00:03.52 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
00:07.52 | telnettech | ok need some help |
00:08.09 | telnettech | I have most of the dialplan done but need help with the system command |
00:08.23 | telnettech | evening jaytee |
00:09.15 | telnettech | I need to create a call file that is then moved to the /var/spool/asterisk/outgoing directory |
00:09.22 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
00:09.39 | telnettech | but i need to create it based on the call to 911 |
00:09.59 | *** join/#asterisk culther0 (i=Admentus@c-75-73-54-42.hsd1.mn.comcast.net) |
00:10.00 | telnettech | anybody help me with the system command for my dialplan |
00:10.08 | baliktad | your calls to 911 are going to depend on a System call to create a .call file?!? |
00:10.10 | culther0 | Howdy everyone. |
00:10.38 | _ShrikE | telnettech: System(echo blah >> /tmp/myfile) |
00:10.52 | telnettech | baliktad: correct.....when someone calls 911, I need to notify another extension of the call |
00:11.01 | _ShrikE | System (mv /tmp/myfile /var/spool/asterisk/outgoing) |
00:11.12 | telnettech | but allow the 911 call to go thru with no interruptions |
00:11.50 | baliktad | I'm not sure I understand, are you going to be doing this .call file moving around before or after your Dial command to 911 |
00:12.06 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
00:12.17 | telnettech | well to complete the call and make it happen, I need to do it before the Dial app |
00:12.29 | baliktad | which is exactly my point |
00:12.42 | telnettech | so I need it to be the priority just before Dial happens |
00:13.06 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
00:13.07 | *** mode/#asterisk [+o russellb_] by ChanServ |
00:13.13 | baliktad | I'm a little confused and a little horrified at what you're trying to do |
00:14.14 | telnettech | i need to notify another extension that a 911 call has been made but the person that is making the 911 needs to not be interrupted....i have most of the dialplan for the call and the notification except the system command part |
00:14.55 | telnettech | so if ext 7103 calls 911, I want to play a recording that tells ext 2100 that 911 was called by 7103 |
00:15.19 | baliktad | do you understand that everything in your dialplan before the Dial() to 911 has the potential to fail and potentially prohibit the 911 call from completing? |
00:15.21 | telnettech | and i want 7103 to continue thru to 911 undisturbed |
00:15.41 | [TK]D-Fender | telnettech: You mean... you've done everything (nothing else needed), except the *1 LINE* of dialplan you need to call before progressing to dialing your 911? :) |
00:16.02 | telnettech | i dont know how to use the system command.....I admit that |
00:16.21 | baliktad | your problem is not with how to use the System command |
00:16.28 | [TK]D-Fender | telnettech: I hand fed you the bloody line :p |
00:16.37 | baliktad | your problem is with prioritizing this notification over the user's actual 911 call |
00:16.48 | [TK]D-Fender | telnettech: and System() it doesn't have parameters! |
00:16.58 | [TK]D-Fender | baliktad: Hardly. |
00:17.10 | culther0 | I've been googling my butt off, but I haven't found any detailed information or maybe a start from 0 point with hardware / software to get going on Asterisk. Currently we've got 4 lines with Comcast; and want to find what equipment( whether comcast is voip to whatever) or maybe kick me in the right way |
00:17.11 | [TK]D-Fender | baliktad: Call System, Dial out. Big friggen deal. |
00:17.15 | [TK]D-Fender | Why is this so hard? |
00:17.22 | culther0 | anyone able to point me in the right direction? |
00:17.46 | telnettech | TK |
00:17.47 | jaytee | what's a phone? |
00:17.53 | culther0 | ^ pretty much |
00:17.54 | [TK]D-Fender | cutlass: Of you have 4 "lines" now, I'll presume regular analog. What do you WANT to do? |
00:18.23 | jaytee | 4 lines with Comcast = proprietary VOIP/ATA |
00:18.23 | [TK]D-Fender | clobbers jaytee with a Bell rotary dial phone. |
00:18.27 | telnettech | TK: remember, I am learning as I go.....I only have this forum and the book to really learn from.....I havent used the system command before |
00:18.33 | culther0 | We have a mix of analog and digital phones, would like an auto attendant to be able to select where to go and the ability to intercom / conference, may buy new phones |
00:18.40 | [TK]D-Fender | telnettech: Go try it |
00:19.00 | [TK]D-Fender | culther0: what is this "digital phone" of which you speak? |
00:19.11 | culther0 | eh, it's at AT&T 1070 |
00:19.12 | telnettech | thats the problem.......I dont know how to write the system command line |
00:19.12 | [TK]D-Fender | culther0: what is it plugged into? |
00:19.27 | culther0 | It's plugged into Cat 5 that's only using 2 wires, so basically standard phone |
00:19.31 | telnettech | that is the only thing i lack before i can test |
00:19.47 | _ShrikE | exten => s,1,System(echo I have no hope) |
00:19.47 | [TK]D-Fender | culther0: this? http://telephones.att.com/telephones_ui/phone_store/dsp_product.cfm?itemID=3609&parent=514 |
00:20.10 | culther0 | that's it |
00:20.24 | [TK]D-Fender | culther0: Seems to be a largely dumb analog phone |
00:20.33 | telnettech | baliktad: If the Dial application is invoked before the call file is written, when the call is hung up, the call file will never be generated |
00:20.48 | culther0 | There's a mishmash of phones, and the owner of the company is open to buying new hardware; but would like some sort of ability to use analog phones for extensions |
00:20.55 | [TK]D-Fender | culther0: that they can be perhaps daisy-chained to each other though |
00:21.00 | culther0 | and yeah it's possible I was tasked with "figuring out what to do" |
00:21.03 | jaytee | [TK]D-Fender, think he'd be good with an 8 port analog card, 4 fxo's for the Comcast interconnect and 4 fxs for the phones? |
00:21.23 | [TK]D-Fender | culther0: Do you like the price you pay for your analog service? |
00:21.38 | culther0 | well it's a contract |
00:21.41 | culther0 | with comcast |
00:21.53 | [TK]D-Fender | culther0: Ok, backed into a corner. Check |
00:21.54 | culther0 | It's like 120 / mo |
00:22.01 | [TK]D-Fender | culther0: Sounds crappy |
00:22.03 | culther0 | D-Fender: Yeah |
00:22.17 | [TK]D-Fender | culther0: Ok, life sucks, but you got in behind instead... |
00:22.22 | [TK]D-Fender | culther0: So moving on... |
00:22.52 | culther0 | D-Fender: The situation is grim and the owner acknowledges it, he wants to keep his phone numbers otherwise he'd be using skype ^_^ |
00:23.18 | [TK]D-Fender | culther0: Yup, you'll want a 4-port FXS card to take in your lines. for that : http://www.telephonydepot.com/Catalog/Sangoma-B-Series/B600D-Analog-Voice-Card |
00:23.38 | [TK]D-Fender | culther0: Now how many PHONES do you want to support? |
00:23.50 | culther0 | no more then 20, currently only like 6 or 7 |
00:25.09 | [TK]D-Fender | culther0: For your phones : http://www.voipsupply.com/linksys-spa8000-g1 |
00:25.19 | [TK]D-Fender | culther0: that + a PC to run * on. |
00:25.24 | [TK]D-Fender | culther0: DONE |
00:25.29 | culther0 | interesting |
00:25.57 | [TK]D-Fender | culther0: cost > $750 + PC |
00:26.00 | culther0 | D-Fender: we do have a punch down block, is that even really worth it? |
00:26.13 | [TK]D-Fender | culther0: Use if you have, but you don't require |
00:26.50 | [TK]D-Fender | culther0: the SPA unit supports RJ21 & RJ11 |
00:26.59 | jaytee | can I ask a stupid question about this? |
00:27.03 | culther0 | ^ yeah |
00:27.11 | jaytee | [TK]D-Fender? |
00:27.12 | [TK]D-Fender | jaytee: Would saying "no" stop you? ;) |
00:28.14 | [TK]D-Fender | jaytee: You know smart questions are always welcome :p |
00:28.15 | *** join/#asterisk Wired_Life (n=Chatzill@mgdb-4db8730b.pool.einsundeins.de) |
00:28.33 | jaytee | you said he needed 4 port FXS card? to take the Comcast lines? if he's using them as analog trunk lines then shouldn't it be FXO ports in the * box? |
00:28.50 | [TK]D-Fender | jaytee: Typo silly! the product is still right! |
00:29.00 | [TK]D-Fender | jaytee: tHAT SHOULDN'T EVEN BE A QUESTION! |
00:29.03 | [TK]D-Fender | Darn caps |
00:29.17 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
00:29.18 | jaytee | so a 4 port FXO card then, ok. |
00:29.19 | [TK]D-Fender | wishes his keyboards all had the key itself light up |
00:29.29 | [TK]D-Fender | jaytee: "duh" :) |
00:29.42 | jaytee | long day, huh? |
00:29.43 | [TK]D-Fender | culther0: So yes, 4-port FXO card. |
00:29.53 | Wired_Life | hello how i make it to dial and if the person click ok jump to next in dial plan? |
00:29.57 | [TK]D-Fender | is pretty sure he doesn't even truly know the terms anyway... |
00:30.12 | [TK]D-Fender | Wired_Life: where on earth do you click "ok"? |
00:30.27 | [TK]D-Fender | Wired_Life: "it" dial? what is "it"? |
00:30.35 | [TK]D-Fender | Wired_Life: Cold you be perhaps a little more vague? |
00:30.47 | [TK]D-Fender | could* |
00:31.03 | culther0 | D-Fender: Rawr, I'm at home doing some off-time research to impress the boss, the cable that comes out of the internet telephony box is a black brick looking cable that plugs into the punch down block. |
00:31.39 | culther0 | from there, the punchdown block has a mishmash of wire and electrical tape and labodomized cat 5 |
00:31.48 | Wired_Life | i have this rule: exten = r,1,Dial(SIP/oli,20) and i search a option to call next thing in dial plan if oli answer the call |
00:31.52 | [TK]D-Fender | culther0: Yeah, they are feeing you analog challes delivered over your internet connection. |
00:32.19 | [TK]D-Fender | Wired_Life: if they answer, then you are TALKING with them |
00:32.28 | [TK]D-Fender | Wired_Life: Typically that is the end of things. |
00:32.31 | jaytee | it would be awesome if Comcast would smell the coffee and sell cheap SIP trunks |
00:32.37 | [TK]D-Fender | Wired_Life: what is there to do "next"? |
00:32.52 | jaytee | but then I ain't waiting for my hair to grow back either |
00:33.13 | telnettech | here comes the dumb question |
00:33.29 | [TK]D-Fender | jaytee: They are supporting CPE while getting to deliver a single circuit. All your savings are belong to them! |
00:33.42 | [TK]D-Fender | jaytee: ch-ch-ch-chia! |
00:34.02 | culther0 | D-Fender: Makes sense; would it be silly to call comcast and ask them if they support this kind of thing or are they just going to tell me to spend 3g's on a proprietary PBX; but since this is all analogue will Asterisk route information such as caller ID and whatnot? |
00:34.08 | Wired_Life | i will make a rule if oli answers (its like a: you are ok) then answer and playback soundfile |
00:34.09 | jaytee | the bit with Joe Pesci about the "drive thru" in Lethal Weapon 2 comes to mind |
00:34.40 | [TK]D-Fender | culther0: Your connection to them is dumb analog. The fact that it walks in the door under IP is transparent to you. No, you do not need their crap. |
00:35.05 | Wired_Life | my problem is if oli answers he has the person on the phone |
00:35.12 | [TK]D-Fender | Wired_Life: what "rule"? Where the hell is this call coming from? |
00:35.44 | [TK]D-Fender | Wired_Life: and if you want him to hear an audio message before the call is bridged, thats what the A() option is for |
00:36.38 | Wired_Life | if a call comes from anonymous i want to call oli and if he say is ok then answer the line and playback sound |
00:37.17 | [TK]D-Fender | Wired_Life: then you want the "M()" option instead and you'll need to make a macro to prompt for the "OK" and return the appropriate MACRO_RESULT |
00:37.43 | [TK]D-Fender | Wired_Life: "core show application dial" <- go read the instructions and come back when you've failed a pile of tests and changes |
00:38.30 | [TK]D-Fender | culther0: You aalready ahve all the analog phones you figure you'll need for this? |
00:38.43 | [TK]D-Fender | culther0: And jsut want to recycle those into your new setup, correct? |
00:42.17 | culther0 | So: Comcast Phone Lines go into FXO card on PC, (4 ports 4 lines?), Line from computer goes to Linksys analog "gateway", gateway has lines to each one of the phone areas.. amirite? |
00:42.24 | culther0 | Yeah correct |
00:42.29 | culther0 | sorry I was reading a site |
00:42.49 | [TK]D-Fender | culther0: Yes, you seem to follow |
00:43.03 | [TK]D-Fender | the Linksys device connects to your LAN and talks SIP to * |
00:43.07 | [TK]D-Fender | ~SIP |
00:43.08 | infobot | sip is probably http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
00:43.43 | culther0 | I see, and I would have the option to add additional phones via ethernet if the boss decides to buy more things, each line would have an extension, etc? |
00:44.18 | [TK]D-Fender | culther0: You can expand in any way you choose. I picked this SPA unit because of your immediate intention to use existing analog phone |
00:44.33 | [TK]D-Fender | sulotherwise I highly recommend getting SIP hard phones. |
00:44.34 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
00:44.34 | *** mode/#asterisk [+o russellb_] by ChanServ |
00:44.36 | culther0 | Sweet, Alright this sounds like a workable solution |
00:44.55 | [TK]D-Fender | culther0: And it is highly cost-effective |
00:44.57 | culther0 | Asterisk isn't too crazy to get a basic configuration setup is there? |
00:45.00 | culther0 | *nod* |
00:45.20 | baliktad | ok TK, one of my customers has a request that I don't really know how to solve |
00:45.23 | [TK]D-Fender | culther0: grab the book, and read the guide for some "inspiration" on how simple a setup could be : |
00:45.25 | [TK]D-Fender | ~book |
00:45.25 | 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 |
00:45.27 | [TK]D-Fender | ~jerjerguide |
00:45.28 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
00:45.44 | [TK]D-Fender | culther0: There is indeed a learning curve to *, but it pays off. |
00:46.13 | baliktad | customer has one site with 6x Linksys SPA-942's, 1 Asterisk box, and 2 DID's |
00:46.21 | telnettech | culther: I agree with the payoff....i am starting to see it but still need some help |
00:46.58 | culther0 | Alright; D-Fender your help is mucho appreciated |
00:47.20 | culther0 | Yeah I will take a look at other pre-fabbed packages or the possibility of finding a local vendor who knows how to configure |
00:47.21 | baliktad | right now the incoming DID's just Dial() everyone (which is fine), but they want to be able to put calls on hold at one station and pick them up at another (ie, call parking without the parking) |
00:47.50 | [TK]D-Fender | baliktad: How do you "have" a DID? :) |
00:48.11 | [TK]D-Fender | telnettech: Ask your consultant if training may be right for you! |
00:48.16 | culther0 | I'd kinda at some point like to come up with a encompasing tutorial cause this stuff isn't readily apparent from what people google |
00:48.17 | [TK]D-Fender | </psa> |
00:48.26 | baliktad | 2 phone numbers routed to their * box provide incoming voice services |
00:48.34 | telnettech | we dont have a consultant |
00:48.57 | [TK]D-Fender | culther0: I haven't seen an encompassing doc that I could jsut trow at someone yet. |
00:49.05 | telnettech | and i have gone to training.....i think i have learned quite abit over the last 6 months starting from scratch |
00:49.17 | telnettech | * toots his own horn |
00:49.28 | [TK]D-Fender | telnettech: get a better trainer young Jedi! |
00:49.41 | telnettech | jsmith was my trainer....lol |
00:49.41 | [TK]D-Fender | telnettech: Not in here, this is a family show! |
00:49.49 | jaytee | oh, god! not that tune again. Brian, you need to work on your repetoire |
00:50.06 | baliktad | problem now is that each phone is independent, if one person answers, it's not easy for someone else to join or finish the call from another phone |
00:50.12 | [TK]D-Fender | telnettech: module unload chan_brokenrecord.so ! |
00:50.28 | jaytee | I can't picture Jared without a Mango Smoothie in his hand |
00:50.41 | telnettech | i am trying guys.... |
00:51.07 | [TK]D-Fender | telnettech: thats what makes this mildly entertaining. Now dance monkey, DANCE!' |
00:51.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
00:51.23 | [TK]D-Fender | draws his 6-shooter |
00:51.27 | baliktad | boss basically wants his SIP phones to function as if they are all sharing the same 2 lines |
00:51.41 | mmlj4 | excuse me while I whip this out |
00:51.57 | [TK]D-Fender | mmlj4: EWWWWWWWWWW!!!!!!!!!! |
00:52.09 | russellb_ | baliktad: Asterisk SLA support will do that |
00:52.17 | [TK]D-Fender | .... |
00:52.25 | mmlj4 | I was expecting laugher not ewwness |
00:52.30 | [TK]D-Fender | russellb_: * SLA makes baby Jesus cry... |
00:52.37 | jaytee | goes off to investigate if the lyrics to "You've gotta be cruel to be kind" were written by someone who's first name starts with A :-) |
00:53.06 | baliktad | I've been looking at SLA, although I'm not sure if it will |
00:53.18 | mmlj4 | [TK]D-Fender: surely you know what movie that's from? |
00:53.24 | jaytee | when it comes to either herpes or SLA, most people choose herpes! |
00:53.26 | [TK]D-Fender | baliktad: It can if you have the speed-dials to support it, but its... FUGLY |
00:53.30 | mmlj4 | sla? |
00:53.50 | russellb_ | sighs |
00:53.53 | [TK]D-Fender | mmlj4: its like "blah", but compiled into an * module ;) |
00:54.03 | mmlj4 | ok |
00:54.04 | baliktad | if I use SLA, one extension will ring and the others just "show" ringing on the BLF, without showing the incoming caller ID |
00:54.04 | [TK]D-Fender | ~sla |
00:54.05 | infobot | sla is, like, service level agreement, or shared line appearances |
00:54.05 | mmlj4 | blah? |
00:54.06 | telnettech | i will say that you have taught me alot TK.....and i do look for the answers BEFORE i come into the chat.....there just isnt anything made for dummies when it comes to the system command |
00:54.33 | mmlj4 | ok, I misparsed... of course I know what an SLA is |
00:54.42 | jaytee | shoots consoling, sympathetic looks in russellb's direction |
00:54.44 | russellb_ | [TK]D-Fender: Your attitude got old with me many months ago |
00:54.56 | [TK]D-Fender | mmlj4: You know "bleh", the sound projectile makes if you don't build up enough pressure first ;) |
00:55.23 | baliktad | russellb_ are you the russell that wrote the SLA pdf I've been poring over for the last month? |
00:55.31 | mmlj4 | but how did key-functionality get morphed into an SLA? |
00:55.38 | [TK]D-Fender | russellb_: Sorry if I rag on this one feature, I know you were steered towards this, I don't want to lay blame or anything |
00:55.56 | [TK]D-Fender | *sigh* |
00:56.10 | telnettech | and i dont have much experience with linux, networking and computers....i am straight from the old world of telephony |
00:56.12 | jaytee | he quit or at least his underscore nick did and the other is grayed out marking away status |
00:56.15 | [TK]D-Fender | mmlj4: synonymous. |
00:56.44 | jaytee | telnettech, do you know how to get to Carnegie Hall? |
00:57.01 | telnettech | take a left at albuquerque |
00:57.05 | telnettech | silly rabbit |
00:57.11 | [TK]D-Fender | jaytee: I'm sure http://maps.google.com will know! |
00:57.12 | jaytee | practice my boy, practice! |
00:57.46 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
00:58.01 | jaytee | and if the answers don't come fast enough then read until your eyes bleed |
00:58.08 | [TK]D-Fender | telnettech: You need some Specialized High Intensity Training! I dole out about as much of it at my office as I can muster! |
00:58.22 | jaytee | SHIT |
00:58.25 | telnettech | are you hiring.....:) |
00:58.38 | knarfly | does asterisk-1.6.09 need zaptel to run with an X101P analog card? |
00:59.09 | [TK]D-Fender | knarfly: No, * 1.6 does not support Zaptel |
00:59.11 | jaytee | nope, because 1.6 won't run zaptel |
00:59.28 | *** join/#asterisk JuStIcIa_ (i=john@cbl-sd-74-96.aster.com.do) |
00:59.49 | knarfly | will my x101P work with *-1.6.0.9 |
01:00.01 | [TK]D-Fender | ~dahdi |
01:00.02 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
01:00.03 | [TK]D-Fender | ^^^^^^^^^^ |
01:00.34 | knarfly | and dahdi installs with the asterisk taeball? |
01:00.45 | knarfly | tarball |
01:00.47 | [TK]D-Fender | knarfly: No, it is a separate package |
01:00.49 | jaytee | nope, dahdi installs with dahdi |
01:01.09 | culther0 | ah ffs now I can't seem to save this chat on IRC |
01:01.10 | culther0 | >_< |
01:01.18 | jaytee | get the tarball that is both the drivers and the tools |
01:01.22 | [TK]D-Fender | culther0: copy / paste |
01:01.32 | knarfly | where do I dahdi |
01:01.33 | jaytee | Xchat logging FTW! |
01:01.59 | culther0 | it won't copy paste! |
01:02.00 | culther0 | >_< |
01:02.47 | knarfly | where do I get the dahdi tarball |
01:02.47 | jaytee | knarfly, http://downloads.digium.com/pub/telephony/dahdi-linux-complete/ |
01:02.54 | culther0 | *stabs own eyes out* |
01:02.57 | [TK]D-Fender | knarfly: www.asterisk.org |
01:03.06 | jaytee | or there |
01:03.13 | telnettech | TK: See they moved all the guys that knows asterisk for my company from US to Singapore.....and the rest of us were basically told to sink or swim.....i am doggie paddling but barely |
01:03.43 | jaytee | doggie paddling all the way to Singapore? wow, that's a rough swim! |
01:04.16 | telnettech | TK: and I am the best of the rest so far....the others dont even try cause there is no one to enforce it cause even the supervisor is scared of Asterisk |
01:06.35 | [TK]D-Fender | telnettech: Then they should welcome your becoming better trained by someone who can show you how to seriously get things done. |
01:07.21 | knarfly | when installing do i need to install dahdi first? |
01:07.27 | jaytee | yes |
01:07.30 | [TK]D-Fender | telnettech: There is "general" and "theory" training, and there is "here's what you need to know, here are the real gotchas & opportunities, and here's some tricks to get the most out of the basics" |
01:07.37 | telnettech | right now the voice part is in limbo with the merger we are going thru......so it is not being looked t....they are eventually going to require us to do hi speed internet and VOD thru IPTV |
01:07.43 | jaytee | knarfly, what distro are you installing it on ? |
01:09.03 | telnettech | i mean i thought i should go to the beginners from digium and was told by supervisor that i need to go to advance....i didnt do too bad but it just opened up more learning that is needed....i agree |
01:09.22 | telnettech | i didnt pass my DCAP but know why |
01:10.00 | jaytee | me neither, passed the written but bombed on the lab cuz I ran out of time. If I'd passed on installing the T1 card I probably would have finished and passed. |
01:10.04 | telnettech | i dont mind book smart but i want real world experience too |
01:10.11 | knarfly | jaytee: fedora 10 |
01:10.20 | telnettech | i failed both parts |
01:10.34 | jaytee | ugh, wrong platform for *, better to use CentOS |
01:10.57 | [TK]D-Fender | telnettech: Book smart would require more "real" books :) |
01:11.13 | knarfly | CentOS sux the big red one |
01:11.27 | telnettech | I have read most of ATOF and i am also reading at same time SIP Demystified |
01:11.37 | [TK]D-Fender | telnettech: TFOT is a kinda loose grab at newbs, and doesn't show either deep detail, or too much practical minimal use |
01:11.58 | [TK]D-Fender | telnettech: just a bit too "spares", but it covers the wider range so IRC, etc can fill in the gaps. |
01:11.59 | telnettech | i also try to get onto voip-info to read as well |
01:12.19 | [TK]D-Fender | telnettech: the WIKI is a seriously mixed bag |
01:12.34 | [TK]D-Fender | knarfly: I use it all the time. jsut fine |
01:12.48 | telnettech | so thats why i come on here and ask questions |
01:13.14 | telnettech | at least I am not as bad a Wired_Life |
01:13.43 | telnettech | i don know how to read the dialplan for the most part :) |
01:13.56 | [TK]D-Fender | telnettech: comparative self-deprecation. SUCKcess! |
01:14.01 | *** part/#asterisk culther0 (i=Admentus@c-75-73-54-42.hsd1.mn.comcast.net) |
01:14.22 | [TK]D-Fender | telnettech: 11 steps to go! |
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01:15.00 | telnettech | even though i know that I am going to get criticized for my question by you, I still ask them so that i can get better |
01:15.11 | jaytee | knarfly, the only difference between Fedora and CentOS is that Fedora is more bleeding edge. they both draw the same code base from Red Hat. |
01:15.22 | [TK]D-Fender | telnettech: Think-skinned, now 10 to go! |
01:15.48 | jaytee | and it's the bleeding edge that will give you major headaches making all the bells and whistles in * work properly. Especially the whistles! |
01:15.57 | telnettech | I just dont have a test system to just play with all the time as I am required to do things on live customer systems |
01:16.04 | [TK]D-Fender | goes to hone his katana... |
01:16.18 | jaytee | know how to tell if it's sharp? |
01:16.20 | culther0 | Lets see if this is logging now |
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01:16.33 | telnettech | so i am trying to be cautious and not make more trouble than I can handle |
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01:19.25 | telnettech | so if all my learning sources are flawed, per your opinion, then how else can i gain knowledge except real world |
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01:23.02 | jaytee | I never said your learning sources are flawed |
01:23.37 | telnettech | im talking about TK |
01:23.57 | jaytee | well, that's a whole nuther subject :-) |
01:24.20 | [TK]D-Fender | telnettech: I didn't say "flawed", they just haven't been "right for you". |
01:24.21 | knarfly | asterisk-1.6.0.9 installed and running...will setup X101P card tomorrow....good night Mrs. Calabash, wherever you are! |
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01:24.42 | jaytee | friggen Fedora, ugh |
01:25.03 | [TK]D-Fender | telnettech: You seem the type to best profit from directed training as opposed to lose forms like TFOT. |
01:25.28 | telnettech | i learn better in a lab as i am instructed, you are correct |
01:25.45 | [TK]D-Fender | Dear God.... "Mrs. Calabash"... like that doesn't (carbon) date you..... |
01:26.02 | telnettech | i like the hands on, I dont read too well and take what i read and put it in practice....never had |
01:26.08 | jaytee | brian, do you have an older PC you can setup at home with a couple cheap X100 cards? |
01:26.32 | [TK]D-Fender | waits for knarfly's return so he can toss him into a gas chromatograph for testing... |
01:26.38 | jaytee | I can mail ya a couple of X100 FXO cards to use on analog lines |
01:26.52 | telnettech | i am trying to get one setup yes....i just dont have the time.....it seems like i am always working on a customer....i am even working now as we speak |
01:27.07 | jaytee | and you've got a SIP phone from class. |
01:27.13 | telnettech | yep |
01:27.27 | jaytee | time? who ever has time. we don't "have" time, we "make" time. |
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01:27.46 | [TK]D-Fender | jaytee: Cards are the real issue, dialplan is. Frankly everyon who thought a card will help them learn * quickly finds out that devies are a dozen lines of setup and dialplan is EVERYTHING |
01:27.56 | telnettech | and i dont like to have a problem that i cant figure out |
01:28.13 | jaytee | and since according to relativity, space and time are intertwined then maybe you just need to find the space first and the time will be there already interwoven. |
01:28.17 | [TK]D-Fender | darn, typing skills degrading... |
01:28.45 | telnettech | I have a system already jaytee....i have red hat loaded on it but i havent loaded the asterisk and all yet |
01:28.50 | [TK]D-Fender | jaytee: Einstein flunked the "practical of relativity" ;) |
01:28.52 | jaytee | dialplan is pretty much at least 98.7 percent of Asterisk. |
01:29.31 | jaytee | Einstein divided by zero in the proofs for one of his early papers. Plus he got a D in Algebra in High School |
01:30.32 | telnettech | and i think i do well with basic stuff in the dialplan....i just have problem with intermediate and advance programming in it....like chan variables, critical thinking Goto statements, and etc |
01:30.59 | telnettech | if it gets to complex it gets confusing to me right now |
01:31.25 | jaytee | that's a skill that takes time to develop and is more programming logic than what most telecom techs get exposed to |
01:31.45 | telnettech | like this system command.....i have sat here tonight and taken what TK told me earlier and struggled thru most of the setup....except the most important part |
01:34.44 | [TK]D-Fender | telnettech: System(/usr/sbin/asterisk -rx "originate SIP/operatortypeperson Local/${CALLERID(num)}@omfg911") |
01:35.54 | telnettech | thank you TK....can you explain what it does a little...so that i can decipher for future reference |
01:36.10 | [TK]D-Fender | telnettech: [omfg911] exten => _X.,1, Playback(thefollowingtwidialed911) exten => _X.,2, SayDigits(${EXTEN}) |
01:36.51 | telnettech | i just needed the system part....i have the dialplan setup |
01:36.53 | [TK]D-Fender | telnettech: makes a call to * to have it trigger an out-call to your operator and when they answer they get this message played to them |
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01:37.20 | [TK]D-Fender | telnettech: actually, drop the "Local/" off. I reversed it |
01:37.57 | DavidBer | Good evening - I was wondering if someone could answer some questions regarding a 1.4.24.1 setup with extensions being busy on inbound calls. |
01:38.12 | Qwell | are they busy? |
01:38.18 | DavidBer | hehe - nah :) |
01:38.37 | DavidBer | calls come in - and asterisk is saying that the extension is busy |
01:39.04 | Qwell | pastebin a log |
01:39.05 | telnettech | thanks TK |
01:39.15 | DavidBer | I have a few dids from Vitelity. One goes to an IVR and one goes to an extension - the extension goes busy and I get a message from Vitelity. The IVR goes to the IVR and then when to the extension goes busy |
01:39.20 | telnettech | i will study it so that i know what it does |
01:39.45 | [TK]D-Fender | telnettech: "help originate" |
01:39.55 | Qwell | your extension likely isn't registered/setup properly |
01:40.08 | DavidBer | sip peer shows them online |
01:40.15 | DavidBer | stupid question - how do I do a pastebin? |
01:40.19 | Qwell | ~pb |
01:40.20 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
01:40.23 | telnettech | oh.....i have been looking at system all this time |
01:40.45 | [TK]D-Fender | telnettech: As I said, System doesn't have parameters. |
01:41.01 | [TK]D-Fender | telnettech: you jsut call * to fire off an independant call so you can resume dialing out. |
01:41.11 | telnettech | correct |
01:41.12 | [TK]D-Fender | telnettech: this spawned call is not attached to the other in any way |
01:41.18 | DavidBer | Qwell - what do you want pasted? |
01:41.46 | telnettech | is there a way to log this call so that they cant say that it was never made? like some type of CDR? |
01:42.37 | telnettech | just asking |
01:45.54 | k-man | i have a linksys spa2102, if i don't want to use the router component (as i already have a router) do i plug the ethernet port or wan port into my network? |
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01:46.52 | otomoto | how can we log to a file certina from a context using "exten => ....." |
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01:52.47 | DavidBer | Qwell - here ya go - http://pastebin.com/d68ddabd9 |
01:54.15 | [TK]D-Fender | telnettech: exten => _X.,3,System(echo "dumbfuck ${EXTEN} called 911 on ${TIMESTAMP}") |
01:54.43 | [TK]D-Fender | telnettech: If CDR does not make you feel warm & fuzzy as it is |
01:54.55 | telnettech | ok thanks |
01:59.44 | telnettech | TK D: does the originate need a sound card in the server? |
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02:06.15 | [TK]D-Fender | GRRR |
02:06.31 | [TK]D-Fender | fsck-ing Flash crash |
02:07.32 | [TK]D-Fender | kicks Adobe/M$ in the nads |
02:09.50 | jaytee | they are nadless nerds, it's an exercise in futility. might as well tilt at windmills, Quixote |
02:10.27 | telnettech | good night huys |
02:10.30 | telnettech | guys |
02:10.38 | jaytee | nite brian |
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02:14.39 | [TK]D-Fender | grabs his lance and mounts his steed. |
02:15.13 | [TK]D-Fender | thinks that sounds entirely too dangerously ambiguous in these parts for his own good |
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02:18.00 | jaytee | hehehe |
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02:33.32 | Octothorpe | Hey, somehow AsteriskNOW 1.5.0 seems to be able to run Apache as apache:apache instead of asterisk:asterisk even though /var/www/html is owned by asterisk:asterisk. What's the score? Didn't they have to be the same owner with FreePBX? |
02:33.49 | Octothorpe | Ah, #asterisknow exists, I'm taking it there. |
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03:02.14 | Wired_Life1 | i hate macros |
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03:10.51 | [TK]D-Fender | Wired_Life1: That's OK, they have no feeling for you to hurt :) |
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03:14.49 | Wired_Life1 | i have a macro test and when im click answer call the call get answers but i dont want a answer call |
03:15.36 | [TK]D-Fender | Wired_Life1: "click"? What "click"? You are talking crazy-talk.... |
03:16.14 | jaytee | gibbledy bibbledy |
03:16.21 | [TK]D-Fender | Wired_Life1: And any YEAR now you might learn to start pastebin-ing your backup when you have problems.... unless of course you jsut want to complain about them and not try to solve them |
03:16.25 | jaytee | argle bargle whoosh! |
03:16.41 | [TK]D-Fender | jaytee: Bah weep grah nah weep nini bong? |
03:17.05 | jaytee | y'know, it's pretty damn scary when I understand that! |
03:17.20 | [TK]D-Fender | jaytee: :D |
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03:18.19 | [TK]D-Fender | knows too much and the ninjas have probably already been dispatched to dispose of him |
03:18.24 | Wired_Life1 | http://pastebin.com/d72d2cc0b |
03:19.33 | [TK]D-Fender | Wired_Life1: Where did you get this variable "ISOK" from? |
03:20.21 | [TK]D-Fender | Wired_Life1: Wired_Life1 and why do you think it means anything? |
03:20.38 | jaytee | "ISOK" "AIN'TOK" |
03:21.04 | [TK]D-Fender | Wired_Life1: "core show application dial" <- read the instructions. |
03:21.13 | drmessano | Did you set IS_NOT_MADE_UP=no in unicorns.conf? |
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03:21.41 | Wired_Life1 | i have read the fucking instructions |
03:21.50 | Wired_Life1 | its all english |
03:21.58 | Wired_Life1 | and very short i want more |
03:22.04 | [TK]D-Fender | I set MILLIONDOLLARSINMYBANKACCOUNTNOW=true , so WHERE'S MY FUCKING MONEY!??!?!? |
03:22.08 | Wired_Life1 | its not helpful to me |
03:23.00 | [TK]D-Fender | Wired_Life1: The instructions tell you what variable has to be set and what its values mean. You INVENTED one out of thin air |
03:23.35 | [TK]D-Fender | Wired_Life1: Now go follow the instruction. You aren't prompting your callee for any input and you aren't determing your result based on it. |
03:25.45 | Wired_Life1 | all with the macro and so on i dont want |
03:26.01 | Wired_Life1 | i dont want to answer a call before check |
03:26.18 | Wired_Life1 | i must check if a person is on phone |
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03:27.35 | [TK]D-Fender | Wired_Life1: What you asked for is to check if they WANT to take the call |
03:28.11 | Wired_Life1 | im so silly |
03:28.21 | Wired_Life1 | english is not my language |
03:29.07 | *** join/#asterisk termas_ (i=paine@creep.bur.st) |
03:30.36 | termas_ | Would anyone like to help me with [what seems to be] an obscure Realtime problem? |
03:31.02 | [TK]D-Fender | termas_: Just provider your details and see who bites |
03:31.46 | termas_ | loki2301: I have a psql DB, extconf.conf points to it for sip users. |
03:31.53 | Wired_Life1 | its now 5:31 in germany and i cant sleep without solving this problem |
03:32.15 | termas_ | When a phone comes online, it registers with the details from the DB fine. |
03:32.34 | termas_ | and can recieve calls, place calls, anything. It all works fine. |
03:32.43 | [TK]D-Fender | Wired_Life1: And you can't solve much of anything with no sleep. visciouscircle=true |
03:32.46 | termas_ | and then, at some point, it stops, and nothing will work until the phone is rebooted. |
03:33.21 | [TK]D-Fender | termas_: tahts really vague. clarify this "nothing will work" bit... a LOT |
03:33.26 | termas_ | "everyone is congested at this time" |
03:33.43 | termas_ | One second, I'll get the exact error. |
03:34.03 | termas_ | [Apr 14 11:32:03] WARNING[9006]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
03:34.04 | termas_ | <PROTECTED> |
03:34.08 | Wired_Life1 | i need more sites with examples and so on and i think the rule i want is already online but i dont know the words to search for @ google |
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03:35.18 | termas_ | I don't know what else to provide, is the DB schema useful? |
03:38.47 | Wired_Life1 | i want it so: if a anonymous call comes i want to check if a person is available behind the phone and then i will answer call and play some playbacks and record voice sample for the person because the caller is anonymous |
03:39.13 | Wired_Life1 | you know what i mean? |
03:39.25 | s14ck | hello |
03:39.28 | Wired_Life1 | but i dont know the words to search for |
03:39.40 | Wired_Life1 | im sure there is already a rule @ google |
03:39.48 | termas_ | Wired_Life1: do you want to test for the status of the end point? |
03:40.14 | termas_ | and then, play to the end point, a sound, when the start point is anon? |
03:42.51 | Wired_Life1 | no |
03:43.53 | termas_ | How do your people log into the phone? |
03:44.04 | termas_ | Agents? |
03:44.27 | Wired_Life1 | if a anonymous call comes in i want to check if a person is available behind the phone (make a call to check or so) and then i will answer the call from anonymous and play some playbacks and record voice sample for the person @ end point because the caller sends no number |
03:45.24 | Wired_Life1 | now i search only the rule to check if end point is there |
03:45.36 | termas_ | So to the caller it would be like "Hi, you have no caller id, say your name." then when it reached the end point, the recording of the name is played to them? |
03:46.00 | [TK]D-Fender | termas_: SIP peer dumps, DB dumps, firewall checks, etc. and you are getting CLI output, there's something there as well |
03:46.17 | jaytee | nite all |
03:46.20 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
03:46.45 | Wired_Life1 | termas_ yes the voice recording stuff i have already |
03:47.12 | s14ck | how can I get one id call in progress and make relationship with id users at my db? |
03:48.13 | termas_ | [TK]D-Fender I can rule out firewall I think, since it works when the phone is freshly rebooted. |
03:48.14 | Wired_Life1 | i search only for the part to check if a person is available behind the phone |
03:48.18 | [TK]D-Fender | terI would think that perhaps you failed a SIP qualify or something and the peer's remote NAT (if applicable) timed out and closed the port behind it, etc. |
03:48.25 | [TK]D-Fender | termas_: I would think that perhaps you failed a SIP qualify or something and the peer's remote NAT (if applicable) timed out and closed the port behind it, etc. |
03:48.35 | termas_ | Wired_Life1, what's wrong with Dial() ? If they are there, they will pick it up? |
03:49.31 | termas_ | [TK]D-Fender There is no NAT, they are on the same subnet. What do you mean by SIP qualify? |
03:50.23 | [TK]D-Fender | termas_: a keep-alive signal. if the host fails to respond * will flag them as uncontactable until they check in again. This of course happens when you restart the phone which is why I suspect it may be the case |
03:51.18 | termas_ | [TK]D-Fender The [DEBUG] stuff in the CLI says things like |
03:51.57 | termas_ | [Apr 14 11:39:52] DEBUG[28140]: res_config_pgsql.c:170 realtime_pgsql: 1Postgresql RealTime: Result=0xb74ffc78 Query: SELECT * FROM sip_friends WHERE name = 'ross.paine' AND host = 'dynamic' |
03:51.57 | termas_ | [Apr 14 11:39:52] DEBUG[28140]: res_config_pgsql.c:178 realtime_pgsql: Postgresql RealTime: Found 1 rows. |
03:51.58 | termas_ | [Apr 14 11:39:52] DEBUG[28140]: res_config_pgsql.c:790 pgsql_reconnect: Postgresql RealTime: Everything is fine. |
03:52.41 | [TK]D-Fender | termas_: enable SIP DEBUG and watch what's actually happening. when a call fails, dump the DB, check your peer at CLI, etc |
03:52.50 | Wired_Life1 | http://pastebin.com/dc114b59 |
03:53.06 | termas_ | ok |
03:54.50 | Wired_Life1 | my problem is i dont want to forward the anonymous call to oli before he said ok |
03:55.24 | Wired_Life1 | but only when oli answer it go next |
03:55.30 | [TK]D-Fender | Wired_Life1: exten => s,n,GotoIf($[${ACCEPT} = 1 ] ?yes:no) <-- extra spaces = BAD |
03:56.32 | Wired_Life1 | i know the rules are buggy they are from http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
03:56.40 | [TK]D-Fender | Wired_Life1: that macro should prompt him asking if he wants to accept it or not. |
03:57.21 | termas_ | http://pastebin.com/d6cf10fa7 |
03:58.23 | Wired_Life1 | i know i know im programming visual basic, php and so on but i dont know this fucking dial plans |
03:58.55 | termas_ | extensions.conf is ugly, and hard :( |
03:58.59 | [TK]D-Fender | Wired_Life1: Go read the book and play around. its just 1 app after another.... |
03:59.10 | [TK]D-Fender | termas_: Ugly yes, hard, no. |
03:59.41 | [TK]D-Fender | termas_: I've writtem more complex languages. Not written IN, WRITTEN. |
04:01.12 | Wired_Life1 | [TK]D-Fender look @ http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial and you know my problem... this rule is answering directly a call... i want to check if anyone is there to make no costs for the caller |
04:01.40 | [TK]D-Fender | Wired_Life1: the CALLER is not #^&%$ing answered, the CALLEE IS. |
04:02.29 | [TK]D-Fender | Wired_Life1: the caller is not answered until it comes back "accepted" by your macro |
04:02.43 | termas_ | It's not 'hard' as in it's difficult, it's hard compared to the alternatives. |
04:03.01 | Wired_Life1 | [TK]D-Fender are you 100% sure? |
04:03.11 | termas_ | the amount of work you have to do in extensions.conf is massive, to achieve little. |
04:03.13 | [TK]D-Fender | termas_: its very easy when you realize its every bit as "dumb" and the 1-page instructions for each app really entails. |
04:03.57 | [TK]D-Fender | termas_: 1 silly step after another. Set a variable, do a comparison. Jump somewhere if X=Y, Dial Z, whatever |
04:04.09 | [TK]D-Fender | termas_: this is "hello world" grade stuff |
04:04.27 | [TK]D-Fender | termas_: there is no "concurrency", its all just linear |
04:04.38 | termas_ | Sure.. Then look up an extension in the db, relate that to a user, log that somewhere else, etc etc.. to hard. :p |
04:04.41 | termas_ | toO^ |
04:05.01 | kc8pxy | termas_: compared to? |
04:05.10 | kc8pxy | termas_: something else that's free? |
04:05.34 | termas_ | I'm going to have to admit at this point, I'm using adhearsion for my dial plan logic.. |
04:06.01 | Wired_Life1 | [TK]D-Fender the call is get answered there is a recording and a recording can only make if a caller is there |
04:06.02 | drmessano | heh |
04:06.18 | *** join/#asterisk Xunie (n=grag@5ED1BFF8.cable.ziggo.nl) |
04:06.53 | termas_ | But, just because I use adhearsion, doesn't make me less of a man... |
04:07.02 | drmessano | Yes it does |
04:07.07 | [TK]D-Fender | ter"relate to a user"? No such thing. Call come in. Who is it auth'd as? Go to their context. everything else is dialplan. loggin... bleh.. no impact on what you DO |
04:07.09 | drmessano | Everyone says that |
04:07.14 | drmessano | "Its shrinkage" |
04:07.17 | drmessano | "No really" |
04:07.27 | drmessano | Sorry, you're less of a man |
04:07.32 | termas_ | :( |
04:07.48 | [TK]D-Fender | 00:05]<termas_>I'm going to have to admit at this point, I'm using adhearsion for my dial plan logic.. <- so what you're really trying to say is "No.. I don't get it either, I've jsut found my cop-out already!" :p |
04:07.59 | termas_ | [TK]D-Fender The numbered extensions relate to different users on different days, in my dial plan. |
04:08.27 | [TK]D-Fender | termas_: if you say so... |
04:08.30 | termas_ | [TK]D-Fender: Well, no, I've done extensive work in extensions.conf, |
04:08.57 | kc8pxy | termas_: callcenter? |
04:09.15 | termas_ | Government, |
04:09.22 | termas_ | small government, |
04:10.37 | kc8pxy | termas_: diferent users use the same phone, but different mailboxen? |
04:10.53 | termas_ | In my dial plan? |
04:11.33 | termas_ | No, the users all use the same phone every day. Captain Hammer, for example, always uses the same username/phone. |
04:11.55 | termas_ | But on wednesday, he's the janitor, every other day he fights crime, so he has a different phone number on wednesday. |
04:11.56 | termas_ | etc. |
04:12.28 | kc8pxy | < termas_> [TK]D-Fender The numbered extensions relate to different users on different days, in my dial plan. |
04:12.28 | *** part/#asterisk otomoto (n=aoun@gateway3.aleks.com) |
04:12.35 | kc8pxy | hmmm |
04:13.19 | Wired_Life1 | i go sleep thx [TK]D-Fender and termas_ ill be back |
04:13.40 | kc8pxy | termas_: i thougth i just read how you can do that in the extensions.conf. |
04:13.41 | termas_ | Night, |
04:14.42 | termas_ | kc8pxy You can do that in extensions.conf, you can ODBC, or natively connect to the database, pull the relivant username and dial it. |
04:16.33 | [TK]D-Fender | yup |
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04:18.34 | Wired_Life1 | http://pastebin.com/d67196c2d one and last |
04:19.49 | [TK]D-Fender | Wired_Life1: You keep showing us dialplan and not telling us what probelm if any you are actually HAVING with it |
04:19.57 | [TK]D-Fender | Wired_Life1: Are we supposed to GUESS? |
04:20.34 | [TK]D-Fender | Wired_Life1: here's one : exten => s,n(yes),SetVar(MACRO_RESULT=CONTINUE) <- in 1.4+ SetVar is permanently gone and replaced by "Set". Another good reason to think twice before copy/paste-ing code from the WIKI |
04:20.43 | [TK]D-Fender | Wired_Life1: Its often outdated crap |
04:22.36 | [TK]D-Fender | Wired_Life1: exten = r,1,Dial(SIP/oli,20,M(test)) exten = r,2,GotoIf($["${ISOK}" = "TRUE"]?10:20)<- and this is still crap. You dial him once with a broken macro and expect to call him AGAIN after |
04:23.04 | [TK]D-Fender | Wired_Life1: r,1 vs r,13 |
04:26.02 | Wired_Life1 | exten = s,1,Set(ISOK=TRUE) ; this is only set if oli answers and this making costs for anonymous i dont want |
04:26.33 | [TK]D-Fender | Wired_Life1: it is set and the call IS COMPLETELY answered. Your 2nd dial is WORTHLESS |
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04:26.50 | Wired_Life1 | thats my problem |
04:26.55 | [TK]D-Fender | Wired_Life1: and the call is bridged immediately. "oli" is never asked to confirm to accept the call or not. |
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04:27.28 | [TK]D-Fender | Wired_Life1: the are 2 DIAL's in your "r" exten. WHY? You left broken crap in there and just pasted from the WIKI after. |
04:28.00 | [TK]D-Fender | Wired_Life1: You aren't testing anything with your first dial macro, so you might as well not have even done it |
04:28.27 | Wired_Life1 | i want to check if oli is there before calling he @ r,13 |
04:29.19 | [TK]D-Fender | R.13 is the dial that DOES THE *#&$ING CHECK |
04:29.29 | [TK]D-Fender | Wired_Life1: r,1 checks NOTHING |
04:30.23 | Wired_Life1 | r,13 do the check with answering my question is how make a check without making costs |
04:30.34 | [TK]D-Fender | Wired_Life1: Wired_Life1 and you can't if someone is there BEFORE CALLING. |
04:32.12 | [TK]D-Fender | Wired_Life1: WTF is "sip/oli"? |
04:32.27 | Wired_Life1 | my softphone |
04:32.32 | termas_ | SIP/oli would be the user "oli". |
04:32.58 | [TK]D-Fender | Wired_Life1: If you call it and it doesn't answer, how is your caller "answered"? |
04:33.11 | [TK]D-Fender | Wired_Life1: I don't see it answering if you IGNORE the call. |
04:33.15 | [TK]D-Fender | Wired_Life1: So what is your problem? |
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04:34.18 | [TK]D-Fender | Wired_Life1: Doesn't matter if you fail to answer. Doesn't matter if the softphone is even contactable. niether will "answer" the caller |
04:34.28 | [TK]D-Fender | Wired_Life1: This is avery large waste of time |
04:35.34 | Wired_Life1 | i want a signal @ my softphone if someone is calling anonymous and then i say ok and then the call go answered and play woman voice |
04:35.52 | termas_ | Busy() and Congestion() signal the Caller, |
04:36.01 | termas_ | everything signals the caller, |
04:36.38 | Wired_Life1 | i want to signal the end point oli |
04:37.05 | Wired_Life1 | if he said ok then the call go answered |
04:37.09 | termas_ | You can't signal the end point, |
04:37.47 | Wired_Life1 | i cant signal the end point without making costs? why? |
04:38.07 | [TK]D-Fender | Wired_Life1: you're calling a soft-phone... at what point are you TALKING to them? |
04:38.41 | Wired_Life1 | can i make only with dial? |
04:38.59 | [TK]D-Fender | Wired_Life1: you're calling a soft-phone... at what point are you TALKING to them? <-------- |
04:38.59 | termas_ | You can dial and add headers, |
04:40.39 | Wired_Life1 | plz say only if my idea should work or not |
04:41.11 | [TK]D-Fender | Wired_Life1: you're calling a soft-phone... at what point are you TALKING to them? <-------- |
04:42.15 | Wired_Life1 | only when i make a call or? |
04:42.29 | leifmadsen | yo! |
04:42.40 | termas_ | Here's what I'd do, I have snom hard phones. When a call comes in, i'd take the call, check it for callerid, if it's good, Dial(SIP/oli), if it's anon, have asterisk answer it, n,background(woman-you-are-anon) |
04:43.04 | termas_ | record() then Dial them into a holding extension. |
04:43.06 | [TK]D-Fender | Wired_Life1: you're making a fucking softphone RING. when are these 2 fucking people supposed to actually START TALKING TOGETHER? |
04:43.38 | leifmadsen | [TK]D-Fender: never! |
04:43.38 | termas_ | the snom phone would be configured with a BLF, so a little light comes on, when oli pushes it, he gets the recording, and if it's ok, he can push 1 to bridge the call. |
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04:43.47 | [TK]D-Fender | goes off to do something productive |
04:44.29 | Wired_Life1 | termas_ but in this case the call are already answered and time is running and making costs |
04:44.34 | [TK]D-Fender | leifmadsen: Another idiot inexplicably trying to use a fork as a hammer when he really needs TOILET PAPER |
04:44.55 | leifmadsen | [TK]D-Fender: no comprende senior |
04:45.02 | termas_ | You must answer the call. |
04:45.06 | [TK]D-Fender | leifmadsen: thats the problem. |
04:45.15 | leifmadsen | [TK]D-Fender: :) |
04:45.23 | leifmadsen | I don't want to meet your mom! I just want... |
04:45.27 | termas_ | Since when do you get charged for receiving a call anyway? |
04:45.36 | [TK]D-Fender | leifmadsen: its the "lets jsut shove apps in, dial shit, and wait, WTF does this have to do with my goal?" |
04:45.40 | leifmadsen | termas_: forever. |
04:45.51 | [TK]D-Fender | leifmadsen: Not tonight... I have a headache :p |
04:46.01 | leifmadsen | [TK]D-Fender: booooooooooooooooooooooooooooooooooo |
04:46.04 | termas_ | So, anyway.. |
04:46.08 | [TK]D-Fender | ~!!! |
04:46.11 | infobot | i heard !!! is BANG BANG BANG at http://www.starterupsteve.com/swf/Group_X_video.html |
04:46.11 | leifmadsen | calls his girlfriend |
04:47.45 | termas_ | Can anyone explain what this debug output means: http://pastebin.com/d6cf10fa7 |
04:48.57 | Wired_Life1 | oh man you are really sure there is no way to ask my softphone before the recording stuff? |
04:49.14 | leifmadsen | ask your softphone what? |
04:49.56 | [TK]D-Fender | Wired_Life1: How many times do we have to say it. If the macro says to reject the call then it is NOT ANSWERED |
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04:51.03 | jblack | Unless he has an answer() before the dial... |
04:51.33 | [TK]D-Fender | jblack: Which we don't see as he has not even shown a TEST CALL or the actual problem HAPPENING |
04:52.00 | [TK]D-Fender | jblack: Just keeps PB-ing broken code he doesn't understand |
04:52.01 | jblack | why ARE we SHOUTING so MUCH? |
04:52.01 | Wired_Life1 | you mean exten => 890,n,Record(${SCREEN_FILE}.gsm|6|25) is not answering the call from anonymous? |
04:52.03 | jblack | :) |
04:52.15 | leifmadsen | jblack: apparently you haven't met [TK]D-Fender |
04:52.21 | [TK]D-Fender | Wired_Life1: Why are you asking them to record their name? |
04:52.32 | jblack | heh. Good point. |
04:52.46 | [TK]D-Fender | jblack: Bloody hell you havent :p |
04:53.20 | drmessano | what THE eff |
04:53.30 | Pan3D | Wired_Life1: it's like this... the call has to have been answered (as in audio flowing) for the record to do anything |
04:53.37 | Wired_Life1 | because the callerid is anonymous and i cant see the number? |
04:53.40 | drmessano | I want TO empasize THE wrong works, ok? |
04:53.44 | drmessano | words too |
04:53.46 | leifmadsen | s/THE/TEH/g |
04:53.48 | [TK]D-Fender | Wired_Life1: exten = r,11,Playback(screen-record) <- THIS will answer the call. Why are you doing it? |
04:53.55 | jblack | [TK]D-Fender: Some day, your heart is gonna explode. And if _it_ holds out, your head is gonna pop off like a cork. Sure you're not american? |
04:54.14 | drmessano | leifmadsen: Sorry, i'm not norse |
04:54.28 | leifmadsen | drmessano: well fuck you then :) |
04:54.28 | Wired_Life1 | screen-record: Please record your name press pound when finished. |
04:54.40 | [TK]D-Fender | jblack: Yes, I'm aiming for tactical-nuclear capacity and then I'm going to pay a visit to his town.... |
04:54.53 | drmessano | jblack: I think IRC is KEEPING him FROM shooting up A McDonalds.. So let it be |
04:55.02 | Pan3D | lol |
04:55.22 | [TK]D-Fender | Wired_Life1: You answered the call to ask their name. YOU. If you don't want to answer the call, WTF are you doing playing audio and trying to record a damn name? |
04:55.36 | [TK]D-Fender | looks areound for a water tower |
04:55.50 | drmessano | [TK]D-Fender: Sounds like some awful spammy project |
04:55.56 | jblack | [TK]D-Fender: That's gonna be messy. |
04:55.59 | Wired_Life1 | i want this after i have said: "ok its ok" |
04:56.09 | drmessano | "If you do not wish to accept this call, give your name and hang up now" |
04:56.13 | drmessano | Telespammer |
04:56.27 | drmessano | Im calling shenanigans |
04:56.38 | [TK]D-Fender | drmessano: this appears to be inbound |
04:56.52 | Pan3D | it's inbound judging by the pb |
04:56.58 | drmessano | [TK]D-Fender: Im calling Nihilism |
04:56.59 | [TK]D-Fender | Wired_Life1: Look when you're asking. You don't get to make this a 2-stage approval process. |
04:57.25 | [TK]D-Fender | Wired_Life1: and what is Oli supposed to base his decision on concerning whether or not to take the call? |
04:57.25 | drmessano | Maybe he just likes pain |
04:57.34 | Pan3D | ha |
04:57.37 | jblack | Heh. Like a cheap robotic secretary that always has to ask whether or not he wants to take the call. |
04:57.39 | drmessano | Nihilomasochism? |
04:57.58 | Wired_Life1 | my english is not so good |
04:58.15 | drmessano | Neither is mine |
04:58.17 | Wired_Life1 | and i dont find good german support |
04:58.29 | [TK]D-Fender | Narcochism = Wanting all the pain to be about you. |
04:58.33 | drmessano | Someone called me a douchebag and I thought it meant "Guy with tight pants" |
04:59.27 | [TK]D-Fender | Wired_Life1: You can't think in a straight line. thats your porblem. You can't even describe steps in order. |
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05:00.50 | drmessano | ONEEEEE |
05:00.52 | drmessano | TWOOOOOO |
05:00.55 | drmessano | FOUR |
05:01.27 | Wired_Life1 | ok... if a call comes from anonymous i want to ring on my softphone if i want to accept and then i will make a record of callers voice |
05:02.48 | Wired_Life1 | of course only if i said ok |
05:02.56 | Wired_Life1 | you know what i mean? |
05:04.18 | [TK]D-Fender | wireWHY would you "accept"? |
05:05.06 | Wired_Life1 | i can only accept if i on my pc |
05:05.28 | [TK]D-Fender | Wired_Life1: Again, WHY would you "accept"? |
05:06.41 | Wired_Life1 | why? because some people call me without callerid(num) |
05:08.33 | [TK]D-Fender | Wired_Life1: So they have no callerid. What makes you decide to ACCEPT TEH CALL OR NOT? |
05:09.08 | termas_ | [TK]D-Fender That's the problem, he doesn't know. He wants to signal the phone, in advance |
05:09.18 | drmessano | Based on WHAT? |
05:09.25 | termas_ | ie, pass the callerid to the end point. |
05:09.26 | [TK]D-Fender | termas_: no... |
05:09.27 | drmessano | Im lost |
05:09.40 | [TK]D-Fender | drmessano: You're not alone, and you're not the first |
05:09.41 | termas_ | He wants the end point to display that the call is anon. |
05:10.07 | [TK]D-Fender | But what fucking human decision is being made on whether to accept it or not? |
05:10.21 | [TK]D-Fender | "Or its TUESDAY, why the fuck not!" |
05:10.54 | termas_ | Well, the idea, (this is what I think he means) is that prior to the call being sent to Oli, the caller is recorded. |
05:11.14 | [TK]D-Fender | termas_: No, that will ANSWER the call <- |
05:11.22 | [TK]D-Fender | ter#1 violation. |
05:11.25 | termas_ | but, Yeah, |
05:11.38 | [TK]D-Fender | termas_: No "buts" |
05:11.39 | termas_ | It will answer the call, you can't record audio from a hung up channel. |
05:11.47 | termas_ | but it doesn't go to end point. |
05:11.49 | Wired_Life1 | i know that |
05:12.04 | Wired_Life1 | oh man you still dont know what i mean |
05:12.15 | [TK]D-Fender | Wired_Life1: WHY would you choose to "accept" this caller? |
05:12.58 | Wired_Life1 | i will know whos behind |
05:13.15 | [TK]D-Fender | Wired_Life1: How will you know? THEY HAVE TO callerid! |
05:13.18 | [TK]D-Fender | no* |
05:13.55 | Wired_Life1 | asterisk will record a voice sample from the person? |
05:14.19 | [TK]D-Fender | Wired_Life1: THAT WILL ANSWER THE FUCKING CALL! |
05:14.37 | termas_ | It will answer the call, but you can drop them into hold music.. |
05:14.38 | [TK]D-Fender | Wired_Life1: How can you record from the caller if you haven't answered? |
05:14.50 | Wired_Life1 | right i will that make asterisk answer the call after i said ok |
05:14.59 | [TK]D-Fender | Wired_Life1: NO |
05:15.11 | Wired_Life1 | why |
05:15.18 | termas_ | Wired_Life1 by the time the call reaches the end point, it's too late. |
05:15.31 | [TK]D-Fender | Wired_Life1: You said you don't want to answer until you accept. You accept based on the recordin. in order to record THE CALL HAS ALREADY BEEN FUCKING ANSWERED |
05:16.35 | termas_ | Within 30 secs 69 calls +10 61.61 % |
05:16.35 | termas_ | Within 45 secs 77 calls +8 68.75 % |
05:16.35 | termas_ | Within 60 secs 84 calls +7 75.00 % |
05:16.35 | termas_ | Within 75 secs 91 calls +7 81.25 % |
05:16.35 | termas_ | Within 90 secs 96 calls +5 85.71 % |
05:16.37 | termas_ | Within 91+ secs 112 calls +16 100.00 % |
05:16.40 | [TK]D-Fender | Wired_Life1: you can't record the name and THEN answer. its too damn late. |
05:16.43 | termas_ | gack. |
05:16.58 | Wired_Life1 | i know |
05:17.15 | Wired_Life1 | you said the playback is answering |
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05:18.15 | *** part/#asterisk HoosierDaddy (n=chatzill@c-67-167-74-95.hsd1.in.comcast.net) |
05:18.49 | termas_ | So, watching [TK]D-Fender working his way to an aneruysm is fun and all, but I'm not getting any closer to my solution.. |
05:18.53 | [TK]D-Fender | Wired_Life1: Seriously... how retarded are you. You can't play a prompt asking for a name, record it, and THEN think "Oh I don't want it to count as answered yet!" |
05:19.45 | [TK]D-Fender | Wired_Life1: You seem to have several key problems with things like "temporal mechanics", and "linear progression" |
05:20.01 | [TK]D-Fender | Wired_Life1: And "logic" and "stuff" |
05:20.08 | Wired_Life1 | nononono |
05:20.13 | Wired_Life1 | hear |
05:20.18 | [TK]D-Fender | Oh very certinly "yes" |
05:20.28 | Wired_Life1 | after i say its ok i can make costs |
05:20.34 | Wired_Life1 | it |
05:20.47 | [TK]D-Fender | Wired_Life1: You have no REASON to make it "ok" since you can't record without CAUSING COSTS |
05:20.54 | [TK]D-Fender | Wired_Life1: You can't go BACK IN TIME |
05:21.00 | termas_ | I can go back in time. |
05:21.16 | termas_ | I feel like i've been going forward and back in time ever since this conversation started. |
05:21.19 | [TK]D-Fender | knocks termas_ back into last tuesday |
05:21.40 | Wired_Life1 | thats it why i have make 2 dials |
05:21.40 | termas_ | 1. State problem. 2. Listen to flaws with concept, 1. State problem. |
05:22.19 | [TK]D-Fender | Wired_Life1: NO |
05:22.20 | Wired_Life1 | 1 dial to me to check if ok and then the answer with record that makes costs |
05:22.29 | [TK]D-Fender | Wired_Life1: You have answered the call. You are FUCKED |
05:22.53 | [TK]D-Fender | Wired_Life1: You have no REASON to accept the call. |
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05:23.25 | [TK]D-Fender | Wired_Life1: Because you can';t get the name without answering. Answering automatically costs you. You can't chang your mind afterwards! |
05:23.56 | [TK]D-Fender | Wired_Life1: And you have nothing to base a decision to answer on without taking a name or something. |
05:24.03 | Wired_Life1 | i know you said serveral times |
05:24.10 | [TK]D-Fender | Wired_Life1: then what don't you get? |
05:24.57 | Wired_Life1 | i want to check first if iam there |
05:25.20 | [TK]D-Fender | Wired_Life1: And if you are? |
05:25.43 | Wired_Life1 | i click ok anywhere |
05:25.53 | Wired_Life1 | and if not the call runs into timeout |
05:26.15 | [TK]D-Fender | Wired_Life1: there is no fucking click. You're on a softphone. You can answer, you can press digits. There is no "OK" |
05:26.43 | [TK]D-Fender | Wired_Life1: If you're there are you always going to pick up the phone? |
05:27.24 | HoosierDaddy | any Broadvoice customers here? |
05:28.32 | Wired_Life1 | if im on the pc i will pick up |
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05:29.24 | [TK]D-Fender | Wired_Life1: Then what? |
05:29.34 | Wired_Life1 | the thing is i will only make costs for the caller if i there to call |
05:29.44 | Wired_Life1 | you know what i mean? |
05:30.08 | Wired_Life1 | its fucking shit if i record and let the caller wait and im not there |
05:32.05 | [TK]D-Fender | Wired_Life1: Now you are only doing this for anonymous calls. So if they aren't anonymous then you are happy to immediately charge them? |
05:33.59 | Wired_Life1 | if they aren't anonymous the call go directly to me |
05:34.29 | [TK]D-Fender | Wired_Life1: they BOTH go to you. How is anonymous "special"? |
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05:35.45 | Wired_Life1 | anonymous can record voice after a trigger from me so i can play the record and know who is there |
05:36.17 | Wired_Life1 | are you still dont know what i will make? |
05:37.23 | [TK]D-Fender | Wired_Life1: Why aren't you checking if you're there for NON anonymous callers? |
05:37.39 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
05:37.52 | [TK]D-Fender | Wired_Life1: Why are only the anonymous ones getting the chance to be ignored by you at no cost? |
05:38.19 | Wired_Life1 | ^^ you are funny |
05:38.33 | Wired_Life1 | because it is so ^^ |
05:39.15 | kc8pxy | [TK]D-Fender: because he only wants to have it bother with anonymous "spamers" or bill collectors :) |
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05:39.45 | [TK]D-Fender | kc8pxy: you are missing the fact they get a BENIFIT from this for no good reason |
05:40.22 | [TK]D-Fender | Anyways, enough bacwards logic and broken descptions for one (lifetime) day. |
05:40.23 | [TK]D-Fender | I'm off. |
05:40.29 | kc8pxy | [TK]D-Fender: mebbe I'm only reading part of the conversation. |
05:41.42 | Wired_Life1 | i know what he mean |
05:42.02 | Wired_Life1 | its not possible to check who is there without making costs |
05:42.09 | Wired_Life1 | lol |
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05:52.17 | kc8pxy | i apologize if I'm even more noob that i seem, but i tried to get my 1.6.0.9 server up, with a single sip channel definition, and i can't seem to get ekiga to register to it. from all the howto's I've read I'm doing things right. |
05:52.34 | kc8pxy | but my asterisk does not seem to be listening.. anyone willing to help? |
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06:05.01 | kc8pxy | admittedly, this is my very first attempt at makign asterisk work. anyone? |
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06:05.42 | HoosierDaddy | I'm no expert either...do you now how to look for sip registration errors? |
06:06.35 | kc8pxy | HoosierDaddy: not really.. asterisk -vvv only complains that i have no modules.conf or features.conf |
06:07.37 | kc8pxy | HoosierDaddy: event_log is empty |
06:08.34 | HoosierDaddy | asterisk -rvvvvvv |
06:08.47 | HoosierDaddy | sip show peers |
06:09.37 | HoosierDaddy | your asterisk box is on your local subnet I presume too |
06:09.39 | termas_ | kc8pxy iptables? |
06:10.34 | kc8pxy | No such command 'sip show peers' (type 'help sip show' for other possible commands) |
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06:11.13 | kc8pxy | termas_: don't think either client or server have any rules, and ther is just a switch between them. |
06:12.05 | wierdo | kc8pxy, If modules.conf is not loaded and chan_sip, you cannot sip show peers |
06:12.06 | termas_ | 1.6.0.9 is much newer than mine, sip show peers works for me.. |
06:12.47 | HoosierDaddy | from * CLI, "sip show ?" doesn't list peers as an option? |
06:13.04 | kc8pxy | wierdo: so i need load chan_sip at a minimum? |
06:13.57 | HoosierDaddy | peers |
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06:14.19 | wierdo | kc8pxy, yes. module load chan_sip.so |
06:14.35 | wierdo | not sure if this is the correct syntax in 1.6.x |
06:15.30 | orangeservice | Hi all, I have two SIP trunks to two different gateways... I am trying to make it so that any inbound call on either of them gets through to the same set of extensions on my gateway... anybody have experience with this? |
06:16.15 | wierdo | kc8pxy, ls /usr/lib/asterisk/modules/ |
06:16.21 | wierdo | is anybody there :) |
06:16.39 | wierdo | or just no modules.conf, or empty, check permissions |
06:16.43 | *** join/#asterisk keebler (n=Christop@h36.247.20.98.dynamic.ip.windstream.net) |
06:16.57 | keebler | Anyone try using Gizmo5 Business with Asterisk? |
06:17.40 | termas_ | orangeservice set both the initial contexts to the same context? |
06:17.56 | termas_ | orangeservice, or set the initial context of one, to goto the other.. |
06:18.41 | kc8pxy | got it |
06:18.57 | orangeservice | termas_: I tried that but strangely enough having them both sent to same context doesnt work... the context works for one provider, but not the other (that being said, I am quite new to asterisk and telephony in general) |
06:19.06 | kc8pxy | sip show peers now shows my sip peer, but it's nto connected.. trying now. |
06:19.26 | kc8pxy | YES! |
06:20.07 | kc8pxy | at least i have a registered peer :) from here, everything is just how to fold the origami, yes? |
06:20.13 | termas_ | can you do anything with the trunk that doesn't work? |
06:20.51 | HoosierDaddy | good deal kc8pxy |
06:20.54 | termas_ | yes kc8pxy, but you should probably look into your iptables.. no rules is odd. |
06:21.27 | orangeservice | I can make outbound calls via it - it does have to be said though, the trunks are quite different in the way they authenticate and so on |
06:21.38 | drmessano | keebler: I have.. works fine |
06:21.51 | kc8pxy | termas_: nothing needed on the lan, it's my house. i have a beefy gateway with paranoid rules. |
06:21.55 | keebler | drmessano: How well does extension calling work? |
06:22.01 | keebler | drmessano: From a mobile phone? |
06:22.14 | drmessano | From a Blackberry, shitty |
06:22.17 | termas_ | kc8pxy oh, my mistake. |
06:22.23 | keebler | drmessano: Damn |
06:22.24 | drmessano | Not sure about Windows Mobile |
06:22.32 | keebler | I only have blackberries |
06:22.40 | drmessano | Its not a SIP client |
06:22.40 | keebler | 8110 |
06:22.48 | keebler | Not true sip? |
06:22.50 | keebler | grr |
06:23.03 | drmessano | It just initiates call bridging |
06:23.07 | keebler | RIM BB_MVS is too fucking expensive. |
06:23.11 | drmessano | There is no SIP client for the BB |
06:23.14 | keebler | Yes |
06:23.15 | keebler | there is |
06:23.16 | keebler | :/ |
06:23.25 | keebler | But its only through RIM |
06:23.41 | drmessano | Thats not a SIP client.. thats native OS support |
06:23.45 | keebler | And they say it won't work with Asterisk. |
06:23.54 | drmessano | You cannot download and install a client |
06:24.10 | keebler | Yeah, thats what I'm discovering. |
06:24.11 | kc8pxy | keebler: intentionally, or ommisively? |
06:25.09 | keebler | kc8pxy: Said they tried testing it, but the configurations can't be standardized for every client, and there are too many factors that would cause it to work improperly, and they said they couldn't get the ATA;s to register properly for extension calling |
06:25.16 | keebler | kc8pxy: Talked to their project lead today. |
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06:25.33 | kc8pxy | _X. matches to "press anything, right? |
06:25.59 | kc8pxy | rather, "pick a number, any number!" |
06:26.01 | keebler | afaik |
06:26.29 | keebler | I haven't worked on my asterisk server in 2 months. |
06:26.35 | keebler | Already started to forget crap. |
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06:27.17 | keebler | drmessano: Can you recommend any GSM/GPRS Phone that properly supports SIP? |
06:27.23 | keebler | We have to use ATT. |
06:27.40 | keebler | I can change from BB if I have to. |
06:28.06 | drmessano | Not sure |
06:28.18 | keebler | The clients just need to be able to dial an extension from the phone properly. |
06:28.26 | keebler | with no diminished call quality. |
06:28.33 | keebler | And still maintain its own extension |
06:29.12 | keebler | http://na.blackberry.com/eng/services/blackberry_mvs |
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06:29.44 | keebler | Its its only stand alone software os that works in conjunction with the BES and PBX |
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06:29.59 | keebler | So theoretically you'd need three servers. |
06:30.05 | keebler | BES/MVS/PBX |
06:30.30 | keebler | And they charge you $8K JUST to install the system. |
06:30.37 | keebler | Won't even let you manage it on your own. |
06:30.57 | aesiamun | hi, is it possible to get Asterisk to dial two numbers at the same time but set the callerid differently for each one? I'm using the system to ring internally to sip phones and to my cell phone. The problem is that i can't control callerid text on my cell. So I would like to use a coding system in my address book. |
06:31.36 | kc8pxy | http://rafb.net/p/bj2Qr310.html |
06:31.54 | kc8pxy | will that be the nice test i hope it will be? |
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06:35.32 | aesiamun | i'd be interested in knowing if anyone has been able to do that before. |
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06:38.51 | kc8pxy | WHEE!!! a split! |
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06:39.42 | k-man | anyone here use pennytel? |
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06:50.57 | k-man | anyone know where i might find documentation on how to write diaplans for a linksys spa2102? |
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06:59.21 | kc8pxy | k-man: wish i knew.. I'm just trying to get my softphone to get noises played at it. |
06:59.52 | kc8pxy | for Playback(), do i need any specific modules? |
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07:07.31 | k-man | kc8pxy: not afaik, do you get any sound or error messages when you try and play a sound? |
07:07.47 | k-man | kc8pxy: its very probably permissions. thats the problem i had when i was trying to do it |
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07:14.12 | k-man | is there some tool i can use to ping a host every 30 minutes to see if its reachable? |
07:16.36 | kc8pxy | k-man: cron? |
07:18.31 | k-man | oh - i meant on my windows mahcine |
07:18.33 | k-man | machine |
07:18.45 | kc8pxy | k-man: all i see on teh sip sotfphone is dialing sip:1 , and then i get it disconnected. asterisk -rvvvvvv shows the following, every time i try to dial that. |
07:19.04 | kc8pxy | <PROTECTED> |
07:19.41 | k-man | kc8pxy: sorry, no idea |
07:20.09 | k-man | kc8pxy: you'll have to post your extensions.conf and sip.conf and a log of what happens when you try and dial |
07:20.19 | k-man | kc8pxy: but I haev to go now so I can't help you |
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07:29.25 | wierdo | k-man, you can use dude: http://www.mikrotik.com/thedude.php |
07:29.26 | kc8pxy | ok, so now i have beeping going on. |
07:29.55 | kc8pxy | but tt-mokeys.gsm should not sound like beeping, should it? |
07:30.05 | kc8pxy | monkeys.gsm |
07:30.09 | wierdo | free tool, does some advanced monitorin,can ran as service, does ping at some period if configured, pretty simple and cool |
07:32.13 | kc8pxy | wierdo: ideas? |
07:34.14 | wierdo | kc8pxy, let me read some posts up |
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07:35.57 | wierdo | kc8pxy, sorry, could you repeat the problem |
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07:39.37 | kc8pxy | wierdo: i can get my sip softphone to connect, and i can dial a number, but I'm not getting what I'm expecting. it seems to get hung up on before i hear monkeys. repost of extensions.conf comming. |
07:40.28 | wierdo | kc8pxy, incopatible codecs ? |
07:40.57 | kc8pxy | wierdo: I'm too noob to know. |
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07:42.19 | wierdo | kc8pxy, pastebin sip.conf for softphone, what softphone are you using |
07:43.16 | kc8pxy | wierdo: ekiga |
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07:44.01 | kc8pxy | http://rafb.net/p/afGrlO88.html is my extensions |
07:44.54 | orangeservice | hi, having a strange one - I am using WaitExten() to grab user input... when asterisk tries to place the call to the exten, ties extention@inbound-context, not extention@local-ext-context like I want it to.... how would I tell WaitExten to use the numbers in a local context? |
07:45.14 | kc8pxy | wierdo: http://rafb.net/p/AxiAw511.html is sip.conf |
07:47.39 | kc8pxy | wierdo: that help? |
07:48.15 | wierdo | allow=alaw |
07:48.15 | wierdo | allow=ulaw |
07:48.49 | mosty | orangeservice, pastebin your dialplan and sip.conf (assuming your local extensions are sip) |
07:49.13 | kc8pxy | DOH |
07:49.20 | wierdo | kc8pxy, add these to sip.conf |
07:49.34 | kc8pxy | i remove the path, and just say tt-monkeys, and it works :-/ |
07:49.45 | kc8pxy | from the playback |
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07:50.54 | kc8pxy | yay me :0 |
07:50.56 | kc8pxy | :) |
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08:00.58 | wierdo | kc8pxy, i have configured it working ekiga, asterisk, no problem |
08:01.21 | wierdo | kc8pxy, ulaw, alaw are configured by default on the softphone |
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08:04.36 | wierdo | kc8pxy, in configuration druid what type of connection you've specified |
08:06.31 | wierdo | kc8pxy, maybe if setup as default 56k, ekiga switch to some lossy codec which is not allowed in sip.conf |
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08:32.19 | jgoo | Is it possible to setup a call group so that people have to dial the extension to take the call (and just picking up won't accept the call) |
08:32.24 | jgoo | or is that the definition of another feature? |
08:33.19 | jgoo | I was thinking to use a queue for this, does that sounds more applicable? |
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08:46.39 | mosty | with a queue, you could dial an extension to join the queue |
08:47.19 | mosty | you would then be able to receive calls from the queue |
08:47.53 | jgoo | mosty: I would like that behaviour but without having to dial an extension to join |
08:48.08 | Chris-NB | hi |
08:48.09 | jgoo | however, I want the extension to ring, and let people 'pickup' the extenion. |
08:48.27 | mosty | jgoo, you can make sip extensions static members of a queue |
08:48.39 | jgoo | aaah. right. so queue is the right way to do that then? |
08:48.49 | jgoo | and they will see the queue extension flashing on their phone, press it to take another call? |
08:49.41 | Chris-NB | anyone using the 'd' option in the dial application? |
08:49.44 | jgoo | I suppose I would also want one actual phone to ring, but just one, then after 20 seconds, pass to another queue or call group that rings all. |
08:49.52 | jgoo | any ideas on what the setup would be? |
08:50.13 | Chris-NB | to dial a 1 digit extension during a call is beeing established, to exit to that extension? |
08:51.06 | mosty | jgoo, find an asterisk queue tutorial/guide- perhaps there is one in the book |
08:51.08 | mosty | ~thebook |
08:51.08 | infobot | from memory, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
08:52.09 | Chris-NB | I've configured the EXITCONTEXT variable and dtmf is transmitted correctly to the server |
08:52.27 | Chris-NB | but the server ignores my key press during an call establishment |
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09:55.27 | orangeservice | mosty - sorry for the delay, manage to solve it... now left with another pickle. WaitExten is only registering the 1st digit of extentions dialed.... booo, anybody seen thins? |
09:56.44 | mosty | like i said before, pastebin |
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10:10.53 | orangeservice | abbreviated sip.conf - http://pastebin.com/d6b8c42aa |
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10:13.13 | orangeservice | @ mosty: and extensions.conf http://pastebin.com/d6cca6c24 |
10:16.33 | orangeservice | and the faily: http://pastebin.com/d24f48571 |
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10:44.12 | mosty | orangeservice, what extension are you trying to dial? |
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11:08.25 | timater123 | can anybody help me, i have an issue with dahdi |
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11:10.15 | timater123 | i setup an asterisk exchange on centos 5.3, the problem is that i have to run dahdi_genconf each time i boot my asterisk server otherwise my analog phones wont work |
11:11.55 | defsdoor | timater123: sounds like you need some modules loading |
11:13.31 | timater123 | defsdoor: i commented out every module except wctdm because i have a tdm400p card |
11:14.11 | defsdoor | timater123: reboot - do lsmod > ~/modlist |
11:14.20 | defsdoor | run dahdi-genconf |
11:14.28 | defsdoor | do lsmod > ~/modlist2 |
11:14.32 | defsdoor | diff modlist modlist2 |
11:14.52 | defsdoor | I'm assuming dahdi-genconf is loading some modules |
11:15.02 | defsdoor | seems most likely reason |
11:16.04 | timater123 | diff returned no output |
11:17.09 | defsdoor | dahdi_genconf just writes a config file (I'm looking at it now) |
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11:17.21 | defsdoor | so unless your config file is being clobbered - I'm clueless |
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11:23.14 | timater123 | defsdoor: are you talking about /etc/dahdi/system.conf |
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11:30.04 | timater123 | <PROTECTED> |
11:30.49 | timater123 | defsdoor: modlist file gives a difference when i restart asterisk after dahdi_genconf |
11:31.02 | defsdoor | what modules ? |
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11:32.48 | timater123 | diff modlist modlist3 |
11:32.48 | timater123 | 5c5 |
11:32.48 | timater123 | < wctdm 39884 2 |
11:32.48 | timater123 | --- |
11:32.48 | timater123 | > wctdm 39884 4 |
11:32.49 | timater123 | 7c7 |
11:32.51 | timater123 | < dahdi 190728 9 dahdi_echocan_mg2,wctdm,wcfxo |
11:32.53 | timater123 | --- |
11:32.57 | timater123 | > dahdi 190728 13 dahdi_echocan_mg2,wctdm,wcfxo |
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11:40.37 | tmjb | hello my asterisk is crashing on ubuntu amd 64 asterisk[28472]: segfault at 31 ip 00007fcb7c50d1bd sp 0000000040cec9b0 error 4 in libpri.so.1.4[7fcb7c505000+2f000] ? |
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11:51.18 | tmjb | U use stable asterisk 1.6.0.8 |
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12:03.16 | leifmadsen | tmjb: have you tried latest 1.6.0 branch? Are you able to reproduce it? Also, make sure you have DONT_OPTIMIZE enabled in menuselect under the Compiler Flags. Once you've done all that, you can open a bug on the tracker at bugs.digium.com |
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12:16.07 | thedonvaughn | ~gsm-bug |
12:16.26 | thedonvaughn | gsm bug still exist with asterisk and gcc-4.2 or greater? |
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12:26.42 | Jakobsen | Hi there. Can anyone point me to a good guide, that describes how to create a High Availability Asterisk cluster with two servers? |
12:26.59 | leifmadsen | Jakobsen: I don't know of any document for that |
12:27.27 | Jakobsen | leifmadsen, same conclusion here :) |
12:27.42 | leifmadsen | I doubt such a document exists; what do you expect something like that to have? |
12:28.21 | [TK]D-Fender | Jakobsen: You'll find it filed somewhere between "Invisible flying unicorns" and Jimmy Hoffa ;) |
12:29.15 | leifmadsen | thedonvaughn: http://bugs.digium.com/view.php?id=13846 |
12:29.40 | Jakobsen | Well.. I'll just create a Heartbeat and DRBD cluster with Asterisk on top of it then :) |
12:29.51 | leifmadsen | Jakobsen: heh, that's pretty much the way to do it |
12:29.56 | Jakobsen | That should do the trick.. Just wanted to know, if there were any easier way.. |
12:30.03 | Jakobsen | Thank you for helping anyways.. :) |
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12:42.40 | Dr-Linux|home | Hi guys, |
12:43.48 | Dr-Linux|home | why my pattren doesn't work with + sign? i.e. exen => _+1NXXXXXXXXX,1,Goto(ivr,s,1) |
12:43.49 | Dr-Linux|home | ? |
12:44.12 | [TK]D-Fender | Dr-Linux|home: show us |
12:44.38 | thedonvaughn | is there a way to unregister a sip provider through the cli? |
12:45.26 | Dr-Linux|home | [TK]D-Fender: I did |
12:45.37 | Dr-Linux|home | <PROTECTED> |
12:45.45 | [TK]D-Fender | Dr-Linux|home: No you didn't. Show a complete dialplan and failed call attempt |
12:46.07 | [TK]D-Fender | Dr-Linux|home: You aren't showing a failure, jsut a single unvalidated line that we should not take at face value |
12:46.48 | Dr-Linux|home | [TK]D-Fender: sure, let me try access again on that server and reproduce the error |
12:47.14 | [TK]D-Fender | Dr-Linux|home: And you appear to be hand-typing that line as well given the fact the spelling on "exten" changed between each. |
12:47.33 | Dr-Linux|home | [TK]D-Fender: BTW, generally this line is okey? the one i shown? |
12:47.49 | Dr-Linux|home | yeah sorry for that |
12:48.46 | Dr-Linux|home | [TK]D-Fender: actually I just wanted to confirm that _+1NXXXXXXXXX is right as it is using + sign |
12:54.07 | mmlj4 | how do you dial a plus sign? |
12:55.01 | [TK]D-Fender | Dr-Linux|home: You're making claims. Back it up |
12:55.35 | [TK]D-Fender | mmlj4: Think "Outside NANPA" |
12:56.00 | mmlj4 | fair enough |
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13:00.32 | Dr-Linux|home | <PROTECTED> |
13:01.18 | Dr-Linux|home | [TK]D-Fender: the problem is _+1NXXXXXXXXX is on top but it when i call it ignored |
13:02.00 | Dr-Linux|home | and it goes below and pick the hard coded number |
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13:02.12 | [TK]D-Fender | Dr-Linux|home: pastebin EVERYTHING. |
13:02.28 | Dr-Linux|home | ok |
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13:02.38 | leifmadsen | Dr-Linux|home: uhhh.... hard coded is always going to match first |
13:02.46 | kaii | mh .. when SIP/10 dials "SIP/11&SIP/12&SIP/13" all of these three peers go into ringing state (given they are not busy or unreachable). does anybody can point me how to obtain this information? (via AMI, CLI, custom app, whatsoever) |
13:02.46 | leifmadsen | most specific will get matched before a pattern match |
13:02.52 | kaii | i need to check somehow if a channel is ringing on more than 1 peer .. |
13:02.56 | kaii | in a stupid "ringall" setup the customer now wants to put priority on transfers and internal calls .. means making internal calls/transfers possible, even if the target is in ringing state with the external ringgroup |
13:03.25 | leifmadsen | kaii: what version? |
13:03.52 | kaii | you dont want to know.. 1.2-bristuffed ... |
13:04.06 | leifmadsen | kaii: not sure you will have any method then |
13:04.08 | kaii | i am able to patch/backport, if necessary. |
13:04.32 | Jas_Williams | Dr-Linux|home: If you do a show dialplan context it will show you the evaluation order |
13:04.53 | Jas_Williams | from cli |
13:05.15 | kaii | leifmadsen: extstate for example can tell me if an extension is ringing or not .. but it does not resolve how many channels are actually ringing |
13:05.56 | Dr-Linux|home | Jas_Williams: nice idea |
13:05.58 | leifmadsen | kaii: you could set a variable that contained that info |
13:06.57 | kaii | leifmadsen: how? if i Dial(SIP/11&SIP/12&SIP/13) and two of them are busy/congested ... how do i detect this during Dial() is executed? |
13:07.43 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
13:09.20 | Dr-Linux|home | Jas_Williams: CLI logs are too fast due to heavy traffic :) also i've lot of self defined conf files i.e. did.conf tf.conf etc .. |
13:09.27 | [TK]D-Fender | kaii: "core show channels concise" <- get parsing |
13:09.29 | kaii | leifmadsen: the external caller goes into Dial() which results in "n" new channels in ringing state ... and after that dial is executed its to late to set the variable .. only app_dial could do this |
13:10.29 | [TK]D-Fender | kaii: then don't ring your devices directly, do it vial a Local channel |
13:10.48 | [TK]D-Fender | v/vial/via/ |
13:10.56 | [TK]D-Fender | s/vial/via/ |
13:10.59 | [TK]D-Fender | sdhshffdsfhbfdg |
13:11.36 | kaii | in the local channel i can increment, dial, and decrement a global variable. ok .. but there is still a race condition for the time dial() needs to detect a busy/congested channel and jump to the next (decrementing) priority |
13:11.43 | mmlj4 | v/ ? |
13:12.58 | [TK]D-Fender | kaii: And I gave you another option already |
13:13.38 | kaii | im still trying to find out what "show channels concise" does and if i somehow can achieve this on asterisk 1.2 ... i'm still bound to that old stuff |
13:14.25 | [TK]D-Fender | kaii: A CLI command, it ain't Raw-Cat Science. |
13:14.30 | kaii | ok 1.2 does support consise too. |
13:15.15 | [TK]D-Fender | never would have guessed... |
13:15.30 | kaii | [TK]D-Fender: seems this could work .. thx for pointing |
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13:16.32 | tamiel | hello, I need to set sip and iax users/peers in postgresql tables. I must use res_odbc or res_config_pgsql ? |
13:16.43 | Katty | drags in |
13:16.53 | [TK]D-Fender | Katty: Mew. |
13:17.12 | Katty | hugs [TK]D-Fender and falls asleep mid-hug |
13:18.31 | Katty | i need sleep. |
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13:19.41 | Jas_Williams | Dr-Linux|home: how about calling just the command eg asterisk -x "core show dialplan context" |
13:19.51 | kaii | i need the missing link between channels in "show channels [concise]" |
13:20.03 | Dr-Linux|home | Jas_Williams: yes I got that |
13:20.29 | Jas_Williams | then will display to your shell session rather than in logs |
13:20.52 | Dr-Linux|home | Jas_Williams: it is showing my desired line at the end, so this line will be executed as last option: |
13:20.52 | Dr-Linux|home | '_+1NXXXXXXXXX' => 1. Answer() [pbx_config] |
13:20.52 | Dr-Linux|home | <PROTECTED> |
13:21.20 | kaii | [TK]D-Fender: parsing show channels isnt a solution too. :-( the link between the initiating channel and the ringing ones is missing .. you can detect "SIP/10" is ringing, but not realiably "Zap/1 is ringing on SIP/10" |
13:21.25 | Jas_Williams | After anything above that may match |
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13:21.29 | Dr-Linux|home | Jas_Williams: but i'm configured it at the top in the context ... this is a problem |
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13:22.05 | eppigy | hello |
13:22.08 | eppigy | i am dave |
13:22.15 | Jas_Williams | Have a read on asterisk parsing orders or put everything else in an include |
13:22.18 | Katty | bops eppigy with pillow. |
13:22.21 | [TK]D-Fender | eppigy: O S U R |
13:22.36 | [TK]D-Fender | Dr-Linux|home: Where's the failed call? |
13:22.47 | Katty | eppigy: you missed spaghetti |
13:22.49 | Dr-Linux|home | [TK]D-Fender: no fialed call |
13:22.53 | eppigy | D: |
13:23.00 | eppigy | spaghettle |
13:23.06 | [TK]D-Fender | Dr-Linux|home: Show us it failing to match as its supposed to. |
13:23.23 | Dr-Linux|home | but the line is not executing that i want . |
13:23.34 | Katty | eppigy: yeah, guess what i had for breakfast. |
13:23.37 | Dr-Linux|home | hhm.. |
13:23.42 | Dr-Linux|home | let me try another thing |
13:24.15 | Dr-Linux|home | Jas_Williams: I've a number to way to get what i want, but just wanted to understand the trick |
13:24.32 | eppigy | Katty: hrmmmm |
13:24.41 | eppigy | Katty: eggs benedict? |
13:25.54 | Katty | <PROTECTED> |
13:25.59 | Katty | plam? |
13:26.02 | rob0 | plum |
13:26.09 | Katty | you're a plum. |
13:26.13 | Katty | PROFESSOR PLUM. |
13:26.25 | Katty | where were you last night, with the candle stick?! |
13:26.30 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
13:26.31 | rob0 | I know. In the Ballroom with the candlestick. |
13:27.23 | Jas_Williams | Dr-Linux|home: look at this what a nightmare :) http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sorting |
13:27.50 | Dr-Linux|home | Jas_Williams: thanks dude |
13:28.09 | kaii | how can i find out which channel initiated another new channel? e.g. Zap/1 executes Dial(SIP/10) .. in show channels i see that Zap/1 is in Dial(..) and see that SIP/10 is in ringing state, initiated by AppDial ... instead of seeing that the channel SIP/10-xyz was initiated by AppDial, i want to go further and know that AppDial was initiated by Zap/1 and therefore know that Zap/1 is calling SIP/10 |
13:28.11 | eppigy | I love eggs benedict |
13:28.31 | Katty | eppigy: i don't think i've ever had eggs benedict. |
13:28.36 | Katty | googles. |
13:28.38 | eppigy | man it is so grood |
13:28.55 | Katty | is that cheese? |
13:29.14 | [TK]D-Fender | Hollandaise = if bacon doesn't kill you, this will |
13:29.35 | [TK]D-Fender | Katty: a cream & egg-yolk based sauce |
13:29.43 | Katty | eww. |
13:29.46 | eppigy | hollondaise sauce |
13:29.58 | eppigy | man it is so good |
13:29.59 | Katty | this might have something to do with my allergy to egg. |
13:30.05 | eppigy | :[ |
13:30.10 | eppigy | you cannot eat egg? |
13:30.10 | leifmadsen | tips his cap towards Ms. Katty |
13:30.13 | Katty | it's just a mild allergy tho. |
13:30.22 | eppigy | I love eggs so much |
13:30.23 | Katty | eppigy: i can. it just makes me naeusous for like an hour |
13:30.27 | eppigy | D: |
13:30.30 | Katty | hugs leifmadsen |
13:30.37 | leifmadsen | is happy he is not allergic to any foods |
13:30.40 | eppigy | I eat at cracker barrel, IHOP, and waffle house |
13:30.45 | eppigy | every week |
13:30.50 | Katty | we have a cracker barrel here. |
13:30.54 | Katty | but.. never been there. |
13:31.01 | eppigy | uncle herchel's favorite |
13:31.06 | Katty | they don't have nutrition info posted on their website )= |
13:31.07 | eppigy | herschel's? |
13:31.15 | eppigy | COUNTRY HAM |
13:31.46 | rob0 | Katty, I'd call that "can" a definite "can't". :) |
13:31.54 | Katty | have you had cocacola ham? |
13:32.08 | *** join/#asterisk SilentGreen (n=chatzill@201.217.56.16) |
13:32.17 | eppigy | negative |
13:32.28 | rob0 | If I wanted to be nauseous for an hour, I'd watch Faux News. |
13:32.44 | Katty | fox news is pretty bad. |
13:32.52 | Katty | it's something about 911 or terrorists every 5 minutes. |
13:33.03 | eppigy | TERRISTS |
13:33.05 | Katty | eppigy: you get a ham. from the store. uncooked. |
13:33.10 | Katty | eppigy: and 2 bottles of coke. |
13:33.18 | Katty | eppigy: and you boil the ham in the coke for like... 4 hours. |
13:33.25 | eppigy | SNAP |
13:33.35 | Katty | eppigy: then cover it in molassas, ground mustard, and cloves. then bake. |
13:33.46 | eppigy | oh wow |
13:33.50 | leifmadsen | eppigy: don't forget about the terrorist democrats! |
13:33.53 | eppigy | and it tastes good? |
13:33.54 | rob0 | diet coke and Mentos |
13:34.02 | Katty | rob0: also fun. |
13:34.08 | eppigy | leifmadsen: that is all part of the partisan elusion |
13:34.16 | Katty | eppigy: it's a nigella bites classic |
13:34.20 | eppigy | every 4 years we are pitted against eachother |
13:34.23 | SilentGreen | i prefer coke zero and mentos... ;o)# |
13:34.29 | eppigy | rather than identifying our true adversary |
13:35.03 | eppigy | *illusion |
13:35.06 | eppigy | jesus |
13:35.32 | eppigy | i guess elusion is corrcet as well though |
13:35.33 | eppigy | NICE |
13:35.56 | Katty | eppigy: http://www.cookstr.com/recipes/ham-in-coca-cola |
13:36.12 | Katty | eppigy: you will never eat ham out again. |
13:37.19 | SilentGreen | need help with asterisk establishing connection from one asterisk to another works but voice isn't send, with a bit luck just for 2 secs |
13:37.37 | eppigy | Katty: it sounds intriguing |
13:38.07 | SilentGreen | maybe there is some intelligent software at the telephone and provider side that anlyses the stream if it is voice or data |
13:38.14 | SilentGreen | any chance to cheat? |
13:39.12 | SilentGreen | is there a possibility to get rtp-stream tunneled? or something else? |
13:39.20 | *** join/#asterisk juanjoc (n=jcomella@200.69.219.113) |
13:39.24 | SilentGreen | that the provider can't block VoIP |
13:39.28 | eppigy | well you can of course tunnel it |
13:39.43 | eppigy | but that has nothing to do with asterisk |
13:40.16 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
13:40.27 | SilentGreen | that's right, it's ssh but is it possible to tunnel an rtp-stream? |
13:40.38 | eppigy | you can tunnel anything |
13:40.55 | [TK]D-Fender | eppigy: Sounds dirty :p |
13:41.02 | eppigy | i know :D |
13:41.27 | eppigy | gurl i wanna tunnel that rtp stream till next july |
13:41.42 | SilentGreen | ok, sorry. i am new to asterisk and voip, thats why i'm asking if it is possible to tunnel an rtp-stream... |
13:41.48 | eppigy | well I mean |
13:42.04 | SilentGreen | by the way. i tunneled my gurl last night... ;o) |
13:42.04 | eppigy | have you checked to make sure there are no firewall rules on your end |
13:42.10 | eppigy | that could eb to blame |
13:42.14 | eppigy | SilentGreen: NICE |
13:43.01 | rob0 | "Tunneled", how? Define what you mean. Some means of tunnelling, like over ssh, won't work for RTP. |
13:43.04 | eppigy | I hope your key length was sufficient to maintain data integrity |
13:43.13 | rob0 | oh hush boys |
13:43.21 | eppigy | lol |
13:44.03 | Katty | eww. |
13:44.15 | Katty | we can't talk about this stuff so early. |
13:44.30 | rob0 | See, now Katty has the ewws. You meanies. |
13:44.36 | eppigy | FLUFFY KITTENS |
13:44.41 | Katty | much better. |
13:44.52 | *** join/#asterisk djMax (n=chatzill@66.92.91.132) |
13:45.01 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
13:45.23 | djMax | Does anyone know how trackable phone numbers (e.g. one phone number per print ad) work? Do the telco's handle it all, or do you need a local PBX? |
13:45.34 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
13:45.39 | SilentGreen | eppigy: ok i check... let's tunnel it... ; |
13:45.45 | Katty | hugs [intra]lanman |
13:45.46 | rob0 | Tunnelling via a VPN like openvpn is as simple as using the right IP addresses with functional routing tables on both ends. |
13:46.01 | eppigy | yes I am a fan of openvpn |
13:46.25 | SilentGreen | you mean "open" for "tunneling"??? lol |
13:46.30 | eppigy | oh boi |
13:46.30 | Katty | eww. |
13:47.00 | eppigy | covers Katty's ears |
13:47.04 | Katty | kthx. |
13:47.17 | SilentGreen | oh sorry... my fault. i mean these nice tunnels for fluffy kittens they can run through and play... |
13:47.27 | eppigy | FLUFFY ORANEG KITTENS |
13:47.30 | eppigy | ORANGE |
13:47.58 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
13:48.06 | Katty | http://icanhascheezburger.files.wordpress.com/2008/04/funny-pictures-happy-fluffy-orange-cat.jpg |
13:48.58 | SilentGreen | CUUUUUTE! |
13:49.17 | eppigy | ^______^ |
13:49.39 | Katty | soda all gone :< |
13:49.54 | eppigy | my rockstar is gone |
13:50.15 | SilentGreen | maybe, because some kind of offtopic for this channel??? :D |
13:50.33 | [intra]lanman | hugs katty... |
13:50.41 | [intra]lanman | Katty: what's up girl? |
13:50.55 | Katty | [intra]lanman: not sure yet. |
13:51.05 | Katty | [intra]lanman: so far i have 1 ticket open. a samsung box is off by 5 minutes. |
13:51.42 | [intra]lanman | off? |
13:51.43 | eppigy | ntpd |
13:51.52 | Katty | we're talking about a samsung 7100 here |
13:51.58 | Katty | it doesn't do ntpd |
13:52.06 | Katty | it does manaul. |
13:52.08 | Katty | manual. |
13:52.22 | eppigy | what r that |
13:52.32 | Katty | what your mom does. |
13:52.36 | Katty | to the clocks in her house. |
13:52.45 | eppigy | oh |
13:52.50 | eppigy | :< |
13:52.52 | rob0 | omg, tmi |
13:52.58 | [intra]lanman | rofl |
13:53.08 | Katty | TOPIC CHANGE: http://icanhascheezburger.files.wordpress.com/2008/04/funny-pictures-happy-fluffy-orange-cat.jpg |
13:53.24 | SilentGreen | manual??? how nasty... i thought its abaout cute fluffy kittens |
13:53.42 | SilentGreen | thx katty |
13:55.33 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
13:56.47 | SilentGreen | ok, please short back to topic ;o) |
13:57.10 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
13:57.34 | SilentGreen | firewall is opened for rtp and the related ports on both sides??? thats annoying... |
13:58.26 | SilentGreen | few months ago it worked perfectly, and now it seems to be blocked... |
13:58.36 | eppigy | lol |
13:58.39 | eppigy | yeah bro |
13:58.56 | SilentGreen | eppigy: what? |
13:59.03 | eppigy | well I mean did you check? |
13:59.14 | eppigy | I thought you were affirming |
13:59.16 | eppigy | that indeed |
13:59.20 | eppigy | they were not open |
13:59.22 | eppigy | on both ends |
13:59.52 | SilentGreen | no, i only checked it again... |
14:00.07 | SilentGreen | and its ok, so far. |
14:00.27 | SilentGreen | would have been strange, because there were no changes |
14:02.02 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
14:02.14 | SilentGreen | and like i said it worked until few weeks ago |
14:02.57 | Katty | hugs jaytee |
14:03.12 | jaytee | hugs Katty |
14:04.06 | eppigy | I LIKE THAT BOOM BOOM POW |
14:04.16 | *** part/#asterisk SilentGreen (n=chatzill@201.217.56.16) |
14:05.36 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
14:05.37 | *** mode/#asterisk [+o Qwell] by ChanServ |
14:07.44 | Great_Anta_Baka | i want to wait till a user enters 4 digits.. is this waitexten(4) ? |
14:08.50 | eppigy | or Read(variable|filename|4) |
14:09.08 | Great_Anta_Baka | ty.. will try that out |
14:09.21 | eppigy | god speed friend |
14:13.42 | *** join/#asterisk grmartin (n=grmartin@c-76-110-3-120.hsd1.fl.comcast.net) |
14:13.52 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
14:14.16 | grmartin | Does anyone have an example of dynamically generating a dial in menu from a db? maybe in perl? |
14:15.11 | Great_Anta_Baka | mmm eppigy i want it to block the outgoing call until i have put in all the digits of my pin code |
14:15.22 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.68.173) |
14:17.21 | [TK]D-Fender | Great_Anta_Baka: So have them enter a value, check it and continue if it matches. |
14:18.14 | xrmx__ | anyone has a sip tapi software to reccomend? |
14:18.17 | Great_Anta_Baka | should i do waitexten(-1) instead.. i am trying to program this in an agi script on an asterisk 1.2 box |
14:19.15 | [TK]D-Fender | Great_Anta_Baka: No, you should be using the AGI commands for getting input. |
14:19.33 | [TK]D-Fender | Great_Anta_Baka: Not typying to call dialplan apps directly for it |
14:21.00 | Great_Anta_Baka | i see |
14:27.08 | pif | hi, has debian stopped updating its asterisk package ? |
14:28.12 | [TK]D-Fender | pif: What version do they list? |
14:28.26 | pif | 1:1.4.21.2~dfsg-3 |
14:28.47 | [TK]D-Fender | pif: and what version does the /topic list? |
14:29.13 | pif | your point? |
14:29.35 | [TK]D-Fender | pif: if (1:1.4.21.2~dfsg-3 != 1.4.24.1) then omfg |
14:29.43 | *** join/#asterisk Wired_Life (n=Chatzill@mgdb-4db8140e.pool.einsundeins.de) |
14:29.53 | Wired_Life | good morning |
14:30.08 | eppigy | hi |
14:30.10 | Wired_Life | hello again [TK]D-Fender |
14:30.29 | keebler | Won't even let you manage it on your own.aaaaaaaaaaaaaaaasdfghjkl;'asdfghjjjjjjjjjkl;'sxdcfghjkl;/'Zsxdfghjk |
14:30.40 | eppigy | calm down guy |
14:30.50 | pif | tzafrir_laptop: you there? |
14:31.03 | [TK]D-Fender | keebler: ? |
14:31.15 | *** join/#asterisk ludan (n=daniele@unaffiliated/ludan) |
14:31.34 | keebler | The aalkhla;h; was my kids. The other was in reference to Blackberry MVS installation. |
14:31.39 | Wired_Life | today i have a simple yes no question.... is its possible to check if a sip softphone is online? |
14:31.48 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
14:31.50 | Wired_Life | via dialplan |
14:32.17 | [TK]D-Fender | Wired_Life: "core show application chanisavail" |
14:32.18 | keebler | [TK]D-Fender: Tring to find a decent SIP capable ATT phone. |
14:32.36 | Wired_Life | thx |
14:32.39 | [TK]D-Fender | keebler: AT&T makes SIP phones? |
14:32.55 | [TK]D-Fender | keebler: and why would you limit yourself to them even if they do? |
14:32.56 | keebler | [TK]D-Fender: Not that I'm aware. |
14:33.06 | keebler | [TK]D-Fender: Our business |
14:33.18 | *** join/#asterisk machoman48 (n=machoman@wifi-eduroam.mendelu.cz) |
14:33.19 | keebler | [TK]D-Fender: I need a dual capable phone |
14:33.33 | [TK]D-Fender | keebler: Is it named "WeWhoMustBuyAT&T"? |
14:34.07 | keebler | Blackberry has a really sweet BES Suite but its too damn expensive and they won't support Asterisk. |
14:34.41 | keebler | [TK]D-Fender: My boss signed the contract before I joined. Its what everyone in this area has. Verizon doesn't work in the field. |
14:34.57 | keebler | So yes. For this part of Texas. ATT is a must. |
14:35.04 | [TK]D-Fender | keebler: What does this have to do with a phone that does SIP & analog? |
14:35.21 | keebler | I need a cell phone that does SIP. |
14:35.33 | [TK]D-Fender | keebler: Oh wait, not its CELLULAR as well? |
14:35.39 | keebler | Yes. |
14:35.46 | [TK]D-Fender | keebler: Perhaps you could provide a new & complete description. |
14:35.49 | keebler | ATT + SIP capable phone. |
14:36.00 | [TK]D-Fender | ATT is a COMPANY, not a TECHNOLOGY |
14:36.01 | keebler | I never really consider ATT for landline anymore. |
14:36.28 | djMax | iPhone does SIP |
14:36.31 | djMax | ish |
14:36.39 | keebler | Okay, my clients want to be able to call the local VOIP Lan from their Cell phone, and have their phone act as a detached extension |
14:37.00 | keebler | djMax: So does BB and many others. But most of the implimentations are too complicated for my clients. |
14:37.18 | djMax | that's a bit trickier, yeah. It would murder battery performance pretty much on all phones |
14:37.29 | keebler | Shit. Late for work. I'll return in 45 minutes. |
14:37.34 | [TK]D-Fender | keebler: So a cell with a VoiP client. |
14:38.53 | *** join/#asterisk jtodd (n=jtodd@144.sub-70-214-184.myvzw.com) |
14:38.53 | *** mode/#asterisk [+o jtodd] by ChanServ |
14:39.05 | *** join/#asterisk curious101 (n=q@125.212.122.188) |
14:40.26 | curious101 | hi! my question is: will asterisk run fine on a server with 2 IP addresses? can it be configured to listen to just one interface? |
14:41.23 | blebleble | curious101: sure |
14:42.38 | curious101 | blebleble: can you explain further please? |
14:44.19 | *** join/#asterisk Wired_Life1 (n=Chatzill@mgdb-4db87385.pool.einsundeins.de) |
14:44.59 | blebleble | just set the bindaddr |
14:47.43 | *** join/#asterisk Gabriel25 (n=gabe@pool-72-68-157-205.nycmny.fios.verizon.net) |
14:49.05 | curious101 | blebleble: ah, I see. Thank you. |
14:51.51 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
14:55.16 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
14:57.43 | Nugget | poor dvossel |
15:03.28 | *** join/#asterisk bakermd (n=bakermd@38.101.225.215) |
15:04.12 | bakermd | How can I change my config so that when it sends email voicemails it connects to a different smtp server? |
15:04.45 | rob0 | That would be an issue in your MTA, not in *. |
15:06.26 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
15:06.45 | *** join/#asterisk jmls (n=jmls@host217-36-208-155.in-addr.btopenworld.com) |
15:06.48 | jmls | hey guys |
15:07.41 | jmls | trying to get 2 instances of asterisk running on the same box (1.6 and 1.4) |
15:07.41 | jmls | got everything made and installed into the appropriate separate directories |
15:07.41 | jmls | started 1.4 ok |
15:07.41 | jmls | trying to run 1.6, I get "Asterisk already running on /var/run/asterisk.ctl" |
15:08.00 | jmls | is there an option to specify where asterisk.ctl can be placed for each instance ? |
15:08.26 | rob0 | I suppose you'd want them to have different settings in asterisk.conf <=== see that |
15:08.38 | Corydon76-dig | jmls: no, but there's an option to source asterisk.conf from different locations, and one of the options in asterisk.conf is where the ctl file is located |
15:09.03 | *** join/#asterisk zapotek6 (n=edpman@mail.comelit.it) |
15:09.19 | jmls | Corydon76-dig, that's what I was looking for. Thanks |
15:09.29 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) [NETSPLIT VICTIM] |
15:10.30 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
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15:11.23 | *** join/#asterisk route_ (n=skarecro@96.57.183.37) |
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15:16.13 | cutlass | hello...can anyone tell me what it means to get a <ZOMBIE> channel? |
15:18.56 | russellb | cutlass: That is a very interesting question, actually. :-) |
15:19.26 | russellb | Essentially, you're seeing what happens to channels when you do something like a transfer. |
15:19.50 | russellb | The way it works internally to Asterisk is that a new channel gets created, all of the "guts" of the old channel are moved to the new one, and the old one becomes a Zombie. |
15:20.01 | russellb | Now, you don't really need to understand why (nor do you need to care) why this is the case ... |
15:20.18 | russellb | but it is expected to see zombie channels with transfers, parking, redirects ... |
15:21.11 | cutlass | ok...so it's not to be interpreted as an error then, right? |
15:21.27 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
15:22.23 | *** join/#asterisk mandd (n=moo@69.172.67.1) |
15:23.47 | cutlass | sorry...I just read your last statement which has the answer to my last question :) |
15:23.59 | cutlass | thanks Russell |
15:24.16 | russellb | you're welcome |
15:26.24 | *** join/#asterisk DGTL_Magician (n=boerg@siona.servers.nosco-ict.nl) |
15:26.29 | jaytee | damn zombies! they're everywhere |
15:26.49 | jaytee | just ask Woody Harrelson |
15:27.00 | outtolunc | says brainsssssss! |
15:27.17 | DGTL_Magician | I like zombies |
15:27.34 | DGTL_Magician | except on my server :) |
15:33.44 | *** join/#asterisk Erol_ (n=x@88.234.108.194) |
15:36.32 | Erol_ | does SIP proxy mean PSTN gateway? |
15:37.46 | [TK]D-Fender | Erol_: No. |
15:37.55 | mmlj4 | no |
15:38.58 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
15:39.33 | *** join/#asterisk ilowe (n=ilowe@modemcable230.43-82-70.mc.videotron.ca) |
15:39.53 | Erol_ | i went to iptel.org web site |
15:39.59 | plundra | One of my sip-providers gives me "T-Bone 2.3.3" as the User-Agent, anyone know what company that is? |
15:40.16 | *** join/#asterisk bbryant (n=olpc@68.208.65.50) |
15:40.48 | plundra | I can't seem to find _anything_ about it :) |
15:41.25 | Erol_ | and I got a bit confused |
15:41.52 | Erol_ | does SER do the same thing asterisk do? |
15:42.27 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
15:43.22 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
15:44.31 | [TK]D-Fender | Erol_: Not at all |
15:45.08 | Erol_ | [TK]D-Fender: uhm. could you give me a crash course, =) |
15:45.26 | *** join/#asterisk c4t3l (n=c4t3l@c-76-31-57-251.hsd1.tx.comcast.net) |
15:45.36 | c4t3l | hello world |
15:48.01 | [TK]D-Fender | ~book |
15:48.02 | 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 |
15:48.04 | [TK]D-Fender | Erol_: ^^^ |
15:48.26 | *** part/#asterisk machoman48 (n=machoman@wifi-eduroam.mendelu.cz) |
15:50.34 | Great_Anta_Baka | lol.. i remember when you did the same thing to me.. although i did have a hardcopy of the 1.2 book at hand |
15:50.36 | eppigy | [TK]D-Fender: when does your lecture on TDM technologies start? |
15:50.54 | Great_Anta_Baka | and does it go online? |
15:52.07 | [TK]D-Fender | eppigy: Usually following the receipt of a personalized training payment :) |
15:52.19 | eppigy | NICE |
15:53.01 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
15:53.16 | deadpigeon | is it better to use an IDE or a SCSI drive on an asterisk server? |
15:53.38 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
15:54.12 | [TK]D-Fender | deadpigeon: SCSI is typically more reliable and faster, so do the math :) |
15:54.45 | mort_gib | deadpigeon: Sata RAID 5 |
15:55.34 | deadpigeon | thanks. im just experiencing some unsatisfactory results with dahdi_test and im trying to narrow down the culprits |
15:55.41 | [TK]D-Fender | MFM RAID 0 :p |
15:55.56 | [TK]D-Fender | deadpigeon: and in other unrelated news... |
15:56.22 | [TK]D-Fender | deadpigeon: Try shoing some useful information because that last bit was a wildly aimed pot-shot |
15:56.35 | [TK]D-Fender | showing* |
15:56.59 | deadpigeon | I think I can handle it myself for now. I was merely asking what was preferred storage devices. |
15:58.01 | mort_gib | deadpigeon: Would dhadi_test have anything to do with HDD performance????? |
15:58.01 | deadpigeon | It could if there was a lot of HD activity or DMA settings were wrong. |
15:58.01 | tzafrir_laptop | pif, yes |
15:58.51 | deadpigeon | Anyone know off hand if I have to recompile dahdi_tool to use watchdog for IRQ misses like we used to with zaptel? |
15:59.41 | Erol_ | I understand that SER is something like an IP router. If I am willing to set up a pbx for for example 10 extensiona would I need SER? |
16:02.32 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
16:02.58 | [TK]D-Fender | Erol_: A SIP router is not a PBX. PBX typically entails backend services like voicemail, IVR's, etc. this is core functionality with *. * also speaks a LOT more than just SIP. |
16:03.17 | [TK]D-Fender | Erol_: If you want a PBX, and want it to speak SIP, then * could very welll be for you |
16:04.59 | djMax | Is 2B channel transfer likely to work on an XO PRI, and/or is there an easy way to find out if it's enabled? |
16:05.23 | Erol_ | [TK]D-Fender: and by the way why people like SIP more than H323? |
16:05.42 | Erol_ | [TK]D-Fender: as far as I can see from the specs H323 has more features |
16:05.59 | djMax | Erol_: I don't think they're really on the same level |
16:06.00 | [TK]D-Fender | Erol_: SIP is in plain-text an there are some other fine points but nothing I'm qualified to elaborate on that much. |
16:06.14 | djMax | SIP is about "rendezvous" more than transfer |
16:06.21 | djMax | transport I should say |
16:06.40 | [TK]D-Fender | Erol_: and SIP is a more "routed" infrastructure than H.3232 IIRC as well |
16:06.54 | _Steve_ | anone want to help me with vlan config on my polycoms? |
16:07.43 | Erol_ | [TK]D-Fender: a website says that best solution for a pbx would be * + SER. |
16:08.01 | djMax | a website? for real? |
16:08.13 | [TK]D-Fender | Erol_: taht is a rather unqualified statement. |
16:08.13 | eppigy | i will argue about this on a forum |
16:08.42 | [TK]D-Fender | Erol_: SER can help in certain large & redundant scenarios, but not for most user's needs |
16:08.46 | eppigy | Erol_: do they mean like multiple asterisk instances with SER for sip registration balancing |
16:08.47 | eppigy | ? |
16:09.05 | eppigy | its uncanny |
16:09.19 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
16:09.19 | eppigy | how [TK]D-Fender and I's thought processes just mesh |
16:09.21 | mmlj4 | I haven't even looked at ser |
16:09.28 | Erol_ | http://www.en.voipforo.com/ser/ser_asterisk.php |
16:09.30 | mmlj4 | guess I should at some point |
16:09.47 | [TK]D-Fender | mmlj4: think "basilisk" :p |
16:09.54 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
16:10.34 | [TK]D-Fender | Erol_: But regardless of that you have not been at all specific about the scope of your needs. |
16:10.35 | eppigy | SER by itself it not very useful but SER teamed with Asterisk is how you make Asterisk scale. |
16:10.45 | eppigy | keyword "scale" |
16:11.02 | djMax | where scale I assume is defined as hundreds or thousands of connections |
16:11.09 | djMax | because under that it would seem overkill |
16:11.12 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
16:11.15 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
16:11.15 | *** mode/#asterisk [+o russellb_] by ChanServ |
16:11.23 | Katty | where are we having lunch today? |
16:11.47 | *** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com) |
16:11.51 | [TK]D-Fender | Katty: General tao.... I feel soo... white..... |
16:12.01 | Erol_ | djMax: so you mean its for large solutions? |
16:12.13 | djMax | from my reading of it, yes. |
16:12.33 | [TK]D-Fender | Erol_: "large" is a matter of perspective |
16:12.36 | mmlj4 | just what I need, scaly software |
16:12.37 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
16:12.50 | [TK]D-Fender | hands mmlj4 a fillet knife |
16:13.05 | _Steve_ | hmm, i don't even see a VLAN setting in the polycom web interface... |
16:13.05 | Erol_ | [TK]D-Fender: pbx with thousand extensions amybe? |
16:13.06 | djMax | which brings us back to the question of what Erol_ actually needs |
16:13.13 | djMax | how many active calls? |
16:13.21 | [TK]D-Fender | Erol_: Yeah, maybe a thousand... |
16:13.29 | [TK]D-Fender | Erol_: Also depends on what they're doing... |
16:13.46 | Erol_ | [TK]D-Fender: just simple phones |
16:13.54 | Erol_ | [TK]D-Fender: talking |
16:13.58 | djMax | I suppose another way to look at it is if you'll need more than one asterisk box, you might consider SER. |
16:14.00 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
16:14.05 | *** join/#asterisk keebler (n=Christop@adsl-99-179-135-231.dsl.rcsntx.sbcglobal.net) |
16:14.13 | keebler | Okay, that was a bit longer than 45 minutes... |
16:14.18 | djMax | (IANASE - I am not a SER expert) |
16:14.27 | [TK]D-Fender | Erol_: all at the same time>? recording? transcoding? other processor intensive stuff? ivr's with ton's of DB lookups? |
16:14.35 | djMax | how many PRI's do you have, or is it all VOIP? |
16:15.04 | [TK]D-Fender | PRI? Never heard it.... |
16:15.16 | keebler | [TK]D-Fender: Back to my original description. I need a GSM/GPRS Cell phone that can do SIP over both WiFi AND GPRS/UMA, and it has to work on ATT network. Any ideas? |
16:15.34 | Erol_ | djMax: not so detailed |
16:15.54 | [TK]D-Fender | keebler: Good to see you managed to boil that down to a complete & coherent single question :) |
16:15.55 | djMax | keebler - "and is easy for noobs to use" right? |
16:16.04 | keebler | djMax: yes. |
16:16.19 | keebler | [TK]D-Fender: Yeah, I had JUST woken up this morning during that chaotic convo. |
16:16.29 | [TK]D-Fender | keebler: And.... no, SIP = plenty of clients... its the quality of network & integration with the phone I'm not so sure of. |
16:16.33 | Erol_ | and what are communigate and sipcat? what is the difference between them and asterisk? |
16:16.42 | eppigy | Katty: I am thinking krystak |
16:16.43 | keebler | djMax: My clients don't even know how to reboot a router. |
16:16.46 | eppigy | krystal |
16:16.55 | djMax | well, that might not earn them noob status. :) |
16:16.57 | keebler | djMax: And they make $120k A YEAR. |
16:16.59 | Erol_ | I mean I know they are for pbx solutions but do they offer anyth,ng more than *? |
16:17.20 | eppigy | no |
16:17.25 | keebler | to them, it just "has to work". |
16:17.28 | eppigy | You need ser |
16:17.30 | djMax | The iPhone sip client is good, IMHO, except no background running |
16:17.38 | eppigy | at the point where your cpu load |
16:17.43 | djMax | (unless you want to recharge every 3 hours) |
16:17.43 | eppigy | is too high |
16:17.45 | keebler | djMax: Is it real SIP? Or just call-through? |
16:17.49 | eppigy | and you need to scale |
16:17.50 | djMax | real sip |
16:17.53 | [TK]D-Fender | Erol_: Do those products let you do whatever you want for any call placed? |
16:17.54 | djMax | jailbroken phones only though |
16:17.58 | keebler | Ah |
16:18.00 | keebler | Damn |
16:18.01 | djMax | there's a non-jailbroken skype client |
16:18.15 | keebler | Not sure Skype will work. and Gizmo5 sucks ass. |
16:18.27 | keebler | (Not real SIP on the Gizmo, I know.) |
16:18.36 | djMax | wonder if android has anything |
16:18.54 | [TK]D-Fender | Erol_: Dial 911 on a tuesday night where the Raiders won their last home game and it happens to be raining and my PBX can decide to boil a pot of coffee rather than terminate your call. Can those other products do that? |
16:19.04 | Erol_ | [TK]D-Fender: i dont know, I just asked maybe you have more exp about them |
16:19.36 | [TK]D-Fender | Erol_: You mean we have to know these other products that you don't as well as * AND provider you a complete comparison? Would you like fries with that sir? :) |
16:19.47 | djMax | android might be more promising and/or palm pre |
16:19.54 | Maliuta | [TK]D-Fender: but can you teach it to use my espresso machine? |
16:19.59 | Erol_ | [TK]D-Fender: i just said maybe man.. |
16:20.05 | [TK]D-Fender | Maliuta : MAYBE |
16:20.10 | djMax | though none of them can touch the iPhone for quality IMHO. |
16:20.18 | Maliuta | [TK]D-Fender: if you can show me the dialplan/AGI for that I'll be your eternal slave ;) |
16:20.32 | *** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27) |
16:20.59 | [TK]D-Fender | djMax: iPhone is nice hardware... and not to say I'm a FOSS-Nazi, I just refuse to have my ass owned by a stupid fruity logo'd company :) |
16:21.35 | [TK]D-Fender | Maliuta: Send me the manual for your machine and we'll takl :) |
16:21.37 | [TK]D-Fender | talk* |
16:22.03 | coppice | All the iPhones here are sold unlocked, but they only unlock the network. you still need to jailbreak them to do anything useful |
16:22.39 | [TK]D-Fender | coppice: and there are bigger processors out there and so much more that can be done... very sad this is the best we've got |
16:22.59 | coppice | what's wrong with the processor? |
16:24.26 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
16:24.48 | [TK]D-Fender | coppice: Just saying there is bigger out there. Many free OS's to base things on. non-proprietary connectors for common stuff. |
16:25.02 | [TK]D-Fender | coppice: jsut that everyone locks you down in some nasty way or another |
16:25.56 | coppice | The only 2 real pains with the iphone are the need to jailbreak, and the lack of real multitasking (and on a Unix platform as well). The other stupid stuff like no cut and paste seems to have been fixed on the latest revision |
16:26.56 | *** join/#asterisk los415 (n=los415@sfca-office.corp.race.com) |
16:27.09 | [TK]D-Fender | coppice: lack of stereo bluetooth, tethering restrictions (more a carrier grievance), etc |
16:27.33 | [TK]D-Fender | coppice: proper flash support, as well |
16:27.38 | coppice | I think the stereo bluetooth is also fixed now. |
16:27.59 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
16:28.04 | coppice | they seem anti-flash, but on a wince machine flash really drains the battery |
16:28.08 | los415 | has anyone used the fax for asterisk product yet? |
16:28.47 | coppice | that's a strange product. why didn't they do V.34 FAX? Commetrex say they have it |
16:29.05 | [TK]D-Fender | los415: Nobody wants to break the Freshness Seal :) |
16:29.15 | los415 | lol |
16:29.49 | los415 | i want to try the t.38 termination right to asterisk and do fax to email but i dont want to beat my head on the desk for hours trying toget it to work |
16:29.52 | coppice | I think they could have got some buzz around a V.34 FAX package, but a me too? |
16:31.08 | los415 | i guess there is only one way to figure out if it works or not |
16:31.40 | coppice | do they supply fax to email software with it? |
16:32.15 | los415 | well it says with 1.6 it will create a tiff |
16:32.23 | coppice | if you look around there are several very simple scripts to do it. doing email to FAX well is rather more involved |
16:32.23 | los415 | so i can figure it out from there |
16:32.38 | coppice | everyone creates a TIFF :-) |
16:33.14 | los415 | well yea but we already have a system that does the entire tiff to pdf to email |
16:33.26 | los415 | it's just getting constant good faxes |
16:33.32 | los415 | and not cut off mangeld |
16:33.50 | coppice | and what gives you cut off mangled FAXes? |
16:34.24 | los415 | then normal fax stuff that comes with trix |
16:35.45 | los415 | our switch supports t.38 and i have it working perfectly going to adtran ta's series cpe's ect |
16:35.59 | los415 | so if asterisk could terminate the t.38 would be nice |
16:36.14 | coppice | app_fax in 1.6 does that |
16:36.52 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
16:38.37 | mazpe | is there an default feature code for *67 in asterisk? |
16:39.43 | mazpe | or how can i block the caller id, for a particular call |
16:40.40 | [TK]D-Fender | mazpe: Depends what you are calling out via |
16:40.55 | [TK]D-Fender | mazpe: and no, there is no "default feature in *" for this. |
16:41.10 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
16:41.28 | mazpe | *67, gives dial tone and allows a call to be made with the caller id blocked |
16:41.45 | [TK]D-Fender | mazpe: Didin't answer my question... |
16:42.01 | mazpe | via a sip trunk |
16:42.07 | [TK]D-Fender | mazpe: You've just described common CLASS functionality without your specific circumstances |
16:42.10 | *** join/#asterisk rafiks (n=rafiks@c-68-56-8-98.hsd1.fl.comcast.net) |
16:42.24 | [TK]D-Fender | mazpe: Depends how they want you to tell them to block CID |
16:42.42 | pif | tzafrir_laptop: are you still packaging asterisk for debian? |
16:42.53 | tzafrir_laptop | pif, yes |
16:43.04 | mazpe | [TK]D-Fender: the provider allows me to manipulate the caller id. I can just make it 00000000 |
16:43.17 | pif | tzafrir_laptop: why are you still at 1.4.21 ? |
16:43.48 | [TK]D-Fender | mazpe: try "core show application setcallerpres" |
16:44.16 | mazpe | [TK]D-Fender: thanks |
16:45.28 | tzafrir_laptop | experimental is at 1.6.1-rc3 |
16:45.43 | tzafrir_laptop | I have testing packages of -rc4 at my repo |
16:45.54 | tzafrir_laptop | for lenny and etch as well |
16:46.16 | tzafrir_laptop | repos actually . the "-experimental" ones under: |
16:46.30 | tzafrir_laptop | http://updates.xorcom.com/pkg-voip/ |
16:46.44 | pif | 'apt-cache show asterisk' only shows 1.4.21 ... |
16:46.51 | tzafrir_laptop | e.g. deb http://updates.xorcom.com/pkg-voip/repo-i386-etch-experimental unstable main |
16:46.52 | pif | the package name has changed? |
16:46.58 | tzafrir_laptop | no |
16:47.06 | tzafrir_laptop | read what I wrote |
16:47.07 | pif | ah |
16:47.15 | rafiks | pif : update ur sources.list |
16:47.23 | pif | got that |
16:47.35 | tzafrir_laptop | the dahdi+libpri (removing bristuff) + asteirsk transition takes time |
16:47.39 | tzafrir_laptop | Lenny has 1.4.21 |
16:47.40 | pif | why not in debian repos? |
16:48.36 | pif | good to hear you are removing bristuff ;) |
16:48.52 | *** join/#asterisk phl4kx (n=supervis@webmailserver.nisira.com.pe) |
16:50.45 | *** join/#asterisk jtodd (n=jtodd@182.sub-70-214-133.myvzw.com) |
16:50.45 | *** mode/#asterisk [+o jtodd] by ChanServ |
16:54.57 | *** join/#asterisk ntbourey (n=ntbourey@c-76-110-3-120.hsd1.fl.comcast.net) |
16:55.09 | ntbourey | Hey every one |
16:55.44 | rafiks | ntbourey ; hey |
16:56.01 | ntbourey | I was wondering if someone could provide me with some assistance |
16:56.39 | Maliuta | no, we won't jump start your car for you |
16:56.49 | ntbourey | Haha I wish it was that simple |
16:57.10 | pif | tzafrir_laptop: Failed to fetch http://updates.xorcom.com/pkg-voip/repo-i386-etch-experimental/dists/unstable/main/binary-amd64/Packages |
16:58.02 | Maliuta | i386-etch-experimental ... with unstable and amd64? |
16:58.14 | Maliuta | sounds like a badly named repo ;) |
16:58.26 | ntbourey | Does anyone know how to integrate Perl into asterisk, res_perl doesn't seem to compile with 1.6 |
16:59.11 | Maliuta | perl for AGI's? |
16:59.37 | tzafrir_laptop | Maliuta, yeah. I should have a proper single repo. but I'm lazy to make this a proper setup |
16:59.39 | ntbourey | I guess I am still kind of new to this |
16:59.55 | RypPn | At a guess... http://asterisk.gnuinter.net/ |
16:59.56 | tzafrir_laptop | pif, use the -amd64 repo |
17:00.22 | Maliuta | ntbourey: define "perl in *" if it's not for AGI's |
17:00.41 | *** join/#asterisk stoffell (n=stoffell@d51A4D629.access.telenet.be) |
17:01.01 | ntbourey | Maliuta: res_perl I belive, allows you to use regular perl scripts |
17:01.02 | coppice | is it really AMD specific, or is it just general x86_64? |
17:01.07 | ntbourey | from within a dialplan |
17:01.11 | ntbourey | correct? |
17:01.20 | tzafrir_laptop | coppice, the name just stuck |
17:01.25 | Maliuta | that's what and AGI is |
17:01.34 | Maliuta | it can be anyone of a number of things |
17:01.43 | ntbourey | Okay |
17:01.45 | Maliuta | perl php ... |
17:01.49 | ntbourey | Right |
17:01.54 | coppice | tzafrir_laptop: Good. Keep reminding intel that their PR machine lies :-) |
17:02.07 | Maliuta | ntbourey: go google for perl agi |
17:02.10 | pif | tzafrir_laptop: better |
17:02.12 | ntbourey | do I just need to install asterisk-perl |
17:02.15 | ntbourey | ? |
17:02.34 | *** join/#asterisk axarob (n=ebash@cpc3-barn8-0-0-cust288.brnt.cable.ntl.com) |
17:02.35 | Maliuta | don't know your distro or how it's packaged |
17:02.45 | tzafrir_laptop | ntbourey, res_perl is quite unmaintained |
17:03.04 | ntbourey | I installed from source Maliuta |
17:03.07 | tzafrir_laptop | asterisk-perl probably refers to the perl asterisk module, which mainly include Asterisk::AGI |
17:03.08 | [TK]D-Fender | ntbourey: can you perhaps explain your expectations for "integration"? |
17:03.42 | ntbourey | I just want to be able to use Perl to handle most of my calling features of my PBX |
17:03.56 | Maliuta | [TK]D-Fender: I don't think he can ... sounds to me like stuff that could be done with Exec() or an AGI |
17:04.22 | [TK]D-Fender | ntbourey: " calling features" is also a very vague statement |
17:04.50 | ntbourey | Maliuta: Yes |
17:04.59 | Maliuta | I want my phone thingy to use the interthingy to do thingy |
17:05.02 | Maliuta | :) |
17:05.08 | ntbourey | up until yesterday I had no knoledge of Asterisk other than what it was |
17:05.19 | Maliuta | ntbourey: read the book |
17:05.23 | ntbourey | Did |
17:05.45 | [TK]D-Fender | ntbourey: then stop trying to hammer oddly shaped blocks into holes you can't even see yet :) |
17:06.01 | Maliuta | [TK]D-Fender: I like that one |
17:06.18 | ntbourey | this is what I am trying to do |
17:06.22 | ntbourey | im looking to derive menus and user number searching from a postgres database using perl |
17:06.28 | Katty | blergh |
17:06.37 | jaytee | phlegm |
17:06.55 | Katty | A... C.... Phlem |
17:07.11 | Maliuta | ntbourey: [TK]D-Fender is right. You should use normal means to achieve your setup, and see if you even need AGI's along the way |
17:07.11 | Maliuta | ntbourey: AGI |
17:07.25 | ntbourey | Okay |
17:07.47 | ntbourey | I already have my basic system working I can dial in, and get responses back |
17:08.05 | ntbourey | I'm trying to use a DB to handle generating menus |
17:09.26 | jaytee | I'm trying to use magic creams and lotions to make my hair grow back |
17:09.26 | [TK]D-Fender | ntIf you want advise you should dummy up a DB table with sample data that you would like to see be used in processing your call and then we can advise on viable ways to do itMaliuta |
17:09.39 | [TK]D-Fender | ntbourey: So show us what the table data would look like FIRST. |
17:09.48 | ntbourey | Can't |
17:09.56 | [TK]D-Fender | jaytee: I told you already.... ch-ch-ch-chia! |
17:10.21 | jaytee | yeah, but I was hoping for a different color than green |
17:10.22 | *** join/#asterisk NOT_guru (n=NOT_guru@24-241-103-142.static.stls.mo.charter.com) |
17:10.25 | Katty | jaytee: you're lovely as you are. you don't need chia hair. |
17:10.36 | Katty | jaytee: it won't make you any more charming. |
17:10.41 | jaytee | Katty, awww, gee. thanks! |
17:10.45 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
17:11.14 | Maliuta | Katty: how come you never say seductive^Wnice things to me? ;) |
17:11.28 | Katty | Maliuta: because you're not as charming as jaytee is. |
17:11.42 | ntbourey | http://pastebin.com/m345327d1 |
17:11.43 | Katty | Maliuta: you should work on that. |
17:11.47 | Maliuta | Katty: thats only because you haven't met me in person |
17:11.51 | jaytee | swells up and swaggers off to slay dragons |
17:12.01 | Katty | Maliuta: i've never met jaytee. |
17:12.01 | Maliuta | Katty: I'm much more charming in real life than online |
17:12.12 | Katty | Maliuta: well isn't that neat. |
17:12.17 | Katty | Maliuta: good for you. |
17:12.22 | Maliuta | Katty: IRC is for blowing off steam and making lude remarks ;) |
17:12.39 | jaytee | lude? as in Qualude? |
17:12.49 | Katty | lude-a-chris speed! |
17:12.50 | jaytee | or lewd as in "hey! nice tits!" |
17:13.03 | Katty | Maliuta: I might be a male. ever think of that?! |
17:13.17 | coppice | lewd? isn't that a compliment? |
17:13.21 | jaytee | ludicrous speed, rofl. I watched that again the other nite |
17:13.26 | Maliuta | Katty: I might not care ... ever think of that? :) |
17:13.31 | Katty | eww. |
17:13.47 | Maliuta | jaytee: go comb the desert for me ;) |
17:13.56 | jaytee | hahaha |
17:14.05 | [TK]D-Fender | ntbourey: How is that a "menu"? |
17:14.15 | [TK]D-Fender | ntbourey: thats jsut a PIN-list, not a menu structure. |
17:14.15 | jaytee | Princess Bride is still better |
17:14.22 | Katty | As you wish! |
17:14.26 | [TK]D-Fender | jaytee: aaassss yoooouuuu wiiiisssshhh!!!! |
17:14.28 | Katty | tumbletumbletumble |
17:15.03 | Katty | That makes me think of twisted. |
17:15.11 | Katty | the dread pirate robberson! |
17:15.16 | Maliuta | ministry of silly walks all over this |
17:15.23 | [TK]D-Fender | Katty: "Roberts" |
17:15.23 | ntbourey | They would be able to login and access a list of contacts |
17:15.35 | Katty | [TK]D-Fender: yes. i know. it was a running gag with twisted. |
17:15.35 | Maliuta | don't make me do the Parrot Sketch |
17:15.42 | [TK]D-Fender | Katty: "the Dread Pirate Robert takes no survivors! HA HA HA" |
17:15.46 | jaytee | back in a few, gots "stuff" to do |
17:15.59 | [TK]D-Fender | </andre> |
17:16.25 | [TK]D-Fender | ntbourey: Ok, this is very trivial stuff you can so far do completely in dialplan. |
17:17.15 | [TK]D-Fender | ntbourey: Go read the BOOK and pay extra attention to the chapter on func_odbc |
17:17.16 | [TK]D-Fender | ~book |
17:17.17 | 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 |
17:17.18 | [TK]D-Fender | ^^^^^^^^^^^^^ |
17:18.00 | telnettech | I hate the company i work for...... |
17:18.14 | Katty | join the club. |
17:18.24 | telnettech | TK all that talking last night about 911 and they shoot me down and tell me to let development work on it |
17:18.32 | Katty | :< |
17:18.50 | Katty | telnettech: our company said they weren't going to sell asterisk systems anymore. |
17:18.50 | Nugget | telnet is eeeeeeevil! |
17:19.00 | Katty | telnettech: because there's "no support" for it. |
17:19.06 | Katty | telnettech: how bout that? |
17:19.14 | Katty | Nugget: shall we hug? |
17:19.34 | telnettech | Katty: what do they mean by no support |
17:19.47 | djMax | anybody know how to setup TBCT on *? |
17:19.48 | telnettech | from the community or within the company |
17:19.50 | [TK]D-Fender | telnettech: If they are afraid of *, and you're "it", who the hell is "development", and can I have some of their crack? |
17:19.57 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
17:20.02 | Katty | telnettech: the real reason they don't want to sell it is because people recognize Samsung and Toshiba. |
17:20.05 | [TK]D-Fender | djMax: "core show application transfer" |
17:20.45 | Katty | telnettech: their excuse is that I'm the only person here who knows asterisk, and there's no "support" i can call if i don't know what to do. regardless of the fact i've offered them at least 10 sources for support. |
17:21.10 | Nugget | huggles Katty |
17:21.20 | coppice | they want more than just the sources :-) |
17:21.24 | [TK]D-Fender | Katty: What are we, chopped liver? |
17:21.34 | Katty | hugs Nugget |
17:21.48 | Katty | [TK]D-Fender: welll... |
17:21.52 | Katty | [TK]D-Fender: heh ;) |
17:22.06 | Katty | [TK]D-Fender: it's just an excuse. |
17:22.14 | telnettech | I am fighting with the support manager and the development mgr....they have sold a product to a customer and told them that they can provide this but havent and the customer refuses to finish paying the contract cuase of it |
17:22.30 | [TK]D-Fender | Katty: Theya re good at that. Coming up with excuses that is. Not good ones... just volume |
17:22.32 | coppice | Katty: free stuff pays no kickbacks |
17:22.43 | telnettech | they want the customer to wait about 4 or 5 months once development has figured out how to do it and then test it |
17:22.46 | Katty | coppice: i'm not going to get into it with them. |
17:22.52 | Katty | coppice: it's like talking to a wall. |
17:23.37 | ntbourey | Thanks for nothing |
17:23.39 | kc8pxy | wierdo: i got monkeys last night. i simply had to specify only the sound name, not it's path. and it worked :) |
17:23.46 | *** part/#asterisk ntbourey (n=ntbourey@c-76-110-3-120.hsd1.fl.comcast.net) |
17:23.54 | [TK]D-Fender | Katty: No... walls reflect sound, its jsut gets lost in their vacuum :p |
17:24.05 | *** join/#asterisk Steak_ (n=steak@85.4.120.35) |
17:24.09 | Steak_ | hello |
17:24.27 | Katty | [TK]D-Fender: that too ;) |
17:24.40 | [TK]D-Fender | "thanks for nothing". Brilliant.. totally. |
17:24.56 | Maliuta | Katty: how big is the firm you work for? |
17:25.14 | telnettech | So i guess all they want are field techs that know how to push buttons and click on drop down menus and cant think for themselves |
17:25.33 | Katty | Maliuta: 30 people, tops. |
17:25.33 | wierdo | kc8pxy, oh, cool :) the obvious |
17:25.39 | Katty | Maliuta: over half of them are sales reps. |
17:25.52 | Maliuta | Katty: I have found there is a size where it will always be crap, and another size where it will always include a brick wall |
17:25.53 | Katty | Maliuta: the company's 'core' business is copy machines. |
17:26.08 | *** join/#asterisk bbryant (n=olpc@68.208.65.50) |
17:26.08 | Steak_ | I have a strange issue (or at least it is strange to me..), I have an asterisk installation with a SPA3102 as PSTN gateway. whenever I receive a phone call from the PSTN, all the internal SIP phone ring correctly (and work perfectly if picked up), but the CLID of the PSTN call is not shown, instead the SPA assigned number is displayed... anybody has an idea? thanks in advance |
17:26.13 | Katty | Maliuta: phone systems is just one of their 'side' projects. |
17:26.25 | Maliuta | Katty: yeah, it's going to be uber crappy then |
17:26.30 | Katty | Maliuta: yep. |
17:26.53 | Maliuta | the size of the sales force is directly proportionate to the size of the crap |
17:26.59 | cutlass | does anyone know how to disable call files? Is it just a matter of making /var/spool/asterisk unwritable, or is there a more elegant way to do this? |
17:27.14 | Katty | Maliuta: usually i get along with people. all sorts of people. i can walk into a party and immediately make friends. |
17:27.24 | Katty | Maliuta: i cannot STAND half the sales reps here. literally, can't even tolerate them. |
17:27.48 | Katty | Maliuta: they open their mouth, and i want to stab them. |
17:29.10 | eppigy | :< |
17:29.16 | eppigy | I am dave in sales |
17:29.17 | Katty | Maliuta: and they wonder why i'm so pissy in the morning! *hee* |
17:30.02 | Katty | eppigy: i think it just has something to do with them being southern missouri hick morons. |
17:30.18 | jaytee | or just because they're in sales |
17:30.30 | Katty | jaytee: doubtful. |
17:30.39 | Katty | jaytee: there are a couple sales reps here i get along with. |
17:31.29 | Katty | i like dave. |
17:31.33 | eppigy | :D |
17:31.48 | telnettech | sorry didnt mean to open a can of worms here |
17:32.00 | eppigy | telnettech: that ship has saled, thanks |
17:32.03 | eppigy | sailed |
17:32.07 | eppigy | lol |
17:32.09 | Katty | we're on to can of pringles now. |
17:32.43 | [TK]D-Fender | cutlass: rm /usr/lib/asterisk/modules/pbx_spool.so |
17:34.35 | cutlass | ok...I wasn't sure if that module was responsible for anything else in addition to call files...is it the same thing to put noload => pbx_spool.so in modules.conf? |
17:34.38 | Maliuta | cutlass: you can unload pbx_spool |
17:35.23 | cutlass | ok |
17:35.31 | [TK]D-Fender | Maliuta : Odds are if they can drop call-files to the FS they can write to *'s configs... best to remove all possibility :p |
17:35.51 | [TK]D-Fender | burns it... WITH FIRE!!!!! |
17:36.13 | cutlass | makes sense... |
17:36.36 | cutlass | thanks to both of you |
17:37.00 | cutlass | :q |
17:37.05 | eppigy | THE CRUCIBLE |
17:37.34 | cutlass | ...sorry...that was intented for vi |
17:37.53 | eppigy | yes i see that |
17:38.45 | [TK]D-Fender | OMMFG GGGGGGGGGGOOOOOOOOOOLLLLLLLLLLLLDDDDDDDDD!!!!!!!!! http://www.collegehumor.com/video:1907543 |
17:38.56 | Pazzo | Can anyone give me a short hint on how to terminate a Call in early media state by Asterisk with some "soft" response (e.g. 480)? |
17:39.39 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-115-7.PHNX.QWEST.NET) |
17:43.34 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
17:43.37 | Pazzo | Hmmm... I discovered that Hangup(18) == 408 |
17:47.01 | Maliuta | hads cutlass a ! for his/her/its :q |
17:49.03 | Pazzo | Hangup(16) doesn't seem to be mapped, I get 602. Hangup(17) gives me 486, as expected |
17:49.11 | cutlass | I like to be alerted of unsaved changes :) |
17:49.18 | Pazzo | No one here able/willing to help me? |
17:50.00 | Katty | I'm having urges to add an S to the beginning of your /nick. |
17:50.26 | *** join/#asterisk Nasra (n=maxshipp@190.166.70.98) |
17:51.23 | eppigy | haha |
17:51.35 | *** join/#asterisk jesselang (n=jesse@h69-21-229-150.mntimn.dedicated.static.tds.net) |
17:53.47 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
17:54.54 | jesselang | Is there a way to use EAGI and channel state (knowing when the call has been answered) simultaneously? I want to have early media, but also know at what point the call is answered. |
17:55.20 | [TK]D-Fender | Katty: Good laugh, watch link! |
17:57.13 | *** join/#asterisk rob327 (i=1001@209.80.7.124) |
17:57.40 | Maliuta | I am so kicking my ISP in the balls in the morning |
17:57.44 | Maliuta | well later in the morning |
17:57.46 | Steak_ | ... can't really understand why the phone is showing the trunk number instead of the incoming caller CID |
17:58.41 | [TK]D-Fender | Steak_: if you set "callerid=" for your sip peer for the PSTN port then that would override anything it passes. Next you ahve to configure the SPA to even pass on the CID. |
17:59.06 | [TK]D-Fender | Steak_: voxilla.com 's forums have plenty of posts on how to configure it accordingly. |
17:59.12 | Steak_ | the PSTN port entry in sip.conf hasn't got the callerid entry, and the SPA is configured to pass the CID |
17:59.41 | [TK]D-Fender | Steak_: You also have to tell it to wait 2 rings or so. |
17:59.48 | Steak_ | the funny thing is that if I look inside of the asterisk call logs, the CID number is displayed perfectly, and asterisk replaces it with the PSTN port name |
17:59.50 | [TK]D-Fender | Steak_: Another gotcha |
17:59.53 | Steak_ | (waiting 3 rings) |
18:00.29 | Katty | [TK]D-Fender: k |
18:00.31 | [TK]D-Fender | Steak_: pastebin a call at verbose 10, SIP debug enabled and NoOp the ${CALLERID(all)} |
18:00.31 | Steak_ | so I am assuming that it is asterisk remapping the incoming call CID with the PSTN port name |
18:00.32 | Katty | watches. |
18:00.46 | Steak_ | ok a sec |
18:00.47 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
18:01.20 | Steak_ | (I have no ${CALLERID(all)} entry in extensions.conf, nor in sip.conf) |
18:01.43 | [TK]D-Fender | Steak_: .. theFUNCTION. noOp it it in your DIALPLAN |
18:02.07 | [TK]D-Fender | ~pb |
18:02.08 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
18:02.11 | [TK]D-Fender | ^^^^^ |
18:02.15 | [TK]D-Fender | Steak_: include your configs |
18:02.18 | Steak_ | k |
18:03.29 | Katty | [TK]D-Fender: cute! |
18:03.51 | [TK]D-Fender | Katty: very well produced and so very real... |
18:05.02 | Katty | [TK]D-Fender: yep. |
18:06.33 | Steak_ | [TK]D-Fender: here the extensions.conf and sip.conf --> http://pastebin.com/d3a60b8df |
18:06.47 | Steak_ | now I'll make the log of a phone call with the output |
18:09.26 | Maliuta | Steak_: ummm you know you left the password in? |
18:09.34 | Maliuta | Steak_: you might want to change that |
18:10.55 | kc8pxy | Steak_: it rings for 60? |
18:11.20 | kc8pxy | Maliuta: i noticed that too. |
18:11.50 | Steak_ | it's all in a private network |
18:12.23 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-64d93ef7e0a84ced) |
18:12.33 | Steak_ | sorry, I got a call from a bakery in the meanwhile that was inquiring about an order of 300 breads :P not made by me :P |
18:12.42 | Steak_ | ok, back again |
18:13.58 | Maliuta | All your breads are belong to us! |
18:14.21 | Katty | not the breads:< |
18:14.21 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
18:14.42 | UQlev | Steak_: that must have been Jesus |
18:15.20 | Maliuta | forces Katty to listen to Bread for 3 hours |
18:15.20 | Maliuta | :) |
18:15.36 | Steak_ | http://pastebin.com/d64f4f42a <-- here the output of a test call from my mobile to the system |
18:15.39 | eppigy | NEIN |
18:15.44 | *** join/#asterisk Joe_CoT (n=joecot@ubuntu/member/pdpc.bronze.joeterranova) |
18:15.51 | Steak_ | problem is that the baker was speaking in german only :O |
18:16.02 | Steak_ | and ich spreche nuer ein bisschen deutsch :P |
18:16.15 | Joe_CoT | hey guys, got a big problem. I updated my kernel and asterisk yesterday, everything seemed fine. Now for some reason, Asterisk won't load sip. Is there anything I should look for? |
18:16.20 | Maliuta | Katty: next on the list ... Chicago, then Air Supply |
18:16.39 | Steak_ | [TK]D-Fender: http://pastebin.com/d64f4f42a <--the call log |
18:17.01 | Ritzerisk | asterisk is like SLOOW cracked out the voice back to me even locally |
18:17.34 | UQlev | Joe_CoT: what OS? |
18:17.38 | Katty | cries over lost bread. |
18:17.44 | Joe_CoT | UQlev, linux. ubuntu |
18:17.44 | Steak_ | sorry katty |
18:17.47 | Ritzerisk | vmware but i had another VMware thats good so im not sure why really |
18:17.53 | *** join/#asterisk The_Lightside (n=Lightsid@41.145.103.193) |
18:18.05 | eppigy | Lets no cry over spilled guineas |
18:18.08 | eppigy | not |
18:19.22 | Joe_CoT | UQlev, wait, wtf, it just loaded |
18:19.28 | [TK]D-Fender | Steak_: add "trustrpid=yes" to your peer entry. |
18:19.47 | Maliuta | Joe_CoT: so sorry you're using a hosed debian system |
18:19.47 | Steak_ | to the pstn trunk ? |
18:19.52 | The_Lightside | hi all, just upgaded to centos 4.7 |
18:19.53 | [TK]D-Fender | steyes |
18:19.59 | [TK]D-Fender | Steak_: Yes |
18:20.01 | The_Lightside | dahdi seems to compile, but wont load |
18:20.19 | eppigy | did you download your new kernel source? |
18:20.23 | kc8pxy | Ritzerisk: vmware for server or client? |
18:20.33 | Maliuta | CentOS 4.7 is an upgrade? I thought it was past 5.0 |
18:20.46 | The_Lightside | afik, the source is there |
18:21.08 | The_Lightside | Maliuta, it is, but some of the legacy stuff is still on 4 |
18:21.27 | Steak_ | [TK]D-Fender: that did not solve the problem |
18:21.34 | Katty | i want to plan a vacation to flordia. |
18:21.38 | Katty | and go to sea world. |
18:21.40 | Katty | it just hit me. |
18:22.08 | Maliuta | Katty: we have a sea world here on the glod coast |
18:22.12 | [TK]D-Fender | Steak_: Reloaded your config? |
18:22.13 | Maliuta | Gold Coast even |
18:22.22 | [TK]D-Fender | Steak_: Also, what # should we be seeing? |
18:22.33 | Steak_ | the number of the incoming call |
18:22.47 | Maliuta | which is? |
18:22.51 | Steak_ | in this case, the 079 etcetcetc |
18:22.57 | Steak_ | 041 is the spa |
18:23.00 | Steak_ | (pstn number) |
18:23.03 | Steak_ | 079 is my mobile |
18:23.08 | [TK]D-Fender | Steak_: Do it again, and do not mask #'s |
18:23.24 | Steak_ | the log you mean? |
18:23.29 | [TK]D-Fender | Steak_: yes, new acll |
18:23.31 | [TK]D-Fender | call |
18:23.36 | Steak_ | ok redoing |
18:23.51 | The_Lightside | how much text can i paste in #? |
18:23.59 | [TK]D-Fender | Steak_: exten => 0417602xxx,1,NoOP() <- and I told you to Noop ther CALLERID |
18:24.25 | Maliuta | The_Lightside: not much |
18:24.31 | Maliuta | ~pb |
18:24.32 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
18:24.40 | The_Lightside | i hope this isnt too much |
18:24.44 | The_Lightside | its like 2 lines |
18:25.06 | The_Lightside | FATAL: Error inserting wct4xxp (/lib/modules/2.6.9-78.0.13.ELsmp/extra/wct4xxp.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
18:25.12 | The_Lightside | lol, not that much |
18:25.36 | Steak_ | [ http://pastebin.com/d37677c86 |
18:25.40 | Steak_ | [TK]D-Fender http://pastebin.com/d37677c86 |
18:25.54 | Steak_ | but I saw something that IMHO needs to be changed on the SPA |
18:26.15 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
18:26.15 | The_Lightside | i get that on dahdi start |
18:26.16 | Steak_ | just look at the first SIP entry in the log |
18:26.19 | Steak_ | From: 0417602486 <sip:0797063898@192.168.25.30>;tag=eb7dce12c13407c4o1 |
18:26.29 | Steak_ | looks like the SPA is masking the real CLID |
18:26.46 | [TK]D-Fender | Steak_: same here |
18:26.51 | Joe_CoT | Maliuta, thanks for actually trying to help with my issue, instead of just making some comment about my Distro |
18:26.58 | [TK]D-Fender | Steak_: From: 0417602486 <sip:0797063898@192.168.25.30>;tag=eb7dce12c13407c4o1 To: <sip:0417602486@192.168.25.30> <- same from/to |
18:27.04 | [TK]D-Fender | Steak_: SPA = misconfigured |
18:27.21 | *** join/#asterisk bpgoldsb (n=bpgoldsb@209.208.6.182) |
18:27.25 | Steak_ | the funny thing is that incoming and outgoing calls work perfectly |
18:27.37 | [TK]D-Fender | Steak_: Sure, CID does not change that |
18:27.48 | bpgoldsb | Is there a replacement for valetparking for Asterisk 1.6? |
18:28.03 | Maliuta | and dmesg says? |
18:28.03 | Maliuta | roll over, roll over. |
18:28.07 | Maliuta | so they all rolled over and <insert_dmesg> fell out. |
18:28.17 | Maliuta | Joe_CoT: that's fine |
18:28.23 | Maliuta | Joe_CoT: anytime |
18:28.43 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
18:28.52 | Steak_ | well nevermind, I'll disconnect everything and start from scratch again, as soon as my right hand will be healed (got injured today while mounting a 4u server on a defective rack) |
18:29.08 | Steak_ | so this is the reason why I am so slow, I can only use my left hand :P |
18:29.22 | Maliuta | The_Lightside: what is in your dmesg? |
18:29.22 | Maliuta | The_Lightside: and did you rebuild the dahdi mods for that particualr kernel? |
18:29.27 | Steak_ | [TK]D-Fender thanks for your help, I'll tackle this thing in the next days |
18:29.29 | Steak_ | bye all |
18:29.36 | [TK]D-Fender | Steak_: You're welcome |
18:30.01 | DavidBer | Good afternoon - I am trying to solve a problem that all my extensions are busy. I checked to see if they had DND set on and do not. They can call out, but any incoming call shows them as busy. |
18:31.08 | [TK]D-Fender | DavidBer: You can't |
18:31.12 | bpgoldsb | I'm trying to upgrade from Ast 1.2 to Ast 1.6. We need to be able to put a call on certain designated spots. In 1.2, we used the ValetParking add/module. Is there a similiar way to do this in 1.6? Via a new module, built-in or not? |
18:31.24 | [TK]D-Fender | DavidBer: "Busy" from the phone is a response, not a "state" |
18:31.42 | The_Lightside | http://pastebin.ca/1391944 |
18:31.51 | DavidBer | Ok - then they are responding busy |
18:32.46 | DavidBer | I had a 1.4.21.2 setup that was acting a little strange with results coming back that the line was busy - so I upgraded to 1.4.24.1 |
18:33.25 | [TK]D-Fender | DavidBer: * doesn't make the other side say "busy"... |
18:34.14 | DavidBer | got sip respons 486 :) |
18:37.53 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
18:38.30 | Maliuta | The_Lightside: are these modules you built yourself? |
18:38.30 | Maliuta | The_Lightside: because it looks like something wasn't linked properly |
18:38.30 | Maliuta | The_Lightside: or there is a lib missing |
18:38.42 | Maliuta | that's it ... I give up. My net connection is being sooooo shitty I can't class it as usable |
18:38.44 | Maliuta | I'm going to try and get more sleep, and then kick the ISP in the balls |
18:39.56 | Katty | wtb summer. |
18:40.37 | Katty | and some palm trees. |
18:40.42 | Katty | two bahama mammas |
18:40.43 | Katty | and a towel. |
18:41.40 | The_Lightside | Maliuta: i did compile them oon the machine, yes |
18:43.35 | *** join/#asterisk oej (n=olle@ns.webway.se) |
18:44.01 | Ritzerisk | you know that lady on asterisk ??/// she sounds like shes on cRACK and its playing really SLOW for some reason but the webmin is not slow at all so im not sure what it is |
18:44.20 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
18:44.42 | *** join/#asterisk WarptwistDK (n=chatzill@0x4dd49295.adsl.cybercity.dk) |
18:44.53 | kc8pxy | Ritzerisk: asterisk on vmware guest? |
18:44.57 | Ritzerisk | is there like a CPU stat command or something i have about 2 gbs of memory to it |
18:45.36 | Ritzerisk | yea on vmware but i had it on another vmware same setup ANd same thing Really odd |
18:46.04 | Ritzerisk | i meant the other Vmware was no problem .... |
18:47.01 | kc8pxy | Ritzerisk: mebbe I've just got too many spare boxen NOT to, but i 'd say toss it on a spare REAL box(like i did), and you should have no proble. in the meantime, try `core show sysinfo` iirc |
18:48.03 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
18:48.57 | Ritzerisk | hmm my other vmware was fine either bad install or what but 2gbs well over .... |
18:51.11 | The_Lightside | or use xen and pass real hardware to the gues |
18:51.18 | The_Lightside | guest |
18:51.40 | Ritzerisk | xen never really used it do i have to have a linux box to do that |
18:52.10 | Ritzerisk | ohhh shoot the only thing i can think of is 64bit windows where the vmware is installed |
18:52.25 | Ritzerisk | i might have to download a 64bit version |
18:55.07 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
18:55.48 | The_Lightside | bleh, winblows.... |
18:58.12 | lesouvage | I'm testing a routine using the cityhall number during out of ffice times. For sme strane reason the channel variables dialtime and answered time don't have a value in the h extension while they have a value when I test the routine on y home number or mobile number. I'm out of options and hoping that one of you can give a hint to get this problem (an h extension that doesn't wrk all the... |
18:58.14 | lesouvage | ...time) solved. Thanks in advange. |
18:58.51 | lesouvage | and excuse for all the typos. |
18:58.52 | eppigy | ALLO |
19:06.30 | *** join/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer) |
19:07.12 | *** join/#asterisk ilowe (n=ilowe@modemcable230.43-82-70.mc.videotron.ca) |
19:07.29 | JerJer | has anyone figured out a dialplan way to know if progress has been made on a call ? Dropping a call file into dialplan - wana kinda wait around until we know progress is being made |
19:08.03 | JerJer | hmm - then again we might not make it to the dialplan until its connected - grrrrrr |
19:09.00 | lesouvage | JerJer: have you checked the status variables? |
19:10.11 | *** join/#asterisk a-s (n=user@92.81.117.113) |
19:10.42 | a-s | I wish to use the echo - canceller from *. Does someone use it? |
19:11.06 | a-s | I did not succeed to start it. |
19:12.56 | JerJer | lesouvage: this box is running biz edition - i don't see anything status |
19:13.00 | JerJer | outside of like AGI |
19:13.02 | JerJer | or AMI |
19:14.32 | a-s | I need to write an echo canceller for a soft written by me. I wish to see one working before... |
19:15.08 | JerJer | dahdi has various echo cans |
19:15.11 | JerJer | not asterisk |
19:15.56 | a-s | JerJer: I looked over dahdi. I cannot start it over a SIP call. |
19:16.09 | UQlev | a-s: prohibit using loudspeakers and that's it |
19:16.27 | a-s | UQlev: I am not writing it for me :) |
19:16.32 | a-s | it's a MUST |
19:16.48 | UQlev | ahh, ok |
19:16.57 | UQlev | keep doing it |
19:17.14 | a-s | UQlev: yes, but where should I start from ? |
19:18.03 | UQlev | a-s: from a client's sound card, it is useless to try to filter it out |
19:18.06 | JerJer | dahdi has nothing to do with sip |
19:18.14 | JerJer | ~book |
19:18.15 | 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 |
19:18.21 | JerJer | ^^^ read it |
19:18.36 | a-s | JerJer: thanks a lot. let's see... |
19:19.07 | *** join/#asterisk ck_28 (i=CK@93.185.235.237) |
19:19.13 | ck_28 | hi ppl |
19:19.35 | ck_28 | How can I stop MOH when status of the dial is ringing and let the user hear the Ring Back Tone from the termination Gateway |
19:21.07 | *** part/#asterisk Joe_CoT (n=joecot@ubuntu/member/pdpc.bronze.joeterranova) |
19:21.46 | ck_28 | Corydon76-dig denon file jtodd leifmadsen putnopvut Qwell russellb Kindly guys i am working on this case for more than 2 weeks can you help me |
19:22.07 | russellb | O.O |
19:22.15 | *** mode/#asterisk [+b %ck_28!*@*] by russellb |
19:22.21 | russellb | Please do not ever do that again. |
19:23.04 | denon | man I hate it when people do that |
19:23.05 | russellb | This is a volunteer help channel. People that are willing to help and have time to help are already here watching. Please do not grab the attention of every op in the channel as your problem is not everyone else's emergency. |
19:23.14 | a-s | JerJer: waw. thanks. seems that the book explains very clearly. |
19:23.25 | *** mode/#asterisk [-b %ck_28!*@*] by russellb |
19:23.39 | Corydon76-dig | If you want paid support, there is a tollfree number available for that. |
19:23.41 | ck_28 | thanks |
19:23.48 | leifmadsen | ~consulting |
19:23.49 | infobot | it has been said that consulting is Having a problem with Asterisk that you need resolved quickly? Hire an Asterisk consultant! You can find several 'for-hire' Asterisk consultants in #asterisk-consultants. |
19:23.55 | leifmadsen | and there :) |
19:24.05 | russellb | Digium also offers support subscriptions. |
19:24.10 | leifmadsen | so many options! |
19:24.18 | ck_28 | its for my univ project |
19:24.35 | russellb | aw, you shouldn't have told me that. That means you should be doing it yourself for sure, or else it's probably cheating! |
19:24.36 | russellb | :-) |
19:24.47 | leifmadsen | ;) |
19:24.56 | Corydon76-dig | Yep, plagiarism |
19:24.57 | ck_28 | ok ser concider it done :) |
19:25.08 | lesouvage | ck_28: Hope you learn from the suggestions. What is your system, plain asterisk? |
19:25.45 | ck_28 | sure asterisk V 1.4.24 |
19:26.39 | lesouvage | Google for dial() cmd asterisk and do some reading. I think you will find your solution. |
19:26.52 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
19:27.19 | ck_28 | lesouvage did you get my quest |
19:27.43 | ck_28 | lesouvage you cant find it at google -2 weeks of searching |
19:28.24 | [TK]D-Fender | ck_28: * does not take audio from the gateway unless the gateway answers *'s call and is passing their progress inband. In short, NO. |
19:29.06 | The_Lightside | [TK]D-Fender, what about early media? |
19:29.23 | [TK]D-Fender | the_Not sure on the fine points for yours... |
19:30.00 | lesouvage | ck_28: Yes, it is about moh when phone is ringing. It seems that you have the m parameter in your dial string. |
19:30.02 | ck_28 | [TK]D-Fender i cant edit chan_sip.c to tellhim that when * recieves 183 or 180 stop early media |
19:31.03 | ck_28 | lesouvage yes _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) |
19:32.16 | [TK]D-Fender | ck_28: Stop using "m", thats why you have MoH |
19:32.44 | *** join/#asterisk Wired_Life (n=Chatzill@mgdb-4db87385.pool.einsundeins.de) |
19:32.44 | lesouvage | ck_28: just remove the m if you like you can replace it for something else. Just google for dial() cmd asterisk and read the info on different paramters. Some say you shouldn't use the r paramter but if you do you will hear a ring. |
19:33.06 | ck_28 | [TK]D-Fender i want to let the caller hear MOH untill it rings |
19:33.21 | [TK]D-Fender | ck_28: what do you mean UNTIL it rings... its already RINGING |
19:33.51 | *** part/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer) |
19:35.32 | ck_28 | [TK]D-Fender some times i have a high pdd so i want to cover the dead air by moh |
19:35.46 | lesouvage | ck_28: more than you already have is not available as an answer to your question. |
19:37.56 | *** join/#asterisk smbrienz (n=guest@host202-204-dynamic.1-87-r.retail.telecomitalia.it) |
19:39.39 | Katty | has... |
19:39.43 | Katty | strawberry |
19:39.46 | Katty | ...limeaide |
19:39.57 | [TK]D-Fender | ck_28: You Early media should be "answered", not "ringing". |
19:40.16 | *** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43) |
19:41.03 | pmhaddad-work | ok, so i have asterisk 1.6.0.9 installed from a tarball, and i just compiled and installed dahdi and dahdi tools from svn, that all works fine, i added the rcbfx module to /etc/dahdi/modules, and restarted dahdi and asterisk |
19:41.09 | ck_28 | [TK]D-Fender i should answer the call and put him on hold and then dial |
19:41.17 | pmhaddad-work | but how can i do like "dahdi show channels" from the asterisk cli? |
19:41.41 | pmhaddad-work | do i have to run dahdi_cfg first? |
19:42.16 | [TK]D-Fender | ck_28: Answer what call? You're the one CALLING |
19:42.53 | [TK]D-Fender | pmhaddad-work: * has to be compiled AFTER DAHDI |
19:43.21 | ck_28 | [TK]D-Fender scenario is like that UserA--->*--->Termination GW ---->User B |
19:43.52 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
19:44.06 | pmhaddad-work | [TK]D-Fender, yep, just noticed that |
19:44.09 | pmhaddad-work | recompiling now |
19:44.25 | [TK]D-Fender | [15:37]<ck_28>what i need is How can I stop MOH when status of the dial is ringing and let the user hear the Ring Back Tone from the termination Gateway |
19:44.44 | [TK]D-Fender | ck_28: As this sys, you DON'T. * can play * INSTEAD of ringing. not BEFore, INSTEAD <- |
19:46.05 | ttl- | does anybody has experience in configuring the Linksys SPA3102 ATA please? |
19:46.38 | [TK]D-Fender | ttl-: www.voxilla.com <- plenty of guides there |
19:47.10 | ttl- | i'm trying to get it to work for 5 days, followed all kinds of guides |
19:47.38 | ttl- | [TK]D-Fender: ok i'll try www.voxilla.com again |
19:48.11 | ttl- | this thing drives me crazy |
19:48.38 | djMax | This line: exten=>9565,1,Transfer(ZAP/1115551212) (number was changed) does that look proper? |
19:48.47 | ck_28 | when I used directrtpsetup=yes I heard the MOH and the ring back tone together when I used directrtpsetup=yes I heard the MOH and the ring back tone together |
19:48.49 | djMax | it just seems to run right over it. |
19:49.16 | [TK]D-Fender | djMax: No, that is certainly not correct. |
19:49.25 | djMax | ok, what'd I blow up? |
19:49.32 | djMax | the Zap invocation? |
19:49.43 | [TK]D-Fender | djMax: does taht look like the proper way to reference a CHANNEL or GROUP of channels to you? |
19:50.03 | djMax | I've been through too many asterisk, freepbx, and sundry configs to remember anymore. |
19:50.10 | djMax | I thought it was. But I go dig now. |
19:50.56 | djMax | used to be easy to see examples in the configs, but with FreePBX is variabled-to-death |
19:52.30 | Katty | deposits strawberry limeaide on jaytee's desk. |
19:53.24 | lesouvage | Can there be any reason that channelvariables doesn't get a value if a channel is set up to certain numbers while with other numbers they got here value as expected. ( ${DIALEDTIME} and ${ANSWEREDTIME} ) |
19:53.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
19:53.49 | ck_28 | [TK]D-Fender by the way why * makes two invites when he want to make a call |
19:54.14 | Katty | what does lesouvage mean? |
19:54.16 | [TK]D-Fender | ck_28: Could be a re-invite through the gateway |
19:54.36 | ck_28 | 1st invite to the GW and when the GW sends ACK * resend an invite to the caller |
19:55.44 | ck_28 | lesouvage even on bye message same behavior ,that because its a b2bua?should be acting like that |
19:56.23 | lesouvage | Katty: I'm testing a routine that uses the ${DIALEDTIME} and ${ANSWEREDTIME} variables in the h extension to get some sort of logging straight. I thought it works like a charm but this evening I found out that the variables doesn't always got values. |
19:56.47 | [TK]D-Fender | lesouvage: "h" often loses access to channel variables |
19:56.50 | Katty | lesouvage: what does that have to do with my question? ^_- |
19:56.57 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
19:57.00 | Katty | lesouvage: oh, i see what you did. |
19:57.06 | Katty | lesouvage: you parsed my question imporperly. |
19:58.06 | Katty | s/imporperly/improperly/ |
19:58.06 | Katty | lesouvage: define lesouvage. |
19:58.06 | ck_28 | [TK]D-Fender you said chan_sip can you pleasw specify in which part |
19:58.07 | ck_28 | or just chan_sip |
19:58.07 | Katty | there are parts ot chan_sip now?! |
19:59.04 | *** join/#asterisk jtexter3 (n=jtexter3@72.242.229.213) |
19:59.46 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
19:59.47 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
20:00.16 | djMax | Ok, so Transfer(DAHDI/g0/12125551212)? |
20:00.51 | [TK]D-Fender | djMax: Looks a lot more valid... assuming you're running DAHDI now and not Zaptel |
20:00.59 | djMax | yes, * 1.6 |
20:01.10 | lesouvage | [TK]D-Fender: I know, but it is kind of strange that must of the times the routine works and sometimes it doesn't because the variables don't have a value. Is there any change that it is because a read() cmd timed out. |
20:01.17 | [TK]D-Fender | djMax: You're samples should be labeled "mixed nuts" :p |
20:01.18 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
20:01.20 | [TK]D-Fender | your* |
20:01.38 | djMax | samples? |
20:01.50 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
20:01.58 | seb- | [TK]D-Fender: are you around? |
20:02.07 | djMax | (i'm sure that was a joke that flew by me) |
20:02.55 | *** part/#asterisk smbrienz (n=guest@host202-204-dynamic.1-87-r.retail.telecomitalia.it) |
20:03.04 | djMax | Hmmm... Received an unknown call with DID set to <x>. I'm assuming that means the transfer failed. |
20:03.06 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
20:04.35 | pmhaddad-work | ok now when i start dahdi up it loads everything but "rcbfx: error"... "no hardware timing source in /proc/dahdi" i looked in syslog, and it shows dahdi loading, but nothing on rcbfx |
20:05.05 | Katty | this makes me want to do shots of vodka. |
20:06.09 | *** part/#asterisk cr4z3d (n=cr4z3d@unaffiliated/cr4z3d) |
20:06.22 | telnettech | Kattty: I am ready too....miserable day |
20:06.33 | Katty | telnettech: let's go have drinks at applebees! |
20:06.47 | Katty | telnettech: we can drown the day with long iceland ice teas. |
20:06.53 | Katty | iceland?! |
20:06.57 | Katty | sighs |
20:07.06 | Katty | i give up. |
20:07.10 | telnettech | im a rim and coke guy |
20:07.22 | pmhaddad-work | is a martini guy |
20:07.23 | telnettech | see i did the same thing.....rum and coke |
20:08.25 | Katty | can't do rum and coke anymore. |
20:08.32 | Katty | bad memories. |
20:08.41 | Katty | got real sick. |
20:08.50 | Katty | but good memories before that! |
20:09.27 | ck_28 | why asterisk sends to invite to complete a method |
20:09.36 | ck_28 | invite or cancle for example,if the gw sends him 486 --* will reply to the Gw with ACK and then send and invite to the sip client |
20:09.42 | djMax | If at priority 1 I have Transfer(|j), and it fails, it should go to priority 102 right? It seems to just quit (I also have n after step 1 to play some monkeys) |
20:10.03 | telnettech | well you drink the long island and i will drink rum and coke |
20:10.03 | Katty | i've never used transfer() before for anything. |
20:10.20 | Katty | oh wait |
20:10.22 | Katty | i have |
20:10.27 | djMax | I assume it's the only way I can get blind transfer |
20:10.39 | Katty | it was a call file, that played a prerecorded message to the number dialed. and then transfered them into a 911 conference call. |
20:11.04 | bpgoldsb | I'm trying to upgrade from Ast 1.2 to Ast 1.6. We need to be able to put a call on certain designated spots. In 1.2, we used the ValetParking add/module. Is there a similiar way to do this in 1.6? Via a new module, built-in or not? |
20:12.08 | Katty | djMax: can you do a transfer to an internal extension? |
20:12.10 | pmhaddad-work | anyone got an idea on my dahdi error? i've googled to no avail |
20:12.24 | Katty | djMax: does this just happen when transfering to dahdi channel? |
20:12.30 | eppigy | 8[] |
20:12.35 | Katty | hi dave. |
20:12.39 | Katty | you should join us at applebees. |
20:12.40 | eppigy | HI |
20:12.43 | djMax | trying. The overall goal is to transfer an inbound DAHDI call to an outbound one. |
20:12.46 | eppigy | i would like to |
20:12.47 | Katty | we are drowning the day away. |
20:12.51 | djMax | i.e. to log the call and shove it off. |
20:12.52 | eppigy | NICE |
20:12.58 | eppigy | i could go for some whiskey |
20:13.03 | Katty | djMax: oh. |
20:13.14 | Katty | djMax: that should be accomplished by a simple Dial |
20:13.33 | Katty | pastebins some stuff |
20:13.35 | djMax | w/o taking 2 chans? |
20:13.40 | Katty | oh. |
20:13.44 | Katty | no. it will take two channels. |
20:14.02 | djMax | yeah, would rather not that. |
20:14.10 | Katty | sounds like..flashhook. |
20:14.14 | Katty | or whatever it's called, i forget. |
20:14.45 | Katty | it's something the samsung 7100 boxes are supposed to be able to do, but the telco has to let you do it |
20:14.53 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
20:15.01 | djMax | XO says they have, but of course I don't believe it |
20:15.25 | Katty | idk how to do it with asterisk |
20:15.43 | Katty | never needed to |
20:15.47 | mweichert | hello. I'm trying to write an extension which dials a number a plays back a sound... |
20:15.51 | mweichert | so far I have the following: |
20:15.52 | mweichert | http://rafb.net/p/wfjL1O14.html |
20:16.16 | mweichert | I can get a sound to play - but it doesn't place the call |
20:16.33 | Katty | mweichert: err that won't work |
20:16.41 | pmhaddad-work | https://support.rhinoequipment.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=92&nav=0,34 <-- lets see if this helps |
20:16.41 | eppigy | STUNTIN LIKE MY DADDY |
20:16.47 | lesouvage | mweichert: just check the M parameter of the dail() cmd. |
20:16.48 | Katty | mweichert: because it won't go to the next step after dial, until the conversation is done. |
20:17.04 | Katty | mweichert: what exactly are you trying to accomplish? |
20:17.26 | Katty | mweichert: play a brief audio snippet at the beginning of a call? |
20:17.47 | mweichert | Katty: when I dial an extension, I want a call to be initiated and when they answer, a message played back to them |
20:18.24 | mweichert | for example, when I dial extension 1111, I want the number 555-555-5555 dialed and a sound played back to them when they pickup |
20:19.01 | mweichert | is this possible? |
20:19.03 | lesouvage | <PROTECTED> |
20:19.10 | Katty | mweichert: well. |
20:19.19 | Katty | mweichert: you mean like... |
20:19.25 | Katty | mweichert: please hold while i connect your call? |
20:19.42 | Katty | mweichert: and then it calls 555-555-5555 |
20:19.54 | jaytee | just got back to my desk. |
20:20.04 | jaytee | Katty, thanks for the strawberry limeade |
20:20.46 | mweichert | Katty, no... calls 555-555-5555, then when they pick up plays "You have just been pinged" |
20:20.51 | *** join/#asterisk mog (n=mog@c-68-62-169-246.hsd1.al.comcast.net) |
20:20.51 | *** mode/#asterisk [+o mog] by ChanServ |
20:21.02 | Katty | mweichert: that would be a call file then. |
20:21.15 | Katty | mweichert: ext 1111 would execute a system() command which copies a call file, and creates a call |
20:21.23 | mweichert | where can I get some documentation that lists what applications such as Dial() are available and what parameters they accept? |
20:21.26 | Katty | mweichert: the call file would dial the number, and then play an audio file to them |
20:21.42 | mweichert | Katty, yes, that's what I'd like to do |
20:21.43 | djMax | hmph, so I'm getting transfer failed (better than unsupported I suppose) |
20:21.54 | Katty | infobot: call file? |
20:21.55 | infobot | ACTION looks around and then screams out file as loudly as possible |
20:22.01 | Katty | file: disregard that. |
20:22.36 | Katty | mweichert: have a look at this: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
20:22.59 | mweichert | ah, I see the Asterisk - The future of Telephony book has "Appendix B. Application Reference" |
20:23.15 | Katty | mweichert: example 6 will probably be most useful. |
20:23.56 | _ShrikE | mweichert: core show application dial |
20:25.07 | Katty | mweichert: per example six... they put in extension 1 (but you could make it 1111), and it copies a call file from a directory into the spool directory which makes asterisk excute it much like an email server would send an email if you dumped it into the spool folder. |
20:25.22 | Katty | mweichert: the call file says to go to extension 10 in context [pa-call-file] |
20:25.43 | Katty | mweichert: in extension 10, you do some like 10,1,Dial(somephonenumber) 10,n,Playback(someaudiofile) |
20:25.57 | Katty | mweichert: then hangup or transfer it or do whatever |
20:27.03 | Katty | mweichert: we use it for 911 here. someone dials 911, andasterisk calls the receiptionist and transfers her into a meetme with the person who called 911 |
20:27.08 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
20:30.24 | *** join/#asterisk propellerhead (n=yogurt2u@host110.190-136-62.telecom.net.ar) |
20:32.32 | Katty | eppigy: what's for dinner. |
20:32.32 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
20:32.38 | eppigy | hrmmmmm |
20:32.38 | Katty | bbkt-trix: ohai |
20:32.42 | eppigy | i do not know |
20:32.45 | Katty | :< |
20:32.48 | eppigy | i know |
20:32.50 | Katty | we are going to starve! /tear |
20:32.57 | Katty | THINK OF THE CHILDREN |
20:33.01 | eppigy | I am thinking a wendys double cheeseburger |
20:33.03 | pmhaddad-work | omg i'm annoyed |
20:33.07 | eppigy | I am on a wnedys kick |
20:33.09 | Katty | pmhaddad-work: whatsammater |
20:33.11 | bbkt-trix | Katty: ohai :-) |
20:33.12 | eppigy | they are so good |
20:33.23 | pmhaddad-work | Katty, trying to get dahdi working with a rhino card |
20:33.27 | pmhaddad-work | with rcbfx |
20:33.32 | mweichert | Katty, thanks - I'm gonna give a couple of things a try |
20:33.35 | Katty | you should talke to jameswf |
20:33.37 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
20:33.37 | Katty | he works for rhino |
20:33.39 | djMax | boy, I have no idea how one goes about debugging the Transfer app. |
20:33.41 | pmhaddad-work | Katty, followed: https://support.rhinoequipment.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=92&nav=0,34 |
20:33.48 | pmhaddad-work | and now i can get the module loaded |
20:33.50 | pmhaddad-work | but |
20:34.13 | pmhaddad-work | the dummy is still the only thing loaded |
20:34.19 | pmhaddad-work | when i do dahdi show channels |
20:34.21 | aiksa[LV] | just a quick question: is bri_cpe signalling available in main dahdi release or is it something I should use bristuff for |
20:34.25 | aiksa[LV] | ? |
20:34.29 | Katty | pmhaddad-work: mmmyeah i'd talk to jameswf |
20:34.31 | pmhaddad-work | and syslog has a LOT of rcbfx errors |
20:34.45 | aiksa[LV] | if tzafrir_laptop were around he would be able to answer this under a sec :) |
20:34.59 | Katty | don't use bri. |
20:35.00 | pmhaddad-work | i don't see him in the channel :| |
20:35.21 | djMax | ok, so Dial(Local/501) works, but Transfer(Local/501) does not. Does that tell anybody anything interesting? |
20:35.26 | Katty | pmhaddad-work: hrmm. |
20:35.29 | Katty | pmhaddad-work: so it seems. |
20:35.32 | tzafrir_laptop | aiksa[LV], bri_cpe is now also part of asterisk >= 1.6.0 |
20:35.34 | Katty | pmhaddad-work: hes usually here tho |
20:35.45 | aiksa[LV] | tzafrir_laptop: but starting with 1.6 |
20:35.46 | aiksa[LV] | ? |
20:35.53 | Katty | djMax: can you transfer to a SIP? |
20:36.05 | aiksa[LV] | and you are online :)) hi friend |
20:36.14 | pmhaddad-work | Katty, could you take a look at the errors in syslog if i pasted them? |
20:36.26 | pmhaddad-work | not sure how hard/easy this is to fix, but its mission critical |
20:36.27 | tzafrir_laptop | this is unrelated to dahdi |
20:36.36 | tzafrir_laptop | it is userspace |
20:36.37 | aiksa[LV] | Katty: were I live and work in most cases bri is the obly option |
20:36.43 | Katty | pmhaddad-work: go visit #asterisk-consultants |
20:36.48 | BlargMaN00 | mweichert: you'd probably be better off having the extension call a script that produces a call file... I think that would be easier for you to implement... |
20:36.50 | aiksa[LV] | tzafrir_laptop: ok. |
20:37.03 | aiksa[LV] | tzafrir_laptop: through libpri? |
20:37.14 | Katty | pmhaddad-work: i don't use rhino equipment, so i'd be useless for you. |
20:37.19 | Katty | pmhaddad-work: someone in there might know. |
20:37.25 | Katty | pmhaddad-work: especially if it's urgent |
20:37.28 | pmhaddad-work | ok |
20:37.33 | pmhaddad-work | thanks Katty |
20:37.34 | djMax | No, Dial(SIP/2001) works, Transfer(SIP/2001) does not. |
20:37.38 | pmhaddad-work | i'll drink a beer for you tonight |
20:37.49 | Katty | djMax: what if you make an auto attendant. |
20:37.55 | Katty | djMax: and then it answers, and then transfers |
20:38.10 | mweichert | BlargMaN00, call a script... what kind of script? AGI? |
20:38.15 | djMax | you mean just answer the call first? |
20:38.28 | Katty | yes. like exten => 444,1,goto(autoattendant,s,1) |
20:38.39 | Katty | and then [autoattendant] s,1,Answer s,2,Transfer(SIP/4001) |
20:38.51 | Katty | <PROTECTED> |
20:39.04 | lesouvage | mweichert: if you give me a moment I can pass something that i working. |
20:39.21 | mweichert | lesouvage, wow, I'd really appreciate that - thanks! |
20:39.26 | tzafrir_laptop | aiksa[LV], this requires libpri >= 1.4.4 |
20:39.46 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
20:40.02 | BlargMaN00 | mwichert: no, just a bash script that will echo everything you need into a tmp call file, and then --> move <-- it to the /var/spool/asterisk/outgoing dir |
20:40.29 | aiksa[LV] | tzafrir_laptop: ok, many thanks. Does that mean that with libpri > 1.4.4 I should be able to compile something rather fresh from 1.4 branch and still get bri signalling for chan_dahdi? |
20:40.50 | djMax | nope, answer doesn't seem to have any affect |
20:40.58 | djMax | effect? can never freakin' get that right |
20:40.58 | tzafrir_laptop | aiksa[LV], no. for 1.4 you still need bristuff |
20:41.05 | Katty | djMax: pastebin me the cli |
20:41.10 | aiksa[LV] | tzafrir_laptop: ok. that explains it. |
20:41.23 | aiksa[LV] | many thanks |
20:41.33 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
20:43.22 | djMax | http://pastebin.com/m413308c3 |
20:45.23 | Katty | what's 1 + 101 |
20:45.38 | djMax | 102 |
20:45.46 | Katty | where is your 102? |
20:45.57 | djMax | don't have one, shouldn't it be 2+102? |
20:46.00 | djMax | Transfer is step 2 |
20:46.07 | Katty | label it as 2 |
20:46.09 | lesouvage | mweichert: see http://www.pastebin.be/17827 |
20:46.35 | djMax | same diff |
20:46.35 | lesouvage | Maybe not the niciest way to do it but it is working without any problems. |
20:46.38 | Katty | djMax: forget the transfer status, just play weasels |
20:46.45 | Katty | and take out the other ns |
20:46.45 | djMax | it doesn't even go there anyhow, just prints failure. |
20:46.46 | *** join/#asterisk profXavier (n=jezus@unaffiliated/neverblue) |
20:46.52 | profXavier | where can I download 1.4.4 ? |
20:47.02 | Katty | recreates aa on her box. |
20:47.19 | djMax | it falls straight through the Transfer line basically. I assume it'd just play weasels. I try now. |
20:47.49 | mweichert | lesouvage, why is 'moving' the file so important? |
20:48.02 | mweichert | why not just write to the outgoing/ directory? |
20:48.10 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) |
20:48.51 | lesouvage | mweichert: the /var/spool/asterisk/outgoing is checked so agrassively that if you copy the file there i a change that the file is processed before completion with unpredictable results. |
20:48.51 | Octothorpe | I know this is #asterisk, not #freepbx... but, that said, FreePBX does seem to be a fairly popular frontend to *. Hopefully someone would be able to help. If I wanted to move the FreePBX content from /var/www/html to, say, /var/www/html/freepbx, I know I have to edit some FreePBX config file like amportal.conf (for example) or somesuch, (1) so that it doesn't break completely and (2) so that future updates don't go sideways. |
20:49.30 | djMax | ok, so doing that from an outbound line to a sip target just quits after Transfer(). Doesn't go to step 3 or 103 |
20:49.45 | djMax | spawn extension exited non-zero |
20:50.03 | Katty | djMax: okay |
20:50.07 | Katty | djMax: here's a working example |
20:50.09 | Katty | pastebins. |
20:50.13 | lesouvage | mweichert: one thig that can happen is that the callfile isn't removed after setting up the call and the call is set up time after time. |
20:51.34 | lesouvage | mweichert: This once cost me my complete sms credit saldo because the callfile keeps on sending sms messages to my phone until I run out of credits while I couldn't do anything. |
20:52.02 | jaytee | quittin time, be back later |
20:52.55 | lesouvage | mweichert: make sure that you have a proper linefeet/return on the last line of the call file because without strange things might happen. |
20:53.02 | djMax | (did I miss the pastebin?) |
20:53.39 | mweichert | lesouvage, heh, that's quite the problem to have occur! |
20:56.22 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
20:56.44 | djMax | either this place just went crazy quiet or I've died. |
20:57.31 | lesouvage | mweichert: does the pastebin make sense to you? |
20:57.37 | mweichert | lesouvage, I don't think the call file is being loaded. |
20:57.44 | mweichert | there is nothing in the log |
20:58.26 | mweichert | what's the ${PAD} variable? |
20:59.40 | djMax | Katty are you messing with me or did you actually pastebin? |
20:59.43 | ttl- | i think my spa3102 is broken |
20:59.46 | ttl- | dunno |
20:59.57 | lesouvage | mweichert: You can start without th mv line and see if the call file is generated. The call file points to a context/extension/priority that has to exist. The ${PAD} variable holds the complete file name wit the directory included. |
21:00.15 | mweichert | and when I want to dial a number using one of my ITSP trunks, am I right to use the following as channel: SIP/name-of-trunk/15551234567 ? |
21:00.36 | ttl- | when pstn line rings it does not send anything to asterisk, even monitoring with wireshark now |
21:01.14 | Katty | djMax: sorry got a call. |
21:01.17 | Katty | djMax: http://pastebin.com/m1524dc7e |
21:01.28 | djMax | thx, thought it was a late APril fools joke |
21:01.29 | Katty | djMax: transfer() doesn't support dahdi/zap transfers |
21:01.38 | ttl- | i don't know what to try anymore |
21:01.42 | lesouvage | mweichert: it is working for me, but you have to adjust it to your setting with a proper local channel or trunk (sip/iax2/dahdi etc.) and pointing to a context/extension/priority that is available. |
21:01.54 | djMax | But there was all this discussion of * support TBCT... how else would that work? |
21:02.22 | mweichert | lesouvage, yes, I know I have the context/extension/priority correct - I'm just not sure about the channel |
21:02.38 | ttl- | thinking of throwing the damn thing into the bin... |
21:03.03 | BlargMaN00 | mwichert: yes, you would use SIP/name-trunk/phone# |
21:03.22 | ttl- | any ATA devices which are good and compatible with asterisk 1.6 ? |
21:03.47 | Katty | djMax: probably uses a different command() |
21:04.42 | rob0 | Throw it here ttl- I've been called a garbage disposal before. |
21:04.52 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
21:05.24 | mweichert | BlargMaN00, do I have to include the pound symbol # at the end? |
21:05.52 | Katty | djMax: maybe you're thinking of ChannelRedirect? |
21:05.55 | BlargMaN00 | mweichert: no... unless your dialplan requires it, then yes |
21:05.57 | ttl- | rob0: It's just one week old, but it's driving me crazy! |
21:06.19 | djMax | I'll check. I see this too: http://wiki.sangoma.com/Asterisk-FAQ#TBCT |
21:06.23 | rob0 | Are you new at this, or have you successfully done it before? |
21:06.25 | ttl- | rob0: There is even no decent documentation |
21:06.38 | djMax | zapata.conf is now dahdi.conf? |
21:06.42 | rob0 | I found a pretty good PDF for my SPA2000. |
21:06.44 | BlargMaN00 | ttl-: such a young ATA... maybe it just hasn't learned yet... 8)~ |
21:07.07 | rob0 | and my SPA 2000's work fine |
21:07.07 | Katty | djMax: yeah i think that's when you use ## |
21:07.13 | Katty | djMax: if it's setup in features.conf |
21:07.15 | Katty | djMax: could be wrong tho |
21:07.43 | rob0 | (under * 1.4 and 1.6, the version is not a factor) |
21:07.46 | Katty | djMax: sangoma people are readily accessible for questions like that |
21:07.58 | djMax | (I don't even know what Sangoma is) |
21:08.03 | ttl- | BlargMaN00: yeah, lol, dunno what i'm doing wrong anymore, maybe it's broken, don't get it anymore... |
21:08.09 | Katty | umm |
21:08.11 | Katty | sangoma is a brand |
21:08.13 | Katty | of cards. |
21:08.22 | Katty | like digium is another brand. |
21:08.25 | rob327 | i just installed the asterisk-gui with svn and i'm getting stuck in some sort of loop while its loading 'creating a config file to store GUI preferences'...right now i am connecting through a firewall however, forwarded a port to the host machine's port 8088 |
21:08.26 | Katty | and rhino. |
21:08.26 | djMax | I have a digium card, wonder if perhaps it doesn't support it |
21:08.28 | rob0 | oh I thought it was a type of cancer. |
21:08.35 | Katty | djMax: you could ask Qwell |
21:08.41 | Katty | djMax: he would probably know |
21:08.43 | rob327 | anyone know if that would make a difference? |
21:08.51 | rob327 | or is this a common issue? |
21:08.53 | rob0 | Lung sangoma ... brain sangoma ... |
21:08.57 | mweichert | here is what I have so far - I feel I'm close: |
21:08.57 | mweichert | http://pastie.org/446582 |
21:08.58 | Katty | rob0: we don't do gui's here. |
21:09.00 | Katty | oh |
21:09.06 | Katty | rob327: we don't do gui's here. they're just problematic. |
21:09.22 | mweichert | lesouvage, can you have a look here please: http://pastie.org/446582 |
21:09.24 | Katty | rob327: i would suggest contacting whoever developed it for support. |
21:09.31 | rob327 | Katty: thanks |
21:09.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
21:09.40 | BlargMaN00 | ttl-: are you using it to connect an analog phone, or analog phone line?? |
21:09.47 | Katty | hi anthony |
21:10.30 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
21:10.47 | ttl- | BlargMaN00: both but i'm now trying to get the pstn work |
21:10.53 | rob327 | i didn't want to go with a gui myself but i'm getting pressure from my boss :\ |
21:11.06 | rob327 | it does seem to be more problematic than it should be |
21:11.34 | djMax | Qwell, out there? |
21:11.38 | ttl- | BlargMaN00: but when the analog line rings i can't get the spa3102 to send it to asterisk, it just does nothing, monitored with wireshark |
21:11.55 | rob0 | Doc, I have a large sangoma growing on my middle digium! What can I do? |
21:12.14 | *** join/#asterisk jblack (n=jblack@pool-71-181-243-204.sctnpa.east.verizon.net) |
21:12.15 | Katty | your doc is a phoney |
21:12.23 | BlargMaN00 | ttl-: you might have to register the fxo side as a seperate trunk in sip.conf... not too sure tho |
21:12.28 | rob0 | ba-da BOOM ching |
21:12.40 | BlargMaN00 | ttl-: does it allow for that?? |
21:13.38 | ttl- | BlargMaN00: don't think so, but the device is not sending anything to anywhere, i looked with wireshark |
21:14.11 | BlargMaN00 | ttl-: found this article --> http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html |
21:14.26 | BlargMaN00 | ttl-: i have never worked with one personally, so see if that helps you out... |
21:14.54 | ttl- | BlargMaN00: followed that one without any success |
21:15.34 | ttl- | BlargMaN00: Thanks anyway :) |
21:15.56 | BlargMaN00 | ttl-: oh... np... i'll keep looking... |
21:17.06 | BlargMaN00 | ttl-: I think this is one of those times where I would have to see it to fix it... 8/ |
21:17.50 | rob0 | Did ttl- not download the PDF docs from linksys/sipura? |
21:18.07 | rob0 | Is the device getting an IP address? |
21:18.16 | rob0 | Can you access its web interface? |
21:18.28 | djMax | ok, so using "##", does it make sense that only 1 dahdi channel would be consumed by a transfer? |
21:18.38 | djMax | I would've assumed it was 2 or 0 |
21:18.58 | ttl- | BlargMaN00: Thanks |
21:19.11 | mweichert | how do I use the originate command in CLI to place a call from 1102 to 1101? |
21:19.18 | ttl- | rob0: yes it has a static i ip |
21:19.34 | ttl- | rob0: otherwise i would not be able to configure it |
21:19.51 | ttl- | rob0: it has like hundreds of options |
21:19.56 | ttl- | lol |
21:20.31 | rob0 | And the defaults are fine for most uses, at least for getting it up and running. |
21:21.20 | ttl- | rob0: Which PDF docs? |
21:21.35 | ttl- | rob0: There are no pdf docs for that device |
21:21.42 | lesouvage | mweichert: console dial extension@context |
21:22.05 | rob0 | I don't have your device, but I did find them for my device, which is probably older than yours. |
21:23.20 | mweichert | lesouvage, "No such command console dial 1111@ext-local" |
21:24.02 | BlargMaN00 | mweichert: the problem is with your dialplan, is that you are trying to go to priority 1 when you only have n priorities... You must have a Priority 1 before anything else... |
21:24.44 | BlargMaN00 | mweichert: change exten => 1111,n,Answer() to exten => 1111,1,Answer() |
21:25.31 | mweichert | BlargMaN00, makes no difference... that's what I had originally, but then I noticed examples had the .call file take action on priority 1 |
21:28.16 | lesouvage | mweichert: it is working for me with console dial 505@inbound |
21:29.00 | mweichert | lesouvage, you're entering that command in Asterisk CLI ? |
21:29.06 | lesouvage | yes |
21:30.03 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
21:30.10 | BlargMaN00 | mweichert: you are creating the call file in another directory, and then moving it into /var/spool/asterisk/outgoing?? |
21:30.12 | *** join/#asterisk SparFux (n=raoul@e182025245.adsl.alicedsl.de) |
21:30.55 | SparFux | Hi! I have an ATA and an analogue phone connected to it. Astoundingly the time of the phone is automatically set when a call arrives, but it is one hour early. What could be the cause of that? |
21:31.25 | mweichert | BlargMaN00, yes |
21:31.47 | BlargMaN00 | mweichert: are you running asterisk as root or as another user?? |
21:32.01 | eppigy | starving |
21:32.11 | mweichert | BlargMaN00, asterisk |
21:32.12 | Katty | :< |
21:32.27 | eppigy | i gotta stay till this att dude gets here |
21:32.37 | eppigy | cause we had another internetr circuit installed today |
21:32.51 | eppigy | and the dude must have knocked power out to one of my smart jacks |
21:32.51 | Katty | :< |
21:32.52 | eppigy | :[ |
21:32.55 | BlargMaN00 | mweichert: are the files disappearing from /var/spool/asterisk/outgoing?? or are they just sitting there?? |
21:33.02 | Katty | my feets hurt. |
21:33.03 | eppigy | his children will pay |
21:33.19 | mweichert | BlargMaN00, they are just sitting there. And I wouldn't want them to go away. |
21:33.21 | Katty | shouldn't have worn heels today. |
21:33.37 | eppigy | me2 |
21:33.41 | djMax | ok, so lemme try a different version of this boondongle: who could/would I pay to solve this TBCT problem? |
21:33.45 | Katty | they're such a pain, aren't they |
21:33.57 | eppigy | yesh |
21:34.00 | BlargMaN00 | mweichert: if they are working correctly, the will disappear as soon as they go in there... that means that * read and processed them, and is done with them... |
21:34.02 | eppigy | its like man |
21:34.07 | eppigy | being sexy comes with such a price |
21:34.15 | mweichert | BlargMaN00, hmm... |
21:34.18 | BlargMaN00 | mweichert: if they are not disappearing, that means * can't read them at all... |
21:34.26 | djMax | (separately, if ## works, what command simulates ##?) |
21:34.35 | mweichert | BlargMaN00, I'll try restarting Asterisk |
21:34.45 | Katty | eppigy: wonderboy. |
21:34.51 | BlargMaN00 | mweichert: do a 'ls -l /var/spool/asterisk/outgoing' and pastebin for me... |
21:35.18 | eppigy | thanks! |
21:35.44 | Katty | whatfordinnerwhatfordinnerhrmmm. |
21:35.59 | eppigy | I will eat ATT guys children |
21:36.08 | eppigy | that is the only way to solve both problems |
21:36.12 | Katty | eppigy: crazy eddie. |
21:36.14 | *** part/#asterisk SparFux (n=raoul@e182025245.adsl.alicedsl.de) |
21:36.19 | mweichert | http://pastie.org/446613 |
21:36.28 | eppigy | THE SINS OF THE FATHER |
21:36.30 | mweichert | BlargMaN00, btw - thanks for your help with this. I appreciate it |
21:36.57 | lesouvage | I found out that if othing is touched, no menu option, no hangup on the called side just a time out then the channel variables dialedtime and answered time aren't set and no info is available for counting the length of the call etc. Does this make any sense? |
21:37.23 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
21:37.39 | BlargMaN00 | mweichert: try 'chmod asterisk:asterisk /var/spool/asterisk/outgoing/1111.call' and see if it disappears... |
21:39.57 | mweichert | nope, still there. Do I have to enable the outgoing directory somehow? |
21:40.03 | mweichert | (even restarted) |
21:40.26 | eppigy | WENDYS |
21:40.46 | Katty | eppigy: http://www.junkfoodnews.net/baconator.JPG |
21:41.01 | ttl- | goodnight |
21:41.08 | eppigy | way too much bacon :[ |
21:41.16 | eppigy | I am not a fan of bacon on burgers |
21:41.27 | eppigy | I like bacon with eggs 8[] |
21:41.56 | BlargMaN00 | mweichert: can you pastebin the CLI output of what happens when you dial the extension that spawns all of this?? |
21:42.06 | Katty | eppigy: http://d1.biggestmenu.com/00/00/24/5a0321911779c1ee_m.jpg |
21:43.54 | mweichert | BlargMaN00, aha, in the log file: Unable to request channel SIP/link2voip/11234567890 |
21:43.57 | eppigy | DUDE |
21:44.05 | eppigy | that looks really grood |
21:44.11 | djMax | is switchtype=national at all similar to 5ess? i.e. blind transfer only works w/5ess seemingly, not sure how screwed I am |
21:44.27 | eppigy | well |
21:44.30 | BlargMaN00 | mweichert: ok... so now you just need to put the real info in there, and it should work... |
21:44.32 | mweichert | BlargMaN00, the file is gone now |
21:44.49 | eppigy | i have found that in most cases 5ess and national are interchangeable |
21:44.53 | Katty | eppigy: http://dixiedining.files.wordpress.com/2008/09/popeye.jpg |
21:45.09 | djMax | thx, exactly what I was wondering. |
21:45.10 | BlargMaN00 | mweichert: also, make sure you change back to 'exten => 1111,1,Answer()' |
21:45.15 | eppigy | SNAP |
21:45.15 | mweichert | BlargMaN00, yes, I put the real info in - just didn't post that number. link2voip is the name of my outgoing route defined in FreePBX |
21:45.27 | eppigy | ARE YOU TRYING TO DRIVE ME IN TO A BEV-RAGE |
21:45.43 | BlargMaN00 | mweichert: oh, ok... |
21:45.57 | mweichert | BlargMaN00, is that not correct? |
21:46.21 | djMax | It's strange because the docs seem to say blind transfer is "automatic" with 5ess, which I don't see why that'd be a good thing |
21:46.55 | Katty | eppigy: http://www.mrpizzaiolo.com/wp-content/uploads/2008/05/cc_pizzabeer.jpg |
21:47.15 | BlargMaN00 | mweichert: you are going to have to show me what your CLI output looks like for me to be able to finish helping you... with out that, i won't be able to see where the issue is now... |
21:47.33 | mweichert | BlargMaN00, okay, thanks. What should the priority be in the .call file? |
21:48.29 | Katty | eppigy: http://www.atoasttothese.com/wp-content/themes/roundbox/roundbox/images/posts/grilled-cheese.jpg |
21:48.43 | BlargMaN00 | mweichert: the call file can stay like it is, you just need to change the first line in your context to have a priority 1 instead of priority n |
21:49.51 | mweichert | waiting for .call file to disappear |
21:51.26 | Katty | eppigy: http://www.littleshamrocks.com/image-files/baked_pineapple_ham.jpg |
21:52.02 | mweichert | the .call file disappeared... when I dial extension 1111, this is what is displayed in Asterisk CLI: |
21:52.02 | mweichert | http://pastie.org/446641 |
21:52.33 | Katty | eppigy: ;) |
21:52.51 | BlargMaN00 | mweichert: then according to the CLI, it should be working correctly... |
21:53.03 | eppigy | 8[] |
21:53.06 | eppigy | D: |
21:53.28 | mweichert | BlargMaN00, the call is initiated as the .call file specified :( |
21:53.35 | eppigy | i do not know how to spell crimanitley |
21:53.36 | mweichert | BlargMaN00, no call is initiated as the .call file specified :( |
21:53.43 | eppigy | crimanittley |
21:54.07 | *** join/#asterisk ingenius (n=alektro@host216.201-253-178.telecom.net.ar) |
21:54.35 | BlargMaN00 | mweichert: are you testing this with a cell phone, or a phone that is also attached to the PBX?? |
21:54.56 | mweichert | BlargMaN00, a phone that is also attached to the PBX |
21:55.44 | *** join/#asterisk juanIMP (n=juan@200.71.41.22) |
21:55.50 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
21:55.54 | BlargMaN00 | mweichert: then this is more than likely why it is not working... try using 'Channel: Local/1234' instead... (replacing 1234 with a valid phone extension) |
21:56.42 | mweichert | BlargMaN00, sorry, the number I'm wanting the .call file to dial is a land line using ITSP trunks... |
21:57.03 | mweichert | but to initiate extension 1111, I'm dialing it from a PBX-attached phone |
21:57.47 | BlargMaN00 | mweichert: oh... so you aren't using the call file??? |
21:58.54 | mweichert | BlargMaN00, I *assume* I'm using it. I thought that if I dial extension 1111, that the .call file will be read at the priority specified in the file, and dial the channel specified in the file. Is that incorrect? |
21:59.20 | BlargMaN00 | sighs... |
21:59.30 | BlargMaN00 | mweichert: i understand where the disconnect is now... |
21:59.58 | mweichert | feels like an idiot |
22:00.19 | BlargMaN00 | mweichert: what happens, is that you want to dial extension 1234 so that it calls 1234567890 and plays the message from extension 1111 <-- is this correct?? |
22:01.03 | mweichert | when I dial extension 1111, I want 123456789 to be dialed and a message played to them |
22:01.33 | BlargMaN00 | mweichert: ok... now i am understanding completely.... we'll get you fixed up now... |
22:01.47 | mweichert | sorry |
22:02.14 | BlargMaN00 | mweichert: no worries... i wasn't seeing the big picture, but now i do, and now I can make sure that you get the help you need... 8)~ |
22:02.22 | mweichert | :) |
22:02.54 | BlargMaN00 | mweichert: can you pastebin your dialplan, so I can see how you are generating the call file??? |
22:03.20 | mweichert | BlargMaN00, I'm manually creating the .call file and then manually moving it |
22:03.58 | BlargMaN00 | mweichert: how are you going to create it when you actually put this into a production environment?? we might as well go ahead and iron that part out too |
22:05.35 | mweichert | BlargMaN00, does it have to be created for every time extension 1111 is dialed? |
22:05.58 | mweichert | btw, this is the contents of my dialplan and such (though I changed the priority as you said) |
22:05.58 | mweichert | http://pastie.org/pastes/446582 |
22:06.09 | BlargMaN00 | mweichert: yes... each call file represents one phone call attempt... it can not make multiple calls... |
22:06.11 | mweichert | I actually have exten => 1111,1,Answer() |
22:06.23 | mweichert | BlargMaN00, ah ok - I didn't realize that. |
22:06.47 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
22:07.03 | BlargMaN00 | mweichert: depending on how clean you want your dialplan, you can use a script, or create it inside your dialplan... i prefer script... but that's just me... |
22:07.18 | knarfly | can I just copy my *-1.4.24 conf files over and make them work on *-1.6.0.9? |
22:07.20 | mweichert | BlargMaN00, script sounds good |
22:07.42 | BlargMaN00 | mweichert: ok... gimme a few minutes to whip something up... |
22:08.09 | mweichert | BlargMaN00, thanks so much! |
22:15.18 | *** join/#asterisk theodred (n=nohost@205.207.102.154) |
22:15.53 | BlargMaN00 | mweichert: i am going to leave and then come back as I have to get on the bus to go home, but I will get back on here as soon as i get on the bus downstairs... shouldn't be more than 10 minutes... |
22:16.23 | BlargMaN00 | mweichert: i almost have everything finished and ready to pastebin... |
22:17.09 | mweichert | BlargMaN00, thanks man |
22:17.31 | BlargMaN00 | mweichert: no worries... |
22:22.34 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
22:25.22 | *** join/#asterisk BlargMaN00-lap (i=BlargMaN@212.sub-70-218-70.myvzw.com) |
22:25.38 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
22:25.38 | *** mode/#asterisk [+o russellb_] by ChanServ |
22:25.39 | BlargMaN00-lap | mweichert: alright... back to work... 8)~ |
22:26.52 | knarfly | I'm new to * on linux...where does linux store the moh files? |
22:27.30 | mweichert | BlargMaN00, that WAS fast |
22:28.10 | BlargMaN00-lap | mweichert: good timing... the bus was waiting on me... |
22:32.54 | eppigy | knarfly: /var/lib/asterisk/moh/ |
22:32.56 | Chainsaw | knarfly: In /var/lib/asterisk by default. moh or mohmp3. |
22:33.08 | eppigy | you can search with "locate" of find |
22:33.17 | eppigy | *or |
22:35.20 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
22:35.23 | mweichert | BlargMaN00, btw, this was my attempt when you were on the bus: |
22:35.23 | mweichert | http://pastie.org/446677 |
22:37.45 | BlargMaN00-lap | mweichert: http://pastebin.com/m7b8cd460 <-- try this and let me know what happens... |
22:37.57 | mweichert | sorry, that was a previous version: http://pastie.org/446679 |
22:38.03 | mweichert | ok -I'll check it out now! |
22:38.49 | seb- | anyone mind testing my * server? i need someone to call and see if they hear "hello world" 5x |
22:41.21 | Chainsaw | seb-: Address? |
22:41.57 | Chainsaw | Hm. How do I dial alphanumerically on a Siemens C485 IP. |
22:42.09 | seb- | Chainsaw: just sent you a msg |
22:42.50 | seb- | Chainsaw: thanks btw |
22:45.06 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
22:45.09 | war9407 | <PROTECTED> |
22:45.09 | war9407 | [Apr 14 18:44:27] WARNING[2643]: file.c:635 ast_openstream_full: File dir-pls-enter.gsm does not exist in any format |
22:45.09 | war9407 | [Apr 14 18:44:27] WARNING[2643]: file.c:936 ast_streamfile: Unable to open dir-pls-enter.gsm (format 0x4 (ulaw)): No such file or directory |
22:45.14 | war9407 | how do I tell where it is trying to play the file? |
22:45.42 | mweichert | BlargMaN00, shit, I'm getting kicked out of the office now - so I'll have to work on this when I get home. Thanks for your help BlargMaN00 |
22:45.45 | seb- | war9407: search for hello-world.gsm and see if you can play that |
22:45.47 | BlargMaN00-lap | war9407: it should be lcated in /var/lib/asterisk/sounds/en/ |
22:46.07 | war9407 | I used --prefix when installing asterisk (latest version) |
22:46.22 | BlargMaN00-lap | mweichert: no worries... i'll be on later tonight, so see if you can catch me then... |
22:46.36 | mweichert | BlargMaN00, thanks a lot |
22:46.40 | war9407 | asterisk-1.6.0.9-x86_64/var/lib/asterisk/sounds/en/hello-world.gsm |
22:46.40 | war9407 | yep its there |
22:46.48 | BlargMaN00-lap | mweichert: if not, i'm always in here, and willing to help... |
22:46.48 | war9407 | how to tell asterisk where the sounds are? |
22:47.06 | seb- | war9407: first see if you can play hello-world |
22:47.18 | war9407 | seb-: ok, |
22:47.31 | BlargMaN00-lap | war9407: just put the file name without the extension (i.e. hello-world) and it automatically looks there |
22:48.04 | mweichert | BlargMaN00, how does asterisk know that the .call file is for extension 1111? |
22:48.32 | BlargMaN00-lap | mweichert: that's what the Extension: 1111 tells * |
22:48.39 | war9407 | kick ass! |
22:48.41 | war9407 | hello world worked! |
22:48.42 | seb- | war9407: what BlargMaN00-lap said |
22:48.53 | mweichert | BlargMaN00, but you have Extension: play |
22:48.54 | seb- | war9407: great! now just put your file in there! |
22:49.11 | *** join/#asterisk _BBV_ (n=buklov@213.138.71.254) |
22:49.26 | BlargMaN00-lap | mweichert: oh yeah... i changed it to that, because they have to be seperate... |
22:49.29 | war9407 | silly question, how do I dial into the pbx from my analog phone? |
22:49.51 | mweichert | BlargMaN00, hmm, but then I don't see how 1111 is linked to the .call file? |
22:50.11 | BlargMaN00-lap | mweichert: 1111 is what you are dialing, and 'play' is where the call file sticks the channel when your SIP truck gets answered... |
22:50.15 | *** join/#asterisk dverzolla (n=dverzoll@proxynet.fcl.com.br) |
22:50.19 | war9407 | [Apr 14 18:50:13] NOTICE[2778] chan_sip.c: Call from 'line1' to extension '*11' rejected because extension not found. |
22:50.19 | war9407 | ah |
22:50.22 | BlargMaN00-lap | s/truck/trunk |
22:50.40 | mweichert | BlargMaN00, so what if I had two .call files and I only wanted one to occur when extension 1111 was dialed? |
22:51.22 | BlargMaN00-lap | mweichert: you will only have the .call files when you dial 1111... they will not show up until you dial 1111 |
22:51.37 | mweichert | ah, right. |
22:52.14 | BlargMaN00-lap | mweichert: if you wanted another one to do something different, then you would create another script, or modify mine to use arguments... |
22:52.28 | BlargMaN00-lap | mweichert: if you need that, let me know, and I will modify it for you... |
22:52.30 | dverzolla | Has anyone running Asterisk in Solaris+SUN? |
22:53.30 | war9407 | the extensions can only be two digits? when I try to dial the extension on an analog phone, it cuts it off after (*)11 |
22:54.38 | war9407 | instead of hello world, how do I get it to get me into a call tree, this is my next step :) |
22:54.55 | BlargMaN00-lap | war9407: what exactly are you trying to accomplish?? |
22:55.14 | war9407 | BlargMaN00: first? make asterisk a nice answering machine |
22:55.19 | war9407 | BlargMaN00: I have it working w/ hello world atm |
22:55.29 | war9407 | BlargMaN00: I would like it to say, dial 1234 for bob and 1235 for sam |
22:55.34 | war9407 | and then leave a voice message |
22:56.04 | BlargMaN00-lap | war9407: ok... so for now, you just want it to be a voicemail system?? |
22:56.15 | war9407 | yes |
22:56.23 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
22:56.26 | war9407 | I have 1.6.0.9 installed |
22:56.29 | war9407 | and asterisk gui 2.0 (svn) |
22:57.10 | BlargMaN00-lap | war9407: i don't use the gui, so i wouldn't be much help there, but i can help you get your dialplan written out to do what you want... |
22:57.17 | war9407 | that'll work |
22:57.26 | war9407 | I assume I edit this in extensions.conf ? |
22:57.59 | BlargMaN00-lap | war9407: that or extensions_custom.conf... depends on how clean you like your files and dialplan... |
22:58.09 | war9407 | what is best practice? |
22:58.10 | BlargMaN00-lap | war9407: but basically yes... |
22:58.50 | war9407 | ; Example "main menu" context with submenu |
22:58.54 | war9407 | something to try? |
22:58.59 | BlargMaN00-lap | war9407: there really isn't a "best practice"... i like seperate files for different things, but that's just because I'm a neat phreak... |
22:59.31 | war9407 | do you have an example template that will work with the default install? |
23:00.11 | BlargMaN00-lap | war9407: yeah... gimme a sec, and i'll whip something up for you... than you can look over it, and see what i did, and hopefully it will point you in the correct direction... |
23:00.13 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
23:00.17 | war9407 | k |
23:01.02 | knarfly | I just switched to Fedora from FreeBSD for my * server...the install + dahdi went well. but I cannot seem to get my sip phone to register |
23:02.23 | knarfly | core show dialplan is not working...has the command changed in *-1.6.0.9? |
23:04.36 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
23:05.58 | knarfly | must have been the firewall with fedora...disabled it and now I'm working but not sure why core show dialplan doesn't work? |
23:07.44 | knarfly | hears only crickets on this channel |
23:07.45 | *** join/#asterisk kerx (n=kerx@adsl-69-104-67-217.dsl.irvnca.pacbell.net) |
23:08.37 | BlargMaN00-lap | war9407: http://pastebin.com/m637318f0 <-- check that out, and see if you understand what i did... |
23:08.45 | war9407 | k |
23:10.35 | BlargMaN00-lap | war9407: BRB |
23:10.40 | war9407 | replace john-exten with 4567 or what not I assume |
23:10.40 | war9407 | k |
23:10.51 | BlargMaN00-lap | war9407: yes |
23:11.42 | war9407 | [pstn] |
23:11.42 | war9407 | include => voicemail-menu |
23:11.42 | war9407 | exten => s,1,goto(voicemail-menu,s,1) |
23:11.44 | war9407 | trying |
23:13.05 | war9407 | [Apr 14 19:12:53] WARNING[5226] pbx.c: Channel 'SIP/pstn-00c639d0' sent into invalid extension 's' in context 'voicemail-menu', but no invalid handler |
23:13.17 | war9407 | s=the extension |
23:13.19 | war9407 | need to fix, |
23:13.20 | war9407 | sec |
23:14.04 | war9407 | [Apr 14 19:13:51] WARNING[5294] pbx.c: Channel 'SIP/pstn-011b1e00' sent into invalid extension 's' in context 'voicemail-menu', but no invalid handler |
23:14.04 | *** join/#asterisk rmod (n=rmod@unaffiliated/rmod) |
23:16.40 | *** join/#asterisk rmod (n=rmod@unaffiliated/rmod) |
23:17.00 | war9407 | it does not like the ,s |
23:17.09 | war9407 | [pstn] |
23:17.09 | war9407 | include => voicemail-menu |
23:17.09 | war9407 | exten => 123,1,goto(voicemail-menu,s,1) |
23:17.22 | *** join/#asterisk telecos (n=sergio@19.167.219.87.dynamic.jazztel.es) |
23:20.10 | war9407 | I did it verbatim like you have it written |
23:20.14 | war9407 | it worked ;) |
23:20.36 | war9407 | I left a voice message but it didnt send it anywhere, I will have to look into it more tomorrow |
23:20.39 | war9407 | one other question though |
23:20.44 | war9407 | it didnt ask me where I wanted to go |
23:20.50 | war9407 | it just said persom at extension 1111 is unavailable |
23:21.01 | war9407 | rather than say, 1111 or 1112 |
23:21.18 | *** join/#asterisk BlargMaN00-lap (i=BlargMaN@94.sub-70-216-251.myvzw.com) |
23:21.29 | BlargMaN00-lap | war9407: back... |
23:22.03 | war9407 | the choose extension part not working |
23:22.08 | war9407 | it just says person at extension is unavaialble |
23:22.33 | BlargMaN00-lap | war9407: how many digits is it letting you put in before it says that? |
23:22.37 | war9407 | also caller id is not getting passed through to the phone |
23:22.39 | war9407 | sec |
23:23.09 | war9407 | calling number.... ring..... person at extension 1234 is unavailable, please leave your message after the tone, when done please hang up or press the pound key |
23:23.47 | war9407 | it does not give me the option tfor one or the other |
23:24.10 | BlargMaN00-lap | war9407: do you have a phone registered to the extension 1234?? |
23:24.19 | war9407 | I am using an FXO (3102) |
23:24.22 | war9407 | no VoIP phones |
23:24.29 | BlargMaN00-lap | war9407: oh... ok |
23:24.33 | theodred | I am new to asterisk, a quarter of the way through the asterisk manual, and am trying to wrap my head around a few of the concepts there.. |
23:24.33 | war9407 | traditional phone line -----> spa3102 -----> inbound |
23:24.52 | war9407 | BlargMaN00: it says 'PSTN' on my caller id, how do I allow that to pass-thru? |
23:24.58 | war9407 | BlargMaN00: and second question, why no voicemail-menu? |
23:25.51 | BlargMaN00-lap | war9407: what context are you sticking the calls from the 3102 into?? |
23:26.11 | war9407 | http://graham.doel.org/knowledge-base/?View=entry&EntryID=9 |
23:26.40 | war9407 | http://blog.pathennessy.org/2009/01/01/configuring-linksys-spa-3102-for-asterisk/ <- this one worked |
23:28.22 | BlargMaN00-lap | war9407: ok... change [incoming] to [pstn] in what i pastebin'd |
23:29.08 | war9407 | that is what it is currently |
23:29.21 | war9407 | [pstn] | #include "voicemail_menu.conf" | include => voicemail-menu | exten => s,1,goto(voicemail-menu,s,1) |
23:29.27 | war9407 | | = return |
23:29.53 | BlargMaN00-lap | war9407: yes |
23:30.23 | BlargMaN00-lap | war9407: except no quotes around voicemail_menu.conf |
23:31.30 | BlargMaN00-lap | war9407: what version of * are you using? |
23:32.19 | war9407 | 1.6.0.9 |
23:32.39 | BlargMaN00-lap | ok |
23:33.15 | war9407 | more serious issue |
23:33.25 | war9407 | messages:[Apr 14 19:20:35] WARNING[5502] file.c: Failed to write frame |
23:33.31 | war9407 | (when I left a voice message) |
23:33.33 | war9407 | it went to /dev/null |
23:33.57 | knarfly | how can I check the status of an X101P card in my system....excuse me but I'm just converting over from zaptel to dahdi |
23:34.07 | war9407 | BlargMaN00: thx for getting my started |
23:34.11 | war9407 | BlargMaN00: will look more into this tomorrow |
23:34.13 | war9407 | my->me |
23:34.18 | war9407 | BlargMaN00: will tty tomorrow if you're around ;) |
23:34.31 | BlargMaN00-lap | war9407: no worries... anything to help |
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