| 00:00.09 | jameswf | ~fax |
| 00:00.10 | infobot | from memory, fax is The honor of designing the first fax *service* in actual use goes to Giovanni Caselli, an Italian abbot, born in Siena in 1815, who turned his hand to science and was, by 1849, editing a scientific magazine. In 1856 he claimed that he had developed a device, which he called a "pantelegraph," that could send facsimiles of images and text. Napoleon III did not come up with the idea, he merely backed it. |
| 00:00.18 | jameswf | ~faxing |
| 00:00.18 | infobot | faxing is, like, 8% knowledge, 5% skill, 11% luck, and 76% voodoo |
| 00:00.26 | IsUp | ~rxfax |
| 00:00.36 | IsUp | :P |
| 00:01.14 | KavanS | lol |
| 00:01.19 | KavanS | 76% voodoo |
| 00:01.41 | jaytee | evening brian |
| 00:02.09 | beek | evening jaytee |
| 00:02.19 | jaytee | evening beek |
| 00:03.15 | jaytee | man, there's nuthin on TV tonight |
| 00:03.33 | StinkyJew | does asterix do fax? |
| 00:04.03 | StinkyJew | JT -> http://www.youtube.com/watch?v=Ct0hBzqeBa0 <- funny |
| 00:04.14 | IsUp | theres an app named "rxfax" but its not working on latest 1.4 i think |
| 00:04.18 | IsUp | i am using it on my old pbx |
| 00:04.20 | jaytee | ok, I'm not gonna answer a question like that from someone with a nick like that or I'm gonna have as many problems with the media as Mel Gibson |
| 00:04.24 | beek | jaytee: www.hulu.com |
| 00:04.48 | jaytee | beek, isn't that run by aliens? |
| 00:05.03 | beek | And an incredible time sink. |
| 00:05.16 | beek | That's what I watch at the office in the evenings when I'm working late. |
| 00:05.16 | jaytee | I thought that was Facebook |
| 00:05.33 | beek | jaytee: Naw, that's just an incredible waste of time. There's a subtle difference. |
| 00:05.40 | nauticalthinker | anyone integrated Asterisk with an old Fujitsu 9600ms ? |
| 00:06.10 | jaytee | "7 friends have sent you drinks you can't actually drink or get drunk on, CLICK HERE to find out who!" |
| 00:06.12 | telnettech | anybody know the best place, beside freepbx.org, to learn how freepbx processes calls and other asterisk functions |
| 00:06.12 | nauticalthinker | I can't find any info online regarding the setup on the F9600 end |
| 00:06.28 | beek | I see that 1.6.1 has been officially released. |
| 00:06.48 | jaytee | really? *.org still had 1.6.0.9 up today |
| 00:07.05 | beek | The message was sent 7:39pm |
| 00:07.12 | jaytee | ah, ok |
| 00:07.36 | jaytee | telnettech, other than googling I'm not sure where you'd get more info. |
| 00:07.36 | telnettech | nautical: are you in sault ste marie, MI |
| 00:08.02 | nauticalthinker | no |
| 00:08.17 | nauticalthinker | why? |
| 00:08.46 | telnettech | we have a customer there that has a fujitsu who has been asking questions about our asterisk product and if we can integrate it |
| 00:09.03 | telnettech | which we havent done before |
| 00:09.08 | nauticalthinker | I c...we are doing the same for one of our clients |
| 00:09.23 | nauticalthinker | located in TN |
| 00:09.54 | telnettech | jaytee: remember i said that we are supposed to get a new version of Asterisk for our customers, well they sprung freepbx on us |
| 00:09.57 | nauticalthinker | what have you found so far on this? |
| 00:10.29 | telnettech | nautical: they are talking with our development team but nothing has been setup or tried |
| 00:11.15 | telnettech | jaytee: so i am looking for more info about the agi scripts and all of the other config info that i can get my hand on |
| 00:11.26 | jaytee | telnettech, it's still asterisk at the core with a web gui wrapper and sql for storing some stuff. I don't think it actually does "realtime" by default though. |
| 00:12.26 | jaytee | but like AsteriskNow or other spinoffs, it kind of restricts what you can do in the dialplan without learning how to override all the defaults. I've yet to see one good book on it out there. |
| 00:12.30 | telnettech | i have played a little with a beta version they gave us for an upcoming install in chicago and as i watch the CLI it bounces thru agi scripts |
| 00:13.23 | telnettech | so that is what im looking for, if I need to make something work, how and where to create the dialplan stuff |
| 00:13.55 | telnettech | i just dont have alot of time before the install to really just play with it right now |
| 00:16.04 | cp5 | 1.6.1.0 is considered stable? |
| 00:16.58 | beek | cp5: It's only 37 minutes old, so draw your own conclusions. |
| 00:17.08 | cp5 | beek, stable it is! |
| 00:17.28 | Qwell | don't deviate |
| 00:17.42 | beek | FWIW, I have been running the 1.6.x series without issue. I'll probably fire up an instance of 1.6.1 to see how that works. It's easy enough to fall back if there are problems. |
| 00:18.11 | beek | Some of the new features listed in CHANGES are compelling enough for me to look at it, for sure. |
| 00:18.17 | beek | ... and soon. |
| 00:18.18 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 00:20.18 | beek | GN all |
| 00:20.30 | beek | heads home for the evening. |
| 00:23.22 | jaytee | nite beek |
| 00:27.00 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 00:29.04 | IsUp | any ideas about faxing on 1.4.24.1?.. |
| 00:33.35 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 00:36.56 | IsUp | i am getting "codec_gsm.c:144 gsmtolin_framein: Invalid GSM data (1)" on every call |
| 00:36.59 | IsUp | any ideas? |
| 00:37.20 | IsUp | ~gsmbug |
| 00:37.21 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 00:39.21 | Qwell | tell me you aren't trying to fax over gsm... |
| 00:39.48 | Qwell | you'll make coppice angry. we don't like it when coppice gets angry. |
| 00:40.06 | coppice | but faxing works over my GSM phone |
| 00:40.55 | IsUp | nope, i am not trying fax atm |
| 00:41.05 | IsUp | rxfax is not working on latest 1.4 i think |
| 00:44.43 | thehar | it works for me |
| 00:45.33 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3a973a5e428d82c7) |
| 00:47.39 | IsUp | i dont have any idea about installing it |
| 00:47.45 | IsUp | can you help me please? |
| 00:48.41 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:52.47 | coppice | Qwell: why are new apps never added to updates? I understand not wanting to tinker with anything in the 1.4 stuff, but a new app which is independent of everything else seems a benign addition |
| 01:06.05 | joobie | guys i want to disable the intro i get for a voicemail |
| 01:06.13 | joobie | is it just directoryintro= ? and leave it blank?? is there a better way? |
| 01:06.54 | *** join/#asterisk t0rrieri (n=Torrieri@nelug/crew/torrieri) |
| 01:10.02 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 01:16.52 | Titanous | Has anyone ever implemented Google Voice/Grandcentral style 'phone switching' with Asterisk? (ie call piked up on desk phone, press * and cell phone rings, and call can be picked up) |
| 01:22.37 | *** join/#asterisk blkry (n=blkry@24-241-112-018.dhcp.gnvl.sc.charter.com) |
| 01:28.14 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 01:46.53 | *** part/#asterisk nauticalthinker (n=mratliff@c-76-122-200-95.hsd1.tn.comcast.net) |
| 01:52.02 | [TK]D-Fender | joobie: "core show application voicemail" |
| 01:54.56 | telnettech | jaytee: I found an IRC chat for freepbx as well |
| 01:56.35 | [TK]D-Fender | telnettech: And you're only finding this out NOW? :p |
| 01:56.56 | [TK]D-Fender | telnettech: After the countless times we've told others to GFO and go there ... |
| 01:57.04 | [TK]D-Fender | heh |
| 01:57.12 | telnettech | yes TK....the development team has released our latest pbx and it us based off freepbx....i am looking for more info about it |
| 01:57.46 | telnettech | i am new to not just asterisk but this IRC chat stuff as well |
| 01:58.58 | telnettech | TK: are you in both chatd? |
| 01:59.17 | telnettech | chats |
| 01:59.21 | [TK]D-Fender | telnettech: And more |
| 02:02.19 | *** join/#asterisk Aiatek (n=munoz@190.6.143.194) |
| 02:02.20 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 02:02.36 | telnettech | i am at least seeking the know how |
| 02:03.06 | jaytee | telnettech, [TK]D-Fender is omnipresent, wherever you turn....there he is |
| 02:03.21 | [TK]D-Fender | jaytee: I prefer the term "stalker" :p |
| 02:03.35 | telnettech | i am still looking for a good understanding of channel variables as well |
| 02:04.01 | [TK]D-Fender | telnettech: There's this wonderful DOC that comes with your source tarball... you should read it |
| 02:04.30 | telnettech | i have started my own "poster" of channel variables that I see when I am troubleshooting a customer site |
| 02:05.01 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:05.04 | telnettech | and then I go to either the TOF or wiki to find out what it does |
| 02:05.18 | telnettech | my supervisor thinks im crazy |
| 02:05.24 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 02:09.08 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-709bf77bb9f590ee) |
| 02:09.30 | jaytee | telnettech, [TK]D-Fender was just trying to give you a hint about the channelvariables.txt file in the tarball. it's chock full o' good stuff. |
| 02:10.03 | [TK]D-Fender | jaytee: I prefer the term "fucking manual" ;) |
| 02:10.47 | telnettech | I understand that |
| 02:11.15 | telnettech | I just dont have access at this time and therefore didnt have a reply |
| 02:11.26 | telnettech | but i have started that poster on the wall |
| 02:11.31 | jaytee | access to what? |
| 02:11.44 | telnettech | to any system at this time of night |
| 02:11.50 | [TK]D-Fender | jaytee: its also best I don't know who's bright idea it was to convert those perfect little files to TEX... like as if anyone gives a crap about TEX :p |
| 02:12.13 | jaytee | telnettech, I carry all the docs on a jump drive, this is the 21st century ya know |
| 02:12.58 | jaytee | [TK]D-Fender, if you find out lemme know, I have a dull rusty spork I want to use to eviscerate them with. |
| 02:13.10 | telnettech | yeah well i am slowly moving into the century...I think we have the Y2k update finally finished :) |
| 02:13.42 | [TK]D-Fender | telnettech: My company finished our Y2K conversion to JD Edwards... in summer 2005 :p |
| 02:14.10 | jaytee | oh, god! JD Edwards (makes sign of the cross) |
| 02:14.22 | jaytee | hey wait! I'm not even religious! |
| 02:14.28 | telnettech | we sent out letters to all 125 customers that we take care of their Avaya systems advising that as of july 1st, we wouldno longer support them |
| 02:14.47 | telnettech | so that is how far we are behind the time |
| 02:15.17 | telnettech | and not VOIP systems |
| 02:15.24 | jaytee | telnettech, that'll work as long as you don't have firm contract dates or already have a "can quit" clause |
| 02:16.01 | telnettech | they all have a can quit clause but it has to be like 90 day notice |
| 02:16.23 | telnettech | and we havent renewed any since jan 1st |
| 02:16.44 | telnettech | but we gave those customers "special" rate for T&M rates |
| 02:17.02 | jaytee | did you support Avaya before you did *? |
| 02:17.08 | telnettech | yes |
| 02:17.16 | telnettech | we have been since 1999 |
| 02:17.27 | telnettech | and I have since 2004 |
| 02:17.56 | jaytee | then now would be a good time to start forming your own company and swoop in to pickup all the support contracts your present employer is dumping. You could make some sweet cake |
| 02:18.02 | telnettech | remember, I work in the hotel industry and they are some of the cheapest business owners to deal with |
| 02:18.21 | telnettech | I cant.....confidentiality clause |
| 02:19.51 | telnettech | The sales group is going to these customers and are supposed to give them a special price to upgrade the infrastructure and pbx with us |
| 02:23.15 | jaytee | telnettech, so what are you installing now for your customers? 1.4 with FreePBX added on or something else? |
| 02:24.18 | telnettech | going forward it will be 1.4.24 with Freepbx as the basis |
| 02:24.45 | telnettech | the Aruba was the last 1.2 install we did |
| 02:25.43 | jaytee | good thing you're not down there now, what with the swine flu goin around |
| 02:25.43 | telnettech | tell me about it |
| 02:26.00 | [TK]D-Fender | jaytee: Men finally have a reason to feel threatened :) |
| 02:26.03 | telnettech | Im supposed to go to Chicago around May 18th thru the 1st week in june |
| 02:26.20 | jaytee | that's my b'day |
| 02:26.23 | telnettech | for phase 1 of an install |
| 02:27.14 | telnettech | it is a 5 star hotel with 564 rooms, the 2nd phase is supposed to be 100 to 150 condos |
| 02:27.46 | telnettech | we will be there with a team of tech from our sister company Nomadix and a group from Singapore |
| 02:28.34 | telnettech | as a whole, customer will have IPTV, VOD, Hi speed Internet and Voip from us |
| 02:31.20 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 02:32.07 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) |
| 02:39.51 | telnettech | ok guys good night |
| 02:40.36 | jaytee | nite |
| 02:40.47 | *** join/#asterisk keith4_ (n=keith@unaffiliated/keith4) |
| 02:41.04 | *** part/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 02:42.18 | [TK]D-Fender | jaytee: In other news : Last week I took my 5th & 4th kyu exams back to back.... I figured the next would be coming shortly, but its this SUNDAY. Eek |
| 02:42.48 | jaytee | must stay focus, Daniel-san!!! |
| 02:42.51 | *** part/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 02:50.29 | eppigy | [TK]D-Fender: i am takign my icnd1 tommorow |
| 02:50.58 | eppigy | oh haha quite different than kyu |
| 02:51.05 | [TK]D-Fender | eppigy: But... do you have Linux+ ? ;) |
| 02:51.15 | eppigy | D: |
| 02:51.32 | eppigy | i can pass any comptia exam without any prior study |
| 02:52.00 | eppigy | possibly never having touched a computer |
| 02:52.08 | [TK]D-Fender | eppigy: :D |
| 02:52.23 | eppigy | i have a network+ |
| 02:52.41 | eppigy | i prepared by getting inebriated 2-3 times a weeks |
| 02:52.52 | eppigy | and putting off studying until the mornign of the exams |
| 02:53.00 | eppigy | and I was like damn |
| 02:53.05 | eppigy | pop3 is tcp 110?? |
| 02:53.10 | eppigy | and I passed it |
| 02:53.25 | eppigy | *exam |
| 02:53.50 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) |
| 02:54.08 | eppigy | i will be working on an rhce after I pass the icnd2 |
| 02:54.48 | jaytee | nite all |
| 03:05.04 | *** join/#asterisk propellerhead (n=yogurt2u@host31.190-227-250.telecom.net.ar) |
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| 03:33.40 | b14ck | sup |
| 03:37.19 | drmessano | Can Asterisk protect me from Swine Flu? |
| 03:43.34 | Nasra | get a Mask |
| 03:44.36 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 03:45.38 | drmessano | Command 'core show function mask' failed. |
| 03:45.41 | drmessano | Apparently not |
| 03:47.37 | [TK]D-Fender | "AIDS infects Herpes, news at 11" |
| 03:48.03 | b14ck | is there a way that any of you know of to get a count of how many extensions are on the system without parsing anything? |
| 03:48.10 | b14ck | im trying to avoid parsing stuff if possible |
| 03:48.13 | *** join/#asterisk Octothorpe (i=octothor@pdpc/supporter/professional/octothorpe) |
| 03:48.32 | drmessano | ROFL |
| 03:48.38 | drmessano | app_osmosis |
| 03:48.49 | [TK]D-Fender | b14ck: "extensions are on the system"? Does "." count as infinite? |
| 03:49.09 | b14ck | how many extensions the user has created |
| 03:49.14 | b14ck | active or not |
| 03:49.14 | drmessano | . would require parsing |
| 03:49.22 | [TK]D-Fender | drmessano: Still valid |
| 03:49.24 | b14ck | if it returns a . thats fine |
| 03:49.28 | b14ck | do you know a way, fender? |
| 03:49.54 | drmessano | [TK]D-Fender:Still means I need to guess it exists.. Since I am not parsing it |
| 03:50.12 | b14ck | in my particular case, if they have 0 extensions, i'll just quit the program early |
| 03:50.14 | [TK]D-Fender | b14ck: As phrased I fail to find a point for your goal let alone a sane set of boundaries |
| 03:50.30 | b14ck | how do you fail to find the point of my goal? lol |
| 03:50.33 | b14ck | its really simple |
| 03:50.38 | drmessano | YEAH |
| 03:50.42 | drmessano | Simple |
| 03:50.50 | b14ck | how many extensions are configured in sip_additional.conf |
| 03:50.58 | [TK]D-Fender | goes back to counting angels dancing on a pin |
| 03:51.06 | b14ck | not including voip trunks or anything like that |
| 03:51.11 | b14ck | whats unclear about that? |
| 03:51.16 | drmessano | How can you avoid parsing? |
| 03:51.21 | b14ck | that's what im asking |
| 03:51.25 | [TK]D-Fender | b14ck: a SIP DEVICE is NOT an "extension" |
| 03:51.25 | b14ck | if there's a function that im not finding |
| 03:51.34 | b14ck | *device* |
| 03:51.41 | drmessano | You cant PARSE something without PARSING |
| 03:51.45 | b14ck | ... |
| 03:51.52 | b14ck | drmessano, you are familiar with functions? |
| 03:51.59 | drmessano | Im familiar with logic |
| 03:52.12 | b14ck | functions are small blocks of logical statements |
| 03:52.46 | [TK]D-Fender | b14ck: Well... I haven't heard too many logical statements from you.. so I guess you serve no function :p |
| 03:52.48 | drmessano | I done care if you call it a parse, a grep, or a scrape, you cannot parse information without parsing |
| 03:52.49 | b14ck | exten => _X.,n,Set(test="test") <-- set is a function |
| 03:52.51 | drmessano | It defies logic |
| 03:52.59 | drmessano | dont* |
| 03:53.01 | b14ck | i'm looking for a function that will return the number of sip devices on the system |
| 03:53.11 | drmessano | Which means parsingh |
| 03:53.13 | drmessano | Which means parsing |
| 03:53.14 | b14ck | thereby, not requiring me to parse anything |
| 03:53.18 | drmessano | .... |
| 03:53.25 | b14ck | parsing is when you have a large collection of data, but need only a select piece of it |
| 03:53.30 | [TK]D-Fender | drmessano: X = MAYBE Y. <- I invented the world's first ILLOGICAL operator. Amrageddon soon to follow! |
| 03:53.36 | b14ck | which isn't what i want to do =/ |
| 03:53.41 | drmessano | This is the dumbest shit I have ever heard |
| 03:53.49 | drmessano | Which is saying a lot |
| 03:53.50 | [TK]D-Fender | b14ck: No, there is no such function because nobody has cared to create it |
| 03:53.51 | b14ck | are you like joking? |
| 03:54.00 | b14ck | fender, OK lol |
| 03:54.12 | drmessano | app_fluxcapacitor |
| 03:54.17 | [TK]D-Fender | b14ck: Since it CAN be pasred out why would anyone write more CORE code bloating the base and wasting memory? |
| 03:54.20 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 03:54.28 | *** join/#asterisk mujah (n=Mohamed_@123.231.20.227) |
| 03:54.30 | b14ck | sigh |
| 03:54.52 | b14ck | they also have gotoif and goto, you dont really need gotoif if you do it properly |
| 03:54.55 | b14ck | but it helps |
| 03:55.15 | drmessano | Would it not need to be parsed for the information to be available? Something has to be parsing it, I dont care if you do it in the dialplan or not |
| 03:55.18 | b14ck | all you really need is increment and decrement |
| 03:55.20 | [TK]D-Fender | b14ck: Why does the dialplan have to know there are 16 SIP peers... OH, ofr that matter, what makes one an "extension" and one a "trunk"? As for as * knows, SIP is SIP. Who knows what's on the other side ? |
| 03:55.24 | *** part/#asterisk mujah (n=Mohamed_@123.231.20.227) |
| 03:55.44 | b14ck | drmessano, not necessarily |
| 03:55.46 | b14ck | example: |
| 03:55.50 | [TK]D-Fender | b14ck: Don't need gotoIf because of Goto? CRAZY |
| 03:55.58 | [TK]D-Fender | b14ck: the REVERSE is tru however |
| 03:56.02 | b14ck | no it isnt |
| 03:56.32 | b14ck | this is becoming a long conversation for a simple question |
| 03:57.00 | [TK]D-Fender | b14ck: Actually I see a way using a few other functions. nassty chain though. I suppose tecnically either could replace the other |
| 03:57.06 | [TK]D-Fender | b14ck: and I answered you already |
| 03:57.12 | b14ck | ya, thanks |
| 03:57.23 | drmessano | There is nothing simple about questions that completely defy logic.. only simple for the one failing to see the lack of logic. |
| 03:57.33 | [TK]D-Fender | b14ck: Serious, WTF are you going to do knowing you have 14 "extensions" defined in the dialplan? |
| 03:57.37 | b14ck | drmessano, i strongly suggest you read a programming book |
| 03:57.44 | b14ck | parsing is a well used term |
| 03:57.47 | b14ck | and clearly defnied |
| 03:58.04 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:58.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 03:58.05 | b14ck | [TK]D-Fender, its for a pbx module |
| 03:58.22 | drmessano | .... |
| 03:58.28 | b14ck | im rewriting the extensions page for freepbx =p |
| 03:58.38 | b14ck | for a bit of fun :) |
| 03:59.27 | drmessano | lol |
| 03:59.38 | [TK]D-Fender | b14ck: So what would you do with this newfound miraculously divined answer? |
| 03:59.49 | drmessano | [TK]D-Fender: parse it |
| 04:00.03 | [TK]D-Fender | drmessano: Parse it like a Polaroid picture! |
| 04:00.21 | drmessano | Puff Puff, parse? |
| 04:00.22 | b14ck | i'm using it for display |
| 04:00.24 | [TK]D-Fender | p-p-p-parse it.........p-p-p-parse it......Parse it like a Polaroid picture! |
| 04:00.33 | b14ck | so im creating a 2d array based on the amount of extensions |
| 04:00.33 | [TK]D-Fender | b14ck: Display? On what? |
| 04:00.36 | b14ck | to keep it looking clear |
| 04:00.51 | b14ck | so if there was a function to count them for display |
| 04:00.57 | b14ck | it'd be a bit easier |
| 04:00.59 | b14ck | thats it really |
| 04:01.03 | b14ck | i have no problem parsing it |
| 04:01.11 | b14ck | just a pain since extensions are in all different files |
| 04:01.17 | b14ck | iax_additional, sip_additional, etc |
| 04:01.19 | [TK]D-Fender | b14ck: You shouldn't have to |
| 04:01.36 | b14ck | why not? |
| 04:01.42 | drmessano | Shame there isn't a db of all that stuff.. like astdb.. oh, wait |
| 04:01.49 | b14ck | drmessano, astdb has no counter |
| 04:01.52 | b14ck | or i would have used that |
| 04:01.55 | b14ck | *sigh* |
| 04:02.12 | [TK]D-Fender | b14ck: Because all of your "extensions" are defined in an SQL database and with half a brain you should already have come up with the 1 line query that would generate a count. |
| 04:02.17 | [TK]D-Fender | b14ck: Do YOU program? |
| 04:02.25 | b14ck | i dont do any sql, but im competent enough |
| 04:02.32 | b14ck | well it isnt sql really |
| 04:02.34 | b14ck | its berkely db |
| 04:02.40 | b14ck | so sql queries dont work |
| 04:02.41 | [TK]D-Fender | b14ck: I'd like some charater witnesses please |
| 04:02.52 | [TK]D-Fender | calls Wingdings to the stand |
| 04:02.54 | b14ck | but you were close enough |
| 04:03.02 | drmessano | Ummm |
| 04:03.33 | drmessano | Yeah [TK]D-Fender, why would you think FreePBX stores its configs in SQLite or MySQL |
| 04:03.36 | drmessano | Wait.. |
| 04:03.45 | [TK]D-Fender | b14ck: your stupid GUI DEFINITIONS are in SQL <---------- |
| 04:03.51 | b14ck | im not talking about gui |
| 04:03.56 | [TK]D-Fender | b14ck: and you WORK for them. Sad... just ... sad |
| 04:04.06 | b14ck | in the sql databases there are only gui options |
| 04:04.13 | b14ck | astdb (berkely db) stores the extensions and information |
| 04:04.15 | b14ck | sql does not |
| 04:04.16 | drmessano | The fucking EXTENSIONS are stores in SQL |
| 04:04.23 | drmessano | .... |
| 04:04.29 | b14ck | drmessano, please chex again! |
| 04:05.01 | drmessano | So how does one regenerate the astdb via the web GUI if the web GUI is deleting its own database? |
| 04:05.06 | b14ck | SELECT * FROM `extensions` WHERE 1 |
| 04:05.17 | b14ck | are you OK? the web GUI doesnt USE astdb |
| 04:05.25 | b14ck | err excuse me |
| 04:05.26 | drmessano | lol |
| 04:05.26 | b14ck | i mean |
| 04:05.32 | b14ck | the web gui uses the sql db |
| 04:05.37 | b14ck | it only pulls the extension list from astdb |
| 04:05.57 | drmessano | There is a GUI command to rebuild the ASTDB when its corrupted |
| 04:05.59 | b14ck | their query is in /var/www/html/admin/core/functions.php |
| 04:06.00 | drmessano | Dont tell me there is not |
| 04:06.10 | b14ck | there isnt, unless you install phpmyadmin |
| 04:06.15 | b14ck | and it doesnt rebuild astdb |
| 04:06.57 | drmessano | Yes, there is.. its done with a function passed via the browser.. undocumented |
| 04:07.04 | b14ck | lol |
| 04:08.46 | carrar | func_oracle_mysql_berkely_db.so |
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| 05:23.38 | joobie | guys i have two queues.. and using addqueuemember() to add people to the queues.. how can i ensure if someone is sitting in one of the queues, it gets priority over the other queue? |
| 05:24.21 | joobie | like agent1, logged into queue1 and queue2.. agent2, logged into queue2.. If a caller comes in queue2, it's split evenly to each other.. if a call comes into queue1, it should always be the next one sent to agent1 (priority) |
| 05:25.01 | leifmadsen | you might need to set a penalty on the agent in one of the queues |
| 05:25.29 | leifmadsen | I don't think it'll work across queues though the way you're probably expecting it to |
| 05:26.02 | joobie | what abotut hte weight= option for the queue.conf |
| 05:26.18 | joobie | just reading about it now.. but http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf says there might be a queue deadlock.. it has a pointer to a bug that doesnt exist |
| 05:27.54 | leifmadsen | aha, right, weights, I couldn't remember what it was called |
| 05:28.04 | leifmadsen | voip-info.org is highly inaccurate when it comes to things like that |
| 05:28.05 | joobie | leifmadsen, when it says the heigher weight |
| 05:28.24 | joobie | does that mean the closer to 0, the more priority.. ie, weight of 1 is more important than 100 |
| 05:28.36 | joobie | or is 100 higher priority to 1? |
| 05:29.06 | leifmadsen | 0 is highest weight I believe |
| 05:29.16 | leifmadsen | 100 is less important than 0 |
| 05:29.23 | joobie | ta |
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| 05:47.11 | joobie | guys having problems trying to find out if EU ISDN is: stereo/mono, bitrate, sampling rate |
| 05:47.16 | joobie | can anyone help? |
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| 06:06.26 | joobie | guys anyone know if i use .. 'out-of-hours|02:00-06:59|sun-mon|*|*' will that do from sunday 2:00 right through to monday 6:59AM? or will it do sunday 2AM-sunday 6:59 and then monday the same? |
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| 06:08.13 | omer | I am trying to compile asterisk but getting this erro "/usr/bin/ld: cannot find -lssl" ?? |
| 06:08.28 | omer | which package do I need? |
| 06:09.49 | omer | openssl-devel? |
| 06:10.35 | omer | Yes , omer you need openssl-devel .... |
| 06:10.39 | omer | ohh thanks ... |
| 06:10.42 | omer | you are welcome |
| 06:10.45 | omer | :-) |
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| 07:11.35 | omer | i have centos 5.2,.....mysql-server 5 and mysql-client is installed.....asterisk 1.4.24 is running ... just installed asterisk-addons-1.4.8 ..... but res_config_mysql.so is still not there....I need it for asterisk realtime |
| 07:12.34 | Pagautas | hi |
| 07:12.38 | Pagautas | anybody alive? |
| 07:12.47 | omer | I dont think so ... |
| 07:13.38 | joobie | guys what's the best way to call forward with asterisk? |
| 07:13.46 | joobie | say i want to forward to a totally different number and just let the call go |
| 07:13.57 | joobie | it's an external number btw |
| 07:14.19 | Pagautas | i have a big problem |
| 07:14.44 | Pagautas | i have a few extensions like exten => _X.,1,Palayback(file) |
| 07:14.56 | Pagautas | i've upgraded to 1.6.1 |
| 07:15.11 | Pagautas | when a call comes from iax the sound is played very lagged |
| 07:15.17 | Pagautas | where could be a problem |
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| 07:15.27 | omer | joobie: check features.conf |
| 07:15.27 | Pagautas | there were no problem with 1.6.0 |
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| 07:21.08 | Pagautas | with sip there is no problem |
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| 07:47.43 | dnikulin | hi all |
| 07:47.55 | dnikulin | I have asterisk and ekiga problems |
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| 07:48.31 | dnikulin | the problem is ekiga has external IP and astersk allows ekiga make only internal calls |
| 07:48.50 | dnikulin | for internal clients :Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING |
| 07:49.05 | dnikulin | and fo r external ekiga client: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE |
| 07:49.15 | dnikulin | can anybody help? |
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| 08:37.56 | DGTL_Magician | Hi |
| 08:38.16 | *** join/#asterisk swc|666 (n=swc@unaffiliated/swc666/x-4934821) |
| 08:38.22 | DGTL_Magician | I have a customer with a problem |
| 08:38.35 | DGTL_Magician | He has an Asterisk 1.4 box with Cisco SIP phones |
| 08:39.04 | DGTL_Magician | When he dials out he can still receive incoming calls, which should be diverted to another phone when he's busy |
| 08:39.37 | DGTL_Magician | I set incominglimit and call-limit to 1 but this allows 1 outgoing and 1 incoming at the same time |
| 08:39.44 | DGTL_Magician | still not fixing his issue |
| 08:39.50 | DGTL_Magician | anyone have an idea? |
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| 08:42.19 | Iskorptix_ | hello |
| 08:42.48 | Iskorptix_ | has anybody seen this warning Dial does not accept L(0), hanging up before ? |
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| 08:48.46 | jeffspeff | i've started getting this error... can anybody help? WARNING[3494]: func_strings.c:652 acf_strftime: C function strftime() output nothing?!! |
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| 09:02.59 | jeffspeff | i've started getting this error... can anybody help? WARNING[3494]: func_strings.c:652 acf_strftime: C function strftime() output nothing?!! |
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| 09:19.56 | Pagautas | i've tried almost all versions from 1.4.17 to 1.6.2-beta1 |
| 09:20.19 | Pagautas | and 1.4.17 is the latest version |
| 09:20.46 | Pagautas | where mixmonitor doesnt stops on call transfer using phone transfer button |
| 09:21.02 | Pagautas | call comes to queue |
| 09:21.57 | Pagautas | monitor insted of mixmonitor doesnt work at all |
| 09:22.59 | Pagautas | maybe wrong channel |
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| 09:42.43 | gambler1 | Hi, I have a little trouble with * 1.6.0.3 having some hanging calls (ie. the user is not talking anymore but * says that channel is up) |
| 09:43.23 | gambler1 | so I was wondering, when packet2packet bridging is used? When there is different codecs on both sides or? |
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| 09:53.35 | frk2 | man I am upto the wall with my local telco regarding CID on outbound PRI calls |
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| 10:00.50 | jes-o-mat | Hi |
| 10:01.25 | jes-o-mat | is there a better way to use SQL statements inside the dialplan beside escaping all spaces? |
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| 10:14.35 | kaldemar | DGTL_Magician: use GROUP and CHECK_GROUP functions |
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| 10:30.53 | gr0mit | anyone recommend a voip provider for incoming numbers in Denmark? |
| 10:31.57 | DGTL_Magician | kaldemar: GROUP and CHECK_GROUP ? |
| 10:32.39 | DGTL_Magician | also calls are being routed through queues.conf |
| 10:36.14 | DGTL_Magician | and Ringinuse is set to no in queues.conf |
| 10:37.32 | DGTL_Magician | I presume the problem is that the SIP phone isn't sending InUse |
| 10:39.16 | kaldemar | DGTL_Magician: they are functions in the dialplan. you can add a call to a group with GROUP and check how many calls are in the group with CHECK_GROUP. using those you can block calls. |
| 10:40.25 | kaldemar | if you have queues with multiple members, then you really can't use those. then look into the phone parameters to allow only one call. |
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| 11:02.43 | Odd_Bloke | Hello all. We're looking to customise the behaviour of part of our dialplan (which outgoing line to use if the dialled number starts with 0) if you are dialling from a given extension (204). I'm currently intending to use a GotoIf on the dialled extension. Is this the most sensible way to do it and, if so, how can I find the extension which is calling? |
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| 11:14.50 | gambler1 | Odd_Bloke: I think it is (but I am not the expert). Hmmmm I think there is an easy way to find out the extension but right now I can think of two not so easy. One is to use SIP_HEADER and get the apropriate field or if you have some agi script that are you calling anyway * will pass a bunch of variables (including extension) |
| 11:24.09 | Odd_Bloke | Turns out I can use CHANNEL for what I want. |
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| 11:59.57 | joobie | guys im going to write a windows app that sits on the screen and displays the number of callers in a call queue, in realtime to an end user |
| 12:00.05 | joobie | just wondering if tehre's a good wya to make this info realtime? |
| 12:00.18 | HenrikBe | hi, I am about to create an predictive dialer with asterisk/sip and wonder if there are any good tutorials on the subject? I will use PHP/javascript/ajax on the client side. |
| 12:00.27 | joobie | been looking at the asterisk management interface, which is OK.. but you have to issue a request for it to respond with how many people are in queue |
| 12:00.44 | joobie | as far as i can tell, there's no way for you to setup an event that will report that |
| 12:01.12 | joobie | problem with having to send a command to get the response is to make it reatime for the end user, i'd have to submit the command every second or two.. which is unneccsary load im trying to avoid |
| 12:01.24 | joobie | is there another way anyone can think of? |
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| 12:13.02 | DGTL_Magician | Maybe for future reference, I fixed the problem kaldemar |
| 12:13.14 | DGTL_Magician | The Cisco phones have an option Call Waiting |
| 12:13.30 | DGTL_Magician | if you turn that off in the config the phone sends SIP In Use messages. |
| 12:15.00 | sulex | scenario: a DAHDI user in answered within a context, he's sent to an AGI where some stuff are done. At the beginning of the AGI I also create a call file to a SIP user, the SIP user aknowlegde the call and stays on phone. The procedures within the AGI finishes and if the SIP user aknowledged the call from the call file the AGI made, I join the two channel to let the caller and the SIP user to talk eachother. Question: is this something |
| 12:15.00 | sulex | possible? (sorry for length and bad english) |
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| 12:19.07 | joobie | sulex, yes |
| 12:21.34 | sulex | joobie: how do i join the two channels? i know how to send a channel to an extension of course the viceversa... but what's the best way to join something like, DAHDI/2-1 to SIP/someiodiot-some_weird_id |
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| 12:27.02 | joobie | there are a few ways sulex |
| 12:27.09 | joobie | call parking is one way that comes to mind |
| 12:30.21 | sulex | joobie: ok thanks, i try to look in that direction |
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| 12:34.39 | joobie | sulex, conference calls also worthwhile looking into |
| 12:36.37 | sulex | joobie: help me understanding if i get it please. A user= DAHDI, B user=SIP, when A calls in I start an AGI, the AGI creates on the fly a "call file" to B. B answers and aknowledge the call, AGI traps somehow the call file is been answered and park the call identified by the SIP channel. A the end of the AGI running on the user A channel i pickup the call from the parking lot... the two peers can now talk eachother... |
| 12:36.39 | sulex | lol |
| 12:36.56 | sulex | i donno if this makes sense just to me and my fantasy :) |
| 12:38.39 | joobie | ya |
| 12:38.48 | joobie | that is pretty much it using parking |
| 12:39.20 | sulex | ok let's try this way and see what happens, thank a lot joobie ;) |
| 12:39.53 | joobie | once the call file has been processed |
| 12:39.58 | joobie | it will end up in the outgoing folder |
| 12:40.09 | joobie | and there is a status in the file that tell su how it went |
| 12:40.27 | joobie | that would be how u can check the call went thru.. just testthe codes tho |
| 12:40.54 | joobie | i did some funky shizz with it before and it wasnt accurate in combination with certain functions.. the status is very basic so it will tell u only very basic info |
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| 12:50.33 | gego | hi there |
| 12:51.33 | gego | I'm trying to load chan_misdn by init before asterisk, which it does, but it's not included in asterisk |
| 12:52.04 | gego | if i manually run the init-scripts it is. who knows why? |
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| 12:58.40 | jaytee | mornin [TK]D-Fender |
| 12:59.26 | [TK]D-Fender | jaytee: blarg. passed out with everything on last night, woke up with my arms asleep, and biked to work. Made good time, but I feel thoroughly tenderized right now |
| 12:59.36 | [TK]D-Fender | is a sack of meat this morning |
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| 13:01.09 | jaytee | [TK]D-Fender, ouch! I've been there and done that. definitely no fun |
| 13:02.00 | jaytee | gego what distro are you running? |
| 13:02.20 | gego | jaytee: lenny |
| 13:03.20 | jaytee | if it works with the init scripts manually but doesn't work when you do a restart it's likely that it's trying to run too early on in the boot process and it's missing a dependency that hasn't initialized yet. |
| 13:04.14 | jaytee | not sure how Lenny prioritizes init scripts for services. RHEL and CentOS do it a little different I think |
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| 13:05.38 | mort_gib | Hi, any idea what ERROR[29631] utils.c: write() returned error: Broken pipe means?? |
| 13:05.45 | gego | jaytee: you meen, if I put it directly before * I have a better chance? symlinks in runlevels rc.x have a priority prefix - is that what you mean? |
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| 13:06.19 | mort_gib | A polycom handset divert incoming calls to his mobile, it work with internal calls but I get this on external calls |
| 13:06.49 | [TK]D-Fender | mort_gib: Every call is jsut a call |
| 13:07.33 | mort_gib | [TK]D-Fender: I know that, still strange behavior, what does it mean?? |
| 13:07.37 | jaytee | gego, yeah, like with CentOS the symlinks start with an S## from 1 to 99 |
| 13:08.07 | jaytee | so I try to start my zaptel or dahdi service as the very last service and then asterisk dead last |
| 13:08.20 | [TK]D-Fender | mort_gib: Show us :) |
| 13:08.42 | jaytee | gego, gotta run out for a bit, be back in a few |
| 13:08.51 | mort_gib | [TK]D-Fender: Hang on, extensions.com is enough right?? |
| 13:09.12 | [TK]D-Fender | mort_gib: No, a complete detailed error and another divert for comparison as well |
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| 13:14.43 | mort_gib | http://www.pastebin.ca/1407311 |
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| 13:15.36 | [TK]D-Fender | mort_gib: WOOMERA? Can't help you there. Whats the underlying protocol? |
| 13:16.05 | mort_gib | [TK]D-Fender: I don't think the problem is WOOMERA or ISDN2 |
| 13:16.37 | mort_gib | I have an identical entry for another number that works 100% |
| 13:16.38 | [TK]D-Fender | mort_gib: I can fully believe that kind of error out of a channel driver |
| 13:16.49 | [TK]D-Fender | mort_gib: Doesn't mean its stable |
| 13:17.06 | mort_gib | [TK]D-Fender: Sangoma A500 card |
| 13:17.23 | [TK]D-Fender | mort_gib: Enable channel debug and see how far it gets in its processing |
| 13:17.29 | mort_gib | [TK]D-Fender: Because it must work... Ehm |
| 13:17.31 | mort_gib | Hang on |
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| 13:19.07 | mort_gib | [TK]D-Fender: How do I enable channel debug in woomera?? |
| 13:19.27 | [TK]D-Fender | mort_gib: no clue. |
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| 13:20.36 | mort_gib | [TK]D-Fender: Well I can enable "protocol debug" but all other incoming calls work fine |
| 13:21.16 | mort_gib | [TK]D-Fender: Useless Polycoms |
| 13:21.22 | mort_gib | :-) |
| 13:22.05 | [TK]D-Fender | mort_gib: You are unable to debug the protocol that your call is out on... and so far I only see a local channel calling a woomera channel. WTF are you doing blaming a SIP PHONE? |
| 13:23.12 | mort_gib | [TK]D-Fender: I'm kidding, testing another handset now |
| 13:23.33 | mort_gib | [TK]D-Fender: In all honesty I haven't updated the firmware on that handset |
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| 13:33.07 | gambler1 | Has anyone tried 1.6.1.o in production? :) |
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| 13:33.43 | [TK]D-Fender | gambler1: Considering its release < 24 hours ago, WTF are you expecting? :) |
| 13:33.52 | mort_gib | [TK]D-Fender: Is it worth while updating the Polycoms?? |
| 13:34.01 | [TK]D-Fender | mort_gib: depends :) |
| 13:34.09 | timgws | Hey all, does anyone know how I can make Asterisk do a bell or sound every say five minutes? |
| 13:34.12 | gambler1 | [TK]D-Fender: miracle :) |
| 13:34.35 | mort_gib | [TK]D-Fender: Figures, a bit too bland question... |
| 13:34.40 | [TK]D-Fender | timgws: where? how? to who? Why? |
| 13:34.53 | [TK]D-Fender | mort_gib: de rigeur.... |
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| 13:35.56 | timgws | [TK]D-Fender: well, I want to have all calls that are made through a dial() command to have a been every five minutes, because there is one person here where I work at the moment who stays on the phone for sales for at least 120 minutes -.-" |
| 13:36.33 | gambler1 | [TK]D-Fender: considering that I already found two bugs... just want to be sure that it's not only me... |
| 13:36.37 | [TK]D-Fender | timgws: And what would a bell do for you? Who is supposed to hear it? |
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| 13:37.01 | [TK]D-Fender | gambler1: Of course its only you... it was released < 24 hours ago |
| 13:37.15 | [TK]D-Fender | gamWe don't have time to have caught up to your silly decisions! |
| 13:37.39 | timgws | [TK]D-Fender: only the person who is making the call, and it would (I hope) help the person to be on the phone for a lesser amount of time |
| 13:38.18 | [TK]D-Fender | timgws: First, do you think this person is unaware they are on the phone for a long time? Do you think they care? |
| 13:38.44 | timgws | well, [TK]D-Fender, said person asked me if I would be able to do it so that he notices it more :) |
| 13:38.57 | [TK]D-Fender | timgws: But of course the solution to your question is "core show application dial" <----- |
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| 13:39.25 | gambler1 | [TK]D-Fender: hehehe... I never said it was in production.. but nevermind.. cheers. |
| 13:41.29 | timgws | [TK]D-Fender: but there is nothing about sending any sounds every x minutes :/ |
| 13:41.42 | timgws | except for maybe L |
| 13:42.01 | timgws | but I don't want to put a hard limit on the call, just a warning bell / whatever |
| 13:42.19 | [TK]D-Fender | timgws: Please use your imagination... |
| 13:42.53 | timgws | [TK]D-Fender: my imagination was lost the second I started writting an open source asterisk billing application xD |
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| 13:43.21 | [TK]D-Fender | timgws: then you hint, and todays magic phrase are "arbitrarily large number" <- |
| 13:43.45 | timgws | <3 well, I was thinking that :P |
| 13:43.49 | timgws | *but*, what if :P |
| 13:44.03 | timgws | I mean, this guy can do *very* long calls just for a small sale xD |
| 13:44.12 | [TK]D-Fender | timgws: 86400000ms = 1 day. If the fucker wants to keep on going let him call back |
| 13:44.23 | [TK]D-Fender | :D |
| 13:44.39 | [TK]D-Fender | timgws: I said LARGE |
| 13:44.54 | [TK]D-Fender | timgws: and I started with an entire day. |
| 13:45.49 | [TK]D-Fender | timgws: Imaging if this guy actually sued a cellphone... I can picture the battery redundant model he'd have to have to handle swaps to support his interpersonal communication ineptitude :) |
| 13:45.58 | [TK]D-Fender | imagine* |
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| 13:48.31 | agx | Does asterisk 1.6 support routing of SIP "MESSAGE" between SIP clients? |
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| 13:52.01 | [TK]D-Fender | agx: last I checked, no. |
| 13:53.03 | agx | [TK]D-Fender: ok ty |
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| 14:19.58 | gandhijee | hey, does anyone here know what AC Ring Trip is? |
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| 14:29.11 | coppice | yes, thanks |
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| 14:30.55 | dnikulin | hi all |
| 14:31.11 | dnikulin | have anybody heard about nexspan? |
| 14:32.36 | [TK]D-Fender | dnikulin: What about it? |
| 14:32.55 | dnikulin | thanks! okay, here's the problem |
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| 14:33.18 | dnikulin | i have installed asterisk 1.4 and trying to call from nexspan client to asterisk client |
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| 14:33.35 | dnikulin | and see in my logs: Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) |
| 14:33.51 | dnikulin | so, nexspan allow only 723 codec! |
| 14:34.02 | dnikulin | but I know it supports 711 and 729 |
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| 14:34.35 | dnikulin | how can I set it up to make nexspan use 729 codec instead on sip trunk? |
| 14:35.10 | dnikulin | peer - audio=0x1 (g723) - that's it |
| 14:35.28 | [TK]D-Fender | dnikulin: THEY are only offering G.723. you have to tell THEM to offer something more. This is not *'s job |
| 14:35.38 | [TK]D-Fender | dnikulin: Go read your manual |
| 14:36.04 | dnikulin | there's no good manual.. |
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| 14:36.19 | [TK]D-Fender | dnikulin: And you should not be offering G.723 unless you have a TC400 in your server |
| 14:36.35 | [TK]D-Fender | dnikulin: Go complain to the manufacturer then |
| 14:36.50 | dnikulin | I just wanna understand if it is possible to make nexspan offer 729, no 723. |
| 14:37.07 | dnikulin | manufacturers are french, I dunno french at all |
| 14:37.18 | dnikulin | mayby somebody had the same problem |
| 14:37.28 | [TK]D-Fender | dnikulin: Aastra sure seems multinational to me. |
| 14:38.30 | [TK]D-Fender | dnikulin: and same "problem"? What problem? You don't know how to manage your device and add "inability to communicate witht he manufacturer" to the list of reasons. |
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| 14:40.19 | HenrikBe | is it possible to run a php script when a SIP-phone (x-lite) connects to asterisk? |
| 14:40.50 | dnikulin | the problem is "I installed the device and it works not proper" and if you hadn't the same problem so just keep quiet and there is no need to show me that can help |
| 14:41.18 | [TK]D-Fender | dnikulin: The odds of anyone here using that proprietary platform is beyond slim. |
| 14:41.23 | dnikulin | i can my device, i did it, i am converting 723 to 729 |
| 14:42.58 | dnikulin | here is 268 user online and I just wanna know if someone got success in setting up nexspan to work with asterisk via 729 codec |
| 14:43.28 | [TK]D-Fender | HenrikBe: Using "regexten" in sip.conf you might be able to poll the dialplan via AMI/etc to check for the arrival of the target exten in that context and use as a trigger, but I don't know anything "event driven" to do this |
| 14:45.34 | HenrikBe | TK: Ok, thanks for the tip! |
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| 15:01.00 | ghento | Morning all. I'm trying to figure out how to get Transfer() to work properly. I place an outgoing call to a mobile (via SIP), once connected, I attempt to use Transfer() to another mobile (SIP). My goal is to connect the two calls together. The first call connects fine, but when I do the Transfer(), it rings once for the second phone, and then stops completely. The first call then just goes onto the next priority in the dialplan. Am I using Transfer prop |
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| 15:04.22 | UQlev | hi, anybody had the issue when clamav reports zoiper.exe (free IAX/SIP client) as trojan.packed-142? |
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| 15:06.28 | lirakis | ahoy |
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| 15:09.14 | telnettech | anybody know how I can capture, besides thru the CLI sip debug, whether an extension is registered or not? |
| 15:09.55 | telnettech | at a particular time when a call is made to it? Anyway to log it? |
| 15:10.21 | UQlev | telnettech: did you try sip show peers? |
| 15:10.27 | lirakis | telnettech: tshark port 5060 -w sip.cap |
| 15:10.59 | lirakis | telnettech: wireshark to view the cap file |
| 15:11.22 | telnettech | here is the scenario: We have an autmated wakeup call sent to a guest room. We record if the call is answered, busy, or no answer |
| 15:11.42 | telnettech | we are getting reports that the wakeup call failed due to the phone being busied |
| 15:12.06 | telnettech | i mean it is possible that the phone is busy at the time of the call but highly unlikely since these are wakeup calls |
| 15:12.18 | djcdjc | heh.. if its busy they are already awake |
| 15:12.27 | djcdjc | unless they knocked the phone off the hook in their sleep |
| 15:12.43 | telnettech | so I am looking for a way to possible capture the registration status of the phone at the time that the wakeup call is placed and log this in our log file |
| 15:12.54 | lirakis | telnettech: capture the sip messaging to see what the actual responce is or if its a 504 timeout |
| 15:13.14 | djcdjc | or make them 'press 1 to acknowledge' |
| 15:13.15 | djcdjc | or something |
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| 15:13.43 | djcdjc | retry three times over 5 minutes, if they dont acknowledge alert a clerk to manually chek the extension? |
| 15:13.48 | telnettech | thats the thing....guest says that they never recieved the call, which is what the logs point to but they also say that they werent on the call |
| 15:13.49 | lirakis | telnettech: use tshark, write to a file. after a call goes out - look at the capture to see if the far end responded. |
| 15:14.09 | muiro | lol, I finally got the asterisk <-> shoretel trunk working |
| 15:14.15 | telnettech | dj: we do have it set for 3 time but all 3 show busy |
| 15:14.23 | lirakis | telnettech: the messaging will tell you what is happening 100% for sure. |
| 15:14.44 | telnettech | but how to I run wireshark 24/7 |
| 15:14.56 | djcdjc | well, add the 'if it fails alert a clerk to manually try/check' |
| 15:14.57 | telnettech | i would think that the file would be huge |
| 15:15.01 | lirakis | telnettech: tshark port 5060 -w sip.cap |
| 15:15.01 | mort_gib | telnettech: Try three times, if unsuccessful call reception and alter them to the sleepy looser |
| 15:15.12 | lirakis | that will run tshark and write output to sip.cap |
| 15:15.13 | mort_gib | s/alter/alert |
| 15:15.22 | djcdjc | alter is good ;) |
| 15:15.38 | mort_gib | djcdjc: Yeah :-) |
| 15:15.44 | [TK]D-Fender | telnettech: "core show function SIPPEER" |
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| 15:15.54 | djcdjc | While I do run asterisk. I actually came here looking for someone that I understand might hang out here some times |
| 15:16.01 | djcdjc | sorry to be OT and all |
| 15:16.28 | telnettech | TK: I can add that to my dialplan and capture it to the log file? |
| 15:16.59 | djcdjc | His name is Eric, but he also goes by Trenton.. friend of mine from a long time ago that I lost track of. If anyone here is him msg me |
| 15:17.25 | [TK]D-Fender | telnettech: Go read |
| 15:17.35 | telnettech | ok i will |
| 15:17.37 | muiro | Question about sip peers. The machine I'm running asterisk on has two NICs. How can I choose which interface to route the data through? I cannot do this with plain linux routing because the destination host is the same. It's a little complex... Is there a way to do this possibly in sip.conf? |
| 15:18.23 | [TK]D-Fender | muiro: Nope. Multi-homed * = severe PITA |
| 15:18.43 | muiro | [TK]D-Fender: damn :(. I hate working with shoretel! |
| 15:20.08 | muiro | I want to trash this shoretel box and make our phone system asterisk so badly |
| 15:21.55 | [TK]D-Fender | muiro: point * to a proxy and have that route accordingly. |
| 15:22.05 | [TK]D-Fender | muiro: Or multiple * instances. VM's perhaps |
| 15:23.03 | UQlev | hi, anybody had the issue when clamav reports zoiper.exe (free IAX/SIP client) as trojan.packed-142? |
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| 15:25.26 | telnettech | TK: Thats exacly what im looking for....thanks once again |
| 15:26.46 | muiro | yeah, I have the multiple instances already set up. No VM required. but setting up and running a new * instance for each trunk I need to go to shoretel ... Proxy is probably the only way. |
| 15:28.20 | muiro | [TK]D-Fender: it is possible to run multiple * instances without vm's. I've been meaning to write up documentation on how to do it. Not hard really. |
| 15:29.00 | [TK]D-Fender | muiro: just need to change a few paths, safe_asterisk script, etc.... |
| 15:29.20 | [TK]D-Fender | muiro: I have run across it before, but never had to personally do it |
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| 15:30.01 | muiro | [TK]D-Fender: yeah, there's zilch docs to help along with it for people who aren't as... I should say... daring about manageing their systems |
| 15:31.32 | [TK]D-Fender | muiro: Its not a very desirable situation to have to do in the first place |
| 15:31.45 | muiro | it's not so bad. Working out well for use |
| 15:31.47 | muiro | *us |
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| 16:14.23 | Katty | eppigy: :< |
| 16:14.26 | Katty | eppigy: i was right. |
| 16:14.29 | Katty | eppigy: doom happened this morning |
| 16:14.43 | Katty | MY ENTIRE WEEK IS RUINED |
| 16:15.03 | muiro | wut |
| 16:16.52 | guaxinim | got swine flu? |
| 16:17.06 | Katty | ... |
| 16:17.09 | Katty | don't piss me off |
| 16:17.13 | Katty | this is a bad week to piss me off |
| 16:17.25 | muiro | what happened little one |
| 16:17.28 | guaxinim | just wondering what is so bad, sorry =( |
| 16:17.39 | Katty | my stomach hurts |
| 16:17.41 | Katty | my back hurts |
| 16:17.43 | Katty | and i am cranky |
| 16:17.45 | Katty | what do you think |
| 16:17.51 | muiro | poor baby :( |
| 16:18.21 | muiro | wow that came out like 30 times more condescending than I meant, lol |
| 16:18.34 | muiro | I only meant to be a little condescending |
| 16:19.11 | watchy | Katty: sounds like you just need hugs |
| 16:19.35 | muiro | or sex |
| 16:19.41 | Katty | <PROTECTED> |
| 16:19.52 | guaxinim | what happened? |
| 16:20.53 | muiro | I can help you out with the sex part if you like |
| 16:20.59 | muiro | maybe |
| 16:21.01 | guaxinim | lol |
| 16:21.04 | *** join/#asterisk Thiago_Lima (n=chatzill@200.159.31.7) |
| 16:21.17 | Katty | that's just gross. |
| 16:21.24 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 16:21.28 | Katty | for all you know i'm a 70 year old man |
| 16:21.37 | watchy | i'm cool with it |
| 16:21.44 | guaxinim | "irc: where men are men, women are men and little girls are FBI agents." |
| 16:21.55 | coppice | Katty: I always assumed you were much older than that |
| 16:22.01 | *** join/#asterisk ingenius (n=alektro@host57.190-138-60.telecom.net.ar) |
| 16:22.49 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 16:22.53 | n3hxs | You look pretty good for a 70 Y/O |
| 16:23.08 | watchy | she works out |
| 16:23.19 | n3hxs | lots of dye for the hair too. |
| 16:23.39 | Katty | :< |
| 16:25.59 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 16:26.09 | *** join/#asterisk bbryant_ (n=bbryant@m425a36d0.tmodns.net) |
| 16:26.27 | [TK]D-Fender | When did this become DalNET? |
| 16:26.38 | Pan3D | heh |
| 16:27.04 | muiro | Katty: that's why I said "maybe" |
| 16:28.36 | jaytee | mmmm, Rally's Big Buford and chili cheese fries. nom nom nom nom |
| 16:28.45 | muiro | now I want that :\ |
| 16:28.55 | Katty | ihad beef stew. |
| 16:28.58 | Katty | and some doritos |
| 16:29.06 | eppigy | poor Katty :< |
| 16:29.15 | muiro | yeah that sounds awful |
| 16:29.32 | eppigy | I passed my ICND1 though! |
| 16:29.37 | eppigy | 8[] |
| 16:29.38 | jaytee | wish I had some of my homemade beef stew but that takes about 8 hours to cook in a crockpot |
| 16:29.54 | jaytee | congrats, dave. what's an ICND1? |
| 16:30.04 | eppigy | 1/2 of the cisco CCNA |
| 16:30.09 | eppigy | or the whole CCENT |
| 16:31.04 | jaytee | cool! |
| 16:31.48 | [TK]D-Fender | eppigy: Congrats |
| 16:31.57 | eppigy | thanks :D!!!! |
| 16:33.16 | [TK]D-Fender | goes for ANOTHER martial arts exam on Sunday morning. |
| 16:33.21 | [TK]D-Fender | thats 2 in 1 month |
| 16:33.23 | [TK]D-Fender | 3* |
| 16:35.28 | *** join/#asterisk telnettech (n=telnette@gw.percipia.com) |
| 16:38.07 | eppigy | DANIEL SON |
| 16:38.15 | eppigy | TOO MANY MIND |
| 16:41.06 | telnettech | anybody know if sip registrations are logged in Asterisk? |
| 16:41.26 | telnettech | and if so where they would be logged |
| 16:42.35 | [TK]D-Fender | telnettech: nope |
| 16:42.59 | rob0 | Speaking of which, are there any other Asterlink users who are having registration failures? |
| 16:45.59 | *** join/#asterisk Mw3 (i=mw3@ip599348cd.rubicom.hu) |
| 16:53.54 | *** join/#asterisk agx (n=badpengu@88-149-224-135.dynamic.ngi.it) |
| 16:59.24 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 17:02.27 | *** join/#asterisk bbryant (n=bbryant@m425a36d0.tmodns.net) |
| 17:05.42 | *** join/#asterisk Ng (n=cmsj@nurukipa.tenshu.net) |
| 17:06.00 | Ng | this might sound like a stupid question, but where does asterisk get its time from? |
| 17:06.52 | Ng | I have a GotoIfTime which stops at 18:00, but it seems to match until abotu 18:02 |
| 17:07.02 | Ng | (going by the system clock, which is synced well with ntp) |
| 17:10.43 | *** join/#asterisk toorima (n=bq@ip68-7-79-241.sd.sd.cox.net) |
| 17:10.54 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 17:13.35 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 17:16.43 | rob0 | the TZ variable can be set before you start asterisk, and controls the time zone |
| 17:16.59 | rob0 | this is a glibc matter, in GNU/Linux. |
| 17:17.40 | rob0 | 2 minutes off, that is strange. |
| 17:19.08 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 17:23.19 | *** join/#asterisk HorizonXP (n=xitij@75-119-225-71.dsl.teksavvy.com) |
| 17:23.44 | HorizonXP | hey, is it relatively straightforward to configure asterisk to dial outbound calls using SkypeOut? |
| 17:23.54 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 17:23.54 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:24.06 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:31.59 | [TK]D-Fender | HorizonXP: No, you need some 3rd party connector because * does not speak "Skype" yet and will soon with a licensed connectory |
| 17:35.03 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.231.210) |
| 17:35.15 | HorizonXP | ah i see |
| 17:35.41 | HorizonXP | ok, so here's another question: is it straightforward to be able to have * send and receive Skype-to-Skype calls? |
| 17:36.32 | [TK]D-Fender | HorizonXP: No, thats the same question effectively. * does NOT speak "Skype" <- |
| 17:36.34 | mykhyggz | <PROTECTED> |
| 17:36.42 | denon | ~skype |
| 17:36.42 | infobot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 17:36.49 | denon | ~skypeforasterisk |
| 17:36.49 | infobot | [~skypeforasterisk] is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.astricon.net/skype for beta details. |
| 17:37.43 | HorizonXP | so the Skype for SIP that was announced in March isn't an option then? |
| 17:38.05 | [TK]D-Fender | HorizonXP: Not yet, and no ETA announced |
| 17:38.08 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 17:38.23 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
| 17:38.43 | HorizonXP | ok, but presumably, that would be a future option |
| 17:38.58 | [TK]D-Fender | HorizonXP: In the future we'll all be telepathic as well. |
| 17:38.59 | HorizonXP | but right now, I'm SOL because I'd need a proprietary driver. |
| 17:39.19 | eppigy | i can help you change tired moments into pleasure |
| 17:39.42 | kc8pxy | HorizonXP: only if you want to make skype calls. there are plent of other channel drivers available :) |
| 17:39.44 | jpcansa | does anybody have an idea why when i change my SIP extension to another context i lost my xfer soft button on my linksys spa942? these are my sip.conf and extensions.conf. test ext. 5004 |
| 17:40.23 | HorizonXP | gotchya |
| 17:40.34 | jpcansa | http://pastebin.com/m33700ffe |
| 17:40.38 | HorizonXP | so i may as well go with a DID provider for now, and deal with their rates |
| 17:40.45 | HorizonXP | for my local and long distance calling |
| 17:42.25 | [TK]D-Fender | jpcansa: * can't make your phone lose those buttons, its controlled by the phone |
| 17:43.30 | *** join/#asterisk SparFux (n=raoul@f050021136.adsl.alicedsl.de) |
| 17:44.30 | jpcansa | [TK]D-Fender, weird, if i set ext 5004 to context phones it shows all buttons, but on simple context "test" it lost xfer |
| 17:44.57 | SparFux | I have a severe problem here. linux-call-router notices DTMF in phone calls from the voice of the girl hanging on the phone. Sometimes it even tries to transfer the call due to this when appropriate DTMF is falsely detected. I cannot find the problem in linux-call-router so I want asterisk to NOT do anything with any detected dtmf tone on external lines. How can I achieve this? |
| 17:46.01 | [TK]D-Fender | SparFux: * only cares about DTMF if you tell it to in your DIAL |
| 17:46.33 | *** join/#asterisk cp5 (n=samy@72.37.252.206) |
| 17:46.37 | cp5 | seanbright: around? |
| 17:46.59 | *** part/#asterisk HorizonXP (n=xitij@75-119-225-71.dsl.teksavvy.com) |
| 17:47.22 | SparFux | I only use TW in dial options, and the calling user is me. not her! |
| 17:48.58 | SparFux | ah, no, SHE is the calling user! |
| 17:49.47 | Chainsaw | wonders what a girl with a dual-modulated voice would sound like |
| 17:50.32 | muiro | like the kind of girl I'd like to not talk all that often when I'm over |
| 17:50.43 | muiro | though, that wouldn't make her really stand out, bahaha |
| 17:50.57 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 17:51.18 | Qwell | cp5: ? |
| 17:51.55 | Qwell | cp5: I hear you need to work on your uppercut |
| 17:52.10 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 17:52.15 | cp5 | Qwell: hey, how's it goin? |
| 17:52.21 | cp5 | Qwell: hah, it's true. i am working on it |
| 17:53.55 | SparFux | That's bad. I would have to differentiate which call to use the option with and which not. It would be much easier if I could say which extensions should use these and that opptions and which not! |
| 17:54.22 | seanbright | cp5: i am |
| 17:55.49 | cp5 | seanbright: hey! fyi, ran into something weird with that patch...if i have autofill enabled, and i send 2 calls into a queue, the 1st call hits an agent, the 2nd call does NOT go to an agent until the first call has ended |
| 17:56.03 | [TK]D-Fender | SparFux: You can. Its your dialplan, so get your hands off your nuts and seize the day! |
| 17:56.07 | SparFux | This is the output of asterisk: http://pastebin.com/d614fbf5 |
| 17:56.08 | cp5 | even though autofill is enabled. i reverted that patch and it behaved normally |
| 17:56.32 | SparFux | Fender: with all this, Callerid and so on, the dialplan gets more and more complicated. |
| 17:58.29 | *** join/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 17:58.31 | mintee | ~book |
| 17:58.32 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:58.38 | *** part/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 18:00.12 | seanbright | cp5: in... what version of asterisk? |
| 18:00.12 | [TK]D-Fender | SparFux: well You seem to leave out all the normal bits like verbose in there. Doesn't help. Also don't see configs. |
| 18:00.29 | SparFux | Well, I think the dtmf problem is common. You can find some issues with several ATAs on the net. |
| 18:00.59 | SparFux | verbose? I did asterisk -r -vvv |
| 18:01.30 | SparFux | hehehe. |
| 18:01.35 | kc8pxy | SparFux: it's common IMHO, to use ~10 v's |
| 18:01.39 | SparFux | she gave me a sound sample and got transferred ;-) |
| 18:01.54 | *** part/#asterisk jes-o-mat (i=jesusch@irc.82110clan.de) |
| 18:02.53 | SparFux | 10 v's don't seem to give me more output. |
| 18:03.17 | [TK]D-Fender | SparFux: And I don't see your DIAL in that output now do I? |
| 18:03.23 | [TK]D-Fender | SparFux: Or anything else |
| 18:03.56 | SparFux | perhaps I simply post my dialplan? |
| 18:04.09 | cp5 | seanbright: in 1.2 (i actually have autofill ported and working in 1.2), i will test in 1.6 though and see if the same thing happens |
| 18:04.49 | [TK]D-Fender | SparFux: I want to see the CALL |
| 18:05.09 | SparFux | I have to search for it. |
| 18:05.13 | SparFux | I will find it... |
| 18:05.49 | jpcansa | [TK]D-Fender, can i control soft buttons with the dialplan? |
| 18:05.55 | seanbright | cp5: ahh, i see. well i can't think of a reason my patch would affect that. |
| 18:06.00 | SparFux | exten => s,1,Dial(${ARG2},${ARG3},${DDOPT}) and DDOPT is DDOPT=TW |
| 18:06.23 | [TK]D-Fender | jpcansa: does the phone say you can signal it live? |
| 18:06.32 | seanbright | cp5: do you still have your autofill backport patch floating around? |
| 18:06.38 | cp5 | seanbright: yeah, me neither. i even went back without the patch to verify but it doesn't happen without the patch, really weird |
| 18:06.52 | [TK]D-Fender | jpcansa: I seriously doubt it. |
| 18:07.03 | SparFux | But the basic call goes like this: exten => s,n(loud),Macro(no-mailbox,${PSTN_MSN4},${BEAKER_SOPH}&${CALL_BEAKER_MOBILE}&${BEAKER_ATA},60,${BEAKER_EMAIL}) |
| 18:07.39 | SparFux | where the phone I pickup is BEAKER_ATA=sip/SPA2 |
| 18:08.16 | SparFux | wait a minute! only if I pickup phones connected to my ATA will I get the falsely detected dtmf tones! |
| 18:08.24 | SparFux | What do I make out of this? |
| 18:08.31 | rob0 | Okay, I've done "sip set debug peer asterlink", and I see they're just ignoring my registration attempts. The host proxy-01.asterlink.com. pings, but "nmap -sU -p5060 proxy-01.asterlink.com." shows "5060/udp closed sip". I guess that's why I can't register! |
| 18:09.09 | *** join/#asterisk Mw3 (i=mw3@ip599348cd.rubicom.hu) |
| 18:10.20 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 18:11.26 | rob0 | Oh yes. It seems that AT&T is blocking SIP!! |
| 18:11.32 | seanbright | cp5: i _may_ know what the problem is |
| 18:11.39 | seanbright | cp5: willing to test out another patch? |
| 18:11.40 | cp5 | oh yeah? |
| 18:11.55 | cp5 | seanbright: sure |
| 18:11.56 | seanbright | cp5: gimme a sec to cook one up |
| 18:11.58 | rob0 | I nmap'ed my own host, which I know is not blocking me, and I get the same result. |
| 18:12.03 | cp5 | seanbright: use me and abuse me |
| 18:13.06 | [TK]D-Fender | rob0: \o/ |
| 18:13.29 | *** join/#asterisk Natanaiel (n=Unknown@unaffiliated/natanaiel) |
| 18:13.46 | seanbright | cp5: http://pastie.org/462819.txt?key=cwsytr2idfkigvodvtajjw |
| 18:13.55 | Natanaiel | where is the main function in the asterisk code? |
| 18:14.05 | Qwell | Natanaiel: asterisk.c? |
| 18:14.22 | seanbright | main/asterisk.c |
| 18:15.12 | Natanaiel | tnx Qwell & seanbright |
| 18:15.32 | seanbright | Qwell did most of the work |
| 18:15.41 | Qwell | it was difficult |
| 18:15.45 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-2c73d86e21a40f0d) |
| 18:15.58 | seanbright | Qwell: still going strong with the e-cig? |
| 18:16.04 | Qwell | mmhmm |
| 18:16.17 | seanbright | where'd you get it? |
| 18:16.25 | Qwell | intarwebs |
| 18:16.27 | Qwell | via china |
| 18:16.36 | seanbright | are they illegal in the states? |
| 18:16.44 | Qwell | no |
| 18:17.03 | seanbright | "In the United States, the Food and Drug Administration (FDA) considers electronic cigarettes to be a nicotine delivery system, subject to its approval. The agency is currently investigating electronic smoking devices, and has barred their import into the US." |
| 18:17.15 | seanbright | don't lie to me, boy. |
| 18:17.20 | cp5 | seanbright: i am a terrible person, i see the issue. the patch got rejected because of my own changes, so i manually patched. on the 2nd ast_copy_string, i used "curint->interface" instead of "cur->interface" |
| 18:17.20 | Qwell | that != illegal |
| 18:17.25 | muiro | doesn't mean they're illegal to posses |
| 18:17.26 | seanbright | semantics |
| 18:17.26 | muiro | just import |
| 18:17.37 | *** part/#asterisk Natanaiel (n=Unknown@unaffiliated/natanaiel) |
| 18:17.37 | seanbright | cp5: how dare you. |
| 18:17.37 | muiro | but you can buy them overseas and get them shipped |
| 18:17.45 | Qwell | you can buy them here too |
| 18:17.53 | seanbright | cp5: the newer patch is actually better, fwiw. |
| 18:17.54 | cp5 | seanbright: :( thank you for accepting my invalid accusations and still trying to help |
| 18:18.01 | cp5 | seanbright: cool, then i'll use it |
| 18:18.02 | seanbright | cp5: no sweat. |
| 18:18.09 | cp5 | thanks again |
| 18:18.12 | seanbright | np |
| 18:19.00 | cp5 | seanbright: it's not too intensive to recreate that char array every single time? this thing may go through *a lot* of queue members |
| 18:19.11 | rob0 | Okay, as usual, I jumped the gun. I can get through on 5060/udp with nc(1) to a host I control. |
| 18:19.21 | cp5 | seanbright: i would assume your original patch is more efficient since it only needs to create it once? |
| 18:19.44 | seanbright | cp5: it's not actually creating the array each time |
| 18:19.50 | seanbright | cp5: it's just scoped differently |
| 18:19.59 | cp5 | seanbright: ok, cool |
| 18:20.00 | [TK]D-Fender | rob0: You're expecting us to basically cheer-lead this right? |
| 18:21.48 | rob0 | haha ... guess what ... it works now! |
| 18:24.33 | seanbright | cp5: i may not be 100% correct on that, actually. |
| 18:24.39 | seanbright | use whichever patch makes you happier :) |
| 18:28.30 | *** join/#asterisk pbx1 (n=pbx1@203.82.38.122) |
| 18:31.27 | *** join/#asterisk b14ck (n=comradeb@72.37.252.50) |
| 18:33.33 | bpgoldsb | Whats the difference betwen cdr_mysql (from asterisk-addons) and cdr_odbc using mysql? |
| 18:33.46 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:33.58 | seanbright | one is native and the other uses odbc |
| 18:34.03 | seanbright | what do i win? |
| 18:34.14 | Qwell | A BRAND NEW CAR! |
| 18:34.15 | bpgoldsb | Hmm. I have some mini snickers on my desk. |
| 18:34.19 | Qwell | err, wrong show |
| 18:34.32 | seanbright | the correct answer is: ANOTHER BEER! |
| 18:34.46 | bpgoldsb | Mmmm, beer. |
| 18:34.54 | bpgoldsb | Beer makes Asterisk so much hotter. |
| 18:34.59 | thehar | yes it does |
| 18:35.02 | Corydon76-dig | hands Qwell the keys to a Yugo |
| 18:35.27 | pbx1 | I have a sangoma A104 all configured to fxs channels and connected to fxs channel banks |
| 18:35.48 | pbx1 | the problem is I have channel banks that restart for some reason under load |
| 18:35.59 | pbx1 | anybody ever have an issue like this? |
| 18:36.35 | pbx1 | span 1 and 2 are fine, had the same channel banks for 3 years |
| 18:36.52 | pbx1 | but the channel banks I got for span 3 and 4 both restart. |
| 18:37.03 | pbx1 | is this a channel bank issue or an asterisk issue? |
| 18:37.08 | pbx1 | thanks for any help |
| 18:37.27 | *** join/#asterisk ddickenson (n=ddickens@67-198-0-5.static.grandenetworks.net) |
| 18:37.51 | Corydon76-dig | pbx1: channel bank issue |
| 18:37.53 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:38.00 | [TK]D-Fender | pbx1: Swap them around and prove if its the port or the CB at fault |
| 18:38.21 | pbx1 | I did that, the channel banks I had for 3 years works on all ports fine |
| 18:38.43 | [TK]D-Fender | pbx1: Does the bad one fail on all ports? |
| 18:38.59 | pbx1 | yes, the bad one is an adit 600 and fails on all ports |
| 18:39.10 | [TK]D-Fender | pbx1: Sounds obvious to me. |
| 18:39.16 | pbx1 | so I ordered 2 CAC ABII and it still fails |
| 18:39.24 | [TK]D-Fender | pbx1: Don't see why this is even a question |
| 18:40.00 | [TK]D-Fender | pbx1: Well you just confirmed the CB failed on all ports and the good one works on all ports. Clearly you are buying dead/ano/or/dying CBs |
| 18:40.23 | pbx1 | is there a way to confirm the channel banks are dead for sure? |
| 18:40.43 | pbx1 | I hear what your saying, must be the channel bank, but it's strang that this happens |
| 18:40.50 | pbx1 | they work for about an hour, then reboot |
| 18:41.41 | ddickenson | anyone have experience with wanpipe install? I recently decided that using sangoma cards was an extra point of failure I didn't need but since I have two matching t1 cards I figured I'd try to use them for the small install I'm doing |
| 18:41.47 | [TK]D-Fender | pbx1: This is "Scientific Method 101" |
| 18:41.50 | n3hxs | Prove the point by swapping, if the problem follows the cable it is the telco. |
| 18:42.25 | pbx1 | I understand, I have to test and all. but I dont' have the experience of doing this often |
| 18:42.37 | n3hxs | you soon will :) |
| 18:42.42 | pbx1 | so I'd like to hear from you the expert if you've experience bad channel banks |
| 18:43.12 | pbx1 | if the behavior of a bad channel bank is to restart for no reason after a an hour or so of use |
| 18:43.32 | seanbright | well i'd hope that wasn't the behavior of a good channel bank |
| 18:43.34 | b14ck | pbx1, i have a bit of advice that may help |
| 18:44.05 | b14ck | monitor the cb, if you notice that it is getting very hot, this may be your problem. many of these devices will reboot automatically if the temp is too high. its built into their mobos |
| 18:44.18 | b14ck | shrug |
| 18:44.38 | pbx1 | thank you b14ck, that helps. see I need that kind of experience. |
| 18:44.41 | pbx1 | I didn't know that could happen |
| 18:44.50 | b14ck | thats a common occurence |
| 18:44.54 | b14ck | happens with routers too |
| 18:44.58 | b14ck | its a safeguard |
| 18:45.04 | b14ck | so if you have high load, you are surely using a lotta cpu |
| 18:45.10 | b14ck | and may stress it out a bit |
| 18:45.18 | pbx1 | so if I have 2 channel banks that work fine on the in the same room |
| 18:45.26 | b14ck | lookup some info about your specific cb and see if you can find anything related to how much load it can handle, etc |
| 18:45.36 | pbx1 | but the others reboot, it could just be one channel bank is more sensitive? |
| 18:45.46 | *** join/#asterisk bbryant (n=bbryant@m425a36d0.tmodns.net) |
| 18:45.50 | b14ck | possibly, it could be that the cb which is rebooting has more traffic over it as well |
| 18:46.17 | pbx1 | the one that is fine has all channels in use, and has never rebooted even when the A/C broke once |
| 18:46.33 | pbx1 | but the new ones I have reboot after about an hour of use under same temp as the working |
| 18:47.03 | b14ck | watch one of the ones that is rebooting. feel it when it reboots (for temperature). if its extremely hot, thats most likely the culprit |
| 18:47.06 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:47.27 | b14ck | maybe the new cb's you purchased dont have decent fans for cooling the cb's |
| 18:47.35 | b14ck | who knows |
| 18:47.48 | pbx1 | the channl banks are fanless, adit 600 |
| 18:47.58 | b14ck | all of them? |
| 18:48.03 | pbx1 | I did feel them and they felt not very too hot |
| 18:48.18 | pbx1 | the others that work ar cac Access bank II |
| 18:48.25 | pbx1 | and they feel warm, but never reboot |
| 18:48.32 | b14ck | hm |
| 18:48.38 | b14ck | its probably not a heating issue then |
| 18:48.48 | b14ck | if they dont feel hot, i doubt that would cause rebooting |
| 18:49.03 | b14ck | but regardless, it shouldnt be an asterisk problem causing them to reboot |
| 18:49.17 | b14ck | asterisk can't do anything to make other devices reboot |
| 18:49.26 | pbx1 | yeah figured it couldnt be asterisk |
| 18:49.46 | b14ck | ya, tricky problems like these are always a pain |
| 18:49.54 | b14ck | because, lets say you figure out that it is the channel banks failing |
| 18:50.01 | pbx1 | could the cause be bad channel bank configuration? |
| 18:50.05 | b14ck | you call the company that you purchased from, and they blame it on other stuff, heh |
| 18:50.13 | *** part/#asterisk ddickenson (n=ddickens@67-198-0-5.static.grandenetworks.net) |
| 18:50.21 | pbx1 | yeah, in the process of complaining |
| 18:50.21 | b14ck | yea |
| 18:50.30 | b14ck | but im not familiar with those cb's |
| 18:50.37 | b14ck | so i have no idea how they are configured |
| 18:50.46 | pbx1 | so if the channel banks are configured the same as my others, |
| 18:50.55 | pbx1 | must be a bad channel bank? |
| 18:51.08 | b14ck | that's what seems to be the correct logical answer. i'd say so, yes |
| 18:51.20 | pbx1 | that was my worry |
| 18:51.42 | pbx1 | I know my questoins seem obvoius to you veterans, but I just needed to throw the quesiton out there |
| 18:51.51 | pbx1 | and hope I could get a fix. |
| 18:51.55 | b14ck | no worries, im by no means a veteran myself |
| 18:52.17 | b14ck | good luck getting it working, if you need other stuff just ask :) |
| 18:52.22 | pbx1 | you helped a lot though |
| 18:52.46 | pbx1 | thanks |
| 18:52.52 | b14ck | no problem |
| 18:56.05 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 18:58.18 | n3hxs | I have seen channel banks work until the number of calls gets so great that the CSU can't correct and the T1 loops resetting its self. |
| 18:59.37 | *** join/#asterisk dany303 (n=dany@mip001.dnetx.net) |
| 18:59.54 | *** part/#asterisk agx (n=badpengu@88-149-224-135.dynamic.ngi.it) |
| 19:05.45 | deeperror | Other than using 1-1 and 1-2 is there any other way of determining if a channel is on instance 2? |
| 19:05.53 | *** join/#asterisk MrGabu (n=gbdurant@200-170-192-195.static.spo.ifx.net.br) |
| 19:06.38 | SparFux | There is not more output with 10 -v's! |
| 19:06.40 | MrGabu | hello, someone know what is the probably cause for a "beep" in QSIG while making a call from Digium to TELCO ? |
| 19:10.20 | *** part/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 19:18.55 | *** join/#asterisk agallo (n=badpengu@88-149-225-212.dynamic.ngi.it) |
| 19:20.00 | *** part/#asterisk agallo (n=badpengu@88-149-225-212.dynamic.ngi.it) |
| 19:20.15 | [TK]D-Fender | deeperror: Since its the same physical link, if you can't trust *'s channel designation, what CAN you trust? |
| 19:20.30 | [TK]D-Fender | deeperror: there's been no response to your report? (Yuo DID place it... right?) |
| 19:20.50 | *** part/#asterisk MrGabu (n=gbdurant@200-170-192-195.static.spo.ifx.net.br) |
| 19:24.40 | *** join/#asterisk Juerd (i=juerd@feather.perl6.nl) |
| 19:24.51 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
| 19:25.11 | Juerd | Looking for someone who knows what a single blinking (really fast) LED on a junghanns quadbri card means |
| 19:25.56 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-2eef988346af42be) |
| 19:26.41 | ariel_ | hello folks |
| 19:29.02 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) |
| 19:29.08 | VaGoNeTaS | hello guys |
| 19:29.34 | VaGoNeTaS | i'm looking for some information |
| 19:29.57 | VaGoNeTaS | i need to make a dialplan or something so i can transfer my calls between my agents |
| 19:30.19 | VaGoNeTaS | outbound callcenter, but sometimes my agents would like to transfer their calls between them |
| 19:30.27 | VaGoNeTaS | anybody knows how to do it? |
| 19:30.40 | VaGoNeTaS | for ex: , i have 40 agents |
| 19:30.53 | [TK]D-Fender | VaGoNeTaS: ... |
| 19:30.55 | [TK]D-Fender | ~book |
| 19:30.55 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:30.57 | [TK]D-Fender | ^^^^^ |
| 19:31.25 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de) |
| 19:31.27 | [TK]D-Fender | VaGoNeTaS: have extensions accessible in their context the can call and just DO IT |
| 19:31.33 | *** join/#asterisk PinkFreud (n=WhyNot@75.147.87.197) |
| 19:32.02 | VaGoNeTaS | from 1001, to 1014 |
| 19:32.21 | VaGoNeTaS | mmm |
| 19:32.25 | PinkFreud | hi folks. I'm a bit new to asterisk. Trying to figure out what the difference between 1.4, 1.6.0, and 1.6.1 are. |
| 19:32.31 | VaGoNeTaS | for example, when the agent is in call |
| 19:32.46 | VaGoNeTaS | he would like to transfer the call to the agent 1005 |
| 19:32.52 | [TK]D-Fender | ping go read the "CHANGES.txt" in 1.6.1 and it will break down the major bits |
| 19:32.54 | VaGoNeTaS | whats your suggestion? |
| 19:33.20 | VaGoNeTaS | your suggestion for my extensions.conf of course |
| 19:33.26 | [TK]D-Fender | VaGoNeTaS: Go read the book. You speak like you have absolutely no understanding of the dialplan whatsoever |
| 19:33.57 | PinkFreud | [TK]D-Fender: will do. are any of these considered 'development' or 'testing'? Or are they all stable releases? |
| 19:34.11 | VaGoNeTaS | i do understand, but i havent made an dialplan for xfers yet |
| 19:34.22 | b14ck | VaGoNeTaS, is this for a business or home setup? |
| 19:34.24 | b14ck | or just for fun? |
| 19:34.53 | [TK]D-Fender | PinkFreud: 1.6.0 series is actually fairly stable although many seasoned users are still paranoid of it. 1.6.1 series JUST got released so I would hold off of critical use for now. |
| 19:35.21 | [TK]D-Fender | PinkFreud: 1.4 is largely predictable and required by many back-water apps that haven't updated in a while |
| 19:35.40 | [TK]D-Fender | VaGoNeTaS: There is no such thing as a dialplan for "transfers" |
| 19:35.47 | PinkFreud | hmmm. |
| 19:35.50 | [TK]D-Fender | VaGoNeTaS: You can either call, or not. |
| 19:36.37 | VaGoNeTaS | bl4 |
| 19:36.40 | VaGoNeTaS | bl4ck |
| 19:37.47 | beek | PinkFreud: Love the nick! |
| 19:37.57 | beek | Afternoon [TK]D-Fender |
| 19:38.02 | PinkFreud | beek: hehe. thanks. |
| 19:38.07 | VaGoNeTaS | is business |
| 19:38.40 | VaGoNeTaS | when my agent is on call, he press pound and the extension |
| 19:38.42 | *** join/#asterisk agx (n=badpengu@88-149-226-198.dynamic.ngi.it) |
| 19:38.44 | VaGoNeTaS | and that's it? |
| 19:39.02 | VaGoNeTaS | i dont think so, im pretty sure that i have to make a new dial plan |
| 19:39.10 | [TK]D-Fender | VaGoNeTaS: What are they talking on? |
| 19:39.33 | VaGoNeTaS | so the agent types on his softphone something like, "#<agent extension>" |
| 19:39.43 | [TK]D-Fender | VaGoNeTaS: WHAT soft-phone? |
| 19:39.45 | VaGoNeTaS | and the call gets tranferred to the other ext |
| 19:39.47 | VaGoNeTaS | Xlite |
| 19:40.27 | VaGoNeTaS | i have configured a dial plan for the logon of the agent |
| 19:40.33 | [TK]D-Fender | VaGoNeTaS: Ok, then your dial command needs to have "Tt" in the options for your user to transfer the call to someone else and have them able to transfer it as well |
| 19:40.55 | VaGoNeTaS | i have configured the agents.conf |
| 19:41.05 | VaGoNeTaS | just a min |
| 19:41.07 | [TK]D-Fender | VaGoNeTaS: this has NOTHING to do with agents.conf |
| 19:41.15 | [TK]D-Fender | VaGoNeTaS: All call processing is extensions.conf <- |
| 19:41.21 | VaGoNeTaS | i know that it has nothing to do with agents.conf |
| 19:41.31 | VaGoNeTaS | and i do know that is processed on extensions.conf |
| 19:41.36 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 19:41.36 | VaGoNeTaS | on the* |
| 19:44.11 | VaGoNeTaS | this is my dialplan |
| 19:44.13 | VaGoNeTaS | [default] |
| 19:44.13 | VaGoNeTaS | include => celulares |
| 19:44.13 | VaGoNeTaS | exten => _ZX.,1,NoOp(${EXTEN}) |
| 19:44.13 | VaGoNeTaS | exten => _ZX.,2,Dial(DAHDI/g1/${EXTEN}) |
| 19:44.19 | [TK]D-Fender | VaGoNeTaS: PASTEBIN |
| 19:44.43 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:44.44 | [TK]D-Fender | VaGoNeTaS>exten => _ZX.,2,Dial(DAHDI/g1/${EXTEN},,T) |
| 19:45.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:46.00 | VaGoNeTaS | ok, and how's my agent transferring the calls ? |
| 19:46.17 | *** join/#asterisk deeperror (n=deeperro@adsl-99-33-114-255.dsl.sfldmi.sbcglobal.net) |
| 19:46.47 | Juerd | Does anyone know what a rapidly flashing red led means on a junghanns quadbri card? |
| 19:47.14 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 19:47.26 | [TK]D-Fender | VaGoNeTaS: # then the extension to transfer to. |
| 19:47.43 | VaGoNeTaS | dude, thank you so much for your help |
| 19:48.12 | VaGoNeTaS | i've added the ,,T to my dialplan, reloaded |
| 19:48.23 | VaGoNeTaS | now im gonna make the tests |
| 19:48.33 | VaGoNeTaS | my 2nd thing to do is: |
| 19:48.40 | VaGoNeTaS | as i told you this is an outbound callcenter |
| 19:48.55 | VaGoNeTaS | when my agents calls to a celphone with no answer |
| 19:49.16 | VaGoNeTaS | and then the customer see the missed call, he is gonna return the call to my number |
| 19:49.52 | VaGoNeTaS | i need that when the customer calls back , the calls go straight to the agent that made the call |
| 19:49.54 | VaGoNeTaS | is that possible? |
| 19:50.21 | VaGoNeTaS | or the call will be redirected "random" to the callcenter? |
| 19:51.11 | watchy | if they dont have direct #s then no |
| 19:51.20 | watchy | but if they got direct #s set your CID to the dudes # |
| 19:51.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:51.42 | VaGoNeTaS | what is #s, |
| 19:51.44 | VaGoNeTaS | numbers? |
| 19:51.46 | watchy | yes |
| 19:51.51 | watchy | im being lazy |
| 19:51.54 | VaGoNeTaS | hahaha |
| 19:51.59 | VaGoNeTaS | got it |
| 19:52.12 | watchy | so when the outgoing call is made it'll show up as your agents number |
| 19:52.23 | watchy | but if you don't have direct numbers for each agent its not going to be possible |
| 19:52.48 | VaGoNeTaS | oh |
| 19:52.48 | VaGoNeTaS | no |
| 19:52.52 | VaGoNeTaS | we have only 1 number |
| 19:52.56 | VaGoNeTaS | its a header number |
| 19:53.11 | watchy | you gonna have to have DIDs dude or it wont work |
| 19:53.18 | watchy | how else would it know what agent to goto? |
| 19:53.25 | VaGoNeTaS | so what you are telling me is: |
| 19:53.31 | watchy | every agent needs a DID |
| 19:53.36 | VaGoNeTaS | the customer will return the call |
| 19:53.55 | watchy | and they'll get a random agent since you dont have DIDs |
| 19:53.55 | VaGoNeTaS | and the call will be redirected random to any of my agents |
| 19:54.00 | watchy | yes |
| 19:54.00 | *** join/#asterisk SparFux (n=raoul@e182018120.adsl.alicedsl.de) |
| 19:54.00 | VaGoNeTaS | ? |
| 19:54.02 | VaGoNeTaS | but |
| 19:54.07 | watchy | pay $10 amonth extra for DIDs |
| 19:54.11 | VaGoNeTaS | haha |
| 19:54.16 | VaGoNeTaS | im located in Chile dude |
| 19:54.18 | VaGoNeTaS | is not that cheap |
| 19:54.24 | watchy | can i come work for you for free |
| 19:54.30 | watchy | i wanna get outta the US |
| 19:54.34 | SparFux | Re again. My DTMF problem is well known and it is the problem of mISDN stack since version v1. http://www.isdn4linux.de/pipermail/isdn4linux/2009-April/004053.html |
| 19:54.39 | watchy | just gimme room and food |
| 19:54.53 | VaGoNeTaS | you gotta be kidding me right |
| 19:55.04 | VaGoNeTaS | whats the matter with your country |
| 19:55.11 | VaGoNeTaS | my mother lives there |
| 19:55.14 | watchy | i need to see the world dude |
| 19:55.22 | VaGoNeTaS | and she havent complain yet |
| 19:55.27 | watchy | im ready to quit what i'm doing now and move somewhere more interesting |
| 19:55.31 | [TK]D-Fender | [15:49]<VaGoNeTaS>is that possible? <- maybe |
| 19:55.44 | VaGoNeTaS | TK, with an IVR? |
| 19:55.59 | watchy | hmm, yea actually you could do it that way |
| 19:55.59 | VaGoNeTaS | watchy yes, but thought you people ... i dont know |
| 19:56.02 | [TK]D-Fender | VaGoNeTaS: Not unless the caller knows which agent called |
| 19:56.03 | VaGoNeTaS | you own the world |
| 19:56.20 | VaGoNeTaS | according to your last president.. that mofu GWbush |
| 19:56.22 | watchy | vag: make a inbound script to check the number calling back |
| 19:56.26 | SparFux | At these times you better get out of the US indeed. It's the biggest economy in the world there and the crisis arises! |
| 19:56.33 | [TK]D-Fender | VaGoNeTaS: If you can trust callerid, then you can compare that to a log of which agent called which # |
| 19:56.37 | watchy | if the # was one dialed before check which agent dialed it, send the call back there |
| 19:56.52 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 19:56.52 | VaGoNeTaS | tk the caller shouldnt know |
| 19:56.54 | watchy | wow tk i'm on the same page as you, i don't feel so dmb |
| 19:57.00 | VaGoNeTaS | this is an big drugstore in Chile |
| 19:57.02 | VaGoNeTaS | Salco Brand |
| 19:57.13 | watchy | vago: check the callers CID when they call back |
| 19:57.21 | VaGoNeTaS | so, my agents call the customers to make some offers like credit cards and shit like that |
| 19:57.24 | watchy | if it matches a number an agent dialed, send it back there |
| 19:58.04 | VaGoNeTaS | agents shoudnt know |
| 19:58.07 | VaGoNeTaS | i mean customers |
| 19:58.23 | watchy | can i message you |
| 19:58.27 | watchy | i dont wanna flood the channel |
| 19:58.32 | VaGoNeTaS | sure go ahead |
| 19:59.04 | SparFux | Hey, I am interested in that conversation. Can we open a new channel? |
| 19:59.22 | SparFux | come to #watchusa perhaps? |
| 20:00.00 | watchy | sure |
| 20:00.04 | VaGoNeTaS | k |
| 20:00.07 | VaGoNeTaS | lets go |
| 20:05.34 | *** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27) |
| 20:08.46 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.242) |
| 20:10.51 | VaGoNeTaS | tk |
| 20:11.05 | VaGoNeTaS | now i need to setup the Inbound |
| 20:11.20 | VaGoNeTaS | my header number is 5879700 |
| 20:11.25 | *** part/#asterisk agx (n=badpengu@88-149-226-198.dynamic.ngi.it) |
| 20:12.39 | VaGoNeTaS | need to make a queue in order to forward the calls to an random agent |
| 20:13.00 | watchy | you talkin in #asterisk not the other chan |
| 20:14.41 | Juerd | I found out what the single flashing red light on the junghanns quadbri is caused by: hcfmulti was loaded, but shouldn't have been. |
| 20:16.52 | *** join/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 20:18.00 | VaGoNeTaS | is away: Fell asleep on keyboard... <<eDK/VgN>> [ Logging, Page: On ] |
| 20:19.23 | VaGoNeTaS | is back from the dead. Gone: 1m 22s |
| 20:21.38 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:22.59 | watchy | anything special need to be done to add a dialplan that uses #? |
| 20:24.52 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 20:26.02 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:29.03 | [TK]D-Fender | watchy: just because I'm pretty sure I know where this is going : add "pedantic=yes" to [general] in sip.conf |
| 20:29.05 | [TK]D-Fender | BBIAB |
| 20:29.54 | watchy | i love you. |
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| 20:32.28 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 20:40.05 | *** join/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 20:42.53 | *** join/#asterisk joseph__ (i=CK@93.185.227.132) |
| 20:42.59 | joseph__ | hi guys |
| 20:44.38 | joseph__ | how to put a condition==33123455633 and get this value 8605627811@ from above Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message |
| 20:44.40 | joseph__ | 192.168.5.189 7109613803 0c9bd9187bf 00102/00000 0x100 (g729) No Init: INVITE |
| 20:44.40 | joseph__ | 192.168.5.197 33123455633 8605627811@ 00101/00001 0x100 (g729) No Rx: INVITE |
| 20:45.45 | *** join/#asterisk Globettrotter (n=eric@ool-457a1c8a.dyn.optonline.net) |
| 20:46.52 | Globettrotter | hey,, what is the setting for "signal ringing" its in the freepbx gui,, but how do i configure that setting via the config file? |
| 20:50.25 | joseph__ | ? |
| 20:50.43 | muiro | oops, I forgot I was still connected to this channel on this network, lol, my bad, sorry |
| 20:50.55 | *** part/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
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| 20:55.07 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 20:57.12 | kb3ien | is there a means to set a variable in sip.conf that it will be inherited in extensions.conf as the call progresses through the dialplan? |
| 20:58.55 | [TK]D-Fender | kb3ien: SetVar=myvariable=12345 |
| 21:00.11 | watchy | tk: that didn't seem to do what i wanted to do, I want to get a Directory when i dial #. |
| 21:01.02 | [TK]D-Fender | watchy: care to show me what # receives? |
| 21:01.09 | [TK]D-Fender | * |
| 21:01.13 | watchy | yea hold bro |
| 21:01.25 | watchy | Apr 29 15:54:54] NOTICE[18018]: chan_sip.c:13885 handle_request_invite: Call from '206' to extension '192.168.1.10:5060' rejected because extension not found. |
| 21:01.57 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 21:02.00 | [TK]D-Fender | watchy: PB the whole invite |
| 21:02.44 | watchy | ok |
| 21:07.10 | watchy | http://pastebin.com/m74d5a48d |
| 21:07.52 | SparFux | What does option relaxdtmf=yes do in sip.conf? |
| 21:07.57 | [TK]D-Fender | watchy: To: <sip:192.168.1.10;user=phone> |
| 21:08.05 | [TK]D-Fender | watchy: Well you've clearly screwed up your phone's dialplan |
| 21:08.11 | b14ck | relaxdtmf makes the dtmf detection a bit more lax |
| 21:08.18 | [TK]D-Fender | SparFux: Same thing it does in others |
| 21:08.20 | b14ck | eg: sometimes female voices set off dtmf detection |
| 21:08.34 | b14ck | relaxdtmf will help so that it wont think a voice is a dtmf tone |
| 21:08.34 | b14ck | :) |
| 21:08.35 | [TK]D-Fender | can whistle up a 300 baud carrier |
| 21:08.59 | SparFux | b14ck: Then what's so relaxed about that? |
| 21:09.12 | [TK]D-Fender | kickin' it old-school biatch! |
| 21:09.14 | b14ck | its relaxed because the rules for what is and what isn't a dtmf tone are more 'relaxed' |
| 21:09.19 | b14ck | they arent as strict |
| 21:09.26 | b14ck | oh wait |
| 21:09.28 | b14ck | other way around =p |
| 21:09.37 | watchy | tk: haha |
| 21:09.45 | watchy | i'll look into that tk |
| 21:09.56 | watchy | thanks tho im trying to fiend off customers |
| 21:10.43 | SparFux | b14ck: other way round? ;-P |
| 21:10.59 | b14ck | there 'ya go =) |
| 21:11.35 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 21:11.39 | SparFux | So, I should NOT be relaxed, right? Set it to NO! |
| 21:12.50 | SparFux | But NO is the default anyway. |
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| 21:18.38 | *** part/#asterisk BajaEd (n=ednagy@72.170.62.90) |
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| 21:40.23 | smash- | =/ |
| 21:56.00 | b14ck | why would anyone use a party line now-a-days? |
| 21:56.04 | b14ck | just for the lolz? |
| 21:58.55 | *** part/#asterisk Juerd (i=juerd@feather.perl6.nl) |
| 22:01.56 | cp5 | to *party* obviously |
| 22:02.10 | cp5 | droppin dox, left and right |
| 22:05.31 | *** join/#asterisk desdesdesdes (n=kg@196.211.34.2) |
| 22:05.58 | desdesdesdes | is there an application for asterisk line dimdim webmeeting ? |
| 22:15.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:16.54 | *** join/#asterisk Nasra (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 22:17.30 | desdesdesdes | is there an application for asterisk like dimdim webmeeting confrencing ? |
| 22:20.39 | *** part/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 22:20.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:21.09 | [TK]D-Fender | desdesdesdes: Probably juust as many as there were 12 minutes and 8 lines ago. |
| 22:22.22 | *** join/#asterisk outtolunc (n=me@166.129.246.169) |
| 22:30.28 | *** join/#asterisk hi365 (n=hi365@94.159.178.139) |
| 22:34.05 | PinkFreud | [TK]D-Fender: no, there's one less. it's authors abandoned it within that space of 12 minutes. |
| 22:34.41 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:38.44 | [TK]D-Fender | PinkFreud: Unless all hosting for the software vanished with them its still available. :) |
| 22:38.53 | PinkFreud | grins |
| 22:38.55 | smash- | hello, does anyone know any west coast sip trunk providers |
| 22:39.28 | smash- | or know a place to that has listings of them im having a nightmare of a problem trying to get, a sip turn up request returned... |
| 22:40.46 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 22:41.57 | [TK]D-Fender | ~itsplist-us |
| 22:41.57 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 22:42.27 | joobie | glad pennytel aint in that list |
| 22:42.41 | joobie | tossers took 8 days to re-enable "mutliple calls" on my account |
| 22:43.33 | *** join/#asterisk bbryant (n=bbryant@m4a5a36d0.tmodns.net) |
| 22:57.01 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 23:15.08 | *** join/#asterisk sirgt (n=leonelre@200.49.177.170) |
| 23:18.50 | joobie | guys is it possible to change the polycom 320 ringtone from the web config? |
| 23:19.03 | joobie | or does that have to be done via tftp |
| 23:19.52 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 23:19.57 | shmaltz | hi everyone |
| 23:20.07 | shmaltz | callaccounting postpaid billing anyone? |
| 23:21.58 | *** join/#asterisk infinity1 (n=brendon@web2.artsopolis.com) |
| 23:22.13 | infinity1 | if i can't use ulaw or g729, whats the next best codec? |
| 23:23.23 | shmaltz | infinit1, why not g729? |
| 23:23.31 | infinity1 | voicepulse doesn't support it |
| 23:23.56 | sirgt | hi, anyone knows if the TE122 card works with vmware? |
| 23:24.17 | infinity1 | sirgt: i'm pretty sure its a bad idea to run asterisk in a VM |
| 23:25.55 | sirgt | @infinity1 why is that? |
| 23:26.39 | [TK]D-Fender | joobie: To what? |
| 23:27.19 | joobie | [TK]D-Fender, not too fussed, just another ringtone |
| 23:27.31 | joobie | looks like if i want a specific ringtone i need to use TFTP to upload the file |
| 23:27.32 | [TK]D-Fender | joobie: Any issue doing it right on the phone? |
| 23:27.51 | joobie | if I must, doing it on the handset is OK.. web interface is ideal |
| 23:28.01 | joobie | handset i'll have to step the user through.. but if there's no web interface option that's ok |
| 23:28.06 | [TK]D-Fender | joobie: You know how this works. |
| 23:28.19 | joobie | i dont have TFTP setup at this place to go down that route |
| 23:28.28 | [TK]D-Fender | People configuring Polycom phones via the web interface should be dragged out and shot. Survivors should be shot AGAIN |
| 23:28.33 | VaGoNeTaS | is away: talk to ya in 45 <<eDK/VgN>> [ Logging, Page: On ] |
| 23:28.36 | joobie | lol |
| 23:28.52 | [TK]D-Fender | joobie: Just go on the phone and change the tone |
| 23:30.21 | [TK]D-Fender | be back shortly. Server overhaul time... |
| 23:31.08 | infinity1 | anyone know what the codec requirements are for polycom HD phones? |
| 23:35.46 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 23:35.56 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:38.41 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 23:48.34 | *** join/#asterisk icel (n=dan@c-68-35-75-64.hsd1.nm.comcast.net) |
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| 23:53.47 | *** join/#asterisk Goldfisch (n=gregturn@user-142gbp1.cable.mindspring.com) |
| 23:54.01 | *** part/#asterisk Goldfisch (n=gregturn@user-142gbp1.cable.mindspring.com) |
| 23:55.10 | Globettrotter | hola,, im getting this error when try to dial out ot SIP response 482 "Loop Detected" back from 0.0.0.0 |
| 23:55.50 | dkdkd | hi, i am not getting a response on SIP OPTIONS or REGISTER to voicepulse.com (I have an account there). I believe it is a firewall issue. I can see Asterisk sending out REGISTER and/or OPTIONS periodically on UDP/5060. I would expect a response to come back in on UDP/5060, is that correct? |