| 00:06.11 | dustybin | I just had a wild thought, i could give all my family and friends there own telephone number, and they could ring eachother using my server |
| 00:06.32 | dustybin | and i could ChanSpy() the conversations |
| 00:06.46 | dustybin | *only joking* :P |
| 00:10.08 | dustybin | blist all |
| 00:41.36 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 00:59.09 | *** join/#asterisk chazbro (n=chaz_bro@adsl-70-234-191-111.dsl.tul2ok.sbcglobal.net) |
| 00:59.26 | chazbro | hey there :) |
| 00:59.52 | chazbro | i'm a total newb to voip |
| 01:00.14 | chazbro | i need a clarification |
| 01:00.51 | chazbro | is Asterisk a kind of competitor to Skype? |
| 01:01.23 | chazbro | can I download Asterisk and just use it by itself |
| 01:01.57 | chazbro | or is Asterisk a development engine that ppl use to develop software for? |
| 01:09.12 | chazbro | ok... to simplify my question(s).... what is Asterisk? and don't refer me to the home page. I've seen the home page... it's gobbldy-gook |
| 01:10.22 | chazbro | nobody? |
| 01:10.26 | chazbro | nothing? |
| 01:10.32 | chazbro | :( |
| 01:11.03 | jaytee | asterisk is a toolkit |
| 01:11.12 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 01:12.08 | chazbro | so ppl use it to develop software for voip |
| 01:12.40 | chazbro | it's not a stand alone software to use for voip |
| 01:12.50 | jaytee | more like use it to create a pbx that can handle voip and traditional telephony circuits |
| 01:15.54 | chazbro | i use Slackware Linux on a regular basis... I'm not afraid of the command line |
| 01:16.50 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 01:16.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:16.56 | chazbro | can a person like myself... provided with enough info use Asterisk as a Skype replacement? |
| 01:18.09 | chazbro | i can and do read man pages |
| 01:18.14 | *** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-xepsvevevbgijojr) |
| 01:18.35 | jaytee | i've read the Bible, doesn't mean I believe all of it |
| 01:18.41 | chazbro | im just new to this whole voip thing |
| 01:19.56 | beek | Hey jaytee -- ChanIsAvail() put me onto a solution. Thanks for the reminder. |
| 01:20.11 | chazbro | infoworld.com called Asterisk one of the all time great software of open source |
| 01:20.52 | *** join/#asterisk leon01 (n=leon01@ool-457c12b2.dyn.optonline.net) |
| 01:20.52 | chazbro | I just thought if it's so great, maybe I should use it instead of other voip apps like Skype |
| 01:21.23 | chazbro | if that's possible |
| 01:21.48 | jaytee | beek, cool. only other thing I could think of would cost money and that's an intelligent T1 failover box that could send a network message to run a script on your asterisk server. then use the script to swap to a dialplan that reroutes. |
| 01:22.39 | *** part/#asterisk chazbro (n=chaz_bro@adsl-70-234-191-111.dsl.tul2ok.sbcglobal.net) |
| 01:23.38 | beek | jaytee: This isn't the most elegant solution, that's for sure. I set it up to try the outgoing PRI. If there are no available channels, then try to get a channel on the inbound PRI, which could be jammed with callers. If that fails, I'll use Teliax. |
| 01:26.47 | jaytee | beek, check out voip-wiki.org for Nagios and PRI. there's a couple bash scripts there that could give you some other ideas. |
| 01:29.27 | beek | I'll do that. I was going to write one -- we use nagios for our networks so that would be a natural. I'll head over there and see what has already been done. |
| 01:29.40 | beek | PRI just came up. |
| 01:30.29 | jaytee | you could modify one of the scripts, it just does an asterisk -rx "pri show spans and dumps it into a string variable, uses grep and awk to modify it and then uses if then for checking values. |
| 01:31.12 | beek | voip-wiki.org? |
| 01:31.17 | jaytee | based on the script result you could set a flag in the astdb and have any queue calls test that flag value and route accordingly |
| 01:31.29 | jaytee | duh! my bad. voip-info.org |
| 01:32.11 | beek | Great list! I'm sure I can get some good ideas in there. |
| 01:32.55 | jaytee | http://www.voip-info.org/wiki/view/Asterisk%20Zaptel%20Nagios%20plugin |
| 01:33.30 | beek | There already... I'm thinking that will work fine. We have a plugin that will run that locally and nagios will get the result, so it should be fairly lightweight. |
| 01:33.33 | jaytee | you'd have to mod the script for dahdi |
| 01:35.07 | beek | Figures this would kick me in the ass. I was all set for failover for my inbound PRI. What the hell was I thinking not to provide for an outbound PRI failure? Sheesh! |
| 01:35.14 | jaytee | actually that might just run as it is |
| 01:35.47 | *** part/#asterisk giovani (n=giovani@unaffiliated/giovani) |
| 01:36.13 | beek | I'll need to tweak it a bit to handle which of the four spans are down. |
| 01:36.48 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:36.54 | beek | I just found a major negative point about hulu.com -- I'm listening to an ad that mentions "spotting." Just what I want to hear. |
| 01:37.22 | jaytee | lol |
| 01:41.28 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 01:47.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:49.52 | Qwell | beek: You forgot to check the gender checkbox. |
| 01:50.08 | beek | Qwell -- Is there such a thing? |
| 01:50.14 | Qwell | probably not |
| 01:50.30 | Qwell | better than Facebook randomly choosing female for me, and my boss asking why it set that :p |
| 01:50.33 | beek | I wish... I've seen that damned thing too many times. The gal looks great, but... |
| 01:52.36 | beek | Qwell -- I just checked my profile. There is a Gender option, I have it set for 'male.' Obviously they're not using that as a basis for selecting commercials! |
| 01:52.47 | Qwell | heh |
| 01:52.50 | Qwell | they should |
| 01:53.06 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 01:53.32 | beek | I agree. They have a "like/dislike" commercial feature and I've been rating these as "dislike", without effect. My TiVo would have gotten the hint by now. |
| 02:05.50 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 02:10.42 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
| 02:13.36 | eppigy | RABAJO |
| 02:17.48 | jaytee | RAPIDO! |
| 02:18.24 | eppigy | yes |
| 02:18.31 | *** join/#asterisk matt_d (n=matt@70.134.104.88) |
| 02:18.43 | matt_d | Hello everyone! |
| 02:19.31 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.103) |
| 02:19.37 | matt_d | hello mumtazah |
| 02:20.08 | mumtazah | hello |
| 02:20.08 | mumtazah | :D |
| 02:20.23 | matt_d | What's going on? |
| 02:21.36 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:21.51 | matt_d | hi riddlebox |
| 02:22.12 | riddlebox | hey |
| 02:24.17 | matt_d | is bored. |
| 02:28.05 | riddlebox | i just got home from doing 2 pri cuts |
| 02:29.14 | matt_d | i just got home too.. and am hungry :) |
| 02:29.29 | riddlebox | yeah I just ate nachos |
| 02:29.57 | matt_d | ..... lucky ........ |
| 02:30.15 | riddlebox | yeah |
| 02:30.34 | matt_d | i started working on a Ruby AGI server a while ago. I think i will work on it a bit tonight. but i'm not motivated :( |
| 02:30.46 | matt_d | I swear I have ADD |
| 02:30.49 | riddlebox | yeah |
| 02:31.43 | matt_d | i havent been on irc for years. i am also connected to efnet. its dead there. no body says anything. whats the deal? jabber taking over everything? |
| 02:31.55 | matt_d | and cell phone text messaging |
| 02:32.04 | riddlebox | i dont know I always log into freenode |
| 02:34.02 | matt_d | times are always changing i guess |
| 02:34.40 | riddlebox | identi.ca |
| 02:35.35 | matt_d | twitter clone? :) |
| 02:35.53 | riddlebox | yeah opensource I believe |
| 02:36.04 | *** join/#asterisk tjz (n=tjz@bb121-7-20-94.singnet.com.sg) |
| 02:38.22 | matt_d | hi tzj |
| 02:59.56 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 03:01.51 | johnakabean | exten => s,1,GotoIf($["${CALLERID(number)}" =! "Unknown"]:blacklisted) |
| 03:01.51 | johnakabean | exten => s,n,GotoIf($["${CALLERID(number)}" =! "Unavailable"]:blacklisted) |
| 03:01.54 | johnakabean | are those correct? |
| 03:03.26 | matt_d | does it not work? :) |
| 03:03.56 | matt_d | exten => s,1,GotoIf($["${CALLERID(number)}" =! "Unknown"]?blacklisted) |
| 03:04.00 | matt_d | exten => s,n,GotoIf($["${CALLERID(number)}" =! "Unavailable"]?blacklisted) |
| 03:04.02 | matt_d | try those |
| 03:04.13 | johnakabean | it catches people that have valid caller id's |
| 03:07.10 | matt_d | it appears so |
| 03:08.33 | matt_d | is bored, again. |
| 03:15.22 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 03:18.08 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 03:22.22 | Nugget | drives around the channel in his new car making vroom vroom noises |
| 03:23.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:26.08 | file | gets run over by Nugget |
| 03:26.28 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 03:27.15 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 03:32.44 | Nugget | my brakes are good, I wouldn't hit you! |
| 03:38.01 | thehar | haw |
| 03:41.22 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
| 03:41.52 | *** join/#asterisk leon01 (n=leon01@ool-457c12b2.dyn.optonline.net) |
| 03:48.44 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 03:51.11 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 03:55.00 | *** join/#asterisk jong2 (n=chatzill@c-71-237-170-100.hsd1.wa.comcast.net) |
| 03:55.01 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 03:55.12 | *** join/#asterisk johnantypas (n=jantypas@mail.antypas.net) |
| 03:56.17 | *** join/#asterisk joako_ (n=joako@opensuse/member/joak0) |
| 03:57.05 | johnantypas | Hello all -- not sure if this is a dahdi-tools item or not, but I'm running Asterisk 1.6.1 and I'm trying to find out two things. One, is the new timing API in place to use the MeetMe() function without my needing a kernel module or Zaptel and two, does anyonehave an example of this new function or the Bridge function? |
| 03:59.26 | johnakabean | john, you need a timing source for meetme; that's why meetme will say "that is not a valid conference number" if dahdi or zaptel is not installed |
| 03:59.27 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 03:59.45 | johnakabean | If you DON"T have any cards from digium, pci pci express, you can run dahdi just fine |
| 03:59.54 | johnakabean | it will automatically use the dummy module |
| 04:00.03 | johnantypas | Wasn't the new timing API supposed to fix that? |
| 04:00.08 | johnakabean | that's what I'm doing it |
| 04:00.16 | johnakabean | no |
| 04:00.23 | johnakabean | asterisk will always require some timing source |
| 04:00.28 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 04:00.42 | johnantypas | Ah -- ok, the blogs had noted a new timing API and hinted that the Bridge command might do this. |
| 04:00.46 | johnakabean | its very easy to install dahdi and it won't use but 0.2 percent of your memory |
| 04:00.58 | johnakabean | briding only works for sip to sip |
| 04:01.03 | johnakabean | bridging |
| 04:01.12 | johnantypas | All I've got is sip. |
| 04:01.21 | johnakabean | and if you bridge a call, it takes the call OFF your asterisk |
| 04:01.28 | johnakabean | so noone can use the asterisk features |
| 04:01.37 | johnakabean | I use all SIP |
| 04:01.46 | johnakabean | but I don't want native bridging |
| 04:02.10 | johnakabean | you can't use ANY asterisk features when the calls get bridged |
| 04:02.21 | johnantypas | OK -- I interpreted bridging as a way to join N channels together. Now that I think about it, it would make one "super channel" and I'd lose control of the bridge :-( |
| 04:02.28 | johnakabean | yes |
| 04:02.31 | johnakabean | you lose all control |
| 04:02.55 | johnakabean | so make sure you set canreinvite=no in your sip trunks |
| 04:02.55 | johnantypas | It's good for handoff, but not for conferneces. OK, so just load the dahdi module and I'm all set. |
| 04:03.01 | johnakabean | yep |
| 04:03.17 | johnantypas | OK -- thanks -- now's a good time to try it, no one on the switch tongiht. THX. |
| 04:03.28 | johnakabean | make sure you installed libpri |
| 04:03.30 | johnakabean | first |
| 04:03.44 | johnantypas | OK -- will do. Even for only SIP? |
| 04:03.47 | johnakabean | yes |
| 04:03.58 | [TK]D-Fender | ... |
| 04:04.00 | johnantypas | (I was so desperate to figure this thing out I even tried FreeSwitch....) |
| 04:04.02 | drmessano | WTF |
| 04:04.05 | [TK]D-Fender | wow, what will people come up with next... |
| 04:04.09 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 04:04.13 | drmessano | That is ALL wrong |
| 04:04.19 | [TK]D-Fender | yup, kinda scary |
| 04:04.20 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 04:04.23 | johnantypas | I'm back home now.... |
| 04:04.29 | [TK]D-Fender | Gotta love the late night crazies |
| 04:04.46 | johnantypas | Thanks all! |
| 04:04.51 | *** part/#asterisk johnantypas (n=jantypas@mail.antypas.net) |
| 04:05.23 | Sandheaver | is a late night crazy |
| 04:05.30 | TJNII | is too |
| 04:06.49 | drmessano | 1.6.1 does remove the need for Dahdi as a timing source, for things like IAX trunking.. But DAHDI is not a timing source for MEETME, its used for MIXING for MEETME |
| 04:07.07 | drmessano | and of course that bit has not been removed |
| 04:07.24 | drmessano | and you DONT need libpri unless you.... wait for it.. need it for a PRI |
| 04:07.34 | TJNII | Lies! |
| 04:08.12 | drmessano | The trixbox forums are NOT the ultimate authority on Asterisk |
| 04:08.31 | Sandheaver | WHAT? |
| 04:08.38 | Sandheaver | fact checks... |
| 04:08.38 | TJNII | Okay, now I wished I had saved the "Lies!" line for that last comment. |
| 04:08.41 | Sandheaver | okay you're right |
| 04:08.48 | Sandheaver | ... this time .... |
| 04:08.53 | drmessano | heh |
| 04:09.47 | Sandheaver | = Naikrovek, btw |
| 04:09.58 | Sandheaver | not that it matters |
| 04:10.55 | jong2 | oh boy wth nerds doing in weekend night. |
| 04:11.12 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 04:11.38 | Sandheaver | my wife and daughters are sleeping |
| 04:11.48 | Sandheaver | that's what i'm doing :) |
| 04:11.58 | Sandheaver | ... that's what THEY'RE doing. i'm xchatting |
| 04:12.07 | jong2 | here is weekend night entertainment for ya. http://www.safisystems.com/ |
| 04:12.39 | jong2 | we need some users to try out. if you bored, give it try. |
| 04:12.53 | jong2 | i promise you it will be good entertainmnt. |
| 04:14.11 | coppice | yet another visual callflow designer |
| 04:14.25 | [TK]D-Fender | Yup, rather Shakespearean. Can't decide whether its a tragedy or a comedy :D |
| 04:14.55 | jong2 | it is open source gpl. |
| 04:15.01 | jong2 | u can swallow it. |
| 04:15.10 | coppice | [TK]D-Fender: The Shakespearean comedies are the ones with the jokes |
| 04:15.27 | jong2 | tommorrow tommorrow tommorrow |
| 04:15.53 | jong2 | was it macbeth... signicant nightin...blah blah |
| 04:16.18 | jql | waits for the page to load... and waits... |
| 04:16.26 | jql | does it not like Safari? |
| 04:16.42 | jong2 | probably not. |
| 04:16.56 | jql | whips out the trusty 'fox |
| 04:16.59 | jong2 | rich nerds runs mac. |
| 04:17.13 | Sandheaver | quite a thing to say to a potential customer |
| 04:17.44 | Sandheaver | besides, i know lots of poor yuppies who want you to think they're rich nerds to have macs |
| 04:17.57 | Sandheaver | s/to have macs/who have macs/ |
| 04:18.03 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:18.13 | jong2 | if u have product to sell, i will be more courteous.. |
| 04:18.21 | jong2 | i mean we |
| 04:19.05 | jong2 | i guess you can donate if you are rich... |
| 04:19.12 | jong2 | Ah... my friday night. |
| 04:19.24 | jong2 | why my kid not sleeping yet. |
| 04:21.18 | Sandheaver | jong2: what is your native language |
| 04:21.44 | jong2 | ah korean ... besides my sticky laptop keyboard does not help |
| 04:22.01 | jong2 | u get the idea... |
| 04:22.07 | Sandheaver | yup |
| 04:22.13 | Sandheaver | south korea i assume? |
| 04:22.20 | jong2 | and had a few booze... |
| 04:22.20 | Sandheaver | do they even have internet in NK? |
| 04:22.27 | jong2 | sure. |
| 04:22.58 | jong2 | i heard they raise hacker army believe or nto |
| 04:23.12 | Sandheaver | oh yeah i heard about that |
| 04:23.22 | jong2 | that might be sweet job for nerds.. |
| 04:36.28 | thehar | eats his late night egg salad sammich. nom nom |
| 05:03.22 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:03.22 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:09.39 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 05:11.23 | linagee | can someone fix voip-info.org please? :( |
| 05:11.32 | linagee | http://www.voip-info.org/wiki/view/Asterisk+QoS <-- still doesn't work |
| 05:14.06 | linagee | n/m. found something awesome. http://www.ctunion.com/node/364 |
| 05:14.18 | linagee | premade voip qos goodness |
| 05:14.36 | *** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus) |
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| 05:43.47 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 05:48.55 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 06:05.45 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:07.49 | TJNII | Sweet! New server is up! |
| 06:07.59 | TJNII | Let's see how weel * runs on VBox. |
| 06:08.06 | TJNII | s/weel/well |
| 06:08.10 | TJNII | s/weel/well/ |
| 06:09.02 | TJNII | Not quite, there infobot. At least I know now why you never corrected me before. |
| 06:09.42 | linagee | what the..... arghhhhh. :( |
| 06:10.05 | linagee | is it possible that the sip server my itsp gives me is not the server that sends/receives voice traffic? :( |
| 06:10.22 | linagee | wow that is fscked. here i thought i had low latency to them. |
| 06:10.33 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 06:14.35 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 06:16.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 06:17.53 | linagee | arghhhhhh. |
| 06:17.58 | linagee | bangs head against wall |
| 06:18.25 | linagee | what the heck is the point in putting a local SIP server some place if it's not the actual box that has the traffic. :( |
| 06:19.09 | linagee | i've never wanted to drop an itsp so bad |
| 06:19.37 | TJNII | Which one? |
| 06:19.42 | linagee | TJNII: voicepulse |
| 06:19.55 | linagee | TJNII: they have sjc servers now. oh wait, no, they don't. :( |
| 06:20.16 | linagee | TJNII: i was so excited that sjc-primary.voicepulse.com got a 2ms ping |
| 06:20.36 | TJNII | Yea, I'd be excited about that too. |
| 06:20.45 | linagee | TJNII: why did I have to do a tethereal and ruin it. :( |
| 06:21.03 | TJNII | Ignorance is bliss, man. |
| 06:21.20 | linagee | TJNII: they have some serious voodoo going on. I ran tethereal, called the number, and I'm like, "wtf is this other IP showing all the RTP packets?" |
| 06:21.32 | TJNII | feels, personally, that if ignorance was actually bliss that there would be a lot more happy people. |
| 06:21.34 | linagee | TJNII: I ping that other IP and it's like, 80ms |
| 06:21.48 | TJNII | I think I have 100ms to Broadvoice. |
| 06:21.55 | TJNII | Never really got below 80 |
| 06:22.08 | linagee | TJNII: the thing that absolutely sucks is that I put a ticket in once for bad service. and they soothed me over. wtf |
| 06:22.43 | TJNII | Heh. |
| 06:23.11 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 06:23.13 | linagee | TJNII: unfortunately I think changing ITSPs is harder than just finding a virtual server closer to this new IP. (I only have like 4 DIDs or so, but still) |
| 06:23.40 | TJNII | Yea, transferring numbers can be a pain. |
| 06:23.58 | linagee | TJNII: I would love it if I can find something in or near SF though. maybe in he.net or something |
| 06:25.17 | TJNII | Broadvoice was really nice when I lived in Iowa because I was (relatively) close to their Chicago proxy. Unfortunately, now in Colorado that same proxy is still the fastest, even with the extra 20ms. |
| 06:25.41 | linagee | TJNII: was that the server where the traffic is actually coming from? |
| 06:26.05 | linagee | TJNII: if you run a tethereal and make a phone call, is the IP the same? |
| 06:27.10 | linagee | TJNII: tell me any place on voicepulse's website where they list their internal RTP server's IP though. hah. not. (it doesn't reverse to anything either) |
| 06:27.46 | linagee | 64.61.93.170 |
| 06:27.55 | TJNII | I honestly don't know. I didn't look into it that close. |
| 06:28.43 | *** join/#asterisk shinao1 (n=shinao1@41.219.251.174) |
| 06:28.46 | linagee | TJNII: I think now I know better. when researching a new itsp provider I will ask, "is voip.whatever.com the same server that will be sending/receiving my voice traffic, or is this just a dumb sip server?" |
| 06:29.00 | *** join/#asterisk Tim_Toady (n=moi@adsl108-116.kln.forthnet.gr) |
| 06:33.16 | linagee | TJNII: how can you know if you're going to an itsp provider that terminates, or just one that is reselling a different company? |
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| 06:35.09 | TJNII | I don't know. |
| 06:35.24 | TJNII | You'd probably want to ask the day crew that |
| 06:40.26 | linagee | TJNII: why is everything a scam? :( |
| 06:45.22 | coppice | where is the scam. they have more than one server. you pinged one that is close. they aren't all so close. what did you expect? |
| 06:49.09 | coppice | my ping time to 64.61.93.170 is 370ms :-) |
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| 07:13.13 | maour | where can i find some information about i , t, h in dialplan ? what's the name of this things ? |
| 07:14.47 | jong2 | http://www.asteriskdocs.org/ |
| 07:15.38 | jong2 | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf |
| 07:16.36 | maour | aha , Predefined Extension Names |
| 07:16.41 | maour | jong2: thanks |
| 07:16.54 | jong2 | np |
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| 07:20.57 | maour | t is time out , but how much ? |
| 07:23.59 | jong2 | i am sure it is specified in conf somewhere |
| 07:30.16 | Tim_Toady | default is 5 sec i think, u can set it up by using Set(TIMEOUT(digit)=foo) |
| 07:32.33 | maour | Tim_Toady: what is =foo ? |
| 07:32.49 | Tim_Toady | some numeric value :P |
| 07:32.56 | Tim_Toady | the timeout that u want to setup |
| 07:33.07 | maour | aha |
| 07:33.22 | maour | i tought digit is the value |
| 07:33.28 | maour | but digit is type |
| 07:33.33 | maour | ok ,thanks |
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| 10:52.13 | wonderworld | hi, i have been trying to get a german ptp ISDN line working for several hours now. could someone have a look on my config to verify that it's correct? Hardware is a Digium B410P. We have 1 ptmp and 3 ptp lines -> http://pastebin.com/m70d3273a |
| 10:58.46 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 10:59.01 | verywiseman | can * work as media proxy? |
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| 11:24.03 | wonderworld | hi, i have been trying to get a german ptp ISDN line working for several hours now. could someone have a look on my config to verify that it's correct? Hardware is a Digium B410P. We have 1 ptmp and 3 ptp lines. ptmp works, ptm doesn't -> http://pastebin.com/m70d3273a |
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| 11:27.21 | dustybin | what are the linksys SPA942 phones like? |
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| 12:57.31 | linagee | TJNII: is broadvoice the same thing as broadviewnet.net? |
| 12:59.08 | linagee | coppice: the scam is where they label the new server SJC (san jose), it is located in san jose, but when you make a connection, the server that sends your audio is on the east coast. |
| 12:59.24 | pugachevcobra | hi there... i need help, a week ago my asterisk was running normally with a few hardware devices with no problem. Right now, when an internal call is made to one of these devices, the device freezes and never answers the call. I've tried changing all nat options possible, removing, tweaking, but still not a clue what could be happening. |
| 12:59.59 | linagee | pugachevcobra: sounds like you changed something |
| 13:01.26 | pugachevcobra | linagee: yes, I tried changing the nat settings for the extensions... but I already reverted back to the original settings that were working, but still no luck |
| 13:02.21 | linagee | pugachevcobra: why did you muck with it if it was working? |
| 13:03.06 | pugachevcobra | linagee: it was having a problem unregistering when in the same lan of asterisk |
| 13:03.42 | pugachevcobra | linagee: the device can call with no problems, only receiving the call is causing trouble |
| 13:04.12 | linagee | which has to do with registering |
| 13:04.38 | linagee | when you have a SIP device and it rings and "has a call", it's because it had a registration with the asterisk server |
| 13:06.22 | pugachevcobra | yes... i'm searching here and find out a setting about the qualify setting, what does the qualify serves? |
| 13:06.26 | linagee | TJNII: how do you like broadvoice? |
| 13:07.11 | linagee | pugachevcobra: no idea, sorry. google is your friend on that |
| 13:07.21 | Pan3D | pugachevcobra: have you cranked up the debugging and taken a look at the console to see where the failure starts? |
| 13:08.54 | pugachevcobra | Pan3D: yes... actually all I can see is that the moment I answer the call asterisk starts retransmitting #1 2 3 4 to the device's ip |
| 13:11.46 | *** join/#asterisk wonderworld (n=ww@mue-88-130-102-173.dsl.tropolys.de) |
| 13:11.47 | Pan3D | what do you mean? |
| 13:11.59 | Pan3D | you should paste your console observation to pastebin |
| 13:12.05 | Pan3D | so folks can look at it |
| 13:12.40 | *** join/#asterisk Gugge (n=Guggeman@vlan2.dlxhosting.dk) |
| 13:12.41 | pugachevcobra | http://pastebin.ca/1554906 |
| 13:13.37 | wonderworld | hi, i am trying to make a german ISDN ptp line work. we have 1 ptmp line and 3 ptp lines. ptmp works fine, ptp doesn't. could someone verify my config for me? thanks a lot -> http://pastebin.com/m70d3273a |
| 13:13.57 | linagee | what the hell? "192.168.0.1010" |
| 13:14.04 | linagee | is that normal? ROFL |
| 13:14.43 | Pan3D | sip:2@192.168.0.1010 |
| 13:14.46 | Pan3D | wtf? |
| 13:14.47 | pugachevcobra | oh no sorry i changed it |
| 13:15.01 | pugachevcobra | i masked it |
| 13:15.18 | linagee | Pan3D: maybe it's a special internet where you can have very big octects |
| 13:15.26 | Pan3D | linagee: lol, awesome |
| 13:15.32 | Pan3D | pugachevcobra: you use NAT? |
| 13:15.42 | linagee | pugachevcobra: are you saying you did a replace to the text before pasting? |
| 13:15.47 | pugachevcobra | Pan3D: well internally i don't need to, but i neet for external devices |
| 13:15.55 | pugachevcobra | linagee: yep |
| 13:16.11 | linagee | pugachevcobra: what is 192.168.0.1010? asterisk server? |
| 13:16.29 | pugachevcobra | linagee: no is the extension number 2 |
| 13:16.33 | coppice | linagee: was your call to a phone line on the east coast? |
| 13:16.42 | linagee | coppice: no |
| 13:16.43 | pugachevcobra | linagee: its 192.168.0.101 |
| 13:17.03 | linagee | coppice: I was wondering how they did the east coast/west coast trick. seems tricky with termination and DIDs. |
| 13:17.38 | linagee | coppice: I think the phone company can do failover with some sort of SS7 magic, but I think that's expensive and unusual as well. |
| 13:18.03 | coppice | linagee: I would expect the SIP server to be not so far away, but the audio to come from a server near the target phone |
| 13:18.20 | coppice | in many places failover is compulsory when using SS7 |
| 13:19.05 | linagee | coppice: that is what doesn't make sense. on their site they say to put sjc-primary.voicepulse.com. so logically you'd think things are terminated in SJC. but when I make a call to the number, it comes from east coast |
| 13:19.14 | linagee | coppice: hrm. I wonder what happens if I call a different DID..... |
| 13:19.31 | linagee | coppice: I wonder if they have the DIDs "planted" to a location somehow |
| 13:19.55 | pugachevcobra | linagee: http://pastebin.ca/1554915 corrected |
| 13:20.29 | linagee | calls pugachevcobra's phone at 192.168.0.101. :-D |
| 13:21.04 | pugachevcobra | huh? |
| 13:21.04 | linagee | pugachevcobra / Cleiton |
| 13:21.15 | pugachevcobra | no thats not the phone |
| 13:21.20 | pugachevcobra | with the problem |
| 13:21.38 | linagee | pugachevcobra: I hate NAT. it's the spawn of the internet devil |
| 13:22.32 | pugachevcobra | linagee: i agree but what could we do behind 1 public ip... |
| 13:22.38 | linagee | pugachevcobra: VPN? |
| 13:23.07 | Pan3D | linagee: indeed. NAT breaks the protocols and spreads evil |
| 13:23.28 | pugachevcobra | well |
| 13:23.36 | linagee | Pan3D: it's also non-portable. your NAT may not be like my NAT. |
| 13:23.44 | pugachevcobra | linagee: vpn wouldnt help me in this case |
| 13:23.47 | Pan3D | hey, don't get personal |
| 13:23.50 | linagee | pugachevcobra: why not? |
| 13:23.51 | Pan3D | ;) |
| 13:23.55 | pugachevcobra | linagee: asterisk and the 2 phones are all on the same lan |
| 13:24.08 | Pan3D | then why is the NAT retransmission coming in? |
| 13:24.15 | linagee | lol. true |
| 13:24.16 | pugachevcobra | nat is activated |
| 13:24.31 | linagee | pugachevcobra: if your asterisk box is local, there is no reason to use NAT.... |
| 13:24.38 | pugachevcobra | linagee: i use it externally as well |
| 13:24.45 | linagee | pugachevcobra: use what externally |
| 13:24.47 | Pan3D | but until you figure out the problem, turn it off |
| 13:24.49 | pugachevcobra | linagee: and rtp wasnt passing through |
| 13:25.02 | Pan3D | pugachevcobra: if you have devices outside, it's understandable you need to do something |
| 13:25.04 | linagee | pugachevcobra: why do you "use it externally as well"? are you at a colo? |
| 13:25.07 | wonderworld | hi, i am trying to make a german ISDN ptp line work. we have 1 ptmp line and 3 ptp lines. ptmp works fine, ptp doesn't. could someone verify my config for me? thanks a lot -> http://pastebin.com/m70d3273a |
| 13:25.13 | Pan3D | in the meantime, turn off the NAT until you find the local problem |
| 13:25.19 | pugachevcobra | i have already turned it off |
| 13:25.23 | pugachevcobra | it didnt help... |
| 13:25.35 | linagee | pugachevcobra: are you working off of a cable or DSL connection? |
| 13:25.46 | pugachevcobra | i mean, i removed the nat=yes from sip.conf and from the extensions... or do I need to put in sip.conf nat=no ? |
| 13:25.57 | pugachevcobra | i didnt try nat=no in sip.conf, just in extensions |
| 13:26.04 | pugachevcobra | linagee: dsl |
| 13:26.19 | Pan3D | and you can't get more IPs? |
| 13:26.21 | linagee | pugachevcobra: what you're doing is a bad idea (TM) then. get a real colo and run out your extensions from there. |
| 13:26.22 | Pan3D | that sucks |
| 13:26.33 | pugachevcobra | whats colo? |
| 13:26.38 | Pan3D | colocation |
| 13:26.42 | linagee | pugachevcobra: a colocated server |
| 13:26.44 | Pan3D | a business that holds servers |
| 13:26.46 | pugachevcobra | ah yes |
| 13:26.54 | pugachevcobra | not feasible |
| 13:26.55 | linagee | Pan3D: the place where the internet is. :) |
| 13:27.00 | Pan3D | hehe |
| 13:27.16 | Pan3D | pugachevcobra: and you can't get more IPs from your upstream? |
| 13:27.53 | pugachevcobra | again, not feasible... |
| 13:27.57 | linagee | pugachevcobra: i would put your asterisk box connected directly to your dsl connection. that might make asterisk happy and not have to do NAT. |
| 13:28.03 | linagee | pugachevcobra: put dual NICs |
| 13:28.05 | pugachevcobra | hmmm |
| 13:28.10 | pugachevcobra | thats an idea |
| 13:28.27 | linagee | pugachevcobra: and use squid proxy or run NAT from linux for your other internet devices |
| 13:28.49 | pugachevcobra | make the asterisk box the router |
| 13:29.03 | pugachevcobra | still |
| 13:29.10 | pugachevcobra | doesnt answer why was it working a week ago |
| 13:29.22 | linagee | pugachevcobra: something changed? |
| 13:29.26 | linagee | pugachevcobra: it had to have |
| 13:29.44 | linagee | pugachevcobra: firmware in phones, asterisk version, something |
| 13:29.44 | pugachevcobra | well... yes, somehow... |
| 13:29.55 | pugachevcobra | not the asterisk version in itself |
| 13:30.03 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 13:30.07 | linagee | pugachevcobra: this is why people in real production environments get things working and throw away the key until the next patching cycle. |
| 13:30.09 | pugachevcobra | do i need to put nat=no in sip.conf, or just removing it is enough? |
| 13:30.36 | pugachevcobra | throw away the key... haha, the ssh keys also then |
| 13:30.52 | linagee | pugachevcobra: you seem to misunderstand. not the ssh key |
| 13:30.57 | pugachevcobra | i understood |
| 13:31.03 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 13:31.05 | pugachevcobra | but we could still access it through ssh |
| 13:31.09 | pugachevcobra | hehe |
| 13:31.19 | pugachevcobra | linagee: about the nat=no, does it make a difference? |
| 13:31.34 | linagee | pugachevcobra: nfi |
| 13:31.50 | linagee | pugachevcobra: asterisk does not like NATs unless you have lots of time to play with it |
| 13:32.13 | pugachevcobra | yes |
| 13:32.39 | pugachevcobra | asterisk was actually strange when the devices werent getting RTP externally... the extension had nat=yes, but it didnt work |
| 13:32.50 | pugachevcobra | only when i put nat=yes in sip.conf it began to work |
| 13:33.24 | pugachevcobra | i honestly dont know the use of the nat option in the extension peer conifg |
| 13:33.40 | pugachevcobra | it seems useless to me |
| 13:33.51 | pugachevcobra | or I might be just doing something wrong |
| 13:34.15 | linagee | pugachevcobra: read the wiki docs |
| 13:34.19 | linagee | pugachevcobra: don't guess |
| 13:34.28 | pugachevcobra | which ones? |
| 13:34.38 | pugachevcobra | asterisk.org? |
| 13:34.54 | *** join/#asterisk SparFux (n=raoul@e182029000.adsl.alicedsl.de) |
| 13:34.56 | linagee | http://www.voip-info.org/ |
| 13:35.17 | SparFux | Hi. Short question. Will asterisk 1.6 run with basically the 1.4 configuration directory, only zaptel updated to dahdi_ |
| 13:35.18 | SparFux | ? |
| 13:37.43 | *** join/#asterisk rossand (n=rossand@173.32.128.218) |
| 13:38.50 | pugachevcobra | linagee: oh joy it worked with nat=no |
| 13:39.18 | linagee | pugachevcobra: "< linagee> pugachevcobra: asterisk does not like NATs unless you have lots of time to play with it" |
| 13:41.51 | pugachevcobra | linagee: thanks for the voip-info.org, ive landed there a lot of times through google, but with the nat stuff, i can fiddle more besides yes or no |
| 13:44.34 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-21-209.acanac.net) |
| 13:46.33 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 13:49.31 | *** join/#asterisk voipmonk (n=voipmonk@67.204.21.209) |
| 13:50.10 | *** join/#asterisk hat_panda (n=peter@c-83-233-7-124.cust.bredband2.com) |
| 13:50.49 | hat_panda | hi is it legal to send invites where the request uri is not the same as the to header field? |
| 13:51.13 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 13:52.50 | hat_panda | or do you have to go to jail if you do such thing |
| 13:55.13 | Gugge | yes, that is perfectly legal |
| 14:00.28 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 14:02.32 | SparFux | With asterisk 1.6 and dahdi 2.2.0.2 I only get app_dial.c:1721 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) |
| 14:02.50 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 14:03.38 | SparFux | but the dahdi_cfg -vvv runs okk. |
| 14:05.09 | SparFux | but dahdi_scan says irq=0 |
| 14:05.15 | SparFux | the irq isn't 0. |
| 14:05.27 | SparFux | irq should be 17. |
| 14:05.33 | SparFux | according to lspci -v |
| 14:08.11 | voipmonk | hrmm |
| 14:08.20 | voipmonk | sparfux - wha kind of card do u have? |
| 14:09.09 | SparFux | HFC-S pci bri card. |
| 14:09.25 | SparFux | I am using some kind of hfc patch I got from a forum I have to admit. |
| 14:10.10 | SparFux | dahdi show channels gives me channel output, though the pseudo channel asn't configured by me and it has default context which is not correct, as my context is from-pstn. |
| 14:10.25 | SparFux | dahdi channels 1 and 2 get from-pstn though correctly. |
| 14:10.40 | SparFux | I think the whole thing goes the right way anyway, but the busy thing is broken. |
| 14:11.47 | *** join/#asterisk asif (n=chatzill@122.166.40.72) |
| 14:12.17 | SparFux | well, I have to look into this some more deeply later on. have to go for now. ... |
| 14:12.26 | asif | hello all! |
| 14:13.46 | asif | have a question about asterisk cdr. which field contains the number that has been dialled? |
| 14:15.03 | voipmonk | look at the column names |
| 14:16.36 | asif | i'm a bit confused since my cdr logs are not proper. the number that's dialled appears nowhere. |
| 14:16.36 | asif | btw i'm using call files to place calls. |
| 14:19.49 | asif | just read the description of the column names in the official documentation. the dialled number is not mentioned anywherre. |
| 14:20.43 | asif | what am i missing here? |
| 14:23.36 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 14:29.09 | voipmonk | u can add some stuff to your dialplan to make sure the dialed number gets inserted |
| 14:29.38 | *** part/#asterisk _wa1cfs (n=jhamm@80.67.64.125) |
| 14:30.43 | asif | i'm already using that, but isn't there any way asterisk cdr would log the dialed number? |
| 14:31.20 | asif | like, that should be a very obvious thing to have, right? is it a bug or a missing feature? |
| 14:31.58 | voipmonk | let me get this right - you added some stuff in the dialplan to insert the cdr - but you still arent getting it? |
| 14:32.16 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 14:32.47 | asif | i've enabled cdr logging. and i expect it to record the number to which i call. |
| 14:32.50 | asif | and it does'nt. |
| 14:33.07 | voipmonk | pastebin what you've added in the dialplan |
| 14:33.54 | asif | how do i do that? |
| 14:34.05 | voipmonk | www.pastebin.ca |
| 14:34.14 | voipmonk | copy and paste it |
| 14:34.23 | voipmonk | then pastebin spits out a url u send here |
| 14:34.31 | voipmonk | so we can click on it and look at your paste |
| 14:36.07 | asif | SetCDRUserField(${dialed_number}) |
| 14:36.30 | asif | so that i'll get the dialed number in the userfield column of cdr |
| 14:37.32 | voipmonk | using cdr mysql? |
| 14:37.41 | asif | yes |
| 14:37.56 | voipmonk | u set userfiled =1? |
| 14:38.02 | voipmonk | in the .conf? |
| 14:38.09 | asif | yes. |
| 14:38.18 | asif | this works fine. |
| 14:38.48 | asif | but i'm checking whether i can get the dialed number automatically recorded, like the other fields. |
| 14:39.17 | asif | asterisk can automatically recorded the call duration etc, right? would it record the dialed number too? |
| 14:39.20 | voipmonk | so it works the way it is but you are looking for another way, why/ |
| 14:39.24 | asif | *record |
| 14:39.31 | voipmonk | ? |
| 14:39.52 | asif | did you get it? |
| 14:40.13 | voipmonk | my cdrs work fine the dialed number is a part of the cdr |
| 14:40.31 | voipmonk | why you arent getting it beyond me without looking at your dialplan which is what I asked for when I was referring to the pastebin |
| 14:40.33 | voipmonk | see how that works? |
| 14:40.36 | asif | oh on which column? |
| 14:41.20 | voipmonk | I use jerjers rating / routing wholesale billing engine |
| 14:42.13 | asif | oh i see. so the engine takes care of managing the cdr? |
| 14:42.50 | asif | hey monk the dialplan i'm referring to is on another server that's the delay fetching it |
| 14:43.44 | voipmonk | yezzir |
| 14:43.47 | voipmonk | from multiple machines |
| 14:43.48 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 14:44.38 | voipmonk | but the cdr stuff that comes with the addons should insert the dialed number |
| 14:44.52 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:44.53 | voipmonk | thats a requirement for a cdr |
| 14:45.08 | asif | hmm yeah |
| 14:45.28 | asif | hey does your cdr data go into a mysql table? |
| 14:45.33 | voipmonk | yes |
| 14:46.16 | asif | can you tell me which column in that table contains the dialed number? |
| 14:46.36 | *** join/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com) |
| 14:47.49 | voipmonk | mine is kinda of proprietary but its the 7th column - but mine has ocn lata and ani in it |
| 14:49.08 | asif | hey check out my mysql cdr table structure - http://www.pastebin.ca/1554970 |
| 14:51.21 | asif | none of these columns contain the dialed number |
| 14:52.49 | *** join/#asterisk jimi_ (n=jimi@unaffiliated/tuxguy) |
| 14:52.56 | jimi_ | Anyone here have the skype unlimited world calling plan? |
| 14:53.03 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 14:53.12 | voipmonk | i had it for 14 days |
| 14:53.49 | voipmonk | dst?? |
| 14:55.03 | voipmonk | how many systems are you using for cdrs? |
| 14:55.11 | asif | dst contains the destination extension |
| 14:55.16 | *** join/#asterisk freakazoid0223 (n=knoppix@pool-71-246-17-206.phlapa.fios.verizon.net) |
| 14:55.20 | voipmonk | well then |
| 14:55.21 | voipmonk | there u go |
| 14:55.31 | asif | umm? |
| 14:55.39 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 14:55.57 | voipmonk | only 1 leg of the call, eh? |
| 14:56.09 | voipmonk | add a field and use mysql to insert it |
| 14:56.11 | voipmonk | and ur done |
| 14:56.13 | voipmonk | next |
| 14:56.13 | pugachevcobra | where does the voicemail email sending app stores the subject message? it isnt in vm_email |
| 14:56.14 | voipmonk | :) |
| 14:56.19 | asif | :) |
| 14:56.20 | asif | destination extension would be s, 1, 2 etc... |
| 14:56.25 | voipmonk | yeah i figured |
| 14:56.28 | voipmonk | one leg of the call |
| 14:56.35 | voipmonk | so u need to add a field |
| 14:56.41 | voipmonk | and use mysql to insert it |
| 14:56.43 | voipmonk | done |
| 14:56.43 | asif | custom cdr field? |
| 14:57.01 | jimi_ | voipmonk, is it truly free up to 10,000 minutes to call russia? |
| 14:57.05 | asif | i'd need asterisk 1.6 to use custom cdr fields, right? |
| 14:57.22 | asif | currently i'm using 1.4 and custom fields are not logged. |
| 15:01.20 | pugachevcobra | where can I change the subject message of the voicemail email app? I mean the "New message 1 in mailbox 123" |
| 15:03.03 | voipmonk | its all there |
| 15:03.08 | voipmonk | check the samplew |
| 15:03.11 | voipmonk | samples |
| 15:03.28 | voipmonk | dont be afraid of the words |
| 15:04.19 | asif | samples? i didn't get that? |
| 15:11.39 | *** join/#asterisk SirColin (n=SirColin@83.216.68.241) |
| 15:11.43 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:11.52 | *** part/#asterisk SirColin (n=SirColin@83.216.68.241) |
| 15:11.58 | *** join/#asterisk fofware (n=Fabian@190.7.25.160) |
| 15:13.29 | pugachevcobra | no one knows that? i've grepped all possible directories that I know asterisk stores stuff |
| 15:21.22 | *** join/#asterisk flohack (n=fhackenb@chello084115131198.3.graz.surfer.at) |
| 15:26.31 | voipmonk | no pugachevcobra needs to check the samples |
| 15:26.37 | voipmonk | for the vmail stuff |
| 15:27.14 | pugachevcobra | ok i found it |
| 15:27.18 | pugachevcobra | its hard compiled |
| 15:27.23 | pugachevcobra | in app_voicemail.so |
| 15:27.30 | pugachevcobra | thats a bummer |
| 15:27.40 | *** part/#asterisk asif (n=chatzill@122.166.40.72) |
| 15:28.57 | pugachevcobra | where could I submit a request for that? |
| 15:29.10 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-104-75-14.pskn.east.verizon.net) |
| 15:34.20 | pugachevcobra | oh man it seems forwarding email through ARI's web gui is messed up... |
| 15:41.47 | voipmonk | hard compiled? |
| 15:41.47 | voipmonk | no |
| 15:41.50 | voipmonk | u can change it |
| 15:41.54 | voipmonk | in the conf file |
| 15:41.59 | voipmonk | pugachevcobra: |
| 15:44.24 | *** join/#asterisk errotan (n=errotan@62.201.122.220) |
| 15:45.48 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 15:48.34 | pugachevcobra | voipmonk: no i cant |
| 15:48.45 | pugachevcobra | voipmonk: i can change the body, not the sender or the subject |
| 15:49.06 | pugachevcobra | voipmonk: and the "pager" notification, i can't even change the body |
| 15:49.09 | voipmonk | yes u can |
| 15:49.10 | pugachevcobra | voipmonk: its all hard coded |
| 15:49.15 | voipmonk | no its not |
| 15:49.23 | voipmonk | keep reading |
| 15:49.33 | voipmonk | it can all be changed in the voicemail.conf |
| 15:49.51 | pugachevcobra | voipmonk: well tell me how, but i've just seen the app_voicemail.so binary file and it has all these in engilsh |
| 15:49.52 | voipmonk | if i do it for you and wlak you through the steps i will have to activate teh spoonfeed 3.0 |
| 15:49.56 | voipmonk | i dont wanna do that |
| 15:50.11 | pugachevcobra | let me check that |
| 15:50.14 | pugachevcobra | but doesnt seem right |
| 15:50.32 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 15:51.03 | *** join/#asterisk raden_work (n=chatzill@68-191-168-32.dhcp.stpt.wi.charter.com) |
| 15:51.04 | pugachevcobra | oh its true |
| 15:52.17 | grandpapadot | ... or you could just use a notifcation script to format it exactly the way you want with perl or php or something ... |
| 15:52.37 | voipmonk | or mysql |
| 16:01.08 | *** join/#asterisk Diblo (n=chatzill@0x573fc6ce.cpe.ge-1-1-0-1101.arcnqu1.customer.tele.dk) |
| 16:01.53 | Diblo | Hey i cannot answer 3rd Parties DialPlan on call out? |
| 16:07.56 | Diblo | Asterisk not send the correct tone or a tone so I can answer. What I know. |
| 16:08.10 | Diblo | What do I know. |
| 16:10.11 | *** join/#asterisk propellerhead (n=yogurt2u@host143.190-228-233.telecom.net.ar) |
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| 16:21.32 | box2 | hmmm |
| 16:21.52 | box2 | now my MeetMe() is giving me "invalid conference number" |
| 16:22.17 | box2 | unable to open psueudo device in log |
| 16:22.49 | box2 | but MeetMeCount() gives me the correct number of users |
| 16:22.51 | jks | anyone knows a channel about fax-over-voip? |
| 16:23.19 | coppice | there's no channel. you need to fax them |
| 16:23.27 | jks | harh harh |
| 16:24.51 | jks | I'm just interested in what affects the first "beep" from the callee fax |
| 16:25.40 | *** join/#asterisk friartuck (n=pmccary@66.162.90.57) |
| 16:30.15 | box2 | ooohh zaptel timer |
| 16:30.19 | box2 | you have failed me again |
| 16:31.00 | *** join/#asterisk hat_panda (n=peter@c-83-233-7-124.cust.bredband2.com) |
| 16:31.14 | drmessano | Thats not Zaptel timer |
| 16:31.41 | drmessano | It is Zaptel, but Zaptel but Zaptel provides the mixing for meetme, not timing |
| 16:31.41 | coppice | jks: call a fax machine and it beeps. nothing much affects that |
| 16:32.13 | jks | coppice, hehe, yeah okay - I probably need to explain a bit better, just though fax-problems weren't well seen in this channel |
| 16:32.30 | *** join/#asterisk wonderworld (n=ww@mue-88-130-102-173.dsl.tropolys.de) |
| 16:33.13 | jks | faxing with G.711a (no T.38) using PAP2T works fine for me... except that in some cases the fax doesn't recognize the first "beep" that comes from the callee fax and thus keeps waiting for it to detect a fax |
| 16:33.25 | wonderworld | hey, does someone know if id'd have to put a BRI card in NT mode if i want to attach it to a ptp line from a telco? |
| 16:33.39 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 16:33.42 | jks | I'm just wondering what will affect the recognition of that beep...do I need to tweak impedance, polarity, gain, or (?) |
| 16:34.30 | coppice | jks: the called machine will keep repeating the first signal, so if it never gets recognised something bad has happened, like a G.729 codec somewhere in the path |
| 16:35.21 | jks | coppice, hmm, odd... it's definetely G.711a the whole way... I'm just trying to fax to same number every time... when it does detect a fax on the other end, it works just fine and transmits the fax without any errors |
| 16:36.15 | coppice | you have a PAP2T. what does the other end have? |
| 16:36.28 | hat_panda | jks: My biggest problem with transmitting tones over ip is echo cancelers that mess up the audio stream |
| 16:36.31 | jks | what I have noticed is that when it does detect the fax the first beep sounds "loud", but when it doesn't detect it the first beep sounds "not so loud" |
| 16:36.40 | jks | coppice, analog connection, no VoIP involved there |
| 16:37.00 | jks | hat_panda, hmm, yes, I have disabled echo cancellation in the PAP2T box |
| 16:37.37 | coppice | jks: there must be an RTP to analogue converter at the other end |
| 16:38.36 | jks | coppice, yes, the setup is: fax - PAP2T - local net server - isdn card - isdn connection to public telephone service... and at the other is regular pstn line and a regular analog fax |
| 16:40.01 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 16:40.12 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 16:42.48 | jks | hat_panda, do you know what the line polarity settings in the pap2t does? - I have noticed that if I use "forward" instead of "reverse", nothing works at all |
| 16:43.27 | jks | on a a related note: anyone knows a fax test number like the Telstra one? (i.e. you send a fax and it reports back on how it went in many details) |
| 16:43.40 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 16:44.44 | wonderworld | hey, does someone know if id'd have to put a BRI card in NT mode if i want to attach it to a ptp line from a telco? |
| 16:46.11 | hat_panda | jks: sry, i dont. If you record the fax audio stream can you hear the fax-beep you looking for? |
| 16:46.58 | jks | hat_panda, hmm, I have speakers in the fax so I can hear what happens, but I haven't actually tried recording it |
| 16:47.06 | *** join/#asterisk flohack (n=fhackenb@chello084115131198.3.graz.surfer.at) |
| 16:47.24 | hat_panda | jks: okay, so do you hear the beep from the speakers? |
| 16:47.31 | jks | hat_panda, I was just thinking if it was normal that you needed to tweak the gain settings with the pap2t to get the recognition working or something |
| 16:47.59 | jks | hat_panda, yes, however it is "really loud" in the cases where the connection works, and "not so loud" when it doesn't work |
| 16:48.42 | jks | hat_panda, I don't know if the machine amplify the sound to say that "I've found the other fax!" or it is actually louder on the line |
| 16:49.58 | jks | I have tried faxing faxtoy.net for example... works just fine, full speed transfer and the received page looks great |
| 16:58.38 | *** join/#asterisk kannan (n=kann@121.246.242.95) |
| 17:08.01 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 17:08.13 | *** join/#asterisk Mango (n=Mango@96.49.69.137) |
| 17:08.58 | Mango | How can I monitor SIP NOTIFY messages sent from my VoIP provider? I turned on verbosity and am looking at the console but I saw nothing. |
| 17:11.02 | ruben23 | hi |
| 17:11.08 | Mango | Hi |
| 17:12.23 | hat_panda | Mango: maybe you need sip set debug if your monitoring from cli |
| 17:12.37 | Mango | thx |
| 17:15.57 | Mango | Excellent!! |
| 17:17.08 | *** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com) |
| 17:17.09 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:17.22 | Mango | Ok. I want to write a plugin that monitors the SIP NOTIFY messages for the Messages-Waiting: header |
| 17:17.54 | Mango | Has this been done before, and if not, can someone give me a push in the right direction? I've been a programmer since 2002 but not with Asterisk. |
| 17:19.38 | hat_panda | Mango: Why do you want to monitor notifies? :) |
| 17:19.57 | Mango | Because my VoIP provider stores Voicemail, and I want my MWI to work :) |
| 17:27.17 | raden_work | anyone use asterisk fax gateways for inbound fax to email and outbound email to fax ? |
| 17:30.23 | Mango | Tried. |
| 17:30.28 | Mango | Nearly went insane. |
| 17:30.34 | Mango | Signed up for http://www.myfax.com/ instead. |
| 17:30.56 | Mango | More expensive, but it does not require perscription drugs. |
| 17:33.38 | *** join/#asterisk mrkiko (n=mrkiko@host12-250-dynamic.2-87-r.retail.telecomitalia.it) |
| 17:34.20 | mrkiko | What does mean the message "Calltoken support required. If unexpected, resolve by placing address..."; I can't find references to this support in the config files, at least or on google |
| 17:37.29 | *** join/#asterisk qdk (n=qdk@0x573d8d8b.bynqu1.dynamic.dsl.tele.dk) |
| 17:51.34 | raden_work | LMAO |
| 17:51.44 | raden_work | fax and sip is a pita |
| 17:51.57 | raden_work | if everyone would get out of the stone age |
| 17:56.40 | hat_panda | raden_work: you mean, if everyone would stop using obsolete fax machines |
| 17:57.05 | coppice | they can't get out of the stone age until they have easy access to an effective alternative |
| 17:57.36 | raden_work | hat_panda: exactly |
| 17:57.39 | TJNII | Of course fax and sip is a PITA. SIP isn't designed to carry data. |
| 17:57.46 | TJNII | It is designed to carry voice |
| 17:58.03 | Qwell | SIP is designed to carry signalling. |
| 17:58.04 | hat_panda | TJNII: isnt it possible to send images over sip? |
| 17:58.16 | TJNII | Qwell: Well, yea. But you know what I mean. |
| 17:58.22 | hat_panda | TJNII: like here you go i want to send you this strange mime type do you accept it ? |
| 17:58.28 | Qwell | no I don't. |
| 17:58.38 | coppice | SIP is actually designed to carry data. it just doesn't do it awfully well |
| 17:59.02 | TJNII | Okay, SIP carries signalling. The RTP carring voice is designed to carry just that, voice. Not digital data from a fax. |
| 17:59.59 | hat_panda | TJNII: thats not correct, rtp is a good carrier for video and text also |
| 18:00.17 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 18:00.19 | TJNII | hat_panda: Not when it is digitizing an audio stream, |
| 18:00.20 | pugachevcobra | raden_work: i tried it and gave up... you should look asterfax |
| 18:00.43 | raden_work | yeah i have a spa on that way but it think i try asterisk fax as well |
| 18:01.16 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-21-209.acanac.net) |
| 18:01.28 | TJNII | hat_panda: If the device he is using doesn't support fax, then it will not decode the data. It will just send a digital representation of the audio of the multiplexed signal. Which doesn't work for shit. |
| 18:02.13 | hat_panda | TJNII: yea, if you dont have a phone and want to call someone that will be a problem too |
| 18:04.14 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 18:05.02 | TJNII | Basically, what I was trying to say is faxing withou T.38 is a hack and doesn't work well. Apparently I expressed myself very, very poorly. |
| 18:05.17 | Mango | Even with T.38 it's hit-and-miss. |
| 18:05.58 | hat_panda | TJNII: yea, i was being rude too sry about that |
| 18:07.42 | *** join/#asterisk martianixor (i=martianx@gateway/shell/blinkenshell.org/x-advdkooupbdaqsua) |
| 18:08.23 | *** join/#asterisk matata (n=bassem@Wikipedia/Bassem-JARKAS) |
| 18:08.39 | *** join/#asterisk polk_ (n=polk@64-135-200-89.FoxValley.net) |
| 18:08.59 | martianixor | hi |
| 18:09.00 | polk_ | I belive my Asterisk Now is blocking port 5060 with centos.. How do I fix this? |
| 18:13.03 | *** join/#asterisk iamturnip (n=joe@S01060002553240a8.vc.shawcable.net) |
| 18:13.25 | iamturnip | Can takecall=> in followme.conf be something other than a number like * or # |
| 18:18.55 | martianixor | guys I'm behind NAT/firewall tried to configure my Asterisk 1.6.1.4, I'm not sure how to debug the issue I'm having |
| 18:19.03 | martianixor | I've enabled debug=3, verbose=3 |
| 18:19.12 | voipmonk | ~sipnat |
| 18:19.13 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:19.37 | martianixor | and I keep getting messages like Scheduling destruction of SIP dialog METHOD=REGISTER at the asterisk CLI |
| 18:20.12 | martianixor | Although the clients I'm using "twinkle, ekiga" claims that registeration is successful |
| 18:20.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:22.17 | martianixor | voipmonk: thanks :-) but is that all? |
| 18:23.00 | voipmonk | no |
| 18:23.04 | voipmonk | that is a start |
| 18:23.28 | voipmonk | where is your Asterisk system in relation to your router and phones? |
| 18:23.57 | voipmonk | how is your network setup? |
| 18:24.21 | voipmonk | is your asterisk system behind a router or does it have a public ip? |
| 18:24.24 | martianixor | It's connected directly to the router, and in the DMZ and I have dyndns |
| 18:24.31 | voipmonk | do your phones have public ips or are they behind a router |
| 18:24.37 | voipmonk | dmz |
| 18:24.43 | voipmonk | :( |
| 18:24.57 | voipmonk | do u have admin access to your router? |
| 18:25.14 | martianixor | one of the phones are at a remote place also behind dyndns |
| 18:25.31 | martianixor | the other is on asterisk's machine |
| 18:25.38 | martianixor | voipmonk: of course |
| 18:25.55 | voipmonk | then begin with this |
| 18:26.04 | martianixor | voipmonk: and sip ports logging has been enabled |
| 18:26.16 | martianixor | everything looks normal |
| 18:26.21 | voipmonk | for the asterisk system behind the router - forward ports 5060 UDP and 10001 - 20000 UDP to the internal ip of your asterisk system |
| 18:26.30 | voipmonk | get it out of the DMZ |
| 18:26.52 | martianixor | yes that's what I've done |
| 18:26.54 | *** part/#asterisk iamturnip (n=joe@S01060002553240a8.vc.shawcable.net) |
| 18:26.56 | voipmonk | then in /etc/asterisk/rtp.conf change the 10000 to 10001 |
| 18:26.58 | martianixor | voipmonk: why out of DMZ? |
| 18:27.12 | voipmonk | you said your current setup isnt working |
| 18:28.09 | matata | voipmonk: I tried the registration with martianixor, and it registerd, (I'm out of his network ) |
| 18:28.36 | matata | voipmonk: but when I call him ekiga said : user not found |
| 18:28.43 | martianixor | and when he tries to call my also registered extension it tells him user not found 404 |
| 18:29.43 | voipmonk | the sky here is very blue |
| 18:29.54 | martianixor | :-( |
| 18:29.55 | voipmonk | water tastes funny |
| 18:30.08 | voipmonk | have you made the router change yet? |
| 18:30.25 | martianixor | getting it out of DMZ you meant? |
| 18:30.37 | voipmonk | that was not the only thing I said |
| 18:30.54 | martianixor | voipmonk: actually I told you port forwarding has been done |
| 18:31.02 | martianixor | for both sip and rtp |
| 18:31.31 | *** join/#asterisk MindTheGap_ (n=MindTheG@187.20.141.72) |
| 18:31.33 | cosmicwombat | Is there known "correct" version of Asterisk/Zaptel -or- DAHDI for a TE420B |
| 18:31.59 | voipmonk | martianixor: no you have not |
| 18:32.34 | martianixor | voipmonk: OK sorry not sure where I've sent that |
| 18:33.25 | martianixor | voipmonk: so now this is why me and my friend matata got stuck, no sign for errors |
| 18:34.15 | voipmonk | so the system is out of the dmz now, 5060 is forwarded and rtp 10,001 - 20,000 udp has been forwarded and u made the change in /etc/asterisk/rtp.conf , yes/ |
| 18:34.20 | martianixor | except for things like user not found although both extensions are registered according to our clients |
| 18:34.35 | voipmonk | yes or no |
| 18:35.00 | martianixor | voipmonk: all except for DMZ, I'll brb, your help is greatly appreciated by the way :-) |
| 18:35.04 | martianixor | brb |
| 18:38.37 | *** join/#asterisk leon01 (n=leon01@ool-457c12b2.dyn.optonline.net) |
| 18:39.23 | martianixor | back |
| 18:39.27 | martianixor | out of DMZ |
| 18:40.11 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 18:40.32 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:a0bd:4fcb:613b:3b75) |
| 18:41.01 | voipmonk | great |
| 18:41.07 | voipmonk | now tell me about your phones |
| 18:41.16 | voipmonk | what kind of phones |
| 18:41.24 | voipmonk | and where are they in relation to your asterisk system |
| 18:41.50 | martianixor | our phones are soft |
| 18:42.01 | martianixor | twinkle and ekiga |
| 18:42.19 | martianixor | ekiga one with matata is remote |
| 18:42.27 | martianixor | mine is twinkle local to the asterisk machine |
| 18:42.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:43.11 | matata | voipmonk: ekiga 3.2.5 |
| 18:43.20 | martianixor | mine which is twinkle is giving me this now 2nd, fetching registrations failed: 404 Not Found |
| 18:43.37 | martianixor | my twinkle is 1.4.1 |
| 18:44.13 | martianixor | 2nd is the name of the profile |
| 18:44.51 | voipmonk | what the hell is a twinkle? |
| 18:46.24 | pugachevcobra | is a linux softphone |
| 18:46.39 | pugachevcobra | martianixor: so you can register... whats the problem then? |
| 18:46.57 | martianixor | voipmonk: http://www.twinklephone.com/ |
| 18:46.59 | matata | pugachevcobra: when I call him it said "user not found |
| 18:47.24 | pugachevcobra | matata: whats his extension number? |
| 18:47.37 | matata | pugachevcobra: 207 |
| 18:47.42 | pugachevcobra | matata: he registers with 207? |
| 18:47.45 | matata | pugachevcobra: and mine is 205 |
| 18:47.50 | pugachevcobra | matata: and you call 207? |
| 18:47.56 | martianixor | pugachevcobra: yes |
| 18:48.33 | voipmonk | ? |
| 18:48.36 | pugachevcobra | pastebin one of your calls |
| 18:48.55 | martianixor | OK |
| 18:49.16 | martianixor | pugachevcobra: awesome nick by the way ;-) |
| 18:49.25 | martianixor | pugachevcobra: you're a gamer? |
| 18:50.01 | pugachevcobra | martianixor: of sims, yes |
| 18:50.57 | martianixor | now after I got Asterisk system out of the DMZ matata can't register with his ekiga |
| 18:51.17 | pugachevcobra | are you forwarding 5060 udp? |
| 18:51.25 | martianixor | I mean matata can't register to my Asterisk now using his ekiga |
| 18:51.31 | martianixor | pugachevcobra: yes of course |
| 18:51.42 | martianixor | pugachevcobra: all the range of 5000 to 5100 |
| 18:52.00 | pugachevcobra | seems a firewall blocking issue |
| 18:52.25 | matata | martianixor: please back to the old configuration |
| 18:53.11 | martianixor | pugachevcobra: what about my softphone should it be 207@127.0.0.1 |
| 18:53.20 | voipmonk | get rid of the authentication nam |
| 18:53.21 | voipmonk | e |
| 18:53.23 | martianixor | pugachevcobra: 207 is my registered extension using twinkle |
| 18:53.23 | voipmonk | in twinkle |
| 18:53.26 | voipmonk | and register |
| 18:53.40 | pugachevcobra | martianixor: well, your softphone is running on the same asterisk box? |
| 18:53.41 | martianixor | voipmonk: authentication name? |
| 18:53.46 | martianixor | pugachevcobra: yes |
| 18:53.53 | martianixor | voipmonk: OK got it |
| 18:54.10 | pugachevcobra | martianixor: still i would recommend putting the nic ip just for testing |
| 18:54.18 | mrkiko | what was the url of the book - astbook^ |
| 18:54.24 | pugachevcobra | martianixor: as in 207@192.168.0.1 |
| 18:54.51 | martianixor | pugachevcobra: that's not a problem |
| 18:55.11 | pugachevcobra | martianixor: can you call yourself? |
| 18:55.19 | TJNII | ~book |
| 18:55.20 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:55.28 | TJNII | mrkiko: ^^^ |
| 18:55.28 | martianixor | pugachevcobra: actually I did all the Asterisk behind nat configuration, like nat=yes, localnet=x.x.x.x, externhost=dynamic |
| 18:55.57 | pugachevcobra | martianixor: externhost=dynamic ? or externhost=blabla.dyndns.com like? |
| 18:56.31 | pugachevcobra | martianixor: localnet should have the netmask, is it there? |
| 18:56.47 | martianixor | yes yes |
| 18:56.58 | pugachevcobra | martianixor: and you can call yourself? |
| 18:57.16 | martianixor | pugachevcobra: externhost=bbla.dyndns.com yes |
| 18:57.19 | martianixor | sorry my bad |
| 18:57.29 | martianixor | pugachevcobra: and localnet has the netmask with the network yes |
| 18:58.17 | pugachevcobra | can you? |
| 18:58.49 | martianixor | pugachevcobra: I couldn't call my self |
| 18:58.58 | pugachevcobra | thats a problem |
| 18:59.14 | pugachevcobra | it says user unavailable? |
| 18:59.25 | martianixor | I'm sorry I didn't see the previous times you asked me about being able to call myself |
| 18:59.33 | martianixor | pugachevcobra: no not found |
| 18:59.45 | pugachevcobra | pastebin you trying to call yourself please |
| 18:59.51 | martianixor | OK |
| 18:59.57 | TJNII | And your configs |
| 19:00.11 | pugachevcobra | yes, sip.conf extensions and such |
| 19:00.29 | martianixor | could that be a problem with dialplans? |
| 19:00.38 | martianixor | or that's just me confused? |
| 19:00.44 | martianixor | I'm pasting |
| 19:02.58 | martianixor | pugachevcobra: this is what I have from the Asterisk CLI http://pastebin.ca/1555167 |
| 19:03.49 | martianixor | I'll try to paste configuration |
| 19:06.14 | pugachevcobra | martianixor: that's all?? |
| 19:06.22 | martianixor | this is my config in sip.conf http://pastebin.ca/1555171 |
| 19:06.56 | martianixor | extensions.conf is the sample one |
| 19:08.57 | pugachevcobra | martianixor: i cant believe thats all the sip debug you got from trying to call yourself |
| 19:08.59 | martianixor | yes It's all I got |
| 19:08.59 | pugachevcobra | just 2 packets? |
| 19:09.25 | martianixor | I'll try to get it from the logs I'm not sure |
| 19:09.47 | pugachevcobra | logs are worse than sip debug |
| 19:09.48 | matata | martianixor: from where is this debug ? from asterisk or twinkle ? |
| 19:10.09 | martianixor | It's from Asterisk CLI |
| 19:10.23 | martianixor | and from twinkle the client it's as follows |
| 19:11.20 | martianixor | http://pastebin.ca/1555177 |
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| 19:15.23 | pugachevcobra | martianixor: your sip.conf file is wrong |
| 19:15.46 | martianixor | pugachevcobra: any hints? |
| 19:16.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.22 | pugachevcobra | martianixor: you have the context phones configured? |
| 19:18.31 | martianixor | pugachevcobra: where should it be? |
| 19:19.35 | pugachevcobra | extensions.conf |
| 19:19.38 | martianixor | pugachevcobra: I'll try to configure it |
| 19:19.43 | pugachevcobra | martianixor: change the context to default |
| 19:19.45 | martianixor | pugachevcobra: actually yes I know |
| 19:19.51 | martianixor | pugachevcobra: I'll reset it to default |
| 19:19.56 | martianixor | pugachevcobra: yes |
| 19:20.26 | pugachevcobra | martianixor: you should pastebin your dialplans |
| 19:20.35 | martianixor | I see |
| 19:22.33 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 19:22.35 | martianixor | pugachevcobra: I think I forgot to set it, I left the sample file |
| 19:22.52 | martianixor | pugachevcobra: would you like to see the extensions.conf sample file? |
| 19:24.39 | martianixor | I'm pastebinning |
| 19:25.39 | pugachevcobra | to make matters simple |
| 19:28.00 | *** join/#asterisk matt_d (n=matt@70.134.104.88) |
| 19:28.45 | ruben23 | hi im getting error like this on my asterisk CI----->http://pastebin.com/m41dd2194 |
| 19:28.59 | ruben23 | but still i have credits on my voip. |
| 19:29.21 | *** join/#asterisk ketema (n=ketema@turtle.ketema.net) |
| 19:29.55 | matt_d | ruben23: the host is busy |
| 19:30.03 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:a0bd:4fcb:613b:3b75) |
| 19:30.52 | matt_d | ruben23: there is nothing you can do about it |
| 19:31.41 | ruben23 | matt_d: the host is the problem..? |
| 19:32.02 | matt_d | ruben23: yes. they are not accepting the connection |
| 19:36.39 | *** join/#asterisk ketema (n=ketema@turtle.ketema.net) |
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| 20:39.07 | *** join/#asterisk dustybin (n=dustybin@174.142.38.197) |
| 20:44.34 | dustybin | would it be difficult to setup my asterisk box to answer incoming calls with this dial plan: |
| 20:44.37 | dustybin | http://paste.debian.net/45806/plain/45806 |
| 20:45.40 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:8593:79a5:b4c6:7ee9) |
| 20:46.15 | matt_d | dustybin: asterisk can do anything, no thats not hard at all. |
| 20:47.42 | dustybin | i guess i could setup asterisk to access my menu straight away if asterisk detects my cellphone number |
| 20:47.54 | dustybin | and everybody else it will do as my paste above |
| 20:49.00 | matt_d | you can have asterisk do anything you want |
| 20:49.15 | dustybin | matt_d: even clean my toilet? :P |
| 20:49.16 | matt_d | you can have it turn on your home lights, prank call your brother and order a pizza for delivery |
| 20:49.19 | matt_d | all from one call |
| 20:49.43 | matt_d | yes, it can, build a simple circuit to turn on those commercial toilet cleaners and have it activate though your dialplan. |
| 20:50.09 | matt_d | there are not that many limitations. |
| 20:50.51 | dustybin | haha |
| 20:50.52 | matt_d | especially when you mix your dialplan with AGI :) |
| 20:50.55 | dustybin | i already use X10 |
| 20:51.15 | matt_d | hehe really? there u go, it can interface with X10 :) |
| 20:51.27 | dustybin | i use Heyu |
| 20:51.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:52.19 | dustybin | when i create a menu, am i right in thinking that i will need to use asterisk festival to create a computer generated voice? |
| 20:52.24 | matt_d | dustybin: nothing about asterisk, but i totally forgot about x10. i am remodeling my home. i think i will use x10 :) thanks for reminding me. |
| 20:52.50 | matt_d | dustybin: you can. would you happen to be running asterisk on a mac? |
| 20:53.04 | *** join/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com) |
| 20:53.04 | matt_d | if so, i think the "say" mac os x voice generator sounds more realistic. |
| 20:53.28 | dustybin | matt_d: here is my howto: http://www.thinkdebian.org/archives/52 |
| 20:54.07 | matt_d | dustybin: when i develop systems, i use "say" and make gsm files to use. festival can do the same, so festival isn't called each time (take a bit of a load off the processor -- if needed) |
| 20:54.08 | dustybin | no i use debian |
| 20:54.21 | dustybin | i see |
| 20:54.32 | matt_d | thanks for the howto, im going to bookmark that to read |
| 20:54.39 | dustybin | :) |
| 20:54.49 | matt_d | i want to be able to control all rooms from me and my wifes iphone. |
| 20:55.08 | dustybin | it isnt too hard to setup |
| 20:55.46 | matt_d | dustybin: in my experience, festival works just fine. the processing cost isn't that big of a deal as long as the system ins't used that much. if not it better to use festival to generate the audio file. |
| 20:55.53 | dustybin | matt_d: why are you making gsm files? those are low quality? why not create ulaw files? |
| 20:56.42 | matt_d | dustybin: gsm or ulaw works fine. i last used it with gsm just to save bandwidth. |
| 20:56.50 | dustybin | ok |
| 20:56.56 | dustybin | my system doesnt get used much |
| 20:57.05 | matt_d | i can't tell the difference (in my left ear, since i have damage) hehe. |
| 20:57.15 | dustybin | one day i might setup a conference with 4 people talking, im sure ulaw and a pentium 3.2 will be more than enough |
| 20:57.26 | dustybin | ok |
| 20:57.30 | matt_d | x10 is all ip based right? |
| 20:57.42 | dustybin | x10 uses x10 |
| 20:57.53 | dustybin | like |
| 20:57.55 | dustybin | heyu on A2 |
| 20:58.00 | dustybin | A2 = my light |
| 20:58.04 | dustybin | heyu off A2 |
| 20:58.33 | matt_d | dustybin: yes. that should be more than enough. |
| 20:59.06 | dustybin | what is the best way to access a menu system? |
| 20:59.18 | matt_d | dustybin: you mean program a menu system? |
| 20:59.22 | dustybin | yes |
| 20:59.35 | matt_d | the easiest way is to set it up in extentions.conf |
| 20:59.39 | dustybin | ok |
| 20:59.47 | matt_d | if you want to use your favorite programming lanauge, you can use AGI and/or FastAGI |
| 20:59.59 | dustybin | ok |
| 21:00.39 | matt_d | that opens it up to any programming language you want; The C languages, Perl, Python, (the best in the whole world) Ruby, Java, JavaScript, COBOL --- anything! :) voip-info.org has a lot of info about it. |
| 21:00.41 | dustybin | i love the way IAX2 only uses 1 port for everything |
| 21:00.47 | dustybin | SIP uses shit loads of ports |
| 21:01.01 | dustybin | ok |
| 21:01.04 | matt_d | yes, IAX2 is nice. |
| 21:01.17 | dustybin | are there many hardware IAX2 phones out there? |
| 21:01.46 | matt_d | yes, there is a good amount. |
| 21:02.06 | dustybin | i dont think polycoms are IAX2 compatible |
| 21:02.52 | matt_d | there is a list over on voip-info.org (i'm sure others as well) |
| 21:02.55 | dustybin | ok |
| 21:03.22 | dustybin | there is a decent iax2 softphone, called efisk |
| 21:03.27 | dustybin | something like that |
| 21:03.49 | dustybin | IDEFISK |
| 21:03.52 | matt_d | i haven't tried that one yet. i used xten, but don't like it. in fact, i don't think it supports iax ... |
| 21:04.02 | voipmonk | idefisk = zoiper |
| 21:04.21 | dustybin | ohh |
| 21:04.22 | *** join/#asterisk shinao1 (n=shinao1@41.219.221.128) |
| 21:04.37 | matt_d | i tried zoiper but it was hard to use |
| 21:04.43 | matt_d | and you have to sign up for their service |
| 21:04.59 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 21:05.14 | dustybin | ok |
| 21:05.51 | matt_d | i only suggest that because voip-info.org is wiki based and info is always being added/updated on there. |
| 21:06.05 | dustybin | zoiper comes with a G729 license |
| 21:06.26 | dustybin | what is the point of me using a G729 codec, will people from the outside be using that codec? |
| 21:06.44 | dustybin | are codecs only beneficial for the internal network |
| 21:07.39 | dustybin | or is audio voip data always raw, and the codecs at the other ends of the telephone convert the signals to whatever you want? |
| 21:07.54 | matt_d | ulaw is the best you will get for the outside |
| 21:08.07 | dustybin | yep i thought so |
| 21:08.20 | matt_d | and should stick with one to prevent transcoding |
| 21:08.27 | matt_d | just to save some processor cost |
| 21:08.28 | dustybin | if i had a internal network of phones, with 50 users, then i could improve the sound quality |
| 21:08.52 | dustybin | but from the outside, ulaw is the best i will get, unless telephone companies change there systems |
| 21:09.11 | dustybin | ok |
| 21:09.52 | matt_d | dustybin: exactly. so its best to stick with ulaw if most calls are accessing the outside. so you don't have to do any transcoding. |
| 21:10.02 | dustybin | excellent |
| 21:10.18 | dustybin | all i need to do now, is get my friends to use IAX2 softphones, so they can call me for free |
| 21:10.43 | dustybin | i look forward to getting 3 of my friends into a conference call :D |
| 21:10.52 | dustybin | and watch my CPU hit 200% :P |
| 21:11.45 | matt_d | you will be surprised how much it can handle. u said 3.2ghz? |
| 21:11.52 | dustybin | yes |
| 21:11.57 | dustybin | i should be ok |
| 21:12.15 | matt_d | i found an article about someone who benchmakred asterisk. with a 2 something ghz procesor and 2gb memory handled 200 calls at once with no problem. |
| 21:12.42 | dustybin | jeeeeeeeeze |
| 21:13.46 | matt_d | and a duel xeon handled about 1500 |
| 21:15.51 | matt_d | right now i'm developing a system that has the potential of handling about 200 calls pretty much all day long. i'm going to split it up using three or four servers, just in case one goes down. thinking about using opensips for the load balancing |
| 21:16.06 | dustybin | aye, redundancy |
| 21:16.38 | dustybin | i guess one would need to use macros when you are using that many channels |
| 21:16.41 | matt_d | havent used opensips before, from what i hear its easy. |
| 21:17.12 | *** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus) |
| 21:17.19 | matt_d | its going to use OpenAGI to handle the dialplan. working on the OpenAGI system as we speak. all in Ruby, man I love ruby :) |
| 21:17.26 | dustybin | nice |
| 21:17.35 | dustybin | its time to fiddle with my dial plan |
| 21:18.44 | matt_d | I used to program professionally full time, but got bored of it. now i do it part time, as I started a business with my friend. still didn't have any fun doing it. until i learned ruby earlier this year. now its fun again :) |
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| 21:50.29 | dustybin | does festival need to be compiled as a option? i cannot locate the binary, however, i do have a festival.conf |
| 21:52.11 | matt_d | no, its seperate. |
| 21:52.18 | dustybin | ohhhh |
| 21:52.18 | matt_d | separate |
| 21:52.29 | matt_d | apt-get install festival --or-- yum install festival |
| 21:52.38 | dustybin | festival - General multi-lingual speech synthesis system |
| 21:52.42 | matt_d | if you don't want to compile from source. |
| 21:52.42 | dustybin | thanks |
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| 22:07.52 | *** part/#asterisk socotra (n=ab@aktaha.plus.com) |
| 22:09.57 | Naikrovek | needs to find a good console font for BitchX |
| 22:10.03 | Naikrovek | on windows... |
| 22:10.05 | Naikrovek | hrm. |
| 22:27.48 | dustybin | i now have communications between asterisk and festival, however, my x-lite phone gives this error |
| 22:27.51 | dustybin | Maximum retries exceeded on transmission |
| 22:27.57 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 22:31.08 | box2 | anyone use ztxen? |
| 22:31.53 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 22:32.10 | Naikrovek | box2: not here |
| 22:35.08 | box2 | my install of zaptel (ztdummy) on a xen VM is going very poorly is why i ask |
| 22:35.49 | box2 | wondering if it's worth the time to modify ztdummy or try ztxen |
| 22:35.53 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
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| 22:47.54 | voipmonk | heheh |
| 22:50.46 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 22:55.49 | dustybin | jeeze, festival is a b***** to setup |
| 22:56.03 | dustybin | if i ring from outside i get: exceptionally long voice queue length queuing to |
| 22:56.15 | dustybin | if i ring internal i get: Maximum retries exceeded on transmission |
| 22:56.45 | Naikrovek | tries to think of a 6 letter swear word that starts with b... |
| 22:57.03 | [TK]D-Fender | BORING |
| 22:57.06 | Naikrovek | lol |
| 22:57.23 | Naikrovek | .... lol more |
| 22:57.53 | dustybin | is this a codec problem? |
| 23:00.38 | dustybin | channel.c:1037 __ast_queue_frame: Exceptionally long voice queue length queuing to <- repeated 10000x |
| 23:02.25 | dustybin | i think it has done that because it crashed |
| 23:11.06 | zamba | decent sip client for linux? |
| 23:11.12 | zamba | ekiga just doesn't do it for me |
| 23:14.42 | TJNII | Aah, gotta love the USPS. I have a letter in my hand postmarked August 12th.... |
| 23:17.11 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 23:19.48 | lesouvage | zamba: zoiper, iax2 and sip client |
| 23:20.45 | *** part/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com) |
| 23:20.53 | zamba | lesouvage: i've heard about zoiper.. let's give that a try |
| 23:27.31 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
| 23:27.33 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 23:27.49 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:33.52 | dustybin | asterisk + festival communication = OK |
| 23:34.12 | dustybin | asterisk festival(hello) = creates .wav in cache = OK |
| 23:34.47 | dustybin | the problem seems to occur when asterisk trys to send that .wav to my softphone |
| 23:35.04 | dustybin | so that could be a codec problem! |
| 23:37.56 | dustybin | nope |
| 23:37.56 | dustybin | Maximum retries exceeded on transmission NDI1NDhhMGEyMGRmMWY0MGRkYjI4Njc5MWExYTYzMWM |
| 23:46.44 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-64-34.kgldga.dsl.dynamic.tds.net) |
| 23:54.31 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:55.24 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |