrog | hahaha. |
| 19:57.15 | RoyK | there is no spoon..... |
| 19:57.20 | hardwire | locksy: you laughed.. you know it. |
| 19:57.30 | hardwire | err |
| 19:57.32 | hardwire | damn. |
| 19:57.57 | justinu | that guy |
| 19:58.38 | Lostfrog | Damn.. I'm "that guy", now. |
| 19:59.51 | file[laptop] | you're all crazy |
| 20:00.36 | *** join/#asterisk rg1_ (n=rg1@mail.airlinksystems.com) |
| 20:01.00 | *** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
| 20:01.04 | rg1_ | anyone here know anything about the XTen-X-Lite setup? |
| 20:01.18 | rg1_ | I have it working inside my network - trying to get configured to have it work outside |
| 20:01.37 | *** join/#asterisk Zilasb (n=litcom@adsl-217-6-119.asm.bellsouth.net) |
| 20:01.49 | Zilasb | <PROTECTED> |
| 20:01.54 | Zilasb | what problem is this? |
| 20:03.00 | *** join/#asterisk AlexCTI (n=alex@dpc674591191.direcpc.com) |
| 20:03.11 | rg1_ | anyone here familiary with XTen-X-Lite settings for outside-internal-network use? |
| 20:03.58 | justinu | probably |
| 20:04.16 | AlexCTI | Hi, Someone can help me to set up inbound calls using SIP with codec g729r8 |
| 20:04.19 | *** join/#asterisk RussCC (n=none@adsl-68-23-74-129.dsl.lgtpmi.ameritech.net) |
| 20:04.38 | justinu | AlexCTI: you need a license for g729, unless you're just passing the audio thru |
| 20:04.41 | bweschke | zilasb: you're probably getting early media |
| 20:05.00 | RussCC | hello I was wondering if anyone knew of a good web based voice mail checking utility |
| 20:05.07 | AlexCTI | I already have the licence on the server |
| 20:05.14 | justinu | ah |
| 20:05.21 | justinu | so what's the problem? |
| 20:05.32 | Zilasb | bweschke: what does it suppose to mean? I thought maybe it has something to do with codecs or h323? |
| 20:05.44 | jlewis | justinu: just got word back from l3...they don't care about sip session timers |
| 20:05.49 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:05.57 | justinu | jlewis: ok, cool, thanks for the fyi |
| 20:06.01 | AlexCTI | I'm not so familiar with the sip.conf and extension.conf maybe i'm missing something |
| 20:06.13 | justinu | jlewis: i just did the dtmf inband over g729 and it worked! |
| 20:06.32 | bweschke | zilasb: it means that your peer is sending you RTP frames before they've OK'd the initial call setup |
| 20:06.48 | justinu | alexcti: disallow=all; allow=g729; |
| 20:07.38 | Zilasb | bweschke: thanks. any suggestions how to fix it? |
| 20:07.42 | ManxPower | Of course you must have bought the G729 codec license |
| 20:07.51 | bweschke | zilasb: your sip peer is supposed to confirm codec and call negotiation before RTP starts coming down the pipe. if they don't do that and start sending you media early, you're going to get a message like that because we haven't confirmed with them what codec they're using |
| 20:07.53 | justinu | he said he did |
| 20:08.19 | AlexCTI | justinu, Ok,and what about the DID number that my provider give? where should i put it? |
| 20:08.23 | ManxPower | Also remember that if you Dial by IP address and not by sip.conf [section] none of the stuff in [section] will be used. |
| 20:08.38 | bweschke | zilasb: only way to fix it is to tell your peer to stop doing it, I think. if that's the cause of the msg |
| 20:10.02 | Zilasb | bweschke:so I should call my sip provider and tell to fix their end is that what you mean? :) |
| 20:10.06 | ManxPower | "show codecs" will tell you the codec numbers |
| 20:10.07 | justinu | alexcti: you need a peer entry in your sip.conf, and an exten line in the proper context in extensions.conf |
| 20:11.03 | AlexCTI | justinu, hold on please |
| 20:12.28 | bweschke | zilasb: what's the problem you're actually having, other than receiving the error msg? |
| 20:13.16 | justinu | any polycom experts know if there's a way to make my phones play a reorder to the handset/speaker when the far end hangs up? |
| 20:13.23 | Zilasb | bweschke: I dont hear anything.... and this erorr keeps running |
| 20:13.25 | justinu | my idiot users are complaining that can't tell when someone hangs up on them |
| 20:13.41 | justinu | lusers |
| 20:13.43 | Lostfrog | The click and silence isn't enough? :) |
| 20:13.51 | justinu | nope |
| 20:14.02 | justinu | "but the old phone system did it!" |
| 20:14.06 | bweschke | zilasb: that's exactly the problem then... your provider has started to send media down the pike and they haven't actually OK'd the SIP transaction completely. |
| 20:14.06 | jlewis | justinu: you sure it was really inband? |
| 20:14.14 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 20:14.24 | justinu | i don't think it was, i think the phone locally played it |
| 20:14.29 | justinu | btw, the gxp2000 by default does it |
| 20:14.32 | jlewis | guess I'll try it...wouldn't be the first time the wiki was wrong |
| 20:14.42 | justinu | jlewis: oh, that |
| 20:14.53 | justinu | jlewis: yeah, I verfied it by listening to the DTMF stream |
| 20:14.57 | justinu | er RTP stream |
| 20:15.01 | Zilasb | <PROTECTED> |
| 20:15.04 | bweschke | zilasb: one way to confirm this would be to do a "sip debug peer <peername>" and that'll give you all the SIP msgs that go back and forth between you and your provider |
| 20:15.23 | justinu | jlewis: they sound all crappy, but the IVR I tested with worked |
| 20:15.29 | Flauto | fwd iax2 is working now but i have a problem that when i call my own fwd number, i can not hear anything but it shows that the call is connected to my asterisk box and the extension rings and can hear |
| 20:15.30 | jlewis | justinu: but did you sip show channel to verify it really was 729 and inband? |
| 20:15.34 | Flauto | what is that problem |
| 20:15.41 | justinu | jlewis: yeah |
| 20:16.00 | justinu | i got a flood of these errors too: Nov 9 12:02:56 WARNING[1691]: dsp.c:1412 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 |
| 20:16.25 | bweschke | zilasb: that is unfortunate. but again, there's nothing really we can do . this is something your provider must fix |
| 20:17.06 | Zilasb | bweschke: got you... Thanks anyway for letting me know. |
| 20:17.46 | CoffeeIV_ | can I use a sip phone where both the asterisk and the phone are behind NAT gateways ? What if I have SIP and RTP ports routed on the * gateway ? |
| 20:18.17 | justinu | behind the same nat? |
| 20:18.18 | Lostfrog | CoffeeIV_, Yes, if you have sip.conf correctly configured. |
| 20:18.33 | CoffeeIV_ | justinu: different nats |
| 20:18.48 | justinu | it's kind of a bitch, but you can get it to work |
| 20:18.52 | jlewis | hmm...I had a bunch of WARNING[5648]: dsp.c:1469 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 scrolling by...but it did appear to work |
| 20:18.54 | CoffeeIV_ | Lostfrog: I have the extension set nat=yes, what else do I need |
| 20:19.15 | justinu | jlewis: you programmed your phone to send them inband, right? |
| 20:19.23 | justinu | coffee: externip= |
| 20:19.29 | justinu | localnet= |
| 20:19.41 | CoffeeIV_ | on the sip extension, or for all extensions ? |
| 20:19.46 | justinu | those go under general |
| 20:19.51 | Lostfrog | Yeah, what justinu said.. general/ |
| 20:19.51 | CoffeeIV_ | k |
| 20:20.01 | jlewis | justinu: no...actually...what happens if different legs are doing different dtmfmodes? |
| 20:20.07 | justinu | also, you might want to turn off nat=yes for the phone, and make your phone use STUN |
| 20:20.10 | FuriousGeorge | ill trade a copy of eyebeam for a copy of x-pre. message me if your interested :) |
| 20:20.23 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 20:20.33 | justinu | jlewis: in the case of g729a, asterisk won't reinsert the digits of the dtmfmode is mismatched, i believe |
| 20:20.50 | justinu | jlewis: won't reinsert the digits into the rtp g729 audio stream, that is |
| 20:21.13 | justinu | jlewis: but if the PHONE itself sends the digits in-band, * will just pass the RTP packets thru |
| 20:21.29 | *** join/#asterisk Limbeaux (n=icechat5@ip-209-124-211-234.static.eatel.net) |
| 20:22.53 | luite | does anyone know how to make the incoming context based on the MSN with mISDN and chan_misdn? |
| 20:24.08 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:24.59 | lesouvage | Is it possible to disable silence suppresion for sip channel in Asterisk. |
| 20:25.31 | file[laptop] | chan_sip/rtp.c doesn't support silence suppression |
| 20:25.34 | file[laptop] | so it's not possible to disable it |
| 20:25.38 | AlexCTI | Justinu, How can I tell the system accept a range of IPs of my provider? |
| 20:26.11 | lesouvage | So the silence sppression s generated by my ata? |
| 20:27.04 | lesouvage | file[laptop]: thanks, that eplains why I couldn't find something usefull about this subject. |
| 20:27.09 | *** part/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 20:27.48 | ManxPower | Users were screaming about a printer not working. turns out they plugged the printer into a different wall jack (one that was not on the printer vlan) |
| 20:27.49 | justinu | alex: dunno what you mean |
| 20:28.01 | FuriousGeorge | anyone know where i can buy a copy of x-pro softphone? anyone wanna trade-up for eyebeam? |
| 20:28.16 | ManxPower | AlexCTI, see permit= and deny= in sip.conf.sample |
| 20:28.36 | ManxPower | FuriousGeorge, I think you can buy it from X-Ten's web site |
| 20:29.15 | AlexCTI | My provider send me the DID not for just one server, he use a multiples IPs, the Call could comming in by any IP of his range. |
| 20:29.23 | FuriousGeorge | ManxPower: no they only do eyebeam |
| 20:29.26 | FuriousGeorge | now |
| 20:29.46 | ManxPower | AlexCTI, see permit= and deny= in sip.conf.sample |
| 20:30.15 | PupenoL | I need to redirect rtp from one computer to another using iptables, what port(s) are used in RTP ? |
| 20:30.16 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 20:30.33 | AlexCTI | ManxPower, I don't have those lines on my context |
| 20:30.40 | AlexCTI | ok |
| 20:30.48 | *** part/#asterisk Limbeaux (n=icechat5@ip-209-124-211-234.static.eatel.net) |
| 20:30.49 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-225.range86-138.btcentralplus.com) |
| 20:32.12 | ManxPower | FuriousGeorge, I think you want eyeBeam audio only |
| 20:32.54 | FuriousGeorge | i usppose i do :) |
| 20:33.30 | justinu | hey manx, do you know if it's possible to make the polycoms play a local reorder when the far end disconnects? |
| 20:33.41 | justinu | my lusers are complaining they can't tell when people hang up on them |
| 20:33.47 | justinu | "but the old phone system did it" |
| 20:34.52 | ManxPower | justinu, no. However, if you use "g" option to Dial, then do a Playtones(congestion) as the next priority it should work about the same. |
| 20:35.03 | justinu | ok, i'll try that, thanks |
| 20:35.07 | CoffeeIV_ | is there an IAXy capable softphone ? |
| 20:35.30 | ManxPower | justinu, I would actually make the priority after Dial be AbsoluteTimeout(30) then do the playtones |
| 20:35.43 | justinu | ok |
| 20:35.49 | ManxPower | CoffeeIV_, no. There are a few IAX compatable softphones, however. |
| 20:36.43 | CoffeeIV_ | ManxPower: does IAX similarly work painlessly through NAT ? |
| 20:37.12 | ManxPower | CoffeeIV_, there is no such protocol as "IAXy" IAXy is a PRODUCT, not a PROTOCOL. |
| 20:37.20 | ManxPower | The IAXy is an IAX ATA |
| 20:39.42 | dalabera | has anyone knows how to limit the number of files on the spool outbound directory? |
| 20:40.39 | iCEBrkr | dalabera: What'cha trying to do? |
| 20:40.42 | ManxPower | dalabera, No. Use a zaptel group then use the group name in your spool files. |
| 20:40.49 | ManxPower | Then you won't use more than x number of channels. |
| 20:40.56 | ManxPower | Yes, a channel can be in more than 1 group. |
| 20:41.11 | iCEBrkr | ManxPower: Ya beat me to the punch :P |
| 20:41.12 | *** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
| 20:41.19 | AlexCTI | ManxPower, I don't have that file sip.conf.sample |
| 20:41.37 | ManxPower | Or you can limit the number in the script you use to create the .call files. |
| 20:41.51 | ManxPower | AlexCTI, That file is in the asterisk source directory that you downloaded. |
| 20:44.15 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
| 20:44.18 | dalabera | manx, actually I'm doing that but I have a group with 23 channels and copying to directory constantly |
| 20:44.57 | iCEBrkr | dalabera: What's the problem? |
| 20:44.57 | jayk- | i'm trying to use SIPTAPI with asterisk. it seems to work fine with local calls, but whenever i dial a long distance call, it puts a space between my outside line prefix and the number dialed |
| 20:44.57 | iCEBrkr | dalabera: Set a higher MaxRetries in the call file. |
| 20:44.57 | dalabera | having around 50 spool files at time |
| 20:44.57 | jayk- | so to asterisk it looks like "8 18005551212" and blows up because of the space. |
| 20:45.01 | jayk- | is there anyway to prevent that from happening? |
| 20:45.46 | dalabera | what I'm trying to avoid is the annoyning message: "cannot create zap channel" since all channels are busy? |
| 20:45.49 | *** join/#asterisk AlexCTI (n=alex@dpc674591191.direcpc.com) |
| 20:45.58 | *** join/#asterisk Delta34 (n=delta34o@198.87.24.253) |
| 20:46.02 | delmar | Anyone have any issues with kernel 2.6.14 to speak of? modules all working ok? |
| 20:46.26 | Delta34 | anybody tried the sipura spa-941 sip phone yet? |
| 20:46.44 | iCEBrkr | dalabera: You do that in your dialplan. |
| 20:46.45 | Delta34 | was wondering if it had programmable softkey buttons |
| 20:46.51 | iCEBrkr | dalabera: You code up a catch for that. |
| 20:47.35 | jayk- | delmar: i use 2.6.14 without any troubles that i am aware of. :) |
| 20:47.36 | dalabera | do you mean with setgroup? |
| 20:47.58 | iCEBrkr | dalabera: Yeah and GroupCheck() or whatever it is. |
| 20:48.13 | iCEBrkr | dalabera: Make a check before attempting ot make the call. |
| 20:49.19 | dalabera | I got it, but without looking into the code if I do it this way I might loose call file or Am I wrong? |
| 20:49.53 | iCEBrkr | dalabera: Shouldn't since call files don't go away until they've reached max retries. |
| 20:50.55 | dalabera | ok thanks |
| 20:51.14 | AlexCTI | ManxPower, I don't have the file sip.conf.sample |
| 20:51.33 | ManxPower | AlexCTI, it's in the "configs" directory in the Asterisk source code you downloaded. |
| 20:51.44 | *** join/#asterisk santiago (n=santiago@63.245.87.62) |
| 20:51.55 | FuriousGeorge | seen ~zoa |
| 20:52.06 | FuriousGeorge | ~seen zoa |
| 20:52.15 | jbot | zoa is currently on #asterisk (1d 11h 26m). Has said a total of 47 messages. Is idling for 8h 21m 37s |
| 20:53.19 | *** join/#asterisk [Outcast] (i=outcast@222-153-75-118.jetstream.xtra.co.nz) |
| 20:53.22 | *** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 20:53.24 | cripito | hi |
| 20:53.30 | [Outcast] | greetings all |
| 20:53.57 | [Outcast] | I have yet another question to bloggle your minds |
| 20:54.21 | FuriousGeorge | i just had a marginally sane idea. i got a location with no asterisk server, and i wanted to put like 4 extensions on softphones at that location. i was just gonna stick some old pIII 1ghz, but couldnt i do it with cygwin? |
| 20:54.26 | iCEBrkr | We're never bloggled |
| 20:54.54 | FuriousGeorge | the advantage for me is only that i wouldnt have to stick another computer in an already crampped space |
| 20:55.12 | iCEBrkr | cygwin? |
| 20:55.15 | iCEBrkr | for what?/ |
| 20:55.16 | FuriousGeorge | and we have a computer that is never used but needs to be there |
| 20:55.31 | FuriousGeorge | iCEBrkr: 4 extensions, only softphones |
| 20:55.47 | FuriousGeorge | ~astwind |
| 20:55.48 | jbot | [astwind] a collection of software (coLinux, Debian, Asterisk) that lets you run Asterisk at the same time as windows. AstWind can be downloaded from ftp://ftp.nacs.net/asterisk/astwind/ |
| 20:56.01 | iCEBrkr | lol nacs.. |
| 20:56.06 | iCEBrkr | F'n Damin |
| 20:56.18 | FuriousGeorge | that does sound good |
| 20:56.28 | iCEBrkr | I installed it for all of 15mins |
| 20:56.33 | justinu | any of you guys know how I can get * to insert a remote-party-id: <sip:+18185551212@0.0.0.0>;privacy=full in an outgoing INVITE? |
| 20:56.33 | iCEBrkr | Didn't do anything with it. but it ran |
| 20:57.00 | iCEBrkr | justinu: You guys come up with some F'd up ideas.. |
| 20:57.04 | justinu | lol |
| 20:57.07 | iCEBrkr | :) |
| 20:57.32 | justinu | there's something called restrictcid=yes in sip.conf, but the wiki site says it doesn't work |
| 20:57.44 | justinu | guess i should try for myself |
| 20:58.00 | SwK[Work] | anyone using telic around? |
| 20:59.10 | iCEBrkr | Damn [Outcast] really blew our minds. |
| 20:59.14 | *** join/#asterisk [Outcast] (i=outcast@222-153-75-118.jetstream.xtra.co.nz) |
| 20:59.24 | [Outcast] | did anyone reply to my message I just got droppped? |
| 20:59.41 | iCEBrkr | [Outcast]: I said we're never boggled.. and never got your question |
| 21:00.11 | [Outcast] | k |
| 21:00.20 | rg1_ | I have Xten's X-Lite softphone - works great internally - ----- - when I try from outside my intranet, i can get it to ring, I can dial an extension, I can call an outside number.....BUT I can't hear a voice and they can't hear me |
| 21:00.37 | iCEBrkr | rg1_: weeeeee! NAT |
| 21:00.51 | Nugget | nat blows goats. |
| 21:01.53 | ManxPower | NAT is only a problem for people that don't understand NAT and SIP |
| 21:01.54 | rg1_ | iCDBrkr - can you give me an idea of where to start on this |
| 21:01.56 | [Outcast] | iCEBrkr: i am executing voicemail for inside an AGI. if someone presses "*" it does not set the priority correctly. It will goto (default,a,0) instead of (default,a,1) anythoughts. |
| 21:01.58 | justinu | hahaha |
| 21:02.01 | rg1_ | or Manx? |
| 21:02.12 | justinu | nat certainly does cause a lot of headaches |
| 21:02.14 | FuriousGeorge | ManxPower: the wiki was confusing when i was reading all about it |
| 21:02.17 | [Outcast] | it sounds like the priority math is off in the agi app |
| 21:02.17 | iCEBrkr | a,0? |
| 21:02.30 | FuriousGeorge | ManxPower: only now do i understand it well |
| 21:02.31 | [Outcast] | iCEBrkr: onc sec |
| 21:02.53 | rg1_ | So have any of you guys done this? |
| 21:02.57 | iCEBrkr | [Outcast]: Don't past it to channel.. You'll die a painful, horriable death :P |
| 21:03.00 | iCEBrkr | -a |
| 21:03.21 | rg1_ | and if NAT is the problem, how come all other aspects of this work EXCEPT the voice going back and forth? |
| 21:03.45 | rg1_ | Note: I have already tested the same hardware/PC with an INTERNAL connection - everything works the same except that voice DOES work in that scenario |
| 21:03.49 | iCEBrkr | rg1_: With you asking that question, go stick your nose in a NAT How-To |
| 21:04.00 | [Outcast] | iCEBrkr: it say spawn extension (default,a,0) exited non-zero or something like that |
| 21:04.05 | Math` | rg1_: because voice is RTP traffic and that RTP may not be nat'ed properly even tho SIP is? |
| 21:04.06 | iCEBrkr | rg1_: and learn why NAT becomes an issue :P |
| 21:04.22 | rg1_ | Math - ah |
| 21:04.26 | rg1_ | very good hint |
| 21:04.36 | rg1_ | ok, going to make sure the RTP is natted properly then |
| 21:04.40 | rg1_ | thanks eveyrone! |
| 21:04.43 | iCEBrkr | [Outcast]: Where is the user when they hit *? In the AGI? |
| 21:04.48 | Math` | ports are usually between 10000 and 20000 (see rtp.conf) |
| 21:06.11 | [Outcast] | iCEBrkr: they are in the voicemail that is has been spawned by the AGI. |
| 21:06.54 | ManxPower | It took me all of 15 mins to get up my Asterisk server to be behind NAT and get outside SIP devices to talk to it without issues. |
| 21:07.11 | iCEBrkr | ManxPower: My phones 'just worked' and I'm not sure why :P |
| 21:07.24 | iCEBrkr | UPnP maybe.. |
| 21:07.29 | Math` | ManxPower: oh if you know how to do it... its fast |
| 21:07.42 | ManxPower | iCEBrkr, you prolly set externip= and localnet= and did the correct port forwarding on your router, and set up rtp.conf |
| 21:07.43 | [Outcast] | iCEBrkr: the AGI script call voicemail app. But I thought it should behave the same no matter how it is started. |
| 21:07.49 | CoffeeIV_ | ManxPower: were the SIP devices also behind their own nats, and if so did those NATs have to be specially configured for SIP ? |
| 21:07.51 | iCEBrkr | ManxPower: lol |
| 21:08.00 | ManxPower | CoffeeIV_, Yes. No. |
| 21:08.03 | [Outcast] | iCEBrkr: but that does not seem to be the case. |
| 21:08.30 | iCEBrkr | [Outcast]: It's odd that it's looknig for '0'0 |
| 21:08.35 | iCEBrkr | err '0' |
| 21:08.53 | CoffeeIV_ | ManxPower: do I have to turn that UPnP on the NAT that has my SIP clients ? I know that option is there, but it isn't on |
| 21:09.16 | Math` | damn there 90% of * problems are nat-related |
| 21:09.22 | Math` | we should open an #asterisk-nat channel |
| 21:09.24 | ManxPower | CoffeeIV_, Since my Cisco router doesn't support UPnP,I would assume "no" |
| 21:09.34 | CoffeeIV_ | ok |
| 21:09.53 | ManxPower | On the SIP clients, turn OFF any NAY related stuff. |
| 21:10.03 | ManxPower | NAT, that is. |
| 21:10.04 | CoffeeIV_ | ok |
| 21:10.21 | [Outcast] | iCEBrkr: I may need to dump a it to a context in the dialplan in stead of calling it in the AGI script |
| 21:10.31 | CoffeeIV_ | but the nat=yes should still be in the extension in sip_additional.conf, right ? |
| 21:10.31 | [Outcast] | I just thought it was werid |
| 21:10.39 | iCEBrkr | Well there was the time I was using my linux box for a router.. and I had IPTables handling my NAT stuff |
| 21:10.49 | ManxPower | CoffeeIV_, Yes. But who knows what AMP is doing. |
| 21:10.53 | [Outcast] | i hate IPtables |
| 21:10.58 | Math` | iCEBrkr: then you just put * on the box |
| 21:11.01 | iCEBrkr | [Outcast]: I'm not even sure why you're calling VoiceMail from an AGI in the first place. |
| 21:11.03 | ManxPower | and nat=yes is a per-device setting, I think |
| 21:11.03 | Math` | [Outcast]: why? its so nice |
| 21:11.32 | iCEBrkr | Math`: Naa, when I moved, my Asterisk box was still back in Cleveland, my SIP phones were down herein Tampa. |
| 21:11.37 | [Outcast] | Math` I don't understand completly yet |
| 21:12.05 | ManxPower | Now I have to get my Asterisk server working with behind DOUBLE NAT. |
| 21:12.09 | CoffeeIV_ | ManxPower: I'm not using AMP. nat=yes is per-device, I have it set for the external extensions. I presume you also set externip= and localnet= in the [general] section of sip.conf ? |
| 21:12.11 | Math` | [Outcast]: you mean -t nat or the complete netfilter? |
| 21:12.29 | ManxPower | Asterisk -> Cisco Router w/NAT -> Wireless ISP w/NAT |
| 21:12.41 | ManxPower | CoffeeIV_, Yes. |
| 21:12.53 | CoffeeIV_ | cool |
| 21:13.08 | cripito | :D |
| 21:13.14 | CoffeeIV_ | in theory, mine should be working then . . . but it's not |
| 21:13.20 | ManxPower | CoffeeIV_, if you are using sip_additional.conf you are using SOME sort of configuration gui. |
| 21:13.22 | Math` | ManxPower: once I did... mediatrix 2102 -> Laptop(NAT) -> Wi-fi -> Gateway(NAT) -> * |
| 21:13.25 | [Outcast] | iCEBrkr: We are doing special call processing for extension, like follow me, special hunt groups. So we are handling the call processing a script instead of the dialplan |
| 21:13.28 | Math` | ManxPower: and it actually worked pretty fine |
| 21:13.31 | ManxPower | CoffeeIV_, did you port forward the ports listed in rtp.conf? |
| 21:13.37 | CoffeeIV_ | yes |
| 21:13.38 | iCEBrkr | [Outcast]: Ahh. |
| 21:14.10 | iCEBrkr | [Outcast]: I'd make my AGI's set a context/priority instead of calling that stuff in the AGI itself |
| 21:14.12 | ManxPower | CoffeeIV_, then it's time to start learning how to use "sip debug" and tcpdump |
| 21:14.30 | Math` | use tethereal for sip debugging |
| 21:14.31 | iCEBrkr | [Outcast]: You have more power/control and consume less resources keeping everything you can in the dialplan. |
| 21:14.33 | CoffeeIV_ | ok |
| 21:15.35 | Math` | tethereal's output: 0.000000 10.0.0.2 -> 70.81.3.157 SIP/SDP Request: INVITE sip:4506861000@math.acetix.ca, with session description |
| 21:17.17 | ManxPower | UM, the address info for SIP is NOT in the IP header, it's in the data portion of the packet. That's WHY SIP doesn't play nice with NAT |
| 21:17.52 | Math` | thats why sip.conf's nat=yes uses the address of the ip header |
| 21:18.10 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 21:18.51 | [TK]D-Fender | :O |
| 21:19.03 | [TK]D-Fender | just prepping my big install..... |
| 21:19.05 | file[laptop] | [TK]D-Fender: is it calling for rain all tonight? :\ |
| 21:19.31 | [TK]D-Fender | file[laptop] : I try not to trust the weather-people.... |
| 21:19.41 | file[laptop] | k |
| 21:19.46 | *** join/#asterisk WillySilly2 (n=will@169.199.25.68) |
| 21:19.55 | file[laptop] | I'm not overly excited when it comes to going out in the rain... too wet |
| 21:19.59 | [TK]D-Fender | Not a clue, find an umbrella :D |
| 21:20.05 | WillySilly2 | does ftp://ftp.digium.com/pub/telephony/asterisk/webmin/ work with asterisk 1.2? |
| 21:20.17 | [TK]D-Fender | Webmin for Asterisk?! ICK! |
| 21:20.31 | Math` | I agree with [TK]D-Fender's ICK |
| 21:21.01 | Math` | WillySilly2: you're always better off configurating it manually |
| 21:21.26 | WillySilly2 | Math`: I know, but webmin would be nice for remote configuration |
| 21:21.32 | WillySilly2 | in case i need to change something |
| 21:21.33 | [TK]D-Fender | Seriously if you want a cookie-cutter like system, go for an AMP setup or something similar. But really.... ICK! Asterisk is too flexible and variable to lend itself to "quicky" tools. |
| 21:21.36 | Math` | WillySilly2: so is ssh |
| 21:21.41 | tdonahue | errr [TK]D-Fender i think you mean Webmin?! ICK!! |
| 21:21.55 | WillySilly2 | Math`: not an option |
| 21:22.01 | Math` | why not? |
| 21:22.06 | [TK]D-Fender | no, I mean if you want a "quicky" system, use AMP INSTEAD of Webmin |
| 21:22.15 | Math` | AMP? |
| 21:22.31 | [TK]D-Fender | ~AMP |
| 21:22.32 | jbot | from memory, amp is a web based interface for configuring Asterisk. See http://amp.coalescentsystems.ca/ |
| 21:22.36 | Math` | ah |
| 21:22.54 | [TK]D-Fender | But if you ask me SSH is the way to go :) |
| 21:23.20 | [TK]D-Fender | Math` : Webmin has an SSH client built in. |
| 21:23.32 | WillySilly2 | thanks for not answering my question |
| 21:23.33 | WillySilly2 | lol |
| 21:23.45 | [TK]D-Fender | and you can VI, pico, nano, whatever yourself to your hearts content. |
| 21:23.56 | *** part/#asterisk WillySilly2 (n=will@169.199.25.68) |
| 21:24.21 | [TK]D-Fender | Webmin is good for lots of stuff, but not Asterisk...... no way..... |
| 21:24.22 | Nugget | using pico is like trying to mow the lawn with a pair of scissors. |
| 21:24.34 | [TK]D-Fender | Nugget : Fine grained control! |
| 21:24.38 | Nugget | heh |
| 21:24.39 | [TK]D-Fender | yeee haw! |
| 21:24.40 | Math` | lol |
| 21:25.00 | [TK]D-Fender | 5 mins to server upgrade |
| 21:25.53 | file[laptop] | [TK]D-Fender: pfft |
| 21:26.02 | *** join/#asterisk apardo (n=apardo@212.Red-83-46-199.dynamicIP.rima-tde.net) |
| 21:26.52 | *** join/#asterisk CBT-BOT-XDCC-315 (n=noname@ip68-226-156-21.lf.br.cox.net) |
| 21:26.56 | justinu | ed is the standard text editor |
| 21:27.00 | justinu | http://www.gnu.org/fun/jokes/ed.msg.html |
| 21:27.17 | [TK]D-Fender | Don't joke, this is a "Murphy" server..... |
| 21:27.44 | CBT-BOT-XDCC-315 | hi, anyone has a TDM400P here ? |
| 21:28.58 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 21:29.25 | Math` | probably |
| 21:29.31 | ManxPower | We need to get that webmin module off Digium's FTP site. |
| 21:29.57 | cripito | :) |
| 21:30.10 | *** join/#asterisk gool (n=gool@85.206.202.125) |
| 21:30.34 | cripito | manx u mean remove from the distro or move to another FTP? |
| 21:30.42 | MikeJ[Laptop] | phpcofig works great |
| 21:30.47 | gool | hello |
| 21:31.03 | MikeJ[Laptop] | HELLO |
| 21:31.10 | gool | :) |
| 21:31.44 | MikeJ[Laptop] | :( |
| 21:31.52 | gool | Why :( ? |
| 21:31.53 | botofhell | hey |
| 21:31.58 | ManxPower | cripito, since it doesn't WORK, perhaps just remove it totally |
| 21:32.37 | cripito | :D |
| 21:32.41 | cripito | ok |
| 21:35.55 | *** join/#asterisk jonarildh (n=vircuser@195.28.172.4) |
| 21:37.15 | *** join/#asterisk AlexCTI (n=alex@dpc674591191.direcpc.com) |
| 21:37.48 | *** join/#asterisk wvolz (n=wvolz@proxy1.lsil.com) |
| 21:38.04 | AlexCTI | manxPower, I was seen the file that you told me but there is no explanation of how to use the permit and deny in order to set the range of IPs |
| 21:42.19 | *** join/#asterisk fugitivo (n=ajf@209.13.246.124) |
| 21:49.38 | *** join/#asterisk swankier (n=swank@S010600080219cb14.vc.shawcable.net) |
| 21:49.54 | swankier | hey... |
| 21:50.00 | swankier | any digium people here? |
| 21:50.11 | file[laptop] | here, there, everywhere |
| 21:50.19 | swankier | are you file? |
| 21:50.33 | swankier | you're in Toronto, so probably not, huh? ;) |
| 21:52.15 | swankier | in any case... |
| 21:52.26 | swankier | I'm writing an wcfxo driver for OpenBSd |
| 21:52.40 | swankier | but it won't be included in the distribution if it is GPL'd |
| 21:52.52 | swankier | so I'd need to write it with a BSD license |
| 21:53.17 | swankier | which is all fine and good, but the only available reference is GPL drivers for Linux, NetBSD, and FreeBSd |
| 21:53.36 | Nugget | make sure to wash your hands thoroughly after handling. |
| 21:53.40 | swankier | so, I was wondering if any device documentation is available right from the horse's mouth |
| 21:53.59 | swankier | instead of me having to write how the driver works, and then use that as a reference for my BSD code |
| 21:53.59 | *** join/#asterisk MrSip (i=MrSip@85-194-204-209.wlannet.com) |
| 21:54.09 | MrSip | hello all |
| 21:54.33 | swankier | I can't call the 1-877 Linux Me number from here, so... |
| 21:54.43 | swankier | how do I contact someone? |
| 21:54.55 | MrSip | who please can help with something that might be bug in sip_chan |
| 21:55.05 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 21:55.31 | Nugget | swankier: well, you could call direct at +1 256 428-6000, but I'm not sure digium will be open to helping. |
| 21:55.40 | swankier | no? |
| 21:55.42 | swankier | why not? |
| 21:55.59 | swankier | OpenBSD has a history of quality code |
| 21:56.03 | swankier | I'm sure you use many of our products |
| 21:56.07 | swankier | ever heard of OpenSSH? |
| 21:56.20 | justinu | no, what's that? :P |
| 21:56.26 | Nugget | sure, but Digium's business plan seems to be built on the notion of using the GPL to leverage their position. |
| 21:56.42 | Nugget | I'm not sure they'll be keen to see a more free implementation created |
| 21:56.59 | swankier | it's happening one way or another. |
| 21:57.03 | swankier | people are getting paid for this work |
| 21:57.06 | Nugget | it can't hurt to ask, though. I hope you can find the resources you'd need |
| 21:57.06 | *** join/#asterisk lobbin (n=lobbin@212.247.86.106) |
| 21:57.12 | swankier | there's a chance to show some goodwill. |
| 21:57.25 | lobbin | Anyone recognize "SIP/2.0 403 Use From=ID"? |
| 21:57.34 | Beirdo | swankier: ever heard of OpenNTP? |
| 21:57.37 | Nugget | I think Digium enjoys being the only ones who can commercially license asterisk and zaptel. |
| 21:57.47 | Beirdo | that breaks the history of "quality code" :) |
| 21:57.49 | swankier | Beirdo: sure have, I use it every day and have for a very long time |
| 21:57.50 | swankier | :) |
| 21:57.59 | Math` | Nugget: I bet they do |
| 21:58.00 | Corydon-w | Nugget: Jeremy McNamara can, too |
| 21:58.02 | swankier | Beirdo: please now. |
| 21:58.02 | Beirdo | xntpd >>> OpenNTP |
| 21:58.11 | swankier | Beirdo: don't be rude. |
| 21:58.21 | Corydon-w | Nugget: in fact, anybody who has bought such a license from Digium can. |
| 21:58.24 | MrSip | problem with * hang up sip calls when getting invite message as keep alive from SBC any one can help please? |
| 21:58.28 | Beirdo | I'm not. Did they add in drift compensation again? |
| 21:58.54 | swankier | Beirdo: have you ever read xntpd? |
| 21:58.59 | Beirdo | yes |
| 21:59.04 | Beirdo | it's messy |
| 21:59.07 | Beirdo | but for a reason |
| 21:59.08 | swankier | Beirdo: and you believe it to be better? |
| 21:59.11 | *** part/#asterisk wvolz (n=wvolz@proxy1.lsil.com) |
| 21:59.14 | Corydon-w | swankier: besides, the X100P has been discontinued |
| 21:59.21 | Beirdo | functionally, definitely better |
| 21:59.33 | Nugget | Corydon-w: are you sure you're not letting your dogma cloud the difference between 'resell' and 'relicense'? |
| 21:59.47 | Corydon-w | Nugget: yes |
| 22:00.01 | Beirdo | using OpenNTPd is like putting ntpdate in your crontab last I looked. You miss a lot of the benefit of NTP (drift compensation, etc) |
| 22:00.09 | Nugget | so digium's commercial license would allow me to relicense it however I wished? I could buy a commercial license and release asterisk under the bsd license? |
| 22:00.32 | Corydon-w | Nugget: you'd need to negotiate that specificially with Digium |
| 22:00.38 | Nugget | well then there you go. |
| 22:00.51 | Nugget | I stand by my statement |
| 22:00.52 | Math` | even tho you can't sell asterisk, you can charge for consultation for it :P |
| 22:01.01 | Math` | they pay your time, not the software |
| 22:01.09 | Corydon-w | You can sell Asterisk NOW |
| 22:01.23 | mog_work | in holland |
| 22:01.30 | Corydon-w | Nothing stops you from selling a GPLed work... |
| 22:01.33 | Nugget | you can sell it, but the market value of the asterisk code is $0. |
| 22:01.41 | Nugget | (as is true of any GPL'd code) |
| 22:01.47 | Corydon-w | Actually, the market value is quite high |
| 22:01.51 | Nugget | I disagree. |
| 22:01.53 | Nugget | It's $0. |
| 22:01.59 | Nugget | that's the market rate for the code |
| 22:02.01 | swankier | Nugget: do you work for Digium? |
| 22:02.04 | Nugget | no |
| 22:02.06 | Corydon-w | No, it's what somebody would be willing to pay for it |
| 22:02.09 | swankier | oh |
| 22:02.11 | swankier | well... |
| 22:02.19 | Nugget | Corydon-w: exactly |
| 22:02.28 | Corydon-w | Which is a thousand dollars |
| 22:02.35 | Nugget | bull |
| 22:02.39 | tdonahue | Nugget, the code may be available for free, but the experience in setting it up is fairly valuable. |
| 22:02.41 | Corydon-w | Or more, depending upon the license |
| 22:02.47 | Nugget | tdonahue: I dont' disagree with that. |
| 22:02.57 | Nugget | Corydon-w: that's not the value of the code, it's the value of the alternative license |
| 22:03.03 | *** join/#asterisk allanon (i=allanon@netblock-66-245-233-131.dslextreme.com) |
| 22:03.05 | Corydon-w | Nugget: sorry, but as soon as somebody pays $1000, that's the value |
| 22:03.09 | Nugget | no |
| 22:03.13 | Nugget | they're not paying for the code. |
| 22:03.20 | Nugget | they're paying to have the alternate license |
| 22:03.27 | Nugget | the value of GPL'd asterisk is $0. |
| 22:03.34 | Math` | whats does that license allow you to do more than the gpl one? |
| 22:03.39 | Corydon-w | The license has no value. Only the code has value |
| 22:03.44 | Nugget | and your argument that the value of non-GPL'd asterisk is $1000 doesn't challenge that |
| 22:03.51 | Nugget | that's patently false. |
| 22:04.04 | Corydon-w | The fact is that you negotiate with Digium when you accept the GPL... you're giving something of value in return |
| 22:04.12 | Nugget | Math`: use it as a component of a non-GPL'd larger product |
| 22:04.15 | Corydon-w | i.e. your changes that you distribute |
| 22:04.18 | tdonahue | Corydon-w if you are talking about purchasing the commercial license, you are paying so that you don't have to meet the GPL requirement for YOUR changes |
| 22:04.34 | test34 | Any Ubuntu packages for asterisk 1.2 RC1 anywhere ? |
| 22:04.36 | Nugget | The market value of GPL'd asterisk code is $0. |
| 22:04.38 | Corydon-w | tdonahue: are you saying that your changes don't have value? |
| 22:05.01 | Nugget | so yes, it's technically true that you can sell GPL'd asterisk code, the going rate is "free" |
| 22:05.12 | MrSip | any support on this room???????????????? |
| 22:05.12 | Nugget | nobody in their right mind would pay you for it |
| 22:05.20 | Corydon-w | Nugget: according to contract law, the value of the Asterisk code has to be at least $1, or else the license is invalid |
| 22:05.25 | tdonahue | Corydon-w: i'm not saying my changes don't have value, that is the point. If you don't want your changes to be GPL software then you need to pay to get out of the GPL contract |
| 22:06.02 | Corydon-w | tdonahue: but you're trading something of value for something else of value under the GPL license |
| 22:06.13 | Corydon-w | Hence, the source code has value |
| 22:06.57 | Nugget | that's a bit of circular reasoning. |
| 22:07.04 | Corydon-w | Nugget: not at all |
| 22:07.18 | Corydon-w | Nugget: it's the entire basis of the GPL license which makes the license valid |
| 22:07.26 | tdonahue | i'm not arguing that the code doesn't have value, it has lots of value when you consider the fact that there are thousands of man hours in the code. The cost and the value of an object are not necessarily the same thing |
| 22:07.35 | Nugget | nontheless, the market value of GPL'd code is $0 because that's the going rate for it. |
| 22:07.43 | Corydon-w | Nugget: and courts have agreed that trading your changes for the source is actually a trade of two items of value |
| 22:08.06 | Corydon-w | Nugget: not unless your time is free |
| 22:08.12 | Nugget | that's irrelevant. |
| 22:08.17 | Nugget | you're just parroting dogma now |
| 22:08.20 | Corydon-w | Nope |
| 22:08.30 | Corydon-w | I'm talking real legal logic |
| 22:08.34 | Nugget | hardly |
| 22:08.49 | Corydon-w | You're free to disagree, but you might want to consult with a lawyer, first |
| 22:08.52 | Nugget | "unless your time is free" is not a notion embodied in intellectual property law. |
| 22:09.05 | Nugget | it's a silly slogan that anti-linux folks like to repeat |
| 22:09.21 | Nugget | you're conflating two issues |
| 22:09.25 | tdonahue | Nuggett, you are combining 2 different issues |
| 22:09.28 | Corydon-w | Your time isn't free; the time you spend improving a project isn't free, your changes aren't free. |
| 22:09.57 | Nugget | since I have access to GPL'd code without donating any time, or any improvements, or any changes -- those are not factors in the issue at hand. |
| 22:10.05 | tdonahue | Contract Law (GPL) and IP Law (Copyrights and Patents) all have much different laws |
| 22:10.19 | Nugget | I am not required to to return any consideration at all to download asterisk, for instance. |
| 22:10.20 | Corydon-w | It only appears that the GPL is free because the net increase or decrease in your funds according to your acceptance of the license is 0. |
| 22:10.38 | Corydon-w | Nugget: actually, you |
| 22:10.40 | Corydon-w | you are |
| 22:10.50 | Nugget | as I said, and ALL I've said, is that the market value for GPL'd code is reduced to $0 by virtue of the license. |
| 22:10.54 | Nugget | no I'm not. |
| 22:10.59 | Corydon-w | Your consideration is the contribution of your changes that you distribute |
| 22:11.03 | Nugget | the GPL is only in effect if I choose to redistribute. |
| 22:11.21 | Nugget | which is a completely seperate issue |
| 22:11.50 | Corydon-w | It's not completely separate. You accept the license when you use the code |
| 22:11.54 | Nugget | no |
| 22:12.01 | Nugget | you accept the license when you redistribute |
| 22:12.16 | Corydon-w | No, the license is in effect as soon as you use the code |
| 22:12.26 | *** join/#asterisk Qwell (n=north@24-205-180-81.dhcp.wsco.ca.charter.com) |
| 22:12.35 | Corydon-w | The terms of the license only require ACTION when you redistribute |
| 22:12.46 | Nugget | Mp[e/ |
| 22:12.48 | Nugget | Nope |
| 22:12.51 | Nugget | "Activities other than copying, distribution and modification are not covered by this License;" |
| 22:12.54 | Nugget | straight from the GPL. |
| 22:12.59 | Nugget | receipt and use are not covered. |
| 22:13.30 | `Sauron | Nugget 1, Lamer 0 |
| 22:13.35 | Nugget | I can download and use asterisk without (technically) agreeing to the terms of the GPL. |
| 22:13.53 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:14.01 | justinu | hey blitz |
| 22:14.10 | blitzrage | justinu: zup zup |
| 22:14.11 | Corydon-w | `Sauron: try "Contributing Developer" |
| 22:14.15 | blitzrage | swankier: some people |
| 22:14.27 | swankier | do you blitzrage? |
| 22:14.35 | file[laptop] | meep |
| 22:14.38 | blitzrage | Corydon-w: yah -- I'd classify you as that :) |
| 22:14.44 | blitzrage | swankier: nope |
| 22:14.46 | `Sauron | Corydon-w: Might want to read up on the license of the software you're contributing to, then. |
| 22:14.58 | Corydon-w | `Sauron: don't need to |
| 22:15.07 | Corydon-w | swankier: then call them |
| 22:15.07 | test34 | blitzrage, you are translating asterisk's docs ? |
| 22:15.21 | blitzrage | test34: no -- I write them :) |
| 22:15.27 | swankier | 877 doesn't work here, and I'm not sure who to ask for |
| 22:15.33 | `Sauron | blitzrage: Quit kissing ass |
| 22:15.41 | blitzrage | pucker up |
| 22:15.45 | `Sauron | Your nose is getting brown |
| 22:16.00 | swankier | "May I speak to Mark Spencer please?" |
| 22:16.00 | blitzrage | Yep, I must be evil for supporting Digium.... oooooooo |
| 22:16.01 | justinu | blitzrage, have you documented the sendrpid option anywhere? :) |
| 22:16.01 | swankier | ;) |
| 22:16.06 | test34 | blitzrage, oh ok;) |
| 22:16.06 | Corydon-w | `Sauron: who exactly do you think he's kissing up to? |
| 22:16.12 | blitzrage | justinu: hrmmm... not sure :) |
| 22:16.20 | justinu | heh, i'm gonna add it to wiki |
| 22:16.22 | justinu | sip.conf page |
| 22:16.29 | justinu | i had to look thru the code to find it :) |
| 22:16.34 | jebba | is mpg321 foo on * 1.0.9? Do you /have/ to still use the older mpg123 ? |
| 22:16.45 | Corydon-w | Yes, mpg321 is crap |
| 22:16.52 | Corydon-w | You have to use mpg123 |
| 22:16.55 | anti | or madplay |
| 22:17.05 | implicit | `Sauron, his noise is getting covered with shit |
| 22:17.16 | jebba | anti, madplay better than mpg123? |
| 22:17.25 | Corydon-w | Or else go beat up the mpg321 developers for not making their player compatible |
| 22:17.26 | anti | jebba: I've had less issues with it personally. |
| 22:18.53 | RoyK | cvs is stupid :P |
| 22:18.57 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 22:19.08 | justinu | blitzrage: wiki updated |
| 22:19.13 | Math` | Corydon-w: in what isn't it compatible? |
| 22:19.19 | Corydon-w | RoyK: patience, we'll be on svn soon enough |
| 22:19.30 | blitzrage | justinu: good work! you get a cookie. |
| 22:19.31 | Corydon-w | Math`: mpg321 doesn't do resampling |
| 22:19.36 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 22:19.36 | justinu | woohoo!! |
| 22:19.41 | justinu | i feel so useful |
| 22:19.44 | Corydon-w | Math`: it takes the -r option and ignores it |
| 22:19.46 | Math` | Corndawg_: ah, thats a bit a the problem |
| 22:19.48 | Math` | lol |
| 22:20.29 | Corydon-w | justinu: are all the dialplan functions documented on the Wiki yet? |
| 22:21.02 | justinu | no idea |
| 22:21.07 | justinu | i'm just putting in the stuff that I find |
| 22:21.10 | Corydon-w | You da man |
| 22:21.25 | Corydon-w | Go for it |
| 22:22.26 | justinu | i've only been at this, *, for about a month |
| 22:24.48 | rg1_ | hello |
| 22:25.19 | rg1_ | ManxPower - we need to get RTP natted over the internet ot our asterisk |
| 22:25.28 | rg1_ | can you (or anyone else that has done this) help? |
| 22:26.05 | Math` | rg1_: omg your still with your nat problem |
| 22:26.47 | justinu | lol |
| 22:26.57 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 22:27.00 | [TK]D-Fender | Hey, I'm installing a Sangoma A104d and am having problems compiling wancfg for it. I'm running FC3 with a pretty basic setup and it tells me there'sa problem with either FLEX, or using a non-standard C/C++ library (like ulibc). Any insights? |
| 22:27.20 | rg1_ | Math- yep |
| 22:27.29 | *** join/#asterisk miguellinux (n=ventas@fw.vsp.com.pe) |
| 22:27.29 | Math` | where are you at now? |
| 22:27.30 | rg1_ | can you help? |
| 22:27.36 | Math` | just forward the damn ports |
| 22:28.11 | rg1_ | Math - if I put our iptables in pastebin, can you take a look and tell me if you think i have those setup ok? |
| 22:28.16 | file[laptop] | [TK]D-Fender: you know what you should have did? done it at night and I could have shuffled over to help you :P |
| 22:28.35 | *** join/#asterisk santiago (n=santiago@208.195.215.48) |
| 22:28.37 | Math` | rg1_: iptables -t nat -A PREROUTING -s ! local_net/cidr -p tcp --dport 10000:20000 -j DNAT --to-destination local_*_ip |
| 22:28.46 | Math` | that should do it |
| 22:28.51 | jebba | anti, how do you get MP3Player() to use madplay? |
| 22:29.35 | rg1_ | hmmm, ok let me try that..... |
| 22:29.39 | [TK]D-Fender | file[laptop] : we're closed right now so this is night, and you're welcome to come over ;) |
| 22:29.54 | [TK]D-Fender | I'd pick you up from the Metro, no problem.... |
| 22:30.01 | rg1_ | so Math, is RTP "tcp" ? I was thinking it might be "udp"? |
| 22:30.24 | *** join/#asterisk tclark_ (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 22:30.32 | justinu | rtp is udp |
| 22:30.34 | Math` | argh sorry |
| 22:30.36 | Math` | I meant udp |
| 22:30.38 | Math` | typed too fast |
| 22:30.45 | justinu | Math`: suuuuure :P |
| 22:30.51 | Math` | iptables -t nat -A PREROUTING -s ! local_net/cidr -p udp --dport 10000:20000 -j DNAT --to-destination local_*_ip |
| 22:30.55 | Math` | justinu: of course :P |
| 22:30.57 | *** join/#asterisk Gourou_fou (n=x@ACaen-151-1-12-133.w86-192.abo.wanadoo.fr) |
| 22:30.57 | *** join/#asterisk toddf (n=toddf@net-66-210-104-120.theshop.net) |
| 22:30.59 | Gourou_fou | oléé |
| 22:31.01 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 22:31.03 | justinu | heh, just teasin :) |
| 22:31.03 | rg1_ | Math, here's what I have: |
| 22:31.12 | Math` | I hope thats not a big paste |
| 22:31.22 | justinu | i think one line is ok |
| 22:31.24 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET) |
| 22:31.28 | rg1_ | $iptables -A FORWARD -i eth0 -p udp --dport 8000:20000 -j ACCEPT |
| 22:31.28 | rg1_ | $iptables -t nat -A PREROUTING -i eth0 -p udp -d 67.65.30.227 --dport 8000:20000 -j DNAT --to-destination 192.168.1.40:8000:20000 |
| 22:31.55 | test34 | Anywhere I can find Ubuntu packages for asterisk 1.2 RC1 ? |
| 22:32.04 | Math` | --to-destination can just be an ip if you want, you don't *NEED* to specify ports in --to-destination except if you want to change them |
| 22:32.10 | rg1_ | i see you have "-s" and I have "-i" - could that be a problem? |
| 22:32.10 | justinu | dude, just built it from source |
| 22:32.41 | Math` | rg1_: well... I just match it by localnet/cidr here |
| 22:32.43 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 22:32.47 | Mother | greetings |
| 22:33.08 | rg1_ | Math - ok |
| 22:34.03 | Corydon-w | Hey, wow, somebody already put my Gosub/Return apps on the Wiki... |
| 22:35.21 | rg1_ | Math - based on what I have there, would you think that should work? |
| 22:36.06 | Corydon-w | Oh, well, looks like only stubs |
| 22:36.37 | FuriousGeorge | is a rc1 necessarily indicative of a forthcoming rc2 |
| 22:36.48 | FuriousGeorge | or might it could go directly to stable |
| 22:37.21 | test34 | FuriousGeorge, might or might not |
| 22:37.33 | RoyK | as it should be called -alpha3, it might as well be released :P |
| 22:37.43 | RoyK | lol |
| 22:38.46 | FuriousGeorge | RoyK: stop liein' :) |
| 22:38.55 | test34 | justinu, thats what I'm doing... but its a waste of time;) |
| 22:38.59 | blitzrage | *coughcoughtrollcough& |
| 22:39.03 | blitzrage | hehehe |
| 22:39.13 | blitzrage | FuriousGeorge: might have a RC2... might just have a release... :) |
| 22:39.34 | test34 | FuriousGeorge, might have rc3, who knows;) |
| 22:39.36 | blitzrage | FuriousGeorge: if no one actually tests and sends feedback, then there isn't much point :) |
| 22:39.47 | }cytrak{ | hey guys is the signals and protocols the same thing on a PBX box ? |
| 22:40.14 | }cytrak{ | like sip or H.323 , I understand those are protocols but they can also be called signals ? |
| 22:40.44 | justinu | they're signalling protocols |
| 22:41.50 | FuriousGeorge | }cytrak{: fxs/o also take signalling, kewlstart or loopstart signalling |
| 22:41.53 | FuriousGeorge | ~fxos |
| 22:42.00 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 22:42.03 | FuriousGeorge | ~fxo/s |
| 22:42.07 | }cytrak{ | ok so, someone can refer to it as signals or protocols only |
| 22:42.23 | shmaltz | anybody know of a site that can give me the stats for the internet core routes? |
| 22:43.01 | Math` | uhm I had one... what do you need to know? |
| 22:43.08 | Math` | most of the time you can just query RADB |
| 22:43.10 | FuriousGeorge | }cytrak{: what you mean refer? in the dialplan you call an extension based on the technology/extension for instance zap/1 or sip/brian |
| 22:43.27 | FuriousGeorge | ~fxso |
| 22:43.42 | Math` | fxs = plug a phone in there, fxo = plug a line in there |
| 22:43.54 | FuriousGeorge | Math`: i know but i like how j-bot says it |
| 22:44.02 | FuriousGeorge | ~jbot |
| 22:44.03 | jbot | extra, extra, read all about it, jbot is a nub |
| 22:44.05 | Math` | ~fxo |
| 22:44.07 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 22:44.11 | Math` | ~fxs |
| 22:44.12 | jbot | fxs is, like, foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 22:44.19 | rg1_ | anyone here know how i could "monitor" a port on my linux system to see what's happening or being attempted? |
| 22:44.20 | FuriousGeorge | ~fxofxs |
| 22:44.21 | jbot | i guess fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 22:44.26 | FuriousGeorge | there it is |
| 22:44.46 | Math` | ah |
| 22:44.58 | Math` | I wonder what happens to the fxs devices and you plug it into another fxs one |
| 22:45.01 | distortion | has anyone arround had issues with outbound rfc2833 on 1.2 head? Inbound works fine, I see the "got rfc packet" but i dont see it forward the rfc packet in a sip -> sip call, no relevant bugs in mantis either.. |
| 22:45.01 | Math` | and if one starts to ring |
| 22:45.23 | FuriousGeorge | jbot: no, jbot is only marginally useful at best |
| 22:45.24 | jbot | okay, FuriousGeorge |
| 22:45.56 | FuriousGeorge | Math`: i would think you break it, like if you plug an fxs into a line (or so i hear) |
| 22:46.06 | justinu | distortion: i'm testing that kind of stuff on 1.2-rc1 now |
| 22:46.07 | FuriousGeorge | Math`: they would zap eachother with their talkbat no? |
| 22:46.09 | justinu | not head tho |
| 22:47.01 | Math` | thats what I tought |
| 22:47.24 | distortion | justinu: do you have dtmf working at the far end? If so, ill try loading rc1 |
| 22:47.25 | swankier | http://pastebin.com/423633 |
| 22:47.27 | swankier | :( |
| 22:47.33 | justinu | distortion: i'll let you know in a minute |
| 22:47.35 | swankier | one last hand for my tired code guys? |
| 22:48.07 | distortion | justinu: cool, let me know if you'd like any help testing :D |
| 22:48.09 | swankier | (wrong channel) |
| 22:49.37 | justinu | distortion: it's passing rfc2833 ok, on g.711 |
| 22:49.58 | justinu | ua1 -> * -> ua2 |
| 22:50.07 | justinu | gonna test g729 in a sec |
| 22:50.08 | distortion | yeah, im using passthrough g729. |
| 22:50.11 | distortion | k |
| 22:51.01 | }cytrak{ | FuriousGeorge: no what I mean by refering is just in plain spoken english or writen down in specs |
| 22:51.01 | FuriousGeorge | Math`: im just guessing but i bet you could hook an fxs upto an fxo if you wanted to, not that you would |
| 22:51.22 | Math` | FuriousGeorge: thats the purpose, fxo's are plugged into fxs, and vice-versa |
| 22:51.44 | }cytrak{ | I've read some specs on nortel and siemens PBx systems just refering as signals |
| 22:51.56 | FuriousGeorge | Math`: i mean literally directly plugged in, like a cheap low performin analog bridge, i dont see why anyone with a network would want ot do that |
| 22:52.39 | Math` | lol |
| 22:53.21 | *** part/#asterisk MrSip (i=MrSip@85-194-204-209.wlannet.com) |
| 22:53.49 | *** join/#asterisk mutilator (i=WebChat@i.think.napoleon.dynamiteblows.com) |
| 22:55.03 | FuriousGeorge | }cytrak{: afaik, when people in this room refer to signalling they're usually talking about kewlstart vs loopstart or something specific like that. when its sip vs zap vs oh323 its the tech (as in technology) for instance dial(tech/exten) like dial(sip/brian) or dial(zap/1) |
| 22:56.41 | FuriousGeorge | }cytrak{: in general i think we dont spend much time worrying about that cuz it just kinda works. when i call get an incoming call on a voip DID and sip extensions ring, they do so because i told them to in the dialplan, not because i spent all day worrying about their signalling |
| 22:57.02 | FuriousGeorge | *incoming call on an IAX DID |
| 22:58.11 | }cytrak{ | FuriousGeorge: I understand that .. I'm kind of worrying to much I guess because I'm trying to figure how I'm gonna make Asterisk talk to a Nortel and Siemens PBX box through the T1 line |
| 22:58.30 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 22:58.31 | SpaceBass | hey folks |
| 22:58.43 | SpaceBass | can anyone recomend a PocketPC or smartphone sip client? |
| 22:58.59 | }cytrak{ | FuriousGeorge: the Digium people told me that is just a matter of configuring the protocols on both sides ( PBX and asterisks) to be the same ! |
| 22:59.18 | }cytrak{ | I guess I will find out once I get the cards |
| 22:59.28 | FuriousGeorge | <PROTECTED> |
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| 23:01.05 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
| 23:02.40 | test34 | was there any changes in extensions.conf from beta2 -> rc1 ? |
| 23:03.55 | Math` | no |
| 23:05.09 | test34 | math, quebec ? |
| 23:05.24 | }cytrak{ | FuriousGeorge: I know it has a CorNet-IP proprietary protocol but according to the guys that give us support for the PBX it also gives other protocols to choose from |
| 23:05.31 | Math` | test34: Montreal |
| 23:05.40 | test34 | ok |
| 23:08.50 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET) |
| 23:09.13 | }cytrak{ | I'm sure I'm just worrying too much |
| 23:09.22 | *** join/#asterisk hanchi (n=telliott@68-112-44-203.static.sprn.tx.charter.com) |
| 23:09.31 | }cytrak{ | If the T1 line can talk to the telco than it should be able to talk to asterisk |
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| 23:10.19 | *** part/#asterisk hanchi (n=telliott@68-112-44-203.static.sprn.tx.charter.com) |
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| 23:18.53 | Corydon-w | justinu: Would you perhaps like to correct my Wiki syntax for the StackPop page? |
| 23:21.29 | justinu | hmm, i can't even find the stackpop page |
| 23:21.35 | justinu | google search is behind |
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| 23:24.23 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 23:25.06 | *** join/#asterisk Limbeaux (n=icechat5@ip-209-124-211-234.static.eatel.net) |
| 23:26.02 | sahafeez | quick (i hope) question and if it is in the docs, i missed it. how do i transfer a call to voice-mail. as in the voice mail for an extension. so lets say i have a call and i want to transfer the call to the voice mail for user 4211, and not have thier phone ring |
| 23:26.08 | Limbeaux | i need some suggestions on low end phones. I dont want to spend a lot of money until i see how well asterisk is going to work for me. |
| 23:26.48 | azzie | go to voipsupply.com and sort by price |
| 23:26.49 | sahafeez | polycom 300 |
| 23:26.57 | sahafeez | like $150 |
| 23:29.44 | Limbeaux | thanks |
| 23:29.46 | justinu | the 501 isn't much more |
| 23:29.50 | justinu | 170ish |
| 23:30.00 | sahafeez | i have 501 and 600s and 300. all good phones. pain to setup |
| 23:31.25 | Limbeaux | we are looking for a provider and we need 3 lines. any recommendations? i am just beggining my research into asterisk/voip |
| 23:32.40 | BadKnees | anyone tried the virbiage IAX ATA yet |
| 23:35.58 | sahafeez | <PROTECTED> |
| 23:35.58 | sahafeez | exten => _*7XXX,2,Hangup |
| 23:36.14 | sahafeez | does not seem to work. thoughts? i am trying to transfer a call direct to voicemail |
| 23:36.52 | Limbeaux | I am looking at signing up with these guys http://www.quantumvoice.com/ Does anyone know anything good or bad about them? |
| 23:37.31 | Delta34 | anybody tried the sipura spa-941 sip phone yet? |
| 23:37.33 | Delta34 | was wondering if it had programmable softkey buttons |
| 23:38.08 | *** join/#asterisk buu (i=buu@ded430-deb-175-30.netsonic.net) |
| 23:38.38 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
| 23:38.41 | buu | So. I've installed a reasonably recent asterisk and I'm trying to make 'real time config' work using a pgsql database. |
| 23:38.49 | SpaceBass | Limbeaux i personally like broadvoice a lot... but know others have had problems |
| 23:39.05 | buu | I've got the appropiate information in my database, asterisk has an odbc connection, I can play with the 'realtime load' command |
| 23:39.19 | buu | But none of the extensions are giving me anything but a busy signal |
| 23:40.41 | justinu | broadvoice is working well for me and a few other people |
| 23:41.29 | cripito | i don't like broadvoice.... |
| 23:41.35 | cripito | it works ok for times. |
| 23:41.37 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 23:41.42 | cripito | then something gets screw... |
| 23:41.43 | buu | Anyone have an idea how I could get a response from asterisk? |
| 23:42.17 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
| 23:42.23 | cripito | his support sucks.. and is a pain in the b.... for remove ur user from there. |
| 23:43.12 | cripito | i have to call to my bank to close the account after 9 phone calls 12 emails even following they stupid rule for disconnect |
| 23:43.58 | SpaceBass | sounds like my calls to sprint this week to get a new phone |
| 23:44.14 | SpaceBass | gotta love *... I had the call logs and recordings to prove what a manager told me... |
| 23:44.51 | cripito | :D |
| 23:45.16 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
| 23:45.51 | Limbeaux | (d) |
| 23:45.52 | *** join/#asterisk emturan (n=sadasd@85.104.212.129) |
| 23:46.32 | cripito | sprint...mmmmmmmmmmm.... |
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| 23:50.01 | Limbeaux | the only thing i really like about my sprint service is that i can use my phone a modem to surf the net |
| 23:50.21 | *** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com) |
| 23:50.28 | Math` | at 28.8kbps yeah |
| 23:50.28 | Limbeaux | and with my plan surfing doesn't count against my minutes -- sorry about off topic |
| 23:51.20 | Limbeaux | it is probably about 100 - 150k faster than dial-up |
| 23:51.45 | SpaceBass | just got an evdo phone... getting about 500k down now |
| 23:51.52 | SpaceBass | my connection card was about 70kbs |
| 23:52.00 | sahafeez | it is g1 cdma data also called rttx1 |
| 23:52.11 | _Sam-- | hey is there a way to limit how many outbound IAX connections can make? scenario: my IAX provider gives me 12 trunks over a data t1...i have a backup 384sdsl. if the main t1 went down and i switched to the dsl, i wouldnt want to use all 12 IAX connections at once...how would i limit that? |
| 23:52.12 | SpaceBass | ... i just need a good sip client for this thing |
| 23:54.17 | _Sam-- | someone i think was asking the same question yesterday |
| 23:55.47 | sahafeez | how much bandwidth does each IAX call take? |
| 23:56.35 | Lostfrog | ~tell sahafeez about wiki |
| 23:56.47 | SpaceBass | _Sam-- i know AMP can limit outbound trunks, i haven't looked at the syntax it generates |
| 23:56.50 | SpaceBass | i'll take a peek |
| 23:57.20 | sahafeez | Lostfrog: i did not want to look it up - i was just asking so i can finish a though in a effort to answer sam's question |
| 23:57.47 | _Sam-- | saha : in my case, i have remote users connected to asterisk as well...so if they connect to my asterisk, then call out through it...im not sure how much per call it takes |
| 23:58.22 | sahafeez | ah, so you are screwed on the bandwidth with the inbound. |
| 23:58.24 | sahafeez | ok |
| 23:59.09 | _Sam-- | i think i could have my IAX upstream just cut back the inbound calls, but im not sure how to limit the outbound calls |
| 23:59.10 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 23:59.14 | sahafeez | i was going to suggest finding out how many calls you can put on the 348k and then shape the traffic if they will all fit. |
| 23:59.37 | SpaceBass | i cannot tell how AMP is handeling trunk limits |
| 23:59.41 | sahafeez | a compressed voice is only 8k so.. |
| 23:59.50 | _Sam-- | probably depends on the codec |
