| 00:01.01 | [TK]D-Fender | echinos: Well I just told you the exact parameter to look for... |
| 00:01.20 | echinos | yeah, I know, so I'm going to go fix it |
| 00:01.20 | drmessano | uh oh |
| 00:01.26 | echinos | tyvm |
| 00:01.35 | [TK]D-Fender | NEXT@!!!@ (c) BKW |
| 00:02.17 | drmessano | "Hmm.. You mentioned SIP. I think I have one of those in my asterisk. Let me go look" |
| 00:02.48 | echinos | :P |
| 00:04.12 | [TK]D-Fender | drmessano: You didn't get the memo? SIP has been officially deprecated in favor of GULP (God-damn UDP Link Protocol) |
| 00:04.21 | drmessano | lol |
| 00:05.04 | [TK]D-Fender | drmessano: It works over the internets now. Both of them. |
| 00:05.27 | drmessano | Just reminds of those overly pretentious artsy douchebags my sister would bring home when we all lived at home. "Oh, VoIP? I think my skype at home has one of those in it" |
| 00:05.47 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-aa2366943e37398d) |
| 00:10.53 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 00:13.35 | drmessano | Dangit |
| 00:13.54 | drmessano | None of these gigabit switches are SIP _OR_ IPV6 compatible |
| 00:13.58 | drmessano | WTF.. |
| 00:16.12 | thedonvaughn | a switch is layer 2. SIP is a high level protocol, the switch shouldn't care. What isn't SIP compatible about it? No QoS? |
| 00:18.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:19.27 | drmessano | "joke" |
| 00:20.17 | drmessano | Nevermind.. I'll be back for "Asterisk after Dark" |
| 00:20.31 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-209-183.phlapa.east.verizon.net) |
| 00:20.34 | thedonvaughn | ah :) |
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| 01:17.18 | ruben23 | hi where am i going to view the call logs of my asterisk server.. |
| 01:17.41 | Akiyuki | try /var/log/asterisk/ |
| 01:18.29 | ruben23 | on cdr-csv |
| 01:18.40 | ruben23 | is it on cdr-csv |
| 01:19.46 | [TK]D-Fender | ruben23: By default, yes |
| 01:21.22 | ruben23 | how about the history calls not the realtime calls... |
| 01:22.12 | [TK]D-Fender | ruben23: huh? |
| 01:22.14 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:22.27 | joat | it appears that set(CALLERID(all)=bob <444>)) doesn't work when connecting to a local (as in Local/456) extension.... (actually, I'm originating the call via the AMI) Any ideas how to set it? |
| 01:23.10 | ruben23 | what i mean is can i also view my old log calls not only the current calls... |
| 01:23.37 | *** join/#asterisk nfi|ermes (i=ErMeS@host74-6-dynamic.8-79-r.retail.telecomitalia.it) |
| 01:24.14 | ruben23 | ok i got it...thanks.. |
| 01:24.18 | [TK]D-Fender | ruben23: what is a "current" call? |
| 01:24.32 | [TK]D-Fender | ruben23: that log shows completed calls |
| 01:24.43 | [TK]D-Fender | ruben23: You need to look at CLI for calls in progress |
| 01:24.49 | ruben23 | yes...i got it |
| 01:24.49 | [TK]D-Fender | joat: Show us |
| 01:25.14 | ruben23 | i have one more problem... |
| 01:26.16 | *** join/#asterisk chendy (n=chatzill@219.134.30.97) |
| 01:28.07 | joat | in my extension I have "exten => 191,1,Set(CALLERIDNAME=MOH)" |
| 01:28.28 | Akiyuki | hmm |
| 01:28.40 | Akiyuki | The VoIP provider that we have right now doesnt do recording. :/ |
| 01:28.41 | joat | when I use AMI to originate a call from that extension to a conference room (or vice versa) the name remains blank |
| 01:28.57 | Akiyuki | I have to tcpdump everything to record the calls |
| 01:29.04 | joat | in other words, CallerID appears to not work for Local/191 to Local/201 |
| 01:29.15 | [TK]D-Fender | joat: Don't show 1 single line like that dump EVERYTHING include CLI output and an attemt to verify it in there as well |
| 01:29.55 | [TK]D-Fender | Akiyuki: And the reason you aren't having * do this is...? |
| 01:30.01 | joat | I'm not using CLI, I'm using AMI |
| 01:30.08 | ruben23 | i got a security issue on my asterisk boxes...im using asterisk 1.4, SIP+sofphones & VOIP...i registered numbers of user on my sip.conf able to do outbound calls-problem is even not on calling time user can able to dial & used my voip..are there ways to restrict them by asterisk configuration on my boxes...in a set of time...ideas only...if possible.. |
| 01:30.12 | [TK]D-Fender | joat: show the CALL IN CLI |
| 01:30.45 | Akiyuki | [TK]D-Fender, Our boss won't use an inhouse phone service |
| 01:31.07 | [TK]D-Fender | Akiyuki: huh? You're no using * at all? |
| 01:31.13 | Akiyuki | Yeah, but not at work |
| 01:31.30 | Akiyuki | Plus, all of our phones are MGCP only |
| 01:31.40 | [TK]D-Fender | Akiyuki: Ah yes, MGCP, I remember now... |
| 01:31.44 | Akiyuki | :) |
| 01:31.45 | [TK]D-Fender | Akiyuki: ... |
| 01:31.49 | [TK]D-Fender | ~wglwat |
| 01:31.50 | jbot | i heard wglwat is well, good luck with all that |
| 01:31.55 | Akiyuki | heh |
| 01:32.17 | Akiyuki | I been using it at home for the last month or so. Softphones only. Waiting until my next check to buy a hardpohne. |
| 01:33.54 | [TK]D-Fender | ruben23: that made no sense. |
| 01:34.04 | ruben23 | :-D |
| 01:34.28 | Akiyuki | ruben23, What is your first language? |
| 01:34.46 | [TK]D-Fender | Akiyuki: GIBBERISH :p |
| 01:34.51 | Akiyuki | rimshots |
| 01:35.12 | Akiyuki | Give it up for [TK]D-Fender everyone! He's gonna be here all week doing 2 shows a night! Don't forget to tip your bots! |
| 01:35.51 | [TK]D-Fender | bows at the drum-hit |
| 01:38.15 | joat | [TK]D-Fender: in one direction --> http://pastebin.com/d61b9d09b |
| 01:38.24 | joat | in the other --> http://pastebin.com/d77cef9db |
| 01:38.43 | joat | caller id fails to show up in status line of AMI |
| 01:39.01 | joat | externally generated calls, however, do show up |
| 01:39.14 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 01:39.31 | [TK]D-Fender | joat: what version of * are you using? |
| 01:39.35 | joat | 1.6 |
| 01:39.42 | [TK]D-Fender | joat: -- Executing [191@default:1] Set("Local/191@default-29dd;2", "CALLERIDNAME=MOH") in new stack |
| 01:39.55 | [TK]D-Fender | joat: Well that variable was deprecated in **1.2** |
| 01:40.06 | [TK]D-Fender | joat: Welcome to 4 years later |
| 01:40.34 | [TK]D-Fender | joat>it appears that set(CALLERID(all)=bob <444>)) doesn't work when connecting to a local (as in Local/456) extension.... (actually, I'm originating the call via the AMI) Any ideas how to set it? |
| 01:40.43 | [TK]D-Fender | joat: And you seem to have reason to have known better |
| 01:40.56 | joat | also tried it this way: Set(CALLERID(all)=Music on Hold <191>) |
| 01:41.16 | joat | still doesn't work |
| 01:41.27 | [TK]D-Fender | joat: Show me something that I SHOULD believe should work then, and output the CID as prrof in the call |
| 01:41.45 | JayTee52 | hands [TK]D-Fender the aspirin bottle |
| 01:42.44 | *** join/#asterisk JayTee52 (n=jforde@unaffiliated/jaytee) |
| 01:44.21 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 01:47.09 | joat | [TK]D-Fender: in one direction --> http://pastebin.com/m1b5c375e |
| 01:47.16 | joat | in the other --> http://pastebin.com/m348dab54 |
| 01:49.49 | [TK]D-Fender | joat: And where do I see any referncing to CID in there? |
| 01:50.03 | nfi|ermes | hi [TK]D-Fender |
| 01:50.16 | nfi|ermes | i m going crazy with the caller id problem |
| 01:50.36 | [TK]D-Fender | nfi|ermes: Go call OpenVox to look at their card |
| 01:52.05 | joat | line 4 of the first example, line 1 of the second |
| 01:52.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:53.11 | [TK]D-Fender | joat: Where do I see that it is NOT successfully set? |
| 01:54.01 | joat | in the AMI interface... externally generated calls show CID there... Local to Local do not even show the extension number from the dialplan |
| 01:54.26 | [TK]D-Fender | joat: I don't see you showing me that it ISN'T set |
| 01:56.04 | JonOnt | hey. where do I set the pin for the Flash Operator Panel? |
| 01:56.13 | joat | suggestion for showing that? core show channels doesn't show CID |
| 01:57.08 | jaytee | JonOnt, probably somewhere in the documentation here: http://www.asternic.org/ |
| 01:57.18 | [TK]D-Fender | joat: You have failed to show that it ISN'T set. |
| 01:57.39 | [TK]D-Fender | joat: joat Core show channels does not care about CID |
| 01:57.53 | [TK]D-Fender | joat: I also fail to see where CID even matter in what you're doing. |
| 01:58.26 | joat | i'm asking for a suggestion (function) to use to display what're you're asking... |
| 01:58.54 | [TK]D-Fender | joat: How can you TELL that it isn't set? |
| 01:58.56 | joat | displaying caller id for external calls (SIP or IAX2) is easy |
| 01:59.10 | [TK]D-Fender | joat: And where does anything in what its calling care about CID in the first place? |
| 01:59.12 | joat | because it isn't showing up in AMI (I've stated this previously) |
| 01:59.26 | [TK]D-Fender | joat: SHOW ME THE DUMP THEN |
| 02:00.06 | joat | forget it... you're a bit too difficult to deal with... |
| 02:00.09 | [TK]D-Fender | joat: And think on why I don't see your Originate either while you're at it |
| 02:00.38 | [TK]D-Fender | jaytee: I'll have that Aspirin now... |
| 02:02.49 | *** join/#asterisk jeffspeff2 (i=Administ@c-98-240-113-191.hsd1.ky.comcast.net) |
| 02:03.27 | jaytee | [TK]D-Fender, I'm thinking something stronger.....like whiskey with a valium chaser :-) |
| 02:03.51 | [TK]D-Fender | jaytee: I just want a ClueBat..... |
| 02:03.59 | jaytee | a real one |
| 02:04.13 | [TK]D-Fender | jaytee: Yup, a good 'ole Louiseville slugger |
| 02:04.39 | nfi|ermes | http://www.openvox.com.cn/bbs/viewtopic.php?p=2301#2301 |
| 02:04.55 | [TK]D-Fender | jaytee: Swear to God man these twits who couldn't actually show anyone the actual problem or evidence of any kind of failure whatsoever is really starting to piss me off. |
| 02:05.33 | jaytee | [TK]D-Fender, yeah, I gave up last night waiting for a pastebin of a sip.conf |
| 02:06.53 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 02:08.54 | [TK]D-Fender | jaytee: Been there, done that... and like the cake ..... there is no t-shirt |
| 02:09.18 | *** join/#asterisk agile (n=mike@24-182-108-29.dhcp.ftwo.tx.charter.com) |
| 02:10.05 | jaytee | we could be in the Bahamas right now with those drinks with the umbrellas sticking out of them. Somewhere I took a wrong turn in life :-( |
| 02:10.45 | JonOnt | Ok, any one familiar with aastra phones? |
| 02:10.47 | [TK]D-Fender | jaytee: was probably at Albuquerque |
| 02:11.06 | *** part/#asterisk bijit (n=chatzill@200.122.188.156) |
| 02:12.10 | jaytee | [TK]D-Fender, that's the "Land of Enchantment" for ya, always messing up someone's future. |
| 02:17.01 | [TK]D-Fender | jaytee: Due to financial constraints, the light at the end of the tunnel has been closed until further notice... |
| 02:18.38 | jaytee | [TK]D-Fender, I think I just saw about 20% of my 401K disappear today. I may have to buy a gun soon while it's still legal to own one. Crime may become the only way to make a living in this country in another 6 months. Seems to be the only way to make a living on Wall Street anyways. |
| 02:19.02 | [TK]D-Fender | jaytee: You mean... while you can still AFFORD one :) |
| 02:19.22 | jaytee | "Give me the pizza and the cheesy bread or the clown get it!!!!" |
| 02:19.34 | jaytee | damn, can't type for shit tonight |
| 02:26.29 | *** join/#asterisk jpastore (n=jpastore@173.9.147.165) |
| 02:29.58 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 02:54.27 | agile | looking for clues as to why doing a GET DATA over agi doesn't seem to receive the dtmf on a newly setup asterisk 1.4.17 (ubuntu hardy) |
| 02:55.10 | [TK]D-Fender | agile: Prove that it works in standard dialplan first |
| 02:55.32 | [TK]D-Fender | agile: And your OS is not responsible for this |
| 02:56.26 | agile | [TK]D-Fender, yes, I'm aware of that but I was just offering more detail as to the version, eg: the version packaged in hardy.. will try straight up in extensions.conf |
| 02:56.45 | jaytee | packages, ugh |
| 02:57.30 | joat | agile: what script |
| 02:57.48 | agile | joat, it's a custom deal |
| 02:58.19 | *** join/#asterisk etfonhomey_ (n=chatzill@74-143-196-254.static.insightbb.com) |
| 02:58.52 | agile | everything was working fine but it was recently moved to a new box, this seems to be the only glitch remaining |
| 02:59.26 | [TK]D-Fender | agile: Ok, go concentrate on getting solid backup. |
| 03:02.42 | joat | agile: it worked previously? |
| 03:03.08 | agile | joat, yes on another box/older asterisk |
| 03:03.30 | agile | 1.4.6 |
| 03:04.17 | joat | probably a change in syntax... |
| 03:04.20 | agile | I hear the file play, but the app never gets my dtmf |
| 03:04.37 | [TK]D-Fender | joat: So 1 hr later are you any closer to actually having some evidence of the CID not being set as advertised? |
| 03:04.53 | *** join/#asterisk sack (n=sack@152.Red-88-24-157.staticIP.rima-tde.net) |
| 03:05.25 | [TK]D-Fender | agile: And are you any closer to verifying that DTMF is functional at all on that channel outside of AGI? |
| 03:05.49 | agile | [TK]D-Fender, I set VoiceMail up on an extension and am able to use it |
| 03:05.51 | [TK]D-Fender | agile: max core debug would show you the DTMF if it was coming in at all... |
| 03:05.51 | joat | I believe I said that you were too difficult to deal with... I don't need to know the answer that badly that I have to put up with condescending comments... to paraphrase you: please consider the effects the tone of your questions have |
| 03:06.08 | [TK]D-Fender | agile: Show us the successful attempt, adn then the failed attempt |
| 03:06.40 | joat | agile: agile: dtmf being generated from where? |
| 03:06.51 | [TK]D-Fender | joat: And please consider that I asked you repeatedly so some sign of failure. A remarkably simple request that you seemed incapable of fulfiilling. |
| 03:06.52 | joat | SIP? |
| 03:07.02 | [TK]D-Fender | for* |
| 03:09.39 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:10.07 | joat | my point exactly |
| 03:10.15 | joat | condescending |
| 03:11.02 | [TK]D-Fender | joat: Condescending? a fair bit off of. |
| 03:12.11 | [TK]D-Fender | joat: So, have you actually found some evidence of your CID "failure" to show for all of this? I'm willing to help if you're honestly willing to try. |
| 03:14.00 | joat | I said forget it... I honestly don't want to through this with you... please drop it |
| 03:14.41 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 03:14.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:14.49 | joat | a remarkably simple request that you seem incapable of comprehending (see how insulting it comes across as?) |
| 03:16.08 | jaytee | it's alot more polite than "Hey DUMBASS!!! HOW MANY TIMES DO I HAVE TO ASK YOU FOR SOMETHING?????" now that's condescending! |
| 03:16.31 | [TK]D-Fender | jaytee: Yup... thats a winner |
| 03:16.34 | joat | sure... |
| 03:16.46 | joat | i'm asking him to ignore my previous question tho |
| 03:17.02 | joat | so the situation is somewhat "in reverse" |
| 03:17.26 | jaytee | no problem, I'm sure you'll find lots of people in here willing to ignore you |
| 03:17.44 | joat | i'm only worried about one at the moment |
| 03:17.46 | joat | heh |
| 03:17.57 | [TK]D-Fender | joat: Well you won't ahve to ask me as often as I asked you. And I'll actually concede to your request. 2 for me I guess. Good luck. |
| 03:18.25 | joat | now agile, where were we? |
| 03:18.45 | joat | ah... where's the dtmf being generated? via sip? |
| 03:18.49 | agile | so my app calls my cell# and asks me to enter a number, but it just times out.. never sees the numbers I give it. |
| 03:18.58 | *** join/#asterisk DigitalIrony (n=eric@nat/digium/x-ec126581e25c565a) |
| 03:19.39 | [TK]D-Fender | agile: And the 2 call attemps I asked you to show us which shows it working in one case, yet not the other? |
| 03:20.01 | joat | agile, have you looked at the dtmf settings in sip.conf? |
| 03:20.05 | agile | [TK]D-Fender, hehe if I could show you one where it was working I wouldn't have a problem :) |
| 03:20.41 | [TK]D-Fender | agile: So it failed with standard * apps as well? |
| 03:21.39 | agile | I did notice some 'errno Resource temporarily unavailable' after I did a agi debug in the console.. http://pastie.org/328455 |
| 03:23.51 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 03:23.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:25.28 | *** join/#asterisk luckyaba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 03:25.29 | [TK]D-Fender | agile: AGI Rx << temp buffer GET DATA /apps/voip/sounds/18_enter_pin 30000 4 - errno Resource temporarily unavailable <-- what is this temp buffer we see a few dozen times before this good looking call? |
| 03:25.49 | [TK]D-Fender | AGI Rx << GET DATA /apps/voip/sounds/18_enter_pin 30000 4 -- <SIP/voipms-00855000> Playing '/apps/voip/sounds/18_enter_pin' |
| 03:26.20 | agile | no idea, it's all the result of a single GET DATA /apps/voip/sounds/18_enter_pin 30000 4 |
| 03:26.38 | joat | agile: what's the GET DATA line in your dial plan look like? |
| 03:27.11 | [TK]D-Fender | agile: Well before we get too embroiled in the AGI part, verify that indeed dialplan apps fail outside of it |
| 03:29.00 | joat | if he didn't change the script when he changed versions, it's likely that the change was in asterisk syntax... one of the changes between 1.4 and 1.6 was that 1.6, by default, no longer tolerates vertical bars ("|") as separators in agi scripts (though it is configurable to do so) |
| 03:29.45 | joat | if he did change the script, then it's probably there |
| 03:29.47 | [TK]D-Fender | joat: Which would matter... if he was runing 1.6 in the first place |
| 03:29.58 | joat | thought he said he upgraded |
| 03:30.00 | [TK]D-Fender | joat: And dialplan has nothing to do with this. |
| 03:30.14 | [TK]D-Fender | joat: And you didn't seem to read where he told us the version. |
| 03:31.04 | [TK]D-Fender | [21:54]<agile>looking for clues as to why doing a GET DATA over agi doesn't seem to receive the dtmf on a newly setup asterisk 1.4.17 (ubuntu hardy) |
| 03:31.06 | joat | [sigh] if you want to be in a private conversation with agile, please take it out of the public forum... |
| 03:31.21 | joat | ah.. |
| 03:31.30 | joat | apologies |
| 03:31.35 | [TK]D-Fender | joat: Who said this has to be private? |
| 03:32.22 | joat | you seem upset that I'm also asking questions of agile |
| 03:32.39 | agile | hugs all |
| 03:32.46 | joat | heh |
| 03:33.08 | [TK]D-Fender | joat: No, correcting since your last to attempts at "assisting" are not at all applicable. |
| 03:33.28 | joat | be my guest then... |
| 03:34.37 | [TK]D-Fender | agile: So do go and try to see if it works 1 step further back before hitting AGI. |
| 03:35.57 | agile | core set debug 999999999, I still don't see anything as I hit the keypad |
| 03:36.52 | [TK]D-Fender | agile: Ok, show the attempt using straight dialplan apps. If you get nothing there this would point to an incorrect DTMFMODE or shaky audio quality if its inband |
| 03:37.10 | [TK]D-Fender | agile: For which or course it'd be nice to know exactly how this call is coming in... |
| 03:39.16 | agile | well that's the thing.. the call with issues is going out not in.. |
| 03:39.29 | agile | http://pastie.org/328455 I prepended some stuff to the paste |
| 03:40.44 | [TK]D-Fender | agile: Don't comapre apples & oragens. If you're working on inbound then make your test the same |
| 03:40.54 | [TK]D-Fender | oranges* |
| 03:41.13 | agile | not sure how I can originate a call from extensions.conf |
| 03:41.13 | [TK]D-Fender | <SIP/209.62.1.2-00857df0> Playing 'vm-password' |
| 03:41.35 | [TK]D-Fender | agile: whatever calls that AGI can jsut as easily call other dialplan |
| 03:41.57 | agile | guess I could go grab my wife's cell |
| 03:42.10 | [TK]D-Fender | agile: You already have the pphone to test with, don't you? |
| 03:42.21 | [TK]D-Fender | agile: So jsut change your dialplan around the AGI |
| 03:42.43 | jaytee | nite [TK]D-Fender |
| 03:42.48 | [TK]D-Fender | jaytee: Nite |
| 03:45.06 | agile | [TK]D-Fender, getting tired, think I'm just going to leave it 'till tomorrow |
| 03:45.19 | [TK]D-Fender | agile: ok |
| 03:45.27 | agile | appreciate the help guys |
| 03:45.38 | [TK]D-Fender | agile: You're welcom |
| 03:45.40 | [TK]D-Fender | e( |
| 03:45.51 | [TK]D-Fender | sigh... darn new keyboard... |
| 03:46.40 | echinos | how do I get a softphone to dial my * and get the default 's' entry point? |
| 03:47.19 | *** join/#asterisk luckyaba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 03:48.07 | [TK]D-Fender | echinos: dial "s" |
| 03:48.22 | echinos | I tried that, no luck |
| 03:48.46 | [TK]D-Fender | echinos: Pastebin the failed call's CLI output with SIP debug enabled |
| 03:48.56 | [TK]D-Fender | echinos: Along with your dialplan. |
| 03:48.58 | [TK]D-Fender | ~pb |
| 03:48.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 03:49.29 | echinos | ok, sec |
| 03:51.12 | echinos | hmm... should probably sanitize my DID out of there, eh? |
| 03:52.00 | echinos | meh. too hard. |
| 03:53.22 | echinos | I'll just get the login out of there |
| 03:53.48 | [TK]D-Fender | echinos: Yes, do mask passwords... |
| 03:54.01 | [TK]D-Fender | echinos: but thats all |
| 03:54.49 | echinos | duh |
| 03:55.12 | echinos | it's called sip, but here's the dialplan :/ |
| 03:55.13 | echinos | http://pastebin.ca/1273009 |
| 03:55.22 | echinos | I feel like a dough-head today |
| 03:55.43 | [TK]D-Fender | echinos: well... |
| 03:56.14 | [TK]D-Fender | echinos: exten 6,s,Playback(carried-away-by-monkeys) <- this is missing the "=>" after "exten", , then "s" is not a priority, it is an Asterisk Standard EXTENSION. |
| 03:56.48 | [TK]D-Fender | echinos: So that and a few other lines are completely ignored by *'s loading the dialplan |
| 03:57.03 | echinos | see? dough-head. |
| 03:57.12 | [TK]D-Fender | echinos: and of course the real clincher would ahve been seeing that call w/ SIP debug. |
| 03:57.28 | [TK]D-Fender | echinos: But, the dialplan does have several clear SNAFU's |
| 03:57.47 | *** join/#asterisk etfonhomey (n=chatzill@mobile-166-214-052-163.mycingular.net) |
| 03:58.02 | [TK]D-Fender | echinos: exten => 7057974908,s,Answer() <- also not a priority |
| 03:58.03 | echinos | Yeah, expected, I'm at the play-around-and-learn stage |
| 03:58.31 | [TK]D-Fender | echinos: Wash. Rinse. Repeat. |
| 03:58.36 | echinos | isn't that the line right after my -in context? |
| 03:58.40 | [TK]D-Fender | echinos: Clean it all up and give it another whirl. |
| 03:58.51 | [TK]D-Fender | echinos: I'm sure you see where it is. |
| 03:59.04 | [TK]D-Fender | echinos: go give everything a good looking over now and try again. |
| 03:59.39 | echinos | that's a normal usage for stuff coming in from my DID tho, no? |
| 03:59.53 | echinos | they do come in properly... |
| 04:00.16 | [TK]D-Fender | echinos: no, that line I showed you is indeed broken. |
| 04:00.27 | echinos | okie |
| 04:00.46 | echinos | thanks for the kind head-shake :) |
| 04:01.18 | echinos | that line was provided by my voip provider ;) |
| 04:01.40 | [TK]D-Fender | echinos: Unlimitel gave you that? Shame on them... and to think I refer customers to them... |
| 04:01.43 | *** join/#asterisk dmhardison (n=dmhardis@cs.roanoke.edu) |
| 04:01.58 | dmhardison | How is the voice quality on the linksys spa3102 as a fxo device |
| 04:02.08 | [TK]D-Fender | dmhardison: Pretty decent. |
| 04:02.21 | [TK]D-Fender | dmhardison: I wouldn't use it for a serious business line however... |
| 04:02.30 | echinos | well, I am a dough-head today, but yeah, I'm pretty sure it was. |
| 04:02.38 | [TK]D-Fender | dmhardison: Residential / infrequent use sure |
| 04:02.54 | dmhardison | [TK]D-Fender if I wanted to bump quality up a notch, what would you recommend? |
| 04:03.11 | [TK]D-Fender | dmhardison Whats your actual need? |
| 04:04.09 | dmhardison | Well, it is residential use, but I will use it probably a fair amount, so I want something fairly decent. |
| 04:05.01 | [TK]D-Fender | dmhardison: Most who I know using one find it OK. I'm fairly sure you'll be happy with it, and the built-in failover is a good bonus |
| 04:05.06 | dmhardison | I am actually using the t-mobile @home voip router, so I guess its going to end up doing a lot of converting. |
| 04:05.28 | dmhardison | Alright then. |
| 04:05.47 | [TK]D-Fender | dmhardison: Considered dumping them and getting your own provider direct? |
| 04:06.02 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 04:06.11 | drmessano | ouch |
| 04:06.18 | Bad_Robot- | evening all |
| 04:06.25 | drmessano | t-mobile is gonna be free :) |
| 04:06.29 | dmhardison | Well I am paying 10 dollars on top of my cell phone bill for it, plus I have no cell coverage at my house, so my blackberry actually connects to it and gives me voip coverage. |
| 04:06.38 | dmhardison | Its actually a really nice deal I think. |
| 04:06.43 | Bad_Robot- | sweet |
| 04:06.46 | drmessano | yeah |
| 04:06.56 | *** join/#asterisk Qorky (n=stephen@202-65-72-37-wireless.bbnet.com.au) |
| 04:07.09 | drmessano | I had considered that as well |
| 04:07.14 | [TK]D-Fender | dmhardison: Guess so... well you should probably be happy with the SPA then. |
| 04:07.26 | dmhardison | Alright, thanks for the information then. |
| 04:07.31 | drmessano | I dont like the cell > Analog > Digital conversion that much, but for $10 |
| 04:07.38 | drmessano | You really cant beat it |
| 04:07.42 | dmhardison | Yeah, I wish I could avoid it. |
| 04:07.43 | dmhardison | but. |
| 04:07.45 | orkid | its a gsm base? |
| 04:07.49 | orkid | the @home router |
| 04:07.52 | echinos | [TK]D-Fender: tyvm, progress made ;) |
| 04:08.03 | [TK]D-Fender | echinos: Glad to hear |
| 04:08.20 | [TK]D-Fender | orkid: No, just a bundled service from the same provider |
| 04:08.21 | dmhardison | orkid no, there is a sim card in it, but it goes via internet. |
| 04:08.29 | [TK]D-Fender | huh? |
| 04:08.45 | drmessano | yeah |
| 04:08.46 | dmhardison | i could actually drop my cell sim card in the router and have two lines via it. |
| 04:08.47 | [TK]D-Fender | dmYou mean your Voip router talks DIRECT to your cell? |
| 04:08.55 | Qorky | Question: I want to play some music or an announcement while I am Dialing a Sip client. Is this what the Dial(SIP/??,90,A()) feature is for? |
| 04:09.04 | Qorky | i dont want moh |
| 04:09.14 | [TK]D-Fender | Qorky: Yes |
| 04:09.14 | dmhardison | [TK]D-Fender Yes. |
| 04:09.27 | [TK]D-Fender | dmhardison: VERY funky... never heard of that anywhere |
| 04:09.31 | dmhardison | It's standard 802.11g |
| 04:09.32 | dmhardison | i think |
| 04:09.41 | [TK]D-Fender | dmum... thats just WiFi |
| 04:09.41 | drmessano | Internet <-- VoIP router <-- GSM to cell --> Cell |
| 04:09.47 | dmhardison | yeah, well. |
| 04:09.54 | drmessano | Its a repeater more or less |
| 04:10.02 | [TK]D-Fender | dmhardison: So why'd you mention GSM in there? |
| 04:10.04 | dmhardison | Basically. but free. |
| 04:10.08 | dmhardison | I didn't. |
| 04:10.13 | dmhardison | someone else did. |
| 04:10.17 | drmessano | It uses the GSM to give coverage to the cell |
| 04:10.29 | dmhardison | It isn't GSM, it is wifi. |
| 04:10.30 | [TK]D-Fender | dmhardison: So basically your cell does WiFi and you want to use your land-line while at home, and your cell service outside? |
| 04:10.42 | drmessano | I thought it gave cell service? |
| 04:10.53 | [TK]D-Fender | I'm getting somewhat confused now... |
| 04:10.54 | dmhardison | Well it does, via voip when there is no coverage. |
| 04:11.13 | dmhardison | Hold on, it is verrrry cool. |
| 04:11.22 | dmhardison | http://www.t-mobile.com/shop/addons/services/information.aspx?tp=Svc_Tab_TMobileAtHome |
| 04:11.27 | dmhardison | I think is the link. |
| 04:11.44 | orkid | so whats the point of putting sim card in the router? |
| 04:12.33 | dmhardison | Well I have two lines.. One from my cell phone, and the other 10 dollars I get a line for my router I can use as a landline. |
| 04:12.38 | kb3ien | app_voicemail.c: Failed to obtain database object for 'asterisk'! i assume this means i somehow told asterisk to keep my voicemails in an rdbms ? |
| 04:12.59 | Corydon76-dig | kb3ien: correct |
| 04:13.02 | dmhardison | And they have a sim card I drop in there. |
| 04:13.20 | Corydon76-dig | kb3ien: compile-time option |
| 04:14.17 | kb3ien | i was thinking about getting something from telular (sp?) to get an FXS port from my cell carrier. |
| 04:17.54 | kb3ien | tellular, and its no more expensive than a regular cellphone. |
| 04:17.55 | Bad_Robot- | do you guys normally setup a few pots lines to be a backup if the T1 goes down |
| 04:18.01 | kb3ien | to operate, not to buy... |
| 04:21.11 | Qorky | <[TK]D-Fender>. I cant get it to play. |
| 04:21.18 | Qorky | its as if it ignores the code.. |
| 04:21.38 | [TK]D-Fender | Qorky: pastebin it... |
| 04:21.39 | [TK]D-Fender | ~pb |
| 04:21.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 04:22.12 | Qorky | its only one line |
| 04:22.47 | [TK]D-Fender | Qorky: I'm sure there's MUCH more.. |
| 04:22.54 | [TK]D-Fender | Qorky: please show teh COMPLETE call. |
| 04:22.57 | Qorky | exten => 88811,1,Dial(SIP/82,90,A(/etc/asterisk/sounds/3mpromo-final)) |
| 04:23.29 | Qorky | so im calling 82 for 90 secs. |
| 04:23.43 | Qorky | and want it to be ringing 82 while it plays the accouncement |
| 04:24.03 | [TK]D-Fender | Qorky: Ah, I did mistake something in there. |
| 04:24.16 | Qorky | or can i background the sound file while it rings. |
| 04:24.22 | [TK]D-Fender | Qorky: No, there is no clean way to play a specific file as the ringing like that |
| 04:24.55 | [TK]D-Fender | Qorky: the closest fake-out is to make a MoH class with only that file available to it and use that as MoH for the call |
| 04:25.09 | Qorky | ah ok. |
| 04:25.16 | Qorky | i was trying to get away from that. |
| 04:25.20 | Qorky | but that can work i guess |
| 04:25.23 | [TK]D-Fender | Qorky: This is something they SHOULD add. |
| 04:25.31 | orkid | hmm, can anyone try 18004664411@tollfree.sip-happens.com ? i get no ring, but asterisk says "Called 18004664411@tollfree.sip-happens.com" |
| 04:25.37 | [TK]D-Fender | Qorky: I understand... |
| 04:25.43 | orkid | it even get the srv and maps to host sometimes.sip-happens.com |
| 04:27.36 | [TK]D-Fender | orkid: Yup, hangs just the same |
| 04:27.59 | orkid | [TK]D-Fender: tx |
| 04:28.32 | [TK]D-Fender | orkid: It keeps retransmitting |
| 04:29.19 | [TK]D-Fender | orkid: Who is that host? |
| 04:30.00 | orkid | sip-happens? free toll-free termination, it used to work (although not for that number). the number is 1-800-GOOG-411 |
| 04:32.29 | orkid | i can ping the host, nmap shows filtered for 5060 tcp and open|filtered for 5060 udp on both tollfree. and sometimes. |
| 04:32.36 | orkid | maybe they're down? |
| 04:32.44 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 04:34.09 | [TK]D-Fender | orkid: Well the only link I found for them has a homepage made in notepad with an obvious 5 minutes of non-effort |
| 04:34.52 | jql | sounds like all of my webpages |
| 04:36.16 | orkid | just b/c it's not fancy doesn't mean it's not good |
| 04:36.45 | orkid | usually b/w all text pages give me the impression that the author cared about the content rather than the delivery |
| 04:37.37 | orkid | i guess i'll go through future-nine |
| 04:37.40 | drmessano | Try tollfreegateway.com |
| 04:37.45 | drmessano | sip-happens has been failing |
| 04:37.50 | orkid | ok i'll check that out too |
| 04:37.51 | orkid | thanks |
| 04:38.12 | drmessano | Sadly it's the first lookup in ENUM and my tollfreecalls via enum have been sucking |
| 04:39.00 | orkid | does tollfreegateway.com set CID to what asterisk sends? |
| 04:40.36 | orkid | got through to google. google records calls? :S |
| 04:40.54 | orkid | they seem to record everything they get their hands on. :) |
| 04:43.49 | drmessano | You want tollfree calls AND CID spoofing? |
| 04:43.54 | drmessano | Wow, want fries too? |
| 04:44.03 | drmessano | [TK]D-Fender |
| 04:44.20 | orkid | not spoofing :P |
| 04:44.27 | orkid | but i guess some would be spoofing stuffz |
| 04:44.28 | drmessano | It is spoofing |
| 04:44.32 | orkid | yea i know |
| 04:44.52 | orkid | technically.. anyway, it doesnt seem to terminate my stuffs in canaduh, like ministry of transportation i just tried calling. |
| 04:45.14 | orkid | "we're sorry, your call did not go through" |
| 04:45.21 | drmessano | [TK]D-Fender: You die? |
| 04:45.27 | [TK]D-Fender | drmessano: Not yet |
| 04:45.31 | drmessano | lol |
| 04:45.34 | drmessano | Quick question |
| 04:45.38 | [TK]D-Fender | drmessano: Figured I'd jsut walk away from this one... |
| 04:47.57 | drmessano | Off the top of your head, unofficially.. If I was testing bandwidth using multiline phones.. Say I wanted to dial 6, 8, 10, 12 calls to a conference somewhere, where the far end had NO bandwidth limits (within reason), and I was using G711.. then wanted to scale that result to G729.. |
| 04:48.20 | drmessano | Ive heard 64k for G711, and 10k for G729, but the G729 has varied in the results I found |
| 04:48.21 | etfonhomey | [TK]D-Fender, I was checking out that T-Mobile router. It must use info off the SIM card(s) to authenticate each line, up to two, over broadband. |
| 04:48.29 | drmessano | So I am thinking roughly 6 to 1 |
| 04:49.00 | orkid | 84 or is it 87 with ethernet overhead for g711 i think |
| 04:49.03 | [TK]D-Fender | drmessano: Conferening G.729 is psychotic |
| 04:49.14 | [TK]D-Fender | drmessano: it demuxes each and mixes. |
| 04:49.17 | drmessano | I dont want to use the conference for G729 |
| 04:49.22 | drmessano | Just testing bandwidth |
| 04:49.24 | [TK]D-Fender | drmessano: Big load |
| 04:49.57 | [TK]D-Fender | drmessano: Not sure what you're asking then |
| 04:50.13 | drmessano | I guess the conference detail was irrelevant |
| 04:50.17 | [TK]D-Fender | drmessano: 8k for G.729 last I heard. + overhead |
| 04:50.46 | [TK]D-Fender | overhead = +/- 20kbps |
| 04:51.02 | drmessano | Multiple calls to one endpoint ---> endpoint has no bandwith limit, near end does.. want to go 7, 10, 12 calls and go "Oh, 8 gets choppy and the bandwidth is maxed"... then scale that result to G729 from G711 |
| 04:51.27 | drmessano | hmm |
| 04:51.45 | jql | speed test would do that |
| 04:52.07 | jql | myspeed has a capacity tester which blasts udp till packet loss |
| 04:52.08 | drmessano | Speedtests are not gonna test the specific endpoint |
| 04:52.23 | jql | I thought it was supposed to have unlimited bandwidth |
| 04:52.27 | jql | speed tests have that |
| 04:52.40 | drmessano | Unlimited bandwidth has nothing to do with routing |
| 04:52.54 | drmessano | and routing to the speedtest locations is all over the place |
| 04:53.21 | jql | if you have a bottleneck worse than the one between you and your ISP, I'm sincerely sorry to hear it. :) |
| 04:53.27 | drmessano | I can run 2 speedtests at any one site that are seperated by the same distance in 2 directions from here and get 25% error either way |
| 04:53.42 | drmessano | you're not making sense |
| 04:53.44 | jql | but you're right -- if you have bottlenecks upstream from your direct ISP link, speed test is a bad idea |
| 04:54.01 | drmessano | I'm not worried about MY ISP or even the far end ISP |
| 04:54.12 | jql | exactly my point. I thought you were |
| 04:54.18 | drmessano | The Speedtests are NOT free of defect where they are located |
| 04:54.25 | drmessano | They are NOT a good metric of the link between |
| 04:54.56 | drmessano | My concern is not that the speedtest will hide a bottleneck at the end points, my concern is the speedtests themselves are sketchy |
| 04:55.01 | jql | we aren't talking about the same type of test. don't sweat it |
| 04:55.04 | drmessano | Which they indeed are |
| 04:55.39 | drmessano | Do 10 speedtests, 2 at each of the common sites, and you'll get 10 different results, with HUGE error |
| 04:56.27 | drmessano | So I would rather test the real link, and forget about latency at some unknown far end |
| 04:56.29 | jql | there's a test I run to determine final-hop capacity, not end-to-end speed. that's why we're not on the same page |
| 04:56.30 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 04:57.01 | drmessano | It's still testing to/from a specific location |
| 04:57.16 | joako | Has anyone replaced a door intercom with something SIP? I was looking at this: http://voip.valcom.com/products/ip_intercomdoorphones.htm and it is perfect, except that the button should ring the door bell |
| 04:57.18 | drmessano | a location that is NOT any more a control point than speedchicken.com |
| 04:57.52 | joako | I was thinking 1) change the button 2) remove the door bell and create a virtual one with paging to the SIP phones |
| 05:00.41 | *** join/#asterisk suahmed (n=Rubel@58.65.224.5) |
| 05:06.16 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 05:06.30 | drmessano | joako |
| 05:06.51 | drmessano | Get a PAP2, cheap.. set the dialplan to autodial |
| 05:07.06 | drmessano | You pick up, and it dials extension X after say 2 seconds of delay |
| 05:07.15 | drmessano | Could be instant, but I would add delay |
| 05:07.38 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 05:07.56 | orkid | seems like there are a lot of options for TF, like tf.callwithus.com , but i just tried goog411 and it gave me congested |
| 05:08.44 | drmessano | hmmm |
| 05:10.45 | orkid | and betamax charges! gheee-tt-ttoooo |
| 05:10.47 | orkid | anyway |
| 05:11.23 | drmessano | betamax sucks |
| 05:11.51 | drmessano | I read up on them after seeing them mentioned.. they get more complains for bait/switch than a drug dealer |
| 05:12.07 | orkid | fo sho |
| 05:12.13 | orkid | but they're cheapest around for some internationals |
| 05:12.24 | orkid | so for phone diailing to such places its good enough for me |
| 05:12.31 | orkid | surely not business type calls |
| 05:13.11 | orkid | they have inconsistencies in billing, stuff like that, its hard to say whether or not u actually lose money on it or not, its all a penny here a penny there |
| 05:13.59 | [TK]D-Fender | Betamax? Didn't that lose to VHS? |
| 05:14.32 | orkid | i'm surprised u guys haven't heard of betamax |
| 05:14.33 | orkid | finarea |
| 05:14.39 | orkid | or are u just pokin fun |
| 05:15.07 | drmessano | A company that steals your money is not worth the risk of "cheap calls" |
| 05:15.08 | [TK]D-Fender | orkid: Both. |
| 05:15.23 | [TK]D-Fender | orkid: You are asking to get bit. |
| 05:15.23 | orkid | they also let u 'steal' some minutes |
| 05:15.26 | orkid | lol |
| 05:15.34 | drmessano | But then again, there is no limit to the extent of what people will subject themselves to, and then justify ad nauseum, for cheap anything |
| 05:15.50 | orkid | i'd have to dump the call records and actually add it up to see how much i'd pay with another provider. right now it's definitely cheaper that using a calling card |
| 05:16.04 | orkid | drmessano: especially when the word 'free' is involved |
| 05:16.05 | orkid | lol |
| 05:16.16 | drmessano | "They listen to your calls and make heavy breathing noises, and sometimes farts, but the termination is .001 cents" |
| 05:16.58 | orkid | lol |
| 05:17.32 | orkid | what's the reason for autofallthrough=yes ? i mean, is it 'safer' or something b/c u can't re-enter extensions or some such (in IVR).. what about if you Hangup() right after? |
| 05:17.41 | drmessano | I seriously hate firefox |
| 05:18.05 | orkid | i'm surprised tf.callwithus.com is congested. |
| 05:18.14 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 05:18.18 | orkid | hmpf. anyone care to try sip:18004664411@tf.callwithus.com ? |
| 05:19.16 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 05:20.37 | drmessano | tollfreegateway works fine |
| 05:21.53 | orkid | yup but try 1800-268-4686 (GENERAL inquiry for ontario transportation ministry) |
| 05:22.24 | drmessano | I really don't care about Canada |
| 05:22.51 | orkid | lol |
| 05:22.56 | orkid | u hate, u dont care |
| 05:22.57 | orkid | what else |
| 05:23.01 | orkid | :P |
| 05:23.26 | drmessano | I'm an American, why the hell should I not show indignant ignorance of anything outside our borders? |
| 05:23.30 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 05:23.42 | drmessano | "Mexico? Nuke em" |
| 05:23.46 | drmessano | "Canada? Nuke em" |
| 05:23.56 | drmessano | "Bru-what? Nuke em" |
| 05:25.19 | drmessano | I think ipkall terminated tollfree as well |
| 05:25.28 | drmessano | voiper.ipkall.com |
| 05:25.55 | drmessano | They're in Washington state.. Maybe they don't hate Canada as much |
| 05:31.51 | orkid | gor gin no answer (duh its late night) but i wonder if they have automated service |
| 05:31.54 | orkid | will see tomorrow |
| 05:33.01 | drmessano | Maybe they hate you |
| 05:36.30 | orkid | calling their 416 number also no answer, so maybe ) |
| 05:37.23 | orkid | lols crap its late, i called the wrong number the no answer one |
| 05:38.17 | orkid | but the proper 416 works hm |
| 05:41.08 | orkid | fault is with TF termination, just called 18002684686 over pots and it worked |
| 05:42.18 | drmessano | ok |
| 05:42.36 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.151.162) |
| 05:43.22 | yidiyuehan | hi, just check with you guys, have you ever tested the impedance of PBX with Digium interface card? |
| 05:45.48 | drmessano | Its probably 600 ohms, +/- 600 |
| 05:49.56 | *** join/#asterisk terracon (n=garry@72.138.175.134) |
| 05:55.59 | *** join/#asterisk flewid (i=flewid@joker.bitchx.org) |
| 06:05.01 | *** join/#asterisk BeeBuu (n=beebuu@218.13.97.130) |
| 06:07.01 | joako | drmessano: You mean a PAP2 with something like a Doorbell Fon? Wouldn't that need something like a X100 card instead?.. I don't want an actual telephone at the door.... |
| 06:07.52 | joako | Or any idea how to interface a NuTone intercom to Asterisk? |
| 06:09.24 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 06:09.55 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 06:10.00 | drmessano | I have no idea what youre trying to do |
| 06:10.17 | Micc | I just installed asterisk on centos 5 and I've got these remote unix connections happening every 3 seconds now. |
| 06:10.41 | drmessano | Did you install FreePBX? |
| 06:11.47 | Micc | what is freepbx? |
| 06:12.18 | drmessano | How did you install Asterisk? From source? |
| 06:12.27 | drmessano | or from an ISO based distro? |
| 06:12.46 | joako | drmessano: right now there is a box at the door with a speaker/microphone and door bell button. I want to make that digital |
| 06:13.12 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 06:13.24 | joako | As amusing as it might be I don't want a Weco 2552 at my front door |
| 06:13.35 | drmessano | joako: PAP2 + 600 ohm transformer.. Button puts the box offhook and it autodials |
| 06:14.14 | Micc | I installed from 1.4.22 source. |
| 06:14.45 | drmessano | Ok, plain vanilla CentOS CD? |
| 06:15.12 | Micc | Not sure, I'm using a virtual private server, it came with centos preinstalled. |
| 06:15.27 | drmessano | Not an asterisk distro? |
| 06:15.35 | joako | 600 ohm *transformer*? |
| 06:15.36 | Micc | uname -r says: 2.6.18-028stab059.5-ent |
| 06:16.02 | drmessano | joako: Thats what I said |
| 06:16.04 | Micc | not an asterisk distro, I didn't know there was such a thing, crazy how far we've come in the last few years. |
| 06:16.37 | drmessano | joako: Radio Shack, under $2 |
| 06:16.48 | drmessano | joako: 600 ohm isolation transformer |
| 06:17.01 | *** part/#asterisk ezrafree (i=ezra@gware/developer/ezrafree) |
| 06:18.31 | joako | How would I wire that? Switch feeds power into the line? Using what sort of speaker and microphone? |
| 06:18.46 | Micc | aha, I think I figured it out, two safe_asterisk processes were running I think. |
| 06:19.10 | drmessano | " Has anyone replaced a door intercom with something SIP?" |
| 06:19.17 | drmessano | Seems you have it all there |
| 06:20.10 | drmessano | joako: It doesnt sound like you're even close to being proficient enough to do this.. What were you expecting here? |
| 06:20.29 | drmessano | joako: This is gonna take some hacking, but not difficult |
| 06:21.03 | yidiyuehan | Drmessano, what's the testing equipments required to measure the AC impedance? I purchased two OEP 1200 Telephone Transformers and setup a circuit, but it doesn't seems right... |
| 06:21.21 | joako | drmessano: I'll admit I am not proficient at creating circuits.... but the rest iI can manage |
| 06:21.41 | drmessano | joako: This isnt creating a circuit, this is wiring a transformer |
| 06:22.05 | drmessano | joako: This is electronics 102 here |
| 06:22.52 | drmessano | yidiyuehan: Who measures? Nominal impedence on US telephone circuits is supposed to be 600 ohms |
| 06:24.57 | yidiyuehan | yes, by right the impedance should be 600 ohms, and I want to measure the impedance of digium cards and calculate the Return loss. |
| 06:25.20 | joako | drmessano: so tell me how you would wire a momentary pushbuton, 600 ohm transformer, microphone and speaker to behave like a telephone set? |
| 06:25.39 | drmessano | Its not gonna vary enough from 600 ohms to expend this effort, yidiyuehan.. If they were far off 600 ohms, you would know |
| 06:25.58 | joako | drmessano: Like I said I am not proficient in ciruits, electronics, electirical engineering or whatever you want to call it |
| 06:26.54 | yidiyuehan | I am staying in Singapore and we are following ETSI TBR038 standards. and I have let one of the testing lab test the PABX with digium card for me, and it shows that return loss is lower than 14dB (min value). Hence I want to measure it myself by seting up a simple circuit. |
| 06:27.34 | *** join/#asterisk synchris (n=synchris@athedsl-04914.home.otenet.gr) |
| 06:30.41 | *** join/#asterisk theskinfox (n=anaxagor@ool-457c5d2e.dyn.optonline.net) |
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| 06:31.40 | drmessano | yidiyuehan: There is NO simple way to test impedence |
| 06:32.21 | orkid | lol UK did starting at 0.34$ |
| 06:32.24 | orkid | 31 |
| 06:33.03 | orkid | yidiyuehan: if return loss is lower than a minimum, can't u put in an attenuator? |
| 06:33.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:34.00 | yidiyuehan | drmessano, you are right, and I have setup the circuit using signal generator, a few 600 resistors, telephone transformer, and oscilloscope to measure the output voltage. but just not sure whether it's the way to measure RL or not |
| 06:34.28 | theskinfox | anybody know much about one touch recording? I have it setup and it works just fine, but I need the ability to play different sound files when the recording stops and starts |
| 06:34.29 | drmessano | If you're gonna test impedence, you need a noise bridge.. |
| 06:34.57 | drmessano | Even then, thats not even close to being scientific |
| 06:35.21 | yidiyuehan | drmessano, according to ETSI TBR038, I only need to use a few 600 ohms resistor to setup the noise bridge |
| 06:36.26 | yidiyuehan | any reference website regarding to impedance measurement circuit? |
| 06:36.39 | drmessano | Google for "Noise bridge" |
| 06:36.50 | drmessano | Lots of good reading.. its more than a few resistors |
| 06:37.07 | drmessano | You cant measure inductive reactance or capacitive reactance with resistors |
| 06:37.44 | yidiyuehan | erm....will do some more home work. thanks man~~ |
| 06:42.24 | *** join/#asterisk BeeBuu (n=beebuu@219.135.43.144) |
| 06:46.34 | orkid | wheatstone bridge? :) |
| 06:46.53 | orkid | so u need a bunch of resistors, capacitors, and possibly inductors :) |
| 06:47.19 | orkid | there are boxes out there that do it, u hook up the stuff u want to test to it |
| 06:47.23 | orkid | (iirc) |
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| 06:56.05 | drmessano | wheatstone bridge can measure an unknown component for resistance, capacitance, or inductance reliably.. It can measure purely resistive impedence, but not inductive or capactive reactance, which is almost certainly going to be involved in a telephone circuit |
| 06:57.42 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-1df8defebf65bcf6) |
| 07:12.17 | orkid | how is capacitive reactance different from capacitance |
| 07:12.21 | orkid | transient? |
| 07:17.13 | drmessano | There's not a comparison.. Capacitive reactance is a capacitive element to what should be a purely resistive line. Pure resistance is mostly an illusion, and all networks have some capacitance or inductance.. Some telephone systems actually build in a capacitive reactance due to it being easier to balance capacitive reactance than high levels of inductive reactance. |
| 07:29.29 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 07:33.45 | *** join/#asterisk tokozedg (n=tokozedg@85.118.98.122) |
| 07:35.04 | tokozedg | i have installed asterisk 1.6 an mysql for cdr+ created database asterisk and table cdr and columns, but cdr doesn`t writes anythins in this table |
| 07:35.19 | tokozedg | so how can i solve it? |
| 07:47.04 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 07:51.39 | *** join/#asterisk somebee (n=somebee@76.84-48-209.nextgentel.com) |
| 07:52.20 | somebee | how do I send variables with the AJAM-interface? (the actual variables in url/?action=....&variables= (?) |
| 07:52.30 | *** join/#asterisk EE_BB (n=chatzill@212.58.24.138) |
| 07:53.27 | EE_BB | hii all I am making a call but I cant see anything on asterisk console like which number is calling ,who is caller status error code ? |
| 07:54.11 | angryuser | hello can someone tell me how do you integrate the stuff with the crm ? let's say i have a queue and with agi i record the ticket number with dtmf, now i want to refresh the support's agent internet page with the link i generate, something like this ,(ticket number 123456) http://my.site.com/staff/index.php?_m=tickets&_a=viewticket&ticketid=123456 , but how send this to agent i want ? |
| 07:54.41 | EE_BB | I could see before |
| 07:54.45 | angryuser | how to make him refresh it's page ;) |
| 07:54.59 | EE_BB | is there any loggin level or something like that ? |
| 07:55.13 | angryuser | EE_BB: yes, try asterisk -vvvvvvvvvvvvvvvvvvvr |
| 07:55.59 | angryuser | EE_BB: but normally you should see it with the minimal verbose |
| 07:56.10 | EE_BB | no verbose |
| 07:56.17 | *** join/#asterisk xrmx__ (n=user@79.0.181.104) |
| 07:58.00 | EE_BB | now -vr is ok |
| 08:04.25 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
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| 08:16.44 | *** part/#asterisk zydoon (n=zydoon@41.225.142.229) |
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| 08:35.06 | tokozedg | anyone here knows how to condifure cdr? |
| 08:36.51 | *** join/#asterisk Daejeo (n=chatzill@114.201.159.94) |
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| 08:37.22 | Daejeo | Meow :) |
| 08:37.38 | Daejeo | any mouse |
| 08:38.44 | Daejeo | Credit Card Fraud Protection with Asterisk |
| 08:39.00 | Daejeo | any developer around ? |
| 08:39.13 | Daejeo | wanna share my idea |
| 08:39.30 | drmessano | Daejeo: You're just itching to make a million dollars off asterisk, arent you? |
| 08:40.01 | *** part/#asterisk genin (i=phrame@ANice-252-1-60-32.w82-122.abo.wanadoo.fr) |
| 08:40.28 | Daejeo | drmessano: I need a telephony platform so i picked up Asterisk |
| 08:40.46 | Daejeo | cisco-- very expensive |
| 08:44.48 | Daejeo | very easy to make API for asterisk , open source , easy to govern what else do i need? |
| 08:46.00 | Daejeo | <PROTECTED> |
| 08:46.27 | Daejeo | But asterisk can protect it |
| 08:47.43 | *** join/#asterisk pcrane (n=pcrane@125-238-255-151.broadband-telecom.global-gateway.net.nz) |
| 08:47.44 | Daejeo | drmessano: isn't it useful ? |
| 08:48.10 | Daejeo | or you can advise something else |
| 08:52.08 | EE_BB | hii all I have a problem about sip, after sip client register itself I can call this client a while but after this period I cant call this client |
| 08:52.32 | EE_BB | if client re-register itself I can call again |
| 08:53.19 | EE_BB | while I cant call client client still calls me |
| 08:55.15 | *** join/#asterisk chendy (n=chatzill@58.251.111.222) |
| 09:02.12 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 09:19.27 | *** join/#asterisk JonOnt (n=Jon@72.34.90.74) |
| 09:20.31 | JonOnt | Hey guys, I set up a page extention on exten 800, when I dial it, all the phones in the office all say that is not a valid *something* number |
| 09:21.57 | JonOnt | Hey any one feel like calling, to test out the system? |
| 09:22.50 | *** part/#asterisk somebee (n=somebee@76.84-48-209.nextgentel.com) |
| 09:24.55 | yang | JonOnt: don't you have a mobile phone? |
| 09:25.48 | JonOnt | yang, not till next week, spent all my money trying to build this office |
| 09:26.00 | yang | query me your number i will call |
| 09:27.43 | JonOnt | 909-740-3976 |
| 09:27.51 | JonOnt | shooot |
| 09:27.58 | JonOnt | that was supposed to be ina querry |
| 09:28.37 | SwK | YAY! More NUmbers to telemarket too! |
| 09:29.00 | JonOnt | turns on zapateller |
| 09:29.04 | SwK | heh |
| 09:29.21 | SwK | that doesnt always work |
| 09:29.37 | SwK | knows how to beat the telezapper |
| 09:29.44 | SwK | and zapateller |
| 09:30.20 | JonOnt | hey some one call that number, so i can see if it rings |
| 09:30.32 | JonOnt | i know i can call out, i called goog411 |
| 09:30.37 | SwK | ring ring |
| 09:31.28 | SwK | hmmm |
| 09:31.44 | SwK | not getting thru |
| 09:32.47 | JonOnt | hum, wierd, my home just rang and rang, answering machine should have picked up |
| 09:33.11 | SwK | ok... hmm i didnt even get early media back on it hah |
| 09:33.21 | donnib | anyone know experience VERY bad ecco when using conference ? I have problems when the third person enter the conference room then it becomes unusable because of the ecco. |
| 09:34.05 | SwK | saw that once with older polycom firmware |
| 09:34.59 | angryuser | can someone tell me if the field FROM in sip messages can be FROM= or the standart require FROM: ? |
| 09:36.49 | SwK | from: or f: |
| 09:37.27 | SwK | you should refer to the RFC as there are also "compact" headers that are things like t: f: l: etc |
| 09:37.55 | *** join/#asterisk dramman (n=Miranda@122.111.59.159) |
| 09:38.41 | dramman | Can anyone help me setup Asterisk with multiple inbound SIP trunks? |
| 09:40.35 | dramman | what's an SBC? |
| 09:45.36 | orkid | session border controller |
| 09:45.39 | orkid | ? |
| 09:45.51 | orkid | or some bloody computer? |
| 09:45.53 | orkid | :) |
| 09:54.25 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 09:55.10 | angryuser | SwK: thank you, some provider asked me to modify : to = in From, i can do it with some tools, but if it is not in rfc i can tell him to ##### his keyboard. |
| 09:56.22 | SwK | http://www.ietf.org/rfc/rfc3261.txt |
| 09:56.35 | Daejeo | any developer around ? |
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| 10:24.12 | dramman | Has anyone been able to get "Dial(Mobile/bluetooth-extension)" working? |
| 10:29.22 | *** join/#asterisk lou_gr (n=lou_gr@212-70-216-131.ath.static.tee.gr) |
| 10:31.30 | Karlitoo | can any 1 help me out with h323 please |
| 10:34.44 | *** join/#asterisk awk (n=awk@41.28.224.86) |
| 10:34.52 | awk | hi... yellow light on FOP usually means what? |
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| 11:08.45 | Daejeo | awk: are you there/ |
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| 12:12.15 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:17.56 | tokozedg | hi i have php script and when i execute it it make his job an than it is closed |
| 12:18.25 | tokozedg | and can i make so that every 1 hour it was automaticaly executed? |
| 12:18.27 | tokozedg | in suse |
| 12:20.11 | angryuser | tokozedg: use cron |
| 12:23.00 | tokozedg | ok thanks |
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| 12:26.17 | *** join/#asterisk Valmon (n=m_dorset@l-a.net.incru.net) |
| 12:26.45 | Valmon | Greetings to all |
| 12:27.48 | Valmon | Valmon probably someone can show me rigth way in my problem |
| 12:27.53 | *** part/#asterisk Valmon (n=m_dorset@l-a.net.incru.net) |
| 12:28.09 | *** join/#asterisk Valmon (n=m_dorset@l-a.net.incru.net) |
| 12:29.48 | Valmon | I trying to setup fax receiving with asterisk 1.4.21 with spandsp-0.0.4-12_pre16.el5 on Centos 5.2 |
| 12:30.46 | Valmon | Extention looks like this: |
| 12:31.04 | Valmon | exten => 1,1,Playtones(ring) |
| 12:31.04 | Valmon | exten => 1,n,Wait(3) |
| 12:31.04 | Valmon | exten => 1,n,StopPlayTones |
| 12:31.04 | Valmon | exten => 1,n,Set(TIMEOUT(absolute)=60) |
| 12:31.05 | Valmon | exten => 1,n,Set(FAXFILE=/tmp/test.tif) |
| 12:31.05 | Valmon | exten => 1,n,RxFax(${FAXFILE}|debug) |
| 12:31.07 | Valmon | exten => 1,n,System(/bin/mail -s "fax arrived from ${CALLERID(name)}" < ${FAXFILE}) |
| 12:31.09 | Valmon | exten => T,1,Hangup() |
| 12:31.19 | angryuser | ~pb |
| 12:31.20 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 12:31.26 | Valmon | sorry |
| 12:34.50 | Valmon | http://pastebin.com/d3345cfff |
| 12:34.58 | Valmon | problem in next. |
| 12:35.11 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 12:35.38 | Valmon | then I calling, or fax do it - where is only silence and fax is hangup on timeout |
| 12:36.18 | Valmon | so no any activity, all stacked on RxFAX("SIP/00040165-09f15f50", "/tmp/test.tif|debug") in new stack |
| 12:36.29 | Valmon | and nothing else. |
| 12:36.56 | Valmon | I just don't know, in which side I should searching |
| 12:37.12 | Valmon | to fix problem |
| 12:42.23 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.195.82) |
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| 12:52.16 | Valmon | sorry for bothering. |
| 12:52.59 | Valmon | do some of you has succesefull configuration asterisk+spand for receivinf faxes? |
| 12:54.22 | klmpencran | Hi. Need help with zaptel compilation/configuration on an Ubuntu server. Am I in the right Place ? Googleing did not work for me. |
| 13:04.41 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:05.46 | Daejeo | angryuser: how do you feel being a angryuser all the times ? |
| 13:08.56 | klmpencran | anyone can help me please ? The problem encountered is a ZT_CHANCONFIG failed |
| 13:10.02 | *** join/#asterisk alibb (n=ali@41.226.86.139) |
| 13:10.28 | Valmon | we are here not in best time :-( |
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| 13:29.39 | Leddy | Is it possible for the voicemail messages to be written as 775 instead of 755 (permissions)? |
| 13:30.02 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
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| 13:36.44 | Valmon | 2Leddy What the point of that? |
| 13:36.48 | thieums | Hi, basic dialplan question : do you know a way to execute dialplan commands after a Dial cmd (hang up by incoming channel) |
| 13:36.56 | klmpencran | Ok, I'll try to be more precise : I've got a ZT_CHANCONFIG failed, no such device or adress with ztcfg command. But wctdm module is loaded. lspci shows an unknown Digium device |
| 13:38.31 | klmpencran | any help about this ? |
| 13:38.44 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:39.19 | Leddy | so I don't have to reconfigure apache to run as asterisk, I can put them in the same gorup |
| 13:39.23 | Leddy | group |
| 13:39.38 | [TK]D-Fender | thieums: read up on the "h" Asterisk Standard Extension |
| 13:40.27 | [TK]D-Fender | klmpencran: pastebin your zapata.conf , zaptel.conf , "cat /proc/interrupts" and "ztcfg -vvvv" |
| 13:40.28 | [TK]D-Fender | ~pb |
| 13:40.29 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 13:40.30 | [TK]D-Fender | ^^^^ |
| 13:40.36 | Valmon | certanly tha could be changed in sources. but probably the most easy way - chown asterisk:apache dir and chmod 770, don't think? |
| 13:41.13 | Leddy | ahh found it |
| 13:41.15 | Leddy | asterisk.conf |
| 13:41.17 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:42.35 | klmpencran | [TK]D-Fender : I'll do that, but just a question before : I learned in Asterisk Bootcamp that zapata.conf was used for zaptel working with Asterisk. My problem seems to happen before the loading of Asterisk no ? |
| 13:43.20 | [TK]D-Fender | klmpencran: True, but I want to make sure the big picture is sane too and if you left legacy stuff it'd tip off for other things I might not think to ask for following. |
| 13:43.22 | thieums | [TK]D-Fender: the h extension seem to work only when called extension hangs up, not calling |
| 13:43.39 | [TK]D-Fender | klmpencran: proactive forensics |
| 13:44.23 | [TK]D-Fender | thieums: you said "inbound". If you are dilaling OUT and and want the dialplan to continue then use the "g" dial option. |
| 13:44.51 | [TK]D-Fender | wow... did that come out right... no matter... |
| 13:45.05 | [TK]D-Fender | thieums: There you go.. the other half of the coin |
| 13:46.59 | tzafrir_laptop | klmpencran, what's the output of: cat /proc/zaptel/* |
| 13:47.09 | klmpencran | [TK]D-Fender: http://www.mibbit.com/pb/YmHzbb . Thanks for help |
| 13:48.13 | [TK]D-Fender | klmpencran: I do not see "wctdm" in your interrupts list like I should. mode probe it agan and repeat |
| 13:48.19 | [TK]D-Fender | modprobe* |
| 13:48.26 | tzafrir_laptop | klmpencran, I don't see the card there . Do you see it on zaptel_hardware ? |
| 13:48.32 | klmpencran | tzafrir_laptop : Span 1: ZTDUMMY/1 "ZTDUMMY/1 (source: HRtimer) 1" (MASTER). The compilation of zaptel (1.4.12.1) worked fine and detected a TDM412P |
| 13:48.41 | *** join/#asterisk zydoon (n=zydoon@41.225.142.229) |
| 13:48.50 | *** part/#asterisk zydoon (n=zydoon@41.225.142.229) |
| 13:49.38 | [TK]D-Fender | klmpencran: And you shold not be loading ztdummy at all, and a tdm412p implies a 2/1 module match and your configs list all 4 channels |
| 13:50.05 | thieums | [TK]D-Fender: i tryed the g option as well, it's the same behaviour than 'h' extension. When the called phone hangs up, it works, but when the calling phone hangs up, it's game over ^^ |
| 13:51.01 | thieums | perhaps depends of asterisk's release |
| 13:51.27 | klmpencran | [TK]D-Fender, tzafrir_laptop : new pastebin here http://www.mibbit.com/pb/KFayc8. The card is seen on zapte_hardware, wctdm is loaded |
| 13:52.16 | tzafrir_laptop | klmpencran, it is not loaded. The '-' indicates that it is not handled by a module |
| 13:52.21 | [TK]D-Fender | thieums: No. When the calling channel dies, "h" is called. When the called channel hangs up dialplan continues processing if the "g" option is used. |
| 13:52.50 | tzafrir_laptop | try: modprobe wctdm24xxp |
| 13:53.11 | Valmon | probably someone can turn me in right side with my problem |
| 13:53.21 | tzafrir_laptop | (the module is wctdm24xxp , not wctdm) |
| 13:53.34 | Valmon | asterisk+spand + receiving fax over sip |
| 13:53.35 | [TK]D-Fender | tzafrir_laptop: if he's got the latest gen |
| 13:53.40 | Valmon | http://pastebin.com/d3345cfff |
| 13:54.05 | Valmon | silence, no any other action after last line from asterisk |
| 13:54.16 | Valmon | so fax is hangup just on timeout |
| 13:54.35 | tzafrir_laptop | [TK]D-Fender, I didn't have to guess. zaptel_hardware told me :-) |
| 13:55.22 | [TK]D-Fender | tzafrir_laptop: Yup, I really didn't look close on that one at all.... feels like Monday still.. |
| 13:55.35 | klmpencran | tzafrir_laptop : I'm beginning to understand. During bootcamp, the card that was given to us was loaded with wctdm module. We were given two cards TDM400P. We plugged all the green and red modules on one card |
| 13:56.02 | [TK]D-Fender | klmpencran: the New TDM410P mode is built on a different PCI design and uses the TDM2400P driver |
| 13:56.07 | [TK]D-Fender | model* |
| 13:56.25 | [TK]D-Fender | klmpencran: the old TDM400P used wctdm |
| 13:56.36 | *** join/#asterisk intralanman (n=lanman@va-67-76-163-209.sta.embarqhsd.net) |
| 13:57.08 | [TK]D-Fender | tzafrir_laptop: For reference, can you say when the zaptel_hardware app was introduced? I am new to that one... |
| 13:58.05 | tzafrir_laptop | it has been included it zaptel in latest revisions, but its hardware db has been lacking until recently |
| 13:58.20 | tzafrir_laptop | in dahdi it's quite relible, and also in latest zaptel |
| 13:59.24 | [TK]D-Fender | tzafrir_laptop: Thanks for the quick background.... never needed some of these to date. |
| 13:59.48 | tzafrir_laptop | The hardware DB is mostly http://svn.digium.com/svn/zaptel/branches/1.4/kernel/xpp/utils/zconf/Zaptel/Hardware/PCI.pm |
| 13:59.56 | *** part/#asterisk intralanman (n=lanman@va-67-76-163-209.sta.embarqhsd.net) |
| 14:01.06 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:01.47 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 14:03.35 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 14:04.41 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:04.41 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:04.52 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 14:05.03 | *** part/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 14:07.18 | klmpencran | tzafrir_laptop,[TK]D-Fender : well, you were right. The wctdm24xxp module is working fine. Issue is closed. Thank you very much for the help |
| 14:07.22 | *** join/#asterisk n3hxs (n=HAMming@pool-70-110-19-76.washdc.fios.verizon.net) |
| 14:07.56 | [TK]D-Fender | klmpencran: Glad to help where I could |
| 14:10.11 | *** join/#asterisk ziram19 (n=chatzill@196.203.52.254) |
| 14:13.55 | *** join/#asterisk seaq (n=seaq@correo.seaq.com.co) |
| 14:19.37 | *** join/#asterisk ChkDigit (n=mike@24.72.71.175) |
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| 14:24.14 | *** join/#asterisk moy (n=moy@201.54.232.194) |
| 14:25.32 | *** join/#asterisk utrui (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 14:25.37 | utrui | hi |
| 14:25.51 | *** join/#asterisk andrebarbosa (n=andrebar@212.13.49.67) |
| 14:26.34 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 14:26.58 | andrebarbosa | hi |
| 14:27.28 | andrebarbosa | i still cannot detect callerid on AT&T lines with a tdm400 :s |
| 14:27.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:28.10 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 14:28.15 | andrebarbosa | does anyone know if there are big differences with asterisk 1.2 and 1.4 regarding callerid detection? |
| 14:30.34 | utrui | hi |
| 14:30.44 | utrui | i don't if i am in a correct channel |
| 14:30.44 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:31.00 | CGMChris | I am trying to use g729 in pass-thru mode. My phones, provider, and sound files all use g729 format. It appears that incoming calls attempt to transcode the ring tone. For example, with 1 license installed: if I have 2 simultaneous incoming calls, only one caller hears a ring. The other, hears silence, and the CLI reports that it cannot convert from g729 to slin. Thoughts? |
| 14:31.23 | utrui | from long time for a good softphone wich is running on windows |
| 14:31.37 | utrui | working with 3g network |
| 14:31.44 | teknoprep | [TK]D-Fender, how much do you charge to do asterisk consulting ? |
| 14:32.30 | utrui | hello |
| 14:32.57 | seaq | hi all, which is the purpose of a "+" in a dial string ?? |
| 14:33.00 | clintc | utrui: what is your question |
| 14:33.07 | utrui | sorry |
| 14:33.18 | [TK]D-Fender | andrebarbosa: And you proved its there with an analog phone/ |
| 14:33.21 | [TK]D-Fender | ? |
| 14:33.38 | utrui | sip softphone which can be runable on Windows mobile 6 os |
| 14:33.39 | andrebarbosa | Yes |
| 14:33.56 | andrebarbosa | I bought one |
| 14:33.56 | andrebarbosa | it shows the callerid |
| 14:33.56 | andrebarbosa | after the second ring |
| 14:34.02 | andrebarbosa | 2\3seconds before it start to ring |
| 14:34.15 | [TK]D-Fender | utrui: Softphones typically care that you have IP... the fact that 3G is your carrier protocol should not matter |
| 14:34.33 | andrebarbosa | But I don't know how to configure asterisk to detect this type of callerID |
| 14:34.33 | [TK]D-Fender | andrebarbosa: pastebin your current configs |
| 14:35.07 | CGMChris | [TK]D-Fender: Do you know if asterisk attempts to transcode the ring-tone sent to incoming callers, and how to alter that behavior? |
| 14:35.25 | andrebarbosa | I got the message: chan_zap.c:6316 ss_thread: CallerID number: (null), name: (null), flags=12 |
| 14:35.35 | andrebarbosa | just after the first ring.. :s |
| 14:35.38 | clintc | utrui: http://www.eyeball.com/products/mobile_voip.html |
| 14:35.45 | [TK]D-Fender | CGMChris: ringing is indeed generated. |
| 14:36.14 | CGMChris | [TK]D-Fender: Are there sound files somewhere that need to be converted to g729, or do I not get the luxury of ringing in passthru mode? |
| 14:36.33 | [TK]D-Fender | CGMChris: No, they are not sound fines... they are **GENERATED** |
| 14:36.45 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 14:36.48 | sehh | hey people |
| 14:36.49 | CGMChris | [TK]D-Fender: So, the answer is 'no'. Rings do not work in passthru mode. |
| 14:36.53 | [TK]D-Fender | CGMChris: Encoded, on the fly from indications.conf |
| 14:37.01 | sehh | i believe i found a bug, when i do more than 3 transfers of the same call via the phone 'transfer' button (not via '##'), then the call is dropped. The log reports: app_macro.c: Macro(): possible infinite loop detected. Returning early. |
| 14:37.12 | sehh | (asterisk 1.4.21 under CentOS) |
| 14:37.13 | andrebarbosa | !pastebin |
| 14:37.14 | [TK]D-Fender | CGMChris: Alternative : record a long amount of ringing, and set it as the MoH class |
| 14:37.16 | [TK]D-Fender | ~pb |
| 14:37.17 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 14:37.18 | [TK]D-Fender | ^^^^^ |
| 14:37.57 | sehh | i'd appreciate some help |
| 14:38.29 | angryuser | CGMChris: Playtone() ? |
| 14:38.44 | [TK]D-Fender | angryuser: Also generated |
| 14:39.12 | angryuser | [TK]D-Fender: yes, but why not to use that on hte fly ;) |
| 14:39.35 | [TK]D-Fender | angryuser: Because that's his PROBLEM |
| 14:40.12 | CGMChris | Ok, I'll just buy the licenses. Thanks for your help guys. |
| 14:41.54 | sehh | how about my problem? anyone care to take a look? apparently, its a well known bug, it has been reported in the freepbx and trixbox forums, but nobody has found a solution |
| 14:41.59 | sehh | seems like an asterisk bug |
| 14:42.08 | *** join/#asterisk ziram19 (n=chatzill@196.203.52.254) |
| 14:42.12 | *** join/#asterisk andrebarbosa_ (n=andrebar@212.13.49.67) |
| 14:42.23 | andrebarbosa_ | [TK]D-Fender, http://www.pastebin.ca/1273527 |
| 14:43.42 | [TK]D-Fender | sehh: * has a normal macro nesting loop limit of 7 IIRC, If you reach this by 3 then Your GUI's dialplan design is calling things recursively as well making it fail in fewer calls due to its poor design |
| 14:44.46 | sehh | [TK]D-Fender, not really, it seems like an asterisk bug, since asterisk doesn't decrement MACRO_DEPTH |
| 14:45.28 | sehh | i think i found some more info about this from someone named "Corydon76" |
| 14:46.10 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 14:46.18 | [TK]D-Fender | sehh: You'll have to paste complete debug and give the monstrosity that is FreePBX... good luck with that part. Good dialplan design circumvents that |
| 14:46.57 | sehh | the bug isn't related to FreePBX, i'm running plain vanila asterisk. I only said that the bug was REPORTED in the freepbx forums, as well as in trixbox |
| 14:47.44 | [TK]D-Fender | sehh: well then your debug should hopefully be a lot cleaner, but since you are the first person reporting it here I'm thinking that your situation would similar at best |
| 14:47.56 | [TK]D-Fender | sehh: otherwise EVERYBODY in here would be screaming. |
| 14:48.30 | sehh | well, the bug doesn't appear when transfering via asterisk's '##' keys, so not everyone will notice the bug |
| 14:48.41 | sehh | aha! |
| 14:48.53 | sehh | it has already been reported and confirmed as a bug by digium! |
| 14:49.03 | sehh | one sec, let me get the bug id |
| 14:49.25 | sehh | there you go: |
| 14:49.27 | sehh | http://bugs.digium.com/view.php?id=13363 |
| 14:49.51 | sehh | resolution: open, status: confirmed, assigned to: Corydon76 |
| 14:50.01 | sehh | ok at least digium is aware of the bug |
| 14:50.39 | [TK]D-Fender | sehh: Interesting... dead channel stuff.. that explains why few are hit with it |
| 14:50.57 | [TK]D-Fender | sehh: Most of us let channels die in peace |
| 14:51.06 | sehh | heh |
| 14:53.36 | utrui | clintc: sorry for the delay |
| 14:54.03 | utrui | mobile voip messenger that soft not free |
| 14:55.25 | *** join/#asterisk mog (n=mog@nat/digium/x-16cffe3105aaff69) |
| 14:55.26 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:55.44 | sehh | i guess i can rest assured that the bug will be fixed.... in this century :) |
| 14:56.49 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.11) |
| 14:59.28 | jasonwoot | <PROTECTED> |
| 14:59.48 | jasonwoot | my ISP insists that I access the SIP proxy from one of their private IPs |
| 15:00.26 | [TK]D-Fender | jasonwoot: This is serious pain that I can't help with. |
| 15:01.25 | jasonwoot | yes, I'll need opiates to get thru this |
| 15:01.31 | jasonwoot | are you holding? |
| 15:06.23 | andrebarbosa | [TK]D-Fender, did you saw my pastebin? |
| 15:06.27 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:07.28 | [TK]D-Fender | andrebarbosa: remove : cidsignalling=bell cidstart=ring |
| 15:08.10 | andrebarbosa | already did |
| 15:08.17 | andrebarbosa | i'm using asterisk 1.2 |
| 15:08.26 | andrebarbosa | any big changes regarding callerid? |
| 15:08.29 | *** join/#asterisk psy0nid3 (n=IT@bookit-dev.com) |
| 15:08.41 | theskinfox | anybody know much about one touch recording? I have it setup and it works just fine, but I need the ability to play different sound files when the recording stops and starts |
| 15:08.46 | [TK]D-Fender | andrebarbosa: "no comment" |
| 15:09.37 | andrebarbosa | ? why? |
| 15:10.01 | seaq | one question if i want to mark SIP packets should i restart asterisk or with a sip reload would suffice |
| 15:11.25 | |p33| | hi asterisk experts. what is the recommended method to get fax/email gateway to work with asterisk/freepbx? |
| 15:11.45 | jasonwoot | now THAT was a painful project |
| 15:13.05 | |p33| | but it works for you now jasonwoot? |
| 15:13.11 | jasonwoot | |p33|: hylafax gateway |
| 15:14.03 | jasonwoot | |p33|: inbound fax-email is much easier to get working than outbound email-fax, in my opinion |
| 15:14.37 | |p33| | jasonwoot: my main priority is fax to email. |
| 15:16.47 | SkramX | also, I can't figure out how to use ackcall=yes (agents.conf) in my scheme which uses addqueuemember() dynamically |
| 15:16.50 | SkramX | anyone done this?? |
| 15:17.41 | coppice | |p33| unfortunately there isn't one all encompassing super duper fax to e-mail and e-mail to fax add on out there. there are several working solutions, though. for fax to email people usually use simple scripts |
| 15:19.13 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:20.08 | seaq | p33 what i've found most reliable is hylafax + iaxmodem + avantfax |
| 15:20.27 | seaq | i've tried a lot with rx_fax and tx_fax and SOMETIMES works like a charm |
| 15:20.38 | seaq | and MOST of the TIME fails miserably.. |
| 15:20.51 | |p33| | seaq: i did to. i coulnt even compile it |
| 15:21.13 | seaq | i've got to problems, |
| 15:21.14 | seaq | thou |
| 15:21.19 | [TK]D-Fender | SkramX: You can't |
| 15:21.21 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:21.36 | SkramX | [TK]D-Fender - ahhhh. i really want to make this work :\ |
| 15:21.40 | [TK]D-Fender | SkramX: For the tenth time, "M()" <----- |
| 15:21.46 | SkramX | ive tried that |
| 15:21.55 | [TK]D-Fender | SkramX: And you haven't shown us your failure |
| 15:22.02 | seaq | one: the personal fax service is somewhat complicated, with rx_fax during voicemail greeting you could sent a fax. with iaxmodem i couldn't find a way, i've got to create fax extensiones |
| 15:22.14 | seaq | something like 8exten for faxes |
| 15:22.15 | SkramX | ill paste stuff later today when i get into work |
| 15:22.15 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 15:22.51 | seaq | two: outbound fax is complicated, mainly because most of the faxes out there are an extension instead of a direct number |
| 15:22.52 | SkramX | i wish AddQueueMember() just took a context for agents.conf and would use those settings. and yes, I know I have the source code. |
| 15:23.18 | seaq | i couldn't find a way to send the extension number after dialing the main number... |
| 15:24.02 | jasonwoot | seaq: how is the quality of your outbound faxes? I've never been able to tweak it to anything other than a barely acceptable level |
| 15:24.27 | SkramX | [TK]D-Fender: seriously, thanks for your help thus far |
| 15:24.35 | SkramX | im going to give macro a go again then post code here (pastie of course) |
| 15:24.36 | coppice | jasonwoot: are you using iaxmodem or app_rxfax? |
| 15:26.11 | jasonwoot | iaxmodem |
| 15:26.28 | coppice | and what kind of PSTN interface? |
| 15:26.48 | jasonwoot | 2400p |
| 15:26.59 | jasonwoot | I hate copper... |
| 15:27.37 | jasonwoot | trying desperately to get SIP trunking to replace copper trunks, PCI cards are killing us with errors |
| 15:27.59 | coppice | with a PRI iaxmodem should be rock solid for both tx and rx. with analogue interfaces some people get trouble. with SIP or IAX results are anyone's guess |
| 15:28.02 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:28.07 | seaq | jasonwoot: to be honest they work, in fact are better than a fax machine.. |
| 15:28.37 | seaq | jasonwoot: but from time to time they are bad... it seems that is highly dependant to IRQ issues on the server. |
| 15:28.52 | *** join/#asterisk Lazybone_DM (n=Lazybone@dslb-084-061-220-162.pools.arcor-ip.net) |
| 15:28.54 | coppice | unexplainable failure rates with the current iaxmodem are <0.5% |
| 15:28.58 | Lazybone_DM | hiii |
| 15:29.34 | psy0nid3 | konnichi wa |
| 15:29.54 | jasonwoot | seaq: I have yet to find the setting which actually controls the quality, bit depth, etc |
| 15:31.45 | Daejeo | psy0nid: japanse? |
| 15:32.21 | coppice | jasonwoot: the bit depth of what? |
| 15:35.07 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:36.05 | seaq | i believe iaxmodem has ECM activated by default... and i'm not sure about quality setting there.... |
| 15:36.54 | psy0nid3 | daejeo: tashou |
| 15:39.23 | coppice | ECM has nothing to do with iaxmodem. its a hylafax feature. it will be used if the other end understands it. |
| 15:42.35 | psy0nid3 | Daejeo: nihongo desu ka |
| 15:42.55 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-c189500dc34f68db) |
| 15:42.55 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:43.09 | *** part/#asterisk Lazybone_DM (n=Lazybone@dslb-084-061-220-162.pools.arcor-ip.net) |
| 15:43.37 | Daejeo | psy0nid: i only know "konnichi wa" |
| 15:44.05 | coppice | æåä¸èçæ¥èª :-) |
| 15:44.28 | [TK]D-Fender | All of the japanese I know relates to killing people :) |
| 15:45.21 | [TK]D-Fender | And paying respects before sending them to their maker. |
| 15:45.48 | *** join/#asterisk tshine (n=tshine@ip70-160-111-108.hr.hr.cox.net) |
| 15:46.09 | psy0nid3 | ????????????! |
| 15:46.13 | coppice | bye bye works well enough for that |
| 15:47.51 | psy0nid3 | :[TK]D-Fender:kiru |
| 15:48.43 | [TK]D-Fender | psy0nid3: I study Tenshin Shoden Katori Shinto Ryu |
| 15:50.17 | coppice | can you write that in Kanji, so I can see what it means? :-) |
| 15:51.00 | [TK]D-Fender | coppice: 天çæ£ä¼é¦åç¥éæµ |
| 15:51.43 | stintel | ã |
| 15:52.22 | psy0nid3 | ??????? |
| 15:53.05 | alibb | is this #asterisk chanel ? or #japan ? |
| 15:53.10 | [TK]D-Fender | psy0nid3: kobudo |
| 15:53.35 | [TK]D-Fender | alibb: #politics starts in 2 hours, don't be late |
| 15:54.15 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:54.17 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 15:54.21 | Katty | morning |
| 15:54.31 | alibb | [TK]D-Fender, never understand politics ! |
| 15:54.37 | Katty | there a way to see the blacklist from the CLI? |
| 15:54.53 | Katty | i'm getting a weird forbidden message when i call an extension. |
| 15:55.30 | [TK]D-Fender | Katty: "database show" |
| 15:55.39 | [TK]D-Fender | Katty: that app is garbage..... |
| 15:55.52 | [TK]D-Fender | Katty: Forbidden is a SIP auth problem |
| 15:56.01 | [TK]D-Fender | Katty: not blacklist |
| 15:56.04 | Katty | ponders |
| 15:56.13 | Katty | let me pastebin this real quick |
| 15:57.33 | Katty | [TK]D-Fender: so... if that's a sip authentication problem... |
| 15:57.39 | Katty | [TK]D-Fender: but she's the only one we can't call.. |
| 15:57.43 | Katty | [TK]D-Fender: but she can call everyone else. |
| 15:57.52 | Katty | [TK]D-Fender: then i need to have a look at her sip registration settings? |
| 15:58.20 | anonymouz666 | show us the log! |
| 15:58.22 | [TK]D-Fender | Katty: Check to see if there is a phone specific refusal as well |
| 15:58.41 | [TK]D-Fender | Katty: And include the full sip debug next time |
| 16:03.05 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:04.56 | Katty | [TK]D-Fender: found the issue (= |
| 16:06.07 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 16:09.52 | |p33| | hmm will 64bit Centos be a problem when compiling SpanDSP? |
| 16:10.57 | [TK]D-Fender | Katty: And? |
| 16:11.07 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 16:12.49 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:13.06 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
| 16:13.44 | *** join/#asterisk RobertLaptop (n=rmiddle@63.68.135.4) |
| 16:13.52 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 16:16.29 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:18.37 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:20.20 | seaq | hi, i'm having this exact same problem http://www.mail-archive.com/asterisk-users@lists.digium.com/msg213987.html |
| 16:20.38 | seaq | i've marked tos_sip=cs5 but when i make the trace via wireshark |
| 16:20.43 | seaq | i've got a cs3 marking... |
| 16:25.31 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:26.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:31.20 | *** join/#asterisk SiberAIR (n=SibRphre@ip67-93-6-162.z6-93-67.customer.algx.net) |
| 16:31.57 | *** join/#asterisk BrokenNoze (n=magic@host81-137-172-233.in-addr.btopenworld.com) |
| 16:33.26 | BrokenNoze | hi all. does anyone know of any patch diagrams for the sangoma a400 card? i need to crimp it onto a rj45 patch panel but can find nothing as to which wire crimps to which rj45 pin |
| 16:35.50 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 16:42.40 | Qwell | |p33|: it shouldn't be |
| 16:46.00 | seaq | ok, i've found that probably is for runinning asterisk as a normal user... i've got this messsage in the logs |
| 16:46.00 | seaq | rtp.c: Unable to set TOS to 184 |
| 16:53.20 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 16:55.47 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 16:57.49 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 16:58.47 | *** join/#asterisk kam_h (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 16:59.03 | *** join/#asterisk stevie[xxx] (n=stevie@e177126184.adsl.alicedsl.de) |
| 17:00.55 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 17:05.14 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:05.27 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 17:13.56 | *** join/#asterisk psy0nid3 (n=chatzill@bookit-dev.com) |
| 17:14.38 | *** join/#asterisk JonOnt (n=Jon@72.34.90.74) |
| 17:17.59 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 17:18.33 | *** join/#asterisk `paul (n=admin@125.60.241.221) |
| 17:19.48 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 17:20.10 | *** join/#asterisk seaq (n=seaq@correo.seaq.com.co) |
| 17:20.14 | `paul | what seems to be the problem if i clicked answer on x-lite softphone but on asterisk it doesnt register as answered.. (keeps on ringing on calling ..... doesnt even show as answered on asterisk CLI) |
| 17:22.19 | JonOnt | hey guys, I need help to debug this incoming route, when I call my sip number, asterisk plays a recording that the number you have dialed is not in service... heres is a pastebin of the CLI log... http://asterisk.pastebin.com/d67ecd9b8 |
| 17:23.51 | stevie[xxx] | (JonOnt) check ur trunk settings |
| 17:24.17 | stevie[xxx] | Received incoming SIP connection from unknown peer to |
| 17:24.47 | stevie[xxx] | did u add context=from-trunk? |
| 17:26.26 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 17:26.48 | JonOnt | stevie[xxx], extentions.conf ? |
| 17:28.07 | stevie[xxx] | sip.conf suppose |
| 17:28.52 | angryuser | JonOnt: what do you mena by sip number ? you external did ? |
| 17:29.48 | JonOnt | angryuser, yes, thanks your for requesting clarificaton, its my external did from bandwith.com |
| 17:30.14 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 17:30.21 | JonOnt | arg, im using freepbx, sip.conf doesnt have my sip stuff in it |
| 17:32.13 | angryuser | JonOnt: try to asigg some action on incoming call to that trunk |
| 17:32.19 | angryuser | assign* |
| 17:33.46 | angryuser | or add external did +19097406161 to some peer |
| 17:36.25 | fogo | on a zap show status, what is 'bpviol' referring to? |
| 17:36.59 | stevie[xxx] | sip_additional.conf JonOnt |
| 17:39.34 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 17:40.35 | echinos | is there a way to make the CLI put timestamps on it's output? |
| 17:41.53 | *** join/#asterisk psy0nid3 (n=chatzill@bookit-dev.com) |
| 17:43.01 | tzafrir_laptop | echinos, basically, yes. |
| 17:43.16 | tzafrir_laptop | However, consider: tail -f /var/log/asterisk/messages |
| 17:43.33 | raasdnil | anyone know of a provider who will do IAX2 and terminate calls in Taiwan? |
| 17:43.44 | raasdnil | terminate directly onto the PSTN that is |
| 17:50.28 | hi365 | anyone auto provisioning snom's? |
| 17:53.38 | JonOnt | stevie[xxx], : hey, does this sound right, my trunk name is Bandwith, and the context for the incoming is from-trunk-Bandwith |
| 17:54.25 | *** join/#asterisk jtodd (n=jtodd@nat/digium/x-2ddd5a5efc4c6812) |
| 17:55.31 | stevie[xxx] | no |
| 17:55.50 | stevie[xxx] | just use from-trunk |
| 17:56.05 | stevie[xxx] | this is a setting for the freepbx enviroment |
| 17:56.28 | stevie[xxx] | if u want to handle it by urself u can choose ur setting |
| 17:56.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:58.13 | JonOnt | wow, my phone just froze |
| 17:58.15 | psy0nid3 | . |
| 17:59.29 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 18:00.52 | echinos | tzafrir_laptop: yeah, I had thought of that, but then I don't have the debug on/off control of what I see |
| 18:01.00 | echinos | I guess I could grep somehow |
| 18:01.12 | tzafrir_laptop | echinos, sure you do |
| 18:01.19 | tzafrir_laptop | in the cli. |
| 18:01.44 | tzafrir_laptop | in a different terminal |
| 18:01.44 | echinos | oh, the messages file reflectsthe debug options? Cool. |
| 18:01.44 | tzafrir_laptop | tail -f /var/log/asterisk/messages | grep whatever |
| 18:01.50 | echinos | I thought it would dump everything. never looked. |
| 18:01.53 | echinos | ducks |
| 18:02.05 | tzafrir_laptop | echinos, see /etc/asterisk/logger.conf and the output of 'logger show channels' |
| 18:02.24 | echinos | sounds good, tyvm |
| 18:02.33 | *** join/#asterisk LND (n=LND@89.193.181.22) |
| 18:04.33 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 18:08.17 | *** join/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 18:14.52 | *** join/#asterisk rednul (n=rednulG5@216.187.133.50) |
| 18:16.48 | rednul | what is the common dial plan setup for picking different outbound lines? something like 01#<number> and 02#<number> etc? or something else? |
| 18:18.06 | *** join/#asterisk JonOnt (n=Jon@72.34.90.74) |
| 18:18.23 | JonOnt | ok, im back in, my phone crashed |
| 18:19.13 | JonOnt | Ok, guys, what do I do next? |
| 18:19.29 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:21.37 | Kobaz | http://pastebin.ca/1273732 |
| 18:21.40 | Kobaz | why would that not work? |
| 18:21.41 | JonOnt | - Executing [+19097406161@from-sip-external:1] NoOp("SIP/4.68.250.148-ac07d5e0", "Received incoming SIP connection from unknown peer to +19097406161") in new stack |
| 18:21.50 | JonOnt | Unknown peer |
| 18:21.51 | Kobaz | 2602 rings, but 2601 won't |
| 18:21.57 | JonOnt | What does that mean? |
| 18:22.31 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:24.23 | *** join/#asterisk synchris (n=synchris@athedsl-4381731.home.otenet.gr) |
| 18:24.41 | agile | now I'm really stumped. Created a replacement asterisk server but had issues where I wasn't getting dtmf with agi's GET DATA call, tried various versions of asterisk (1.4.17, 1.4.21, 1.4.22) and always the same issue.. |
| 18:25.07 | agile | so I moved everything back to the old server, and now I have the same issue there, too.. |
| 18:25.20 | agile | possibly my sip provider? |
| 18:25.26 | *** join/#asterisk stevie[xxx] (n=stevie@e177155175.adsl.alicedsl.de) |
| 18:26.36 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 18:27.08 | UnixDawg | asterisk does mgcp trunking correct ? |
| 18:27.20 | *** join/#asterisk key2 (n=key2@olm03.cvf.fr) |
| 18:27.28 | key2 | is there any asterisk operator switchboard that rocks ? |
| 18:27.43 | UnixDawg | isyphony |
| 18:28.30 | key2 | mmh |
| 18:29.44 | key2 | also, my meetme conf dont work for some reason |
| 18:30.01 | key2 | <PROTECTED> |
| 18:30.01 | key2 | <PROTECTED> |
| 18:30.01 | key2 | <PROTECTED> |
| 18:30.01 | key2 | <PROTECTED> |
| 18:30.06 | key2 | i get this |
| 18:30.11 | key2 | any idea ? |
| 18:33.00 | Kobaz | why wouldn't I be able to go a Goto() from within a Macro() ? |
| 18:33.06 | Kobaz | er i mean |
| 18:33.13 | Kobaz | Goto() within a macro definition |
| 18:35.21 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 18:35.23 | ZaVoid | hey guys |
| 18:36.16 | ZaVoid | trying to figure out the channel variable so i can grab the username registered via an iax channel |
| 18:36.36 | ZaVoid | and not ${CALLERID(name)} or num |
| 18:37.08 | ZaVoid | any thoughts? |
| 18:38.19 | Kobaz | ZaVoid: use setvar= in iax.conf |
| 18:38.42 | ZaVoid | how does that help? each phone registering via iax will have a different username |
| 18:38.57 | ZaVoid | i want to pull out that info |
| 18:39.01 | Kobaz | like, i have: [2604] username=2604 ....etc... setvar=ExtenPhoneNum=2604 |
| 18:39.17 | ZaVoid | yeah but i'm not definiing the username |
| 18:39.22 | ZaVoid | i use realtime |
| 18:39.27 | ZaVoid | so it authenticates via realtime |
| 18:39.29 | Kobaz | so no matter what the callerid is, i always have a var that gives me the username |
| 18:39.37 | Kobaz | you still have a username if you use realtimer |
| 18:39.42 | Yourname` | Like autopause in queues.conf, which other setting actually logged out the agent instead of pausing when the agent didn't take a call? |
| 18:39.48 | seanbright | ZaVoid: core show function IAXPEER |
| 18:39.55 | ZaVoid | right but i need to definite it in a variable so when i dial out i can use it in sip |
| 18:40.21 | seanbright | hmmm, that might not have the info you need... |
| 18:40.24 | *** join/#asterisk sprite-- (n=sprite@12.228.1.97) |
| 18:40.24 | ZaVoid | sean that won't work |
| 18:40.45 | ZaVoid | lets say for example.... just talking theory ok? you register with iax account 1111 and pass 111 |
| 18:40.53 | ZaVoid | but you dont set/define callerid settings |
| 18:41.00 | seanbright | right, you're screwed. |
| 18:41.03 | ZaVoid | then i still need to be able to put 1111 |
| 18:41.06 | ZaVoid | lol |
| 18:41.15 | seanbright | are you averse to changing asterisk code? |
| 18:41.20 | JonOnt | guys, how do I set up a trunk wirth more than one host ip? |
| 18:41.26 | ZaVoid | i enjoy drinking yes sean |
| 18:41.51 | seanbright | eh? |
| 18:43.04 | sprite-- | Hey! I am doing a site similar to niteflirt.com. I need to be able to call 2 parties, bridge the connections, but keep them in seperate contexts. I am planning on doing the webpage in RoR and it looks like I can easily intergrate it with adhearsion. Does the Asterisk 1.6 bridge support what I need to do? If not what is the best way to do it? From what I read MeetMe conference is not very performance effective. What is the best way to handle a large number |
| 18:43.21 | Yourname` | Or is it autopause only now? |
| 18:44.10 | Yourname` | Can't seem to find it in queues.conf.sample |
| 18:44.42 | Kobaz | sprite--: phone sex |
| 18:44.44 | *** join/#asterisk digime (n=digi@adsl-75-3-206-197.dsl.sndg02.sbcglobal.net) |
| 18:44.45 | Kobaz | sprite--: ? |
| 18:45.41 | sprite-- | Kobaz: yeah pretty much |
| 18:46.17 | digime | hi, i have ip phones running through a sonicwall. the issue is if someone calls me, they CAN hear me but I CAN'T hear them. This happens on any IP phone. I changed the SIP maxexpirey in asterisk to 3600 (to match the setting on the sonicwall). Could this be why this is happening? |
| 18:46.23 | sprite-- | I would like to support ~200 concurrent conversations. An Asterisk cluster seems to be the way to go. |
| 18:47.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:47.59 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-147-101.telkomadsl.co.za) |
| 18:48.13 | sprite-- | Should I perhaps build off app_conference? Trying to find the best solution before I get started on anything. I have a decent amount C/C++ programming experience on Windows, but have done zero linux programming before. |
| 18:52.02 | |p33| | digime: in some nated fw you have to create static outbound mapping |
| 18:52.09 | Kobaz | sprite--: do you have a good supply of "operators" |
| 18:52.40 | Qwell | Kobaz: looking for a job? O.o |
| 18:52.52 | xacatecas | hi guys, i'm busy with a custom cdr backend, my question is simple, since I _need_ to connect to some other system via tcp, if that system isn't available I'd like to keep retrying until it is available (ie, make the call to the cdr handler "block", in a way. is this safe? or can I set a timer of sorts to call-back at a later stage (and make a copy of the cdr struct - and how?) |
| 18:53.11 | mm2knet | has anyone yet compiled asterisk 1.6.0.2? |
| 18:53.23 | Qwell | mm2knet: ask a more specific question, and people can help you |
| 18:53.34 | lmadsen | digime: I've never ever had any lucky getting VoIP working through sonicwalls and gave up on them |
| 18:53.40 | mm2knet | i get the following error: |
| 18:53.41 | mm2knet | manager.c: In function âaction_getvarâ: |
| 18:53.41 | mm2knet | manager.c:1732: error: âSENTINELâ undeclared (first use in this function) |
| 18:54.03 | mm2knet | google knows nothing -.- |
| 18:54.05 | Qwell | much better |
| 18:54.21 | mm2knet | ^^ |
| 18:55.03 | sprite-- | How does the Bridge cmd in Asterisk 1.6 work? I can't find any documentation on it. |
| 18:55.06 | digime | <@lmadsen: yeah |
| 18:55.35 | digime | talking to sonicwall support right now |
| 18:56.33 | digime | |p33|: for what |
| 18:56.52 | |p33| | digime: have you created a manual outbound nat rule? |
| 18:56.53 | digime | note: i have sip transformations turned on.. that handles signalling, timeouts, etc. with natting.. |
| 18:57.13 | mm2knet | my system is Debian etch x64 |
| 18:57.38 | Qwell | mm2knet: How did you download that? |
| 18:57.48 | digime | i have the sip service specified in outbound rules. i found that my asterisk box sip signalling/timeout was set at 300, and the sonicwall at 3600. so i changed asterisk to 3600. |
| 18:57.56 | mm2knet | wget http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz |
| 18:58.03 | agile | what sip providers do you guys <3 these days? |
| 18:58.07 | digime | i have a lan to wan access rule for bandwidth allocation. maybe i will eliminate that first |
| 18:58.10 | lmadsen | agile: Unlimitel |
| 18:58.31 | digime | |p33|: sip transformations should handle that |
| 18:58.34 | lmadsen | agile: they actually own their own network and don't resell Verizon, Level3, DIDX, etc... like all the other providers |
| 19:00.23 | Qwell | mm2knet: not even seeing SENTINEL usage in manager.c... |
| 19:00.55 | |p33| | digime: perhaps, i had that exact issue with pfsense (bsd based) and manual outbound rules solved it. |
| 19:01.20 | Qwell | mm2knet: can you show me line 1732 of main/manager.c? |
| 19:01.33 | mm2knet | y, just a sec |
| 19:01.36 | Yourname` | Oh it was in agents.conf in case anyone is wondering. However, it doesn't work.. autologoff=1 |
| 19:01.38 | Yourname` | 15 |
| 19:02.17 | Yourname` | autopause works however, just not autologoff |
| 19:03.47 | mm2knet | c = ast_channel_alloc(0, 0, "", "", "", "", "", 0, "Bogus/%p", SENTINEL); |
| 19:04.24 | jasonwoot | digime: can asterisk see SIP trunks via a 2nd NIC, or must I do it via router? |
| 19:04.34 | digime | |p33|: you mean it resolved the issue whereby you couldn't hear someone calling into you, but they could hear you? |
| 19:05.24 | seanbright | Qwell: that's been fixed in the branch already methinks |
| 19:05.36 | Qwell | mm2knet: It is broken in that release. We're going to have to respin it most likely. |
| 19:05.54 | mm2knet | ok |
| 19:06.15 | Qwell | seanbright: mmhmm |
| 19:06.15 | mm2knet | hsould i continue with 1.6.0.1? or is there an easy pathc? |
| 19:06.47 | digime | jasonwoot: dunno? |
| 19:06.49 | *** part/#asterisk killfill (n=killfill@200.63.96.244) |
| 19:06.53 | seanbright | mm2knet: change to: |
| 19:07.00 | seanbright | c = ast_channel_alloc(0, 0, "", "", "", "", "", 0, "Bogus/manager"); |
| 19:07.27 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:08.03 | mm2knet | ok, works now |
| 19:08.05 | mm2knet | ty :) |
| 19:08.16 | seanbright | you're welcome. tip your waitress. |
| 19:08.31 | *** part/#asterisk ixx (n=ixx@70.114.153.233) |
| 19:08.34 | Katty | tips seanbright |
| 19:08.44 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 19:11.53 | *** part/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 19:12.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:12.39 | *** join/#asterisk stevie[xxx] (n=stevie@g224123230.adsl.alicedsl.de) |
| 19:13.05 | stevie[xxx] | does asterisk support "comfort noise"? |
| 19:15.44 | farkus | I wish cellphones had comfort noise. I'm constantly asking "are you still there?" like every minute or two. they're quiet. too quiet. |
| 19:16.20 | stevie[xxx] | im using asterisk 1.6.0.1, i dont find some good docus for comfort noise and asterisk |
| 19:16.38 | *** join/#asterisk xheliox (n=jeff@fl-71-55-214-178.dhcp.embarqhsd.net) |
| 19:17.55 | *** join/#asterisk ddfire (n=dfaerman@201.216.241.193) |
| 19:18.01 | ddfire | hi |
| 19:18.03 | xheliox | Hey all.. quick question, perhaps someone can help. Using res_cepstral (SayText).. I want to insert things like <break strength='medium' /> into the text, but the text always stops playing at that point. Is there some sort of escape character I should be using before and after it? |
| 19:19.46 | ddfire | question, i cant find the way to get the state of the peers, if they are in a call and whit who from the manager.... |
| 19:19.53 | ddfire | how i get that info |
| 19:20.02 | ddfire | the idea es make a callcenter panel control |
| 19:21.10 | ddfire | any one???? |
| 19:24.32 | seanbright | what kind of peers? |
| 19:24.33 | seanbright | SIP? |
| 19:24.48 | ddfire | sip or/and iax |
| 19:24.56 | seanbright | there is a SIPshowpeer manager action |
| 19:25.07 | ddfire | yes and it give very poor info |
| 19:25.59 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 19:26.02 | ruben23 | hi to all |
| 19:26.03 | ddfire | i can paste it here i will be kicked |
| 19:26.06 | ddfire | hi ruben23 |
| 19:26.23 | seanbright | ddfire: you don't need to paste it, and if you did you would paste it to a pastebin |
| 19:26.28 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:26.52 | ddfire | seanbright: have any address of a pastebin>? |
| 19:27.03 | seanbright | ddfire: /msg jbot pb |
| 19:27.31 | farkus | stevie[xxx]: main/rtp.c:781 shows incomplete support for comfort noise generation |
| 19:27.45 | ddfire | lol too much techno |
| 19:28.17 | stevie[xxx] | farkus , which asterisk version do u refer? |
| 19:29.05 | farkus | stevie: sorry, I was looking at 1.4.21.2 -- I'll look at 1.6 |
| 19:29.31 | ddfire | http://pastebin.com/de322ca9 |
| 19:29.51 | ddfire | its show very poor info about the current activity |
| 19:30.09 | seanbright | ddfire: the Status AMI action is probably what you want |
| 19:30.23 | farkus | stevie: it' |
| 19:30.28 | farkus | oops |
| 19:30.35 | ddfire | seanbright: sorry but i dont understand |
| 19:30.39 | farkus | stevie: it's main/rtp.c:1003 |
| 19:30.55 | digime | where i can find out what RTP ports asterisk is using? |
| 19:31.12 | seanbright | ddfire: in the asterisk CLI type: manager show command Status |
| 19:31.19 | ddfire | ho |
| 19:32.14 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 19:32.21 | stevie[xxx] | farkus, thx, so there are not settings to be made to enable it? |
| 19:33.15 | farkus | stevie: it's not a matter of settings. there doesn't appear to be a complete implementation of the technology |
| 19:33.30 | *** join/#asterisk sprite-- (n=sprite@12.228.1.97) |
| 19:34.03 | stevie[xxx] | ok, so i think my settings for rxgain and txgain is too low, because i thought this came from asterisk |
| 19:35.05 | farkus | there appear to be cards that support it, according to a google search for "comfort noise asterisk" |
| 19:35.14 | stevie[xxx] | yeah |
| 19:35.15 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 19:36.14 | ddfire | seanbright: thanks you very much thas what i was looking for |
| 19:36.20 | seanbright | ddfire: yup |
| 19:37.34 | digime | found! rtp.conf |
| 19:37.47 | ddfire | seanbright: by the way what is a better idea, have many supervisors pc connected to the manager or only one whit the astrisk proxi? |
| 19:38.04 | seanbright | ddfire: the latter |
| 19:39.43 | digime | are RTP ports TCP or UDP? |
| 19:40.12 | jaytee | udp |
| 19:41.57 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 19:44.03 | jaytee | jbot seen ManxPower |
| 19:44.04 | jbot | manxpower <n=manxpowe@221.sub-75-251-108.myvzw.com> was last seen on IRC in channel #asterisk-dev, 1h 59m 19s ago, saying: 'Maybe I'm just spoiled by TMPGEnc eXpress.'. |
| 19:44.37 | sprite-- | What has better performance? app_confcall or app_conference? |
| 19:46.17 | sprite-- | err app_confcall is only for 1.2? :( |
| 19:47.21 | *** join/#asterisk stevie[xxx] (n=stevie@e177123084.adsl.alicedsl.de) |
| 19:48.30 | jaytee | sprite--, if you only need 3 way calling then the conference function of most voip phones will suffice and use less resources on the * server than using MeetMe. |
| 19:48.37 | stevie[xxx] | i got an issue with my DID incoming numbers, all incoming calls have the main DID number for any number of my DID block. How can asterisk get the right incoming DID? |
| 19:49.31 | jaytee | stevie[xxx], these are analog and not PRI I'm guessing? |
| 19:50.00 | stevie[xxx] | analog means in a block? |
| 19:50.19 | stevie[xxx] | i got number 0815+0-99 block |
| 19:50.28 | jaytee | no, analog means analog, PRI is digital |
| 19:50.37 | stevie[xxx] | no its PRI |
| 19:51.06 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:51.25 | jaytee | stevie, where are you located? |
| 19:51.35 | *** part/#asterisk xheliox (n=jeff@fl-71-55-214-178.dhcp.embarqhsd.net) |
| 19:51.45 | stevie[xxx] | germany |
| 19:51.51 | ddfire | people thanks you all |
| 19:55.57 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:57.46 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:58.02 | [TK]D-Fender | stevie[xxx]: DID's do not coming in as the pilot # |
| 19:58.22 | bijit | is there another way to see last reload or restart apart from logs? |
| 19:58.23 | [TK]D-Fender | stevie[xxx]: they land on the PRI associated with it, but the # should always be visible as the target |
| 19:58.53 | stevie[xxx] | i dont understand |
| 19:59.24 | jaytee | stevie[xxx], you need to call the provider of your PRI |
| 19:59.42 | [TK]D-Fender | stevie[xxx]: You need to stop worrying and look at your PRI debug |
| 19:59.55 | stevie[xxx] | this is no configuration fault of mine? |
| 20:00.01 | [TK]D-Fender | stevie[xxx]: I'm betting you're neurosing over nothing. |
| 20:00.16 | [TK]D-Fender | stevie[xxx]: You haven't shown us anything <- |
| 20:00.25 | [TK]D-Fender | (since my arrival anyways) |
| 20:00.45 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:03.40 | *** join/#asterisk Howie69 (n=Owner@host-12-173-142-207.nctv.com) |
| 20:03.56 | stevie[xxx] | ill try to get the dumb |
| 20:09.04 | digime | What protocol do RTP ports use in asterisk? I assume it's UDP but RTP ports can use other protocols... |
| 20:11.05 | jtodd | UDP, yes. |
| 20:12.56 | digime | besides the SIP signalling ports 5060-5061, and the RTP ports specified in rtp.conf, does asterisk use any other ports? |
| 20:16.44 | stevie[xxx] | [TK]D-Fender http://nopaste.org/p/aNydr0fqV |
| 20:18.08 | stevie[xxx] | i didnt called 69666719 i called 696667192 |
| 20:18.24 | stevie[xxx] | the DID block isnt transmitted |
| 20:23.19 | ruben23 | hi can i ask if what is SVN...? |
| 20:23.55 | lmadsen | subversion |
| 20:24.10 | lmadsen | it is a code management system |
| 20:28.26 | ruben23 | ok im having hardtime doing this.....during installation of asterisk 1.4.22 do i need to install also SVN.? |
| 20:28.42 | *** join/#asterisk af_ (n=getsmart@88-149-241-38.dynamic.ngi.it) |
| 20:28.48 | ruben23 | - svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 20:29.00 | ruben23 | its not working |
| 20:29.07 | kb3ien | will _1NXXNXXXXXX match before 17185551212 if they are competing in the same dialplan? |
| 20:30.01 | kb3ien | i was hoping bypass the outbound trucks for any call that should return to my own pbx. |
| 20:30.40 | ruben23 | the command is not valid.. |
| 20:31.36 | kb3ien | oops rather, inbound is a context included from outbound. outbound has '_1NXXNXXXXXX' inbound has 17185551212. the call is currently in the outbound context. how do i make 17185551212 win ? |
| 20:32.26 | kb3ien | naturally i dont want to include the outbound context in the inbound context. i'd open myself up for much abuse... |
| 20:33.33 | *** join/#asterisk pecanha (n=e@189.106.103.88) |
| 20:34.15 | pecanha | hello, can anyone give me some help. I can talk between my extensions, but when someone calls me it's not finding any extension available. |
| 20:34.31 | [TK]D-Fender | stevie[xxx]: Could be they are sending the trailing part of your block as DTMF. I ahve seen this before on Euroisdn with overlap dialing |
| 20:34.32 | hardwire | so if I'm just bridging from a channel bank, through asterisk via a 2 port PRI card, to a telco PRI, would you think I'd want echo cancellation? |
| 20:34.40 | hardwire | or at least onboard echo can? |
| 20:34.53 | hardwire | I suppose it all depends on the telco's network.. |
| 20:34.57 | Alan_Hicks | Update on my odd echo/static problem. |
| 20:35.06 | [TK]D-Fender | hardwire: Yes, if you know whats good for you |
| 20:35.11 | stevie[xxx] | [TK]D-Fender so what do i have to do? |
| 20:35.44 | hardwire | [TK]D-Fender: thats not what I wanted to hear! |
| 20:35.52 | [TK]D-Fender | stevie[xxx]: First confirm with your telco, second check your PRIDIALPLAN settings as it is clearly trunating a digit |
| 20:35.53 | hardwire | would you mind changing your reply? |
| 20:35.55 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 20:36.04 | Alan_Hicks | If y'all recall from yesteday, I've got a box that gets echo and/or static on all phone calls over the PSTN (not straight SIP or SIP->IAX2->SIP). This wasn't fixed by changing the TDM card (even cahnged brands) or by changing the echo canceller. |
| 20:36.28 | [TK]D-Fender | hardwire: Ok, how's "Get HWEC or you'll regret it" sound instead? |
| 20:36.34 | hardwire | god I'm losing money on this deal. |
| 20:36.50 | Alan_Hicks | I put another asterisk box in its place as a temporary server and there are no problems with it, even though it's using the very same card the other server used. So I guess I'm looking at some sort of weird hardware bug in my server. |
| 20:36.51 | stevie[xxx] | [TK]D-Fender , PRIDIALPLAN, where is this lcoated? or do u men my capi.conf? |
| 20:37.30 | [TK]D-Fender | stevie[xxx]: I was hoping for Zaptel, but then again, this started off being a PRI conversation which has turned to BRI |
| 20:37.37 | [TK]D-Fender | stevie[xxx]: Start by talking with your telco |
| 20:37.52 | stevie[xxx] | yeah, i am trying atm |
| 20:38.17 | [TK]D-Fender | steAnd what you have seems to be BRI, not PRI. please don't confuse the two |
| 20:38.29 | stevie[xxx] | i sorry -.- |
| 20:38.32 | kb3ien | i think this one comes down to good ol' #include again... |
| 20:39.26 | *** join/#asterisk macno (n=michele@adsl-150-160.adsl.metrolink.it) |
| 20:46.17 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 20:46.29 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 20:46.40 | ruben23 | hi |
| 20:46.50 | digime | besides the SIP signalling ports 5060-5061, and the RTP ports specified in rtp.conf, does asterisk use any other ports? |
| 20:47.04 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 20:47.22 | digime | My phone uses RTP port 3000 in its config, but if I set the RTP ports in asterisk to 10,000, does it ignore the phone? |
| 20:47.49 | macno | Hi all, I can register my Nokia E71 but I can't do or receive call... |
| 20:47.54 | file | no, there are two medium streams going on... one from Asterisk to the phone and one from the phone to Asterisk |
| 20:47.56 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 20:48.04 | file | rtp.conf controls the ports Asterisk will use and tell the phone to send to |
| 20:48.25 | macno | no nat, no firewall i'm just another subnet |
| 20:48.32 | file | Asterisk will send to whatever the phone says to |
| 20:48.34 | digime | @file: thanks. long time no talk. |
| 20:49.00 | x86 | hey guys chan_dahdi is not seeing my card properly |
| 20:49.10 | digime | @file: so in theory i should have the rtp ports on the phone match the config in asterisk's rtp.conf |
| 20:49.17 | x86 | I'm trying to use it like zaptel, using /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 20:49.34 | file | digime: it does not matter |
| 20:49.38 | [TK]D-Fender | x86: don't |
| 20:49.40 | x86 | zap show channels from the CLI is not showing me any of the channels on my TDM402E |
| 20:49.57 | file | digime: Asterisk looks at the SDP for the IP address and port to send the audio stream to, it uses whatever the device (in this case your phone) tells it to |
| 20:50.02 | digime | ok |
| 20:50.08 | [TK]D-Fender | x86: use the new proper config files and back it up for us |
| 20:50.08 | file | of course this all goes out the window with symmetric RTP |
| 20:50.24 | x86 | [TK]D-Fender: it says it's still compatible (1.4.22) |
| 20:50.39 | [TK]D-Fender | x86: Stop asking for trouble |
| 20:50.45 | x86 | hmm, ok... |
| 20:50.52 | x86 | is there a transition guide? |
| 20:51.13 | Kobaz | 2008-12-02 15:49:33 EST LOG: statement: SELECT * FROM asterisk.voicemail WHERE mailbox = E'2601' AND context = E'internal' |
| 20:51.17 | Kobaz | asterisk is sending that to postgres |
| 20:51.21 | Kobaz | what's up with the E? |
| 20:51.32 | *** part/#asterisk Alan_Hicks (i=alan@cardinal.lizella.net) |
| 20:51.53 | stevie[xxx] | [TK]D-Fender my provider said this should be a configuraitone rror of my TK |
| 20:51.53 | x86 | [TK]D-Fender: actually, I'm using zaptel 1.4.12.1 |
| 20:51.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:52.04 | x86 | [TK]D-Fender: dahdi-linux-all or whatever isn't installed at all... |
| 20:52.22 | x86 | [TK]D-Fender: ztcfg -vvv shows both channels just fine |
| 20:52.31 | digime | @file: so if Asterisk looks at the SDP for the IP address and port to send the audio stream to, then the RDP port on the phone doesn't matter. |
| 20:52.31 | [TK]D-Fender | stevie[xxx]: they say they send all the digits in the direct transmission? not post-answer DTMF? |
| 20:52.43 | [TK]D-Fender | x86: Please make up your mind then. |
| 20:52.54 | file | digime: well it matters in that you need to have one set... but the value doesn't matter |
| 20:53.06 | digime | ok |
| 20:53.11 | stevie[xxx] | no, and i figured out my error |
| 20:53.20 | x86 | [TK]D-Fender: I want to use chan_zap with zaptel |
| 20:53.27 | stevie[xxx] | i didnt set up correctly msn=XXX in my capi.conf |
| 20:53.41 | stevie[xxx] | i have to set up each DID |
| 20:53.43 | digime | so asterisk will only use the sip signalling ports and the RTP ports as I have configured, no other ports? I am setting up qos on my firewall |
| 20:54.23 | Kobaz | er, never mind |
| 20:54.25 | [TK]D-Fender | x86: Stop talking in tiny unusable little bits and pastebin the whole damn mess |
| 20:54.33 | file | digime: correct |
| 20:54.59 | x86 | [TK]D-Fender: can I use chan_dahdi with zaptel libraries? chan_zap.so doesn't exist it seems |
| 20:55.19 | [TK]D-Fender | x86: How can you use DAHDI when you installed ZAPTEL |
| 20:55.34 | [TK]D-Fender | x86: on eor the other. Period. |
| 20:55.35 | x86 | [TK]D-Fender: because they're backwards-compatible? |
| 20:55.42 | stevie[xxx] | no, this wasnt it |
| 20:56.02 | [TK]D-Fender | x86: NO.l Strop screwing stuff up and do a COMPLETE job on ONE |
| 20:56.10 | digime | @file: also, the firewall has a SIP Signaling inactivity time out (seconds) of 3600. This is the default. But my sip extensions were set much less and I was having different issues. I set the timeouts in sip.conf to the same 3600. Does this make sense? |
| 20:56.17 | x86 | no chan_zap in the asterisk source tree anywhere, nor in the zaptel source tree |
| 20:56.34 | x86 | [TK]D-Fender: dude, I'm running asterisk 1.4.22 that has no chan_zap |
| 20:56.35 | digime | @file: it seems to me that the firewall SIP timeout and asterisk SIP timeout should be the same |
| 20:56.45 | file | digime: without knowing more about your setup I can't answer that, but I would be considered if your firewall even knows about SIP... often times they mess the signalling up |
| 20:56.58 | [TK]D-Fender | x86: Should have a chan_zap |
| 20:57.02 | [TK]D-Fender | n't* |
| 20:57.05 | x86 | [TK]D-Fender: would just migrate to dahdi but can't find it on downloads.digium.com |
| 20:57.16 | [TK]D-Fender | x86: oh WTF... |
| 20:57.23 | *** join/#asterisk rig (n=hrm@tony11-128-74.inter.net.il) |
| 20:57.26 | digime | @file: ok |
| 20:57.30 | rig | hello |
| 20:57.41 | [TK]D-Fender | x86: http://www.asterisk.org/downloads |
| 20:57.46 | rig | anyone here using a phillips ap200? |
| 20:58.24 | digime | @file: Is 802.1p worth exploring? I have everything set by bandwidth priority now, but I can also tag the packets with 802.1p Marking Actions (preserve/explicit/map). Is 802.1p recommended? |
| 20:58.58 | file | unfortunately I must now return to doing work |
| 20:59.04 | Howie69 | digime: what are you using for your QoS? |
| 20:59.15 | digime | i am using a sonicwall |
| 20:59.21 | Howie69 | ah |
| 20:59.22 | digime | i think i got it working |
| 20:59.34 | Howie69 | I was trying to use mastershaper on a linux box or QoS and routing of the network |
| 20:59.43 | Howie69 | but, having trouble getting it to work :) |
| 20:59.44 | x86 | [TK]D-Fender: it sucks that they claim it's backwards-compatible and it's not... |
| 20:59.48 | Howie69 | seems it does not want to route |
| 20:59.56 | digime | yeah |
| 21:00.13 | [TK]D-Fender | x86: No, you're jsut not reading the docs. |
| 21:00.21 | rig | how about the aastra sip-dect thing? anyone using that? |
| 21:00.34 | [TK]D-Fender | x86: nobody said it used chan_zap, it can simply fake out some of the syntax |
| 21:00.53 | [TK]D-Fender | x86: Now stop doing a half-assed job and jsut migrate over properly. |
| 21:00.56 | x86 | [TK]D-Fender: what part did i miss? I've got the drivers (zaptel) installed properly, I've got the 'zap' command in the cli, ztcfg -vvv shows my channels configured correctly.... |
| 21:01.04 | [TK]D-Fender | x86: Yuo seem to fish for trouble at every turn. |
| 21:01.14 | x86 | [TK]D-Fender: I never said it used chan_zap either |
| 21:01.32 | bijit | confusing... |
| 21:01.54 | x86 | [TK]D-Fender: i'm ripping out the zaptel stuff right now and installing dahdi |
| 21:02.38 | harry_v | x86, guess you didnt know like me :) |
| 21:03.06 | x86 | right |
| 21:03.16 | x86 | first time I've been forced to use dahdi |
| 21:04.38 | rig | is there a way to get asterisk to ignore a # when trying to transfer calls? it seems to think that a hash sign signifies the end of the number |
| 21:04.39 | stevie[xxx] | [TK]D-Fender direct transmission |
| 21:04.48 | stevie[xxx] | but can be switched |
| 21:04.56 | jasonwoot | rig: try the metric system |
| 21:05.04 | rig | ? |
| 21:05.09 | jasonwoot | royale with cheese |
| 21:05.21 | x86 | [TK]D-Fender: where do i get some default dahdi configs from? |
| 21:05.23 | rig | ah. a joke. how droll. |
| 21:05.39 | jasonwoot | i kid, i kid |
| 21:05.59 | stevie[xxx] | [TK]D-Fender should i prefe DTMF transmision or direct? |
| 21:06.02 | stevie[xxx] | prefer |
| 21:07.01 | rig | so? |
| 21:08.56 | Kobaz | ftp is tehsux |
| 21:10.32 | x86 | [TK]D-Fender: ok, switched over to dahdi, and dahdi show channels still only shows a single channel called pseudo |
| 21:11.31 | [TK]D-Fender | x86: And you haven't pastebin'd a thing as usual. |
| 21:11.40 | x86 | http://pastebin.ca/1274050 |
| 21:11.47 | x86 | [TK]D-Fender: was working on it |
| 21:11.55 | [TK]D-Fender | rig: on what phone? |
| 21:12.13 | rig | any phone |
| 21:12.24 | x86 | http://pastebin.ca/1274051 |
| 21:12.31 | [TK]D-Fender | rig: What are you using? |
| 21:12.34 | rig | i have a couple of snoms, an aastra, sipura/linksys, what have you |
| 21:12.37 | rig | all behave the same |
| 21:12.44 | rig | i think it's in the pbx |
| 21:13.05 | [TK]D-Fender | rig: You should not be using DTMF transfers on ANY of these phones |
| 21:13.38 | [TK]D-Fender | x86: And you're missing a config file <- |
| 21:13.56 | x86 | which? |
| 21:14.05 | x86 | and why is it still trying to use zapata.conf?! |
| 21:14.14 | x86 | Dec 2 15:13:49 asterisk asterisk[11332]: ERROR[11332]: chan_dahdi.c:11696 in setup_dahdi: Unable to load zapata.conf |
| 21:14.35 | *** join/#asterisk invalidrecord (n=fares@87.114.172.187.plusnet.thn-ag1.dyn.plus.net) |
| 21:14.36 | harry_v | unload the zap module |
| 21:14.52 | invalidrecord | hi anyione here built asterisk with the macports port? |
| 21:15.21 | [TK]D-Fender | x86: chan_dahdi.conf <--- |
| 21:15.38 | invalidrecord | got my new mac today but asterisk keepd complaining about not having config files even though they are in /opt/local/etc/asterisk |
| 21:15.39 | [TK]D-Fender | x86: You don't seem to haev read some VERY "fine" manuals.... |
| 21:16.00 | x86 | [TK]D-Fender: ah right, I forgot that file... I did configure it thought |
| 21:16.02 | x86 | though* |
| 21:16.05 | rig | i'm not following you |
| 21:16.08 | invalidrecord | the pid file and ctl are in the right /opt/local/var/run dir |
| 21:16.23 | x86 | http://pastebin.ca/1274060 |
| 21:16.27 | x86 | [TK]D-Fender: there it is |
| 21:16.40 | x86 | [TK]D-Fender: i did notice that when i reload the chan_dahdi, i get "ignoring signalling" |
| 21:17.26 | x86 | well i did, when I had a zapata.conf |
| 21:17.32 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 21:17.36 | x86 | not sure why it's looking at zapata.conf instead of system.conf.... |
| 21:18.00 | x86 | err, chan_dahdi.conf anyway |
| 21:18.35 | x86 | http://pastebin.ca/1274064 |
| 21:18.58 | stevie[xxx] | [TK]D-Fender i had to argue with my telephone provider if they send it with DTMF or direct |
| 21:19.23 | stevie[xxx] | they say i didnt configured it correct |
| 21:19.44 | x86 | sweet, removed signalling from chan_dahdi.conf (after symlinking it to zapata.conf), and now it doesn't give me "ignoring signalling error", but I also get NO channels at all now, not even pseudo |
| 21:20.29 | rig | how am i supposed to send it on a linksys ata for example? |
| 21:21.05 | x86 | had to totally restart asterisk (instead of reload chan_dahdi from the CLI), now I've got pseudo back... but still niether of my usable FXO lines appear |
| 21:21.27 | *** join/#asterisk pecanha (n=e@189.106.103.88) |
| 21:21.38 | [TK]D-Fender | x86: and your chan_dahdi.conf has not CHANNELS defined |
| 21:23.10 | x86 | [TK]D-Fender: when i put channels => 1-2 and/or signalling = fxs_ls, it gives me WARNING: Ignoring channels or WARNING: Ignoring signalling |
| 21:23.27 | x86 | oh wait, im retarded |
| 21:23.41 | [TK]D-Fender | x86: No.. that information we knew ;) |
| 21:23.46 | x86 | [TK]D-Fender: heh |
| 21:23.53 | x86 | [TK]D-Fender: ok, how do i specify channels? |
| 21:24.01 | [TK]D-Fender | channel => 1 |
| 21:24.05 | pecanha | Please, somebody help me! :/ Can't receive external calls -> virtual number -> on my extension. I'm behind a nat (my server is outside - valid ip). I have firewall disabled, and my softphone is on DMZ. I can talk between extensions. Errors: -- Couldn't call 1 |
| 21:24.09 | x86 | gah, channel, not channels |
| 21:24.16 | pecanha | and Everyone is busy/congested at this time (0:0/0/0) |
| 21:24.19 | [TK]D-Fender | x86: You might want to consider looking at those magical SAMPLES too! |
| 21:24.48 | [TK]D-Fender | pecanha: READ UP : |
| 21:24.50 | [TK]D-Fender | ~SIPNAT |
| 21:24.51 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:24.51 | x86 | [TK]D-Fender: unable to reconfigure channels 1-2 |
| 21:25.02 | [TK]D-Fender | heading home. BBIAB |
| 21:25.24 | rig | so, i'm pretty sure it's asterisk that's reading the hash as the end of the number, not my phone. |
| 21:25.54 | x86 | GAH! |
| 21:25.59 | rig | MOO! |
| 21:26.09 | x86 | chan_dahdi now says "unable to open /dev/zap/channel" |
| 21:26.11 | x86 | WTF?!?! |
| 21:26.15 | x86 | WHY ZAP?!!?! |
| 21:26.19 | x86 | kicks dahdi in the nuts |
| 21:26.40 | rig | and you're surprised dahdi doesn't like you? |
| 21:27.05 | pecanha | my server is not behind nat, so the tutorial doesn't apply... |
| 21:29.24 | bijit | x86: I would start from scratch compile dahdi then asterisk.. |
| 21:34.30 | stevie[xxx] | which shourtcut relase my locked screen session? |
| 21:36.14 | *** join/#asterisk telecos (n=sergio@154.166.219.87.dynamic.jazztel.es) |
| 21:36.31 | seanbright | ^A D |
| 21:38.52 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 21:40.04 | stevie[xxx] | doenst work |
| 21:40.04 | x86 | stevie[xxx]: ^A Q generally, if you've scroll-locked it (^A S) |
| 21:40.17 | stevie[xxx] | yeah , i thought so too |
| 21:40.25 | stevie[xxx] | seems to be crashed |
| 21:40.39 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 21:40.53 | seanbright | C-a C-q |
| 21:41.29 | stevie[xxx] | must be crashed |
| 21:41.41 | seanbright | screen is bug free |
| 21:41.43 | seanbright | it doesn't crash |
| 21:41.44 | seanbright | :P |
| 21:41.59 | *** join/#asterisk cryptnix (n=andrew@216.111.201.3) |
| 21:42.36 | cryptnix | when doing a pattern ... for any # ... upon an incoming route ... what would be a default pattern to pass anything to a route |
| 21:43.18 | stevie[xxx] | if i try to screen -x from other console it freezes too :d |
| 21:43.25 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:43.39 | seanbright | cryptnix: you mean like _.? |
| 21:43.44 | seanbright | (which is evil, btw) |
| 21:45.22 | seanbright | if you want to match any dialed number, do _X. |
| 21:45.30 | seanbright | although you should always be as specific as possible |
| 21:45.31 | cryptnix | inc . ? |
| 21:45.57 | seanbright | . matches one or more characters (not just numbers) |
| 21:46.03 | seanbright | so yes, it would also match a literal . |
| 21:46.06 | seanbright | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 21:46.24 | rig | so, nobody knows how i get the attended transfer to ignore the hash? |
| 21:46.50 | cryptnix | nice. thank you |
| 21:47.33 | seanbright | rig: let me look at the code |
| 21:47.35 | seanbright | hold |
| 21:47.46 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 21:47.59 | ruben23 | hi |
| 21:48.16 | *** join/#asterisk xorl (n=xorl@li30-130.members.linode.com) |
| 21:48.20 | rig | holding |
| 21:48.21 | xorl | ok, so why would I get |
| 21:48.27 | xorl | dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 21:48.38 | seanbright | xorl: client not registered? |
| 21:48.42 | cryptnix | Call from 'xxxx' to extension 's' rejected because extension not found. man i just seen #'s come thru ... asterisknow blows. |
| 21:48.49 | xorl | seanbright: Do I need to reboot the phones? |
| 21:48.52 | xorl | i swapped out our NAT |
| 21:49.02 | seanbright | xorl: no clue |
| 21:49.08 | rig | i think the "no route to destination" is a good hint |
| 21:49.13 | ruben23 | i have problem compiling asterisk 1.4.22 during installation i run make clean i have errors.... |
| 21:49.13 | seanbright | the error just means that it can't find the phone |
| 21:49.19 | seanbright | ruben23: pastebin them |
| 21:49.23 | cryptnix | its a group. |
| 21:49.23 | seanbright | ~pb |
| 21:49.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 21:49.57 | seanbright | rig: bad news... |
| 21:49.59 | seanbright | rig: if (res == '#') |
| 21:50.08 | rig | i had a feeling that was it |
| 21:50.10 | rig | damnit |
| 21:50.20 | rig | which file is that in? maybe i'll patch it |
| 21:50.33 | seanbright | you running 1.4? |
| 21:50.37 | rig | yeah |
| 21:50.42 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 21:50.43 | ruben23 | ok.. |
| 21:50.46 | seanbright | ast_app_dtget in main/cli.c |
| 21:50.49 | seanbright | err |
| 21:50.52 | seanbright | ast_app_dtget in main/app.c |
| 21:51.30 | Katty | can someone do a lil testy poo for me? |
| 21:51.34 | seanbright | can |
| 21:51.39 | Katty | querying |
| 21:51.41 | seanbright | i'll need $1 however |
| 21:51.59 | *** join/#asterisk tshine (n=tshine@ip70-160-111-108.hr.hr.cox.net) |
| 21:52.08 | rig | you think there's a reason it's in there? |
| 21:52.12 | rig | doesn't look like it to me |
| 21:52.27 | seanbright | can't hurt to comment it out and test it |
| 21:52.35 | seanbright | unless that testing is happening in a production system :) |
| 21:52.46 | ruben23 | hmm..this is hard im doing my asterisk boxes in virtual machine...cannot copy/paste my display errors...too long.. |
| 21:53.06 | rig | i'll give it a try |
| 21:53.57 | seanbright | ruben23: ok. copy and paste a single error into the channel then. |
| 21:54.08 | ruben23 | seanbright:cant pastebin..the error... |
| 21:54.35 | ruben23 | ok |
| 21:54.40 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:56.08 | ruben23 | line 1: Makefile:43: *** missing separator. stop. Line2: make: ***[apps-clean] Error 2 |
| 21:56.34 | seanbright | ruben23: yikes. what OS/distro? |
| 21:56.55 | ruben23 | its Centos 5.0 and asterisk 1.4.22 |
| 21:57.03 | rig | well, that fixed it |
| 21:57.12 | rig | should we tell someone? |
| 21:57.18 | seanbright | rig: if you consider it a bug, post it to bugs.digium.com |
| 21:57.29 | rig | seems like a bug to me, no? |
| 21:59.25 | jaytee | quittin time, be back later |
| 21:59.42 | rig | anyway, thanks for the help |
| 22:00.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:01.08 | ruben23 | seanbright: i got it with make clean....but no luck with ./configure & make |
| 22:02.29 | ruben23 | on ./configure errors= configure error: C++ preprocessor /lin/cpp fail sanity check |
| 22:02.36 | seanbright | yikes |
| 22:02.49 | seanbright | is this a fresh install? |
| 22:02.54 | ruben23 | yes |
| 22:03.01 | seanbright | did you install g++? |
| 22:03.06 | ruben23 | whst could be wrong |
| 22:03.29 | seanbright | yum install gcc-c++ |
| 22:04.29 | harry_v | btw, if a sip service is registered which includes did account wouldn't there be cli output if a did caller trying to call in if for some reason sip.conf or extentions.conf was misconfigered? |
| 22:05.02 | SkramX | what does this mean? -- Local/79210201@agents-f643,2 requested special control 20, passing it to SCCP/79210201-00000042 |
| 22:05.12 | harry_v | Never thought that would be a issue but getting a slow busy tone when testing it. |
| 22:05.56 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 22:06.10 | *** join/#asterisk granite12 (n=garryj@adsl-216-103-86-221.dsl.snfc21.pacbell.net) |
| 22:06.12 | *** part/#asterisk macno (n=michele@adsl-150-160.adsl.metrolink.it) |
| 22:06.24 | [TK]D-Fender | harry_v: Depends.. verbose if the call is accepted at all, SIP debug if you need to prove that they are even getting through to you in the first place |
| 22:06.33 | harry_v | true |
| 22:07.14 | granite12 | does 'telephone reminders v4' still have that problem where it starts playing its message before the external call is connected? |
| 22:08.00 | harry_v | uless my isp is blocking it. |
| 22:08.18 | [TK]D-Fender | harry_v: A possibility. |
| 22:09.06 | [TK]D-Fender | granite12: Very vague description of a problem for a 3rd party solution.... |
| 22:09.36 | [TK]D-Fender | granite12: Feel free to pastebin actual CLI output of your failed call attemp at verbose 10 |
| 22:09.46 | [TK]D-Fender | ~pb |
| 22:09.46 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 22:09.48 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 22:11.30 | nny_1 | [TK]D-Fender: got to run, but thanks for the tip yesterday, (ITSP) Looking good so far and very well built it seems |
| 22:11.44 | [TK]D-Fender | nny_1: You're welcome |
| 22:11.57 | *** part/#asterisk nny_1 (n=scott@64.203.244.146) |
| 22:12.15 | seanbright | ruben23: status update? |
| 22:12.45 | [TK]D-Fender | seanbright: Never trouble troubel till trouble troubles you. |
| 22:12.59 | stevie[xxx] | [TK]D-Fender , i dont make steps forward, with my bri card |
| 22:13.17 | [TK]D-Fender | stevie[xxx]: Go try and see if they sent post-answer DTMF |
| 22:13.24 | stevie[xxx] | i did |
| 22:13.47 | stevie[xxx] | the woman couldnt anwser this question |
| 22:13.58 | stevie[xxx] | she blamed me taht my config is wrong |
| 22:14.22 | *** join/#asterisk StephenF (n=none@cust-69-19-242-54.static.o1.com) |
| 22:14.35 | [TK]D-Fender | stevie[xxx]: Sorry, but I can't help beyond this. |
| 22:15.46 | stevie[xxx] | u have an idee why there are the last 2 digits missing in the DID ? |
| 22:15.50 | stevie[xxx] | do |
| 22:16.00 | *** join/#asterisk cianmaher (n=Cian@84.203.233.126) |
| 22:16.18 | [TK]D-Fender | stevie[xxx]: I told you my guess. You haven't seemed to have tried to test this theory |
| 22:16.33 | stevie[xxx] | i really called the provider |
| 22:16.43 | [TK]D-Fender | stevie[xxx]: TEST IT |
| 22:17.19 | stevie[xxx] | i gess i missed sth |
| 22:18.09 | Katty | [TK]D-Fender: can i borrow you for a moment? |
| 22:18.13 | Katty | [TK]D-Fender: i need to test my firewall |
| 22:18.35 | [TK]D-Fender | h4x0rz teh Katty |
| 22:20.18 | stevie[xxx] | how do i check for post-answer DTMF? pls |
| 22:21.53 | harry_v | TK, found what the issue was. My account being that it is a dynamic ip was not being recognized by my service provider so switched the IP address with account info. |
| 22:23.17 | [TK]D-Fender | stevie[xxx]: Go make some dialplan that will answer, and do a read and watch for digits received |
| 22:24.04 | cianmaher | then you need to decide what you want to do with that dtmf |
| 22:24.28 | [TK]D-Fender | harry_v: ew... fixed-only? |
| 22:24.36 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 22:24.42 | [TK]D-Fender | cianmaher: We're not sure thats what's happening. |
| 22:25.18 | harry_v | yes. I knew something wasnt right to the fact that no info was passing across the cli. Called them to confirm my suspitions so helpdesk was helpfull. |
| 22:26.06 | [TK]D-Fender | harry_v: point them to a host if you can and dyndns it or something |
| 22:26.29 | harry_v | Now my next issue, Cli at times hangs on reload or fresh install on sip.conf and users.conf. |
| 22:26.37 | harry_v | I have though of doing that. |
| 22:27.09 | harry_v | Ive looked at my wirewall records. Keep seeing shaw.abuse. in the hostname :) |
| 22:27.24 | stevie[xxx] | i am using FreePBX , and i set up different DIDs for incomming calls |
| 22:27.26 | cianmaher | How would one play a sound file to any active call, both parties (Called Party, Caller)? |
| 22:27.32 | [TK]D-Fender | harry_v: usual reason for hang is DNS problems |
| 22:28.08 | harry_v | Ohh well, the that may be the problem. your not talking about /etc/resolve.conf ? |
| 22:28.22 | [TK]D-Fender | harry_v: and so forth |
| 22:28.37 | [TK]D-Fender | harry_v: routeing, the who kit & kaboodle |
| 22:29.26 | x86 | [TK]D-Fender: thanks for the help earlier... I ended up calling digium as it was a bug in 1.4.22... to fix the problem, had to upgrade to 1.6.0.1 |
| 22:29.45 | x86 | [TK]D-Fender: took an hour for digium to solve it, so I understand why you couldn't help |
| 22:30.29 | [TK]D-Fender | x86: Well As usual I didn't get everything to go on in a nice clean shot so debugging is a bitch. |
| 22:30.47 | [TK]D-Fender | x86: Why do you have to be so damned difficult all the time? :p |
| 22:33.06 | x86 | I gave you everything you asked for :) |
| 22:33.19 | x86 | anyone ever setup SLA? |
| 22:33.29 | x86 | I'm having a hell of a time getting SLA working with Polycom IP330 phones |
| 22:33.45 | x86 | afaik, sla.conf is setup correctly (as per voip-info.org) |
| 22:35.03 | *** join/#asterisk FruitBasket (n=a@host-72-175-240-62.static.bresnan.net) |
| 22:35.38 | harry_v | never used sla.conf |
| 22:35.46 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 22:36.10 | FruitBasket | I have sip debugging on and I see a weird message... NOTIFY sip:1011@192.168.99.100, ... Subscription-state: terminated;reason=noresource. What does that mean? |
| 22:36.23 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 22:38.12 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:39.32 | [TK]D-Fender | x86: there is NO point to SLA on that model... assuming you can even get presence working on that 1 free line-key |
| 22:39.33 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 22:41.26 | *** join/#asterisk dahunter3 (n=dahunter@pool-72-67-222-109.lsanca.fios.verizon.net) |
| 22:49.45 | dahunter3 | What permission do I have to have to use the manager API originate command? |
| 22:50.14 | dahunter3 | (because I keep getting Permission denied) |
| 22:50.46 | seanbright | dahunter3: does doc/manager.txt hold any clues? |
| 22:51.11 | x86 | [TK]D-Fender: oh, man :( |
| 22:51.20 | x86 | [TK]D-Fender: no way to make the extension take no line keys? |
| 22:51.31 | harry_v | I dont know if this is a common practice among asterisk devs but is it common to set up the configs where the user can log into ANY phone of the same make and model and the configs follow them? |
| 22:51.36 | x86 | [TK]D-Fender: the phone's extension is always going to take up a line appearance? |
| 22:52.05 | [TK]D-Fender | x86: the phones REGISTRATION, yes |
| 22:52.13 | x86 | hmm |
| 22:52.13 | [TK]D-Fender | x86: * does not support SLA |
| 22:52.31 | [TK]D-Fender | x86: * supports a FUGLY hack that PRETENDS to be SLA. |
| 22:52.32 | x86 | [TK]D-Fender: i thought russelb put some SLA stuff into 1.4/1.6 recently? |
| 22:52.41 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:52.55 | x86 | [TK]D-Fender: ah, I was going to say, I was finding nothing about conflicting information regarding SLA + Asterisk |
| 22:53.12 | [TK]D-Fender | x86: And only to the over-zealous & desperate |
| 22:53.26 | x86 | [TK]D-Fender: I basically want old key system functionality, where I have two CO line appearances on the phones, and they can put it on hold, and pick it up on another phone, etc |
| 22:53.42 | x86 | [TK]D-Fender: so it's not ready for prime-time yet? |
| 22:53.45 | [TK]D-Fender | x86: WGLWAT |
| 22:53.51 | x86 | ? |
| 22:54.06 | [TK]D-Fender | x86: Prime time? Its a complete hack-job unworkable on smaller phones and... BLEH! |
| 22:54.16 | x86 | :( |
| 22:54.18 | [TK]D-Fender | x86: its only good for LINES anyways... not even stations |
| 22:54.26 | x86 | yeah that's what I want |
| 22:54.39 | x86 | I want two CO lines visible on all of my SIP stations |
| 22:56.10 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 22:58.28 | Yourname` | Help needed, SIP no route to host error out of nowhere.. http://pastebin.ca/1274285 |
| 22:59.07 | *** part/#asterisk `paul (n=admin@125.60.241.221) |
| 23:00.08 | [TK]D-Fender | Yourname`: sip show peer border" |
| 23:00.17 | [TK]D-Fender | Yourname`: and next time, include SIP debug |
| 23:00.30 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-a21309eda9c0bd2a) |
| 23:00.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 23:02.06 | *** join/#asterisk stevie[xxx] (n=stevie@e177159254.adsl.alicedsl.de) |
| 23:05.42 | *** join/#asterisk kerx (n=prepro@adsl-68-123-206-108.dsl.irvnca.pacbell.net) |
| 23:05.51 | kerx | hi, what is the bare minimum modules necessary to run asterisk? |
| 23:06.02 | kerx | or is there a list of modules and there descriptions of what they do somewhere? |
| 23:06.07 | jql | it boots up with zero |
| 23:06.16 | jql | just doesn't doo much withour res_pbx and chan_sip |
| 23:06.49 | kerx | what about AMD where is this module? |
| 23:07.02 | jql | shrugs |
| 23:07.32 | Yourname` | [TK]D-Fender: Funny enough, a stop now, delete astdb, and reboot seems to have fixed it :S |
| 23:08.53 | SkramX | [TK]D-Fender: around? :) |
| 23:09.33 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 23:09.36 | SkramX | so I have http://pastie.org/329294 - I want the user to continue in the queue if the called party (agent) hangs up. Right now, it hangs up both callers |
| 23:11.55 | *** join/#asterisk sasargen (n=chatzill@68-245-72-5.pools.spcsdns.net) |
| 23:16.08 | granite12 | hello! i am building an appointment reminder application for a doctor's office. i would like to use open source products. does anyone have recommendations of a "software stack" that they'd recommend? |
| 23:16.49 | granite12 | the stack would need to call patients at a scheduled time and play a customized message and log its results. |
| 23:20.35 | SkramX | [TK]D-Fender: nevermind. thanks for your help earlier |
| 23:22.05 | *** join/#asterisk quentusrex23 (n=nobody@209.90.233.58) |
| 23:22.25 | quentusrex23 | Does anyone know of an end users cheat sheet for how to use basic features? like voicemail, transfering calls, etc? |
| 23:24.09 | harry_v | granite, my wife is sort of in your position. Make sure the patients approve the message option. :) |
| 23:24.26 | harry_v | Granite, sounds like a php case to me. |
| 23:26.55 | quentusrex23 | Does anyone know of an end users cheat sheet for how to use basic features? like voicemail, transfering calls, etc? |
| 23:28.05 | harry_v | to my knowlege there is none you can just print out. But there is a list of features online you can use to make one. |
| 23:28.29 | granite12 | harry_v: lol! do you think asterisk is the right platform to initiate outbound calls? i'm coming from the webdev world so i'm confortable with that piece, just unsure about the telephony. |
| 23:28.50 | digime | Is 802.1p worth exploring? I have everything set by bandwidth priority now, but I can also tag the packets with 802.1p Marking Actions (preserve/explicit/map). Is 802.1p recommended? |
| 23:28.50 | harry_v | well what do I know |
| 23:28.51 | harry_v | :) |
| 23:29.59 | digime | Does anyone here have a snom m3 wireless sip phone? All my other IP phones can handle multiple calls but asterisk won't send multiple calls to the M3. It thinks the M3 is "busy". |
| 23:30.12 | granite12 | harry_v: ideally, there'd be a service i pay for by the minute/call. they could run asterisk or whatever. i just want to make sure the appointments get confirmed! |
| 23:31.16 | harry_v | look up asterisk call list. But I do not know if you can control the control list to call out at a given time. Others here make know more about that aspect. |
| 23:31.27 | granite12 | harry_v: will do, thanks! |
| 23:33.23 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:35.50 | ruben23 | hi |
| 23:37.16 | ruben23 | i have problem installing asterisk during ./configure heres my error: configure error: *** termcap support not found. im using centos 5.... |
| 23:38.11 | seanbright | ruben23: install ncurses |
| 23:39.06 | seanbright | ruben23: yum install ncurses-devel libtermcap-devel |
| 23:39.18 | ruben23 | ok thanks |
| 23:39.23 | seanbright | yup |
| 23:41.00 | kb3ien | anyone else got a limit of 10 buddies on the polycom? |
| 23:42.17 | kb3ien | maybe the the <bw> tag ist as i understood it? |
| 23:48.35 | ruben23 | seanbright:hi soory got another error on compiling...its on the make command.. |
| 23:49.22 | ruben23 | its on my app_amd.c |
| 23:52.37 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 23:53.55 | seanbright | ruben23: what is the error? |
| 23:53.55 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-a21309eda9c0bd2a) |
| 23:54.40 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 23:55.23 | ruben23 | oh goodness...its along one...what would i do.. |
| 23:55.54 | seanbright | pastebin. time to figure out how to do it in the VM. |
| 23:56.10 | ruben23 | :-D |
| 23:57.45 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |