| 00:04.37 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 00:10.36 | Dr-Linux | fuck israel |
| 00:11.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:14.21 | Nivex | fuck war |
| 00:14.51 | hads | So how about that Asterisk... |
| 00:15.13 | robl^ | Asterisk?!?!?1 what's that? |
| 00:15.22 | KDan | it's everything |
| 00:15.24 | KDan | or so the manual says |
| 00:15.56 | Dr-Linux | Nivex: this is not a war |
| 00:16.06 | KDan | apparently there's a coffee.so module that you can load which starts your coffee machine when you dial in, and a kitchen_sink.so which opens your tap |
| 00:16.46 | KDan | however the first one is a bit buggy so occasionally the codec for your coffee is tasteless |
| 00:17.06 | Nivex | right, we haven't had an honest to $DEITY war in, how long? |
| 00:17.17 | Nivex | When is the last time one soverign nation declared war on another? |
| 00:17.20 | Nivex | WWII? Korea? |
| 00:17.36 | Nivex | We need a good old fashioned "You said what about my mother?" war! |
| 00:18.00 | Nivex | war on drugs, war on terror |
| 00:18.00 | KDan | Nivex: oh, all right then, go on and have it, but be back by tea time. |
| 00:18.01 | Nivex | what bullshit |
| 00:18.12 | Nivex | if you're gonna do it, DO IT RIGHT! |
| 00:18.21 | KDan | what are you suggesting? |
| 00:18.38 | Nivex | </rant> |
| 00:18.43 | Dr-Linux | Nivex: you from Israel? |
| 00:18.55 | Dovid | ~seen dovid |
| 00:19.06 | jbot | dovid is currently on #asterisk (15h 17m 34s). Has said a total of 1 messages. Is idling for 11s, last said: '~seen dovid'. |
| 00:19.06 | Nivex | Dr-Linux: nope. |
| 00:19.12 | Dovid | Dr-Linux: I am |
| 00:19.17 | *** join/#asterisk riddlebox (n=james@206.80.73.2) |
| 00:19.43 | Dovid | can the playback feature use .wav or is it better to convert it to GSM |
| 00:19.44 | Dovid | ? |
| 00:19.52 | Dr-Linux | Dovid: what you guys want from labenon? |
| 00:20.05 | Dr-Linux | Dovid: it works |
| 00:20.11 | Dovid | Dr-Linux: Nothing from them |
| 00:20.19 | Dovid | we want chizbulah out of there |
| 00:20.26 | KDan | Dr-Linux: I suspect that it's actually stuff that they don't want from them |
| 00:20.26 | rob0 | http://www.pixelscapes.com/spatulacity/button.htm |
| 00:20.29 | KDan | i.e. rockets |
| 00:20.31 | Dr-Linux | Dovid: then why killing them? |
| 00:20.42 | KDan | Dr-Linux: where are you from? |
| 00:21.02 | hads | So! How about that Asterisk... |
| 00:21.03 | Dovid | what would the US do if one day Mexico started bombing the US over the border ? How long would it take them to react ? |
| 00:21.15 | Dr-Linux | Dovid: you guys are calling a big war .. and war is not a good |
| 00:21.24 | Dr-Linux | KDan: i'm from Pakistan |
| 00:21.26 | Dovid | Israel dosent wana kill civilians. they droped leaflets number of times to tell them to leave |
| 00:21.28 | Nivex | Dovid: 'bout 2 seconds. then mexico wouldn't be there anymore. |
| 00:21.48 | Dovid | Dr-Linux: no one wants war. but what option do they have ? |
| 00:21.48 | rob0 | War sucks. |
| 00:21.54 | Dr-Linux | Dovid: why chizbulah? |
| 00:22.03 | rob0 | Big red buttons are cool. |
| 00:22.08 | Dovid | Nivex: thats the point. but if Israel defends its self its bad, however the US can and its good |
| 00:22.11 | Nivex | hads: it rules! |
| 00:22.14 | Dovid | docelmo: cause they cause trouble |
| 00:22.14 | KDan | Dr-Linux: because they bomb israel? |
| 00:22.23 | Nivex | anyone here messed with Pound Key? |
| 00:22.46 | Dovid | Nivex: topic change ? |
| 00:22.56 | burnproof | :p |
| 00:23.04 | Nivex | Dovid: topic resync :) |
| 00:23.14 | Dr-Linux | KDan: NO, bcoz they have nothing .. |
| 00:23.26 | Dovid | ok |
| 00:23.40 | Dovid | can i use background to play .wav or is it better to convert it to GSM ? |
| 00:23.42 | KDan | Dr-Linux: why would israel bomb hizbollah because they have nothing? that doesn't make sense to me, Dr-Linux. |
| 00:23.49 | [hC] | Wow the polycom ip430 boots fast |
| 00:23.49 | Dr-Linux | Dovid: did you see the 2nd tape from Alqueda? "Holly war against israel"? |
| 00:23.52 | Dovid | alison just did work for a client of mine |
| 00:24.01 | Dovid | no Doc I didnt |
| 00:24.14 | burnproof | Dovid: it slinear wav file i guess not just plain wav file? am i correct ? :) |
| 00:24.21 | Dovid | not sure |
| 00:24.39 | Dr-Linux | KDan: don't you think isreal is killing innocent people and kids as well? |
| 00:24.42 | Dovid | but i think i am gona convert cause it's easier for orginizational purposes |
| 00:24.48 | Dr-Linux | where is America now? |
| 00:24.51 | Dovid | whats the command to convert em over ? |
| 00:24.53 | KDan | Dr-Linux: oh yes, they are. http://www.news.com.au/heraldsun/story/0,21985,20003158-661,00.html << this is why they're doing it |
| 00:25.35 | *** join/#asterisk Trazz (n=traderz@c-67-163-92-37.hsd1.il.comcast.net) |
| 00:25.36 | burnproof | Guys, could we just change the topic of this war, since no one wins the war |
| 00:25.47 | Dr-Linux | KDan: Israel has all hand on America, Media etc as well. |
| 00:26.05 | KDan | Dr-Linux: see pvt |
| 00:26.08 | Dovid | Dr-Linux: THEY DONT MEAN TO DO IT. THEY ASKED THE CIVILIANS TO LEAVE. They are targeting the terorists. on the other HAND CHIZBULAH IS TARGETING CIVILIANS !!!! |
| 00:26.09 | sevard | To err is human -- but it feels divine. -- Mae West |
| 00:26.23 | Dovid | ok. enough with my rant here |
| 00:26.37 | Dr-Linux | Dovid: those are all politics man .. |
| 00:26.56 | Dr-Linux | so just leave , i just wanted to tell you, you guys are calling big war |
| 00:27.17 | sevard | king, for your mother country. |
| 00:27.22 | KDan | Dr-Linux: pls respond to pvt |
| 00:27.32 | KDan | these guys don't want us to discuss politics here, clearly |
| 00:27.39 | KDan | and it's their right since this is not a politics channel |
| 00:27.50 | KDan | I'm very much willing to debate this topic with you, so let's take it off the chan |
| 00:28.20 | Dovid | ok |
| 00:28.32 | Dovid | so now how do i convert .wav to gsm ? |
| 00:28.39 | burnproof | Dovid: i think if you are using a trunk there is a convert command there |
| 00:28.48 | [TK]D-Fender | [hC] : Told it was"zippy" :) |
| 00:28.48 | burnproof | on the console |
| 00:28.53 | Dovid | i know. forgot the command |
| 00:28.58 | Dovid | something wit the word sox in it |
| 00:29.04 | Dovid | too tired. been working days on end |
| 00:29.58 | burnproof | Dovid: sox foo.wav -r 8000 foo.gsm resample -ql |
| 00:30.39 | Dovid | thanks |
| 00:30.43 | burnproof | np |
| 00:31.29 | Snake-Eyes | Any one had any experience with using softhangup() ? |
| 00:32.22 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 00:33.40 | Snake-Eyes | I have tried different options and it doesnt seem to hang up a call |
| 00:38.36 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.226.132.telnor.net) |
| 00:41.17 | *** join/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 00:42.46 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:43.23 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 00:44.03 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 00:44.07 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 00:44.19 | CrashHD | hola |
| 00:58.04 | angom_h | hello, somebody hear using an HP DL server with digium E1/T1 cards ? |
| 00:58.12 | angom_h | *here |
| 00:59.56 | *** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net) |
| 01:00.35 | babyju | anyone here using asterisk with dynamic public ip address? |
| 01:00.54 | *** join/#asterisk rene- (n=rene-@200.93.193.198) |
| 01:01.08 | angom_h | babyju: not now but I have done it |
| 01:01.21 | rene- | hey |
| 01:03.49 | rene- | a quick question: in an agentlogin scenario, say i got a call connected to the agent (B) originated by (a) the agent can finish the call by means of hitting * in their phones, i would like to end the call via a hangup, i cant hangup the agent channel since that means the agent logs out of the queue. if i hangup (a) channel does the agent is sent back to wait for calls? |
| 01:04.56 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 01:07.12 | carrar | rene-, thats a settable time |
| 01:07.13 | *** join/#asterisk truz_`24 (n=truz_`24@74.129.166.232) |
| 01:08.03 | carrar | wrapuptime |
| 01:09.15 | carrar | or your question isn't clear |
| 01:11.48 | *** join/#asterisk Trakkasure (n=Sgemtum@24-50-26-239.atlsfl.adelphia.net) |
| 01:21.22 | *** join/#asterisk knarfly (n=bmorris@c-69-180-98-189.hsd1.fl.comcast.net) |
| 01:26.11 | CrashHD | what codec uses the least amount of bandwidth and would work the best over a cellular ip connection? |
| 01:27.09 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 01:30.25 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 01:31.02 | *** join/#asterisk tempest1 (n=asf@adsl-144-35-31.chs.bellsouth.net) |
| 01:31.52 | rene- | carrar: i am sorry what i mean is: agent logged via agent_login gets call, if i hangup (via manager interfase or cli) the caller channel, will the agent (called channel) be cleanly sent to listen to music on hold ?? |
| 01:36.24 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 01:39.01 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 01:39.05 | n9urk | can anyone help with one way audio problems? |
| 01:39.29 | n9urk | idefisk ext -> * -> idefisk ext works fine |
| 01:39.51 | n9urk | teliax -> * -> idefisk ext does not work |
| 01:40.04 | n9urk | teliax -> * -> teliax works |
| 01:40.08 | n9urk | any idears? |
| 01:40.10 | n9urk | thanks |
| 01:40.34 | Delta239 | hey a question |
| 01:40.45 | Delta239 | where can i find an AGI? |
| 01:41.31 | websae | asterlink :) |
| 01:42.00 | Delta239 | yeah but |
| 01:42.11 | Delta239 | if i have like VD_amd_post.agi |
| 01:42.19 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 01:42.27 | *** join/#asterisk mvdk (i=mvdk@60-240-70-160-nsw-pppoe.tpgi.com.au) |
| 01:42.35 | Delta239 | i find them :D thanks |
| 01:45.22 | mvdk | Does anyone here have any clue how to use the PPPD command in the dialplan to establish an external connection? |
| 01:46.19 | mvdk | (Or any other mechanism) |
| 01:48.45 | mvdk | The http://www.voip-info.org/wiki/index.php?page=Asterisk%20cmd%20ZapRAS page states about server side, but says nothing about client side :( |
| 01:50.24 | n9urk | I have a serious problem, can someone help? I am having one way audio problems and the ports have been forwarded in the router |
| 01:50.43 | *** join/#asterisk ikey (i=ikey@220.226.51.130) |
| 01:52.55 | *** join/#asterisk Skarmeth (n=Skarmeth@201008252201.user.veloxzone.com.br) |
| 01:53.48 | mvdk | n9urk: Are you certain the problem isn't on the other end? |
| 01:54.02 | mvdk | It really all depends..... |
| 01:54.11 | n9urk | mvdk: It is baffleing me |
| 01:54.22 | n9urk | <PROTECTED> |
| 01:54.26 | mvdk | What are you actually attempting to do, are you using SIP or IAX, or what? |
| 01:54.32 | n9urk | IAX |
| 01:54.35 | mvdk | Ah, OK |
| 01:54.46 | n9urk | is there some problems with that? |
| 01:55.31 | n9urk | Ok, I am trying to have POTS to ring 1 of 2 extensions |
| 01:55.36 | mvdk | OK, you're saying that idefisk -> * -> idefisk works fine (over your internet link) |
| 01:55.39 | mvdk | Or what? |
| 01:55.41 | n9urk | right |
| 01:55.56 | mvdk | And what do you want to work, ASCII is good here :) |
| 01:56.11 | n9urk | so say I dial extension 200 from one idefisk ext I get 2way audio |
| 01:56.18 | mvdk | idefisk -> * -> Digium Card -> POTS? |
| 01:56.32 | n9urk | asterisk is on a server |
| 01:56.39 | n9urk | at hurricane electric |
| 01:56.51 | mvdk | Sorry, hurricane electric is what? |
| 01:56.59 | mvdk | I'm sorry, I'm in AU |
| 01:57.13 | n9urk | oh ok it is a data center |
| 01:57.22 | mvdk | Oh, OK, that makes sense |
| 01:57.55 | mvdk | What facilities do you have in this box? |
| 01:58.14 | mvdk | No doubt you *don't* have a PRI or other access mechanism in this box.... |
| 01:58.31 | n9urk | what is a pri? |
| 01:58.48 | n9urk | I called another DID which accesses teliax with sip and I get 2way audio |
| 01:58.50 | mvdk | Primary Rate Interface, think of it as a set of 31 phone wires |
| 01:59.00 | mvdk | OK, that makes sense |
| 01:59.25 | _Vile | 23 usable in the states |
| 01:59.36 | mvdk | My mistake, only 30 are usable here |
| 01:59.43 | mvdk | 1 is used for signalling :) |
| 02:00.10 | n9urk | Any ideas on what I can do? |
| 02:00.19 | mvdk | Anyway, n9urk, what is the ascii diagram of what *doesn't* work? |
| 02:00.36 | mvdk | idefisk -> * -> SIP? |
| 02:01.10 | n9urk | Pots -> teliax -> iax -> * -> iax -> idefix == one way audio |
| 02:01.15 | nextime | in usa PRI are over T1, 23+1 channels , in japan is over J1, i think 23+1 channels too ( but i'mnt sure ), in EU and others are over E1, 30+1 channels |
| 02:01.35 | mvdk | nextime: I am aware of this.... |
| 02:02.05 | n9urk | pots -> teliax -> sip -> * -> iax -> idefisk == two way audio |
| 02:02.11 | n9urk | (which is on another did) |
| 02:02.22 | mvdk | n9urk: Oh, I see..... |
| 02:02.49 | mvdk | asterisk is trying to bridge itself out of the connection, but something isn't working |
| 02:02.55 | n9urk | idefisk -> iax -> * -> iax -> idefisk works |
| 02:03.22 | mvdk | Tell me, if you put canreinvite=no in the teliax side, does it work? |
| 02:03.26 | n9urk | is there a "canreinvite" in IAX? |
| 02:03.35 | mvdk | I believe there is |
| 02:04.04 | CrashHD | notransfer |
| 02:04.07 | mvdk | Sorry, notransfer |
| 02:04.11 | mvdk | Just looked it up |
| 02:04.32 | mvdk | Yeah, notransfer=yes in the teliax iax.conf declaration |
| 02:04.40 | n9urk | will try |
| 02:06.09 | n9urk | trying |
| 02:06.18 | n9urk | let's see if Jeremy answers |
| 02:06.24 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 02:07.04 | mvdk | Did it work? |
| 02:07.39 | n9urk | mvdk: jeremy didn't answer |
| 02:08.15 | mvdk | Ah, so it signalled, but he didn't pick up the phone? |
| 02:08.33 | n9urk | it was signalling ealier |
| 02:08.43 | n9urk | He could answer then we would only get one way audio |
| 02:09.03 | mvdk | What I'm saying is, can you confirm that he didn't pick it up? |
| 02:09.50 | n9urk | not just yet mvdk |
| 02:10.00 | n9urk | in about 60 seconds i will know |
| 02:10.08 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:10.08 | nextime | any news about the release of * 1.4? |
| 02:10.08 | mvdk | OK, cool |
| 02:10.20 | mvdk | I've been wondering that myself, nextime.... |
| 02:10.46 | Qwell | Has anything been posted on the many mailing lists? Does asterisk.org say anything? |
| 02:10.47 | mvdk | I got the impression it was going to be released last month.... |
| 02:10.49 | n9urk | maybe a little longer than 60 seconds |
| 02:10.50 | Qwell | No, so please stop asking :P |
| 02:11.02 | Qwell | it's already been said "When it's ready, and no sooner" |
| 02:11.15 | mvdk | Ah, OK |
| 02:11.25 | mvdk | Standard release schedule, in other words :) |
| 02:11.40 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 02:12.01 | mvdk | Pity that that's not going to include the codec negotiation patch :( |
| 02:12.19 | budmang | whats a good toll free voip provider? |
| 02:12.26 | n9urk | exgn.net |
| 02:12.28 | budmang | i just want to prepay 5-10 bucks to play around |
| 02:12.30 | n9urk | teliax.com |
| 02:12.44 | budmang | teliax |
| 02:12.44 | mvdk | All kinds, budmang |
| 02:12.56 | mvdk | However, none of those work well for me |
| 02:12.59 | budmang | just a random toll free. |
| 02:13.08 | n9urk | exgn.com is pretty cheap |
| 02:13.08 | budmang | teliax ive never had any issues |
| 02:13.13 | budmang | I heard their sip is bad |
| 02:13.15 | budmang | i use iAx |
| 02:13.17 | mvdk | I like my voice transmitted without bouncing halfway around the world and back..... |
| 02:13.22 | nextime | Qwell : right, sorry but on the site there something about a "latest july release of 1.4", and i come back just today from holydays, so i was hoping that it was ready and very close to the release date |
| 02:13.31 | n9urk | notransfer=yes solved the problems |
| 02:13.35 | n9urk | thanks guys!!!!!! |
| 02:13.37 | budmang | exgn.com doesnt load |
| 02:13.49 | n9urk | sorry budmang exgn.net |
| 02:13.53 | mvdk | n9urk: That means that your NAT is incorrectly set up |
| 02:14.11 | mvdk | Usually, you should be able to convince it to work.... |
| 02:14.12 | n9urk | mvdk: any suggetsions? |
| 02:14.27 | mvdk | What is your router? |
| 02:14.49 | n9urk | wrt54g |
| 02:14.53 | n9urk | linksys |
| 02:14.55 | mvdk | It could just be that your router is a PoS, in which case you'll just have to put up with it.... |
| 02:15.02 | mvdk | Ah, No, maybe not |
| 02:15.18 | mvdk | Is that the Linux or VxWorks model? |
| 02:15.36 | n9urk | don't know what you mean |
| 02:15.46 | mvdk | If it's the VxWorks model, it's definitely a PoS |
| 02:15.57 | n9urk | what is a vxworks model? |
| 02:16.11 | mvdk | It refers to the OS on the router |
| 02:16.22 | n9urk | how would I determine that? |
| 02:16.28 | mvdk | When did you buy it? |
| 02:16.33 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 02:16.35 | n9urk | ca mar 2005 |
| 02:16.56 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 02:18.20 | budmang | im looking for a $5 prepay toll free |
| 02:18.20 | mvdk | What's the Serial Number prefix? |
| 02:18.24 | budmang | no setup fees. |
| 02:18.30 | budmang | just something to call and play with for a bit |
| 02:18.43 | n9urk | cdf8 |
| 02:19.29 | mvdk | OK, it's good |
| 02:19.41 | mvdk | If it were cdfb, OTOH.... |
| 02:20.06 | mvdk | Hrm.... |
| 02:20.31 | mvdk | And you're forwarding all relevant ports to your internal hosts? |
| 02:20.40 | n9urk | mvdk: yes |
| 02:20.46 | n9urk | 10k to 20k |
| 02:20.48 | n9urk | 1020 |
| 02:20.50 | n9urk | 5060 |
| 02:20.51 | n9urk | right? |
| 02:21.10 | mvdk | Ah, you're forwarding the RTP ports, I don't recall IAX working that way.... |
| 02:21.12 | Qwell | 1020? |
| 02:21.40 | n9urk | am I missing some portS? |
| 02:22.14 | Qwell | are you having trouble getting incoming calls? |
| 02:22.31 | mvdk | 4569 |
| 02:22.53 | n9urk | Qwell: one way audio |
| 02:22.57 | Qwell | with iax? |
| 02:23.14 | Qwell | budmang: I like nufone and asterlink, personally |
| 02:23.19 | *** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net) |
| 02:23.27 | sudhir492 | hi |
| 02:23.30 | mvdk | Qwell: It sets up connections with a remote asterisk host just fine, but when it tries to transfer, he gets one way audio |
| 02:23.34 | budmang | asterlink is hell. |
| 02:23.35 | budmang | :-) |
| 02:23.41 | mvdk | Setting notransfer=yes makes everything work |
| 02:23.52 | sudhir492 | on my asterisk console, I constantly see -- Remote UNIX connection |
| 02:23.52 | sudhir492 | <PROTECTED> |
| 02:24.04 | Qwell | sudhir492: tell your admin to stop connecting to the console |
| 02:24.30 | sudhir492 | admin who? |
| 02:24.33 | mvdk | I suspect that forwarding 4569 with notransfer=no will leave everything working while taking his asterisk server out of the media path.... |
| 02:24.35 | Qwell | you tell me |
| 02:25.49 | sudhir492 | I starte the asterisk, by safe_asterisk, and in the same window occasionally I do "asterisk -vvvr" to check some things |
| 02:26.21 | Qwell | sudhir492: well...uhh...yeah... |
| 02:26.26 | Qwell | it's...supposed to...do that |
| 02:26.37 | mvdk | n9urk: Did you get that? |
| 02:26.39 | sudhir492 | :-) |
| 02:26.44 | n9urk | mvdk: yeah |
| 02:26.45 | n9urk | thanks |
| 02:26.55 | n9urk | I have about 1001 things going on at once |
| 02:26.57 | n9urk | ;) |
| 02:27.08 | mvdk | It should reduce the delay in the media path |
| 02:27.19 | mvdk | Which is somewhat important |
| 02:27.19 | budmang | what about 8/month for unlimiting incoming ? |
| 02:27.25 | budmang | i know i saw a site with that |
| 02:27.26 | budmang | :-) |
| 02:27.51 | sudhir492 | Qwell: where is the remote UNIX connection? there is not remote machine involved. Besides, I do not see such message on other asterisk boxes. |
| 02:28.19 | sudhir492 | sure would like to know what that means |
| 02:28.27 | *** join/#asterisk evilbit (i=hhoffman@gateway/tor/x-f0f6eb297498ae36) |
| 02:28.39 | *** join/#asterisk angom_h (n=Angel@red-corp-200.76.226.132.telnor.net) |
| 02:29.10 | mvdk | I know that for me, calling through a US server to get to a UK server without a transfer would introduce intolerable lag.... |
| 02:29.42 | evilbit | hi, it's been a little bit since I've messed with asterisk... I've just upgraded to the latest version and configs seemed to move across just fine... I am getting " Operating with different codecs 2[0x2 (gsm)] 4[0x4 (ulaw)] , can't native bridge..." Should I just force a particular codec? |
| 02:29.45 | sudhir492 | Qwell: Is there a way to locate from which IP the remote connection is happening? |
| 02:30.15 | mvdk | evilbit: That means it's transcoding |
| 02:30.25 | Qwell | sudhir492: always localhost |
| 02:30.28 | mvdk | That may or may not matter to you |
| 02:30.43 | evilbit | mvdk: ah, ok... that's rather cpu intensive, correct? |
| 02:30.52 | mvdk | Depends on the codec |
| 02:31.08 | evilbit | fair enough |
| 02:31.12 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 02:31.13 | mvdk | In your case, it's taking gsm to ulaw, that's not terrible |
| 02:31.30 | mvdk | If you're handling 50 calls, it adds up, though :) |
| 02:31.35 | evilbit | I suppose I should start going through all of the doco again |
| 02:31.46 | Qwell | ~book |
| 02:31.48 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:31.58 | evilbit | mvdk: nah, it won't be much... for my consulting business |
| 02:32.01 | mvdk | Well, what's really important is "Are you using any low bandwidth connections?" |
| 02:32.03 | evilbit | Qwell: thx! |
| 02:32.33 | L|NUX | mog : arround ? |
| 02:32.36 | evilbit | mvdk: nah, two 1GB connections to I1 phone calls come into my provider and back out to my cell phone |
| 02:32.38 | L|NUX | Qwell : hey |
| 02:32.46 | Qwell | what? |
| 02:32.52 | L|NUX | Qwell : howdy |
| 02:32.54 | mvdk | Goodness, where do you come by those kind of links? |
| 02:32.55 | Qwell | hi |
| 02:32.56 | ManxPower | 1Gbps? I HATE YOU! |
| 02:33.03 | sudhir492 | Qwell: I did not see this before, even on the same asterisk box. The messages Remote UNIX connection, and then Remote UNIX connection disconnected is so frequent |
| 02:33.17 | mvdk | I may have to murder you in your sleep - bandwidth, must have bandwidth :) |
| 02:33.21 | ManxPower | I'd be happy with connections with less than 300ms latency to the default gateway. |
| 02:33.22 | Qwell | sudhir492: I assure you, it's always been there |
| 02:33.40 | evilbit | I'm allowed to run servers at my full time job as long as I don't kill the network and don't attach dns names outside of our org names :-) |
| 02:34.06 | mvdk | Ah, I see |
| 02:34.14 | mvdk | To whom are you using GSM then? |
| 02:34.17 | sudhir492 | Qwell: I believe you. Does it have somehting to do with the debug level then? |
| 02:34.22 | Qwell | sudhir492: no |
| 02:34.33 | evilbit | and with the amount of p2p traffic going on I'm sure that I'm not getting great throughput |
| 02:35.02 | mvdk | evilbit: Just QoS your voice traffic up, screw the p2p stuff.... |
| 02:35.10 | evilbit | mvdk: not sure, I call my iax provider did, it connects to my * box and calls back out to my cellphone... so I guess it's the connection from my iax provider to my * box |
| 02:35.16 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-205.cybersurf.com) |
| 02:35.21 | evilbit | hehe, that would be great! |
| 02:36.11 | mvdk | Anyway, from your iaxprovider to * - Yeah, I'd imagine you'd likely use gsm, speex or g729 across that |
| 02:36.13 | evilbit | is there a "this is a great codec if you've got good bandwidth" recommendation? |
| 02:36.21 | mvdk | g711 |
| 02:36.22 | Qwell | ulaw |
| 02:36.33 | mvdk | But it burns bandwidth..... |
| 02:36.35 | sudhir492 | Qwell: I have 2 boxes, exact same hardware, running Asterisk 1.2.7.1. Why do I see this message -- Remote UNIX connection, on only 1 box? |
| 02:36.50 | Qwell | sudhir492: because you only pay attention to one |
| 02:36.53 | denon | because something else is connecting to that box |
| 02:37.01 | mvdk | 80 fscking kbps/calll |
| 02:37.06 | mvdk | s/calll/call |
| 02:37.10 | denon | or maybe the one is in a more verbose mode |
| 02:37.16 | evilbit | hmm, so my * box will only pick up the 1st time... guess I need to review my configs |
| 02:38.10 | JT | mvdk: is it pretty much flat on 80.0? |
| 02:38.20 | mvdk | evilbit: In any case, you pay for g729 |
| 02:38.42 | mvdk | g729 costs US$10/concurrent transcoding session |
| 02:38.44 | evilbit | in that it's propritary? |
| 02:38.51 | evilbit | ah, wow...prolly don't want that |
| 02:38.53 | mvdk | Yes, that's right |
| 02:38.57 | sudhir492 | I am paying attention to both the boxes. both of them have asterisk console open. I want to find out what is connecting to this box |
| 02:38.59 | JT | is that fee once off or per annum? |
| 02:39.03 | mvdk | Once off |
| 02:39.03 | *** join/#asterisk Nitrus^ (n=Nitrus_@72-34-76-86.skyriver.net) |
| 02:39.20 | JT | mvdk: so g711, is it flat on 80kbit/s? |
| 02:39.29 | mvdk | As I said, US$10/concurrent transcoding session is the Digium charge... |
| 02:39.32 | Qwell | JT: 64k/s plus overhead |
| 02:39.34 | Nitrus^ | is it possible to do a call transfer without a transfer button on your phone? |
| 02:39.36 | mvdk | JT: It depends.... |
| 02:39.49 | mvdk | JT: IP overhead makes it 80k/s |
| 02:40.14 | mvdk | If we're talking about a IPoPPPoA session, then it's about 104k/s |
| 02:40.40 | mvdk | These links are *really* not optimised for small packets.... |
| 02:41.01 | JT | hrm |
| 02:41.13 | JT | that's pretty bad for PPPoA |
| 02:41.26 | mvdk | Well, PPPoEoA is even worse :) |
| 02:41.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 02:41.59 | JT | and i realise the codec is 64000bits/s |
| 02:42.12 | JT | PPPoEoA?? |
| 02:42.22 | JT | never heard of a connection described like that |
| 02:42.24 | Qwell | over ethernet over ATM |
| 02:42.24 | mvdk | PPP over Ethernet (over ATM) |
| 02:42.30 | JT | sure |
| 02:42.43 | JT | but the over atm bit, is that just a given in dsl |
| 02:42.48 | mvdk | This is another common mechanism for a dsl connection.... |
| 02:43.20 | JT | where i'm going is, aren't dsl connections always atm related, so is that where it comes from? |
| 02:43.28 | mvdk | Yes, more or less |
| 02:43.36 | mvdk | It doesn't have to be that way, though.... |
| 02:43.59 | mvdk | I mean to say, if you controlled the DSLAM, you could use Ethernet instead, if you wished.... |
| 02:44.17 | JT | i'm curious as to what the linespeed means then, does that exclude any underlying atm overhead |
| 02:44.29 | mvdk | JT: ATM overhead is variable |
| 02:44.39 | JT | we have a bridged ethernet ADSL connection here |
| 02:44.41 | JT | no PPPoE |
| 02:44.44 | JT | no PPPoA |
| 02:44.49 | mvdk | It really depends on getting the 48 byte alignment.... |
| 02:44.50 | evilbit | so, searching through the docs and not seeing much... it there a common reason for asterisk only aswering the first call and then needing to do a reload before it will answer again? |
| 02:45.38 | mvdk | evilbit: Weird.... |
| 02:45.53 | evilbit | yeah :-( |
| 02:45.56 | mvdk | JT: There's a minimum of 5 bytes/48 bytes |
| 02:46.12 | mvdk | That is to say, every 48 bytes needs a 5 byte header |
| 02:46.25 | JT | yep, 53byte cells |
| 02:46.29 | mvdk | Add the SNAP headers |
| 02:46.37 | evilbit | old version is 1.2.0-rc2 new version is 1.2.10... so I guess I should expect some oddities |
| 02:46.44 | mvdk | (That's the EoA part of the equation) |
| 02:47.04 | JT | hmm |
| 02:47.19 | JT | is the overhead difference between SIP and IAX noticable? |
| 02:47.41 | fearnor | jt: i've done the math before. a 0byte TCP ACK will end up as 3 ATM frames. |
| 02:48.13 | fearnor | jt: only if you are drinking iax coolaid |
| 02:48.26 | JT | hmm |
| 02:48.27 | mvdk | JT: Only when you use trunking |
| 02:48.34 | JT | ah |
| 02:48.38 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 02:48.41 | mvdk | That is to say, if this is an inter-office connection, probably |
| 02:48.47 | fearnor | even then, i strongly doubt. |
| 02:48.54 | Dico_ | hello everybody |
| 02:48.58 | JT | connection to a voice provider |
| 02:49.08 | Dico_ | I would like to know what is an extension TBD in iax ? |
| 02:49.13 | JT | a few support iax, unfortunately the best value for money one only does SIP |
| 02:49.14 | mvdk | fearnor: If you're running 3 calls simultaneously, you spend a fair bit less.... |
| 02:49.21 | Qwell | Dico_: to be determined |
| 02:49.23 | fearnor | define fair bit. |
| 02:49.30 | Qwell | Dico_: it means your iaxy is waiting for you to dial |
| 02:49.34 | fearnor | jt: iax is for uberghetto. |
| 02:49.42 | JT | fearnor: what? |
| 02:49.43 | Dico_ | is it ? I see. Thanks |
| 02:49.47 | fearnor | iax is for 'not enough volume to justify real switches' |
| 02:49.48 | mvdk | fearnor: fair bit means 24kb/s for g711 |
| 02:49.59 | fearnor | mvdk: uhhh...for how many calls? |
| 02:50.10 | mvdk | fearnor: When we're talking about 4 calls |
| 02:50.26 | JT | what are you talking about fearnor? what protocol would you suggest for lots of connections between 2 points then? |
| 02:50.26 | mvdk | (Yes, these are small offices) |
| 02:50.37 | mvdk | JT: I'd suggest multiple IAX trunks |
| 02:50.41 | fearnor | JT: SIP, if you want to eventually move to proper equipment ;) |
| 02:50.46 | fearnor | define lots anyway |
| 02:50.51 | fearnor | is lots 10? 100? 1000? |
| 02:50.59 | fearnor | i dare you to do iax trunking with 1000 calls. |
| 02:51.00 | fearnor | or 100. |
| 02:51.03 | mvdk | You can, of course, use the CRTP protocol, but almost noone implements that.... |
| 02:51.05 | JT | whatever you think lots is, heh |
| 02:51.14 | fearnor | for asterisk ninjas, lots is 10 :P |
| 02:51.16 | JT | mvdk: why multiple trunks, reliability? |
| 02:51.42 | mvdk | JT: because iax trunking doesn't work so well above 40 calls in a trunk.... |
| 02:51.43 | Dico_ | Qwell, and what is a request DPREP ? |
| 02:51.46 | fearnor | real telco people talk in terms of ds3s worth of voice (672 trunks) |
| 02:51.51 | JT | ah ok |
| 02:51.52 | Dico_ | is it for presence ? |
| 02:52.05 | mvdk | fearnor: It may interest you to know that there are many many small businesses about |
| 02:52.08 | JT | yes fearnor but i am talking about voip solutions, not circuit switched tdm |
| 02:52.22 | mvdk | Not all of them are in the telco business, you understand.... |
| 02:52.39 | JT | voip over atm may come some way to being at the level of tdm |
| 02:52.40 | fearnor | thats why i want to know what people mean by 'lots' :) |
| 02:52.52 | JT | so things mightn't be tdm forever |
| 02:52.53 | fearnor | jt, atm is another technological mirage. |
| 02:53.06 | fearnor | please throw it out the garbage. |
| 02:53.15 | JT | i love telco technology |
| 02:53.17 | fearnor | yes, you can do CES on ATM (CBR and AAL1) |
| 02:53.18 | JT | but let's be realistic |
| 02:53.19 | mvdk | Anyway, VoIAX with trunking is acceptable for low volume stuff |
| 02:53.22 | JT | times are changing |
| 02:53.22 | fearnor | but nobody does |
| 02:53.34 | mvdk | And you can use it with somewhat high-volume stuff, too.... |
| 02:53.43 | fearnor | people do PPPoEoATM (shoot me now) |
| 02:54.01 | fearnor | sprinkle L2TP to taste.' |
| 02:54.03 | mvdk | As long as you define high-volume as a sufficiently small quantity, like an E1 worth :) |
| 02:54.53 | mvdk | But for many people, that's just fine |
| 02:55.51 | mvdk | Anyway, IAXoIPoPPPoA is a cheap solution, for all that :) |
| 02:56.06 | fearnor | then there's whole subject of voice over intarweb |
| 02:56.11 | fearnor | vs voiprivatenetwork |
| 02:56.32 | mvdk | fearnor: Let's define a new term: chequebook :) |
| 02:56.48 | JT | voip is pretty good for non telco business from a cost savings perspective |
| 02:56.57 | fearnor | not at a matter of money, matter of clue |
| 02:56.58 | JT | mvdk: there's already a term to go with that: TDM |
| 02:56.58 | mvdk | JT: my point precisely |
| 02:57.12 | fearnor | TDM rocks. nothing wrong with TDM. |
| 02:57.22 | JT | except chequebook |
| 02:57.29 | mvdk | fearnor: Except when you're the one paying for it..... |
| 02:57.32 | fearnor | not checkbook either, if you know what you are doing. |
| 02:57.32 | JT | it rocks |
| 02:57.35 | fearnor | i do TDM when I can. |
| 02:58.10 | fearnor | you aren't saving money. you are trading quality for money. |
| 02:58.14 | mvdk | Over here, it's 10 circuits for $305 a month |
| 02:58.35 | mvdk | fearnor: That's far more expensive than the VoIP circuits.... |
| 02:58.44 | fearnor | yes, between two offices its cheaper to get ip over intarweb, *but* you lose quality. |
| 02:59.15 | mvdk | So, we get 1 fractional PRI per office as a backup mechanism, and use the intarweb for everything we can |
| 02:59.15 | fearnor | you can't guarantee business quality service. |
| 02:59.24 | mvdk | Yes, we lose quality - but it's acceptable |
| 02:59.49 | JT | quality loss wouldn't be too bad if you use g711 |
| 02:59.53 | mvdk | People talk over mobile phones all the time, right? |
| 03:00.00 | JT | with no silence detection |
| 03:00.10 | JT | and the link was reliable |
| 03:00.19 | fearnor | then you get into faxes not working |
| 03:00.22 | fearnor | and t38 being brokeass |
| 03:00.35 | mvdk | fearnor: We use the PRI for faxes, this is not difficult.... |
| 03:00.43 | JT | well depends if you're interoffice faxing |
| 03:00.52 | JT | and still have SOME pstn lines |
| 03:00.52 | mvdk | The point is, iaxmodem works acceptably well.... |
| 03:00.54 | fearnor | my TDM shit just works (tm) |
| 03:00.59 | fearnor | and never have to think about it. |
| 03:01.07 | JT | we know it just works |
| 03:01.21 | JT | but there's more factors that need to be considered in a business, heh |
| 03:01.25 | benjk_ | faxes should be banned anyway |
| 03:01.45 | mvdk | Yes, but surely you understand that most small business owners don't much like being raped, fearnor.... |
| 03:02.04 | fearnor | it all depends heh |
| 03:02.08 | fearnor | we drop T1s to our customers |
| 03:02.25 | mvdk | Charges for E1s amount to bending over and copping it sweet |
| 03:02.27 | fearnor | used to put asstricks and just do PRI and zapras for IP |
| 03:02.34 | fearnor | but zapras sucks bricks |
| 03:02.42 | fearnor | life sucks. |
| 03:03.28 | mvdk | Well, we're proposing to use ADSL2+ w/ Annex M on 2 sites |
| 03:03.35 | mvdk | And a fractional PRI at each |
| 03:03.39 | benjk_ | Try Wanpipe instead of Zapras |
| 03:04.24 | babyju | Anyone know if cablevision's optimum voice compatible with asterisk? |
| 03:04.26 | mvdk | Yes, we use a backup PRI connection for just the voice stuff |
| 03:04.55 | mvdk | So there's 10 timeslots, of which 2 are used for a backup data circuit |
| 03:05.05 | mvdk | (Exclusively for voice) |
| 03:05.18 | mvdk | (encoded as g729, urgh :( ) |
| 03:05.21 | benjk_ | as for faxes, just get a single POTS line, plug your fax machine directly into the POTS line, do not plug in anything else, done! |
| 03:05.23 | *** join/#asterisk overworked554 (n=overwork@atlantis.clearshout.com) |
| 03:05.25 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 03:06.03 | overworked554 | hey all, does anyone know if ast-ax-snmpd works with asterisk 1.2.x? |
| 03:06.20 | mvdk | Well, the thing is, benjk, we have a single POTS line, but we can support faxing over the PRI with iaxmodem - it makes for slightly cheaper calls |
| 03:06.35 | JT | mvdk: you normally use g711? |
| 03:06.39 | mvdk | JT: no |
| 03:06.48 | fearnor | mv: "faxing" - its not an actual fax machine, is it? |
| 03:07.02 | mvdk | fearnor: No, it's hylafax |
| 03:07.05 | fearnor | ok. |
| 03:07.08 | benjk_ | vote benjk for prez and I promise you faxes will be eradicated within 4 years |
| 03:07.16 | JT | mvdk: so what do you normally use? |
| 03:07.29 | mvdk | JT: g729, unfortunately |
| 03:07.29 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.78) |
| 03:07.29 | fearnor | just checking if anyone has haxored iaxmodem to work with t38modem or something leet like that ;) |
| 03:07.31 | mvdk | Trying to kick the habit :) |
| 03:07.36 | JT | oh right |
| 03:07.57 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 03:08.03 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 03:08.05 | mvdk | Only thing is, most PSTN VoIP providers don't allow use of gsm or speex or ilbc.... |
| 03:08.07 | benjk_ | sending a fax: $5 federal tax per fax, $1 tax per page |
| 03:08.14 | JT | is there anything that sounds better than g729 that still uses a bit less bandwidth than g711? |
| 03:08.21 | mvdk | JT: speex |
| 03:08.22 | Newbie___ | hi, anyone using zhone channel bank with * ? or try zhone before |
| 03:08.24 | JT | yeah indeed most don't |
| 03:08.25 | mvdk | Or ilbc |
| 03:08.29 | benjk_ | receiving a fax from a country that doesn't collect fax taxes, $1 per page |
| 03:08.54 | benjk_ | double the rate every six months |
| 03:09.30 | mvdk | benjk: Get over yourself - many businesses fax stuff to this strata management firm I work for |
| 03:09.42 | mvdk | They're fscking plumbers |
| 03:09.54 | mvdk | Or electricians, or other tradesmen |
| 03:10.06 | file | I faxed something today, felt like I sold my soul |
| 03:10.06 | benjk_ | yeah I know some ppl still use fax, but that's no reason to still use the technology |
| 03:10.09 | file | or was that lastnight? I forget |
| 03:10.16 | mvdk | Most, unfortunately have not taken the time to familiarise themselves with technology... |
| 03:10.34 | mvdk | Of course, the postal service is even more evil :) |
| 03:10.38 | benjk_ | fax should be actively phased out |
| 03:11.10 | mvdk | 2 girls in the office here spend all their time scanning and filing documents that come in |
| 03:11.19 | mvdk | We're working on reducing that time.... |
| 03:11.39 | overworked554 | heh, um yeah... fax isn't going away anytime soon.. people like paper |
| 03:11.41 | mvdk | Fortunately, faxes are filed in less time than that :) |
| 03:11.42 | benjk_ | you can give ppl a fax machine that looks and talks and walks like a fax machine, but what it really is is a scanner that creates PDFs and sends them as a digitally signed email to the recipient |
| 03:11.55 | benjk_ | that would make sense |
| 03:12.01 | mvdk | benjk: That's what we use for interoffice faxes |
| 03:12.24 | benjk_ | but the existing analog protocols for faxing should be abandoned |
| 03:12.26 | evilbit | why signed? |
| 03:12.32 | mvdk | Or rather: it sends it to themselves, and they forward it to whomever |
| 03:12.38 | mvdk | But no, it's not signed |
| 03:13.01 | benjk_ | signed because the only reason why ppl still use fax in the first place is that in some jurisdictions courts accept faxes as evidence |
| 03:13.28 | Dico_ | in IAX, what is the difference between PING/PONG and LAGRQ/LAGRP ? |
| 03:13.37 | Dico_ | isn't it the same goal ? |
| 03:13.40 | benjk_ | request/response |
| 03:13.46 | evilbit | ah, ok |
| 03:14.33 | evilbit | isn't it still fairly easy to trick ppl by replacing the content of signed messages that haven't also been encrypted? |
| 03:14.34 | Dico_ | i knoww it's req/rep |
| 03:15.04 | JT | evilbit: smart people would check the signing with their pgp program |
| 03:15.10 | benjk_ | the entire transmission will be signed |
| 03:15.15 | benjk_ | would be |
| 03:15.27 | benjk_ | thus you couldn't replace the content |
| 03:15.52 | benjk_ | not without the signature being recognised as not matching the content |
| 03:15.54 | Dico_ | yes, with the fingerprint |
| 03:15.57 | evilbit | hehe, relying on ppl being smart is never a good thing ;-) |
| 03:16.23 | JT | evilbit: email programs setup for pgp automatically verify the signature |
| 03:16.47 | benjk_ | I am not even talking email programs |
| 03:16.48 | evilbit | well, I belive that it worked by taking a signed message... replacing the contents and then forging a new message as the original sender... the signature checks out but the content is now different |
| 03:17.03 | benjk_ | I am talking a true replacement of a fax machine that feels exactly like a fax machine |
| 03:17.20 | evilbit | gotcha |
| 03:17.21 | JT | wtf are you talking about evilbit |
| 03:17.23 | mvdk | benjk: Perhaps you're talking about t.37? |
| 03:17.23 | benjk_ | just that it doesn't have a phone jack, and it doesn't speak any analog fax protocol |
| 03:17.31 | JT | when you sign a message, is signs the whole thing |
| 03:17.43 | JT | so if you change the content, the pgp prog will redflag it |
| 03:18.09 | mvdk | And the thing, benjk, is that you'll require t.37 -> analogue bridges for a fair while yet.... |
| 03:18.15 | evilbit | there was some way around it... I'll find the paper as it was really informative |
| 03:18.39 | mvdk | Of course, it would be good if this proliferation of proprietary fax services stopped :( |
| 03:18.44 | *** join/#asterisk robbie2 (n=rob@howzat.dsl.onthenet.net) |
| 03:18.55 | benjk_ | I am talking about an internet-fax-machine that would send a fax in form of a signed PDF from one machine to another, but it wouldn't use any existing fax technology whatsoever |
| 03:19.06 | robbie2 | can i set a number to dial so any calls to my extension get redirected to another extension ? |
| 03:19.11 | robbie2 | in extensions.conf |
| 03:19.15 | mvdk | benjk: But how is it going to send it to existing fax machines? |
| 03:19.34 | JT | yeah, the only analogue stages should be, the light bouncing off the scanner imager, and the light eminating from a monitor to user, or creation of printout |
| 03:19.37 | benjk_ | mvdk, hence my $5 federal tax + $1 per page proposition ;) |
| 03:20.03 | overworked554 | mvdk: run it through an outsourced fax provider |
| 03:20.05 | mvdk | benjk: Ah - the main problem with that is all the senators that use faxes.... |
| 03:20.14 | benjk_ | existing fax machines would need an adapter at the receiving side or they are door stops |
| 03:20.18 | mvdk | overworked554: My point precisely |
| 03:20.33 | mvdk | That's what I was proposing, T.37 -> analogue bridges |
| 03:21.17 | benjk_ | yes, but thius would only be for transitional technology, not the actual technology to be used once the transition is complete |
| 03:21.45 | mvdk | benjk: I'd envisage it working somewhat like SMTP does today.... |
| 03:21.55 | benjk_ | yes, sure |
| 03:22.21 | benjk_ | but without any possibility to send anything without signatures and sender verification |
| 03:22.49 | JT | what, iris scan |
| 03:22.52 | JT | dna sample |
| 03:23.08 | benjk_ | no, something along the lines of a SIM card in the fax machine |
| 03:23.28 | benjk_ | you get your key to participate in the system, don't lose it |
| 03:23.37 | benjk_ | if you lose it, it can be cancelled |
| 03:23.47 | JT | like pgp |
| 03:24.15 | benjk_ | yes, but perhaps on a card or dongle so you can take it along |
| 03:24.20 | JT | sure |
| 03:24.34 | *** join/#asterisk mitcheloc (n=mitchelo@69-167-145-62.lmdaca.adelphia.net) |
| 03:24.43 | benjk_ | in no event should fax noises be transmitted anywhere |
| 03:24.44 | overworked554 | people will find it a nuisance though, you need to not think like a techie :-) |
| 03:25.03 | overworked554 | people wont use a card or fob |
| 03:25.08 | benjk_ | in a company, you would have a fax that is programmed with the key of the company or of the department |
| 03:25.24 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.78) |
| 03:25.24 | JT | yeah people would rather send unsigned faxes |
| 03:25.26 | benjk_ | you wouldn't need to stick a key device in |
| 03:25.32 | JT | especially companies |
| 03:25.38 | JT | plausable deniability |
| 03:25.52 | benjk_ | if that was so, companies wouldn't use faxes |
| 03:25.59 | JT | not really |
| 03:26.04 | JT | faxes aren't very secure |
| 03:26.11 | benjk_ | the only justification for faxes is that courts accept them as evidence |
| 03:26.12 | JT | anyone can set the sender line at the top |
| 03:26.20 | benjk_ | that doesn't matter |
| 03:26.24 | JT | and anyone can walk in and use the machine |
| 03:26.31 | benjk_ | what matters is that the courts accept it |
| 03:26.33 | mvdk | JT: Yeah, but when you go to court, you can sub peona their phone records |
| 03:26.51 | JT | call records from the telco would prove that a fax was sent from a particular location to a particular location |
| 03:26.51 | overworked554 | benjk: that is only one reason. |
| 03:26.55 | benjk_ | you could do the same with a new digitally based system |
| 03:26.55 | JT | NOT who sent it |
| 03:26.59 | mitcheloc | mvdk, you'll be able to subpoena their e-mail records from the NSA soon as well |
| 03:27.11 | mvdk | mitcheloc: lol |
| 03:27.13 | benjk_ | it wouldn't be ordinary email records |
| 03:27.39 | JT | so the court would look at the circumstances of the case in determining who sent a fax, blah blah blah |
| 03:27.39 | mvdk | What makes you think that the NSA would respond, mitcheloc? |
| 03:27.44 | JT | it's not cut and clear |
| 03:27.44 | overworked554 | one of the reasons why many of the fax service providers are still in business today is because of the phobias people have with computers. they are afraid of email. |
| 03:27.45 | benjk_ | you could build a specific fax store-and-forward system with all the features that are required |
| 03:28.02 | mvdk | benjk: We have that already - look at t.37 |
| 03:28.08 | benjk_ | but that's exactly what I was addressing |
| 03:28.15 | benjk_ | no its not like t37 |
| 03:28.31 | mvdk | benjk: How isn't it? |
| 03:28.33 | benjk_ | t37 is a transitional technology which is still partially based on the old fax stuff |
| 03:28.52 | JT | that is true |
| 03:28.59 | benjk_ | what I am talking about is to give people the exact same end user experience but not use any of the existing protocols whatsoever |
| 03:29.40 | JT | it is possible, but the replacement needs to be somwhat of a killer app |
| 03:29.53 | benjk_ | and since there is a notion to see a fax as something that can stand up in court, that system should add signatures and sender verification properly, in a way that it cannot be easily faked |
| 03:29.55 | JT | i mean, telegraphs and DXes have gone |
| 03:30.09 | mvdk | benjk: t.38 is based on the old fax stuff.... |
| 03:30.19 | mvdk | benjk: t.37 is a store-and-forward system |
| 03:30.31 | JT | the problem is you need independant third party verification for the best chances in court |
| 03:30.37 | JT | like telco call records |
| 03:30.38 | benjk_ | but it still has backwards compatibility ballast |
| 03:30.45 | mvdk | Of course, t.37 does use e.164 numbers |
| 03:30.52 | fearnor | t.37 is trash, done/done |
| 03:30.53 | benjk_ | well, that's alright |
| 03:31.01 | fearnor | t.38 is proper. |
| 03:31.11 | mvdk | benjk: I think that that "backwards compatibility ballast" makes a lot of sense.... |
| 03:31.13 | benjk_ | you can always use ENUM to translate those into ip addresses |
| 03:31.21 | fearnor | there's no point in store-and-forward faxing. |
| 03:31.31 | benjk_ | I think the backwards compatibility shouldn't be built into the infrastructure and the protocols |
| 03:31.34 | mvdk | Do you really *not* want to be able to send faxes to existing machines? |
| 03:31.53 | benjk_ | it should be in the form of an adapter that turns your old fax machine into one that can speak the new protocols |
| 03:31.54 | mvdk | fearnor: Perhaps you could use actual arguments to go with your babblings? |
| 03:31.56 | benjk_ | like an ATA |
| 03:32.19 | fearnor | mvdk, show me anyone who does bothers to do t.37 in this century |
| 03:32.20 | benjk_ | t.38 is anything else BUT proper |
| 03:32.34 | benjk_ | its 170 years old crap |
| 03:32.44 | overworked554 | fearnor: pgs :-) |
| 03:32.45 | benjk_ | still based on the existing fax crap |
| 03:32.52 | fearnor | t.38 is *encapsulation of all the data parts of fax calls*, thus compatible with every fax machine. |
| 03:32.57 | benjk_ | indeed |
| 03:33.01 | benjk_ | which makes it total crap |
| 03:33.02 | fearnor | you want something new? use PDF or scanned TIFs or whatever. |
| 03:33.08 | fearnor | don't pretend its a fax. |
| 03:33.27 | mvdk | fearnor: I happen to think that pretending it's a fax is the *right* thing to do.... |
| 03:33.48 | fearnor | it aint. either it looks like a fax, sends like a fax, feels like a fax |
| 03:33.55 | fearnor | or its store-and-forward, aka email. |
| 03:33.58 | overworked554 | mvdk: I agree, only way you'll convince people that they should hop on board with it |
| 03:34.00 | benjk_ | with the same logic we could say, don't pretend its a POTS line |
| 03:34.05 | fearnor | fax is immediate. store-and-forward is email. |
| 03:34.09 | benjk_ | and thus, don't use fax in the age of VoIP |
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| 03:34.28 | fearnor | big difference. my customers who get voip don't have to care if its pots or voip |
| 03:34.35 | *** join/#asterisk marcster (n=marc@203.87.184.218) |
| 03:34.47 | fearnor | fax that is store-and-forward isn't really fax. |
| 03:34.50 | benjk_ | because t.38 pretends its a POTS line, which is a totally flawed approach from the start |
| 03:34.57 | fearnor | incorrect |
| 03:35.06 | fearnor | it is the *only* approach that works and is backwards compatible. |
| 03:35.14 | fearnor | and preserves same user experience. |
| 03:35.16 | benjk_ | nonsense |
| 03:35.23 | fearnor | t.37 is *not the same user experience* |
| 03:35.30 | mvdk | fearnor: What happens when the other party signals busy? |
| 03:35.35 | marcster | hi. i wish to learn asterisk and finally i was able to save some cash for the hardware requirements. what do i need to buy to setup a working asterisk box? |
| 03:35.38 | benjk_ | you can replicate the user experience with a totally new system that it designed for IP from the ground up |
| 03:35.43 | fearnor | mvdk: your hear a busy tone. |
| 03:35.54 | mvdk | fearnor: precisely |
| 03:35.57 | fearnor | humans are used to busy tones. |
| 03:36.04 | benjk_ | and you'd have less trouble to do that since you don't need to work around all that ballast |
| 03:36.11 | mvdk | fearnor: to my mind - just send a message when it's done |
| 03:36.18 | fearnor | good luck in whatever reality you live in |
| 03:36.22 | mvdk | And send one if it isn't done after an hour of trying |
| 03:36.44 | benjk_ | what matters to people is the ease by which they can put a piece of paper on a fax machine, press a button and have it sent |
| 03:36.51 | benjk_ | thats the user experience to replicate |
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| 03:37.01 | overworked554 | guys, fax service providers sell e2f and f2e all day long and its basically store and forward :-) |
| 03:37.14 | mvdk | benjk: And I think that that's replicable with t.37 |
| 03:37.24 | benjk_ | and since faxes are seen as standing up in court, you also want to add proper sender verification |
| 03:37.32 | mvdk | overworked554: Yes, that's basically t.37 |
| 03:37.50 | mvdk | t.37 is essentially SMTP with a little bit added |
| 03:37.52 | benjk_ | you could always devise a realtime pdf transmission protocol |
| 03:38.13 | overworked554 | t37 IS smtp with a tiff attachment |
| 03:39.34 | mvdk | Yes - I'd use authenticated SSLised SMTP |
| 03:39.52 | benjk_ | whats important is that the infrastructure is set up do so sender verification and to be able to send directly from machine to machine, unless you actually want it in your email box |
| 03:40.57 | benjk_ | if that's available on the market, people will buy it and use it |
| 03:41.51 | benjk_ | but the user experience has to be the same, machine-to-machine, no computer needed, only an ethernet jack with an internet hook up |
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| 03:43.29 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:44.31 | mvdk | Point is, someone's going to want to bill you for a fax that needed an analogue hop somewhere |
| 03:44.59 | mvdk | Hence the need for authentication - it really just comes down to billing |
| 03:48.01 | benjk_ | charging is also an anti-spam measure of sorts |
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| 03:51.12 | *** join/#asterisk bmg505 (n=leon@c1-218-6.rndf.isadsl.co.za) |
| 03:52.31 | mvdk | Well, I've got to go, see you later :) |
| 03:53.05 | *** join/#asterisk madlynx (n=sleever@195.206.63.10) |
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| 04:01.13 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 04:01.28 | Mattwj2005 | good evening everyone! :) |
| 04:02.31 | madlynx | and U |
| 04:03.42 | Mattwj2005 | well I am rebuilding my Asterisk server....not using the sample configurations this time |
| 04:04.17 | ManxPower | I wonder if Postfix supports T.37 or if Outlook / Thunderbird support T.37 |
| 04:04.58 | ManxPower | I guess I wonder if ANYTHING my users might use support T.37. |
| 04:05.37 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 04:08.16 | marcster | hi. i wish to learn asterisk and finally i was able to save some cash for the hardware requirements. what do i need to buy to setup a working asterisk box? |
| 04:08.19 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 04:08.58 | Mattwj2005 | depends on what you want to do |
| 04:10.05 | marcster | what would a noob want to do? |
| 04:10.16 | trelane | marcster, how many phones do you intend to have connected? |
| 04:10.26 | trelane | what interface to the PSTN or VoIP do you intend to use? |
| 04:10.46 | marcster | trelane: i can start with one softphone and one analog and decide from there once i get things working |
| 04:11.25 | marcster | from pstn? im not sure if my answer's correct but the one with rj-11 terminator. for the voip, ethernet. |
| 04:11.43 | trelane | marcster, how many phone lines? |
| 04:11.55 | trelane | how many concurrent VoIP calls? |
| 04:12.00 | trelane | SIZE not the bloody plugs |
| 04:12.45 | marcster | trelane: say 5 phone lines and 5 concurrent voip calls |
| 04:12.56 | *** join/#asterisk Trakkasure (n=Sgemtum@24-50-26-239.atlsfl.adelphia.net) |
| 04:14.02 | ManxPower | marcster, Learning Asterisk is a long and complex process. Well, learning telephony is a long and complex process. Well, learning networking is a long and complex process. Learning Linux is a long and complex process. You should expect to learn telephony, networking, and linux before you'll really accomplish much with Asterisk |
| 04:14.21 | *** join/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer) |
| 04:14.33 | JerJer | ok who's trolling? |
| 04:14.44 | Qwell | JerJer: bunch of people |
| 04:14.48 | marcster | ManxPower: im halfway |
| 04:15.09 | russellb | i am! |
| 04:15.22 | russellb | I want to be the next vonage, who can be helping me? |
| 04:16.08 | trelane | russellb, well lets see, do you have shite for infrastructure? |
| 04:16.27 | Qwell | russellb: It's only a troll if somebody takes the bait |
| 04:16.31 | Qwell | so...troll |
| 04:16.33 | russellb | I have a computer and ADSL, is that enough? |
| 04:16.43 | ManxPower | Qwell, otherwise it's just carrion? |
| 04:16.49 | Qwell | ManxPower: exactly |
| 04:16.49 | JerJer | we have the best infrastructure - cup and string |
| 04:17.16 | russellb | JerJer: one of these days I'm seriously going to set that up between two boxes |
| 04:17.24 | trelane | russellb, no, you still need a propriatary protocol that doesn't make sense, and an IPO that looses money, and make sure that you under no circumstances have a business plan that make sense to anyone... that part's critical. |
| 04:17.42 | Qwell | trelane: SIP is proprietary? |
| 04:18.01 | Qwell | because like...that's what Vonage uses |
| 04:18.01 | trelane | Qwell, vonage isn't quite sip |
| 04:18.03 | Corydon76-home | I think you're thinking about Skype |
| 04:18.05 | Qwell | sure it is |
| 04:18.24 | trelane | really? when did that change, I knew it to be propriatary at one point because I ran ethereal on it and it sure as fsck wasn't sip |
| 04:18.28 | JerJer | russellb: we should do it for a booth display :) |
| 04:18.38 | russellb | trelane: never |
| 04:18.41 | russellb | JerJer: word. |
| 04:18.51 | Qwell | chan_cas? |
| 04:19.00 | russellb | I was just thinking chan_alsa is enough |
| 04:19.05 | *** part/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 04:19.17 | trelane | JerJer, I had a client _insist_ on a local VoIP carrier, and after 5 hours of trying to figure out their screwed up sip proxy, I handed the guy two dixie cups with cat 5e run between them and told him I'd finished his internet-enabled phone system. |
| 04:19.28 | russellb | it will just need a little AGI to connect to the other box via manager to make it go off hook / hangup |
| 04:20.46 | JerJer | lol |
| 04:21.38 | Qwell | just make it always offhook, and...stick it in a Wait() loop :p |
| 04:22.08 | Qwell | You might have issues with early media though |
| 04:22.09 | trelane | JerJer, they're being redundant by having 6 sip servers that all connect to the same PSTN interface... and the IP of the PBX is hardcoded into the switch instead of using register =>. The switch is a layer 5 managed jobbie that as best I can tell is misconfigured so that when one end dies instead of failing over it falls over |
| 04:22.24 | ManxPower | Bah! Screw the research. I'll torture newbies before going to bed. |
| 04:22.25 | JerJer | oh god |
| 04:22.45 | trelane | redundancy by 1. putting all the equipment in the same facility 2. terminating it all on the same place in the PSTN 3. all the equipment on the same managed switch |
| 04:23.43 | JerJer | someone needs to be beat with a wet noodle |
| 04:23.45 | trelane | I reasonably believe that the sip servers themselves are sun entry-level servers with fiber interconnects to the PSTN |
| 04:24.09 | trelane | and I have gotten calls from the provider at 8pm informing me they need help becase a harddrive took out their asterisk cluster |
| 04:24.20 | trelane | (I'm praying they aren't running an asterisk cluster as their sip switch) |
| 04:24.38 | JT | are they running asterisk on sun? |
| 04:24.47 | trelane | JT, sun opteron kit |
| 04:24.54 | JT | ah |
| 04:24.58 | trelane | at least from what I gather |
| 04:25.01 | trelane | that part's conjecture |
| 04:25.22 | trelane | I have no clue WHAT they run but "the asterisk server is down" == our entire sip network has disintegrated" |
| 04:25.34 | *** join/#asterisk alephcom (n=alephcom@h72.n66-112-187.mcsnet.ca) |
| 04:28.19 | JerJer | too funny |
| 04:28.35 | *** join/#asterisk icyfire0573 (n=icyfire@u1016342.ul.warwick.net) |
| 04:30.28 | JT | trelane: so no raid on their server(s)? |
| 04:31.10 | icyfire0573 | I've got a lot of noise on my phone line, I'm currently working on remedying this. However for the time being I'm getting a lot of false pickups on the asterisk server where the phone that is also connected to the line isn't ringing. I'm using fxs_ks signaling. I don't know if changing that would make a difference, I'm just asking to find out. Would a different signaling method make it have less false positives in regards to rings with li |
| 04:31.11 | icyfire0573 | ne noise? |
| 04:32.16 | *** part/#asterisk budmang (n=budman@12.206.134.162) |
| 04:32.18 | FuriousGeorge | JerJer: can i merge accounts or not |
| 04:32.29 | JerJer | can you? |
| 04:32.42 | Qwell | JerJer: Forge my signature :P |
| 04:32.54 | FuriousGeorge | JerJer: i give up |
| 04:33.06 | FuriousGeorge | icyfire0573: are you using call progress detection |
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| 04:34.24 | FuriousGeorge | JerJer: can i merge my nufone accounts into one account? by that i mean basically combining balances |
| 04:35.11 | JerJer | can you? |
| 04:35.34 | FuriousGeorge | JerJer: i figured you would know, as it is your company. |
| 04:35.35 | FuriousGeorge | my bad |
| 04:36.49 | icyfire0573 | FuriousGeorge, its not turned on to my knowledge, but I dont know what file that would be in or if its a default either |
| 04:36.51 | trelane | JT yeah there was, but this drive ground the array to a halt |
| 04:37.45 | FuriousGeorge | icyfire0573: that would be in zapata and would only affect analog calls. now that i think about it, it doesnt cause false rings, just might disconnect your call prematurely |
| 04:38.01 | JerJer | usually you ask "may i do this" |
| 04:38.18 | Qwell | JerJer: May I have you forge my signature? |
| 04:38.22 | Qwell | :P |
| 04:38.23 | FuriousGeorge | JerJer: only when i go to the bathroom in middleschool |
| 04:38.28 | icyfire0573 | :-/ Do you know if a different signalling method would make it false detect more? |
| 04:38.33 | FuriousGeorge | the rest of the time i subscribe to "can i" |
| 04:38.40 | icyfire0573 | or even less for that matter :-) |
| 04:39.09 | FuriousGeorge | icyfire0573: im not expert (that's qwell) but i dont think it would do that |
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| 04:39.20 | FuriousGeorge | ks is supposed to detect that sort of thing |
| 04:39.30 | Qwell | nah, I don't know much about zap |
| 04:39.31 | FuriousGeorge | when i used to get false rings was when: |
| 04:39.42 | Qwell | I have a feeling I'll learn it real soon though |
| 04:39.47 | FuriousGeorge | a> i had an old switched pbx sitting on top of asterisk that wouldnt play nice |
| 04:39.47 | icyfire0573 | So, basically "Clean up the line noise" is the answer? |
| 04:39.48 | FuriousGeorge | and |
| 04:41.00 | FuriousGeorge | b> when i would answer and wait for more than 2 seconds, the pbx would interrupt the calling party's second ring, making him/her think the call had been answered, sometimes they would then hang up when not hearing anything by the next ring |
| 04:41.24 | FuriousGeorge | i fixed the former by removing the switched pbx and the latter by using wait(2) and not wait(3) |
| 04:42.16 | FuriousGeorge | JerJer: may i merge two of my nufone accounts into one |
| 04:42.23 | icyfire0573 | FuriousGeorge, Your suggesting I add a 1 second delay before signalling the SIP phone that there is a call on the line then? And using the delay to help prevent the false positive? |
| 04:43.04 | FuriousGeorge | icyfire0573: what i am saying is that depending on your wait() after your answer, * may interrupt the calling party's second ring |
| 04:43.52 | FuriousGeorge | if it does calling party may not allow enough time for the next ring, and may just think his/her call got disconnected |
| 04:44.33 | icyfire0573 | Alright, Well right now it just sends the ring to my SIP phone and dosen't stop it from ringing on the other end. I'm going to try and add exten => s,1,wait(1) to see if the one second delay fixes the false positives for now. |
| 04:44.54 | FuriousGeorge | icyfire0573: use answer and wait(2) |
| 04:45.01 | FuriousGeorge | then dial(sip/device) |
| 04:45.33 | icyfire0573 | I'd rather not answer it, because there are other phones connected to the trunk which would stop ringing if I answered it. |
| 04:45.54 | FuriousGeorge | icyfire0573: other phones where? behind your pbx or on other pbx's |
| 04:46.27 | icyfire0573 | its phone line Splitter PBX/Cordless |
| 04:46.32 | JerJer | FuriousGeorge: typically when you want support from a company you use that companies resources to submit a request |
| 04:47.06 | FuriousGeorge | JerJer: did it. how long until, typically, your company responds. |
| 04:47.43 | FuriousGeorge | icyfire0573: iow, if you answer with * your cordless wont ring |
| 04:47.57 | JerJer | FuriousGeorge: i have not seen any requests from you |
| 04:48.04 | icyfire0573 | Exactly. So I'm gonna just try the wait(1) then ring. |
| 04:48.54 | FuriousGeorge | icyfire0573: if thats the case, the false ring has nothing to do with anything. what do you mean, by the way? you answer on the cordless and * keeps ringing? |
| 04:49.13 | FuriousGeorge | JerJer: should i ask again over e-mail? |
| 04:49.32 | JerJer | should you? |
| 04:49.33 | icyfire0573 | The cordless phone dosen't ring at all. But the asterisk box starts ringing my SIP phone |
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| 04:49.45 | FuriousGeorge | JerJer: i use nufone for the customer support |
| 04:50.05 | JerJer | i bet you do |
| 04:50.10 | FuriousGeorge | icyfire0573: you got a zap interface on there |
| 04:50.17 | icyfire0573 | Yea |
| 04:50.27 | FuriousGeorge | who did the wiring |
| 04:50.29 | FuriousGeorge | ? |
| 04:50.46 | icyfire0573 | Me, its hardly wiring as it were. Just a splitter on the main line. |
| 04:51.02 | FuriousGeorge | i meant the wiring from the splitter to the NID |
| 04:51.09 | FuriousGeorge | (network interface device) |
| 04:51.17 | FuriousGeorge | (punch down thingy) |
| 04:51.22 | Qwell | (green box) |
| 04:51.31 | FuriousGeorge | (thing with the metal clippies) |
| 04:51.32 | JT | MDF |
| 04:51.33 | JT | IDF |
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| 04:51.52 | icyfire0573 | That's probably as old as the house. |
| 04:52.40 | *** part/#asterisk alephcom (n=alephcom@h72.n66-112-187.mcsnet.ca) |
| 04:52.46 | FuriousGeorge | icyfire0573: if you are getting false rings, and there is no other PBX sitting on the line, im at a loss |
| 04:53.17 | icyfire0573 | Alright. Like I said there is an excessive amount of line noise, like a really loud hum on the line. I was thinking that was the cause. |
| 04:53.18 | FuriousGeorge | like i said, ive had a legacy PBX stting between * and ma'bell do that, but if you dont have the same thing going on... |
| 04:53.22 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 04:54.14 | FuriousGeorge | icyfire0573: im gonna lean toward agreeing with you, only cuz i dont know any better. this is outside of my scope of understanding, but it seems like something is making the line voltage act like the line is ringing, and myabe you hear that with the hum |
| 04:54.33 | FuriousGeorge | my advice would be to hook up right to the NID, and see if the hum, or the false rings go away |
| 04:54.38 | JT | well if it's a hum |
| 04:54.56 | JT | there's a high likelyhood it's AC power induced |
| 04:54.57 | FuriousGeorge | and if the hum happens with a regular phone call ma'bell and complain |
| 04:55.19 | JT | which is AC, 60HZ in the US, line rining is normally about 20Hz though |
| 04:55.30 | JT | ringing |
| 04:55.31 | icyfire0573 | Yea, its definatly occuring with a regular phone. I'll have to look around for AC lines that would be causing the problem. |
| 04:55.33 | FuriousGeorge | JT: like a groung loop problem right? |
| 04:55.50 | JT | maybe, or AC and phone cables running parallel |
| 04:56.35 | FuriousGeorge | which goes back to my asking who did your wiring. old houses tend to daisy chain telephone wiring. who knows what goes on along that |
| 04:56.52 | icyfire0573 | I think the incoming phone line is right next to the incoming power line if that could be it, The issue with the humming came up about a month ago I think. We never really got around to diagnosing it and I was too cheap to pay for someone to come out and look into it. |
| 04:57.13 | JT | if you had some test gear it would be nice |
| 04:57.21 | JT | oscilloscope, or if you're lucky, TDR |
| 04:57.49 | FuriousGeorge | icyfire0573: hook up to directly to the nid, unhook everything else, and let me know how that does you |
| 04:57.50 | icyfire0573 | Yea, I'll just borrow the "Phone box" from work. Its got all the good stuff. That phone with the clips for ends, the toner. |
| 04:58.07 | FuriousGeorge | but set and line tracer? |
| 04:58.10 | FuriousGeorge | *butt |
| 04:58.39 | JT | buttinski |
| 04:58.39 | icyfire0573 | Probably, I don't know the names for them. |
| 04:58.48 | JT | scope and TDR are usually more expensive than a buttinski and cable tracer |
| 04:58.56 | FuriousGeorge | niether of those will detect the problem JT is describing (other than hearing the hum on the testing devices) |
| 04:59.21 | icyfire0573 | Haha, but I would know the EXACT frequency of the hum. :-) |
| 04:59.38 | JT | your "phone box" detects that? |
| 04:59.43 | FuriousGeorge | however, if you pick the lines up from the street at the NID/DMARC, as i suggested, assuming the interference is comming off the street, you can isolate the problem |
| 05:00.03 | FuriousGeorge | * should be chilling out by the nid any-damn-way for this very reason |
| 05:00.07 | *** part/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 05:00.08 | FuriousGeorge | :) |
| 05:00.08 | icyfire0573 | Its a good a plan as any. I got plenty of free time on friday to work that out. |
| 05:00.20 | FuriousGeorge | now, who loves whiskey as much as me |
| 05:00.21 | FuriousGeorge | ? |
| 05:00.31 | icyfire0573 | What kind of whisky? |
| 05:00.35 | FuriousGeorge | any |
| 05:00.43 | FuriousGeorge | but ill put burbon before scotch for now |
| 05:00.48 | icyfire0573 | SoCo then! |
| 05:00.56 | FuriousGeorge | followed by american, irish, and canadian in that order |
| 05:01.09 | FuriousGeorge | soco is a liqueur, did you know that? |
| 05:01.20 | icyfire0573 | Not at all, although I did know it was candy :-) |
| 05:01.56 | FuriousGeorge | you are absolutely right. soco is full of sugar which makes it a liqueur. not a whiskey |
| 05:02.02 | icyfire0573 | I do well with scotch and water. |
| 05:02.12 | FuriousGeorge | same thing with cap't morgan. it's not real rum |
| 05:02.30 | FuriousGeorge | icyfire0573: i wait till the ice floats then its scotch and water :) |
| 05:02.38 | icyfire0573 | hahah. awesome |
| 05:02.48 | icyfire0573 | Well, personally I'm a Gin kinda guy. Bombay Sapphire. |
| 05:03.13 | brookshire | Gin & Tonic #1 |
| 05:03.18 | FuriousGeorge | <PROTECTED> |
| 05:03.23 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 05:03.25 | icyfire0573 | G&T fo life! :-D |
| 05:03.42 | FuriousGeorge | but it sure is delicious with tonic |
| 05:03.51 | icyfire0573 | FuriousGeorge, Agreed on both points. |
| 05:04.11 | icyfire0573 | HAHA! |
| 05:04.55 | JT | will voip for feed |
| 05:04.59 | JT | or food |
| 05:05.16 | icyfire0573 | Cattle feed for you then! |
| 05:05.16 | FuriousGeorge | some guy asked me to sell vdka for Remmy Martin, i explained that I couldnt because I need to be available to restart my various asterisk servers every few months |
| 05:06.09 | JT | heh, impression of reliability there |
| 05:06.50 | icyfire0573 | Alright, I need sleep, Work in 6 hours. |
| 05:07.07 | FuriousGeorge | icyfire0573: get to know ur nid, let me know how it goes |
| 05:07.10 | FuriousGeorge | later |
| 05:07.17 | icyfire0573 | Will do. Later! |
| 05:07.28 | JT | anyway, just tell me you were joking and asterisk never need restarting once setup :) |
| 05:07.54 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 05:08.21 | FuriousGeorge | JT: yes, asterisk never needs restarting. |
| 05:08.27 | JT | cool |
| 05:09.11 | FuriousGeorge | ~qwell |
| 05:09.13 | jbot | well, qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 05:09.36 | JT | heh |
| 05:10.26 | JT | so realisticly, what's reliability like? |
| 05:11.05 | FuriousGeorge | JT: let me preface this by saing that of the half dozen asterisk servers ive installed only one is on propper hardware, the rest are PCs ive installed linux and some TDM400Ps on |
| 05:12.22 | FuriousGeorge | i'd say cron it to restart asterisk daily, and the server once a month for the MPG123's that spin out of control. otherwise train your customers to restart the server when they a> stop receiving incoming analog calls (zap fails), or nothing works at all (* fails) |
| 05:12.24 | JT | proper hardware being server grade computer and not TDM400P? |
| 05:12.37 | FuriousGeorge | no, ive confused you |
| 05:12.52 | FuriousGeorge | propper hardware being a "server quality motherboard" |
| 05:13.14 | FuriousGeorge | the tdm400p is the digium produced analog telephone interface which i find fails every 8 weeksw |
| 05:13.49 | JT | yeah i think i'm understanding you perfectly :) |
| 05:14.18 | L|NUX | mog : arround ? |
| 05:14.39 | FuriousGeorge | my main issue is that incoming analog channels (on the tdm400p) become in use, even when they are not. |
| 05:14.46 | FuriousGeorge | after a while, that is |
| 05:14.49 | JT | hmm |
| 05:15.10 | FuriousGeorge | ~lastspoke mogorman |
| 05:15.23 | FuriousGeorge | ~lastseen mogorman |
| 05:15.29 | FuriousGeorge | ~jbot |
| 05:15.32 | jbot | i heard jbot is only marginally useful at best, He got a C- on his Turing Test, or a complete idiot, or a dolt |
| 05:16.01 | FuriousGeorge | the "he got a C- on his turing test" was me |
| 05:16.11 | FuriousGeorge | i think we can all agree thats the best part |
| 05:16.38 | EyeCue | jt@austnet for the win |
| 05:16.39 | EyeCue | :] |
| 05:17.01 | L|NUX | can some one help me with chan_jingle |
| 05:17.25 | FuriousGeorge | L|NUX: there's a chan_jingle? u must be using svn, no? |
| 05:17.37 | JT | jingle....what is that |
| 05:17.47 | FuriousGeorge | google voip over xmpp |
| 05:17.54 | L|NUX | FuriousGeorge : i am using svn |
| 05:17.58 | L|NUX | FuriousGeorge : svn trunk |
| 05:18.11 | FuriousGeorge | yeah, trunk, thats what i shoulda said |
| 05:18.13 | L|NUX | FuriousGeorge : but some how i am unable to configure it :( |
| 05:18.30 | FuriousGeorge | L|NUX: did mog document it yet? |
| 05:18.46 | L|NUX | no idea :( |
| 05:19.07 | Newbie___ | hi, anyone using zhone channel bank with * ? or try zhone before |
| 05:19.14 | ManxPower | ~docs |
| 05:19.15 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 05:19.16 | ManxPower | ~mailinglist |
| 05:19.18 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 05:19.18 | ManxPower | ~book |
| 05:19.20 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:19.29 | FuriousGeorge | ~furiousgeorge |
| 05:19.31 | jbot | hmm... furiousgeorge is a knife-fighting (cable) monkey last seen with The Man with the Yellow Bat |
| 05:19.50 | FuriousGeorge | did i add cable? either way that is so true and hilarious |
| 05:20.13 | JT | Newbie___: no, but i WILL be... |
| 05:20.18 | FuriousGeorge | ~manxpower |
| 05:20.21 | jbot | manxpower is, like, your God. someone you should send lots of money to because he helps so many people. |
| 05:20.37 | FuriousGeorge | f that noise :) |
| 05:25.11 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 05:27.39 | Newbie___ | JT: been googling and the general feedback is a pain to setup. is that true ? |
| 05:28.03 | JT | Newbie___: haven't started, but have heard a few complaints |
| 05:28.20 | JT | main complaint of the guy i bought it off is don't try to get caller id working |
| 05:28.24 | JT | because he tried everything |
| 05:28.42 | Newbie___ | JT: i dont mind the callerid stuff, but is the echo that worry me |
| 05:29.00 | JT | ah not sure |
| 05:29.10 | JT | guy didn't mention anything |
| 05:30.00 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 05:30.26 | Newbie___ | JT: it needs to upgrade to a version b too |
| 05:31.16 | JT | ? |
| 05:32.10 | *** join/#asterisk parag_ast (n=root@dxb-b16259.alshamil.net.ae) |
| 05:32.48 | dlynes_laptop | parag_ast: you know you shouldn't irc as root, right? |
| 05:33.00 | parag_ast | ohh ya sorry |
| 05:33.03 | parag_ast | wait |
| 05:33.04 | *** part/#asterisk parag_ast (n=root@dxb-b16259.alshamil.net.ae) |
| 05:33.43 | Newbie___ | JT: firmware |
| 05:34.01 | JT | yes, firmware |
| 05:34.03 | JT | but what for? |
| 05:35.49 | Newbie___ | http://www.voip-info.org/wiki/view/Asterisk+Channel+Bank |
| 05:36.12 | *** join/#asterisk parag_ast (n=parag_as@dxb-b16259.alshamil.net.ae) |
| 05:37.01 | parag_ast | Can anybody note http://pastebin.ca/112763 and line no.s 93,274,,286 |
| 05:37.24 | parag_ast | native bridge is getting created between Zap/1-1 and Zap/2-1 |
| 05:37.30 | parag_ast | which should not happen |
| 05:39.28 | parag_ast | dlynes_laptop, help me |
| 05:39.50 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:39.51 | parag_ast | My two lines are getting busy... |
| 05:41.46 | dlynes_laptop | parag_ast: Looks to me like SIP/30, Zap/2-1 and Zap/1-1 were all in a conference together, and then SIP/30 left the conference, right? |
| 05:49.52 | *** join/#asterisk parag_redhat (n=parag_re@dxb-b111502.alshamil.net.ae) |
| 05:50.48 | parag_redhat | Can Anybody help me with http://pastebin.ca/112763 Note the lines 93,274,,286 |
| 05:51.38 | parag_redhat | dlynes_laptop, ?? |
| 05:54.09 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 05:54.37 | JT | Newbie___: ? |
| 05:54.53 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 05:55.51 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 05:56.24 | Un1x | parag_redhat that looks like FreePBX |
| 05:56.29 | smackus | ok, configuring wanpipe for a T1, what option do i select? 1,2,3,4 or 5? none of them make sense to me. I would expect to see one of those for a T1 |
| 05:56.42 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 05:56.52 | smackus | i have a 4port sangoma card where I will have a combo of pri and t1s |
| 05:59.16 | Corydon76-home | Please contact Sangoma for hardware support |
| 05:59.56 | websae | anyone used OpenPBX? |
| 06:00.11 | Corydon76-home | This is #asterisk, not #openpbx |
| 06:00.22 | websae | let me rephrase that question |
| 06:00.41 | websae | has anyone used t.38? in conjunction with openpbx? |
| 06:00.47 | *** join/#asterisk daysmen3 (n=primus@host86-139-117-242.range86-139.btcentralplus.com) |
| 06:01.11 | Corydon76-home | Again, please take those questions to #openpbx |
| 06:02.04 | websae | i figure there is a large fax over IP base here....with experience and input on such |
| 06:02.24 | Corydon76-home | Yes, and we use it on Asterisk |
| 06:02.45 | *** join/#asterisk angom_h (n=Angel@red-corp-200.76.226.132.telnor.net) |
| 06:02.51 | Corydon76-home | Besides, if you haven't noticed yet, the OpenPBX project is dead |
| 06:03.11 | parag_redhat | Un1x, its a zap problem |
| 06:03.14 | parag_redhat | help me |
| 06:13.20 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 06:38.10 | ousk | any sip guru here? |
| 06:39.00 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 06:47.49 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 06:55.39 | *** join/#asterisk WildPikachu (n=WildPika@unaffiliated/wildpikachu) |
| 06:55.59 | WildPikachu | if I pick up a call from a call queue, how do I allow it to be transferred? |
| 06:57.19 | *** join/#asterisk asterboy (n=kevin@S01060204ee2b6007.ed.shawcable.net) |
| 06:57.36 | asterboy | irssi rocks! |
| 06:57.50 | Qwell | asterboy: more than you know |
| 06:58.30 | asterboy | ya, I won't touch another irc client now |
| 06:58.51 | asterboy | just built a new lfs system and that was one of the first things to go on it. |
| 07:05.14 | *** join/#asterisk firstknight (n=firstkni@202.155.150.55) |
| 07:05.55 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
| 07:05.57 | tengulre11 | hi,all |
| 07:06.02 | rowter | Anyone has by chance NVLineDetect.c here so I could test it out? |
| 07:06.06 | tengulre11 | which OS suit to asterisk? |
| 07:06.40 | firstknight | Hi, does asterisk support logging sms, video, text, voice chat ( to make it short, all data ) of its clients? |
| 07:07.12 | tengulre11 | Debian,Redhat,Gentoo...? |
| 07:08.05 | tengulre11 | am i offline too? |
| 07:08.07 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:08.23 | tengulre11 | who can give me tips? |
| 07:09.45 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:10.15 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 07:10.21 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 07:11.19 | WildPikachu | anyone have "call transfer" problems with the budgetone 102 and its transfer button? |
| 07:12.59 | *** join/#asterisk ptblank (n=MURDER1@69-166-222-110.lmdaca.adelphia.net) |
| 07:15.13 | L|NUX | any one successfully connected google talk with Asterisk ? |
| 07:15.28 | asterboy | make sure you upgrade that budgetone to the latest firmware |
| 07:19.05 | JT | does anyone know offhand whether T1 crossover cables are meant to use shielded twisted pair cables? |
| 07:19.13 | rowter | anyone here had used NVlinedetect? |
| 07:21.02 | asterboy | nmap compile output has a cool ascii art message |
| 07:21.28 | hads|home | So does Asterisk :) |
| 07:21.53 | asterboy | seriously?...not noticed that if there is |
| 07:22.08 | hads|home | It might only be in trunk. Not sure. |
| 07:25.12 | daysmen3 | is there a way of creating QUEUES dynamically |
| 07:25.29 | *** join/#asterisk angom_w (n=Angel@red-corp-200.76.226.132.telnor.net) |
| 07:25.53 | daysmen3 | without having them statically written and defined in queues.conf |
| 07:27.54 | L|NUX | daysmen3 : yes there is |
| 07:31.28 | L|NUX | daysmen3: exten => 100,1,AddQueueMember(sales|SIP/${CALLERIDNUM}) |
| 07:31.39 | L|NUX | like this :) |
| 07:32.38 | creativx | so addqueuemember creates the queue too if its non existant? |
| 07:32.45 | L|NUX | yeah |
| 07:36.38 | *** join/#asterisk ailantian (n=ailantia@219.142.133.95) |
| 07:36.55 | rowter | Anyone has by chance NVLineDetect.c here so I could test it out? |
| 07:37.59 | daysmen3 | <L|NUX> - WOW - i didnt think it was possible - so let me get this straight - rather than create queue by inserting definition in queues.conf you can creat it directly from extension.conf |
| 07:40.54 | *** join/#asterisk ailantian (n=ailantia@219.142.133.95) |
| 07:43.27 | L|NUX | yeah |
| 07:43.46 | L|NUX | daysmen3: i have done it by my self and its working perfect |
| 07:44.21 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-59-55.cybersurf.com) |
| 07:44.21 | L|NUX | and if you want to logged out from queue you can use this |
| 07:44.21 | L|NUX | exten => 101,1,RemoveQueueMember(sales|SIP/${CALLERIDNUM}) |
| 07:44.23 | L|NUX | :) |
| 07:44.45 | Qwell | creativx: No, it doesn't |
| 07:45.11 | L|NUX | Qwell : am i wrong ? |
| 07:45.14 | Qwell | yes |
| 07:45.19 | L|NUX | awww |
| 07:46.33 | ailantian | may I ask a question? I have a problem with SUBSCRIBE |
| 07:46.41 | Qwell | bbl |
| 07:47.35 | firstknight | does asterisk can record all information in its network ( voice recording, video recording, sms logging, etc )? |
| 07:47.37 | ailantian | my asterisk is 1.2.10, it always reply 200 OK with expires:0, why? someone help me ? thanks |
| 07:47.48 | daysmen3 | <Qwell> OK so this aint true or it is |
| 07:49.32 | creativx | my queues are defined static in queues.conf |
| 07:49.39 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 07:49.42 | creativx | but they are populated dynamically from a db |
| 08:00.29 | daysmen3 | what im trying to do is create queues dynamically for each user and use queue functionality for Follow me or hunt dialing |
| 08:01.01 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 08:01.39 | *** join/#asterisk asterboy (n=kevin@S01060204ee2b6007.ed.shawcable.net) |
| 08:05.27 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 08:09.02 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:11.00 | Un1x | anyone have any suggestions for some good cards |
| 08:11.14 | hads|home | Hallmark? |
| 08:11.17 | Un1x | coz after i had bought the tdm400p everyone said i shouldn't have and it wasn't a good card etc |
| 08:11.21 | Un1x | so new suggestions :P? |
| 08:11.36 | hads|home | There's nothing wrong with the TDM400 |
| 08:14.04 | Un1x | hads|home you think you could help me a bit the rxgain isn't working if you'd like to help please.. |
| 08:14.55 | hads|home | What exactly is the problem? In question form. |
| 08:17.16 | Un1x | well i tried rxgain isn't working |
| 08:17.38 | Un1x | if im correct does the rxgain only work if the input is from a pstn line and not from sip etc? |
| 08:17.48 | hads|home | Yes |
| 08:20.07 | Un1x | fuck |
| 08:20.13 | Un1x | thats why rxgain aint working for me |
| 08:20.21 | Un1x | is there a way to increase volume from sip? |
| 08:20.46 | hads|home | Swearing isn't nessecary |
| 08:21.44 | *** join/#asterisk AuPix (n=jfp_mart@81.171.226.203) |
| 08:22.45 | hads|home | rxgain/txgain can only effect things that are plugged into the card, so it's not going to effect anything SIP |
| 08:22.54 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 08:23.15 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 08:24.08 | Un1x | hads|home well i have a analoge phone plugged into, the FXS port... |
| 08:24.22 | Un1x | shouldn't it still increase the volume even tho the call is being placed via SIP... |
| 08:24.31 | hads|home | Well you didn't say that. |
| 08:24.39 | Un1x | sorry :p |
| 08:25.19 | hads|home | I don't know much about setting gains on the FXS ports. |
| 08:26.57 | asterboy | jbot libnewt |
| 08:27.09 | asterboy | ~libnewt |
| 08:27.34 | asterboy | jbot where is libnewt |
| 08:27.36 | jbot | asterboy: what are you talking about? |
| 08:27.41 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 08:28.08 | asterboy | jbot I want libnewt |
| 08:28.10 | jbot | You want libnewt? |
| 08:28.16 | asterboy | jbot yes |
| 08:28.18 | jbot | You don't say! |
| 08:28.27 | asterboy | jbot you ass |
| 08:28.50 | hads|home | Heh |
| 08:29.07 | asterboy | :P |
| 08:29.45 | L|NUX | any one successfully connected google talk with Asterisk ? |
| 08:30.42 | mitcheloc | i tried, but had poor call quality |
| 08:30.49 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 08:31.00 | mitcheloc | what is so cool about using gtalk as a soft phone anyway? |
| 08:37.50 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402BD8F.pool.t-online.hu) |
| 08:43.19 | L|NUX | mitcheloc : can you help me in that :) |
| 08:44.31 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 08:45.39 | mitcheloc | L|NUX, can you be more specific? |
| 08:45.56 | L|NUX | mitcheloc : i have installed * from svn trunk |
| 08:46.09 | L|NUX | and now i want to use chan_jingle along with asterisk |
| 08:46.12 | L|NUX | but i am failed :( |
| 08:46.26 | L|NUX | i have searched alot but could not find much docs |
| 08:46.45 | tuck3r | i thought gtalk wasn't jingle compatible yet |
| 08:46.54 | mitcheloc | there really aren't any docs |
| 08:46.59 | *** join/#asterisk TypMic (n=TypMic@outland.cmf.nrl.navy.mil) |
| 08:47.09 | mitcheloc | tuck3r, google wrote the first jingle spec |
| 08:47.16 | L|NUX | hummm |
| 08:47.19 | mitcheloc | there is a new one now that is official |
| 08:47.27 | L|NUX | but i was digging from past few days |
| 08:47.34 | L|NUX | and got ill and come back to done thi |
| 08:47.38 | L|NUX | thi/this |
| 08:48.47 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 08:49.15 | L|NUX | mitcheloc : can you help me with that ? |
| 08:50.17 | mitcheloc | L|NUX, you need to be more specific |
| 08:50.25 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:50.28 | L|NUX | mitcheloc : can i pm you ? |
| 08:50.40 | mitcheloc | just chat here, nobody else is awake ;) |
| 08:50.50 | L|NUX | ok |
| 08:50.52 | L|NUX | wait |
| 08:50.55 | puzzled | morning |
| 08:51.07 | L|NUX | let me show your my jingle.conf and jabber.conf |
| 08:52.00 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402BD8F.pool.t-online.hu) |
| 08:52.19 | L|NUX | http://pastebin.ca/112876 |
| 08:52.28 | L|NUX | mitcheloc : check this http://pastebin.ca/112876 |
| 08:52.41 | *** join/#asterisk kannan (i=neoo@221.134.167.199) |
| 08:53.33 | L|NUX | but its not connected |
| 08:53.39 | L|NUX | when i do jabber show connected |
| 08:53.43 | mitcheloc | L|NUX, what error messages are you getting? and is res_jabber loaded or not? |
| 08:53.44 | L|NUX | it say disconnected |
| 08:54.04 | L|NUX | its loaded |
| 08:54.15 | L|NUX | jabber show connected |
| 08:54.15 | L|NUX | JABBER: User: dost4u@gmail.com is Disconnected |
| 08:55.39 | mitcheloc | try setting the serverhost to talk.gmail.com |
| 08:55.43 | L|NUX | ok |
| 08:56.48 | L|NUX | JABBER ERROR: No DNS asterisk for client to talk.gmail.com |
| 08:57.25 | L|NUX | and dig return no record |
| 09:00.07 | mitcheloc | echo "64.233.167.125 talk.google.com" >> /etc/hosts |
| 09:01.01 | L|NUX | ok |
| 09:01.09 | L|NUX | doing it |
| 09:01.17 | asterboy | Seems my Linux source is missing Modules.symvers, Zaptel gets really pissed: |
| 09:01.18 | asterboy | http://pastebin.ca/112883 |
| 09:01.29 | asterboy | any insight? |
| 09:04.12 | L|NUX | same |
| 09:04.13 | L|NUX | :( |
| 09:04.28 | L|NUX | mitcheloc : no luck |
| 09:04.50 | mitcheloc | L|NUX, it may be trying srv lookups (but i doubt it), you need to fix your dns then so talk.google.com resolves properly |
| 09:05.12 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 09:05.44 | asterboy | http://www.linuxfromscratch.org/lfs/view/stable/chapter08/kernel.html |
| 09:05.46 | asterboy | broken link |
| 09:05.58 | L|NUX | talk.google.com. 43149 IN CNAME talk.l.google.com. |
| 09:05.58 | L|NUX | talk.l.google.com. 61 IN A 216.239.37.125 |
| 09:06.25 | mitcheloc | i'm sure they have more then one server ;) |
| 09:06.27 | L|NUX | its connected now |
| 09:06.28 | L|NUX | :) |
| 09:06.34 | mitcheloc | okay great |
| 09:06.36 | L|NUX | mitcheloc : nah onlye one server :) |
| 09:06.45 | L|NUX | how can i call f4fahmed@gmail.com |
| 09:06.45 | L|NUX | ? |
| 09:06.58 | asterboy | Zaptel won't compile because it complains my linux source is missing the Modules.symvers |
| 09:07.00 | mitcheloc | open google talk, add it to your buddy list and dial away |
| 09:07.01 | asterboy | http://pastebin.ca/112883 |
| 09:07.06 | mitcheloc | (using the call button) |
| 09:07.27 | asterboy | now how do I get the Modules.symvers file? |
| 09:07.37 | tuck3r | what distro? |
| 09:07.41 | mitcheloc | change autoprune to no though, it should work better |
| 09:07.45 | L|NUX | <mitcheloc> : its say Call with dost4u@gmail.com not answered |
| 09:08.07 | asterboy | lfs |
| 09:08.13 | asterboy | Linuxfromscratch |
| 09:08.22 | tuck3r | then you should know... |
| 09:08.29 | mitcheloc | did the buddy dost4u go online from asterisk starting? |
| 09:08.55 | L|NUX | yeah |
| 09:09.00 | L|NUX | its in my list :) |
| 09:09.03 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:09.16 | mitcheloc | watch the CLI for helpful information ? |
| 09:10.47 | L|NUX | checking |
| 09:12.20 | L|NUX | mitcheloc : http://pastebin.ca/112938 |
| 09:13.13 | mitcheloc | [Aug 3 09:17:31] WARNING[11000]: pbx.c:2313 __ast_pbx_run: Channel 'Jingle/f4fahmed-fa3a' sent into invalid extension 's' in context 'default', but no invalid handler |
| 09:13.18 | *** join/#asterisk effectiveape (n=nick@82.153.22.16) |
| 09:15.22 | L|NUX | hummm |
| 09:15.37 | L|NUX | exten => 1,1,Dial(Jingle/asterisk/f4fahmed@gmail.com) |
| 09:15.46 | L|NUX | this is the only line i have in [default] |
| 09:16.25 | mitcheloc | well, you need a dial plan, especially to handle the "s" extension since asterisk is looking for it |
| 09:17.06 | L|NUX | hummm |
| 09:17.15 | effectiveape | I'm having problems in getting my cisco 7940 phones to transfer (when the call has come in from the outside (Zap)). But not when it's from internal. Also the music on hold only activates on internal too. I'd thought it might be something to do with tT but isn't that just for the *2 feature and not a phone feature? |
| 09:17.36 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 09:17.51 | L|NUX | mitcheloc : hummmm |
| 09:17.57 | L|NUX | got it |
| 09:17.57 | L|NUX | :) |
| 09:22.23 | L|NUX | done :) |
| 09:22.50 | mitcheloc | cool |
| 09:23.03 | mitcheloc | is it good quality? |
| 09:24.04 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.130) |
| 09:24.34 | effectiveape | can anyone help? |
| 09:26.06 | mitcheloc | The 7940's have a transfer button on them, it "just works"? |
| 09:26.09 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.156) |
| 09:26.11 | tengulre11 | hi,al |
| 09:26.35 | effectiveape | mitcheloc: It doesn't 'just work' for me |
| 09:26.45 | effectiveape | I just get transfer failed when i use it |
| 09:26.47 | tengulre11 | which codecs suit to me ? I running asterisk on public network, and client using X-let! |
| 09:26.54 | mitcheloc | effectiveape, is everything on the same LAN? |
| 09:27.21 | tengulre11 | I using GSM codecs, but the voice is not nice! |
| 09:27.25 | effectiveape | mitcheloc: Yep i just have all the phones on 10.0.2.* and an incomming isdn |
| 09:29.12 | mitcheloc | effectiveape, not sure then, it has always "just worked" for me! |
| 09:29.19 | asterboy | what does this mean? http://pastebin.ca/112965 |
| 09:30.05 | asterboy | Trying "make install" zaptel 1.2.7 give me this error: No suitable location for module rules can be found... exiting. |
| 09:30.09 | effectiveape | can you show me your extensions/zap/sip configs? |
| 09:30.16 | effectiveape | Maybe something i'm missing |
| 09:30.31 | effectiveape | mitcheloc: which phones you using |
| 09:32.31 | mitcheloc | effectiveape, 7940's, 7910's and 7960's |
| 09:32.34 | mitcheloc | you are using SIP right? |
| 09:32.57 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 09:33.08 | effectiveape | mitcheloc: yeah |
| 09:33.13 | effectiveape | 8.2 |
| 09:33.19 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 09:33.23 | tengulre | hi,all |
| 09:33.27 | mitcheloc | nice, i'm on 7.*, anything neat in 8? |
| 09:34.11 | tengulre | which codecs suit me ? my asterisk runing on public network. and the client using X-lite. |
| 09:35.35 | tengulre | I have a other problem, which OS best suit to asterisk? |
| 09:35.54 | tuck3r | linux? |
| 09:36.03 | L|NUX | tuck3r : yup |
| 09:36.05 | tengulre | tuck3r: yes! |
| 09:36.06 | mitcheloc | OSX! |
| 09:36.11 | tuck3r | windows! |
| 09:36.24 | mitcheloc | no stupid, you can't run it on there |
| 09:36.25 | tengulre | which linux? |
| 09:36.29 | L|NUX | mitcheloc : have not checked quality |
| 09:36.37 | tuck3r | tengulre, one you like? |
| 09:36.56 | tengulre | tuck3r, I like debian, redhat ...... |
| 09:37.12 | tengulre | and so on. |
| 09:37.21 | tuck3r | use it then? |
| 09:37.21 | asterboy | build_tools/genmodconf linux26 "" "tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp pciradio ztd-loc ztdummy" |
| 09:37.38 | asterboy | No suitable location for module rules can be found... |
| 09:37.41 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:37.42 | tengulre | tuck3r: which linux are u using? |
| 09:37.44 | asterboy | genmodconf? |
| 09:37.55 | effectiveape | mitcheloc: appart from transfer it seems ;) |
| 09:37.58 | tuck3r | tengulre: gentoo |
| 09:38.03 | L|NUX | wow |
| 09:38.09 | L|NUX | gentoo is great ;) |
| 09:38.13 | tengulre | COOL! |
| 09:38.13 | tuck3r | yes ;) |
| 09:38.14 | asterboy | genpoo |
| 09:38.18 | ailantian | someone could help me ? asterisk 1.2.10 reply to my SUBSCRIBE message with expire:0,I can not subscribe to asterisk |
| 09:38.18 | L|NUX | not every one can install it ;) |
| 09:38.19 | L|NUX | hehe |
| 09:38.30 | asterboy | try lfs |
| 09:38.34 | tuck3r | ast_freak: this from the person that is having problems installing zaptel |
| 09:38.34 | effectiveape | mitcheloc: any chance you could pastebin your confs? |
| 09:38.44 | tengulre | but I think gentoo is hard to config. I m a beginner. |
| 09:39.00 | mitcheloc | effectiveape, sorry, not tonight, reformatting my laptop... |
| 09:39.09 | tuck3r | then use something else |
| 09:39.57 | mitcheloc | tengulre, you can use the automated.it guide and the suggested OS |
| 09:40.21 | tengulre | mitcheloc: oh! |
| 09:40.26 | mitcheloc | oh wait, that is such a last year response |
| 09:40.28 | mitcheloc | ~book |
| 09:40.30 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:40.33 | mitcheloc | ^^ thats what you want |
| 09:41.27 | tengulre | mitcheloc: which audio codec are you using? |
| 09:41.50 | tuck3r | really there is a chart on the 'net |
| 09:42.09 | tuck3r | gsm or g729 are the lower bandwith codecs |
| 09:43.01 | mitcheloc | tengulre, start with the book ;) |
| 09:45.42 | tengulre | tuck3r,mitcheloc: thanks.. |
| 09:46.26 | yxa | if i have a sip account [isp] that i can dial using sip/prefix+number@isp, how can I let another * box and its users dial using that too? |
| 09:46.35 | tengulre | but when I using gsm codec, the voice is not nice. why? in sip.conf, I writed disallow=all allow=gsm |
| 09:46.47 | tuck3r | yxa: trunks, dundi... |
| 09:47.10 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 09:47.17 | yxa | tuck3r can sip be trunked? |
| 09:47.25 | tuck3r | afaik yes |
| 09:47.51 | JT | first i've heard of that tuck3r |
| 09:48.08 | asterboy | ok, looks like zaptel wants modules.conf in /etc |
| 09:48.16 | tuck3r | then use IAX :P |
| 09:48.38 | yxa | how do I go about doing that? |
| 09:48.38 | asterboy | don't have modules loading so I just created the file and it installed...although I don't know if that will be a problem later on trying to run. |
| 09:48.39 | tuck3r | does it really matter? it is connecting the asterisk boxes not making the call |
| 09:49.22 | tuck3r | asterboy: thats what you get for using LFS :P |
| 09:49.39 | effectiveape | the transfger problem shows in the logs as "Attempting native bridge of SIP/14-9c46 and SIP/12-6ca3" |
| 09:49.46 | effectiveape | but then fails. |
| 09:50.00 | tengulre | tuck3r: do u know why? :-D) |
| 09:50.12 | tuck3r | know what? |
| 09:50.29 | asterboy | lol |
| 09:50.37 | asterboy | ya, it's a tight system alright |
| 09:51.05 | asterboy | 15 processes running |
| 09:51.05 | tuck3r | gentoo is as far as i will go, LFS is just a waist of time |
| 09:51.18 | asterboy | lol, no way its a waste of time |
| 09:51.44 | asterboy | I used jhalfs to automate the install. |
| 09:51.44 | yxa | is trunking the only solution to my requirements? Can I just let user dial like number@box@isp@remote box? |
| 09:51.44 | tengulre | tuck3r: but when I using gsm codec, the voice is not nice. why? in sip.conf, I writed disallow=all allow=gsm |
| 09:51.47 | tuck3r | no point at all... I can run the same system with compiling my hand |
| 09:52.09 | tuck3r | tengulre: get a better net connecting? |
| 09:52.23 | tengulre | tuck3r: yes! |
| 09:52.39 | effectiveape | anyneen? |
| 09:52.43 | effectiveape | argh ;) |
| 09:53.45 | tengulre | ?? |
| 09:54.04 | tuck3r | ??? |
| 09:54.26 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 09:54.28 | tengulre | my IP: 61.152.252.29, SIP user: 0212005 secret:0212005, you can test it! |
| 09:54.50 | fourcheeze | Got weird system-wide problem here |
| 09:55.00 | tengulre | other man don't attack it! :) |
| 09:55.00 | fourcheeze | it seems to be random and intermittent |
| 09:55.19 | tengulre | my ID is 0212002 |
| 09:55.24 | tengulre | you can calling it me! |
| 09:55.31 | fourcheeze | from time to time 2/3 of my sip clients find themselves unable to register |
| 09:55.43 | fourcheeze | asterisk just loses them |
| 09:55.46 | fourcheeze | I'm on 1.2.7 |
| 09:55.54 | fourcheeze | (realtime) |
| 09:56.00 | fourcheeze | any thoughts/clues/ideas/shots in the dark? |
| 09:57.21 | tengulre | tuck3r: did u logon in my asterisk? |
| 09:57.29 | tuck3r | tengulre: uh no |
| 09:57.39 | tengulre | :( |
| 09:58.03 | tengulre | anybody can help me to see what problem? |
| 09:58.25 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 09:58.28 | *** join/#asterisk lorinc (n=ang@caracas-3961.adsl.interware.hu) |
| 10:00.02 | tengulre | tuck3r: do you busying now? |
| 10:00.14 | tuck3r | uh yes |
| 10:00.47 | tengulre | sorry for trouble you! |
| 10:00.47 | mitcheloc | tengulre, please read the book, after that, people will be more keen to help you |
| 10:02.19 | *** join/#asterisk somegeek_ (i=levin@tor/regular/somegeek) |
| 10:03.00 | tengulre | mitcheloc: thank you very much! < my english is lower, crying.......> |
| 10:04.43 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 10:04.57 | *** join/#asterisk phearless (n=phearles@host81-138-68-106.in-addr.btopenworld.com) |
| 10:05.29 | E-bola | What should i do if the ppl i call complain they can barely hear my (volume too low) while they are more than loud enough for me to hear.? |
| 10:05.33 | E-bola | Adjust the phones gain to +6 |
| 10:05.39 | E-bola | or adjust it in asterisk? |
| 10:05.42 | E-bola | or something else |
| 10:08.15 | ailantian | someone could help me? my sip phone can not subscribe to asterisk 1.2.10, asterisk always reply the subscribe package with 200 OK ,but the expires is 0,why? |
| 10:08.26 | fourcheeze | phew I'm back |
| 10:08.41 | fourcheeze | anyone cast any light on my lost registrations issue? |
| 10:08.45 | fourcheeze | could that be a routing issue? |
| 10:09.02 | pif | hi, is someone using an isdn gigaset connected to a quadBRI ? |
| 10:09.23 | l-fy | hi pi |
| 10:09.26 | l-fy | hi pif |
| 10:09.31 | l-fy | where is Hercule? |
| 10:09.33 | phearless | hello folks |
| 10:09.44 | phearless | I do NOT have any questions :-) |
| 10:09.57 | l-fy | phearless > i don't even use asterisk :) |
| 10:10.13 | fourcheeze | phearless: then don't ask them here ;-) |
| 10:10.23 | phearless | I will use it, but not now.. |
| 10:10.38 | phearless | I am starting to investigate about * |
| 10:11.01 | ailantian | can i tell asterisk route all my rtp streams? |
| 10:11.01 | pif | hi l-fy |
| 10:11.19 | fourcheeze | l-fy: what you doing here? |
| 10:11.26 | l-fy | fourcheeze > trolling :) |
| 10:11.30 | fourcheeze | cool |
| 10:11.34 | l-fy | and looking for good looking guys |
| 10:11.37 | l-fy | do you know any? |
| 10:11.42 | tuck3r | wtf? |
| 10:11.43 | l-fy | and is not gay |
| 10:11.46 | tuck3r | lol |
| 10:12.19 | l-fy | so what? |
| 10:12.19 | fourcheeze | so I'll declare myself out |
| 10:12.21 | l-fy | i live in Europe |
| 10:12.26 | fourcheeze | yeah, me too |
| 10:12.26 | l-fy | :)))))))))) |
| 10:12.51 | l-fy | fourcheeze > where you from? |
| 10:12.56 | tuck3r | you two get a room(channel?) |
| 10:12.57 | fourcheeze | England |
| 10:12.58 | *** join/#asterisk tdonahue-laptop (n=tdonahue@seymour-cuda1-69-173-87-106.albyny.adelphia.net) |
| 10:13.14 | l-fy | fourcheeze > and since when being married was a problem for an english man? :)))) |
| 10:13.15 | tdonahue-laptop | morning all |
| 10:13.27 | l-fy | hi tdonahue-laptop |
| 10:13.31 | fourcheeze | oh, being married is no problem - I quite like it |
| 10:13.50 | l-fy | tuck3r > actually i don't use * i use the competition of * and that was a friendly chat :) |
| 10:13.51 | fourcheeze | anyone using sipx on the front of * ? |
| 10:14.14 | l-fy | fourcheeze > why in the earth will someone use sipx? |
| 10:14.26 | tdonahue-laptop | l-fy, i find being single much less expensive ;-) |
| 10:14.28 | fourcheeze | well, that's what I want to know |
| 10:14.47 | fourcheeze | apparently some people use it on the front of * |
| 10:15.04 | l-fy | tdonahue-laptop > well, i have to admit that since making money is a natural skill for me that wasn't a problem in the last 2 years since i have the company |
| 10:15.33 | tdonahue-laptop | well then it all works out for the best, doesn't it :) |
| 10:17.11 | firstknight | can I send data ( something like attachment ) between 2 sip clients and have asterisk recorded the data in server? |
| 10:17.26 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:17.45 | tuck3r | night all |
| 10:22.52 | L|NUX | mitcheloc : you still arround ? |
| 10:23.10 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 10:23.22 | tdonahue-laptop | has anyone heard of a problem with cisco phones not turning on the MWI until nearly 1 hour after the message arives? |
| 10:24.09 | tdonahue-laptop | the only thing i can come up with is it might be related to "qualify=no |
| 10:24.10 | tdonahue-laptop | " |
| 10:28.07 | *** join/#asterisk rdsousa (n=RDS@213.205.87.88) |
| 10:28.18 | pif | pop quizz, what is the best wi-fone available? |
| 10:28.41 | pif | I'm pretty happy with the Utstarcom f1000G but is has no desktop charger |
| 10:29.30 | rdsousa | there's any dect ip solution available? |
| 10:29.59 | rdsousa | so that's possible to use siemens gigaset dect phones? |
| 10:34.35 | l-fy | yes rdsousa |
| 10:34.40 | l-fy | www.yoda.com.tw |
| 10:34.48 | l-fy | and look for the access point with dect |
| 10:35.06 | pif | l-fy: have you tried it? |
| 10:36.17 | l-fy | pif > nope but yoda is a great company |
| 10:36.54 | pif | the web site is a bit shitty |
| 10:37.07 | l-fy | true but their equipments are some of them best |
| 10:37.08 | pif | can't find the dect |
| 10:37.26 | RoyK | l-fy: do you use yoda stuff? |
| 10:37.37 | *** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net) |
| 10:37.39 | rdsousa | but it allows roaming? |
| 10:37.45 | l-fy | RoyK > just that that you gave me |
| 10:37.47 | RoyK | ah |
| 10:37.48 | RoyK | ok |
| 10:37.49 | rdsousa | between antenas |
| 10:37.59 | l-fy | and i still remeber their great support |
| 10:38.12 | RoyK | hehe |
| 10:39.23 | l-fy | RoyK > how are those ata's? |
| 10:39.26 | l-fy | aren't great? |
| 10:39.55 | pif | analog>digital>analgog == shit |
| 10:40.11 | pif | yeah, anal_gog ;) |
| 10:41.22 | RoyK | l-fy: sure, they work |
| 10:41.29 | RoyK | but not really easily configurable |
| 10:41.33 | WildPikachu | sorry, stupid question, is FXS connected to a phone or PSTN? |
| 10:41.46 | RoyK | i mean, you can't do a lot of tweaking, and provisioning them sucks big time |
| 10:43.38 | rdsousa | i want a solution for 40-50 dect phones |
| 10:43.45 | rdsousa | not just 4 or 5 |
| 10:44.02 | rdsousa | to cover big areas like warehouses |
| 10:44.11 | pif | use wi-fones then |
| 10:44.27 | rdsousa | i've already tested the f1000 |
| 10:44.35 | pif | and? |
| 10:44.35 | rdsousa | and they sucked big time |
| 10:44.48 | pif | i use the f1000g with good results |
| 10:44.53 | pif | what sucked? |
| 10:44.56 | rdsousa | very low volume |
| 10:45.12 | rdsousa | losing registry |
| 10:45.19 | pif | not the 1000g |
| 10:45.34 | pif | and it roams fine betw AP's |
| 10:48.53 | effectiveape | Time to ask again ;) - I'm getting a transfer failed (on cisco 7940) when i'm attempting a transfer from Zap->Sip to another Sip. This is the log of what happens... On the phone i just get "transfer failed" |
| 10:49.04 | effectiveape | http://pastebin.ca/113065 |
| 10:49.35 | effectiveape | Also i don't get music on hold through the Zap channel to the calling party which could be related |
| 10:52.25 | pif | effectiveape : so incoming call from zap, then transfer attempt to another cisco? |
| 10:53.25 | effectiveape | pif: yeah |
| 10:53.33 | rdsousa | i don't know if it is the 1000g |
| 10:54.30 | rdsousa | i think mine is the f1000 |
| 10:54.33 | rdsousa | not f1000g |
| 10:54.41 | rdsousa | there's any difference |
| 10:54.42 | rdsousa | ? |
| 10:55.40 | pif | firmware is different and chiset too, same shell |
| 10:56.06 | pif | chipset |
| 10:56.24 | pif | to make sure, watch to bootup logo |
| 10:56.34 | *** join/#asterisk kiddy (n=achu@59.93.14.150) |
| 10:57.03 | kiddy | I am not able to use the festival feature in asterisk |
| 10:57.11 | pif | effectiveape : using * or cisco transfer? |
| 10:57.30 | effectiveape | pif: cisco |
| 10:57.39 | pif | * version? |
| 10:57.45 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 10:57.49 | kiddy | when I dial to a particular extension the log shows " == Spawn extension (from-internal, 4444, 2) exited non-zero on 'SIP/103-09c40e20'" |
| 10:57.54 | pif | other working phones on the same * ? |
| 10:57.54 | effectiveape | bristuff 1.2 |
| 10:58.00 | kiddy | Please help me to solve the problem |
| 10:58.02 | effectiveape | none are |
| 10:58.12 | pif | other brands? |
| 10:58.18 | kiddy | I didn't find any documents for solving the problem |
| 11:02.42 | kiddy | no idea ? |
| 11:04.18 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 11:11.05 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-59-55.cybersurf.com) [NETSPLIT VICTIM] |
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| 11:15.04 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 11:18.08 | icyfire0573 | kiddy, try dialing an extension that dosen't exist and see if the same thing happens. If it does the extensions aren't set up right. Also, make sure someone/something is logged in on that extension. I don't know too much about Asterisk, but there is some place to start. Also, im leaving for work now so I can't be much more help than that. |
| 11:27.20 | *** join/#asterisk DiG__ (n=dgetz@pd95b7911.dip0.t-ipconnect.de) |
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| 12:32.32 | MrChimpy | hello |
| 12:32.48 | MrChimpy | 13:39:02) swtashcandie943893: dont tell anyone but i want to suck you off so bad...ive seen your profile and i live near you...lets get together msg my other name crazypartygurl4u |
| 12:32.52 | MrChimpy | she sounds nice |
| 12:33.41 | *** join/#asterisk Trakkasure (n=Sgemtum@69-163-145-203.atlsfl.adelphia.net) |
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| 12:46.08 | expat_iain | Anyone have a link on how to setup a Grandstream GXP2000 phone to hotdesk logins? |
| 12:52.08 | *** join/#asterisk coppice (n=chatzill@185.202.17.210.dyn.pacific.net.hk) |
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| 13:17.57 | fourcheeze | are there any ulimits I should be setting on my * server? |
| 13:25.23 | *** join/#asterisk Skaag (n=hintza@212.199.180.157.static.012.net.il) |
| 13:25.47 | *** join/#asterisk popvoxdave (n=popvoxda@c-71-206-59-174.hsd1.md.comcast.net) |
| 13:26.52 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 13:27.05 | Skaag | is destar recommended for managing an asterisk? |
| 13:27.10 | Skaag | or should I stick with AMP? |
| 13:30.07 | fourcheeze | Emacs |
| 13:31.06 | FlatFoot | hello all |
| 13:31.07 | [TK]D-Fender | Skaag :vi |
| 13:31.37 | hmmhesays | i'm having a mental block here |
| 13:31.47 | hmmhesays | how do I list my hard disks in linux? |
| 13:31.58 | creativx | /dev |
| 13:32.07 | creativx | also known as.. devices |
| 13:32.19 | hmmhesays | lol |
| 13:32.27 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 13:32.46 | hmmhesays | we'll not speak of this |
| 13:33.07 | Skaag | I want to take a wooden club, or maybe a chainsaw, and hack at people who say vi |
| 13:33.18 | Skaag | but hey i'm sure deep, deep inside you guys mean well |
| 13:33.34 | fourcheeze | Skaag: well emacs |
| 13:34.03 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 13:34.05 | Skaag | I want to learn from example |
| 13:34.21 | Skaag | I thought it would be nice to play with a GUI and see what it generates |
| 13:35.10 | Skaag | i'm having problems connecting Ekiga to my asterisk |
| 13:35.19 | hmmhesays | which gui? |
| 13:35.24 | [TK]D-Fender | Skaag : Actually.... I suck at vi, emacs, and just about every other linux CLI editor out there. I suppose I could "make do" with nano but can't be bothered. I use MC personally :) |
| 13:35.25 | Skaag | anyone here has a sample user block that works for Ekiga? |
| 13:35.39 | Skaag | [TK]D-Fender: I use emacs |
| 13:35.45 | Skaag | it's really great |
| 13:35.54 | Skaag | for quickies I use Joe, or Vi |
| 13:36.10 | Skaag | hmmhesays: I tried destar but it can't see my instance of asterisk |
| 13:38.03 | hmmhesays | ever tried this great gui known as VI? |
| 13:38.44 | [TK]D-Fender | hmmhesays : We've just been over that.... get some coffee |
| 13:41.10 | Skaag | hmmhesays: are you implying you know the configuration files of Asterisk so well that you really don't need the GUI and that VI is good enough? |
| 13:41.37 | [TK]D-Fender | Skaag : Umm.... virtually everyone here does.... |
| 13:41.38 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 13:41.52 | hmmhesays | [TK]D-Fender: haha |
| 13:42.00 | ManxPower | Yes, very few people here use gui config |
| 13:42.01 | hmmhesays | yeah pretty much, although I do use freepbx |
| 13:42.20 | rob0 | <== mc :) |
| 13:42.28 | ManxPower | VI: The User Hostile Editor. |
| 13:42.30 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 13:42.36 | ManxPower | I use "joe" or "jEdit" |
| 13:42.37 | hmmhesays | nano is good too |
| 13:42.46 | tuck3r | VI kills kittens |
| 13:42.48 | hmmhesays | i use kwrite if I'm in gnome or kde |
| 13:42.59 | hmmhesays | I rather like kwrite |
| 13:44.08 | hmmhesays | VI gives you hairy palms |
| 13:44.34 | hmmhesays | so angus picks out every note on the thunderstruck riff |
| 13:44.41 | hmmhesays | that... SUCKS |
| 13:44.54 | phearless | if I well understood : |
| 13:45.05 | phearless | trixbox is a linux distro that include asterisk and some tools |
| 13:45.12 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 13:45.19 | phearless | and freepbx is a php application that control asterisk ? |
| 13:45.24 | hmmhesays | I went to play it last night, my speed picking needs work |
| 13:45.28 | [TK]D-Fender | hmmhesays : Funny that the video shows it as hammer-on/offs |
| 13:45.38 | phearless | but freepbx is included in trixbox ? |
| 13:45.46 | [TK]D-Fender | phearless : Correct on both counts |
| 13:46.00 | phearless | ok cool |
| 13:46.01 | hmmhesays | [TK]D-Fender: watch some live clips, he's picking it all |
| 13:46.07 | [TK]D-Fender | phearless : And with any hope you'll realize that YOU want to control * instead of some GUI |
| 13:46.09 | phearless | I will try to find a pc to install trixbox |
| 13:46.19 | [TK]D-Fender | hmmhesays : It did sound a little plucked out... |
| 13:46.23 | FlatFoot | been searching for an age for an answer to this ..... chan_sip.c:681 retrans_pkt: anyone got an idea ? |
| 13:46.27 | hmmhesays | [TK]D-Fender: oh come on, I like freepbx |
| 13:46.34 | [TK]D-Fender | phearless : Just remember not to come here with problems on it. |
| 13:46.39 | hmmhesays | FlatFoot: you have a call failing |
| 13:46.44 | hmmhesays | sip debug |
| 13:46.53 | phearless | [TK]D-Fender: promised. |
| 13:46.59 | [TK]D-Fender | hmmhesays : You know better and know when to use it and not to nag here about it. You're forgiven :) |
| 13:47.00 | FlatFoot | ok that puts me in the right direction thanks |
| 13:47.09 | hmmhesays | [TK]D-Fender: haha true |
| 13:47.19 | noname32 | anyone know if there is a way to use asterisk or any other programs that are comparable to something like webex or other online meeting rooms |
| 13:47.52 | hmmhesays | I tried out app_conference yesterday on 1.2.10, it segfaults when ilbc is used |
| 13:48.43 | [TK]D-Fender | hmmhesays : Dang, and I was preparing to test it myself... |
| 13:48.46 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 13:49.00 | [TK]D-Fender | hmmhesays : Oh wait... on on that codec? |
| 13:49.26 | hmmhesays | it compiles clean, make install doesn't work, g.729 / ulaw / alaw work, but you get a second call into a conf using ilbc and it takes asterisk down |
| 13:49.31 | [TK]D-Fender | ilbc was the alternate name for a song by Styx :) |
| 13:49.47 | *** join/#asterisk steveaj (n=steve@82-71-15-37.dsl.in-addr.zen.co.uk) |
| 13:50.07 | hmmhesays | haha |
| 13:52.33 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.22.Dial1.SanJose1.Level3.net) |
| 13:52.34 | noname32 | hmmhesays, hmm i am looks for something that can do a kind of like video relay like sharing desktop of the leader so that you can give like presentations and stuff it doesnt look like that is as complex as i am looking any other sugestions? |
| 13:52.40 | *** join/#asterisk destructure (n=irish@rrcs-24-173-126-174.se.biz.rr.com) |
| 13:53.13 | *** part/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer) |
| 13:53.49 | [TK]D-Fender | noname32 : VNC with web-deployed Java viewer |
| 13:54.18 | noname32 | i tryed that but it seemed kinda jumpy at times |
| 13:54.49 | noname32 | so far netmeeting has worked the best but realy would prefer a web based option |
| 13:54.58 | [TK]D-Fender | noname32 : Use a compressed server like TRicghVNC. Best odds. that or a screenshot based web updater maybe |
| 13:55.11 | hmmhesays | or tightvnc |
| 13:55.22 | noname32 | hmm |
| 13:55.30 | noname32 | not sure i looked in to those |
| 13:55.34 | noname32 | thanks guys |
| 13:56.00 | MrChimpy | what auto generates include/asterisk/version.h? |
| 13:56.07 | hmmhesays | sadly enough microsofts rdp puts the smack down on vnc |
| 13:56.42 | [TK]D-Fender | TightVNC .... blarg... can't type today |
| 13:57.05 | [TK]D-Fender | hmmhesays : Yeah MS opens all sorts of doors, esp in the back ;) |
| 13:57.06 | ManxPower | VNC can't really compare to pcAnywhere or, I imagine, RDP |
| 13:57.39 | *** join/#asterisk Unistim_junky (n=rover@216-230-88-10.client.cypresscom.net) |
| 13:57.53 | [TK]D-Fender | ManxPower : Does the job just fine multi-platform and free. Just not a great "presentation" tool |
| 13:58.07 | ManxPower | [TK]D-Fender, VNC is slow, even when it's compressed. |
| 13:58.42 | nortex | What do most people do for music on hold content? |
| 13:58.46 | tuxick | there's several vncs |
| 13:58.55 | tuxick | think tightvnc was rather quick |
| 13:59.01 | [TK]D-Fender | nortex : MP3's seem to be popular... |
| 13:59.13 | tuxick | alst setting 8bit colors helps :) |
| 13:59.24 | Skaag | ManxPower: Joe is my favorite of all, small, simple, powerful enough for mos |
| 13:59.26 | Skaag | most |
| 13:59.46 | Skaag | [TK]D-Fender: so you all know the configs by heart but nobody can answer my questions ;-) |
| 14:00.04 | ManxPower | tuxick, run it over a 384k frame relay connection sometime. |
| 14:00.11 | [TK]D-Fender | Skaag : Funny I don't remember you asking anything.... |
| 14:00.27 | tuxick | ManxPower: no thanks :) |
| 14:00.30 | [TK]D-Fender | Skaag : Not since joining 45 mins ago..... |
| 14:00.39 | Skaag | I have a problem connecting Ekiga to asterisk |
| 14:00.42 | Skaag | I was here last night |
| 14:00.46 | Skaag | ;-) |
| 14:00.46 | *** join/#asterisk iq (n=IQ@unaffiliated/iq) |
| 14:01.03 | [TK]D-Fender | Skaag : Sorry.. I left my flux capacitor at home.... some details would be helpful... |
| 14:01.27 | ManxPower | What the hell is an Ekiga? |
| 14:01.34 | hmmhesays | you think I could get 100+ transcoded calls across a single 3.0ghz xeon box? |
| 14:01.35 | Skaag | well right now i've moved the cdr to sqlite3 and my asterisk can't find the file cdr_sqlite3_custom.so so i'm trying to locate the package that contains it |
| 14:01.42 | Skaag | ManxPower: It's a softphone for linux |
| 14:01.59 | [TK]D-Fender | Skaag : Sounds like part of asterisk-addons. |
| 14:02.17 | e-ddie | does anyone know where to define what number to put as src, when transferring a call? like, if i call from one nr, and the destination transfers the call, the src number for the transfer will be registered as the src for the one calling to the person transferring |
| 14:02.45 | [TK]D-Fender | ManxPower : New name for GnomeMeeting |
| 14:03.01 | nortex | [TK]D-Fender, I guees my question was more about the license/copywrite protections, do most people go out and pay for the royalty free stuff or what. |
| 14:03.07 | hmmhesays | I haven't done any load tests in awhile |
| 14:03.40 | Skaag | [TK]D-Fender: I have no such package in the list |
| 14:03.45 | [TK]D-Fender | nortex : No, most people use stuff illegally. on the WIKI there is a page linking to all sort of royalty free music you can use theough. Mostly student produced stuff in the PD. |
| 14:03.56 | [TK]D-Fender | Skaag : "packages" in what "list"? |
| 14:04.11 | Skaag | in debian |
| 14:04.29 | [TK]D-Fender | Skaag : Most of us here compile from source.... do try to avoid using packaged versions.... you'll likely not find everything you're looking for. |
| 14:04.32 | Skaag | when I search for packages containing the name "asterisk" there is no "asterisk-addons" |
| 14:05.05 | [TK]D-Fender | Skaag : Nobody said the package maintainers were any good did they? USE THE SOURCE LUKE! |
| 14:05.07 | Skaag | [TK]D-Fender: but then I have to recompile everything there's a new version instead of just upgrading the box itself, and I have almost 30 boxes to maintain :-) |
| 14:05.12 | Skaag | ok |
| 14:05.17 | [TK]D-Fender | Skaag : 30 * boxes? |
| 14:05.18 | Skaag | source it is, then |
| 14:05.24 | Skaag | 30 linux boxes |
| 14:05.34 | Skaag | if I had to maintain everything from source, i'd be in trouble |
| 14:05.35 | [TK]D-Fender | Skaag : Fine... use deb's for everything else. |
| 14:05.42 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 14:05.50 | [TK]D-Fender | Skaag : But for your 1 measly little box, get off your ass a bit ;) |
| 14:05.55 | Skaag | my mail system is already something I maintain from source, for example, and it's a pain in the ass but hey it's the best there is |
| 14:06.09 | Skaag | true, * is worth working a bit harder for ;-) |
| 14:06.13 | Skaag | exciting software |
| 14:06.59 | hmmhesays | frustrating software |
| 14:07.15 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:07.22 | [TK]D-Fender | hmmhesays : For so many other people than me. I seem somewhat blessed so far... |
| 14:09.27 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 14:10.16 | hmmhesays | [TK]D-Fender: i've been fed up frustrated, angry |
| 14:10.21 | hmmhesays | all the way to the bank though |
| 14:10.59 | *** join/#asterisk tessier_ (n=treed@gw.drjays.com) |
| 14:12.15 | Skaag | what's the advantage of using freepbx over vanilla asterisk? |
| 14:12.17 | [TK]D-Fender | hmmhesays : You'll be able to afford all sorts of therapy now! |
| 14:12.39 | [TK]D-Fender | Skaag : You don't need an IQ and if your expectations are low you might not be disappointed. |
| 14:13.09 | iq | iq ? |
| 14:13.15 | [TK]D-Fender | Skaag : And it does make some of the slightly more complicated stuff very easy. Like fax-email. but the rest... screw it. |
| 14:13.24 | [TK]D-Fender | iq : SHH! |
| 14:13.38 | [TK]D-Fender | iq : Trixbox users don't need you! |
| 14:13.50 | iq | this is rude :-( |
| 14:13.51 | [TK]D-Fender | (and/or don't HAVE you) |
| 14:13.59 | [TK]D-Fender | lol |
| 14:13.59 | tuxick | iq is overrated! |
| 14:14.02 | Skaag | I'll just stay with vanilla |
| 14:14.06 | iq | lol |
| 14:14.09 | [TK]D-Fender | Skaag : Good call.... |
| 14:15.29 | Skaag | I want to learn it properly and thoroughly |
| 14:16.13 | [TK]D-Fender | Skaag : ... |
| 14:16.15 | [TK]D-Fender | ~docs |
| 14:16.16 | jbot | i guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 14:16.17 | [TK]D-Fender | ~book |
| 14:16.18 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:16.31 | [TK]D-Fender | Start with the book, and use the WIKI for reference |
| 14:16.38 | Skaag | thanks that will be helpful |
| 14:18.16 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 14:18.22 | *** join/#asterisk devoo (n=al@124.6.183.0) |
| 14:19.05 | [TK]D-Fender | jbot : docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 14:19.07 | jbot | ...but docs is already something else... |
| 14:19.12 | [TK]D-Fender | jbot : no, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 14:19.15 | jbot | [TK]D-Fender: okay |
| 14:19.36 | [TK]D-Fender | Time to seriously ditch that hopelessly outdated handbook draft.... |
| 14:19.48 | coppice | jbot: docs can be found at a clinic near you |
| 14:20.02 | [TK]D-Fender | coppice : *FAP* |
| 14:20.04 | *** join/#asterisk Sid99 (n=simon@80.88.192.113) |
| 14:20.14 | [TK]D-Fender | ~docs |
| 14:20.15 | jbot | extra, extra, read all about it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 14:20.30 | devoo | hello. i'm an undergrad student and my thesis is related to asterisk. since asterisk has a lot of features already, are there any feature that are still lacking and underdeveloped that could be created and one that would help a university? |
| 14:20.34 | *** join/#asterisk Crappynick (i=Swoosh@66.173.103.100) |
| 14:20.46 | [TK]D-Fender | devel : SIP-B <----- |
| 14:20.52 | [TK]D-Fender | devoo : SIP-B <----- |
| 14:21.06 | coppice | devoo: all the really important features are missing |
| 14:22.03 | devoo | right |
| 14:22.08 | *** join/#asterisk lehel (n=t@82.77.41.206) |
| 14:22.55 | *** join/#asterisk rec (n=rec@c-67-182-205-227.hsd1.ut.comcast.net) |
| 14:23.06 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.22.Dial1.SanJose1.Level3.net) |
| 14:23.19 | coppice | what is SIP-B? |
| 14:23.33 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.22.Dial1.SanJose1.Level3.net) |
| 14:23.56 | hmmhesays | hmm sixtel has been quite reliable lately |
| 14:24.02 | [TK]D-Fender | coppice : The extensions including SLA (Shared Line Appearances), amongst other functionality. |
| 14:24.23 | vader-- | hmm i need to make a training doc for how to use this new telephone system |
| 14:24.29 | coppice | SIP over TLS seems more interesting than that |
| 14:24.37 | [TK]D-Fender | coppice : SLA is a feature a lot of people turn * down because of its lack of. |
| 14:24.54 | [TK]D-Fender | coppice : Thats back end, not user-visible. |
| 14:25.06 | fourcheeze | [TK]D-Fender: is SLA where more than one person can see a "line" ringing? |
| 14:25.16 | coppice | its still vital and missing |
| 14:25.31 | [TK]D-Fender | fourcheeze : Yes and answer other peoples calls, pick up held calls, etc. |
| 14:25.49 | [TK]D-Fender | coppice : Secure SIP over TCP? |
| 14:25.50 | fourcheeze | sounds useful |
| 14:26.12 | fourcheeze | [TK]D-Fender: how do you define who can/can't pick up another call? |
| 14:26.17 | [TK]D-Fender | fourcheeze : Its the universally requested feature people coming from a Key-System background demand. |
| 14:26.22 | coppice | yep. its actually a SIP requirement, and not an option, these days |
| 14:26.28 | hmmhesays | is there any good phones for SLA these days? |
| 14:26.36 | [TK]D-Fender | fourcheeze : By allowing them to log in as a shared ID for it. |
| 14:28.36 | *** join/#asterisk Un1x (i=Sean@72.61.82.242) |
| 14:28.55 | *** join/#asterisk vlt (n=dm@p54B34C6B.dip0.t-ipconnect.de) |
| 14:29.20 | MrChimpy | ast_log(LOG_NOTICE, "Attempting to start dial sample %s\n",opt_args[OPT_ARG_DIAL_SAMPLE]); |
| 14:29.41 | hmmhesays | I know the snoms were able to subscribe to the hint extensions |
| 14:31.35 | [TK]D-Fender | hmmhesays : Not talking about hints. Hints are a lower grade solution to seeing status. SLA allows you to pick up calls going to and held by the user you have access to. |
| 14:31.55 | hmmhesays | [TK]D-Fender: are there any ip phones out there right now that can do this? |
| 14:32.57 | [TK]D-Fender | hmmhesays : Ton's of them. Linksys, Polycom, etc. |
| 14:33.03 | [TK]D-Fender | hmmhesays : Its * thats behind |
| 14:33.19 | hmmhesays | I wonder what is required to make them work, is there an rfc defined? |
| 14:33.21 | hmmhesays | is it part of sip? |
| 14:33.29 | [TK]D-Fender | hmmhesays : Its also the only way you're gonna get any kind of presence on a Linksys. |
| 14:33.39 | [TK]D-Fender | hmmhesays : Yup.. SIP-B |
| 14:33.49 | hmmhesays | I haven't read up on sla in a long time |
| 14:33.54 | bkw_ | lalala |
| 14:34.12 | hmmhesays | how's cluecon going |
| 14:34.59 | vlt | Hello. In my extensions.conf I have a block of lines for one extension. Prio 1 dials one phone for 15 seconds, prio 102 is Busy(), and prio 2 dials several other phones in case no one answers the first call after 15 sec. Prio 3 then is Hangup(). How can I prevent asterisk from executing prio 2 (dialling all the other phones) after the phone call finishes normally on the first called phone? |
| 14:35.32 | bkw_ | CLUECON ROCKS!!!! |
| 14:35.43 | hmmhesays | i know, i'm sad I can't be there |
| 14:35.54 | [TK]D-Fender | vit : normally dial WILL hang up the whole call with the dirst one is completed. That is unless you get it option "g". |
| 14:35.54 | Un1x | wow |
| 14:36.00 | Un1x | free hardware at cluecon :D |
| 14:36.16 | hmmhesays | unfortunately the cash flow is at at trickle right now |
| 14:36.25 | Un1x | but sad thing is it's on the same days as blackhat. and other conferences |
| 14:36.38 | Un1x | so it kinda cross intersects and doesn't let people attend all of em :p |
| 14:36.40 | hmmhesays | i'm not finding much info on this sip-B |
| 14:37.43 | vlt | [TK]D-Fender: I dial the first one with options "gtT" (without knowing what tat means) ... |
| 14:38.03 | hmmhesays | remove g |
| 14:38.14 | hmmhesays | and send me a bottle of vodka |
| 14:38.41 | vlt | What does "g" stand for? |
| 14:38.43 | [TK]D-Fender | hmmhesays : RFC 3265 |
| 14:39.02 | [TK]D-Fender | vit : (G)o on processing whe the call ends. |
| 14:39.07 | hmmhesays | g stands for get me a sandwich |
| 14:39.23 | vlt | [TK]D-Fender: Aah, ok, thank you. |
| 14:39.24 | [TK]D-Fender | hmmhesays : Section 5.2.3. of the Polycom Admin Guide. |
| 14:39.59 | hmmhesays | i'll check it out |
| 14:41.17 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:41.40 | Skaag | [TK]D-Fender: ok i've installed from source, still can't find cdr_squlite3_custom.so... |
| 14:42.25 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:45.16 | *** join/#asterisk ^HeLL^ (n=root@175.Red-88-8-72.dynamicIP.rima-tde.net) |
| 14:45.21 | ^HeLL^ | hello all |
| 14:45.37 | [TK]D-Fender | Skaag : ... and did you download and compile the addons tarball as well? |
| 14:45.48 | Skaag | yes of course |
| 14:46.39 | Skaag | these are the modules in the addons: format_mp3.so app_saycountpl.so cdr_addon_mysql.so app_addon_sql_mysql.so res_config_mysql.so |
| 14:46.49 | Skaag | cdr_sqlite3_custom.so is not part of the addons |
| 14:47.01 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:47.58 | l-fy | hey marv[work] |
| 14:48.15 | marv[work] | hi |
| 14:48.17 | [TK]D-Fender | Skaag : What exactly is requesting that object? |
| 14:50.00 | Skaag | asterisk itself, in startup |
| 14:50.37 | [TK]D-Fender | Skaag : I'm guessing if may have failed because of missing libs |
| 14:50.47 | Skaag | Aug 3 17:50:33 WARNING[8034]: loader.c:499 load_modules: Loading module cdr_sqlite3_custom.so failed! |
| 14:51.00 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+cdr+SQLite |
| 14:51.01 | lehel | Skaag: noload [?] |
| 14:51.38 | *** join/#asterisk jac]Z[oby (n=me@62.218.44.178) |
| 14:51.39 | [TK]D-Fender | Skaag : And how did you "move" CDR to it? |
| 14:51.41 | jac]Z[oby | Hi |
| 14:51.57 | Skaag | I did not |
| 14:52.00 | jac]Z[oby | anyone can help me with a dialplan thingy regarding country codes |
| 14:52.05 | Skaag | you see, destar uses sqlite3 |
| 14:52.10 | jac]Z[oby | (+49 +43 and so on) |
| 14:52.12 | Skaag | it creates version 3 cdr files |
| 14:52.55 | [TK]D-Fender | Skaag : Sorry.. can't account for cleaning up a b0rked install of a GUI.... I suggest not loading it for the time being |
| 14:53.11 | hmmhesays | cisco have subscribe support in their 7960 firmware yet? |
| 14:53.11 | [TK]D-Fender | Skaag : Did you verify the DB's existance and format? |
| 14:53.27 | Skaag | yes |
| 14:53.30 | *** join/#asterisk ^HeLL^ (n=root@175.Red-88-8-72.dynamicIP.rima-tde.net) |
| 14:54.08 | ^HeLL^ | hello again |
| 14:54.37 | jac]Z[oby | how can i manipulate every outgoing number form +43 to 00 |
| 14:55.21 | [TK]D-Fender | jac]Z[oby : that's all in your pattern matches in your extens... |
| 14:55.38 | jac]Z[oby | ? |
| 14:56.13 | jac]Z[oby | n00b on trixbox , please explain |
| 14:56.13 | [TK]D-Fender | jac]Z[oby : What does your dialplan look like now? |
| 14:56.22 | [TK]D-Fender | jac]Z[oby : Please read the channel topic.... |
| 14:56.32 | jac]Z[oby | oh |
| 14:56.33 | jac]Z[oby | sry |
| 14:56.38 | jac]Z[oby | ok will try there |
| 14:56.41 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:00.05 | *** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com) |
| 15:00.22 | *** part/#asterisk madlynx (n=sleever@195.206.63.10) |
| 15:00.28 | ManxPower | jac]Z[oby, exten => _43XXXXXXX,1,Dial(Zap/g1/00${EXTEN:2}) |
| 15:00.46 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 15:03.23 | hmmhesays | da da da da da |
| 15:03.30 | jac]Z[oby | ManxPower: where do i add this? |
| 15:03.32 | hmmhesays | hmm app_dev state looks interesting |
| 15:03.40 | hmmhesays | app_devstate even |
| 15:03.44 | ManxPower | jac]Z[oby, extensions.conf of course. |
| 15:03.59 | ManxPower | jac]Z[oby, I doubt it will work for you unless you understand how extensions.conf works. |
| 15:04.19 | [TK]D-Fender | ManxPower : .... Trixbox |
| 15:04.30 | jac]Z[oby | ;) i know im sorry |
| 15:04.37 | jac]Z[oby | i am on it |
| 15:04.41 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:04.55 | ManxPower | [TK]D-Fender, I know. I can't help him with the horrible things TrixBox does to Asterisk config files. I cannot help with Trixbox, only with standard Asterisk installs |
| 15:08.34 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 15:08.51 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 15:09.02 | *** part/#asterisk vlt (n=dm@p54B34C6B.dip0.t-ipconnect.de) |
| 15:09.40 | *** join/#asterisk iq (n=IQ@unaffiliated/iq) |
| 15:11.10 | anonymouz666 | coppice: is it hard to implement blocking of collect calls in MFC/R2? |
| 15:11.21 | *** join/#asterisk apardo (n=apardo@87.217.147.51) |
| 15:12.00 | coppice | where are you? |
| 15:12.07 | anonymouz666 | brazil |
| 15:12.23 | MrChimpy | tudo bom? |
| 15:13.02 | anonymouz666 | MrChimpy: hehe |
| 15:13.30 | MrChimpy | my new asterisk patch is ace. it makes asterisk crash when a call is answered. |
| 15:13.58 | coppice | people use a trick for this in brazil. they hang up and reseize quickly. I did something about this, but I can't remember the status. |
| 15:14.25 | *** join/#asterisk s0lid (n=jlq@gr-153-4.eglobalreach.net) |
| 15:15.22 | anonymouz666 | np and thanks. |
| 15:15.29 | anonymouz666 | I will test |
| 15:19.58 | smackus | has anyone set up the sipura spa1001 for provisioning via ftp server? |
| 15:20.00 | *** join/#asterisk tonys (n=tonys@201.216.253.149) |
| 15:20.01 | smackus | is it possible? |
| 15:22.25 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 15:22.30 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 15:22.51 | [TK]D-Fender | smackus : heard of TFTP and HTTP, but not FTP. Might be possible (likely I would think) |
| 15:23.22 | smackus | I dont see anywhere to do this and the instructions with this are worthless... where do you set up the tftp at? |
| 15:23.39 | [TK]D-Fender | smackus : Do you have the admin & deployment guides? |
| 15:23.52 | smackus | for the sipura? |
| 15:23.55 | watchy | tk: i heard it was a bitch to get the deployment forware for sipuras |
| 15:23.56 | [TK]D-Fender | smackus : clearly |
| 15:24.09 | smackus | i only have the little quick start guide... that is all it came with |
| 15:24.26 | [TK]D-Fender | smackus : Go download those guides. The WIKI links them I believe |
| 15:24.27 | watchy | smackus: you gotta contact sipura for all the stuff |
| 15:24.36 | smackus | ok |
| 15:24.39 | [TK]D-Fender | smackus : The quick start guide is a LEAFLET. virtually worthless |
| 15:24.44 | smackus | yep |
| 15:24.52 | watchy | they have some special software from what i understand to create a file with all the settings |
| 15:24.53 | [TK]D-Fender | smackus : Check the WIKI |
| 15:25.07 | smackus | headed there now |
| 15:25.09 | smackus | thanks |
| 15:25.40 | coppice | The quick start leaflet for most things these days might as well be a small scrap of paper that says "YOU ARE ON YOUR OWN" :-) |
| 15:25.50 | watchy | yea it sucks |
| 15:26.11 | watchy | im on a diet tk :( |
| 15:26.24 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 15:27.41 | [TK]D-Fender | watchy : Same here.... |
| 15:27.42 | *** join/#asterisk kfudge (n=kzymxa@24-217-137-21.dhcp.stls.mo.charter.com) |
| 15:28.12 | watchy | im eating wraps everyday they are tasty |
| 15:28.37 | kfudge | Anyone having an issue with putting a call on hold? Apparently on one of my servers, if a call is on hold longer than two minutes it occasionally drops the call |
| 15:30.04 | jbalcomb | Is there an industry term for a server used to test a production for potential implementation? |
| 15:30.33 | eKo1 | jbalcomb: test a production? |
| 15:30.48 | jbalcomb | s/production/product |
| 15:31.02 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 15:31.10 | generalhan | whats up all ?! |
| 15:31.20 | eKo1 | jbalcomb: yes, it is called a test server |
| 15:33.07 | jbalcomb | eKo1: it would be that simple i suppose |
| 15:34.08 | [TK]D-Fender | jbalcomb : Tonight I'm picking up configs for processing. |
| 15:34.52 | *** join/#asterisk salviadud (n=ralfalfa@201.123.130.150) |
| 15:34.55 | jbalcomb | [TK]D-Fender: right on. you test your phone? |
| 15:35.20 | [TK]D-Fender | jbalcomb : Will now. |
| 15:35.47 | generalhan | anyone in here used cisco 7960s with a firmware other than SIP with * ? |
| 15:37.04 | tonys | mgcp |
| 15:37.05 | tonys | _? |
| 15:37.12 | generalhan | i HATE the newest firmware for SIP and i have an extra phone i would like to test other options with ... i just wanna know what the best one is |
| 15:37.15 | [TK]D-Fender | generalhan : Ask Qwell :) |
| 15:37.22 | [TK]D-Fender | jbalcomb : PM |
| 15:37.33 | generalhan | i rember hearing Qwell talking about chan_skinny |
| 15:37.35 | generalhan | or sccp |
| 15:40.03 | Sid99 | ok prob being a bit daft here but can someone answer this ? |
| 15:40.21 | Sid99 | doing a sip debug i can see this line |
| 15:40.51 | Sid99 | Looking for %23 followed by the exten number any idea whats going on ? google was not my friend on this occasion |
| 15:44.04 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 15:49.53 | fourcheeze | any ideas why my asterisk would have lost nearly all its registrations twice this morning - it hasn't done it in the last 6 hours |
| 15:50.19 | eKo1 | fourcheeze: network problems |
| 15:50.45 | fourcheeze | no, I was logged in to the console at the time |
| 15:50.52 | fourcheeze | and my clients are at lots of different sites |
| 15:51.13 | fourcheeze | different ISPs etc |
| 15:51.26 | fourcheeze | unless there's a kind of network problem that only affects UDP on port 5060 |
| 15:52.40 | fourcheeze | eKo1: do you know of such a thing? |
| 15:53.41 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 15:54.11 | sevard | What is the Zap tool to clear dialtone on the zap channels to calibrate the zap lines? |
| 15:54.22 | sevard | I can't find it :/ |
| 15:54.24 | fourcheeze | any idea what sort of things I should be doing to diagnose the problem if/when it happens again |
| 15:54.49 | eKo1 | fourcheeze: do you have any kind of network monitoring tools available? |
| 15:54.58 | fourcheeze | sure |
| 15:55.05 | fourcheeze | I can do a pcap dump I suppose |
| 15:55.14 | fourcheeze | or ethereal or something |
| 15:55.16 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:55.55 | eKo1 | I'm thinking more in the lines of mrtg or smokeping |
| 15:56.04 | eKo1 | or nagios |
| 15:56.21 | *** join/#asterisk Slugs_ (i=1000@eacb02-00-cmmgga-70-34-178-219.atlaga.adelphia.net) |
| 15:58.00 | sevard | fxotune :) |
| 15:58.07 | anonymouz666 | coppice: the trick I know is using flash() for FXO, but this is not applicable to E1 trunks |
| 16:00.34 | blaylock | attempt_thread: Call failed to go through, reason 8 |
| 16:00.43 | blaylock | has anyone ever had that? |
| 16:01.02 | blaylock | trying to figure out if its asterisk screwing up or the other end |
| 16:01.12 | blaylock | its over a PRI |
| 16:04.51 | anonymouz666 | People here lost big projects, just because nobody knows how to block an incoming collect call with R2. |
| 16:05.55 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 16:05.55 | *** join/#asterisk saftsack (n=saftsack@p54A7CB43.dip.t-dialin.net) |
| 16:06.54 | hmmhesays | [TK]D-Fender: you got any good websites for increasing your picking speed? |
| 16:07.08 | sevard | just hold the pick between your fingers. |
| 16:07.08 | sevard | loser. |
| 16:07.09 | hmmhesays | i just watched a youtube video with synaster gates |
| 16:07.15 | hmmhesays | I want to be able to do that |
| 16:08.00 | Qwell | hmmhesays: Yngwie Malmsteen... |
| 16:08.16 | Qwell | haven't seen the guy you said, but...yeah |
| 16:08.22 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 16:08.29 | sevard | !seen dlynes |
| 16:08.41 | Qwell | bbl |
| 16:08.45 | dlynes_laptop | I haven't seen him either |
| 16:08.46 | sevard | where has that squirly guy gone to |
| 16:08.51 | Qwell | dlynes_laptop: should |
| 16:08.51 | ManxPower | anonymouz666, Since Digium does not support R2, that is no suprize. |
| 16:09.11 | hmmhesays | synaster gates is the dude from avenged sevenfold |
| 16:09.16 | sevard | ManxPower: are you sure that's not a surprize |
| 16:09.30 | dlynes_laptop | sevard: did you miss me? ;) |
| 16:09.37 | sevard | by the time fxotune finishes i'll be dead in my grave |
| 16:09.41 | anonymouz666 | Asterisk does support R2 |
| 16:09.45 | sevard | dlynes_laptop: i missed you like a whore misses her highest paying client |
| 16:09.48 | Alric | cpp |
| 16:09.51 | anonymouz666 | coppice wrote the support. |
| 16:09.55 | ^HeLL^ | anybody has tested B410P Digium BRI card? |
| 16:09.59 | ManxPower | sevard, no surpize at all. It would be just as suprizing to find out Microsoft does not support running Windows on MIPS |
| 16:10.13 | sevard | ManxPower: http://www.m-w.com |
| 16:10.31 | sevard | Did you learn how to spell on the back of a box of cherios? |
| 16:10.37 | ManxPower | sevard, prolly |
| 16:10.47 | dlynes_laptop | What's R2? |
| 16:10.56 | sevard | dlynes_laptop: where have you been? |
| 16:10.58 | anonymouz666 | MFC/R2 |
| 16:10.59 | dlynes_laptop | Google's never heard of this specific definition |
| 16:11.01 | dlynes_laptop | ah |
| 16:11.04 | dlynes_laptop | ok |
| 16:11.07 | hmmhesays | R2 evil |
| 16:11.21 | ManxPower | dlynes_laptop, it's a non ISDN protocol for E1 lines. |
| 16:11.23 | sevard | dude, R2 was tight when he kicked that dude's butt in the last movie. |
| 16:11.23 | dlynes_laptop | sevard: sprinkler blew a head in the house |
| 16:11.27 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 16:11.33 | sevard | dlynes_laptop: smoking indoors? |
| 16:11.45 | dlynes_laptop | sevard: no...sprinkler installer screwed up |
| 16:11.56 | sevard | 80 hours later and 2/8 fxo lines tuned |
| 16:12.02 | ManxPower | dlynes_laptop, R@ is done slight different in each country. |
| 16:12.06 | ManxPower | ..er.. R2 that is. |
| 16:12.07 | sevard | dlynes_laptop: did said installer carry insurance? |
| 16:12.25 | dlynes_laptop | sevard: the house owner's thinking about taking him to court...blew a head off the sprinkler system less than four years after it was built |
| 16:12.57 | dlynes_laptop | sevard: normally those things are supposed to be good for 20-25 years |
| 16:12.58 | sevard | lots of damange? |
| 16:13.07 | dlynes_laptop | sevard: about $25K worth, yeah |
| 16:13.13 | sevard | yeah, I know, it's a big issue when one pops |
| 16:13.33 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) |
| 16:13.34 | dlynes_laptop | sevard: well, the water comes out of there at about the same pressure as it comes out of a fire hydrant |
| 16:13.44 | sevard | they're good firesupressors |
| 16:13.55 | dlynes_laptop | sevard: the cable modem and router were directly across the room from the sprinkler head |
| 16:13.57 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 16:14.04 | dlynes_laptop | so needless to say, they got toasted first |
| 16:14.19 | sevard | what about server racks/tvs/smallchildren |
| 16:14.30 | dlynes_laptop | It's a house, not an office |
| 16:14.31 | sevard | i put all of those in a box |
| 16:14.44 | sevard | i have server racks, tvs, and small children in my house |
| 16:14.45 | dlynes_laptop | but no kids upstairs |
| 16:14.52 | dlynes_laptop | neighbors were quite pissed though |
| 16:15.01 | dlynes_laptop | they had two kids |
| 16:15.03 | ManxPower | I don't think you'll get much for the small children. There are plenty of them. |
| 16:15.09 | dlynes_laptop | Yeah, no kidding |
| 16:15.44 | dlynes_laptop | And thankfully none of the leaks came near any electronic equipment (except for the cablemodem and router) |
| 16:16.07 | dlynes_laptop | We had a computer in the same room as the router before, but moved it into the kitchen about 2 months ago :0 |
| 16:16.15 | dlynes_laptop | that was a stroke of luck :) |
| 16:17.23 | sevard | carpet, sheetrock, paneling, trim, paint, all of those cost money |
| 16:17.57 | dlynes_laptop | sevard: not to mention the hardwood laminate flooring |
| 16:18.01 | *** join/#asterisk scorn (n=tim@eclipse.sigint.com) |
| 16:18.13 | dlynes_laptop | sevard: and the ceramic tiles |
| 16:18.21 | scorn | am I in the right place for zaptel support? |
| 16:18.27 | dlynes_laptop | scorn: no |
| 16:18.33 | dlynes_laptop | scorn: you need to go to #windows |
| 16:18.40 | scorn | ouch |
| 16:18.43 | dlynes_laptop | heh |
| 16:18.45 | dlynes_laptop | j/k |
| 16:18.53 | dlynes_laptop | what's the issue? |
| 16:18.54 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-246-132-81.aubnin.fios.verizon.net) |
| 16:18.59 | scorn | gimme a sec, I'm still stunned |
| 16:19.23 | hmmhesays | snap out of it |
| 16:19.38 | coppice | anonymouz666: a few months ago people were asking for collect call handling, and I implemented it. then they stopped asking, and I don't think it got tested properly. If they have lost projects it is entirely their own fault, as I was very willing to make the necessary changes |
| 16:19.41 | dlynes_laptop | hmmhesays: always so subtle |
| 16:20.04 | scorn | I'm trying to get ztmonitor to at the very least acknowledge that I'm sending data over a channel to a digium board. |
| 16:20.19 | hmmhesays | dlynes_laptop: woot |
| 16:20.27 | dlynes_laptop | scorn: is it a tdm400p or tdm2400p board? |
| 16:20.48 | scorn | dlynes_laptop: 205P |
| 16:21.00 | scorn | I can cat /dev/zap/1 and see data |
| 16:21.06 | dlynes_laptop | does ztmonitor even work on pri cards? |
| 16:21.10 | *** join/#asterisk s0lid (n=jlq@gr-153-4.eglobalreach.net) |
| 16:21.14 | scorn | but ztmonitor opens /dev/zap/pseudo |
| 16:21.16 | dlynes_laptop | I thought it only worked on tdm cards? |
| 16:21.43 | dlynes_laptop | scorn: I would use zttest and patlooptest if it's a pri card |
| 16:22.17 | scorn | dlynes_laptop: I'll give it a shot, patlooptest will receive data? |
| 16:22.41 | dlynes_laptop | scorn: patlooptest needs all your channels to be cleared, and it also requires a PRI loop adapter |
| 16:23.04 | dlynes_laptop | scorn: IIRC, that means the orange pair needs to be looped back into the blue pair |
| 16:23.38 | dlynes_laptop | scorn: assuming you're using A standard, that is, and not B standard |
| 16:23.41 | scorn | dlynes_laptop: worth mentioning that I'm not using this in a telephony environment. I'm sending straight PCM data over an E1 channel, no asterisk involved, and I'm wanting to play it out on the sound card. |
| 16:23.55 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 16:24.14 | dlynes_laptop | scorn: patlooptest makes sure that your pri port is even working |
| 16:24.26 | dlynes_laptop | scorn: it tests to make sure you don't have a faulty card |
| 16:24.35 | anonymouz666 | coppice: I totally agree with you. You implemented that in the last version? unicall0.0.3pre9? |
| 16:24.50 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 16:24.51 | *** part/#asterisk KranZ (n=user@sme.bestline.net) |
| 16:24.52 | scorn | dlynes_laptop: okay. I'll give it a try. |
| 16:24.53 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 16:24.59 | scorn | dlynes_laptop: thanks |
| 16:25.05 | dlynes_laptop | scorn: you've already tried zttest then? |
| 16:25.13 | CunningPike | Morning, dlynes_laptop |
| 16:25.18 | dlynes_laptop | scorn: and you're getting scores of 99.975 and higher? |
| 16:25.24 | dlynes_laptop | good morning, cp |
| 16:25.24 | KranZ | ack! http://scoreboard.keynote.com/scoreboard/Main.aspx?Login=Y&Username=public&Password=public |
| 16:25.34 | KranZ | anyone having probs getting to certain sites? |
| 16:25.40 | hmmhesays | so i'm going to try this app_devstate for sla now |
| 16:25.49 | scorn | dlynes_laptop: I tried before, but it didn't seem to work for me, kept giving and error I think, I'm going to try it again because I can't remember the details. |
| 16:25.59 | coppice | anonymouz666: I can't remember. you are the first person to ask about this for months |
| 16:26.22 | dlynes_laptop | scorn: if you're getting errors on that, you might not be invoking it properly |
| 16:26.28 | scorn | dlynes_laptop: I have been using zttool to verify L1 is up, and it doesn't seem to be seeing any data come over either. |
| 16:26.31 | *** join/#asterisk salviadud (n=ralfalfa@201.123.130.150) |
| 16:26.34 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 16:26.49 | *** join/#asterisk apardo (n=apardo@87.217.147.51) |
| 16:27.00 | scorn | dlynes_laptop: But it is acknowledging a connection and a timing source. |
| 16:27.08 | dlynes_laptop | scorn: never used zttool, cause i'm too lazy to install its prerequisites |
| 16:27.24 | scorn | dlynes_laptop: apt a god send. |
| 16:27.25 | anonymouz666 | coppice: heh |
| 16:27.44 | dlynes_laptop | scorn: i've been thinking about installing debian on my laptop |
| 16:27.53 | scorn | dlynes_laptop: God send, sry. |
| 16:27.59 | dlynes_laptop | scorn: i set aside some extra space on it when I installed windows and slackware |
| 16:28.17 | scorn | dlynes_laptop: I love debian, never tried slack. |
| 16:28.22 | dlynes_laptop | scorn: it was specifically for installing other distros...was thinking about trying debian and gentoo on there |
| 16:28.46 | coppice | Unicall was undergoing a major revamp, then I got diverted. Hopefully in a few weeks I will get back to it. I'm doing the EL7 protocol for Unicall at the moment, but that's a kind of sideline activity |
| 16:29.48 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:29.48 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:29.52 | scorn | dlynes_laptop: never tried gentoo either, I went from Mandrake to Debian 4 years ago, so you can probably understand my enthusiasm :) |
| 16:29.59 | dlynes_laptop | yep |
| 16:30.11 | dlynes_laptop | I'm not a big lover of Redhat or anything based on it |
| 16:30.20 | scorn | me either. |
| 16:30.28 | scorn | depends are a nightmare. |
| 16:31.14 | sevard | word to that. |
| 16:31.16 | sevard | fuck the rh. |
| 16:31.20 | anonymouz666 | coppice: I understand. I can help you to test collect call handling, IF you want. Just drop me a line. |
| 16:31.22 | KranZ | gentoo is for the brave |
| 16:31.23 | scorn | I remember spending 3+ hours just chasing down dependencies and sub-dependencies if you want to call them that. for single package. |
| 16:31.48 | dlynes_laptop | I've used Ubuntu and some other debian based distro (one that's only available as a live CD) |
| 16:31.53 | scorn | hairs stand up on the back of my neck remembering it. |
| 16:31.58 | dlynes_laptop | I like using them for rescue CD's |
| 16:32.33 | scorn | Linux RIP |
| 16:32.41 | h3x | <PROTECTED> |
| 16:32.41 | scorn | another God send. |
| 16:32.43 | *** join/#asterisk bkidney (n=bkidney@nat01-128.stratos.ca) |
| 16:33.04 | sevard | slackware man :) |
| 16:33.22 | Qwell[] | Why would Dial play a ringtone? |
| 16:34.01 | MrChimpy | because you're calling a remote handset and it plays a dial tone to the caller? |
| 16:34.08 | MrChimpy | sorry, ringing tone |
| 16:34.10 | Qwell[] | Dial doesn't play the tone |
| 16:34.28 | nortex | Just to be sure before ordering it, does a Linksys SPA-1001 connect to asterisk? |
| 16:34.29 | Qwell[] | the remote end does |
| 16:34.33 | Qwell[] | nortex: yes |
| 16:34.43 | bkidney | Has anyone seen this problem. When I dial from my SIP phone to an external line, I still hear the sip phone ringing in my ear after the other extension has picked up (we can talk and he does not here the ringing)? |
| 16:34.46 | Qwell[] | nortex: there are better though, for about the same price |
| 16:34.51 | mog | Qwell[], ! |
| 16:34.54 | Qwell[] | mog: ! |
| 16:34.55 | MrChimpy | qwell: um.... |
| 16:35.01 | generalhan | Qwell[]: how do you like the cisco 7960 on sccp ? |
| 16:35.14 | Qwell[] | generalhan: sccp sucks |
| 16:35.14 | mog | Qwell[], is skinnny crazy |
| 16:35.17 | Qwell[] | generalhan: skinny however... |
| 16:35.28 | Qwell[] | generalhan: I'm biased though |
| 16:35.32 | MrChimpy | qwell: how on earth is music on hold achieved then? |
| 16:35.56 | nortex | Qwell[], Which would you recommend, I have used IAXy and SPA-3000 |
| 16:36.04 | generalhan | Qwell[]: whether youre biased or not ... this latest SIP firmware is PISSING ME OFF, and i have an extra that i want to test out other options on |
| 16:36.05 | wunderkin | it is not done on earth, it is done on mars |
| 16:36.05 | [TK]D-Fender | MrChimpy : Ritual sacrifice of a goat is customary ;) |
| 16:36.24 | generalhan | [TK]D-Fender: hahaha typically ! |
| 16:36.34 | MrChimpy | tkd: i ran out of goats some time ago |
| 16:36.44 | generalhan | MrChimpy: then youre SOL i think ! |
| 16:36.47 | [TK]D-Fender | dlynes_laptop : Probably Mepis |
| 16:36.52 | generalhan | no MoH for you |
| 16:37.11 | Qwell[] | generalhan: chan_skinny in trunk :) |
| 16:37.14 | [TK]D-Fender | MrChimpy : Here, go fill up :) www.goatse.cx |
| 16:37.24 | Qwell[] | nortex: I believe it's the SPA 2002 |
| 16:37.29 | generalhan | Qwell[]: what about the firmware ? |
| 16:37.33 | Qwell[] | [TK]D-Fender: which of the 2xxxx SPAs are better? |
| 16:37.36 | [TK]D-Fender | omg, the page is gone! |
| 16:37.37 | Qwell[] | generalhan: what about it? |
| 16:37.41 | mog | yeah |
| 16:37.44 | mog | its been years |
| 16:37.52 | generalhan | i was looking on cisco's site for it and i couldnt find anything other than SIP that was not CCM |
| 16:37.53 | mog | they got banned |
| 16:37.55 | mog | by cx domain |
| 16:38.01 | [TK]D-Fender | nortex : Forget the SPA-1001 and get the 2002 instead |
| 16:38.21 | bkidney | Qwell[] and generalhan: I have a Cisco 7960 problem. When I dial from it, I still hear ringing in my ear after the other party picks up. How you experienced this? |
| 16:38.22 | [TK]D-Fender | mog : behold the death of an era |
| 16:38.29 | mog | yeah |
| 16:38.33 | Qwell[] | bkidney: Stop using the r option to Dial |
| 16:38.34 | mog | there are loads of mirrors |
| 16:38.39 | mog | long page on it on wikipedia |
| 16:39.01 | tzanger | there's always tubgirl |
| 16:39.05 | [TK]D-Fender | nortex : And Sipura/Linksys' full lineup works pretty well with * |
| 16:39.10 | mog | she is gone too from what i hear |
| 16:39.17 | [TK]D-Fender | tzanger : o>O ? |
| 16:39.18 | mog | but i dont generally go looking for it |
| 16:39.19 | Qwell[] | mog: You could check... |
| 16:39.23 | mog | heh |
| 16:39.24 | mog | no |
| 16:39.24 | scorn | dlynes_laptop: now I remember, zttest from what I can tell only reports back the accuracy of the interface(?), just reports back 99.98..... over and over, then gives an average. |
| 16:39.44 | bkidney | Qwell[]: I am not using it. The exten string I am using is: exten => _9XXX, 1, Dail(Zap/4/${EXTEN:1}) |
| 16:39.55 | Qwell[] | mog: easy solution. send it to somebody within earshot, and wait for the "OH...MY...GOD!" |
| 16:40.00 | mog | lol |
| 16:40.03 | mog | so evil |
| 16:40.10 | [TK]D-Fender | bkidney : You could also try typing DIAL properly... |
| 16:40.19 | nortex | [TK]D-Fender, How about the 2100 or 2002 any thoughts? |
| 16:40.46 | [TK]D-Fender | nortex : AVOID. they are routers and SIP is only active on the WAN interface. |
| 16:40.53 | bkidney | [TK]D-Fender: Typo. I'm not on the * box. |
| 16:41.27 | [TK]D-Fender | nortex : Only for use with shcmucks with home PC's and a direct internet connection. Gives them a router and direct FXS. Not the style you should aim for. |
| 16:45.59 | Alric | Whats wrong with just using them as dual port ATAs? |
| 16:46.14 | Alric | and just shrugging at the LAN port? |
| 16:49.06 | watchy | today has been a nice quite day |
| 16:49.12 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 16:49.17 | watchy | i am overjoyed in the wonderful peace i am having |
| 16:50.22 | nortex | [TK]D-Fender, Thanks for the help :) |
| 16:50.31 | watchy | i may buy tk a steak if life keeps up this good |
| 16:50.37 | scorn | I'll be overjoyed when fall gets here. |
| 16:50.58 | watchy | scorn r u the scorn from lwz |
| 16:51.06 | TripleFFFF | quickie.. can one #define.. proper,fixed etcetc ? |
| 16:51.09 | TripleFFFF | i mean i assumed proper was landline, and fixed was global for htat area |
| 16:51.12 | scorn | watchy, n |
| 16:51.13 | [TK]D-Fender | Alric : Double NAT possible issues (they won't forge since they assume they are public), etc. Just not worth the trouble, plus that fact they cost more. |
| 16:51.28 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 16:51.46 | [TK]D-Fender | watchy : Blue & seared filet mignon! |
| 16:51.52 | Alric | I thought the 2100 was just like $75 or so? Doesn't seem so bad for an ATA that does/will do T.38, right? |
| 16:52.05 | TripleFFFF | whats the 2100 make ? |
| 16:52.08 | scorn | watchy, why, do I look like him |
| 16:52.11 | TripleFFFF | linksys ? |
| 16:52.17 | [TK]D-Fender | Alric : If t.38 is a difference and matter then you do what you must. |
| 16:52.17 | watchy | scorn: your nick does |
| 16:52.33 | watchy | plus your kinda pudgy like him to |
| 16:52.52 | scorn | y |
| 16:53.01 | scorn | getting pudgy-er |
| 16:53.09 | watchy | i guess cuz u drink to many sodas |
| 16:53.14 | watchy | drink water instead |
| 16:53.27 | scorn | n, I retain water. |
| 16:53.33 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.180) |
| 16:53.43 | watchy | whats t38? |
| 16:53.49 | dlynes_laptop | TripleFFFF: pap2-na, sipura 2002 I think(?) |
| 16:53.49 | mog | faxing over ip |
| 16:53.55 | TripleFFFF | t38 is 1 over t37.. |
| 16:54.01 | dlynes_laptop | TripleFFFF: maybe also grandstream 486 |
| 16:54.01 | TripleFFFF | the bit that makes faxing work |
| 16:54.03 | TripleFFFF | j/k |
| 16:54.11 | TripleFFFF | but wahts 2100 ? |
| 16:54.12 | watchy | mog: would faxing include normal data? |
| 16:54.16 | noname32 | we cogent is eating it at the moment |
| 16:54.21 | mog | not modem data |
| 16:54.24 | *** join/#asterisk jbsolutios (n=jbenson@62.168.75.180) |
| 16:54.25 | scorn | dlynes_laptop: did you see my comment above? |
| 16:54.26 | mog | thats a different spec |
| 16:54.27 | dlynes_laptop | TripleFFFF: 2100 is 2002 with a router built in i think |
| 16:54.27 | brad_mssw | TripleFFFF: sipura 2100 does as long as you have firmware 3.2 or higher |
| 16:54.31 | TripleFFFF | oh |
| 16:54.31 | mog | a. something |
| 16:54.31 | watchy | mog: oh, you guys need to make modem data work |
| 16:54.35 | brad_mssw | TripleFFFF: the 2002 does NOT |
| 16:54.36 | dlynes_laptop | scorn: about being pudgier? |
| 16:54.39 | watchy | i want to update my freakin tivo over asterisk |
| 16:54.40 | jbsolutios | does anyone here use XTunnels please? |
| 16:54.41 | mog | well we do have it working over our hw |
| 16:54.44 | mog | just not over ip |
| 16:54.47 | scorn | dlynes_laptop: heh, no, zttest. |
| 16:54.47 | dlynes_laptop | scorn: or about retaining water? |
| 16:54.49 | mog | you can try watchy |
| 16:54.53 | mog | if you do ulaw |
| 16:54.55 | mog | it might work |
| 16:54.56 | dlynes_laptop | scorn: oh...no |
| 16:55.00 | mog | not very likely though |
| 16:55.10 | watchy | oh i did, it dials out but it just doesnt connect, i dont think it can negotiate on what to do with the peer |
| 16:55.19 | watchy | so the tivo just says cant connect |
| 16:55.19 | mog | eh |
| 16:55.26 | mog | get a mythtv box |
| 16:55.27 | scorn | dlynes_laptop: now I remember, zttest from what I can tell only reports back the accuracy of the interface(?), just reports back 99.98..... over and over, then gives an average. |
| 16:55.29 | mog | youll be happier ^_^ |
| 16:55.36 | [TK]D-Fender | dlynes_laptop : When people use words like that I tend to tack on "Like what... Lake Superior!?" :D |
| 16:55.36 | watchy | mog: not if i got directv |
| 16:55.43 | dlynes_laptop | scorn: ok, that's fine, then |
| 16:55.44 | watchy | and hd dtv at that |
| 16:55.46 | TripleFFFF | thanks |
| 16:55.48 | mog | well there is your first problem |
| 16:55.49 | mog | ^_^ |
| 16:55.56 | dlynes_laptop | scorn: if you get low scores there, it's usually because you have interrupt problem |
| 16:56.08 | TripleFFFF | so funny that voipsuply shipped me 300$ atas and charged 500$ shipping |
| 16:56.18 | Qwell[] | watchy: there are HD tuners... just get another box |
| 16:56.18 | watchy | mog: i live in a town of 11,000 our cable co aint got hd and never will have it |
| 16:56.21 | scorn | dlynes_laptop: anything in the arsenal that will just watch the incoming channels and tell me if data is coming in? |
| 16:56.29 | TripleFFFF | witouth asking they made it like super urgent like next hour deliv |
| 16:56.34 | dlynes_laptop | scorn: not that i can think of, no |
| 16:56.35 | TripleFFFF | poor poor them |
| 16:56.35 | mog | eh |
| 16:56.41 | mog | at least yet |
| 16:56.45 | mog | to expensive |
| 16:56.45 | dlynes_laptop | scorn: if zttest and patlooptest both pass |
| 16:56.50 | watchy | mog: we have a small town but like 500 .gov bomb contractors |
| 16:56.50 | TripleFFFF | watchy get sat |
| 16:56.53 | TripleFFFF | FTA's |
| 16:56.54 | TripleFFFF | ;0 |
| 16:56.56 | mog | ahh |
| 16:56.59 | TripleFFFF | they down ATM but will go UP soon |
| 16:57.01 | watchy | if something blows up about 15miles away |
| 16:57.01 | dlynes_laptop | scorn: then your next step is to see if there's any signalling happening |
| 16:57.05 | watchy | yuo wont see me anymore |
| 16:57.17 | TripleFFFF | [12:57] watchy: mog: we have a small town but like 500 .gov bomb contractors |
| 16:57.19 | dlynes_laptop | scorn: normally you do that by making a call into your pri |
| 16:57.22 | TripleFFFF | lol .. you live in BUsh state |
| 16:57.25 | dlynes_laptop | scorn: but you're not running a pri |
| 16:57.33 | watchy | TripleFFFF: camden arkansas |
| 16:57.45 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.137) |
| 16:57.48 | TripleFFFF | oh its still close to bushville |
| 16:57.50 | watchy | lemme get you a picture from google maps to show you all the crazy ass bomb bunkers here |
| 16:57.53 | watchy | its nuts |
| 16:57.53 | scorn | dlynes_laptop: zttool is acknowledging Layer 1 up, and I know that the card that is the source for the data is sending out data, |
| 16:57.53 | dlynes_laptop | watchy: damn dem dar arkansawrians |
| 16:58.34 | scorn | dlynes_laptop: so after a loopback test, then I guess I need to start looking at writing some code. I'm guessing to do what I'm wanting to do, patlooptest is a good start? |
| 16:58.46 | dlynes_laptop | scorn: no |
| 16:58.51 | scorn | oh |
| 16:58.54 | scorn | heh |
| 16:58.55 | dlynes_laptop | scorn: patlooptest just tests a loopback adapter |
| 16:59.12 | dlynes_laptop | scorn: and for that to work, you have to set your pri adapter up in clear mode |
| 16:59.24 | scorn | dlynes_laptop: right, but it will show me how to receive data on a channel, no? |
| 16:59.27 | dlynes_laptop | scorn: that's not the normal mode of operation |
| 16:59.29 | scorn | oh |
| 16:59.37 | scorn | hm |
| 16:59.46 | scorn | anything else you can think of? |
| 17:00.30 | scorn | anything that needs to happen to the data after I pull it out of /dev/zap/XX? |
| 17:01.05 | TripleFFFF | anyone have t38 pass trough to tdm /pri's ? |
| 17:01.23 | TripleFFFF | cant find my darn x100 |
| 17:01.53 | mog | you can TripleFFFF with spandsp |
| 17:01.57 | mog | i would think |
| 17:02.00 | TripleFFFF | nono |
| 17:02.06 | scorn | TripleFFFF: I live the THE Bush state, I even have the T-shirt to prove it. |
| 17:02.16 | TripleFFFF | i mean i need someone that has a asterisk t38 PTrough enabled. to a pstn |
| 17:02.17 | scorn | and bumper sticker |
| 17:02.22 | TripleFFFF | so i can send him a test fax.. |
| 17:02.45 | TripleFFFF | so me FAX --> SIP --> YOU SIP--> YOU PSTN --> EFAX --> ME EMAIL == YAY! |
| 17:02.51 | watchy | http://maps.google.com/maps?f=q&hl=en&q=spectra+camden,+ar&ie=UTF8&ll=33.59421,-92.616377&spn=0.016301,0.040255&t=k&om=1 |
| 17:02.55 | watchy | chedk that out triple |
| 17:03.05 | watchy | look at all those fucking underground bunkers |
| 17:03.12 | watchy | thats 15miles away from me |
| 17:03.18 | watchy | they are all full of explosives and bombs |
| 17:03.25 | TripleFFFF | that is not a good place to be |
| 17:03.41 | TripleFFFF | <PROTECTED> |
| 17:03.41 | TripleFFFF | Spectra Technologies Llc |
| 17:03.51 | TripleFFFF | with a name like that oyu know your a shadow enterprise |
| 17:04.03 | watchy | haha |
| 17:04.04 | TripleFFFF | spectra lol.. fuck this world is going to an end soon.. |
| 17:04.38 | TripleFFFF | you guys should use your .whatever # ammendment and get things straghten up |
| 17:04.48 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.215) |
| 17:05.01 | kfudge | Anyone having an issue with putting a call on hold? Apparently on one of my servers, if a call is on hold longer than two minutes it occasionally drops the call. |
| 17:05.20 | KranZ | kfudge: look at your sip.conf |
| 17:05.30 | watchy | TripleFFFF: yea we have a whole area full of bunkers and explosives plants |
| 17:05.37 | KranZ | there's a timer which drops a call after the rtp stream stops for so long |
| 17:05.55 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 17:05.58 | [TK]D-Fender | kfudge : that'd be rtp.conf |
| 17:06.10 | KranZ | rtpholdtimeout=300 |
| 17:06.17 | KranZ | its in sip.conf |
| 17:06.24 | [TK]D-Fender | KranZ : interesting |
| 17:06.28 | TripleFFFF | or rtptimeout |
| 17:06.53 | KranZ | i was messing with it yesterday |
| 17:07.28 | kfudge | change it to that? |
| 17:07.41 | KranZ | try setting it low to see it in action |
| 17:07.44 | KranZ | then set it to what you want |
| 17:08.35 | KranZ | also, you could disable VAD so rtp traffic never stops |
| 17:08.35 | TripleFFFF | not really |
| 17:08.35 | KranZ | ...but, you eventually want vad |
| 17:08.35 | TripleFFFF | i got a 13 hour call stuck in asterisk last night |
| 17:08.36 | TripleFFFF | asterisk DID NOT hangup |
| 17:08.36 | TripleFFFF | so i added rtptimout |
| 17:08.59 | KranZ | and |
| 17:09.06 | kfudge | so is the going to help? lol |
| 17:10.11 | watchy | hmm |
| 17:10.27 | watchy | callerid aint passing to my cisco 7960 when someone calls in |
| 17:10.27 | kfudge | Because I really don't want the call disconnected at all |
| 17:10.35 | watchy | what do i have to do to enable caller id? |
| 17:10.48 | Qwell[] | watchy: don't use 8.x firmware |
| 17:11.08 | watchy | i'm not i think i;m on 7.2 |
| 17:11.14 | Qwell[] | oh |
| 17:11.16 | watchy | is been forever since i set this cisco up though |
| 17:11.30 | generalhan | i like the 8.2.00 |
| 17:11.35 | generalhan | but the 8-3-00 is GARBAGE |
| 17:12.04 | kfudge | Sometimes the caller needs to be kept on hold longer than 5 min |
| 17:12.14 | KranZ | kfudge: then set that for 5x60seconds |
| 17:12.14 | *** join/#asterisk edorsey (i=edorsey@216-63-192-130.ded.swbell.net) |
| 17:12.26 | KranZ | kfudge: or higher |
| 17:12.31 | kfudge | that is what I was thinking |
| 17:12.40 | kfudge | Thanks alot for that tip though |
| 17:12.46 | KranZ | but set it low to make sure it will trigger |
| 17:12.56 | KranZ | then set it up |
| 17:13.05 | KranZ | if it doesnt trigger, then it probably wont help |
| 17:13.41 | kfudge | but if it does then i can set it for a higher amount |
| 17:13.56 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 17:17.05 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 17:19.12 | *** join/#asterisk terje (n=joem@67.41.208.129) |
| 17:20.48 | terje | hi, I'm creating a call file in /var/spool/asterisk/outgoing with the following listed for the Application: |
| 17:20.50 | terje | application: Flite("Wake me up") |
| 17:21.01 | terje | the call is made but Flite doesn't execute. |
| 17:21.05 | [TK]D-Fender | generalhan : Cisco is only grudgingly acknoledging SIP it seems... |
| 17:22.49 | watchy | sata drives in linux are known as /dev/sdX? |
| 17:22.56 | bkidney | My TDM400 FXO port does not seem to be detecting when a called party picks up. It is configured for loopstart and connected to line using this signalling. Anyone know what the problem might be? |
| 17:22.57 | Qwell[] | watchy: usually |
| 17:23.10 | watchy | qwell: never loaded a linux box with sata heh, thanks |
| 17:23.20 | [TK]D-Fender | watchy : Typically because SATA was fist implemented using the SCSI framework in the kernel IIRC |
| 17:23.28 | [TK]D-Fender | first* |
| 17:23.28 | watchy | yea thats what i noticed |
| 17:23.31 | generalhan | [TK]D-Fender: well i loved it until this last release now i hate it .. even though i just downgraded to the one before this .. |
| 17:24.01 | generalhan | i would use skinny or sccp but im not too familiar witht the set up and i have a TON of things on my "to-do-list" |
| 17:24.30 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:25.23 | *** join/#asterisk s0lid (n=jlq@gr-153-4.eglobalreach.net) |
| 17:25.49 | *** join/#asterisk xming (n=xming@gentoo/user/xming) |
| 17:27.41 | ManxPower | bkidney, FXOs cannot detect when the called party picks up. That's life in the analog world. |
| 17:29.16 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 17:29.48 | bkidney | ManxPower: When I call out through the FXO and the other party picks up, I can hear them talk, but Asterisk does not recognize the connection and continues ringing. Is there a way to fix this? |
| 17:30.04 | generalhan | lol |
| 17:30.51 | rowter | Anyone has by chance NVLineDetect.c here so I could test it out? |
| 17:32.38 | *** join/#asterisk meshuga (i=meshuga@c-71-231-141-145.hsd1.or.comcast.net) |
| 17:33.30 | s0lid | !ping |
| 17:33.39 | *** join/#asterisk Assid (i=assid@203.115.64.10) |
| 17:36.03 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 17:39.09 | *** join/#asterisk Taker (n=GLA@u5-78.dsl.vianetworks.de) |
| 17:39.41 | Taker | hello ;-) |
| 17:40.33 | *** join/#asterisk enjay- (n=enjay@209.181.130.105) |
| 17:45.16 | *** join/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 17:45.53 | Optic | hey, I've got asterisk on a T1 with a digium card... is there anyway to block caller id entirely in the dialplan? |
| 17:46.08 | Optic | we have one phone user whom I don't want sending caller id |
| 17:46.16 | mog | yes |
| 17:46.56 | Optic | how does one go about it? |
| 17:47.52 | watchy | i didnt realise you could send callerid |
| 17:47.57 | Juggie | set callerid blank in sip.conf |
| 17:48.05 | Juggie | or in realtime whatever you use |
| 17:48.47 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 17:48.59 | Optic | that's not the dialplan? |
| 17:49.09 | Juggie | you could do it in dialplan if you wanted |
| 17:49.15 | Juggie | but you can just assign it to the phone |
| 17:49.18 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 17:49.56 | Juggie | it would certainly be easier to just set it in sip options |
| 17:50.50 | Wazb^ | is there way to get ringback through some configuration |
| 17:50.51 | Optic | I don't want to disable it for all users, just a particular extension |
| 17:51.06 | Juggie | are you listning to the words comming out of my mouth |
| 17:51.09 | Juggie | or in this case my hands |
| 17:51.13 | Juggie | SET IT IN SIP OPTIONS |
| 17:51.19 | terje | in the dialplan for an extension I have 'exten => 555,1,Answer' (as an example). Is the 'Answer' considered an 'application' by asterisk and if so, I can I specify it in a call file as an 'application: Answer' ? |
| 17:51.19 | Juggie | i've said that THREE times |
| 17:51.49 | terje | MusicOnHold seems to work, where Flite doesn't seem to work in a call file. |
| 17:52.10 | edorsey | techincally he probbably isnt listening to them...no |
| 17:52.37 | Juggie | sip.conf [whateverphone] callerid="" |
| 17:53.19 | Optic | juggie: I see, sorry :) |
| 17:53.30 | Juggie | or, you could do |
| 17:53.43 | Juggie | callerid="Unknown Name <Unknown Number>" or whatever |
| 17:53.56 | Juggie | i dont think * has a privacy flag |
| 17:53.58 | dlynes_laptop | Juggie: You don't even need the quotes, for that matter |
| 17:54.00 | Juggie | i googled and cant find it |
| 17:54.04 | ManxPower | DON'T USER QUOTES |
| 17:54.21 | Juggie | really? i've allways used quotes i think without problem... |
| 17:54.26 | Optic | hmm |
| 17:54.37 | dlynes_laptop | Juggie: putting quotes in there, adds an extra set of quotes that aren't necessary |
| 17:54.42 | Juggie | ahh k. |
| 17:54.43 | salviadud | what's wrong with quoting? |
| 17:54.46 | Juggie | i'll keep that in mind. |
| 17:54.49 | kfudge | damn rtpholdtimeout didn't work |
| 17:55.03 | ManxPower | callerid="unknown name <123455>" would se the CALLERIDNAME to be unknown name <123455> |
| 17:55.19 | kfudge | Any other suggestions? |
| 17:55.42 | dlynes_laptop | kfudge: what exactly are you trying to do? |
| 17:56.03 | Juggie | ManxPower, k, thanks. |
| 17:56.11 | dlynes_laptop | kfudge: put somebody on hold and have it remind you every so often that they're on hold? |
| 17:56.54 | kfudge | No I am getting random disconnects after a few minutes when a caller is on hold |
| 17:57.04 | dlynes_laptop | damn |
| 17:57.11 | dlynes_laptop | remind me never to call your business |
| 17:57.16 | kfudge | lol |
| 17:57.18 | dlynes_laptop | I don't want to be put on hold for three minutes |
| 17:57.44 | Optic | hmm, what about SetCallerPres? |
| 17:57.49 | *** join/#asterisk CrazyYoss (n=luther@adsl-69-236-44-222.dsl.pltn13.pacbell.net) |
| 17:57.55 | dlynes_laptop | Optic: that's for privacy features |
| 17:57.58 | dlynes_laptop | Optic: not caller id |
| 17:58.17 | Optic | ok |
| 17:58.32 | dlynes_laptop | Optic: using certain options in there is the same as *67 |
| 17:59.39 | CrazyYoss | whats a "best guess Mr. Sulu" on the cost per minute for a 1800 number in the 50k minutes range? Just a general range..5-10? 10-15? |
| 17:59.40 | [TK]D-Fender | Juggie : "Man... ain't NOONE understanding the words commin outta yo' mouth!" - Chris Rock |
| 18:00.09 | [TK]D-Fender | CrazyYoss : Sub .02$ |
| 18:00.34 | Optic | dylnes: that's basically what I want... the called party to basically see that caller-id was blocked |
| 18:00.36 | CrazyYoss | [TK]D-Fender: Thank you |
| 18:00.46 | *** join/#asterisk iq (n=IQ@unaffiliated/iq) |
| 18:00.50 | dlynes_laptop | Optic: yeah...you want SetCallerPres() then |
| 18:00.57 | Optic | I want basically *67 :) |
| 18:01.07 | Hughes | Well, I'm baffled. |
| 18:02.07 | Hughes | Have a grandstream 100 and sipura SPA-2000. They can both communicate with the asterisk voicemail, make calls out my FXO, but if they call each other there's no audio. |
| 18:02.17 | Hughes | No NAT involved at all. |
| 18:03.08 | [TK]D-Fender | Hughes : Pastebin up your sip.conf |
| 18:03.10 | [TK]D-Fender | ~pb |
| 18:03.12 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 18:03.13 | dlynes_laptop | Hughes: are they negotiating codes properly? |
| 18:03.20 | dlynes_laptop | s/codes/codecs/ |
| 18:03.48 | Optic | thanks, i've some good ideas now. |
| 18:04.02 | Optic | sorry to raise your blood pressure, Juggie :) |
| 18:04.23 | Hughes | jbot I've been looking for some good debug output. Can you point me in the right direction? |
| 18:04.46 | Hughes | I'm looking at asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvgr sith sip debug on. |
| 18:04.50 | Hughes | with |
| 18:05.38 | *** join/#asterisk Ding_Aling (n=none@nv-71-49-133-50.dhcp.embarqhsd.net) |
| 18:05.54 | salviadud | that's a lot of verbose there |
| 18:06.11 | Hughes | It's not generating a lot of traffic. Want to see everything I can right now. |
| 18:07.34 | dlynes_laptop | Hughes: jbot is a bot |
| 18:08.01 | *** part/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 18:08.02 | dlynes_laptop | ~lart hughest |
| 18:08.05 | dlynes_laptop | ~lart hughes |
| 18:08.20 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 18:08.23 | Ding_Aling | I need a new VOIP provider. My current one Junction Networks is down again. Any recomendations? We don't do mutch long distance so any local unlimiteds would be great. |
| 18:08.45 | Hughes | Yeah, I caught on. |
| 18:09.35 | Hughes | sanitizing sip.comf for pastebin |
| 18:09.42 | dlynes_laptop | Hughes: anyways, just do set verbose 100 and then sip debug |
| 18:09.50 | dlynes_laptop | Hughes: that'll give you plenty of debug info |
| 18:10.07 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 18:10.34 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402BD8F.pool.t-online.hu) |
| 18:10.45 | visba | Ding_Aling: give Teliax a try |
| 18:10.49 | *** join/#asterisk |dennis| (n=dummie@200.32.215.82) |
| 18:10.56 | Ding_Aling | I will take a look |
| 18:11.02 | *** join/#asterisk pigpen2 (n=mark@fwipsec.tmi-sa.org) |
| 18:11.05 | *** join/#asterisk [bsd] (n=bsd@59.144.251.151) |
| 18:11.46 | nortex | What is the best wi-fi phone out there right now? |
| 18:11.59 | Juggie | theres no answer to that question |
| 18:12.04 | mog | none |
| 18:12.08 | mog | softphone maybe |
| 18:12.12 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 18:12.23 | Juggie | there are some that work |
| 18:12.26 | Juggie | but none that are fantastic |
| 18:12.35 | fourcheeze | damn my asterisk just ditched a load of my sip clients |
| 18:12.45 | fourcheeze | anyone got any faint idea why it does this |
| 18:13.06 | Juggie | are you using * realtime? |
| 18:13.28 | nortex | I was afraid of that. The UTstarcomm I have messed with is not the best for sure. |
| 18:13.45 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:13.48 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 18:13.52 | Ding_Aling | What voip provider has the best network peering. I am looking for a company with more than one boix sitting in a colo |
| 18:13.53 | terje | Hughes, what happens if you dial by IP address? |
| 18:13.57 | watchy | anyone got polycom sip 1.6.7 layin around? |
| 18:14.14 | fourcheeze | Juggie: yes |
| 18:14.28 | Hughes | terje - haven't tried it. |
| 18:14.30 | fourcheeze | Juggie: do you know what it is? |
| 18:14.33 | Juggie | fourcheeze, they arnt really ditched... |
| 18:14.47 | fourcheeze | well they can't make/receive calls |
| 18:14.51 | Juggie | they haev just expired from the realtime cache. |
| 18:14.53 | Juggie | hmmmmm |
| 18:14.57 | visba | Ding_Aling: where are you located? Teliax is just about everywhere |
| 18:14.59 | Juggie | try a |
| 18:15.02 | Juggie | 'sip reload' |
| 18:15.06 | Juggie | and then see if they can |
| 18:15.25 | Ding_Aling | I am in vegas |
| 18:15.43 | visba | try tracing to voip-ca1.teliax.com, that's their Cali gateway |
| 18:15.49 | Ding_Aling | I am just looking for a bigger dog than junction networks |
| 18:16.12 | [TK]D-Fender | nortex : They ALL suck.... some a little less than others. |
| 18:16.13 | Ding_Aling | Multiple gateways already sounds nice |
| 18:16.26 | fourcheeze | Juggie: it seems to be particular sites become disconnected all together, like * can't find it's way back to them |
| 18:16.39 | Juggie | sites? |
| 18:16.43 | visba | Ding_Aling: i'm in Denver, using the voip-co1 gateway |
| 18:16.47 | Juggie | are they all connected via sip? |
| 18:16.47 | fourcheeze | I have clients on about 15 sites |
| 18:16.50 | fourcheeze | yes |
| 18:16.54 | Juggie | only one * server? |
| 18:16.56 | fourcheeze | yes |
| 18:17.36 | Wazb^ | i am not getting ringback when i forward call to carrier ? |
| 18:17.48 | Juggie | how do you have realtime configured |
| 18:17.51 | fourcheeze | Juggie: this morning it happened twice and then didn't happen at all again until about an hour ago |
| 18:17.52 | Juggie | are you using cacheing? |
| 18:18.03 | Ding_Aling | visba: thanks for the recomendation |
| 18:18.03 | fourcheeze | Juggie: which parameter is that? |
| 18:18.05 | Juggie | have you changed something lately? did it ever happen before? |
| 18:18.08 | Juggie | rtcachefriends |
| 18:18.22 | Juggie | can you contact the client site even though sip doesnt work. |
| 18:18.25 | fourcheeze | Juggie: all that happened was that I had to fail over servers |
| 18:18.42 | fourcheeze | Juggie: everything else internet-wise is normal |
| 18:18.44 | Juggie | so then the server has changed? |
| 18:18.49 | fourcheeze | the server has changed |
| 18:18.58 | fourcheeze | the hardware has |
| 18:18.58 | Juggie | are you sure its an exact copy of the config? |
| 18:19.01 | Juggie | same version of asterisk? |
| 18:19.11 | fourcheeze | I copied over /etc/asterisk to the new server |
| 18:19.18 | *** join/#asterisk speedracer (n=speedrac@gateway.digium.com) |
| 18:19.19 | fourcheeze | and restored the database from a backup |
| 18:19.41 | Juggie | are the clients nat? |
| 18:19.46 | fourcheeze | yes |
| 18:19.50 | fourcheeze | I think this is the problem |
| 18:19.56 | Hughes | dlynes: http://pastebin.com/762828 |
| 18:19.57 | Juggie | so you have nat=yes & qualify=yes |
| 18:19.58 | Juggie | set? |
| 18:20.02 | fourcheeze | from time to time for particular sites the ability to traverse nat is lost |
| 18:20.02 | fourcheeze | yes |
| 18:20.08 | fourcheeze | that's set in the realtime database |
| 18:20.23 | Juggie | make sure that shows up as being set when you do sip show peers |
| 18:20.30 | Juggie | how many clients do you have. |
| 18:20.46 | fourcheeze | 60 |
| 18:20.53 | fourcheeze | rtcachefriends=yes |
| 18:21.02 | fourcheeze | I'm tearing my hair out with this now |
| 18:21.04 | Juggie | i assume they have voicemail? |
| 18:21.06 | fourcheeze | yes |
| 18:21.13 | fourcheeze | everything works perfectly 99% of the time |
| 18:21.31 | Juggie | you can try changing the maxexpiry/default expiry |
| 18:21.33 | Juggie | to something low |
| 18:21.34 | fourcheeze | then seemingly randomly 50% of them vanish from sip show users |
| 18:21.43 | fourcheeze | I have lowered it from 300sec to 60 |
| 18:21.47 | Juggie | to force the clients to register alot |
| 18:21.49 | fourcheeze | problem still happens |
| 18:22.42 | Juggie | do the clients go unreachable in the log? |
| 18:22.56 | Juggie | eg does it say like <peer> is now unreachable? |
| 18:23.06 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 18:23.32 | fourcheeze | Juggie: yes I see: 060803-183923 NOTICE[8461] chan_sip.c: Peer '20060067' is now UNREACHABLE! Last qualify: 75 |
| 18:23.46 | Juggie | yep |
| 18:23.48 | Juggie | something is amiss |
| 18:23.52 | Juggie | causing your nat to break |
| 18:23.54 | fourcheeze | yeah, tell me about it :-) |
| 18:24.32 | *** part/#asterisk speedracer (n=speedrac@gateway.digium.com) |
| 18:24.34 | Juggie | perhaps its another problem that concidentally happened at the same time? |
| 18:24.55 | Juggie | because if * is loosing the client.. not sure what to tell ya |
| 18:24.57 | fourcheeze | Juggie: I'm very pleased to have someone to think through this with BTW :-) |
| 18:25.08 | Juggie | i accept money. |
| 18:25.14 | fourcheeze | :-) |
| 18:25.15 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 18:25.28 | fourcheeze | that would be taking advantage of someone in need ;-) |
| 18:25.48 | Juggie | i would blame your nat device |
| 18:25.56 | fourcheeze | which one? |
| 18:25.58 | Juggie | nothing in * can cause your clients to go unreachable |
| 18:26.16 | fourcheeze | ok, but I have this happening on many sites |
| 18:26.19 | *** join/#asterisk angom_w (n=angom@red-corp-200.79.158.218.telnor.net) |
| 18:26.19 | Juggie | with the exception of like the network config saying which ip range is local and which is remote. |
| 18:26.24 | *** join/#asterisk twisted[asteria] (n=twisted@pdpc/supporter/active/twisted) |
| 18:26.24 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 18:26.27 | Juggie | is * behind nat locally? |
| 18:26.32 | Juggie | eg behind a firewall? |
| 18:26.35 | fourcheeze | asterisk is out in the wild |
| 18:26.46 | fourcheeze | clients are behind nat |
| 18:26.46 | *** part/#asterisk [bsd] (n=bsd@59.144.251.151) |
| 18:26.56 | Juggie | no sort of firewall? |
| 18:27.04 | fourcheeze | nothing between asterisk and the internet |
| 18:27.19 | fourcheeze | it's straight on the end of a 2Mbit/sec pipe |
| 18:27.21 | Juggie | and i assume its running on the same ip. |
| 18:27.29 | fourcheeze | it doesn't change IP number |
| 18:27.44 | fourcheeze | there are no obvious things like collisions etc on switches |
| 18:27.52 | *** join/#asterisk SwK (n=Silik0nJ@65.169.134.2) |
| 18:28.02 | fourcheeze | current asterisk version is 1.2.7.1 |
| 18:28.04 | Juggie | then i am not sure... if your * has multiple networks |
| 18:28.12 | Juggie | make sure you set your networks in sip.conf |
| 18:28.16 | fourcheeze | * has 3 IP numbers |
| 18:28.30 | Juggie | hmmmmmmmmmmmmmm |
| 18:29.09 | Juggie | make sure you set localnet= |
| 18:29.16 | ManxPower | so much for my nap. |
| 18:29.32 | Juggie | maybe manx has an idea :) |
| 18:29.58 | Juggie | make sure your box is configured the same way, eg, the internet has the primary gateway and is the default route. |
| 18:30.08 | Juggie | * has a habit of sending packets down the wrong network |
| 18:30.22 | fourcheeze | Juggie: do I need to set localnet = if I'm not behind nat? |
| 18:30.34 | Juggie | it coudnt hurt |
| 18:30.42 | fourcheeze | hm |
| 18:30.43 | Juggie | it will make sure nat=yes clients dont go out those routes. |
| 18:30.50 | fourcheeze | ok |
| 18:30.57 | fourcheeze | * is on a public IP though |
| 18:31.03 | Juggie | yes |
| 18:31.05 | Juggie | but as i said |
| 18:31.13 | Juggie | * has a nasty habbit of using the wrong network card in some situations |
| 18:32.13 | fourcheeze | ok |
| 18:32.29 | Juggie | the cards are potentially in the box in a different order now which could be causing this unexpected behaviour |
| 18:32.34 | fourcheeze | well there is only one card and it has 3 IP numbers on the same network |
| 18:32.46 | Juggie | ohh ok, i misunderstood then |
| 18:32.49 | Juggie | the box has no local ips? |
| 18:33.06 | fourcheeze | none at all |
| 18:33.11 | Juggie | oo also |
| 18:33.18 | Juggie | the ip that the phones use to talk to * |
| 18:33.22 | Juggie | is that eth0 |
| 18:33.26 | fourcheeze | eth0:2 |
| 18:33.29 | Juggie | ahhh |
| 18:33.32 | Juggie | ding ding |
| 18:33.32 | fourcheeze | ohhh? |
| 18:33.35 | fourcheeze | is that bad? |
| 18:33.40 | Juggie | * is dumb sometimes. |
| 18:33.44 | Juggie | try making it eth0 |
| 18:33.48 | fourcheeze | how come it's not dumb all the time? |
| 18:33.54 | Juggie | i dont know. |
| 18:34.01 | fourcheeze | i.e. how come it's only dumb at random intervals? |
| 18:34.04 | Juggie | but on your old box was the main ip eth0 |
| 18:34.14 | fourcheeze | yes, as far as I can remember |
| 18:34.18 | Juggie | yeah... try that. |
| 18:34.21 | fourcheeze | ok |
| 18:34.22 | Juggie | just switch em. |
| 18:34.23 | fourcheeze | anything else? |
| 18:34.27 | Juggie | thats all i can think of. |
| 18:34.45 | Juggie | if it sends packets on th wrong ip address thast whats causing problem |
| 18:34.49 | Juggie | and it may very well be. |
| 18:34.51 | *** join/#asterisk tasat- (n=tasat-@c-67-180-181-221.hsd1.ca.comcast.net) |
| 18:35.03 | Juggie | so switch eth0:2<->eth0 |
| 18:35.06 | Juggie | and restart * totally. |
| 18:35.19 | *** join/#asterisk ComputerWarm (n=donc@209.29.158.52) |
| 18:35.37 | ComputerWarm | HEllo question how can i go about setting a Var of the phone number that the person called in on? |
| 18:35.43 | Juggie | if that solves the problem, please find some way to put in a bug report |
| 18:35.55 | Juggie | as that is not the proper behavior |
| 18:35.55 | *** join/#asterisk burnproof (n=burnproo@210.213.198.37) |
| 18:36.19 | [TK]D-Fender | ComputerWarm : "show application set" |
| 18:36.19 | fourcheeze | believe me, when I find out what it is I will file as many bug reports as I can :-) |
| 18:36.32 | Juggie | i suspect qualify=yes |
| 18:36.32 | fourcheeze | Juggie: ok, networking restarted, about to restart * |
| 18:36.38 | Juggie | is sending packets from the main ip. |
| 18:36.45 | Juggie | rather then eth0:2 |
| 18:36.50 | Juggie | so the nat isnt beign kept open properly |
| 18:37.10 | Juggie | you might have also tried setting externip= |
| 18:37.16 | Juggie | but lets try this for now |
| 18:37.53 | fourcheeze | ok, I've got lots of unreachable hosts now |
| 18:38.20 | Juggie | you shoudnt really have any until they start registering? |
| 18:38.26 | fourcheeze | hmm |
| 18:38.30 | Juggie | assuming you are using realtime dynamic peers |
| 18:38.32 | fourcheeze | yeah |
| 18:38.35 | fourcheeze | that's what I thought |
| 18:38.40 | *** join/#asterisk mfdutra (n=marlon@201.37.64.151) |
| 18:38.44 | fourcheeze | I can see them though |
| 18:38.52 | Juggie | they will only show up in the list once they register or someone tries to call them |
| 18:39.03 | fourcheeze | maybe asterisk didn't properly die |
| 18:39.12 | Juggie | must not have. |
| 18:39.25 | mfdutra | is it possible to proceed with the dialplan *while* Dial() is in place? |
| 18:39.38 | Juggie | yes |
| 18:39.42 | Juggie | use a macro inside dial. |
| 18:39.43 | mfdutra | I wanna do Dial(), Wait(1), SendDTMF(1234)... |
| 18:40.01 | fourcheeze | Juggie: ugh |
| 18:40.03 | Juggie | then do Dial(device/${EXTEN}ww1234) |
| 18:40.06 | fourcheeze | I'm getting a load of these now: |
| 18:40.08 | fourcheeze | 060803-193950 NOTICE[22101]: chan_sip.c:6297 check_auth: stale nonce received from 'Ringer <sip:20060027@voip2.callmaster.co.uk>' |
| 18:40.24 | *** part/#asterisk ComputerWarm (n=donc@209.29.158.52) |
| 18:40.26 | mfdutra | Juggie: it's an E1 device |
| 18:40.33 | mfdutra | can I use 'w' in a digital line? |
| 18:40.36 | Juggie | mfdutra, no. |
| 18:40.46 | Juggie | theres a parm on app_dial |
| 18:40.51 | Juggie | for sending dtmf's after a call is connected |
| 18:40.52 | Juggie | look it up |
| 18:40.56 | mfdutra | oh |
| 18:41.09 | Juggie | fourcheeze, probally related to the ip changing |
| 18:41.13 | fourcheeze | Juggie: I had this when I failed over |
| 18:41.15 | Juggie | they might have been seeing the eth0 |
| 18:41.25 | Juggie | it will probally take a few minutes for the reigstrations to clear |
| 18:41.40 | fourcheeze | when I failed over I had to get everyone to use a different hostname |
| 18:41.46 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:42.11 | mfdutra | D |
| 18:42.13 | mfdutra | I'll try it |
| 18:42.53 | Juggie | fourcheeze, as i said most likely related to the ip changing |
| 18:43.14 | [hC] | juggie doesnt know what he's talking about... hes from canada! (shhhh) |
| 18:43.16 | Juggie | once the phones do a re-register it should clear i would think. |
| 18:43.19 | [hC] | :) |
| 18:43.36 | [hC] | er i mean, he's from canada eh. |
| 18:43.36 | Juggie | [hC], at least i'm not from BC. |
| 18:43.55 | fourcheeze | Juggie: it didn't last time |
| 18:44.04 | Juggie | fourcheeze, have someone reboot their phone |
| 18:44.09 | Juggie | i dont see why. |
| 18:44.10 | fourcheeze | there is a bug about this somewhere |
| 18:44.11 | Juggie | it woudnt. |
| 18:44.13 | fourcheeze | I can reboot one here |
| 18:44.18 | Juggie | ok try that. |
| 18:44.32 | Juggie | * was most liekly confusing the phones/nat devices |
| 18:44.38 | Juggie | sendnig some packets from one ip |
| 18:44.40 | Juggie | and some from another. |
| 18:44.42 | mfdutra | it didn't work |
| 18:44.42 | CunningPike | Juggie: That's fightin' talk - where are you from, then? |
| 18:44.45 | mfdutra | I'll try a macro |
| 18:45.01 | Juggie | CunningPike, originally NFLD, but currentally ottawa, ontario |
| 18:45.22 | CunningPike | Juggie: Ah, the Center of the Universe(TM) |
| 18:45.27 | Juggie | no thats toronto. |
| 18:45.31 | CunningPike | :D |
| 18:45.49 | *** join/#asterisk mcolgin (n=g0dsmite@dsl254-018-172.sea1.dsl.speakeasy.net) |
| 18:45.51 | [hC] | no no, its vancouver. |
| 18:45.52 | [hC] | :) |
| 18:46.02 | Juggie | vancouver is where all the hookers go |
| 18:46.12 | [hC] | :| |
| 18:46.13 | jbalcomb | Is there a syslog server option for DHCP? I don't see one. |
| 18:46.25 | fourcheeze | Juggie: I've restart my phone but asterisk just keeps sending 401 unauthorised |
| 18:47.18 | Juggie | why |
| 18:47.21 | Juggie | how is it failing |
| 18:47.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:47.54 | fourcheeze | Juggie: Transmitting (NAT) to 82.153.23.79:37839: |
| 18:47.54 | fourcheeze | SIP/2.0 401 Unauthorized |
| 18:47.59 | fourcheeze | it just keeps sending that |
| 18:48.09 | Juggie | whos generating that message |
| 18:48.10 | watchy | hack the planet |
| 18:48.12 | fourcheeze | Juggie: * |
| 18:48.13 | [TK]D-Fender | fourcheeze : You may want to try using the right user/pass |
| 18:48.13 | Juggie | * or the remote peer |
| 18:48.18 | fourcheeze | [TK]D-Fender: haha |
| 18:48.31 | *** join/#asterisk tempest1 (n=asf@adsl-230-225-143.chs.bellsouth.net) |
| 18:48.34 | fourcheeze | [TK]D-Fender: this happened to me on monday and it's not at all funny |
| 18:48.55 | fourcheeze | somehow the * refuses to believe that these are genuine credentials |
| 18:49.04 | Juggie | fourcheeze, i dont know why * would deny the registration it shoudnt unless as obvious the name/pass. |
| 18:49.13 | ManxPower | Apparently I'm very wicked, since when I went to take a nap the phone started ringing. |
| 18:49.17 | CunningPike | fourcheeze: Have you pastedbin your sip.conf? |
| 18:49.19 | fourcheeze | Juggie: I know, but it's familiar ground for me |
| 18:49.29 | fourcheeze | CunningPike: my config is in realtime |
| 18:49.43 | CunningPike | fourcheeze: Oh. :-| |
| 18:49.51 | watchy | whats real time |
| 18:50.00 | CunningPike | 11:49 PDT |
| 18:50.08 | Juggie | fourcheeze, is there an * error |
| 18:50.10 | Juggie | to the command lnie |
| 18:50.14 | Juggie | saying reg failed? |
| 18:50.27 | [TK]D-Fender | watchy : Like GMT except it fits into the Imperialist Viewpoint ;) |
| 18:50.34 | fourcheeze | Juggie: no, just the stale nonce one |
| 18:51.03 | Juggie | who's reporting the stale noonce |
| 18:51.07 | Juggie | * or the end point? |
| 18:51.31 | fourcheeze | * |
| 18:51.48 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 18:51.49 | *** join/#asterisk scorn (n=tim@eclipse.sigint.com) |
| 18:52.04 | fourcheeze | Juggie: how do I wipe all the authentication data? |
| 18:52.12 | fourcheeze | wherever it's cached? |
| 18:52.26 | Juggie | i'm not sure the only place it should be cached is the sip_buddies table |
| 18:52.29 | scorn | Does anyone know what the device files /dev/zap/1, 2 etc are outputting? |
| 18:53.30 | watchy | i wish i could print food using a color laser printer |
| 18:53.44 | Juggie | you might try emptying regseconds/ipaddr/fullcontact |
| 18:53.58 | Juggie | those are the only ones i know of that are edited on the fly by realtime |
| 18:54.02 | Qwell[] | watchy: paper is edible |
| 18:54.10 | scorn | If I send audio into my 205P card on channel 1, cat'ing /dev/zap/1 shows a varying pattern, but is it possible to take this and correlate it to the data I put in? |
| 18:54.32 | watchy | qwll but not tasty |
| 18:54.38 | Juggie | Qwell, what causes a stale nuonce and how can you clear it... * has been restarted etc |
| 18:55.27 | fourcheeze | Juggie: ok |
| 18:56.13 | Juggie | fourcheeze, are any of your phones registering? |
| 18:56.18 | fourcheeze | Juggie: some |
| 18:56.32 | Juggie | apparentally when a stale nuonce is received * tells the phone |
| 18:56.37 | Juggie | and it is forced to generate another and register |
| 18:56.40 | Juggie | so its not a fatal error |
| 18:57.02 | fourcheeze | yeah, but the same clients are getting the same error |
| 18:57.11 | scorn | is there an IRC channel for more zaptel specific questions? |
| 18:57.44 | fourcheeze | Juggie: some clients logged straight back in |
| 18:57.51 | fourcheeze | others have repeatedly not logged in |
| 18:57.59 | Juggie | fourcheeze, clients that ignore the request for a new nonce? :) |
| 18:58.05 | Juggie | are they all the same phone? |
| 18:58.28 | fourcheeze | no |
| 18:58.39 | fourcheeze | some snoms and some sipuras |
| 18:58.46 | fourcheeze | interesting some clients didn't log in since last time |
| 18:58.47 | *** join/#asterisk tempest1 (n=asf@adsl-153-42-29.chs.bellsouth.net) |
| 18:58.52 | fourcheeze | those are now logged in |
| 18:59.02 | Juggie | how many out of the 60 are failing? |
| 18:59.03 | fourcheeze | so it has reversed the situation |
| 18:59.15 | Juggie | intresting :) |
| 18:59.15 | fourcheeze | 40 |
| 18:59.18 | fourcheeze | yeah |
| 18:59.23 | Juggie | are some clients directed towards the second ip? |
| 19:00.32 | fourcheeze | some are |
| 19:00.39 | Juggie | well. |
| 19:00.43 | fourcheeze | not many |
| 19:00.47 | Juggie | the ones that arnt working? :) |
| 19:00.52 | fourcheeze | no |
| 19:01.02 | fourcheeze | the ones that aren't working are the ones that are directed to the old IP number |
| 19:01.02 | Juggie | you should haev them all pointed to the same ip. |
| 19:01.06 | hmmhesays | howdy folks |
| 19:01.06 | fourcheeze | sorry |
| 19:01.17 | fourcheeze | Juggie: forget the ones on the old IP number |
| 19:01.20 | fourcheeze | those are the minority |
| 19:01.24 | fourcheeze | and they are all working now I think |
| 19:01.30 | *** join/#asterisk deb_user (n=debian_u@albuquerque.agroinnovations.com) |
| 19:01.37 | fourcheeze | the question is how to clear the nonce issue |
| 19:01.43 | hmmhesays | oej around here? |
| 19:01.46 | Juggie | are the phones hard coded to the ip or to a hostname? |
| 19:01.46 | deb_user | an anybody recommend a good, open source sip softphone |
| 19:01.47 | hmmhesays | ~seen oej |
| 19:01.50 | jbot | oej <n=oej@213.115.215.54> was last seen on IRC in channel #asterisk, 48d 12h 29m 36s ago, saying: 'But morning anyway'. |
| 19:01.53 | Juggie | hmmhesays, oej is on the missing list |
| 19:01.56 | deb_user | I don't really like x-lite that much |
| 19:02.00 | fourcheeze | Juggie: a hostname |
| 19:02.05 | Juggie | presumabbally vacation. |
| 19:03.21 | Juggie | fourcheeze, you are playing with fire having multiple phones at dif ip's same box. when its the same external network. |
| 19:03.31 | fourcheeze | Juggie: that wasn't the plan |
| 19:03.46 | *** join/#asterisk s0lid (n=jlq@gr-153-4.eglobalreach.net) |
| 19:03.53 | fourcheeze | there are only very few on the other IP number/hostname and those were basically the customers we couldn't contact to change |
| 19:03.59 | *** join/#asterisk ctaloi (n=caloi@65.169.134.2) |
| 19:04.05 | fourcheeze | I don't care if they can log in or not right now |
| 19:04.20 | Juggie | your primary * ip |
| 19:04.22 | Juggie | should be eth0 |
| 19:04.27 | fourcheeze | yes it is |
| 19:04.27 | Juggie | which everone you consider that to be |
| 19:04.35 | Juggie | and move all your customers to that. |
| 19:04.39 | fourcheeze | I will do |
| 19:04.47 | fourcheeze | can't do it now - it's after work hours here |
| 19:04.48 | deb_user | Can anybody recommend a good, open source sip softphone? |
| 19:05.18 | Juggie | as i said * has problems with multiple external ips... eg, it can reply to the right address but how does it know which to send qualify packets from |
| 19:05.20 | Juggie | for example. |
| 19:05.46 | Juggie | qualfiy is what keeps nat open but it wont if its comming from a dif source ip then what the nat expects |
| 19:05.48 | fourcheeze | ok, I can see that some clients will need moving |
| 19:06.01 | fourcheeze | what about the ones that don't |
| 19:06.05 | fourcheeze | those are still not registerd |
| 19:06.24 | Juggie | is there anyone of those ppl you can contact to have them reboot their phone? |
| 19:06.29 | Juggie | to see if that will solve it |
| 19:06.31 | mfdutra | Juggie: how can I use a macro to proceed my dialplan while Dial() is in place? |
| 19:06.31 | nortex | deb_user, It is not an exact match, but idefsk from astguru.com is a good little iax softphone |
| 19:06.47 | Juggie | mf Dial(blah,M(macroname)) |
| 19:06.53 | deb_user | nortex: i'm looking for somethign that supports iax and sip |
| 19:06.55 | mfdutra | hmmmm |
| 19:07.05 | Juggie | RTFM :) |
| 19:07.06 | mfdutra | trying |
| 19:07.08 | deb_user | i've just checked out yate...looks cool |
| 19:07.13 | fourcheeze | Juggie: I've got a few phones here |
| 19:07.17 | deb_user | i'm downloading it now |
| 19:07.22 | Juggie | www.voip-info.org look up 'app dial' |
| 19:07.41 | *** join/#asterisk asterboy (n=kevin@S01060204ee2b6007.ed.shawcable.net) |
| 19:08.07 | asterboy | Anyone know if the Thomson SpeedTouch 2030 Phone is in the NorthAmerican market yet? |
| 19:08.26 | asterboy | http://www.thomson.net |
| 19:08.52 | mfdutra | Juggie: but this is before connection. I need to execute commands after connecting |
| 19:09.15 | Juggie | mfdutra, your going to make me look |
| 19:09.18 | Juggie | and i'm going to find it |
| 19:09.22 | Juggie | and then i might not tell you |
| 19:09.41 | Juggie | ok, found it. |
| 19:09.44 | mfdutra | I hope you find it :) |
| 19:09.57 | Juggie | D(digits) |
| 19:10.01 | *** part/#asterisk muh-die-kuh (n=hco@muh-die-kuh.de) |
| 19:10.04 | asterboy | never mind...they obviously don't know how to run a web site |
| 19:10.07 | Juggie | originator wont hear them, but the called party will. |
| 19:10.10 | mfdutra | yes, I tried that, but it didn't work |
| 19:10.13 | Juggie | it does |
| 19:10.16 | Juggie | you just dont hear them |
| 19:10.24 | asterboy | http://www.thomson.net/EN/Home/MiniSites/BAP/Telecom/SubCategory.html?category=Telephony |
| 19:10.28 | asterboy | errors out |
| 19:10.28 | mfdutra | the pbx in the other side was still playing the dialtone |
| 19:10.42 | asterboy | no one using the Thomson speedtouch 2030 |
| 19:10.43 | asterboy | ?? |
| 19:10.44 | mfdutra | as if it hadn't received the dtmfs |
| 19:10.49 | Juggie | try D(ww123456) whatever |
| 19:10.58 | mfdutra | hmm maybe |
| 19:10.59 | mfdutra | trying |
| 19:11.14 | Juggie | call your own phone, and see if you can hear dtmf's when it calls |
| 19:11.23 | fourcheeze | Juggie: I've restarted a snom and a sipura and neither registered |
| 19:11.25 | mfdutra | asme thing |
| 19:11.32 | nortex | asterboy, Never heard of it sorry. |
| 19:11.34 | Juggie | same nuonce error? |
| 19:11.34 | mfdutra | same.. |
| 19:11.42 | Juggie | mfdutra, then call one of your phones |
| 19:11.44 | Juggie | with that option |
| 19:11.48 | Juggie | and see if you can hear the dtmfs |
| 19:11.53 | mfdutra | ok |
| 19:12.34 | fourcheeze | Juggie: * sends repeated 401 |
| 19:13.07 | Juggie | fourcheeze, there is no auth rejected message on the console? |
| 19:13.19 | fourcheeze | Juggie: nope |
| 19:13.26 | asterboy | ok thanks for the feedback nortex |
| 19:13.34 | Juggie | could it possibly have something to do w/ the domain i wonder? |
| 19:13.48 | mfdutra | Juggie: I can hear it |
| 19:14.28 | mfdutra | the problem seems to be that the other part in the E1 channel doesn't send the "answer" bit |
| 19:14.52 | Juggie | fourcheeze, make sure all your domains are listed in sip.conf |
| 19:15.19 | Juggie | you could optionally also enable allowexternaldomains=yes |
| 19:17.02 | hmmhesays | anyone tried to patch asterisk with the latest bristuff? |
| 19:17.10 | Juggie | have you? |
| 19:17.28 | hmmhesays | i just did |
| 19:17.35 | hmmhesays | and chan_sip fails to compile |
| 19:17.40 | tasat- | can any recommend a good low overhead gui monitor? |
| 19:17.48 | hmmhesays | ssh |
| 19:18.00 | CunningPike | tasat-: To monitor what? |
| 19:18.44 | tasat- | CunningPike: asterisk call status: not really ncessary to control, most importantly to see what's happening on multiple channels... |
| 19:18.56 | CunningPike | tasat-: Try FOP |
| 19:18.59 | CunningPike | ~fop |
| 19:19.00 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 19:19.16 | tasat- | CunningPike: I'll check it out... any particular reasons why you like it? |
| 19:19.25 | CunningPike | Silly jbot. Flash Operator Panel - www.asternic.orfg |
| 19:19.34 | CunningPike | s/orfg/org/ |
| 19:19.55 | CunningPike | tasat-: It was easyish to set up and seems to work quite well |
| 19:20.11 | tasat- | sounds good |
| 19:20.31 | CunningPike | tasat-: Tried HUDLite - still can't get it working, and the list of dependancies is painful |
| 19:21.38 | fourcheeze | Juggie: I didn't have my domains listed they now are - |
| 19:21.52 | fourcheeze | Juggie: doesn't seems to have changed things |
| 19:21.56 | tasat- | CunningPike: what about AstGUIMan or something like that? |
| 19:21.57 | Juggie | did you restart? |
| 19:22.02 | fourcheeze | Juggie: I still don't understand why after restarting I see peers |
| 19:22.05 | fourcheeze | immediately |
| 19:22.08 | nortex | CunningPike, No joke, plus you duplicate all your config to get the buttons in the HUD |
| 19:22.25 | Juggie | that would only happen if you were using static realtime, or if you had peers defined in sip.conf |
| 19:22.33 | CunningPike | tasat-: Not intimately familiar with that one, but iirc it's a config manager rather than a monitor |
| 19:22.40 | CunningPike | nortex: Yah, it bites |
| 19:23.48 | fourcheeze | Juggie: nope definietly dynamic |
| 19:24.14 | Juggie | then i haev no idea. |
| 19:24.41 | fourcheeze | Juggie: ok, after a sip reload it does show no peers to start with |
| 19:24.47 | Juggie | :) |
| 19:24.56 | fourcheeze | Juggie: I think you must know ;-) |
| 19:25.01 | fourcheeze | who would know? |
| 19:25.11 | Juggie | whats the problem |
| 19:25.17 | Juggie | still nonce? |
| 19:25.35 | Juggie | maybe turn your debugging up higher |
| 19:25.39 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-8.getnet.net) |
| 19:25.39 | Juggie | see if theres more errors your not seeing |
| 19:25.52 | Juggie | i'm not sure what would be causing the problem your having i've never seen it... |
| 19:26.39 | *** join/#asterisk arkonadev (n=chatzill@65.203.186.131) |
| 19:27.23 | arkonadev | so if i am calling an bash script on the asterisk server and append the variable like a query string would that variable be available through the STDIN? |
| 19:27.34 | arkonadev | so AGI(myscript.agi?var) |
| 19:27.45 | fourcheeze | Juggie: yeah it's a bad one |
| 19:27.54 | Juggie | theres a reason the auth is failing |
| 19:27.58 | Juggie | you must find it :) |
| 19:28.18 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 19:28.31 | fourcheeze | Juggie: ok one clue is that it goes if you change the hostname people register with |
| 19:29.08 | Juggie | did you try allowexternalhostnames or whatever the setting is? |
| 19:29.28 | arkonadev | how would i pass just one variable to my agi script with the AGI() applicatoin? |
| 19:29.36 | Juggie | RTFM |
| 19:30.09 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 19:30.10 | arkonadev | yeah it says AGI(program[,arguments]) so it would be AGI(script.agi,var) |
| 19:30.20 | Juggie | AGI(script|arg1|arg2....) |
| 19:30.29 | Primer | Anyone using polycom phones with the agent login button? |
| 19:30.31 | arkonadev | k thanks |
| 19:30.44 | Primer | trying this branch where it supposedly works...but doesn't |
| 19:30.49 | *** join/#asterisk tempest1 (n=asf@adsl-230-226-81.chs.bellsouth.net) |
| 19:31.10 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:31.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:31.18 | Juggie | RUSSEL! |
| 19:31.29 | Juggie | +L, its about time you got out of bed. |
| 19:31.36 | Primer | SplasPood: and it's not as simple as configuring the button to dial a specific extension. Because there are other buttons to effect availability once logged in |
| 19:32.13 | Primer | what'd be nice is if the polycom just just send its xml to another port and have a customized daemon to interpret the XML and send AGI to asterisk |
| 19:32.34 | *** join/#asterisk _alex_mx_ (n=alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
| 19:33.32 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402BD8F.pool.t-online.hu) |
| 19:34.18 | *** join/#asterisk twisted[asteria] (n=twisted@pdpc/supporter/active/twisted) |
| 19:34.18 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 19:34.21 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402BD8F.pool.t-online.hu) |
| 19:34.34 | watchy | is it possible to stream video with *? |
| 19:35.05 | *** join/#asterisk azzie (n=az@azzie.net) |
| 19:35.11 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402BD8F.pool.t-online.hu) |
| 19:36.34 | *** part/#asterisk _alex_mx_ (n=alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
| 19:37.12 | azzie | anybody familiar with Cisco MGC (pgw2200) ? |
| 19:37.25 | Primer | watchy: I use ekiga with video on asterisk |
| 19:37.37 | De_mon | anyone used the SIP trunking provider broadband.com? I'm not quite sure how setup asterisk to receive their calls |
| 19:37.41 | watchy | whats ekiga? |
| 19:37.53 | Primer | SIP video program for Linux |
| 19:38.06 | watchy | uses webcam to send video? |
| 19:38.06 | Primer | open sores, standards compliant |
| 19:38.13 | Primer | aye |
| 19:38.20 | watchy | wow, it run on winderz? |
| 19:38.23 | Primer | no |
| 19:38.32 | watchy | is there a windows variant |
| 19:38.36 | Primer | windows has its own standards compliant SIP video programs |
| 19:38.39 | Primer | but they're pay |
| 19:38.42 | Primer | xten has one |
| 19:38.54 | Primer | don't know of any free ones |
| 19:39.36 | watchy | im suprised sjphone dont do video |
| 19:39.44 | fourcheeze | Juggie: it's very weird now - I'm not getting any messages at all |
| 19:39.51 | fourcheeze | Juggie: and hardly anyone logging in now |
| 19:41.02 | watchy | wow that eyebeam is cool |
| 19:41.55 | watchy | i just found a demo of it imma leech it |
| 19:42.27 | [TK]D-Fender | watchy : I use eyebeam w/ video over SIP. Works great |
| 19:42.47 | watchy | any major config required? |
| 19:43.35 | GerbilWrk | what variable is used to get the called number? like if you have numbers with wildcards, and need to transfer it back to the correct one that was dialed? |
| 19:45.47 | watchy | i wonder why my nzb om atic is broke |
| 19:45.50 | watchy | i hate windows |
| 19:48.30 | watchy | i hope they replace my internet at work tommorow |
| 19:48.41 | [TK]D-Fender | watchy : nope, pretty easy |
| 19:48.52 | watchy | i run a wisp and im not even on our backbone :( |
| 19:51.07 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-8.getnet.net) |
| 19:51.21 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 19:53.27 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 19:54.51 | fourcheeze | ok, can anyone help |
| 19:54.54 | fourcheeze | ? |
| 19:55.04 | fourcheeze | I will buy anyone a real crate of beer :-) |
| 19:55.29 | fourcheeze | if they can let me know how to stop asterisk just saying 401 to almost every client |
| 19:55.33 | fourcheeze | after changing IP number |
| 19:55.37 | russellb | what, beer?! |
| 19:55.45 | Juggie | russ, try. |
| 19:55.45 | fourcheeze | your choice |
| 19:55.46 | Juggie | i give up. |
| 19:56.03 | Juggie | he has * running w/ multiple external ips :) |
| 19:56.07 | russellb | fourcheeze: after chaning IP on your own box? |
| 19:56.10 | russellb | ah. |
| 19:56.15 | fourcheeze | yep |
| 19:56.18 | russellb | sip reload? |
| 19:56.21 | fourcheeze | hehe |
| 19:56.24 | fourcheeze | if only |
| 19:56.38 | Juggie | hes getting unable to auth problems, i thougth it might be domain related |
| 19:56.48 | Juggie | fourcheeze, did you try the allowexternal domains setting? |
| 19:56.49 | fourcheeze | I can remove 2 IP numbers if that helps |
| 19:56.55 | fourcheeze | Juggie: yes |
| 19:57.06 | Juggie | explain the current problem. |
| 19:57.13 | Juggie | so russ can hear it |
| 19:57.14 | russellb | my mind is not in a state capable of debugging problems of any complexity. |
| 19:57.24 | fourcheeze | russellb: that may be an advantage |
| 19:57.43 | Juggie | russ did you commit the fix to channel.c? |
| 19:57.55 | fourcheeze | I had a problem with keepalives going from the wrong interface |
| 19:57.59 | fourcheeze | so I swapped IP numbers |
| 19:58.07 | fourcheeze | and now people can't register |
| 19:58.13 | fourcheeze | about 75% of them can't |
| 19:58.23 | fourcheeze | asterisk says 401 Unauthorised |
| 19:58.33 | fourcheeze | and I sometimes get a tonne of stale nonce messages |
| 19:58.35 | Taker | Hi! Got no error compiliing bristuff, but loading the module: http://nopaste.debianforum.de/3816 |
| 19:58.38 | [TK]D-Fender | fourcheeze : did you bind to 1 IP or is it at 0.0.0.0? |
| 19:58.46 | fourcheeze | 0.0.0.0 |
| 19:58.58 | fourcheeze | so the main IP address of the box changed |
| 20:00.05 | watchy | why have multi ips? are they used for voip? |
| 20:00.25 | *** join/#asterisk ikey (i=ikey@220.226.41.90) |
| 20:00.28 | fourcheeze | watchy: I don't need them any more - it was from a failover arrangement |
| 20:00.28 | arkonadev | so im tyring to set a varible in the dialplan through a bash script from what i can tell from the book all you do is echo "SET VARIABLE var value" |
| 20:00.36 | watchy | then bind 1 ip to asterisk |
| 20:00.40 | fourcheeze | ok |
| 20:00.42 | fourcheeze | I'll try that |
| 20:00.43 | [TK]D-Fender | fourcheeze : CHANGE THAT |
| 20:00.49 | arkonadev | then in the dialplan i go ${var} and it doesnt seem to be working |
| 20:00.52 | arkonadev | am i doing this wrong |
| 20:00.56 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 20:00.57 | watchy | because i dunno how the hell * knows what ip to use |
| 20:01.06 | watchy | i guess if 0000 is binded its gonna use any ip it feels like |
| 20:01.11 | watchy | which isnt a good idea |
| 20:01.40 | watchy | because you may get a sip client come in on the voip ip and outgoing packets out on another to the sip client i think |
| 20:01.42 | nortex | I thought 0.0.0.0 uses all IP's |
| 20:01.47 | watchy | i dont think that would work very well |
| 20:01.57 | fourcheeze | udp 0 0 212.187.197.142:5060 0.0.0.0:* |
| 20:02.03 | watchy | nortex: but when * sends an out going packet its not gonna send it on EVERY ip |
| 20:02.09 | watchy | i wouldnt think it would |
| 20:02.20 | file | it sends it on the first bound IP, which makes things... interesting sometimes |
| 20:02.36 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 20:02.39 | watchy | id just bind one ip to * so it has no choice |
| 20:02.42 | watchy | its the ips bitch |
| 20:02.44 | fourcheeze | ok guys you are great |
| 20:02.49 | watchy | did that fix it |
| 20:02.55 | fourcheeze | I mean you are the best |
| 20:03.01 | watchy | where me and tks crate of beer |
| 20:03.08 | fourcheeze | watchy: pm me :-) |
| 20:03.10 | watchy | i want guinness |
| 20:03.12 | fourcheeze | ok |
| 20:03.20 | *** join/#asterisk jcwunder (n=chris@ppp-82-135-70-201.dynamic.mnet-online.de) |
| 20:03.29 | nortex | I guess on mine since the interfaces are on seperate subnets it does okay. |
| 20:03.43 | dlynes_laptop | yeah..same with if you buy g729 codec licenses from asterisk...they generate license keys based on your first nic adapter |
| 20:03.58 | dlynes_laptop | so if you change out that nic and replace it with another one, you have to reregister |
| 20:04.06 | watchy | i am guessing the voip came in on one ip and packets to the peer was going from another and that was causing the issue |
| 20:04.10 | fourcheeze | yeah |
| 20:04.13 | fourcheeze | terrible |
| 20:04.24 | fourcheeze | so the bug is in the comment in sip.conf |
| 20:04.33 | fourcheeze | which should say "always fill this in" |
| 20:05.04 | SplasPood | Primer: A) respond to me in the same channel on the same day B) So you have those buttons invoke other dialplan logic.. |
| 20:05.19 | dlynes_laptop | Has anyone checked out mcc? |
| 20:05.21 | watchy | yea if i had multiple ips i would fill it in |
| 20:05.26 | Primer | SplasPood: they simply send XML |
| 20:05.40 | dlynes_laptop | I'm just curious how mcc is for postpaid billing |
| 20:05.54 | charles___ | dlynes_laptop: I've done |
| 20:05.58 | Primer | SplasPood: and expect XML to know that that actions have been taken |
| 20:06.08 | dlynes_laptop | charles___: how is it for postpaid? |
| 20:06.14 | dlynes_laptop | charles___: or do you only use it for prepaid? |
| 20:06.15 | SplasPood | Primer: I have no idea what you're talking about |
| 20:06.17 | charles___ | dlynes_laptop: didn't worked for me |
| 20:06.26 | charles___ | dlynes_laptop: no LCR available |
| 20:06.27 | dlynes_laptop | charles___: what didn't work about it? |
| 20:06.35 | TrixVox | Who do you guys use for supermicro-type boxes? |
| 20:06.50 | charles___ | dlynes_laptop: but it's one of the closest ones to work |
| 20:07.08 | Primer | SplasPood: the menu options reflect the status of the agent. i.e. logged in or not, available or not |
| 20:07.11 | dlynes_laptop | charles___: no lcr available? I don't follow you (I know what lcr is) |
| 20:07.15 | fearnor | hi splas! |
| 20:07.15 | *** join/#asterisk apardo (n=apardo@87.217.147.51) |
| 20:07.22 | SplasPood | menu options? |
| 20:07.26 | SplasPood | fearnor: hola |
| 20:07.31 | charles___ | dlynes_laptop: the software doesn't do LCR |
| 20:07.40 | *** join/#asterisk mikebwilliams (n=mike@64.184.24.94) |
| 20:07.41 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 20:07.41 | charles___ | dlynes_laptop: and has issues |
| 20:07.45 | mikebwilliams | hey all |
| 20:07.46 | dlynes_laptop | ah |
| 20:07.50 | charles___ | dlynes_laptop: but it's one of the closest ones to be fully working |
| 20:07.54 | Primer | SplasPood: ok, I don't know what your polycom phone looks like, but mine has an LCD screen |
| 20:08.01 | dlynes_laptop | I'm not worried about LCR, personally, at this point |
| 20:08.12 | dlynes_laptop | It's something I'd like to consider for the future though |
| 20:08.18 | charles___ | dlynes_laptop: I tried others too |
| 20:08.23 | SplasPood | Primer: yea, but there are buttons down the side... When you originally asked you said 'redefine the buttons' |
| 20:08.24 | mikebwilliams | i think i may have found a bug in Asterisk, and was wondering if anyone could give me some pointers on where to look in the source |
| 20:08.28 | Primer | SplasPood: there are 4 buttons at the bottom of said screen. button 3 is labelled "login", button 4 is unlabeled |
| 20:08.31 | watchy | how hard is it to stream local radio station to asterisk MOH? |
| 20:08.32 | charles___ | dlynes_laptop: but mcc was one of the top 2 |
| 20:08.39 | dlynes_laptop | charles___: yeah...I was in the process of writing my own |
| 20:08.40 | SplasPood | Primer: Oh well I have that turned off. |
| 20:08.49 | Primer | SplasPood: THAT is what we're trying to use |
| 20:08.52 | hmmhesays | ok manually patching anything reall really sucks |
| 20:08.55 | hmmhesays | *really really really |
| 20:09.08 | dlynes_laptop | charles___: had almost all of it written, but then I ran into issues trying to figure out how to code certain parts in perl |
| 20:09.09 | SplasPood | Primer: Ok, you cannot, and thats not how you explained it originally |
| 20:09.10 | charles___ | dlynes_laptop: me too |
| 20:09.17 | charles___ | dlynes_laptop: I believe it's a great step |
| 20:09.18 | SplasPood | Primer: You can, however, redefine the hard buttons |
| 20:09.28 | charles___ | dlynes_laptop: I only need to rate cdr's and take out money, nothing special |
| 20:09.31 | mikebwilliams | asterisk is sending a bad cancel uri: CANCEL :18122228221@64.184.24.13:5070 SIP/2.0 |
| 20:09.50 | dlynes_laptop | charles___: I used perl because i wanted something lightweight running on the server; it would have been easier in Java as I know Java backwards and forwards, but I didn't want that kind of a load on the server |
| 20:10.11 | charles___ | dlynes_laptop: most of the software tries to do to many things and ends up doing nothing |
| 20:10.19 | charles___ | dlynes_laptop: for example the astbill that one doesn't do anything |
| 20:10.35 | dlynes_laptop | charles___: yeah...I need a full post-paid billing solution, and a prepaid billing solution would also be nice |
| 20:10.54 | dlynes_laptop | charles___: however, I need to keep historical data, and output bills in pdf format |
| 20:11.13 | charles___ | dlynes_laptop: let me read you |
| 20:11.49 | dlynes_laptop | charles___: plus keep track of their monthly line charges, depending on what options they've ordered on their lines |
| 20:11.57 | charles___ | dlynes_laptop: second |
| 20:12.44 | charles___ | dlynes_laptop: okay I read you |
| 20:12.46 | dlynes_laptop | charles___: well, and allow for adjustments on an individual bill basis, for when customers bitch about service problems and want their bill lowered |
| 20:12.53 | charles___ | dlynes_laptop: I've done that stuff in PHP |
| 20:13.18 | charles___ | dlynes_laptop: I believe php or perl or any other language can do that |
| 20:13.23 | dlynes_laptop | Yeah..php would've been useful, too |
| 20:13.28 | dlynes_laptop | I just don't know php well enough |
| 20:13.35 | charles___ | dlynes_laptop: do you know perl ? |
| 20:13.42 | dlynes_laptop | charles___: somewhat |
| 20:13.45 | charles___ | dlynes_laptop: not that different have you made perl cgi's ? |
| 20:13.56 | dlynes_laptop | charles___: i learned it a lot more working on that billing application :) |
| 20:14.09 | charles___ | dlynes_laptop: which application ? |
| 20:14.17 | dlynes_laptop | charles___: our own billing application |
| 20:14.22 | charles___ | dlynes_laptop: the thing is mcc is going to v 1.5 |
| 20:14.32 | charles___ | dlynes_laptop: I believe it will be good in V.1.5 |
| 20:14.38 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 20:14.47 | dlynes_laptop | charles___: I just ran into a roadblock with it, so I'm trying to get a guy that does perl programming for us from time to time to finish it |
| 20:14.52 | charles___ | dlynes_laptop: I don't really have the time now to invest on my own application but I know it can be done in a week |
| 20:15.18 | charles___ | dlynes_laptop: mcc v 1.5 may be a good switch for u |
| 20:15.28 | dlynes_laptop | Maybe |
| 20:15.39 | dlynes_laptop | I'm evaluating 1.4 atm |
| 20:15.39 | charles___ | dlynes_laptop: the guy who develops it Mindaugas, he is into making the software work |
| 20:15.52 | charles___ | dlynes_laptop: his design has some glitches to make it work in US |
| 20:15.58 | charles___ | dlynes_laptop: but still may be modified |
| 20:16.23 | charles___ | dlynes_laptop: where're you from ? |
| 20:16.33 | *** join/#asterisk brijn (n=bas@204.244.176.116.net-conex.com) |
| 20:16.37 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
| 20:16.39 | brijn | Hello all |
| 20:16.52 | cbiggers | need a hug? |
| 20:17.19 | brijn | Me? :) |
| 20:17.25 | cbiggers | anyone |
| 20:18.07 | dlynes_laptop | charles___: canada |
| 20:18.08 | GerbilWrk | what variable is used to get the called number? like if you have numbers with wildcards, and need to transfer it back to the correct one that was dialed? |
| 20:18.09 | charles___ | cbiggers: you gotta chill |
| 20:18.18 | dlynes_laptop | charles___: yeah, i noticed he's into making it work |
| 20:18.19 | charles___ | dlynes_laptop: same dialing style as in US |
| 20:18.24 | dlynes_laptop | charles___: correct |
| 20:18.30 | macTijn | grr |
| 20:18.49 | dlynes_laptop | charles___: I'm a programmer, so if I run into problems, I can always submit a patch back to him that'll fix it |
| 20:19.04 | charles___ | dlynes_laptop: but you gonna notice his stuff oriented to england with VAT and 1 in front of the country code, kind of sucks to import rate tables, but it can be done |
| 20:19.09 | dlynes_laptop | charles___: I can fix php code...I just don't know it well enough to write soemthing from scratch |
| 20:19.26 | dlynes_laptop | charles___: that's not a problem...we have GST (same thing as VAT) |
| 20:19.53 | charles___ | dlynes_laptop: we have STD's and all kinds |
| 20:20.05 | dlynes_laptop | charles___: where i come from, STD is something nasty |
| 20:20.11 | charles___ | dlynes_laptop: :P |
| 20:20.21 | dlynes_laptop | charles___: what did you mean by STD? |
| 20:20.23 | [TK]D-Fender | :O |
| 20:20.26 | [TK]D-Fender | STANDARDS |
| 20:20.32 | dlynes_laptop | ah |
| 20:20.33 | dlynes_laptop | heh |
| 20:21.03 | charles___ | :P |
| 20:21.12 | file | who has an STD?!? |
| 20:21.18 | dlynes_laptop | file: does |
| 20:21.30 | dlynes_laptop | file got it from Corydon-w |
| 20:21.32 | charles___ | file: yes file tell us about it. does it hurt ? |
| 20:21.37 | fourcheeze | Juggie: now I have my working registrations how can I test my losing keepalives problem? |
| 20:21.41 | macTijn | anyone here familiar with ShoreTel PBX stuff ? |
| 20:21.49 | dlynes_laptop | macTijn: yeah...expensive ass shit |
| 20:21.51 | file | eep |
| 20:22.02 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 20:22.12 | dlynes_laptop | macTijn: it's an IP-based system |
| 20:23.03 | dlynes_laptop | [TK]D-Fender: enough...he's already been poked enough by Corydon-w |
| 20:23.07 | hmmhesays | haha |
| 20:23.08 | cbiggers | this damn emerge --sync is takin awhile |
| 20:23.09 | [TK]D-Fender | :O |
| 20:23.42 | charles___ | dlynes_laptop: are you using mcc already ? |
| 20:24.00 | dlynes_laptop | charles___: no |
| 20:24.05 | dlynes_laptop | charles___: just evaluating it |
| 20:24.19 | charles___ | dlynes_laptop: he just migrated mcc |
| 20:24.25 | dlynes_laptop | charles___: but at least it uses postgresql, not mysql |
| 20:24.56 | charles___ | dlynes_laptop: doesn't matter, both are working solutions |
| 20:25.09 | charles___ | dlynes_laptop: I've been dealing with mysql and postgresql for yeras |
| 20:25.19 | dlynes_laptop | charles___: yeah...I just prefer postgresql |
| 20:25.38 | dlynes_laptop | charles___: besides...I only want one db server running on my machine |
| 20:26.04 | charles___ | dlynes_laptop: did he backed from gpl and is going to release mcc only for paid customers ? |
| 20:26.24 | dlynes_laptop | charles___: nope...still gpl |
| 20:26.35 | charles___ | dlynes_laptop: look the website, has been changed today |
| 20:26.41 | charles___ | dlynes_laptop: kolmisoft |
| 20:26.43 | dlynes_laptop | charles___: funny |
| 20:26.49 | dlynes_laptop | charles___: I just downloaded it this morning |
| 20:27.19 | *** join/#asterisk tempest1 (n=asf@adsl-150-198-150.chs.bellsouth.net) |
| 20:27.29 | dlynes_laptop | http://www.paskambink.lt/mcc/ |
| 20:27.48 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:27.58 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:27.58 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 20:28.06 | De_mon | I've got sip privoder that does ip based authentication.. I setup a [sip-provider] definition in sip.conf but when i make calls though the hat provider I just get disconnected and asterisk doesn't log anything |
| 20:28.19 | Dr-Linux | hey all |
| 20:28.44 | dlynes_laptop | hey |
| 20:29.17 | [TK]D-Fender | bbiab |
| 20:29.44 | De_mon | if sip debug ip <provider ip> doesn't show any traffic when I try to receive a call from them, does that mean asterisk is misconfigured or they didnt provision it correctly? |
| 20:29.59 | dlynes_laptop | De_mon: or perhaps your firewall is blocking it? |
| 20:30.19 | De_mon | its possible, but 5060 isn't blocked at fw |
| 20:30.22 | cbiggers | anyone here run OS x? |
| 20:30.26 | cbiggers | i got a quick question |
| 20:30.38 | jhiver | de_mon, you could do an ngrep -Wbyline host <provider_ip> to check if the traffic is reaching your box |
| 20:31.52 | *** join/#asterisk XARiUS (n=bdarcy@71.142.177.252) |
| 20:32.02 | brijn | QoS question: I have astshaper configured, it checks based on port nr it seems (4569), when I look on the line traffic to the other SIP party does not go over that port |
| 20:32.16 | brijn | Does astshaper just look at the tos field, and not at the port nr? |
| 20:32.33 | XARiUS | hey guys, anyone seen odd issues with 7960's, where in a app queue environment, the phone just never stops ringing??? |
| 20:32.37 | De_mon | nada from provider how nice |
| 20:33.10 | XARiUS | (like even after the caller has gone to vm, and the queue was exited) |
| 20:33.23 | GerbilWrk | Anyone know how to send a variable from one context to another? |
| 20:33.27 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 20:33.36 | generalhan | XARiUS: what FW version you using ? |
| 20:33.44 | XARiUS | I've tried 7.4. 7.5.. 8.2... 8.3.. |
| 20:33.45 | XARiUS | lol |
| 20:33.47 | *** join/#asterisk Alex (n=hauntedu@about/mandrake/pdpc.student.HauntedUnix) |
| 20:33.52 | Alex | Woo Hoo! My zaptel card works! :) |
| 20:33.59 | dlynes_laptop | wow |
| 20:34.03 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 20:34.08 | generalhan | oh lol |
| 20:34.19 | jhiver | De_mon, then it's not your * config |
| 20:34.25 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 20:34.31 | XARiUS | 8.2 and 8.3 still do it but are unusable due to a problem with qualify= |
| 20:34.36 | charles___ | XARiUS: wich version ? |
| 20:34.38 | jhiver | ngrep is your friend, it's also nice to debug SIP :) |
| 20:34.42 | charles___ | XARiUS: 8.3 ? |
| 20:34.51 | jhiver | like ngrep -Wbyline port 5060 |
| 20:34.59 | XARiUS | 8.2 and 8.3 have a qualify= issue, phone unreg's and won't re-reg (documented on wiki also) |
| 20:35.02 | jhiver | = all SIP stuff, no added fluff |
| 20:35.03 | XARiUS | running 7.4 right now |
| 20:35.05 | charles___ | XARiUS: I use 8.2 here with no quality problems |
| 20:35.07 | dlynes_laptop | charles___: i guess you weren't downloading it from the lithuanian site? |
| 20:35.18 | generalhan | i use 8-2 here no issues either |
| 20:35.20 | XARiUS | not quality, QUALIFY= in sip.conf, these phones are natted at home offices. |
| 20:35.23 | generalhan | 8-3 on the other hand |
| 20:35.26 | generalhan | STUPID 8-3-00 |
| 20:35.45 | charles___ | XARiUS: no nat here |
| 20:35.52 | charles___ | XARiUS: but it works behind nat too |
| 20:36.13 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 20:36.13 | charles___ | XARiUS: I'm using 8.2 didn't know that 8.3 was available already |
| 20:36.30 | XARiUS | yeah the nat part is fine, it's just lately if a caller hit's a queue for say 60 seconds (which rings say 3 7960's), when they exit the queue after timeout, and get dumped into vm.. the 7960's just keep on ringing. |
| 20:36.33 | XARiUS | oddest damn thing. |
| 20:36.47 | generalhan | yes it is |
| 20:36.58 | charles___ | XARiUS: try 8.2 |
| 20:37.04 | XARiUS | yeah I did, same issue. |
| 20:37.05 | charles___ | generalhan: where did you get 8.3 ? |
| 20:37.15 | XARiUS | actually, I might have skipped 8.2 and went right to 8.3. |
| 20:37.19 | XARiUS | brb. :) |
| 20:37.20 | charles___ | XARiUS: which asterisk ? |
| 20:37.37 | XARiUS | SVN-branch-1.2-r38761 |
| 20:37.53 | charles___ | XARiUS: phones have an issue downgrading , I've read something that says it can't downgrade from 8 not sure exactly |
| 20:38.23 | XARiUS | charles: this one went from 8.3 to 7.5, then 7.4 just fine. |
| 20:38.35 | charles___ | XARiUS: only occurs with the 7960 or does with softphones or other phones ? |
| 20:38.52 | XARiUS | just 7960's.. queue config is identical to other customers.. only difference? other customers are running Polycom 501's. |
| 20:39.05 | charles___ | where did you guys got 8.3 ? |
| 20:39.13 | XARiUS | the cisco support site.. have to have an account. |
| 20:39.19 | *** join/#asterisk Olobola (n=casper_s@adsl-75-2-154-84.dsl.pltn13.sbcglobal.net) |
| 20:39.37 | charles___ | XARiUS: I found 8.2 there only |
| 20:39.45 | XARiUS | I just snagged 8.3 off there this morning |
| 20:39.49 | XARiUS | must be very new |
| 20:40.22 | Olobola | is knew phone incoming OK? I can't tell if it's me or not. |
| 20:45.07 | arkonadev | has anyone ever done a tcpping in a bash agi script? |
| 20:45.46 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 20:45.48 | Wazb^ | hi |
| 20:46.02 | hmmhesays | hyroken |
| 20:46.08 | hmmhesays | right in the junk |
| 20:46.09 | hmmhesays | ouch |
| 20:46.43 | Wazb^ | could anyone tell me what file asterisk use for ringback when we use r option in Dial command ? |
| 20:47.14 | russellb | Wazb^: it's from the information in indications.conf |
| 20:48.26 | arkonadev | when you are doing AGI debug what is tx and what is rx |
| 20:50.13 | Wazb^ | there is no file name mentioned over there |
| 20:51.19 | XARiUS | *bashes all the 7960's with a bat* |
| 20:51.29 | XARiUS | I'm really starting to dislike cisco in favor of polycom. |
| 20:51.37 | jcwunder | is there someone around using debian asterisk-bristuff-1.2.7.1.dfsg-3 and zaphfc ? |
| 20:52.49 | charles___ | XARiUS: only occurs with the 7960 or does with softphones or other phones ? |
| 20:53.35 | arkonadev | anyone here have much experience right bash agi scripts? |
| 20:53.39 | arkonadev | writing |
| 20:54.30 | charles___ | arkonadev: #!/bin/sh \n echo "What are you looking for"; |
| 20:55.22 | arkonadev | well i am using a bash script to ping an ip before i send an AGI script to that ip the problem is i dont think that the dial plan can call an agi script that uses tcpping because it always returns nothing |
| 20:55.32 | arkonadev | but when i just use a normal ping it works |
| 20:55.38 | *** join/#asterisk edorsey (i=edorsey@216-63-192-130.ded.swbell.net) |
| 20:55.40 | arkonadev | but i need to ping a certain port so tcp ping is a must |
| 20:55.49 | edorsey | anyone ever heard of a company called arelnet? |
| 20:56.01 | charles___ | arkonadev: you can't ping a port |
| 20:56.09 | charles___ | arkonadev: there is no such thing ping a port |
| 20:56.27 | arkonadev | tcpping -x 1 ipaddress port |
| 20:56.27 | charles___ | arkonadev: if you want to check if the port is opened is other thing totally different |
| 20:56.36 | XARiUS | charles: only the 7960's |
| 20:56.37 | arkonadev | will ping a particular port |
| 20:57.02 | edorsey | its not technically a ping |
| 20:57.21 | arkonadev | when i call the command through the command line it works perfectly but when i try to call a bash script with that command with the AGI() application it doesnt work |
| 20:57.28 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 20:57.35 | charles___ | I would never get a tool called tcpping on my server, it may get my server 0wn3d and rootkited in seconds |
| 20:58.02 | charles___ | arkonadev: you have to look for the answer |
| 20:58.07 | arkonadev | im not running the server...it was already there so im using it |
| 20:58.20 | arkonadev | what do you think i am in here for |
| 20:58.23 | charles___ | arkonadev: software is writting to standard output redirect it and see what it returns |
| 20:58.43 | arkonadev | it returns and empty string when i call it from the dialplan but returns the correct info when i call it through the command line |
| 20:58.57 | arkonadev | and=an |
| 20:58.59 | charles___ | arkonadev: did you checked permissions ? |
| 20:59.21 | charles___ | arkonadev: maybe your tool needs root capabilities to do that kind of powerfull ping |
| 20:59.31 | *** part/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 21:00.11 | arkonadev | the script has nobody root |
| 21:00.35 | charles___ | arkonadev: +x to nobody ? |
| 21:00.53 | charles___ | arkonadev: what does your ping thing requires ? |
| 21:01.21 | charles___ | arkonadev: do your tests in some other way get snmp to check the stuff |
| 21:01.34 | arkonadev | not sure im not really the server admin and knew to the whole unix thing but i think your right because script that right has the correct permission but the tcpping command might not |
| 21:01.41 | arkonadev | knew =new |
| 21:01.47 | hmmhesays | app_devstate.c:65: error: too many arguments to function ‘ast_device_state_changed_literal’ |
| 21:01.52 | hmmhesays | what does that mean in general |
| 21:02.03 | charles___ | arkonadev: ask the admin why he doesn't use SNMP instead |
| 21:02.21 | charles___ | arkonadev: snmpget can return the value that you want |
| 21:02.29 | arkonadev | aright im going to see if he is available |
| 21:02.45 | charles___ | hmmhesays: buffer overflow on your ast_device :P |
| 21:02.57 | hmmhesays | its a compile error bah |
| 21:04.05 | charles___ | hmmhesays: see the line, probably it's passing too many parameters to ast_device_state_changed_literal() |
| 21:04.20 | charles___ | hmmhesays: readd app_devstate.c line 65 |
| 21:04.31 | charles___ | hmmhesays: are you getting from SVN ? |
| 21:04.59 | hmmhesays | yeah, how many arguments is ast_device_state_changed_literal looking for |
| 21:05.16 | kfudge | Damn I just can't figure out this on hold issue that I am having |
| 21:05.17 | charles___ | hmmhesays: get stable version, won't have that trouble |
| 21:05.19 | hmmhesays | apparently not 3 |
| 21:05.26 | hmmhesays | heh |
| 21:05.34 | hmmhesays | don't give bad advice |
| 21:05.36 | charles___ | hmmhesays: to answer that I will have to read the headers of ast_device_state_changed_literal |
| 21:05.44 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 21:06.43 | smackus | what does a blue/red alarm mean with a sangoma t1 card? |
| 21:06.43 | charles___ | hmmhesays: what bad advice ? I'm telling you to get the stuff that is stable and will work |
| 21:06.44 | hmmhesays | this is not bleeding edge |
| 21:06.44 | hmmhesays | this is a patch out of bristuff |
| 21:06.44 | *** part/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 21:06.44 | charles___ | smackus: it's carnival time. |
| 21:06.44 | charles___ | smackus: bling bling |
| 21:06.44 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 21:06.44 | *** join/#asterisk rob0 (n=rob0@cardinal.lizella.net) |
| 21:06.45 | hmmhesays | so getting no.. it won't work |
| 21:06.45 | smackus | what? |
| 21:07.02 | charles___ | hmmhesays: if you want to deal with non release you gotta check the headers or wait until tomorow maybe someone will commit the SVN fixing it |
| 21:07.11 | hmmhesays | heh |
| 21:07.19 | charles___ | hmmhesays: may be a bad prototype for the funcion or a bad call to the function |
| 21:09.05 | hmmhesays | heh again |
| 21:09.12 | charles___ | smackus: I haven't used sangoma cards, but blue alarm is crc's if I don't mistake, and red is when it's dead |
| 21:09.26 | russellb | hmmhesays: it could be a problem due to having trunk headers on the system, but building for 1.2 |
| 21:10.08 | hmmhesays | or ast_state_device.... has changed since 1.2.7 |
| 21:10.52 | russellb | there has only been one API change in the 1.2 series that I know of, and that isn't it |
| 21:11.46 | hmmhesays | nope, my bad, patch error forgot to patch devicestate.c |
| 21:12.01 | russellb | lame! |
| 21:12.33 | hmmhesays | whwhaha |
| 21:12.38 | russellb | oh yeah?! |
| 21:12.41 | russellb | /kick hmmhesays take that! |
| 21:12.45 | hmmhesays | lol |
| 21:13.01 | hmmhesays | there will be a coup if you kick the loveable fuzzy hmmhesays |
| 21:14.32 | *** join/#asterisk jmang (n=not@24.79.192.187) |
| 21:14.36 | jmang | Hello, |
| 21:14.46 | jhiver | aaaah guys compiling openh323 is slow and boring |
| 21:14.53 | hmmhesays | yes |
| 21:14.53 | GerbilWrk | anyone know how to get userinput on a call and turn it into a variable? |
| 21:15.32 | hmmhesays | read |
| 21:15.35 | hmmhesays | cmd read |
| 21:16.42 | jhiver | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read |
| 21:17.54 | Bullseye_Network | So what is the easyest way to get the channel an agent is bridged with...(Reliabily) |
| 21:18.42 | GerbilWrk | hmmhesays, thanks |
| 21:21.17 | hmmhesays | np |
| 21:21.30 | hmmhesays | now show em them t1tt1es |
| 21:21.48 | Bullseye_Network | (.)(.) |
| 21:21.53 | x86 | hah |
| 21:22.20 | hmmhesays | ahh nothing like the ballad of chasey lain to brighten a day |
| 21:25.43 | *** join/#asterisk queuetwo (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 21:27.18 | *** part/#asterisk iq (n=IQ@unaffiliated/iq) |
| 21:27.20 | Alric | Has anyone tried hooking asterisk up to a Metaswitch via SIP using G.726? |
| 21:28.13 | queuetwo | Hi. I've got a new asterisk installation connected to a VoicePulse IAX2 account. I can dial the number and watch asterisk answer, and the console claims to be playing "welcome.gsm", which does exist. Over the phone, i don' |
| 21:28.26 | queuetwo | t shear anything but dead air, though. |
| 21:28.27 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 21:28.39 | queuetwo | Where would I start to debug this? |
| 21:29.03 | fourcheeze | if I have box A which has asterisk and box B that doesn't and a sip call comes into box B could I do something clever with iptables to shove anything over to my asterisk on box A? |
| 21:29.33 | fourcheeze | IE redirect all packets on box B's IP number to box A? |
| 21:29.38 | fourcheeze | but do it at the IP level? |
| 21:29.54 | Nugget | just set up a SRV record in DNS for box b's hostname/ |
| 21:30.04 | Nugget | have it direct SIP calls to box a |
| 21:30.19 | TrixVox | queuetwo: try something else like playing dtmf and then hanging up |
| 21:30.26 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.244) |
| 21:30.26 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 21:30.36 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 21:30.46 | fourcheeze | Nugget: this is just a quick and dirty thing |
| 21:30.55 | fourcheeze | until I can contact our incoming provider |
| 21:30.57 | Nugget | why not a quick and clean thing? |
| 21:31.00 | queuetwo | TrixVox, If I have a person pick up, I can hear them fine. (Is DTMF test still worth it?) |
| 21:31.10 | hads | telnet |
| 21:31.10 | Nugget | telnet is eeeeeeevil! |
| 21:31.18 | TrixVox | oh, ok, then it's not a voicepulse problem... |
| 21:31.21 | fourcheeze | Nugget: mainly because incoming stuff from our provider comes to an IP number |
| 21:31.27 | fourcheeze | not a hostname |
| 21:31.41 | fourcheeze | just trying to bridge a gap while they wake up an engineer |
| 21:32.01 | TrixVox | hmm, must be something about playing the file... try a different file? |
| 21:32.17 | TrixVox | or something longer than one word... |
| 21:32.17 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 21:32.19 | vader-- | hola |
| 21:32.35 | fourcheeze | Nugget: would it work in principle? |
| 21:32.52 | Nugget | not if the other end is calling by IP, no. |
| 21:32.53 | vader-- | question you can send SIP/EXTEN1&SIP/EXTEN2 into the dial command can you also do the same thing for voicemail? |
| 21:33.03 | vader-- | can you dial one extension and leave voicemail in two boxes? |
| 21:33.13 | Alex | Hello everyone - I've got my extensions.conf "properly" set up, as per the Asterisk O'Reilly book, yet when I try and make an outbound call, it just dies straight awa\y, doesn't pick up a channel or anything? |
| 21:33.31 | mountainm2k | OK, so I have extension 699 going to VoiceMailMain() -- any way it can look at the Caller-ID and take that if you just hit # for the mailbox? |
| 21:33.53 | nortex | vader--, If I remeber correctly the first box to answer will get the message, not both. |
| 21:34.43 | *** join/#asterisk frenzy (n=frenzy@196.46.104.216) |
| 21:35.05 | frenzy | How do I increase MOH volume? |
| 21:35.51 | mountainm2k | frenzy: depends -- what mode is your MOH? |
| 21:35.53 | queuetwo | TrixVox, I changed from welcome to you-dialed-wrong-number, same problem... |
| 21:36.00 | frenzy | files |
| 21:36.06 | frenzy | I tried loud |
| 21:36.12 | frenzy | completely stopped working |
| 21:36.28 | mountainm2k | If you're using "files", the best thing would be to use "sox" to change the volume of the files |
| 21:36.51 | DrkShdw | anyone here good with "expect"? |
| 21:36.51 | *** join/#asterisk slima (i=slima@syriusz.slnet.uu3.net) |
| 21:36.58 | frenzy | how do I do that? (command please) |
| 21:37.02 | frenzy | I know sox is installed |
| 21:37.07 | x86 | frenzy: man sox |
| 21:37.22 | mountainm2k | sox -v (somenumber) <inputfile> <outputfile> |
| 21:37.24 | queuetwo | If I change it to XXXXXXXXX, I get File XXXXXXXXX does not exist in any format, so it must be finding the files... |
| 21:37.37 | mountainm2k | where (somenumber) is larger than 1.0 to increase the volume, less than 1.0 to decrease |
| 21:37.45 | mountainm2k | and make sure you don't overwrite -- |
| 21:38.00 | frenzy | :) |
| 21:38.01 | frenzy | thanks |
| 21:38.19 | mountainm2k | I suggest copying your files elsewhere, and then use those as <inputfile>, making <outputfile> be the correct location... |
| 21:38.53 | mountainm2k | <PROTECTED> |
| 21:39.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:39.21 | mountainm2k | <PROTECTED> |
| 21:39.28 | mountainm2k | sox won't work with mp3's |
| 21:39.36 | supjigatr | Anyone know how to get ISDN to using unknown dialing plan? I am hitting a DMS100 and it expects 10 digits since the * boxs aways sends national type. |
| 21:39.37 | Qwell[] | mountainm2k: sure it will |
| 21:39.54 | frenzy | i'm using freepbx if I upload .wav it gets turned in .mp3 |
| 21:40.39 | mountainm2k | <PROTECTED> |
| 21:40.52 | mountainm2k | perhapps my version of sox is outdated or something... |
| 21:41.21 | queuetwo | Hi. I've got a new asterisk installation connected to a VoicePulse IAX2 account. I can dial the number and watch asterisk answer, and the console claims to be playing "welcome", which does exist in gsm format. Over the phone, I don't hear anything but dead air, though. When I set the dial plan to ring an actual extension, the person and I can speak fine. Where would I start to debug this? |
| 21:42.19 | x86 | queuetwo: if you switch to my service I'll fully support you :) |
| 21:42.32 | x86 | queuetwo: http://www.shellshark.net/voip/ |
| 21:43.00 | queuetwo | Anyone else? |
| 21:43.15 | TrixVox | It has nothing to do with his service... it's his box. Someone should know how to debug problems playing gsm files (I don't know) |
| 21:43.25 | TrixVox | Try playing a .wav? |
| 21:43.26 | x86 | i do know |
| 21:43.39 | x86 | and i'll gaurantee it to work if he uses my service |
| 21:43.50 | TrixVox | Yeah, but you're a nobody :) |
| 21:43.50 | x86 | i'll set it up for him and fully debug |
| 21:43.56 | x86 | TrixVox: eh? |
| 21:44.12 | mountainm2k | TrixVox: paste your dialplan |
| 21:44.24 | mountainm2k | particularly the incoming context, extension 's' |
| 21:44.27 | mountainm2k | ~pastebin |
| 21:44.29 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts |
| 21:44.34 | queuetwo | x86, You have just guaranteed that many people will never try "your service." Stop spamming, and don't try to win business by concealing information from other members of the channel. |
| 21:44.55 | queuetwo | If you don't want to help, then just keep quiet. |
| 21:45.06 | mountainm2k | amen |
| 21:45.11 | x86 | i do want to help |
| 21:45.15 | TrixVox | And try dishing out the money for a real, working SSL certificate. |
| 21:45.28 | x86 | but i dont provide free support |
| 21:46.06 | mountainm2k | x86: then don't offer your paid support |
| 21:46.11 | queuetwo | x86, The rest of us do. you are in the wrong place. |
| 21:46.19 | x86 | i didnt know that was against the AUP here |
| 21:46.41 | TrixVox | queuetwo: I think mountainm2k was talking to you -- post your extensions.conf to pastebin... |
| 21:46.47 | x86 | queuetwo: how are you magically able to speak on behalf of the populous of the whole channel? |
| 21:46.52 | queuetwo | Ah, sorry. Missed that. |
| 21:47.01 | TrixVox | I don't see anyone disagreeing with him... |
| 21:47.29 | mountainm2k | queuetwo: Sorry, copied the name incorrectly... |
| 21:47.30 | TrixVox | You're trying to sell voip service to a room full of voip service providers haha |
| 21:47.37 | supjigatr | setup_zap: Ignoring prilocaldialplan |
| 21:47.49 | supjigatr | What does that mean exactly? |
| 21:48.18 | mountainm2k | queuetwo: I was going to check, but you can -- make sure the application you're playing the file from, such as Playback(), does _NOT_ include the filename extension... |
| 21:48.46 | mountainm2k | queuetwo: In other words, make sure if your greeting is themaingreeting.wav, you should say Playback(themaingreeting) |
| 21:48.58 | mountainm2k | queuetwo: and * will figure out which file to play |
| 21:49.06 | mountainm2k | queuetwo: (if there are more than one available) |
| 21:50.34 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.149) |
| 21:50.36 | edorsey | x86 this is the wrong place to get people buy a service from you goto the dumb company execs not asterisk/freepbx users... |
| 21:50.56 | GerbilWrk | hmmhesays, know of anyway to pass a veriable to another context? |
| 21:51.05 | x86 | edorsey: no one asked you :) |
| 21:51.06 | queuetwo | mountainm2k, pastebin coming up - had to strip out API keys and such. Now pastebin is slow as molasses. |
| 21:51.21 | edorsey | no one asked you to offer paid support |
| 21:51.32 | hads | queuetwo: use pastebin.ca - it's usually faster |
| 21:51.40 | x86 | edorsey: he asked for support, he didnt specify paid or not |
| 21:51.43 | TrixVox | edorsey: He's using the nufone marketing plan -- 1. insult the users, 2. ???, 3. Go broke |
| 21:51.43 | mountainm2k | queuetwo: You could just pastebin that particular section -- or read above, only one thing I wanted to look at |
| 21:52.00 | x86 | TrixVox: who did i insult? |
| 21:52.11 | mountainm2k | x86: ME |
| 21:52.17 | queuetwo | mountainm2k, http://pastebin.ca/114062 |
| 21:52.25 | x86 | mountainm2k: how? |
| 21:52.28 | edorsey | your no better than the bots that send me viagra spam |
| 21:53.16 | mountainm2k | <PROTECTED> |
| 21:53.20 | *** join/#asterisk enjay- (n=enjay@209.181.130.105) |
| 21:53.45 | enjay- | whats the best way to troubleshoot echo from internal to internal spa942->spa942 its intermittent. |
| 21:53.50 | queuetwo | x86, Want to drum up business? Then pick a decent nickname (one people will search for on the internet) and then prove your stuff by helping everyone you can. people will respect you and want to buy things form you. What you are doing now is just causing agitation. |
| 21:54.34 | edorsey | personally i prefer the viagra spam |
| 21:54.40 | x86 | queuetwo: i'm not asking for business advice :) |
| 21:54.58 | queuetwo | mountainm2k, Mine are in /usr/share/asterisk/sounds/welcome ... /usr/share/asterisk/sounds/welcome.gsm |
| 21:55.00 | edorsey | you should be, trying to sell your services in IRC channels... |
| 21:55.06 | TrixVox | haha |
| 21:55.14 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:55.14 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 21:55.18 | eKo1 | might as well sell viagra |
| 21:55.39 | TrixVox | You mean chat for 8 hours, make 1 sale isn't a scalable marketing model? |
| 21:56.16 | mountainm2k | queuetwo: Interesting... The console says that file doesn't exist? Can you try specifying the full path to it? Can you substitute other files in its place? |
| 21:56.44 | edorsey | Well....no |
| 21:56.50 | queuetwo | mountainm2k, No, console doe snot say file does not exist. When I use a "bad" file, it does report as missing. (Just a test I ran.) |
| 21:57.03 | *** join/#asterisk tessier_ (n=treed@gw.drjays.com) |
| 21:57.16 | mountainm2k | queuetwo: OK, so it finds it, just won't play it... |
| 21:57.29 | TrixVox | Is it because he's trying to play a gsm file over a g711 connection? Or is that not a problem? |
| 21:57.32 | queuetwo | mountainm2k, I think it is playing it, I just can't hear it. |
| 21:57.55 | edorsey | i think you meant: chat for 8 hours, make everyone think your stupid, lose reputation, maybe make a sale |
| 21:58.07 | queuetwo | TrixVox, Hrm... That could be, although I'm using VP's stock config for that... |
| 21:58.12 | Qwell[] | edorsey: and probably lose a customer or two |
| 21:58.40 | edorsey | yeah that too |
| 21:58.47 | queuetwo | A good example would be some of the guys selling Rails Hosting. They field questions in the #rubyonrails room, earn people's trust, and people want to host and do jobs with them. |
| 21:58.49 | x86 | edorsey: i think you meant: "you're" |
| 21:59.11 | edorsey | and i think you meant to capitalize your "I" but you didnt who cares. |
| 22:00.22 | edorsey | now you are critiquing me on my grammar? what run out of other things to make fun of me for? |
| 22:00.42 | queuetwo | Is there a problem paying gsm over specific connection/protocol types? |
| 22:01.02 | mountainm2k | queuetwo: Sorry, I stepped away |
| 22:01.15 | TrixVox | queuetwo: Is the call even staying up long enough for you to hear it? Does Background() keep the call up? |
| 22:02.01 | queuetwo | TrixVox, I think so. The console seems to know when I hang up. Should I be using something other than Background()? |
| 22:02.13 | queuetwo | I thought that was how to handle an IVR menu. |
| 22:02.23 | TrixVox | Or do you need s,1,Answer(), then s,2,Background(...) |
| 22:03.20 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 22:03.38 | TrixVox | John Todd, of loligo.com advises: |
| 22:03.38 | TrixVox | <PROTECTED> |
| 22:04.04 | mountainm2k | ahhha, yeah, queuetwo isn't doing that |
| 22:04.07 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 22:04.09 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:04.15 | mountainm2k | Wonder if that.s it... |
| 22:04.32 | queuetwo | Nope. |
| 22:04.40 | queuetwo | Added Answer and Wait(2) - no difference. |
| 22:04.49 | TrixVox | did you reload? |
| 22:05.00 | queuetwo | Yes. |
| 22:05.12 | queuetwo | actual format = ulaw, ... |
| 22:05.48 | TrixVox | Playback() instead of Background() ? |
| 22:05.53 | queuetwo | I'll try. |
| 22:06.06 | dioedu | there is a way to use asterisk like a bridge (or transit, I don't know how is called out of Brazil) to other pbx ? |
| 22:06.38 | queuetwo | No difference. |
| 22:06.52 | *** join/#asterisk dasenjo_ (n=dasenjo@208.195.215.229) |
| 22:06.53 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 22:08.55 | *** part/#asterisk kfudge (n=kzymxa@24-217-137-21.dhcp.stls.mo.charter.com) |
| 22:09.48 | *** join/#asterisk daysmen3 (n=primus@host86-139-117-242.range86-139.btcentralplus.com) |
| 22:10.01 | queuetwo | Any other ideas? |
| 22:13.46 | mountainm2k | queuetwo: I'm out... My thought was the file extension, but I was incorrect... :-p |
| 22:13.49 | queuetwo | actual format = gsm makes no difference... |
| 22:15.17 | Hughes | Is there a much of new sip client caching or something that wasn't present in the near 1.0 days (the last time I had asterisk on the bench)? I'm going nuts not being able to register sip clients after making topology changes. |
| 22:16.04 | Hughes | It's like the clients try to register but asterisk just throws the data away. no debug, no nothing. |
| 22:17.05 | Hughes | It all seemed to "just work" last time. |
| 22:23.38 | queuetwo | If I use MP3Player, I can hear it just fine.... |
| 22:23.54 | queuetwo | (I can hear the mp3 I select, that is.) |
| 22:24.17 | queuetwo | WTH is wrong with Background / Playback? |
| 22:26.31 | queuetwo | Playtones works as well... |
| 22:26.36 | generalhan | can some one explain to me how to use ztmonitor? i keep getting a command not found when i try to run it... |
| 22:26.43 | generalhan | i need to get this echo prob fixed ASAP |
| 22:27.44 | *** part/#asterisk dioedu (n=dioedu@200.207.150.85) |
| 22:29.18 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:29.18 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 22:29.27 | *** join/#asterisk tempest1 (n=asf@adsl-150-206-88.chs.bellsouth.net) |
| 22:29.33 | syzygyBSD | generalhan: are you root? |
| 22:30.52 | syzygyBSD | queuetwo: well, I dont' know if this has anything to do with your problem, but MP3 isn't a free format, so support isn't built into many GPL items, unless there is another library already on the system |
| 22:31.20 | syzygyBSD | ie sox doesn't support MP3 unless you had lame on your computer when you compiled |
| 22:31.40 | queuetwo | syzygyBSD, Mp3Player a) was the format that worked and b) Uses mpg123 externally to stream the sound. |
| 22:32.18 | queuetwo | syzygyBSD, I'm having aproblem with not hearing .gsm files playing. The mp3 thing was a test, and it worked. |
| 22:32.28 | syzygyBSD | ya, I have problems with mpg123 before, but that could have been sox issues, I don't know |
| 22:32.50 | syzygyBSD | hmm, gsm has always worked for me |
| 22:32.57 | queuetwo | I don't have nay problems with mp3s. I have a problem with gsm. mp3 works fine. :) |
| 22:33.00 | syzygyBSD | are you trying to play the default sounds? or some you made |
| 22:33.06 | queuetwo | Defaults. |
| 22:33.12 | queuetwo | Playback(welcome) |
| 22:33.31 | syzygyBSD | huh, never had problems with them |
| 22:36.40 | *** join/#asterisk dbradley (n=dbradley@tangram.gotadsl.co.uk) |
| 22:37.46 | queuetwo | Is there some additional package required for the gsm codecs? (Ubuntu Dapper) |
| 22:37.48 | *** join/#asterisk mcolgin (n=g0dsmite@dsl254-018-172.sea1.dsl.speakeasy.net) |
| 22:37.57 | dbradley | hey folks, anyone built Asterisk for *BOTH* BRI and PRI ISDN cards using the same binaries? I thought I'd seen somewhere that the bristuff patches affected PRI from working correctly |
| 22:40.01 | generalhan | syzygyBSD: yea im root |
| 22:40.50 | syzygyBSD | locate ztmonitor |
| 22:41.37 | syzygyBSD | how did you become root? |
| 22:41.43 | syzygyBSD | su? |
| 22:42.37 | *** join/#asterisk Assid (i=assid@dialup-mum-203.94.221.122.mtnl.net.in) |
| 22:43.55 | sevard | Question, My callerid on my incoming fxo lines on my TDM400P is sending callerid as Incoming/Asterisk even though I have my callerid=Incoming usecallerid=no in my zapata.conf under that group |
| 22:44.08 | sevard | I have no callerid on these zap lines |
| 22:46.54 | *** join/#asterisk watchy2 (i=wiit@h140.184.255.206.cable.cmdn.cablelynx.com) |
| 22:47.06 | watchy2 | is there anything i can run on a tdm400p to check if a fxo port is bad? |
| 22:47.49 | sevard | watchy2: you can run fxowatch or fxotune -i 4 |
| 22:48.15 | watchy2 | ah |
| 22:48.37 | watchy2 | well i got 4 lines plugged in but i think we have an issue with it but im not sure |
| 22:49.08 | watchy2 | i know the 4 lines tested good today but i disabled one yesterday because it was wigging out |
| 22:49.15 | watchy2 | but i dunno if it was the line or the fxo |
| 22:49.35 | watchy2 | cant really enable it after checking the phone lines either without restarting * |
| 22:53.27 | queuetwo | sevard, Where would I find fxowatch? |
| 22:55.16 | sevard | queuetwo: in your zaptel source directory |
| 22:55.45 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.229) |
| 22:56.04 | generalhan | syzygyBSD: no i AM root ! lol |
| 22:56.11 | generalhan | i dont ever su, though i know i should ! |
| 22:56.11 | queuetwo | sevard, I see fxotune there, but no fxowatch... Is it not in stable? |
| 22:58.54 | queuetwo | No one knows why .gsm files would play silently over IAX2? |
| 22:59.16 | charles___ | queuetwo: IAX only or SIP too ? |
| 23:00.10 | charles___ | queuetwo: using stable latest ? |
| 23:00.22 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:00.22 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 23:02.38 | *** join/#asterisk tempest1 (n=asf@adsl-230-225-130.chs.bellsouth.net) |
| 23:06.27 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 23:07.51 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 23:07.59 | ManxPower | Can anyone here recommend cheap rack mount systems? They may netboot, may not have an HD, need sound and RCA or S-Video out. |
| 23:08.54 | ManxPower | It will basically play video to an RF Modulator. |
| 23:10.52 | *** join/#asterisk Djeli (n=djeli@58.171.89.237) |
| 23:12.05 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 23:12.09 | *** part/#asterisk Djeli (n=djeli@58.171.89.237) |
| 23:13.02 | *** join/#asterisk evilbit (i=hhoffman@gateway/tor/x-381ac03d06c1717f) |
| 23:13.10 | evilbit | hi, anyone using teliax? |
| 23:14.30 | *** join/#asterisk lorinc (n=ang@caracas-3871.adsl.interware.hu) |
| 23:14.52 | *** join/#asterisk aiksa (n=aiks@83.223.131.104) |
| 23:14.53 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-149-117.dsl.pltn13.sbcglobal.net) |
| 23:14.57 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 23:15.00 | aiksa | Hi everyone |
| 23:15.27 | aiksa | sorry for newb question, but I have a problem with grandstream budgetone |
| 23:15.42 | darkgamer20 | what the best place to buy a X100P card? |
| 23:15.50 | aiksa | as it happens on all of the phones - i believe that most of you have delt with this |
| 23:15.51 | evilbit | ebay? |
| 23:16.00 | ManxPower | darkgamer20, X100P cards have not been sold in at least a year -- only clones |
| 23:16.14 | Hughes | Looks like it was iptables. |
| 23:16.42 | aiksa | user A calls user B (both on budgetones) if user B drops the receiver first; the phone hangs up for user A |
| 23:16.47 | darkgamer20 | ManxPower: i know, but like where can i buy a clone that comes close to the quality of the original X100P |
| 23:16.52 | aiksa | anyone experienced it? |
| 23:17.12 | XARiUS | ah HA.. I think I finally nailed this bastard cisco issue |
| 23:17.15 | ManxPower | darkgamer20, I doubt you'll find any. |
| 23:17.42 | *** part/#asterisk angom_w (n=angom@red-corp-200.79.158.218.telnor.net) |
| 23:17.57 | darkgamer20 | ManxPower: you doubt i will find any of the original X100P? |
| 23:17.59 | *** join/#asterisk sumasuma (n=sumase@cm222.omega183.maxonline.com.sg) |
| 23:18.25 | sumasuma | i was trying to integrate cepstral into asterisk, facing compilation problems |
| 23:18.27 | aiksa | at the moment of the drop of receiver the date is displayed as if line has been hanguped, but as soon A puts the receiver down BT starts counting second as if it had active channel |
| 23:18.27 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 23:18.36 | sumasuma | anybody here to help me please |
| 23:18.48 | ManxPower | darkgamer20, I doubt you'll find anything cheap that comes close to the X100P. |
| 23:18.58 | sumasuma | i getting this error, /usr/include/asterisk/file.h:27:2: error: #error You must include stdio.h before file.h! |
| 23:19.08 | evilbit | sumasuma: post the errors somewhere |
| 23:19.22 | file | sumasuma: what are you trying to build? |
| 23:19.27 | ManxPower | sumasuma, I'll get you don't have the kernel dev source installed |
| 23:19.47 | evilbit | gcc-devel? |
| 23:19.53 | sumasuma | file, i want to include sepstral into my asterisk |
| 23:19.59 | aiksa | anyone could help me with that problem, please? |
| 23:20.15 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 23:20.17 | file | that's why, the module is old and needs to be modified to... include stdio.h before file.h |
| 23:20.18 | darkgamer20 | ManxPower: hmm with a clone card, will i still have a good pbx? |
| 23:20.35 | darkgamer20 | ManxPower: im planning on just a home system |
| 23:20.40 | ManxPower | darkgamer20, Even with the X100P you will not have a good PBX. |
| 23:21.01 | aiksa | i had searched the google as far as I could yet I couldnt find the source of the problem |
| 23:21.03 | sumasuma | darkgamer20: there is a card I bought, professional card, which is very good compared to clones from ebay |
| 23:21.34 | darkgamer20 | sumasuma: what was it? |
| 23:21.43 | sumasuma | ManxPower: kernel-dev will get it resolved? |
| 23:22.02 | sumasuma | darkgamer20: check it out in ebay, professional x100p |
| 23:22.11 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 23:22.13 | russellb | professional x100p? wtf? |
| 23:22.20 | JT | haha yeah |
| 23:22.26 | russellb | don't let marketing crap like that make you think it's any different |
| 23:22.37 | darkgamer20 | hey sumasuma are you joking around |
| 23:22.42 | darkgamer20 | ? |
| 23:23.06 | sumasuma | darkgamer20: it was better than the clones, I really experienced it |
| 23:23.13 | darkgamer20 | oh |
| 23:23.15 | Qwell[] | oh...man... |
| 23:23.18 | hads | Haha, there are 'professional' ones listed on ebay. |
| 23:23.22 | darkgamer20 | yea |
| 23:23.25 | Qwell[] | That's awesome |
| 23:23.30 | Qwell[] | suckers |
| 23:23.31 | file | right... |
| 23:23.35 | darkgamer20 | hey guys whats the difference? |
| 23:23.41 | Qwell[] | darkgamer20: about $40 I bet |
| 23:23.45 | Qwell[] | ...nothing else |
| 23:23.48 | russellb | $40 for an extra word |
| 23:23.48 | darkgamer20 | oh |
| 23:24.01 | sumasuma | darkgamer20: yes, $40 i paid |
| 23:24.20 | darkgamer20 | yea guys it actually says asterisk logo on the card |
| 23:24.28 | sumasuma | darkgamer20: I bought clones before, it got stucked me before, this was working fine |
| 23:24.30 | Qwell[] | darkgamer20: uhh...no |
| 23:24.34 | darkgamer20 | that means its the original right? |
| 23:24.36 | russellb | darkgamer20: then that is trademark infringement |
| 23:24.44 | darkgamer20 | oh |
| 23:24.54 | hads | http://i4.ebayimg.com/04/i/06/c8/b9/f1_1.JPG |
| 23:25.10 | darkgamer20 | yea |
| 23:25.15 | russellb | that's not on the card ....... |
| 23:25.25 | Qwell[] | haha, that's photoshopped |
| 23:25.30 | russellb | that's photoshop, and trademark infringement none the less |
| 23:25.34 | darkgamer20 | oh |
| 23:25.40 | ManxPower | darkgamer20, The comppany that makes that card should be sued by Digium. |
| 23:25.44 | darkgamer20 | oh |
| 23:25.50 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 23:25.53 | darkgamer20 | thats not cool to buy that huh? |
| 23:26.08 | ManxPower | No matter how hard you wish it was not true, there are no real digium X100P cards available any more. |
| 23:26.11 | hads | Haha "Asterisk , Digium, X100P, Wildcard may be the trademark of Digium Inc., Our Professional Card is no way affiliated to Digium." |
| 23:26.23 | darkgamer20 | lol |
| 23:26.35 | russellb | s/may be/is/ |
| 23:26.40 | ManxPower | You can flap your arms as hard as you want, but you will have a better chance of flying than finding a real, new digium X100P card on eBay. |
| 23:26.45 | file | ZOMBIE WANTS BRAINS |
| 23:27.30 | aiksa | so no ideas as regards that budgetone bug? |
| 23:27.33 | ManxPower | file, Digium really has to start doing a smackdown on eBay X100Ps |
| 23:27.37 | JT | did digium actually use a from scratch design for the X100P, or is it just a particular winmodem rebranded/customised? |
| 23:27.46 | De_mon | considering they dont make X100P's any more |
| 23:27.57 | Juggie | its just a modem. |
| 23:27.59 | ManxPower | JT, they did not design X100P from scratch. |
| 23:28.04 | darkgamer20 | hmm ok i need a fxo card and i dont have the money for a tdm400p so what are my options? |
| 23:28.16 | ManxPower | darkgamer20, A SIP FXO box. |
| 23:28.17 | evilbit | so, I'm having this problem with teliax where the first time I call my DID it gets routed to my * box but subsequent attempt result in a telco errmsg that my call cannot be completed... any ideas? |
| 23:28.32 | ManxPower | I need a full cable tv headend, but I'm not going to get that cheap either. |
| 23:28.55 | sumasuma | file: Will kernel-dev will solve my problem ?? |
| 23:29.00 | JT | ManxPower: it was basically just a winmodem still? |
| 23:29.08 | ManxPower | JT, Correct. |
| 23:29.09 | file | sumasuma: no, the module has to be modified... to do exactly what the message says |
| 23:29.23 | JT | just of a particular quality |
| 23:29.26 | file | stdio.h must be included before file.h |
| 23:29.34 | darkgamer20 | hey what do you think about the digitnetworks x100p card? |
| 23:29.39 | sumasuma | oh |
| 23:29.46 | ManxPower | JT, But apparently only Mark was smart enough in the entire world to write a telcom driver for it. |
| 23:29.54 | Qwell[] | darkgamer20: It's all the same POS |
| 23:29.56 | JT | heh |
| 23:30.07 | darkgamer20 | damn man |
| 23:30.08 | ManxPower | darkgamer20, As I understand it the reason the X100P is no longer available is that THE PARTS WERE NOT BEING MADE ANY MORE. |
| 23:30.17 | darkgamer20 | oh |
| 23:30.30 | ManxPower | So anything else is NOT a real card and is NOT fully compatable and will require CUSTOM NON-DIGIUM DRIVERS. |
| 23:30.46 | sumasuma | ManxPower: I heard you can get upto millions of the same chipset in china |
| 23:31.02 | ManxPower | And you'll get bitchslapped if you want support for those cards. |
| 23:31.17 | darkgamer20 | lol |
| 23:31.29 | ManxPower | sumasuma, Good! How much will a million of them cost? |
| 23:31.42 | darkgamer20 | manxpower: can i get the sip fxo box? |
| 23:31.45 | Qwell[] | ManxPower: about 8 cents |
| 23:31.49 | aiksa | ok, looks like canreinvite=no solved the issue |
| 23:31.52 | ManxPower | The people selling "X100P cards" are small time people without the funds to actually have cards made. |
| 23:32.07 | ManxPower | Qwell[], 8 cents for a million of them! Wow! That's cheap! |
| 23:32.07 | aiksa | that could be the source of error? |
| 23:32.13 | sumasuma | Manxpower: I dummp sir, I'just discussed here the other day and a chinese man explained me |
| 23:32.35 | ManxPower | That is like 0.00000008 cent each |
| 23:32.44 | Qwell[] | ManxPower: sounds about right |
| 23:32.54 | darkgamer20 | damn' |
| 23:32.57 | darkgamer20 | wtf |
| 23:33.14 | darkgamer20 | that must be very bad POS cards |
| 23:33.33 | ManxPower | Anyway, I'll let darkgamer20 flap his arms all he wants, I'll sit back and point and laugh. |
| 23:34.08 | darkgamer20 | ManxPower: lol please don't guide me please |
| 23:34.23 | ManxPower | darkgamer20, you want a cheap FXO solution. There is none. |
| 23:34.32 | darkgamer20 | damn |
| 23:34.47 | JT | SIP ethernet based embedded box would probably be the cheapest |
| 23:34.54 | ManxPower | Now, if you worked in Telecom then a TDM400P card would be considered cheap. |
| 23:34.57 | darkgamer20 | JT whats that? |
| 23:35.11 | ManxPower | since you can expect to spend about $1,000 for a 4-port analog card for many PBXs |
| 23:35.35 | JT | darkgamer20: a hardware box |
| 23:35.45 | JT | iirc, a Sipura 3000 has an FXO port? |
| 23:36.11 | darkgamer20 | ManxPower: do you run asterisk at home? |
| 23:36.32 | ManxPower | darkgamer20, I did until Katrina hit and I moved to the top of a mountian |
| 23:37.12 | darkgamer20 | oh sorry about that, what did you have a PSTN or a voip line? |
| 23:37.46 | ManxPower | both. |
| 23:37.59 | *** part/#asterisk slima (i=slima@syriusz.slnet.uu3.net) |
| 23:38.11 | darkgamer20 | how did you handle the calls from the PSTN line? |
| 23:38.24 | ManxPower | I had PSTN for local calls, VoIP to an asterisk server that is a local call to my boyfriend, and a VoIP telco for the ocassionaly long distance and for local numbers to where my family lives. |
| 23:38.42 | ManxPower | darkgamer20, I had a REAL X100P for a while, then a TDM400P |
| 23:39.01 | ManxPower | TDM400P w/1xFXO and 2xFXS |
| 23:39.38 | ManxPower | You do realize that a TDM400P w/1xFXS costs about as much as many people spend on their cabletv bill each month, right? |
| 23:39.50 | darkgamer20 | yea i know that |
| 23:40.16 | *** join/#asterisk tempest1 (n=asf@adsl-230-225-130.chs.bellsouth.net) |
| 23:40.22 | ManxPower | somegeek, cancel your cabletv service and start saving. |
| 23:40.27 | TrixVox | evilbit: I used to have problems like that with Teliax... it's on their side, but they're non-responsive to requests to fix it. They always said it way my problem, but everything's been rock-solid since I switched to VoicePulse Connect. Not sure what other help to provide, if it's not hitting your switch on subsequent calls, it can't be your box. Might want to double-check your firewall... |
| 23:41.08 | darkgamer20 | ManxPower: how dose the TDM400P compare with the X100P? is it about the same? |
| 23:42.05 | TrixVox | brb, vpn time |
| 23:42.24 | *** join/#asterisk juice (i=1000@mo-71-50-31-176.dhcp.embarqhsd.net) |
| 23:46.11 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 23:46.37 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:47.34 | evilbit | TrixVox: thanks for the heads up. Hopefully I can transfer my number, but probably not :-( |
| 23:50.14 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 23:53.14 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 23:53.32 | XARiUS | can someone help me straighten my brain out please.. :) |
| 23:53.50 | XARiUS | I've been staring at the same problem for too long |
| 23:53.59 | Nivex | there, nice and straight |
| 23:54.02 | Nivex | :) |
| 23:54.03 | XARiUS | lol.. spanks. |
| 23:54.34 | XARiUS | has queue logic changed recently? |
| 23:54.39 | XARiUS | (as in application queue) |
| 23:55.11 | Bullseye_Network | dont think so |
| 23:55.49 | XARiUS | I can't get this queue to stop ringing and fall back to the dialplan and it's pissin me off |
