| 00:01.21 | joat | ManxPower: I used the Festival wrapper from the Cepstral site |
| 00:01.36 | *** join/#asterisk coppice (n=chatzill@53.157.17.210.dyn.pacific.net.hk) |
| 00:02.05 | joat | it generates the sound file but doesn't play in the dialplan (still messing with it) |
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| 01:12.29 | drclue | I've started a Google hosted project for FastAGI/AMI http://code.google.com/p/fastagi-php-drclue/ . While the code is actually intended for a particular project I'm working on , I'm designing it it a project independent fashion. Both php scrips operate as daemons. The shared daemon class will also auto generate the init.d scripts for both the existing PHP scripts or any other daemon one might want to create with it |
| 01:26.50 | *** join/#asterisk LiNeTuX (n=LiNeTuX@119.153.205.68.cfl.res.rr.com) |
| 01:34.48 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 01:38.39 | drclue | While FastAGI/AMI is but one way to skin the cat, it does have two basic benefits over doing things strictly in the dial plan. One is scalability, as one can easily offload FastAGI to another server on the network, allowing asterisk to focus on being asterisk. The other big benefit is in the development cycle , where once something is in the editor , one can simply save/dial to test as opposed to reloads and the like |
| 01:44.30 | drclue | Unlike a lot of Asterisk tool development , this is currently funded development with yet an open-source result. The benefactor has some specific goals in mind but I was able to convince them that making things open source and listening to the needs of the asterisk community would give the tools a life of their own. So there it is , if you have always wanted some alternatives for PHP tools and have some ideas , this is the time |
| 01:56.35 | *** join/#asterisk WindBack (n=quassel@200-122-74-15.cab.prima.net.ar) |
| 01:57.20 | WindBack | If in any moment i want to return from a macro to the place it was called, is correct to come back using ${MACRO_PRIORITY} |
| 01:57.39 | WindBack | If in any moment i want to return from a macro to the place it was called, is correct to come back using:exten => s,9,Goto(${MACRO_CONTEXT},${MACRO_EXTEN},${MACRO_PRIORITY}) |
| 02:01.35 | WindBack | or do in need to add 1 to to MACRO_PRIORITY? |
| 02:09.09 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 02:11.43 | ectospasm | WindBack: it will go to that priority |
| 02:11.51 | ectospasm | If you add one, it will go to that priority + 1 |
| 02:12.51 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 02:12.57 | WindBack | ectospasm: to continue executing the next step after the macro calling, do i neet +1 or just MACRO_PRIORITY? |
| 02:13.07 | ectospasm | Normally you label the priorities, so instead of s,9,Goto, you'd have s,n(label),Goto |
| 02:13.20 | ectospasm | WindBack: you're jumping into the macro |
| 02:13.38 | ectospasm | so execution will continue from that point, until you jump out and go somewhere else |
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| 02:14.41 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:15.01 | WindBack | ectospasm: using Goto(${MACRO_CONTEXT},${MACRO_EXTEN},${MACRO_PRIORITY}) i'm not jumping into macro because i'm using contex, extension and priority, doen't it? |
| 02:15.46 | ectospasm | Wait, you're right |
| 02:15.54 | ectospasm | You're jumping into the macro improperly |
| 02:16.22 | WindBack | ectospasm: I'm trying to return from my macro to the next step after the macro calling |
| 02:17.05 | WindBack | ectospasm: another option is jumping to the last step in the macro and put there a Nop aplication |
| 02:17.23 | *** join/#asterisk InfoNutz (n=infonutz@S01060014bfdb2716.cg.shawcable.net) |
| 02:17.27 | ectospasm | It depends on what you need to do that for |
| 02:17.46 | WindBack | ectospasm: more or less I'm trying to create switch case inside my macro |
| 02:17.48 | ectospasm | Why are you exiting out of the macro then jumping back in? Couldn't you do all of it in the macro? |
| 02:19.34 | WindBack | ectospasm: at certain point of my dialplan i need to call a macro a inside the macro there are many returns depending of some condition. At any of this returns, I want to come back to the next step where the macro was calling |
| 02:19.46 | ectospasm | The language doesn't have that capability, you'll have to do it with IF, GotoIf, etc. |
| 02:20.10 | WindBack | ectospasm: sure, i did it using GotoIf application |
| 02:20.12 | ectospasm | look at the way FreePBX does stuff like that |
| 02:20.19 | ectospasm | I know they have macros that do that |
| 02:21.30 | WindBack | ectospasm: I'm just asking you, using inside the macro Goto(${MACRO_CONTEXT},${MACRO_EXTEN},${MACRO_PRIORITY}), is a correct way to return to the next step where the macro was calling?? |
| 02:21.58 | WindBack | s/calling/called |
| 02:22.02 | ectospasm | WindBack: You want GosubIf |
| 02:22.20 | ectospasm | asterisk -rx "core show application GosubIf" |
| 02:24.38 | ectospasm | Doing it with Goto/GotoIf is probably wrong, and won't work |
| 02:24.48 | [TK]D-Fender | WindBack: core show application macroreturn" |
| 02:24.51 | [TK]D-Fender | CRAZY PEOPLE |
| 02:25.09 | [TK]D-Fender | Nesting works. |
| 02:25.24 | [TK]D-Fender | and a GOTO is a broken concept to return from a macro |
| 02:25.58 | ectospasm | [TK]D-Fender: GosubIf would work, too |
| 02:26.20 | ectospasm | ...wouldn't it? |
| 02:26.25 | ectospasm | I dunno, I've never used it |
| 02:26.26 | [TK]D-Fender | ectospasm: that has nothing to do with returning |
| 02:26.32 | WindBack | ectospasm: not in my case |
| 02:27.02 | ectospasm | [TK]D-Fender: yes it does, it stores the value for when you call return in the dialplan you jump to |
| 02:27.29 | ectospasm | "core show application return" |
| 02:27.55 | WindBack | ectospasm: because you are propusing my to use gotoifsub instead macros, and inside my macro i 'm calling again goto aplication |
| 02:28.28 | ectospasm | Gotos considered harmful, anyway |
| 02:28.29 | WindBack | ectospasm: so I wouldnt return to the first calling |
| 02:28.41 | WindBack | [TK]D-Fender: macroExit |
| 02:28.50 | WindBack | [TK]D-Fender: not macroReturn |
| 02:28.52 | ectospasm | It basically pushes the label onto the stack |
| 02:29.01 | ectospasm | ...when you jump |
| 02:29.11 | ectospasm | when you return, it pops the first one off the stack |
| 02:29.22 | ectospasm | You'd have to return until you're back to your original position |
| 02:29.27 | WindBack | [TK]D-Fender: but yes, this is the thing i was looking for |
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| 02:30.10 | WindBack | ectospasm: yes, so i would need as many returns as goto calls i would have |
| 02:30.27 | WindBack | [TK]D-Fender: thanks |
| 02:31.13 | ectospasm | WindBack: however you design it, it's up to you |
| 02:31.23 | ectospasm | OK, with that I'm going to bed. |
| 02:34.32 | WindBack | ectospasm: good dreams |
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| 02:59.25 | *** join/#asterisk robito (n=robito@200.106.103.130) |
| 03:00.31 | robito | hey guys how can i use the meetme with the option "p" to leave a meetme press 1 Meetme(ROOM,p[(1)]) ?? |
| 03:04.04 | [TK]D-Fender | robito: no [] in parameters |
| 03:05.02 | *** join/#asterisk Scunizi (n=Scunizi@ip72-197-240-157.sd.sd.cox.net) |
| 03:05.04 | robito | thanks you =) |
| 03:07.33 | Scunizi | I'm learning about ippbx's and found several that suggest that a client needs a "win Proxy server" to help make a remote phone connect to the ip-pbx. I've tried connecting a phone/handset (not sip based) remotely and was able to get line status indication and ring but no dial tone or voice.. where do I begin investigating? |
| 03:08.47 | shido6 | ~answers |
| 03:08.48 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6: /path/to/src/asterisk/UPGRADE*.txt |
| 03:08.56 | shido6 | ~sipnat |
| 03:08.57 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:09.07 | shido6 | Scunizi take a peek at the info for NAT |
| 03:09.22 | Scunizi | shido6: thanks.. |
| 03:13.24 | phuff | If I wanna do "authorized numbers only can ring this extension" where would I start? |
| 03:13.33 | phuff | googling "Call screening" isn't working for me :) |
| 03:13.52 | phuff | I don't really think I want like 300 lines in extensions.conf for each number I want to authorize. |
| 03:14.10 | InfoNutz | hey everyone... wuts the best work around to have the chan_h323 module work with Opal with fedora, i searched everywhere with google and many forums with no luck. Many indicating its not supported. please help |
| 03:14.18 | [TK]D-Fender | phuff: "core show application read" ,"core show application authenticate" , "core show application gotoif".... |
| 03:16.26 | phuff | [TK]D-Fender: Thanks |
| 03:17.38 | phuff | [TK]D-Fender: Ah, not quite. |
| 03:17.48 | phuff | [TK]D-Fender: I want to have my phone only ring if the call is from somebody I've authorized. |
| 03:18.04 | [TK]D-Fender | phuff: And how do you "authorize" them? |
| 03:18.08 | phuff | By sticking their number in a file or a database |
| 03:18.19 | phuff | Or, in an extreme case, in a dialplan, I suppose. |
| 03:18.31 | [TK]D-Fender | phuff: "core show application mysql" |
| 03:18.59 | [TK]D-Fender | phuff: Or go make some AGI script that will do whatever lookup you feel like. |
| 03:19.20 | phuff | Ok, cool, thanks :) |
| 03:32.56 | WindBack | From my dialplan I'm calling a macro, this macro is calling the Dial() application. After I end the call I return from the macro using MacroExit application. Then outside the macro i'm using the DIALSTATUS variable. It appears to be working fine and not loosing the value returned by the Dial (), but is it correct? |
| 03:33.15 | WindBack | Shoud the Varibles remain the value after exit the macro? |
| 03:34.09 | WindBack | I think yes, because they are channel variables. Am i Right? |
| 03:34.16 | [TK]D-Fender | WindBack: There is no sense of scope in * |
| 03:35.54 | *** join/#asterisk joako_ (n=st0n3d@opensuse/member/joak0) |
| 03:36.03 | joako_ | where can I acquire more information on installing HUDlite w/o trixbox? |
| 03:36.14 | WindBack | [TK]D-Fender: |
| 03:36.14 | WindBack | thanks. That was my question |
| 03:37.11 | [TK]D-Fender | joako_: http://www.hudlite.org/linuxserver.htm |
| 03:38.12 | joako_ | [TK]D-Fender: That document is outdated. Not only that if you search the site there is no download for the server. FWIW I have the server installed from the trixbox RPMs on a CentOS 5 machine. |
| 03:38.20 | tgunr | I'm confused by something in the Makefile |
| 03:38.41 | tgunr | ASTSBINDIR=$(sbindir) but sbinddir is not in the makefile so where does it get defined? |
| 03:40.01 | *** join/#asterisk jfranco (n=ikono@190.128.108.172) |
| 03:40.15 | jfranco | Hi |
| 03:41.19 | *** part/#asterisk feld (n=feld@games.coffeenet.org) |
| 03:44.13 | jfranco | I have a question about the application Park. I'm using Asterisk 1.4.17. When i try to Park a channel in an parking extension that is already in use the channel doesnt never hangs up. Tha documentation says park() will park the call on that extension, unless it already exists. In that case, execution will continue at next priority. But i don't know if this is a bug |
| 03:44.44 | jfranco | Can anybody help me with this issue? |
| 03:46.35 | [TK]D-Fender | jfranco: PASTEBIN a failed parking attempt for us to look at. Also realize that your * version is very outdated |
| 03:46.37 | [TK]D-Fender | ~pb |
| 03:46.38 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 03:46.39 | [TK]D-Fender | ^^^^^^^^^^^^^ |
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| 04:10.43 | *** join/#asterisk cobra2599 (n=cobra259@cpe-66-25-60-236.tx.res.rr.com) |
| 04:11.10 | cobra2599 | Hey all, anyone know why i would received "Permission Denied" when trying to use the Manager API to orginate a call |
| 04:11.47 | drclue | Perhaps one did not do a proper login? |
| 04:12.05 | cobra2599 | login is working ok |
| 04:12.17 | cobra2599 | but when i try to orginate it throws me an error |
| 04:12.30 | drclue | Do you get a confirmation message from AMI at login? |
| 04:12.52 | cobra2599 | Authentication Accepted |
| 04:13.35 | drclue | Does the configuration for the log-in afford the activities? |
| 04:14.08 | cobra2599 | i believe so i am allowing all actions for the login |
| 04:15.12 | cobra2599 | read = system,call,log,verbose,command,agent,user |
| 04:15.12 | cobra2599 | write = system,call,log,verbose,command,agent,user |
| 04:15.22 | drclue | How are you trying to originate the call? |
| 04:15.42 | cobra2599 | let me post my code |
| 04:16.04 | drclue | Not in the chat I hope |
| 04:16.15 | cobra2599 | http://pastebin.org/9590 |
| 04:16.16 | cobra2599 | lol no |
| 04:16.20 | *** part/#asterisk cobra2599 (n=cobra259@cpe-66-25-60-236.tx.res.rr.com) |
| 04:16.26 | drclue | Use something like http://pastebin.com/ |
| 04:16.31 | *** join/#asterisk cobra2599 (n=cobra259@cpe-66-25-60-236.tx.res.rr.com) |
| 04:16.48 | drclue | Looking |
| 04:18.03 | cobra2599 | i tried to orginate in two plces to be sure it wasnt a coding problem |
| 04:18.31 | cobra2599 | Response is : Error and Message is Permission denied |
| 04:19.04 | drclue | Just for fun , try this code http://code.google.com/p/fastagi-php-drclue/downloads/list just to see if something will change |
| 04:19.31 | drclue | Asterisk can be way too much fun at times |
| 04:21.06 | WindBack | [TK]D-Fender: you have just toldme that in asterisk there is no scope sense. But the macro arguments ${ARG1} ${ARG2} etc they have scope sense. |
| 04:21.08 | drclue | I'm atill looking at your code , but it is a little hard to think in windows , since I stopped doing so about a decade ago. |
| 04:21.31 | [TK]D-Fender | WindBack: O RLY? Show me. |
| 04:21.41 | cobra2599 | same for me with PHP |
| 04:21.56 | WindBack | [TK]D-Fender: Sorry, I'm asking you |
| 04:22.44 | WindBack | [TK]D-Fender: for example if i call a macro inside another macro ${ARG1} is not the same in the inner macro |
| 04:22.48 | cobra2599 | the command is on line 23 of that pastie |
| 04:22.50 | drclue | Get your tickets now WindBack vs. Fender on pay per view!!!!!!!! |
| 04:23.51 | drclue | That sip address looks a little weird |
| 04:24.28 | drclue | emm magi jack |
| 04:24.33 | WindBack | [TK]D-Fender: I can pass to the inner macro the ARG2 value as first argument to the inner macro |
| 04:25.18 | drclue | magi is like a whole norther conversation |
| 04:25.39 | cobra2599 | yeah no matter what i put there i get the same error |
| 04:25.54 | cobra2599 | hey nothing wrong with free calling for testing |
| 04:25.57 | cobra2599 | lol |
| 04:26.17 | WindBack | [TK]D-Fender: Am I wrong? |
| 04:26.21 | drclue | The error may actually be from magi who really does not like asterisk folk |
| 04:26.45 | cobra2599 | nah i tried making it call an internal extension as well no go |
| 04:26.53 | [TK]D-Fender | WindBack: And watwindSure you can pass them, but see if your originals stay intact on your RETURN |
| 04:27.13 | drclue | I get lots of free calling w/o arguing with magi |
| 04:27.49 | WindBack | [TK]D-Fender: watwindSure?? |
| 04:28.17 | [TK]D-Fender | WindBack: Sure you can pass them, but see if your originals stay intact on your RETURN |
| 04:30.07 | WindBack | [TK]D-Fender: Now I understand, If i pass to the second macro the arg2 as first argument, I'm loosing the arg1 original value, isn't it? |
| 04:30.45 | [TK]D-Fender | WindBack: Go see |
| 04:32.59 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
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| 04:41.48 | cobra2599 | ok i figured it out |
| 04:42.03 | cobra2599 | you have to add "originate" to the list of allowed options |
| 04:42.16 | cobra2599 | and freepbx does not support this yet |
| 04:54.19 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 04:54.32 | joobie | hey guys.. is it possible to stop a while loop in extensions.conf? |
| 04:54.41 | joobie | i just want to stop a particular instance, not the loop all together |
| 04:55.01 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 04:59.01 | ManxPower | EndWhile will end the innermost loop, IIRC |
| 04:59.26 | joobie | it does |
| 04:59.30 | joobie | but it ends the loop all together |
| 04:59.37 | joobie | i just want to skip to the next iterration |
| 05:00.24 | [TK]D-Fender | joobie: So Goto just before the end |
| 05:00.27 | ManxPower | Isn't there a Next or something like that? |
| 05:00.46 | cobra2599 | Continue |
| 05:00.57 | joobie | TK, how can i goto a point in the middle of the dialplan? .. like Goto,context,line-num .. but i use 1,n,n,n for my dialplan |
| 05:01.27 | joobie | hmm |
| 05:01.30 | joobie | duno about continue |
| 05:01.37 | joobie | google not finding it |
| 05:02.10 | ManxPower | Is there a ContinueWhile? |
| 05:02.18 | ManxPower | "core show applications like while" |
| 05:02.37 | ManxPower | you don't use google to look up asterisk apps, you use Asterisk to look up Asterisk apps. |
| 05:02.43 | joobie | ContinueWhile: Restart a While loop |
| 05:02.45 | joobie | nice |
| 05:02.52 | joobie | good to know that core show cmd - ta |
| 05:03.04 | Tim_Toady | joobie exten => 123,n(tag),kokolala and use goto(cont,exten,tag) |
| 05:03.13 | ManxPower | "core show applications" is your friend |
| 05:03.22 | joobie | yer.. |
| 05:03.27 | joobie | well google is a better friend to me |
| 05:03.29 | joobie | we go way back |
| 05:03.37 | ManxPower | not for Asterisk applications. |
| 05:03.48 | joobie | for everything we are best buddies |
| 05:03.58 | joobie | but thanks :Pi like the "like while" bit - didn't know it was possible |
| 05:04.12 | ManxPower | *shrug* If you don't want to use the best tool for the job.... |
| 05:04.59 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-31-36.dsl.lsan03.sbcglobal.net) |
| 05:06.48 | [TK]D-Fender | joobie: and DUH... **LABELS*** |
| 05:07.40 | joobie | ahh k |
| 05:07.46 | joobie | thnx |
| 05:07.57 | joobie | i havent played with my dialplan in a while.. starting to forget |
| 05:08.24 | joobie | im trying to call my local computer shop and they are not picking up.. so just wrote a nice loop to ring them 300 times before hanging up.... |
| 05:09.07 | joobie | quit |
| 05:09.10 | joobie | woops |
| 05:09.14 | ManxPower | I did a similar thing to get thru to a cab company during events in New Orleans. |
| 05:09.29 | Tim_Toady | teenagers... |
| 05:09.33 | Tim_Toady | :P |
| 05:09.58 | joobie | heh cool |
| 05:10.23 | joobie | Tim_Toady, i just had enuf of dialing these guys.. have rung them manaully about 5 times today |
| 05:11.44 | Tim_Toady | next step is ping -f |
| 05:12.05 | joobie | heh |
| 05:12.40 | joobie | hmm.. is there a way to handle the busy tone to make it redial? |
| 05:12.57 | joobie | i tried using ${DIALSTATUS} but that only populates after the dial() function finishes |
| 05:13.13 | ManxPower | joobie: need to know the technology you are using first. |
| 05:13.14 | joobie | need some sorta realtime catch so when it's busy it doesnt wait for the timeout |
| 05:13.18 | joobie | 1.4 |
| 05:13.21 | ManxPower | you must be using analog |
| 05:13.24 | joobie | standard dialplans |
| 05:13.40 | joobie | im passing the call through to a sip provider |
| 05:14.07 | ManxPower | Dial should exit as soon as it gets back the SIP "Busy Here" message. |
| 05:14.15 | joobie | yea it's not |
| 05:14.22 | joobie | i removed the timeout and tested.. no joy. |
| 05:14.24 | ManxPower | joobie: then your carrier sucks |
| 05:14.29 | joobie | pennytel |
| 05:14.48 | joobie | no other way to achieve it? |
| 05:14.53 | ManxPower | I don't care who they are. If they are not sending back a Busy Here message for busy numbers then they suck |
| 05:15.15 | *** part/#asterisk cobra2599 (n=cobra259@cpe-66-25-60-236.tx.res.rr.com) |
| 05:15.23 | joobie | that's nice - but my problem still exists. |
| 05:15.25 | ManxPower | I bet if you watch your dialplan you'll see that the carrier answers your call before dialing out. |
| 05:15.27 | joobie | :P |
| 05:15.43 | ManxPower | joobie: you cannot do what you want to do with the limitations of the carrier you are using. |
| 05:16.15 | joobie | it's not answering it man.. it says "is making progress passing it to .." then waits |
| 05:16.32 | joobie | but the thing is, if i NoOp() $DIALSTATUS, even on a busy call it says 'no answer' |
| 05:16.45 | ManxPower | looks over his glasses at joobie |
| 05:17.32 | joobie | thanks :P *kicks pennytel* |
| 05:18.04 | ManxPower | I wish you the best of luck. |
| 05:18.09 | joobie | cheers |
| 05:18.25 | joobie | it's ok.. with a 300 dial attempt loop it will do the trick |
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| 06:04.32 | alunca | how can I setup a callback? |
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| 08:10.19 | siera08 | i have one problem which asterisk forwards voicemail to email. |
| 08:11.02 | siera08 | i use asterisk 1.4.16, and after i got email with voicemail, it's sender address is "asterisk@localhost.localdomain" |
| 08:11.16 | siera08 | somebody can change this address?? |
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| 08:22.58 | mort_gib | siera08: Have a look at voicemail.conf and your "MTA pfogram" config |
| 08:23.26 | siera08 | mort_gib: first, thank u for your reply. |
| 08:23.43 | siera08 | i use asterisk pbx, and i saw voicemail.conf before. |
| 08:24.06 | siera08 | but i can't find any field related with sender's address.. |
| 08:24.34 | siera08 | through googling, i think it's about sendmail's configuration. |
| 08:25.42 | mort_gib | siera08: I promise you it's one of the options in voicemail.conf |
| 08:27.15 | siera08 | mort_gib: i found that option in http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf |
| 08:27.23 | siera08 | but i can't find it. |
| 08:27.30 | mort_gib | Type it in.... |
| 08:29.18 | siera08 | ?? |
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| 08:30.28 | mort_gib | siera08: If it's missing from your actual config, add it! |
| 08:31.02 | siera08 | mort_gib: which option? |
| 08:31.30 | tengulre | how to transfer in agent mode? |
| 08:33.48 | mort_gib | siera08: Look fro Servermail in the link you sent.... |
| 08:34.17 | siera08 | mort_gib: |
| 08:34.29 | siera08 | servermail option isn't sender's address. |
| 08:34.37 | siera08 | i tried it before.. |
| 08:38.10 | mort_gib | Ok, if you say so :-/ |
| 08:39.19 | siera08 | mort_gib: thank u for your kind. :) |
| 08:40.38 | mort_gib | siera08: np, but please remember that we spend a lot of time in here helping out where we can... I might not be TK, but we all have to start somewhere. |
| 08:40.55 | mort_gib | I can PROMISE you that servermail is what you are looking for |
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| 08:46.22 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 08:46.27 | tengulre | hi,all |
| 08:46.39 | tengulre | how to transfer current calling in agents mode? |
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| 09:01.22 | joako | Am I correct in assuming backing up /etc/dahdi would be about the same as backing up /etc/zaptel.conf, correct? |
| 09:02.55 | siera08 | mort_gib: i found it. it's my mistake, |
| 09:03.23 | siera08 | spelling problem, "serveremail" option is true, but i test "serviermail" (omitted "e").. |
| 09:03.29 | siera08 | thank u.. |
| 09:03.36 | mort_gib | siera08: np |
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| 09:44.13 | angryuser | hi all, i have a strange issue, asterisk 1.2 music on hold plays once and then stops. i.e i call first time i heard the beginning of the song, i call second time, i hear the song from the place i left it, and the at the end, nothing, i am using mp3 128 fixed rate. Any help ? here is my musiconhold class "predecwengo => mp3:/var/lib/asterisk/mohmp3/predecwengo,-z" |
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| 09:55.10 | mort_gib | angryuser: Maybe you should consider moving to a new version of asterisk... |
| 09:56.04 | angryuser | mort_gib, lets consider this not as option |
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| 12:07.46 | Diblo | I have a problem with the asterisk to work with text (im). I can find a lot of sms but not abut text im. :p |
| 12:08.37 | cjk | hi, are there any mobile phones where you can save contacts in a special format like number##dtmf_sequence. so the phone dials number and sends the dtmfs once the call has been answered |
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| 12:24.58 | Naikrovek | what are you guys using for spam control on Linux these days? When I used to do this, Spamassassin reigned surpreme |
| 12:26.35 | [TK]D-Fender | Naikrovek: Still the same, and wrong channel... |
| 12:26.58 | Naikrovek | well pardon me for breaking into all the conversations that are going on |
| 12:27.14 | Pan3D | #network |
| 12:27.33 | Pan3D | #networking* |
| 12:28.08 | ariel_ | morning folks |
| 12:29.34 | *** join/#asterisk coppice (n=chatzill@52.204.17.210.dyn.pacific.net.hk) |
| 12:35.50 | MariusKarthaus | Hi, we have a number of different company brands, each with it's own phone number. If a customer calls us at brandX, i want asterisk to start calling all our regular-, mobile- and sipphones. Once one of those phones is picked up, drop the rest of the attempted new calls, playback some text-to-speech info about the calling customer and the brandname that he called on to the employee that picked up the phone and than transfer the call f |
| 12:35.50 | MariusKarthaus | <PROTECTED> |
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| 12:36.58 | alunca | weird! when I called out, it call itself. please help! |
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| 12:42.16 | [TK]D-Fender | MariusKarthaus: Originate in no way connects to another call already on the system |
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| 12:53.02 | Naikrovek | alunca: elaborate |
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| 12:58.27 | MariusKarthaus | [TK]D-Fender: but if my sip-phone can do it why should asterisk not be able to do it. If a customer calls me on my sipphone directly and I pick up the phone and tell him to hang on, open a second line call my colleague, tell him about the customer and transfer the customer to my colleague it work. So I assume that asterisk should be able to somehow be 'me' in that scenario |
| 12:59.42 | [TK]D-Fender | MariusKarthaus: And just because a screwdriver works very well at putting in screws doesn't mean its a HAMMER |
| 13:00.16 | [TK]D-Fender | MariusKarthaus: Originate spawns completely separate calls. NOTHING to do with any other. |
| 13:00.28 | [TK]D-Fender | ~assume |
| 13:00.29 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav It makes an (ass) out of (u) and (me) |
| 13:00.29 | leifmadsen | MariusKarthaus: you probably want Bridge() |
| 13:00.57 | coppice | Screwdriver? |
| 13:00.59 | coppice | Thank's miss, but I've got to take this bus out |
| 13:01.38 | MariusKarthaus | I know it spawns different calls. But isn't that just what my sipphone does when I do this sort of thing manually and use transeer to connect the two lines. I was trying to mimick the way I do it manually |
| 13:02.01 | leifmadsen | MariusKarthaus: you probably want Bridge() |
| 13:02.10 | *** join/#asterisk ice_croft (n=nolan@mail.kubkurort.ru) |
| 13:02.17 | MariusKarthaus | I'm looking into bridge jes |
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| 13:02.53 | [TK]D-Fender | coppice: http://pastebin.com/m1a75b958 |
| 13:05.09 | alunca | Naikrovek what do you mean by > <Naikrovek> alunca: elaborate ?? |
| 13:05.31 | Naikrovek | i mean I want you to elaborate on your problem. |
| 13:06.12 | [TK]D-Fender | Naikrovek: You just used a word he didn't understand to xplain the word he didn't understand |
| 13:07.15 | Naikrovek | well i assumed he was smarter than that, and didn't understand what I wanted to elaborate |
| 13:07.44 | alunca | I did double check all the setting; evyething should be okay. Also, I was testing with Vmware, and it was working fine. but now I am working on a real pc, and I can only received calls; whenever I call out, it will call myself. |
| 13:08.56 | Naikrovek | alunca: well something isn't configured correctly. Check your phone dialplan and your outbound route from the asterisk server. |
| 13:09.00 | MariusKarthaus | leifmadsen: if I use bridge, can I still 'say' (text-to-speech) things to both channels before connecting them together? |
| 13:09.05 | [TK]D-Fender | alunca: And that tells us precisely nothing about your failed calls... |
| 13:09.27 | [TK]D-Fender | alunca: pastebin it and it would be nice to know what the call is coming in on, etc |
| 13:09.31 | leifmadsen | shrugs |
| 13:09.49 | MariusKarthaus | leifmadsen: as I am on asterisk 1.4 i'd have to upgrade manually and i do not want to do that unless i know it will get the result I need |
| 13:10.07 | [TK]D-Fender | ~pb |
| 13:10.07 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:10.32 | leifmadsen | MariusKarthaus: I'd not suggest you upgrade at all until you've tested that (and everything else) on a separate server. You can get the answer to your question that way. |
| 13:12.21 | MariusKarthaus | leifmadsen: sure but still that is quite a big setup. Do you know if I use bridge, do I still need to create a outgoing call with originate? Or does brige do that for me? |
| 13:12.33 | leifmadsen | shrugs again |
| 13:13.08 | MariusKarthaus | ok sorry :P |
| 13:13.47 | alunca | Naikrovek I follow this guide from this link, http://forums.slickdeals.net/showthread.php?t=1480557 ... please search for "[FreePBX Configuration]" |
| 13:17.57 | [TK]D-Fender | alunca: FreePBX is NOT supported here... |
| 13:17.59 | [TK]D-Fender | ~freepbx |
| 13:18.00 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:19.32 | alunca | oh :( |
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| 13:33.42 | mort_gib | What is the experience with SwitchVox?? |
| 13:34.22 | Naikrovek | i couldn't get it to recognize that my polycom phone was there; it couldn't figure out the https (as opposed to http) admin page so it wouldn't provision the phone |
| 13:34.26 | Naikrovek | i got mad and stopped there. |
| 13:34.44 | mort_gib | Naikrovek: Was that for me?? |
| 13:35.01 | Naikrovek | it could ping the phone and all the other checks it does, but wouldn't bring the phone online because it couldn't get through htttps |
| 13:35.07 | Naikrovek | mort_gib: yes |
| 13:35.11 | mort_gib | Damn |
| 13:35.19 | Naikrovek | feel free to prove me wrong though |
| 13:35.31 | Naikrovek | is not the font of SwitchVox knowledge |
| 13:35.56 | mort_gib | So my client, on a 160 user site wants to go for SwitchVox "To take the risk out of the project..." |
| 13:36.22 | Naikrovek | mort_gib: well it is supported; i'm certain you could get help from digium |
| 13:36.27 | [TK]D-Fender | mort_gib: That just puts a pretty name badge on it :) |
| 13:36.43 | Naikrovek | it had a good UI, i liked it |
| 13:37.00 | mort_gib | [TK]D-Fender: Well they DID have a nice powerpoint presentation, that ALWAYS sells... |
| 13:37.11 | cjk | hi, if a user is in a call and receives a second call, blf state is not updated to "ringing" is there a way to change this behavior? |
| 13:37.44 | [TK]D-Fender | cjk: Show us... and describe your * version, and the devices involved in the calls |
| 13:38.24 | [TK]D-Fender | cjk: Ah, think I jus got it.... IIRC the state is that of the highest state of any call associated with the device |
| 13:38.29 | *** join/#asterisk SuPrSluG (n=chatzill@firewall-a.buf.ny.i-evolve.net) |
| 13:38.38 | mort_gib | Naikrovek: So, easy to do "complex" dial plans in??? |
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| 13:39.07 | Naikrovek | mort_gib: it runs asterisk, so anything possible with * should be possible with switchvox |
| 13:39.32 | merlin8282 | Hi people. Can I use mISDN with a Junghanns QuadBRI IDSN card ? |
| 13:39.36 | mort_gib | Yeah, but for a lot of things you would need the cli -right?? |
| 13:40.00 | Naikrovek | mort_gib: don't think so; they expose all that stuff through a custom UI that isn't available anywhere else |
| 13:40.31 | cjk | [TK]D-Fender, yep, now it would be good to update it to "ringing" in order that other users can do a pickup |
| 13:40.32 | Naikrovek | mort_gib: i'm sure that my polycom problem could be sorted out easy, and i'm sure you can do anything you need with it |
| 13:41.07 | Naikrovek | download it and try it; they have a free version |
| 13:41.14 | Naikrovek | maybe it'll work for you |
| 13:41.19 | mort_gib | Naikrovek: Yeah, I'll still be doing the implementation, we start demos and stuff next week |
| 13:41.48 | Naikrovek | mort_gib: please let me know your thoughts if you see me in here |
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| 13:42.19 | MrTelephone | what is the proper way to send unknown name/private number with Set Callerid or Callerid= in sip.conf? |
| 13:42.19 | mort_gib | -Will do, I just prefer "compile from source" Also I have not yet implemented an Asterisk yet where I didn't need a lot of other things... |
| 13:42.39 | [TK]D-Fender | cjk: There is some possibly massive hackery possible for this where you can set a custom DevState flag on call-in and unflag on M(), |
| 13:42.46 | *** part/#asterisk Carlos_PHX (n=carlos@ip68-108-193-174.ph.ph.cox.net) |
| 13:42.46 | [TK]D-Fender | cjk: Fugly at best... |
| 13:42.55 | MrTelephone | Some automated phone systems do not like UNKNOWN <0> or alpha characters between the brackets.. |
| 13:43.16 | [TK]D-Fender | MrTelephone: "core show application setcallerpres" |
| 13:43.46 | cjk | [TK]D-Fender, no the best way, do you think there is a need to introduce this feature in the core? |
| 13:44.15 | MrTelephone | where can I find a better description of the options? |
| 13:44.40 | Naikrovek | mort_gib: i've never build an asterisk system via "build from source" method, my employer uses TrixBox CE and it's working great for us. |
| 13:45.35 | mort_gib | Naikrovek: Good for you, I ended up compiling from source, and now that works great for me (and my clients) |
| 13:47.55 | *** join/#asterisk Ex_peter (n=peter@unaffiliated/expeter/x-019426) |
| 13:48.53 | [TK]D-Fender | mort_gib: Its a GUI, expect to find more than enough walls to bang your head on... |
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| 13:50.03 | mort_gib | [TK]D-Fender: So I do, I didn't select the version :-) -But then I hardly ever do. I'm rolling out a ridiculous Citrix 4.0 install these days... |
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| 13:51.08 | MrTelephone | hey fender thanks for the insight on that application |
| 13:51.15 | MrTelephone | it seemed to work fine |
| 13:51.31 | [TK]D-Fender | MrTelephone: You're welcome |
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| 14:11.52 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
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| 14:19.32 | merlin8282 | mmm... |
| 14:19.45 | merlin8282 | Is there any way to use mISDN under 2.6.26 (Lenny) ? |
| 14:20.45 | merlin8282 | I'm trying to compile mISDN 1.1.9.1 under Lenny, but in the kernel "make menuconfig" it doesn't appear :( |
| 14:22.53 | [TK]D-Fender | merlin8282: http://www.google.ca/#hl=en&q=debian+lenny+misdn&btnG=Google+Search&meta=&aq=0&fp=6393c88910a58a95 |
| 14:22.58 | *** join/#asterisk RaDiC_rs (n=quassel@p50908ED7.dip0.t-ipconnect.de) |
| 14:24.34 | merlin8282 | [TK]D-Fender: sorry, I forgot to say that I try to use * 1.6 |
| 14:26.23 | tzafrir_laptop | merlin8282, shouldn't it be built as a separate module? |
| 14:26.48 | merlin8282 | tzafrir_laptop: yes, it should. |
| 14:27.03 | tzafrir_laptop | out of the kernel tree, that is |
| 14:27.18 | tzafrir_laptop | do you have linux-headers-`uname -r` installed ? |
| 14:27.21 | merlin8282 | mmm, ok. |
| 14:27.37 | merlin8282 | yes |
| 14:28.48 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:29.11 | Da-Geek | Hi All... Quick Hardware recommendation question... |
| 14:29.20 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 14:29.37 | Da-Geek | i need a single PCI card that can run 2 PSTN Lines and 1 fax.... |
| 14:30.08 | Da-Geek | anyone got any suggestions (supplier in the UK if pos....) |
| 14:30.27 | Da-Geek | TDM type thing would be good... |
| 14:30.37 | merlin8282 | I would say a Digium TDM400p, but i'm far not an expert. |
| 14:30.41 | Da-Geek | just wondered what people would recommend ? |
| 14:30.54 | [TK]D-Fender | Da-Geek: aNY EXPANSION EXPECTATIONS? |
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| 14:31.43 | Da-Geek | [TK]D-Fender, tbh with this install not really.... 99.9% of outgoing with be SIP |
| 14:32.29 | Da-Geek | [TK]D-Fender, the card with be for the legasy PSTN inbound lines + 1 incoming/outgoing Fax |
| 14:32.36 | [TK]D-Fender | I'd recommend a Sangoma B600 |
| 14:33.26 | Da-Geek | has used TDM's before, never used the Sangoma's.... |
| 14:33.32 | Da-Geek | heads over to Google :-) |
| 14:33.33 | TSM | A series sangoma are better, lifetime warranty, more expensive than the B series |
| 14:35.14 | Da-Geek | umm the B600 looks interesting... |
| 14:36.03 | joako | Asterisk cdr-csv times are in GMT? |
| 14:37.10 | *** join/#asterisk coolthreads (n=coolthre@203-97-238-71.cable.telstraclear.net) |
| 14:37.49 | joako | TSM: I love the Sangomas just don't forget to update wanrouter when you update your kernel! Last night I was about to start reimaging a system until I had the clever idea to bust out a loopback plug (bad wiring inside a wall) |
| 14:38.35 | TSM | joako: up, its compiled against kerne |
| 14:38.55 | TSM | joako: up, its compiled against kernel, no probs wioth PRI for me as i have the NTE on the wall next to the box |
| 14:40.19 | joako | Yes, I know. Rule #1 don't assume anything (just because the cable was working for Ethernet doesn't mean it will work for T1) . I'm just saying wanrouter is just one more piece of software that just like any other software can cause problems. |
| 14:43.02 | mort_gib | joako: Yeah, and sangoma didn't have a stable Wanrouter for BRI until a month ago... |
| 14:43.02 | Da-Geek | is being very stupid.... |
| 14:43.29 | Da-Geek | cant remember what way round the FXS/FXO are.. |
| 14:43.51 | Da-Geek | witch do you need for a normal voice line and witch do you need for a Fax line / |
| 14:43.52 | Da-Geek | ? |
| 14:44.14 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:44.38 | *** join/#asterisk propellerhead (n=yogurt2u@host7.190-225-41.telecom.net.ar) |
| 14:44.42 | Naikrovek | ~fxofxs |
| 14:44.43 | infobot | from memory, fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 14:45.40 | Naikrovek | i remember it by remembering that the "S" in FXS provides Service |
| 14:45.52 | Naikrovek | where service == dialtone |
| 14:47.07 | [TK]D-Fender | S= Station |
| 14:47.13 | [TK]D-Fender | X = eXchange |
| 14:47.27 | Naikrovek | that's fine but i remember it as s=service |
| 14:47.32 | [TK]D-Fender | O = Office |
| 14:47.52 | Da-Geek | so if i want a PCI card to run 3 incoming PSTN Lines and also be able send Fax'es... |
| 14:47.53 | *** join/#asterisk af_ (n=getsmart@88-149-230-54.dynamic.ngi.it) |
| 14:48.04 | [TK]D-Fender | Naikrovek: Ok... I'll remember you as "Tim" :) |
| 14:48.10 | Da-Geek | i need 3 X FXO + 1 X FXS |
| 14:48.22 | Naikrovek | [TK]D-Fender: you can call me susie if it makes you happy. |
| 14:48.34 | Naikrovek | but i probably wont' remember to respond to it |
| 14:48.38 | [TK]D-Fender | spins up "A Boy Named Sue" |
| 14:50.26 | Da-Geek | so 3 Red and 1 Green ? is that right... |
| 14:50.41 | Da-Geek | = confused.com |
| 14:50.44 | *** join/#asterisk linuxgeek (n=uppal@119.59.81.198) |
| 14:51.00 | jaytee | Blankman: "Slap me around and call me Susan" |
| 14:51.05 | linuxgeek | Hi Everyone |
| 14:51.19 | Naikrovek | Da-Geek: if your fax is going to be plugged directly into your * box, yes. |
| 14:51.19 | jaytee | Da-Geek, yes 3 red (FXO) and 1 green (FXS) |
| 14:51.48 | linuxgeek | i am playing an announcement via agi , why do i keep on getting ast_openstream_full: File Welcome_welcome-english.wav does not exist in any format |
| 14:51.56 | linuxgeek | i have tried every format possible :( |
| 14:52.07 | jaytee | the answer is right there in the error message |
| 14:52.16 | linuxgeek | but it exists :( |
| 14:52.17 | jaytee | do a locate on the filename |
| 14:52.24 | jaytee | R U SURE? |
| 14:52.28 | [TK]D-Fender | LinSHOW uS |
| 14:52.35 | [TK]D-Fender | linuxgeek: SHOW uS |
| 14:53.08 | Da-Geek | Naikrovek, jaytee, so if i want this box to 'Send' faxes that it gets via a print driver can i use the FXO, since it is making the call ? |
| 14:53.11 | linuxgeek | /var/lib/asterisk/sounds/Welcome_welcome-english.wav |
| 14:53.11 | linuxgeek | just one sec |
| 14:53.26 | linuxgeek | http://pastebin.ca/1532596 |
| 14:54.23 | [TK]D-Fender | linuxgeek: Show us the failed CALL with AGI debug |
| 14:54.29 | Da-Geek | linuxgeek, how did you make the .wav ? |
| 14:54.44 | [TK]D-Fender | is already 99% certain of the problem |
| 14:54.46 | Naikrovek | Da-Geek: yes, in that case you could use one of the three FXO ports since there's no fax machine that needs dialtone |
| 14:54.49 | linuxgeek | audicity :-) |
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| 14:55.38 | linuxgeek | http://pastebin.ca/1532598 <-- updated the same paste |
| 14:56.23 | Da-Geek | Naikrovek, so could i just get a card with 3 or 4 FXOs and just use the * box to do the Fax'ing ? |
| 14:56.30 | linuxgeek | sorry , just one sec , updating it again |
| 14:57.22 | Naikrovek | Da-Geek: if it's a virtual, not physical, fax machine, it would simply dial out to the analog telephone network like any of the desk phones. |
| 14:57.34 | joako | mort_gib: Well FWIW didn't digium themselves have a BRI card with poor driver support... I never read anything about a BRI card with excellent driver support, then again BRI isn't too common where I am at the main use here really was before broadband internet was popular and 28.8k modems were common |
| 14:58.23 | linuxgeek | [TK]D-Fender , http://pastebin.ca/1532601 |
| 14:58.53 | [TK]D-Fender | linuxgeek: As i suspected... |
| 14:58.59 | Da-Geek | Naikrovek, aha... so if the customer still wants a big hunk of plastic/metal on the dest to send the faxes, i should get an FXS and plug the Fax in2 it so that the machine can Send faxes.. |
| 14:59.02 | [TK]D-Fender | linuxgeek: Never EVER specify the EXTEnSION |
| 14:59.47 | Naikrovek | Da-Geek: yes, if you have a physical fax machine that wants dialtone, you'll need a FXS module for that fax machine. |
| 14:59.54 | linuxgeek | oh but extensions is used to identify the call coming in on which shortcode |
| 15:00.18 | [TK]D-Fender | linuxgeek: NO |
| 15:00.21 | mort_gib | joako: Yes, but where I am, BRI is the only option, PRI is extremely expensive. I can't get incoming (local) numbers so we are pretty much stuck with BRI |
| 15:00.24 | Da-Geek | Naikrovek, perfect answer and now makes alot more sence to me :-) |
| 15:00.33 | Naikrovek | Da-Geek: when a fax is sent or received, one FXO port and that FXS port will both be in use at once, so maybe a better option is to just skip asterisk if the fax is going to have its own number all the time |
| 15:00.33 | Da-Geek | has only ever used FXO's |
| 15:00.34 | linuxgeek | ohkay , what should i use instead of that? |
| 15:00.49 | [TK]D-Fender | linuxgeek: You do not put ".wav" etc on the file to play. * picks it FOR YOU |
| 15:00.57 | linuxgeek | oh |
| 15:01.03 | linuxgeek | extension , file extension |
| 15:01.04 | linuxgeek | lol |
| 15:01.11 | linuxgeek | i thought asterisk extension |
| 15:01.14 | linuxgeek | let me fix that |
| 15:01.19 | Naikrovek | Da-Geek: and if that fax machine is busy. if it's not always busy then it makes sense to use one of the other lines for that. |
| 15:01.23 | [TK]D-Fender | reaches for his ClueBat |
| 15:01.27 | [TK]D-Fender | ... (tm) |
| 15:01.35 | Naikrovek | [TK]D-Fender: that was what i was going to say - drop the file extension |
| 15:01.36 | Da-Geek | Naikrovek, better to have it all virtual IMHO then i dont need to worry about the FXS |
| 15:01.41 | Naikrovek | didn't want to get yelled at again |
| 15:02.00 | [TK]D-Fender | QUIET SUE!!!! |
| 15:02.02 | [TK]D-Fender | :p |
| 15:02.05 | Naikrovek | lol |
| 15:03.35 | joako | mort_gib: I know they are popular in other places of the world... I actually looked at the digium specs for their BRI card and it was that it doesn't support North American BRI and actually that was the issue when I looked into it |
| 15:04.13 | mort_gib | joako: As it turned out they didn't support Gibraltar at first either. They do now but... |
| 15:04.29 | jaytee | "What if I don't want to be Mr. Pink?" |
| 15:05.13 | beek | morning [TK]D-Fender , jaytee |
| 15:05.41 | jaytee | morning beek |
| 15:06.15 | [TK]D-Fender | beek: Mornin' |
| 15:07.18 | joako | mort_gib: Well I am hearing BRI prices are becoming very expensive in some places in the US perhaps in some cases double, I don't think the telcos ever wanted to sell it I rememeber in the mid 1990's some telcos would not even know what BRI/ISDN is, at least their sales depts. |
| 15:07.34 | mort_gib | joako: I started looking into providing DDI, as the first, but the Telco license required was some £50K |
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| 15:09.15 | mort_gib | joako: Here it's for practical reasons... PRI's are just SO expensive. The minimum you can get is half a PRI, where BT sells them in blocks of 8 or so |
| 15:14.57 | yang | hello mort_gib |
| 15:15.02 | *** join/#asterisk skrusty (n=root@83.166.170.138) |
| 15:15.10 | mort_gib | Hi Yang, how are things?? |
| 15:15.13 | skrusty | hi everyone |
| 15:15.51 | skrusty | does anyone know if lockmode is implemented in 1.6.1 ? |
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| 15:29.54 | skrusty | anyone here know much about app_voicemail and lockmode? |
| 15:30.40 | [TK]D-Fender | skrusty: http://www.google.ca/#hl=en&q=asterisk+lockmode+voicemail&btnG=Google+Search&meta=&aq=f&fp=6393c88910a58a95 |
| 15:30.55 | [TK]D-Fender | skrusty: you'd be amazed what you can find in about 3 seconds flat when you look... |
| 15:31.01 | skrusty | yeah, i found that too |
| 15:31.08 | [TK]D-Fender | skrusty: And? |
| 15:31.16 | skrusty | but, it doesn't seem to make any difference putting lockmode into asterisk.conf |
| 15:31.28 | skrusty | it still fails when locking a file over smbfs |
| 15:31.44 | [TK]D-Fender | skrusty: PB your config |
| 15:31.55 | skrusty | so, i am assuming that either, asterisk.conf is the wrong place to reference it... |
| 15:31.58 | skrusty | one moment |
| 15:33.39 | skrusty | http://pastebin.com/d639e2f41 |
| 15:34.39 | skrusty | oh im such a pleb |
| 15:35.22 | skrusty | one moment |
| 15:35.53 | skrusty | right, corrected the '=' with '=>' |
| 15:35.57 | skrusty | and tested |
| 15:36.07 | skrusty | but still the same problem |
| 15:36.24 | [TK]D-Fender | skrusty: http://www.mail-archive.com/svn-commits@lists.digium.com/msg17766.html |
| 15:36.30 | [TK]D-Fender | skrusty: Compare vs your code |
| 15:36.54 | skrusty | one moment.. |
| 15:36.55 | [TK]D-Fender | skrusty: A more permanent solution might be to reverse the default value in source |
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| 15:39.49 | skrusty | i can see the AST_LOCK_TYPE flags in app.h |
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| 15:45.19 | chnuebi | Hey guys, is it possible to set the dahdi channel for incoming calls? I have an E1 line and I hope this is possible setting in the incoming extension in extensions.conf... Im really new in this topic =/ I have asterisk 1.6.2.0 but i dont that is a version-related question |
| 15:45.20 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:47.03 | [TK]D-Fender | chnuebi: ... |
| 15:47.04 | [TK]D-Fender | ~book |
| 15:47.05 | 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 |
| 15:47.06 | [TK]D-Fender | ^^^^^^^ |
| 15:47.14 | hardwire | indeed |
| 15:47.32 | [TK]D-Fender | chnuebi: Incoming calls go to the context you specified for your channels in chan-dahdi.conf |
| 15:47.41 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:48.04 | rene- | chnuebi: i believe you can not choose which channel from your E1 trunk you can receive the call |
| 15:48.45 | rene- | you can choose which channel you want to -send- calls, not sure about receiving them |
| 15:50.11 | chnuebi | there are 100 numbers which are terminated on this line and i want to say, if someone calls this number move it to channel 25 for example |
| 15:50.23 | citywok | to choose the inbound channel, your provider would need to configure that. to choose the inbound context, as TK said you set that in the dahdi/zaptel config file for the channel |
| 15:51.21 | citywok | chnuebi: you can't transfer a call from one channel to another afaik, and i'm not sure why it would ever even matter. |
| 15:51.36 | [TK]D-Fender | chnuebi: No, and what would be the point? |
| 15:52.03 | [TK]D-Fender | chnuebi: the reason for PRI + DID si so you can receive whatever DID is dialed on an available channel |
| 15:52.23 | [TK]D-Fender | chnuebi: Now * has no control over this, but your telco might |
| 15:52.47 | [TK]D-Fender | chnuebi: Though almost anyone I can think of would say this is a pretty crazy sounding idea |
| 15:53.22 | chnuebi | that was not me, who decided to do that^^ We want to allocate several channels for several projects... |
| 15:53.54 | [TK]D-Fender | chnuebi: Then you have to ask your TELCO |
| 15:53.59 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:54.07 | [TK]D-Fender | chnuebi: You do not allocate channels to DID's, they do |
| 15:54.38 | [TK]D-Fender | chnuebi: And for that reason, yes I can see a need, and I had 1 client who needed this. |
| 15:54.44 | grandpapadot | chnuebi: Or just simply route them accordingly once they get to your asterisk box. |
| 15:54.46 | chnuebi | yea thank you, i'll try. I thought maybe I can set something through the signalling channel... |
| 15:54.52 | bmoraca | yep...telco determines to which channel an inbound call is sent. |
| 15:55.19 | *** join/#asterisk neverblue (n=chatzill@unaffiliated/neverblue) |
| 15:55.37 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 15:55.43 | neverblue | what is the distinct difference between the Polycom Soundpoint IP320 and the 330 ? |
| 15:56.13 | [TK]D-Fender | grandpapadot: No, you missed the point... |
| 15:56.30 | [TK]D-Fender | neverblue: 330 has a passthrough port for PC |
| 15:56.42 | bmoraca | neverblue: the 330 has an integrated switch |
| 15:56.52 | neverblue | ah |
| 15:56.55 | neverblue | right, thank you |
| 15:57.05 | [TK]D-Fender | neverblue: And avoid both and get the 321/331 instead (respectively). revision with more RAMN, longer support, etc |
| 15:58.12 | TSM | from what i was told the 321/331 were the same as 320/330 but just packaged with a PSU |
| 15:58.28 | *** part/#asterisk chnuebi (n=patrick@42-204.203-62.cust.bluewin.ch) |
| 16:00.34 | TSM | ahh just seen, some site just packaged them diffrently and the diffrent part numbers, yep they are diffrent revisions |
| 16:01.45 | [TK]D-Fender | TSM: there may be a PS-bundled box for those as well, but the base is itself different |
| 16:03.27 | TSM | [TK]D-Fender: if im only ever gona use the default SIP stuff, none of the productivity package, apart from the added ram is there any diffrence, im seeing about a $10 price increase of the 3x1 versions over the 3x0 versions |
| 16:04.07 | *** join/#asterisk jpcansa (n=jpbenavi@216.25.165.166) |
| 16:04.19 | grandpapadot | They are the same price at http://telephonydepot.com, $106 |
| 16:04.25 | [TK]D-Fender | TSM: the older ones will get EOL'd sooner, you needs may change, and so will their firmwares and you'll find yourself losing updates faster |
| 16:04.39 | [TK]D-Fender | TSM: Feel free to paint yourself into a corner over $10 however... |
| 16:05.18 | [TK]D-Fender | TSM: there may be other benificial changes, but I didn't do an overly thorough comparson |
| 16:05.19 | TSM | right back to my supplier and get them to give me a price, oh dont get me on price and budget i have enough pain in my office to order a $50 item |
| 16:06.06 | [TK]D-Fender | TSM: yes, and what is the cost of a bad business decision? |
| 16:06.17 | [TK]D-Fender | TSM: there is a difference between frugal & stupid |
| 16:06.50 | TSM | yep thats not my decision, i want the better stuff, but somtimes they dont see that |
| 16:06.54 | TSM | they=directors |
| 16:06.57 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 16:07.11 | [TK]D-Fender | lunch time, BBIAB |
| 16:08.37 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:08.41 | *** join/#asterisk scud (n=scud@69.73.19.194) |
| 16:09.42 | jaytee | http://www.youtube.com/watch?v=8r1CZTLk-Gk |
| 16:11.05 | scud | i have a bounty for $200 (that is all I can afford) for someone to port asterisk to Windows Mobile so I can resell it. What is the status on that? |
| 16:11.13 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 16:11.15 | TSM | why |
| 16:11.26 | TSM | too much work |
| 16:11.28 | putnopvut | $200? |
| 16:11.33 | grandpapadot | scud: It's done, http://127.0.0.1/dev/null |
| 16:11.36 | TSM | what you mean is that you want a SIP client |
| 16:11.43 | russellb | scud: are you serious? |
| 16:11.46 | jaytee | lol |
| 16:11.52 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
| 16:12.29 | outtolunc | waits for the bidding wars to start |
| 16:12.35 | scud | russellb: yeah, I'm hurting financially and I need this shit ASAP. |
| 16:13.01 | grandpapadot | jaytee: great vid |
| 16:13.03 | russellb | Well, in all seriousness, there is _no_ way you're going to get that, 1) for $200, 2) period. |
| 16:13.13 | russellb | so, I would look into another route to solve your financial woes. |
| 16:13.19 | jaytee | http://www.osburn.com/asterisk-sign.png |
| 16:13.27 | grandpapadot | scud: although porting to iphone has had some limited success already ... |
| 16:13.37 | rene- | iphone is unix like |
| 16:14.25 | jaytee | I heard a rumor that the folks at NORAD ported the WOPR to Windows Mobile |
| 16:15.30 | rene- | why would someone would buy an asterisk based windows mobile system, i also think you mean softphone |
| 16:15.31 | grandpapadot | jaytee: omg, they made a wargames 2 and it was a terrible movie |
| 16:15.33 | coppice | I thought it was Burger King who ported the WOPR to WIn Mobile |
| 16:15.47 | jaytee | never saw the second one |
| 16:16.32 | rene- | scud: sjphone runs on windows mobile |
| 16:17.00 | jaytee | I never eat at Booger King anymore. Everytime I'd order a Whopper it would come out looking like it was run over by a car and leaking runny mayo everywhere |
| 16:17.03 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 16:17.55 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:17.58 | coppice | is the Mayo Clinic an obesity treatment centre? |
| 16:18.44 | hardwire | .... |
| 16:18.57 | hardwire | no. |
| 16:18.57 | hardwire | google? |
| 16:18.57 | hardwire | :P |
| 16:19.11 | hardwire | You're thinking of the Hamsammich Clinic. |
| 16:19.18 | jaytee | lol, seriously no. the Mayo Clinic is one of the leading cancer research centers and hospitals in the US |
| 16:19.22 | Qwell | mmm, sammich |
| 16:19.37 | scud | rene-: that looks perfect to resell to canadians and my south african clients. |
| 16:19.40 | Qwell | where's Joel? he could school us on Mayo |
| 16:19.41 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 16:20.28 | coppice | americans get so fat on mayo they are too lazy to even say its full name |
| 16:20.32 | jaytee | I love mayo, I'd rather risk a heart attack using mayo than smear that leper jizm called Miracle Whip on my sammich |
| 16:20.34 | rene- | hahha |
| 16:20.42 | hardwire | scud: you're in the hospital? |
| 16:21.00 | coppice | jaytee: you could have a sandwich with neither |
| 16:21.15 | jaytee | coppice, or spicy brown mustard |
| 16:21.19 | Qwell | jaytee: or both |
| 16:21.31 | Qwell | also, what's with the miracle whip hate? |
| 16:21.42 | *** join/#asterisk ice_croft (n=nolan@mail.kubkurort.ru) |
| 16:21.43 | *** part/#asterisk dustybin (n=dustybin@thinkdebian.org) |
| 16:21.56 | jaytee | I've just never liked it. I always thought it tasted "unnatural" |
| 16:22.11 | Qwell | don't think of it as mayo (it isn't) |
| 16:22.47 | grandpapadot | D9 rocked. |
| 16:22.59 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:23.10 | jaytee | i'd never think of it as mayo, a substitute for mayo or any kind of condiment. You can keep all of it for yourself. |
| 16:23.36 | jaytee | D9? District 9? |
| 16:24.06 | grandpapadot | Yup, loved it. |
| 16:24.26 | Naikrovek | would love to see that |
| 16:24.31 | jaytee | it's gotten great reviews and placed #1 at the box office this weekend |
| 16:24.42 | russellb | guys, we were having a productive discussion about asterisk on windows mobile, why do you bring us off topic with mayo and movie talk? :-p |
| 16:25.07 | Qwell | D9 has more chance of becoming reality. |
| 16:25.16 | jaytee | lol |
| 16:26.12 | grandpapadot | ;) |
| 16:27.54 | Naikrovek | oh i wish aliens WOULD land on earth. |
| 16:28.10 | Naikrovek | I want to know how all my Christrian friends would explain that |
| 16:28.24 | grandpapadot | Anyone familiar with http://howlertech.com and their G.729 codecs? |
| 16:28.27 | phuff | fwiw, mormon's wouldn't have much of a problem with it ;) |
| 16:28.37 | Naikrovek | neither would scientologists |
| 16:28.55 | Qwell | Naikrovek: depends on how much they've paid |
| 16:28.56 | jaytee | any truly intelligent species out there would avoid us like the plague once they picked up our television broadcasts and saw one episode of American Idol or watched Glen Beck. |
| 16:28.59 | phuff | touche' |
| 16:29.00 | Qwell | must. bite. tongue. |
| 16:29.10 | Naikrovek | qwell: lol |
| 16:29.16 | grandpapadot | I'm not a scientologiest, but scientology is not a religion of itself, it's a gateway to a deeper understanding of your religion and how it applies to your existence, I've done some reading. |
| 16:29.48 | [TK]D-Fender | grandpapadot: 2 words... Farenheit 451 <- |
| 16:29.49 | jaytee | L. Ron Hubbard founded the Church of Scientology over a fifty dollar bet. |
| 16:30.06 | grandpapadot | lol |
| 16:30.13 | jaytee | I'm serious! |
| 16:30.18 | *** join/#asterisk awkfu (n=awkfu@166.205.7.8) |
| 16:30.24 | grandpapadot | I'm just stating that it's not a religion, I didn't really offer my personal opinion. |
| 16:30.30 | bmoraca | anyone have any experience with Lucent TNTs? |
| 16:30.33 | coppice | a bet with robert heinlein |
| 16:31.23 | jaytee | and I agree that it's not a religion yet it claims the tax exempt status of one. personally I don't think any church should be exempt since most of them "meddle" in political issues which violates separation of church and state. |
| 16:32.25 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 16:32.31 | IBC_Jkenney | Should we change the title of the room to Asterisk a discussion of theology and Life maybe? |
| 16:32.44 | Naikrovek | eh maybe not; my fault for subject change |
| 16:32.44 | IBC_Jkenney | :) |
| 16:33.00 | Naikrovek | i'm not surprise how quickly we strayed from asterisk, though |
| 16:33.05 | jaytee | Heinlein was a great author once you got past his rabid chauvinism. L. Ron Hubbard was just a hack author and everything I've read by him was just pure shit. |
| 16:33.52 | box2 | at least it was pure |
| 16:34.24 | jaytee | Let's get back to this talk of porting Asterisk to Win Mobile. Anyone want to volunteer to work on the C# coding? :-) |
| 16:35.08 | IBC_Jkenney | how do you know the difference between pure shit and just regular shit. Is there a unit of measure? or some since to determine shits purity? |
| 16:35.23 | IBC_Jkenney | since=science |
| 16:35.25 | Naikrovek | it's measured by the amount of people that smell it |
| 16:35.25 | IBC_Jkenney | sorry |
| 16:35.45 | IBC_Jkenney | well that good be just gas not necessarily shit |
| 16:35.46 | Naikrovek | asterisk on windows would be cool, though |
| 16:35.51 | IBC_Jkenney | good = could |
| 16:35.53 | Qwell | Naikrovek: no |
| 16:35.56 | IBC_Jkenney | I cannot type today |
| 16:36.24 | IBC_Jkenney | Blaspheme! (now founding the church of asterisk) |
| 16:36.35 | rene- | teliax or gafachi? which is best? |
| 16:36.45 | Naikrovek | qwell: sure it could; it's software. |
| 16:36.47 | jaytee | yeah, Asterisk on Windows would rock because everyone wants to run a telephony system on an OS with a kernel that's has so much latency you could balance your checkbook waiting for a response to an interrupt. |
| 16:36.49 | *** join/#asterisk b14ck (n=comradeb@72.37.252.50) |
| 16:36.54 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 16:36.55 | b14ck | yo |
| 16:36.59 | box2 | i always use a properly calibrated micrometer to measure my shits purity |
| 16:37.07 | b14ck | anyone using the tc400b transcoding card? i'm thinking about getting one |
| 16:37.14 | *** join/#asterisk awkfu (n=awkfu@166.205.7.8) |
| 16:37.35 | coppice | b14ck: do you need to transcode G.723.1? |
| 16:37.42 | b14ck | i need it for g729 |
| 16:37.47 | jaytee | box2, impossible with a micrometer, you need a gas chromatograph |
| 16:38.05 | b14ck | i'm kinda afraid of using a 3.0GHz dual xeon to handle 100 g729 calls |
| 16:38.05 | coppice | b14ck: what's wrong with the host CPU? |
| 16:38.15 | b14ck | i just dont want to stress out the cpu too much i suppose |
| 16:38.25 | b14ck | and having that card seems like it would be a nice way to defer the loda |
| 16:38.28 | box2 | perhaps a petrometer? |
| 16:38.31 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 16:39.16 | coppice | A 3.0GHz dual xeon will do 100 calls of G.729 without problems, unless you mean a olde worlde P4 based xeon |
| 16:39.39 | b14ck | on digiums site it says that you need at 2.8GHz dual xeon to handle ~80 calls |
| 16:40.01 | b14ck | im kinda assuming that they are rounding up rathre than down =p |
| 16:40.13 | Naikrovek | if they're all being transcoded in software, yes |
| 16:40.30 | coppice | that might be old info based on a P4 based Xeon |
| 16:40.35 | Naikrovek | if each phone is set to use G.729 then you can handle thousands possibly |
| 16:40.37 | b14ck | oh |
| 16:40.44 | b14ck | I'm not using phones, actually |
| 16:40.49 | Naikrovek | okay |
| 16:40.51 | b14ck | It's just scripts running |
| 16:40.55 | b14ck | So calls come in, hit the server |
| 16:41.03 | b14ck | and we do some audio processing |
| 16:41.05 | b14ck | etc |
| 16:41.11 | Naikrovek | yes that's CPU bound |
| 16:41.17 | b14ck | ya |
| 16:41.28 | b14ck | I'm just wondering if anyone has experience with that card and/or likes it. |
| 16:41.40 | b14ck | I'm just worried about owning my host CPU and having poor call quality or something. |
| 16:41.52 | Qwell | yet you'll happily run trixbox O.O |
| 16:42.03 | b14ck | i dont run trixbox =p |
| 16:42.15 | Naikrovek | what's the matter with trixbox? asterisk + freepbx + linux |
| 16:42.32 | Qwell | ~trixbox |
| 16:42.32 | infobot | trixbox is, like, a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 16:42.33 | bpgoldsb | Is there a way to get a list of or call all registered sip phones via dialplan? Or do I need to use some AGI? |
| 16:42.38 | Qwell | plus a whole bunch of other useless bloated crap. |
| 16:42.47 | rene- | ~freepbx |
| 16:42.48 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:43.02 | Naikrovek | quantify this "other useless bloated crap." |
| 16:43.08 | Qwell | Naikrovek: trixbox. |
| 16:43.16 | b14ck | Qwell, so what do you think of the tc400b? oo |
| 16:43.20 | Naikrovek | can you parse english |
| 16:43.29 | Qwell | Naikrovek: all of it. |
| 16:43.34 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 16:43.52 | Naikrovek | then you understand what i'm saying, but screwing with me, which makes your opinion useless |
| 16:43.55 | grandpapadot | Naikrovek: Basically this channel is designed to support Asterisk proper, not everybody uses it as a PBX solution. |
| 16:44.35 | box2 | i use it just for conf with friends |
| 16:44.45 | box2 | and we just soundboard eachother |
| 16:44.48 | box2 | good times |
| 16:44.54 | Naikrovek | qwell: you said trixbox is asterisk + linux "plus a whole bunch of other useless bloated crap." i would like to knwo what the "other useless bloated crap" is |
| 16:45.06 | Qwell | Naikrovek: everything beyond "asterisk + linux" |
| 16:45.09 | grandpapadot | Naikrovek: "useless crap" of no interest to people in #asterisk |
| 16:45.23 | Qwell | the GUIs, the scripts. |
| 16:45.28 | Naikrovek | so, since you don't liek it, you can't even tell me WTF it is |
| 16:45.31 | grandpapadot | ... 3rd party modules |
| 16:45.42 | Qwell | grandpapadot: ...buggy 3rd party modules. |
| 16:45.49 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 16:45.56 | Naikrovek | you're so opposed to something you can't even quantify, i wonder if you even know what your'e talking about |
| 16:46.02 | grandpapadot | Naikrovek: No man, you're getting bowed up over nothing, the reason #freepbx or #trixbox *exists* is to help you with all the crap that doesnt' belong here. |
| 16:46.25 | bpgoldsb | Naikrovek: everything beyond "asterisk + linux" the GUIs, the scripts. |
| 16:46.31 | bpgoldsb | I'm pretty sure he answered your question |
| 16:46.32 | b14ck | anyone kow if the tce400p fits into a 1u? :D |
| 16:47.39 | Naikrovek | bpgoldsb: linux has guis and scripts all on its own, he did not answer my question. WHICH ... you know what? never mind. it's a religious thing to him and therefore he doesn't need to knwo anything about anything to openly dislike it |
| 16:48.11 | box2 | why is silly guy acting so silly |
| 16:48.15 | Naikrovek | and i'm not looking for help on it, which is why i was asking here |
| 16:48.22 | grandpapadot | Naikrovek: lol, dude, not at all. You wouldn't go talk about things specific to the Debian Linux Distro in the #kernel channel. |
| 16:48.23 | [TK]D-Fender | Naikrovek: Linux does not have GUI's |
| 16:48.24 | *** join/#asterisk davidandgoliath (n=David@S0106001d60d4e488.vn.shawcable.net) |
| 16:48.30 | Qwell | Naikrovek: BAD packaging standards. The whole distro is just crap. |
| 16:48.34 | Qwell | What more do you want? |
| 16:48.48 | Naikrovek | that's a start - i just wanted to see if you knew WTF you were talking about |
| 16:49.15 | Qwell | No, I talk randomly about things I don't understand because I like to hear myself talk. |
| 16:49.23 | bpgoldsb | Qwell, he wants you to fax him your resume and birth certificate. |
| 16:49.24 | Naikrovek | that i believe |
| 16:49.42 | [TK]D-Fender | Naikrovek: And Trixbox runs forked *, and forked FreePBX, and proprietary shit that none of the original makers give a shit about wanting to support, and their attempt to dumb it down so much for the masses attracts idiots. We see their kind in here all the time. |
| 16:50.01 | Naikrovek | "their kind" |
| 16:50.08 | Naikrovek | oohhhkaayy |
| 16:50.12 | [TK]D-Fender | Naikrovek: Naikrovek Yes... IDIOTS |
| 16:50.21 | [TK]D-Fender | Naikrovek: Droves upon droves |
| 16:50.44 | [TK]D-Fender | Naikrovek: Everyone going "why can't I jsut click here and do it?" |
| 16:51.03 | jaytee | There is no God but Asterisk and Spencer is it's Prophet! Infidels and heretics must die.........unless they bribe us with cheese. |
| 16:51.30 | [TK]D-Fender | Naikrovek: Most things they come in here for has nothing to do with *, but rather the 10 tons of other crap it comes bundled with |
| 16:51.30 | box2 | i'm so allergic to cheese i would probably explode |
| 16:51.51 | Naikrovek | it doesn't come bundled with anything |
| 16:51.55 | ariel_ | I feel we need a disclaimer about trixbox... oh the name says it. trix on you if you use it.... |
| 16:51.56 | jaytee | allergic to cheese? wow, that totally sucks dude! I'm so sorry for you. |
| 16:52.06 | Naikrovek | it's the distros you're railing on |
| 16:52.16 | Qwell | trixbox *IS* the distro. |
| 16:52.19 | coppice | cheese is so bloody expensive here |
| 16:52.24 | Naikrovek | we're off trixbox now you dork |
| 16:52.26 | box2 | jaytee: :(( |
| 16:52.31 | jaytee | coppice, where is here? |
| 16:52.58 | coppice | HK |
| 16:53.26 | box2 | i should move to china, they hate dairy there |
| 16:53.27 | jaytee | puts note in journal, "never go to Hong Kong, the cheese is too expensive" |
| 16:53.43 | *** join/#asterisk jtodd (i=lf1dcbo8@ns.fox-den.com) |
| 16:53.43 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:53.46 | Naikrovek | i lived in HK for a while, never will I see a city more crowded |
| 16:54.03 | coppice | box2: there's a pizza hut on every corner |
| 16:54.13 | box2 | ! |
| 16:54.29 | Naikrovek | no there isn't. there are cheap, cheap cabs to nearby pizza huts tho |
| 16:54.59 | Pan3D | hehe |
| 16:55.19 | Pan3D | I saw a picture of a HK cab the other day |
| 16:55.25 | coppice | dunno. cabs are too pricy for me |
| 16:55.35 | Pan3D | the inside looked like an airplane cockpit -- with advertising |
| 16:55.56 | box2 | they have TV's on the busses too |
| 16:56.00 | Naikrovek | i remember looking down from my office at a sea of red cabs all over all the streets |
| 16:56.06 | Naikrovek | every day |
| 16:56.12 | coppice | A HK cab looks nothing like an airplane cockpit :-\ |
| 16:56.22 | Naikrovek | and taking cabs all the way across town for USD$4 |
| 16:56.40 | coppice | when was that? in the 19th century? |
| 16:56.50 | Naikrovek | 2001-2002 |
| 16:57.18 | coppice | $4 == HK$31. That doesn't take you very far |
| 16:57.41 | Naikrovek | i remember avoiding the chicken because it had neither the flavor nor consistency of chicken. |
| 16:58.30 | coppice | are you sure it was HK you lived in? :-\ |
| 16:58.43 | Naikrovek | positive, it stank insanely |
| 16:59.02 | jaytee | lots of Chinese people with British accents |
| 16:59.03 | Naikrovek | when i was there USD$1 was HK$4 |
| 16:59.07 | box2 | maybe you were on the Kowloon side |
| 16:59.25 | Naikrovek | lived in sydney for two years as well |
| 16:59.29 | Naikrovek | sydney was awesome |
| 16:59.32 | Naikrovek | but expensive |
| 16:59.59 | Pan3D | coppice: I was being metaphoric, but yes, they look like very busy interiors -- http://www.johnnyjet.com/images/PicForNewsletterJapan2005TokyoInsideTaxi.JPG |
| 17:00.14 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 17:00.21 | box2 | tokyo taxi? |
| 17:00.33 | Pan3D | whooops, wrong one |
| 17:00.35 | Pan3D | hold on |
| 17:01.18 | Pan3D | hmmm... it's not in my history anymore. It was a particular article on Hong Kong on the BBC I think. |
| 17:01.38 | coppice | Naikrovek: the HK dollar has been locked at US$7.8 for the last 25 years or more |
| 17:02.06 | coppice | sorry US$1/7.8 |
| 17:02.08 | Naikrovek | my memory is fuzzy on it. I worked 18-20 hour days while i was there |
| 17:02.15 | Naikrovek | didn't pay too much attention to my surroundings |
| 17:02.31 | Naikrovek | i remember it was fixed, yes, but i would have sworn it was 1:4 ratio |
| 17:02.57 | Pan3D | ahhh, here it is |
| 17:03.01 | Pan3D | http://www.norcalminis.com/2009/05/proof-cellphones-can-be-driving.html |
| 17:03.32 | Pan3D | Gizmodo, not BBC |
| 17:04.49 | coppice | Pan3D: that cab driver must have some stramge sideline business :-) |
| 17:04.59 | Naikrovek | yeah |
| 17:05.22 | Pan3D | It struck me as interesting at the time because I'm reading a book on Cell phones and social connectedness. |
| 17:05.35 | Pan3D | coppice: haha, something |
| 17:07.22 | grandpapadot | Is there a way to force asterisk 1.4 to send out current subscription information (in the case where a reboot leaves Polycom phone's watchbuddy status flipped)? |
| 17:10.10 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096590413.dsl.bell.ca) |
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| 17:14.45 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 17:14.56 | manxpower | ~answers |
| 17:14.57 | infobot | somebody said answers was Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6: /path/to/src/asterisk/UPGRADE*.txt |
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| 17:19.27 | *** join/#asterisk maour (n=quassel@unaffiliated/maour) |
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| 17:21.58 | maour | i have a problem ! i can dial 101 with X-Lite but i can't dial any other numbers (users) ! :p , more info and configs >> http://pastebin.com/m64672c76 |
| 17:22.54 | maour | should i register all of users in sip proxy in X-Lite ?! |
| 17:23.29 | [TK]D-Fender | maour: Pastebin a failed call with SIP DEBUG enabled |
| 17:23.31 | [TK]D-Fender | ~pb |
| 17:23.32 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:23.35 | [TK]D-Fender | ^^^^^^^^^^ |
| 17:24.17 | *** join/#asterisk rossand (n=rossand@CPE0018f8360b13-CM0013711c94ee.cpe.net.cable.rogers.com) |
| 17:24.48 | maour | ok , but just anwer this question ?! should i register all of users in sip proxy in X-Lite ? because after i registers a user in X-Lite / Sip Proxy i can call it |
| 17:25.07 | maour | <PROTECTED> |
| 17:25.13 | maour | :-/ |
| 17:28.25 | maour | seems my problem solved :p |
| 17:33.08 | *** part/#asterisk jpcansa (n=jpbenavi@216.25.165.166) |
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| 17:40.33 | grandpapadot | Is there a way to force asterisk 1.4 to send out current subscription / hint information (in the case where a reboot leaves Polycom phone's watchbuddy status flipped)? |
| 17:43.34 | *** join/#asterisk tripps (n=sean@76.31.197.242) |
| 17:45.36 | bpgoldsb | Is there a way to get a list of or call all registered sip phones via dialplan? Or do I need to use some AGI? |
| 17:46.08 | grandpapadot | bpgoldsb: What are you trying to accomplish (endgame)? |
| 17:46.15 | bpgoldsb | Dial all connected devices. |
| 17:46.18 | Naikrovek | pretty easy with one of those inferior asterisk linux distributions |
| 17:46.20 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) |
| 17:46.45 | bpgoldsb | All connected SIP devices, rather |
| 17:47.03 | eppigy | all your face |
| 17:47.29 | grandpapadot | Naikrovek: lol, wow |
| 17:47.47 | Naikrovek | wow what |
| 17:49.44 | *** join/#asterisk jmworx__ (n=jeval@mail.octasic.com) |
| 17:51.11 | manxpower | bpgoldsb: 'asterisk -rx "sip show peers" |
| 17:51.25 | bpgoldsb | That gets me the list, sure. |
| 17:51.35 | bpgoldsb | But that doesn't do anything with regards to dialing. |
| 17:51.51 | manxpower | Nope. You'll have to write that feature yourself. |
| 17:51.52 | leifmadsen | you need to create a script to do that |
| 17:52.01 | bpgoldsb | Thats what I was asking :) |
| 17:52.28 | manxpower | I'd suggest using an AGI, have that ask the Manager interface (asterisk -rx can throw junk in the results), then parse it all. |
| 17:52.28 | bpgoldsb | Thanks |
| 17:52.41 | bpgoldsb | Thats option 2. |
| 17:52.53 | bpgoldsb | I'm using realtime for my sip information, so I might just query that. |
| 17:53.26 | *** join/#asterisk rossand (n=rossand@CPE0018f8360b13-CM0013711c94ee.cpe.net.cable.rogers.com) |
| 17:53.27 | grandpapadot | bpgoldsb: If you're using realtime, then just loop your results with the MySQL command and build a dial-string. |
| 17:53.42 | grandpapadot | bpgoldsb: don't forget to get rid of the final & in your dial string |
| 17:53.50 | bpgoldsb | grandpapadot, Thats the plan. |
| 17:53.55 | grandpapadot | bpgoldsb: Plus you can check if each device is online during the loop |
| 17:54.08 | bpgoldsb | Really? hmm. |
| 17:54.10 | grandpapadot | bpgoldsb: We do this with remote call groups and before sending calls to phones. |
| 17:54.25 | Qwell | Naikrovek: I never claimed they were all inferior. I only made claims about one in particular. |
| 17:54.38 | grandpapadot | lol |
| 17:54.44 | *** join/#asterisk benklop (n=ben@97-118-197-210.hlrn.qwest.net) |
| 17:54.57 | benklop | hello everyone |
| 17:55.21 | benklop | is there a way to have the parking lot play ring tones instead of hold music? |
| 17:56.25 | grandpapadot | benklop: Not that I know of, you can hack the code or just use a musiconhold class that has a signel file that's just ringing. |
| 17:59.06 | *** join/#asterisk MarkBest (n=chatzill@66.236.6.194.ptr.us.xo.net) |
| 17:59.15 | benklop | grandpapadot: that's what i figured |
| 17:59.34 | benklop | grandpapadot: thanks |
| 17:59.54 | justdave | I'm not getting any CALLERID(name) data at all, on any channel... considering I have incoming channels from three different phone companies (one of them is a PRI) that seems odd... is there anything I should look at in asterisk before I assume all three phone companies aren't sending it and complain to them? |
| 18:00.17 | justdave | (internal extensions have caller id names - just incoming stuff from outside doesn't) |
| 18:01.03 | justdave | on the very first line in my dialplan for that context I have it NoOp with the callerid, and the number prints, and name is an empty string |
| 18:01.44 | Naikrovek | yeah that's how PSTN works I think. No names are sent, you must look them up |
| 18:01.51 | Naikrovek | only numbers are sent |
| 18:02.04 | justdave | how do the little boxes on home phone lines get the names then? |
| 18:02.25 | grandpapadot | justdave: cnam is the tech, check out http://www.listyourself.net |
| 18:02.44 | justdave | actually pretty sure that was all transmitted with it, because I've used it before on generic modems before |
| 18:02.51 | grandpapadot | justdave: basically all carriers share a distributed database and when calls come in they lookup the caller id name info from the database. |
| 18:03.12 | grandpapadot | justdave: it's not, just the number goes from carrier to carrier |
| 18:03.14 | Naikrovek | if you're have three physical POTS lines, you should get caller ID |
| 18:03.29 | justdave | ah, so it's because I'm on a PRI they're expecting me to look it up myself |
| 18:03.49 | Naikrovek | but I have a voip provider, I just get the number, so I use callerid-superfecta or whatever it's called to search the net for name |
| 18:04.04 | grandpapadot | justdave: no, your carrier should be able to lookup the cnam info and send it along ... |
| 18:04.06 | justdave | and the other two are VOIP, and they probably just pass on what they get, which probably comes in the same way |
| 18:06.39 | Naikrovek | http://nerdvittles.com/?p=609 |
| 18:09.56 | MarkBest | Can anyone assist in troubleshooting a 'Q931' Normal Unspecified Error? |
| 18:11.04 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 18:12.57 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 18:16.00 | [TK]D-Fender | [14:03]<justdave>ah, so it's because I'm on a PRI they're expecting me to look it up myself <- no |
| 18:22.01 | benklop | grandpapadot: what part of the source would that code be in? I'm thinking maybe the "r" option from queue() could be added to it or something |
| 18:22.30 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 18:22.53 | benklop | grandpapadot: i'm doing the google voice setup actually, is there another way to do this that would ring? |
| 18:23.09 | benklop | i've only been working with asterisk for about a week now |
| 18:25.26 | grandpapadot | benklop: Not sure. |
| 18:27.35 | grandpapadot | benklop: You would be better off modifying a non-asterisk module like valetparking.c and then just compiling and loading that ... |
| 18:27.40 | grandpapadot | benklop: Let me find the source, one sec. |
| 18:28.18 | grandpapadot | benklop: http://www.voip-info.org/wiki/view/AGX+Extra+Addons+for+Asterisk |
| 18:28.19 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 18:28.30 | *** part/#asterisk neverblue (n=chatzill@unaffiliated/neverblue) |
| 18:28.32 | grandpapadot | benklop: Hack on the valetparking.c |
| 18:29.01 | grandpapadot | benklop: Then just drop in in your asterisk source/apps dir and compile away |
| 18:29.31 | grandpapadot | benklop: instead of make install, just copy the resulting apps/valeparking.so into your asterisk modules dir and then just "module load valetparking" |
| 18:30.05 | grandpapadot | benklop: There's probably a way to just compile the module ... |
| 18:37.52 | *** join/#asterisk Cyon (n=cyon@fxp1.dmz1.cro1.bestweb.net) |
| 18:43.27 | *** join/#asterisk ethicx (n=ethicx@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 18:43.30 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 18:43.42 | ethicx | hi you all. |
| 18:44.43 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:46.54 | *** join/#asterisk rossand (n=rossand@CPE0018f8360b13-CM0013711c94ee.cpe.net.cable.rogers.com) |
| 18:49.52 | Naikrovek | omg my company newsletter has a friendly article about dress code |
| 18:50.04 | Naikrovek | wtf am i doing with my life working here |
| 18:51.00 | Naikrovek | section called "know your mate" W T F |
| 18:51.44 | [TK]D-Fender | Mating in the office is generally frowned upon... |
| 18:52.27 | *** join/#asterisk maour_ (n=quassel@unaffiliated/maour) |
| 18:52.48 | Naikrovek | this is the lowest rent newsletter ever |
| 18:53.38 | maour_ | which softphone is better to use in Linux ? seems X-Lite don't work on Linux anymore ! i mean in ver 3.0 or 4.0 :p |
| 18:53.56 | [TK]D-Fender | ~ekiga |
| 18:53.57 | infobot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 18:54.35 | *** join/#asterisk dustybin (n=dustybin@thinkdebian.org) |
| 18:54.45 | maour_ | [TK]D-Fender: seems nice , thanks |
| 18:54.57 | dustybin | why do you need to re-boot after compiling asterisk? |
| 18:55.13 | [TK]D-Fender | dustybin: Who said you did? |
| 18:55.47 | dustybin | [TK]D-Fender: a lot of howtos say so.. maybe they are wrong! |
| 18:56.00 | [TK]D-Fender | dustybin: I don't see you showing me one... |
| 18:56.26 | dustybin | doesnt matter |
| 18:56.46 | [TK]D-Fender | dustybin: Of course not, that might simply show an inability to read or interpret ;) |
| 18:56.55 | dustybin | :( |
| 18:57.06 | dustybin | <---- [FAIL] |
| 18:58.38 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.231.210) |
| 19:03.25 | *** join/#asterisk RaDiC_rs (n=quassel@bchm-4d090f29.pool.mediaWays.net) |
| 19:10.19 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
| 19:11.30 | *** join/#asterisk zmb (n=zm23@zaara.cuit.columbia.edu) |
| 19:14.27 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 19:15.47 | elitecoder | I'm wondering how to implement "Configure outbound routes" in this: http://forum.bandwidth.com/showthread.php?tid=139&pid=173#pid173 |
| 19:15.53 | elitecoder | could anyone tell me where I do that / how? |
| 19:17.43 | [TK]D-Fender | elitecoder: #freepbx <- thats where |
| 19:17.46 | [TK]D-Fender | ~freepbx |
| 19:17.47 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:17.51 | elitecoder | what |
| 19:17.56 | elitecoder | that's asterisk isn't it |
| 19:18.13 | [TK]D-Fender | elitecoder: No, it isn't |
| 19:18.13 | Naikrovek | no |
| 19:18.13 | elitecoder | it says right on the top |
| 19:18.24 | [TK]D-Fender | elitecoder: Thats like saying yuo drive a KENWOOD just because thats the brand of STEROE in your car |
| 19:18.25 | elitecoder | asterisk + freepbx .. is that not asterisk lol? i thought it was both |
| 19:18.32 | Naikrovek | you can go ask in #freepbx or be ignored here |
| 19:18.39 | manxpower | elitecoder: OK. You configure an "outbound route" by editing extensions.conf, adding the exten => line matching your route (usually _1NXXNXXXXXX) and Dial() for the Tech you want. |
| 19:19.00 | grandpapadot | lol, omg |
| 19:19.02 | grandpapadot | ya'll |
| 19:19.09 | elitecoder | so what do I need to do, to have a fallback outgoing sip peer? |
| 19:19.15 | [TK]D-Fender | route... what a horribly worthless word in * context... |
| 19:19.18 | elitecoder | they have two so I want to use both somehow like in that config |
| 19:19.26 | [TK]D-Fender | elitecoder: You need a primary. |
| 19:19.26 | elitecoder | yeah well they suck a bit |
| 19:19.49 | grandpapadot | elitecoder: There's always a heated debate regarding #asterisk vs #freepbx, basically, in a nutshell, any help you get here will very likely break your ability to use the GUI tools that make freepbx useful. Best bet, join #freepbx and ask your questions. |
| 19:20.02 | manxpower | elitecoder: Check the value of DIALSTATUS or HANGUPCAUSE to determine the exit status of Dial, then run Dial again if the result is something that indicates a failure. |
| 19:20.05 | elitecoder | I'm using asterisk |
| 19:20.09 | elitecoder | plain old f'in aserisk |
| 19:20.10 | elitecoder | lol |
| 19:20.13 | elitecoder | that's why im here |
| 19:20.17 | manxpower | elitecoder: then you have your answer. |
| 19:20.37 | elitecoder | k ty ill read everything and see what might help me |
| 19:20.48 | manxpower | ~answers |
| 19:20.49 | infobot | extra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6: /path/to/src/asterisk/UPGRADE*.txt |
| 19:21.00 | manxpower | elitecoder: Also look thru the Asterisk Book. |
| 19:21.14 | manxpower | grumbles at VMware. |
| 19:22.03 | Naikrovek | what's the vmware problem |
| 19:22.40 | elitecoder | manxpower: Question then... when do I know to stop trying? |
| 19:23.15 | elitecoder | I aleady have thinks .. mostly working. I have a couple problems. 1 is congestion and 2 is failures. but we called the #'s and they seem to be working fine. |
| 19:23.19 | elitecoder | things * |
| 19:24.06 | manxpower | Naikrovek: The problem is that I have VMware on a slow mahine. |
| 19:24.17 | Naikrovek | that could be a problem |
| 19:24.21 | elitecoder | Yesterday I spent some time sorting out the congestion and got it down to 1%, the failures are occurring on about 30% of outbound calls. Today congestion is back to 20% and I'm positive it's because of bw.com's call volume or something |
| 19:24.42 | manxpower | What are your HANGUPCAUSE valuse? |
| 19:24.45 | Naikrovek | 32-bit vmware is horrible in comparison to 64-bit vmware |
| 19:25.02 | elitecoder | manxpower: I only have failure REASON's |
| 19:25.22 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:25.36 | manxpower | No, the value of HANGUPCAUSE is a number. I don't think DIALSTATUS has "FAULURE" as a reason |
| 19:25.46 | elitecoder | Well, .. one second |
| 19:26.14 | elitecoder | http://www.voip-info.org/wiki/view/Asterisk+Reason+variable |
| 19:26.29 | elitecoder | I'm using .call files in the outgoing directory |
| 19:27.09 | manxpower | elitecoder: I wish you the best of luck. |
| 19:27.11 | elitecoder | so when a call fails, I use that value to determine what happened |
| 19:27.15 | grandpapadot | elitecoder: We use bw.com as one of our wholesale carriers and aren't having any problems, we send approximately 5,000 calls an hour their way. |
| 19:27.29 | elitecoder | f. |
| 19:27.52 | elitecoder | manxpower: can you elaborate as to why you gave up right there |
| 19:27.53 | elitecoder | ? |
| 19:28.26 | manxpower | elitecoder: 1) I don't get paid to help people 2) I'm not interested in debugging .call files today |
| 19:28.29 | elitecoder | grandpapadot: are oyu using astgerisk? |
| 19:28.45 | grandpapadot | elitecoder: Yes, 1.4.26 compiled against debian 4 (etch). |
| 19:28.57 | elitecoder | well I don't got a lot of money to spend on it right now but i'd send someone 20 bucks if they helped me figure it out |
| 19:29.40 | elitecoder | grandpapadot: could I see your sip conf lines for the bw peer? |
| 19:31.19 | grandpapadot | elitecoder: http://pastebin.com/m60daf8ac |
| 19:32.15 | elitecoder | any reason you chose to have qualify 3000? |
| 19:32.24 | elitecoder | their forum says 300 so i'm wondering what the difference is |
| 19:32.29 | grandpapadot | Yea, so our calls will fall through if the peer is offline. |
| 19:32.43 | grandpapadot | 3000 is 3 seconds |
| 19:32.55 | elitecoder | ok |
| 19:33.00 | elitecoder | mind if i pm you? |
| 19:33.20 | grandpapadot | sure, but you'll likely get more opinions in channel |
| 19:33.30 | elitecoder | alright well, fall through toooo? |
| 19:33.45 | grandpapadot | Alternate providers/routes/etc |
| 19:33.53 | citywok | grandpapadot: 5000 calls an hour? what kind of deal did you work out with them, are you on the unlim(3000minute) turnks? |
| 19:34.14 | manxpower | Use Local/ aka chan_local aka doc/localchannel.txt instead of a real Channel/Tech |
| 19:34.38 | elitecoder | yeah that's what i need to figure out. manxpower said to check the hangupcause and dialstatus but i don't know if i can since i use .call files |
| 19:34.49 | manxpower | You can when you use Local/ |
| 19:34.52 | grandpapadot | So if the bw.com peers go offline because they are latent the call will still go out through another provider .. in fairness, they are our third provider, we go outbound through a bank of 96 PRI's, then to Level 3/XO, then to bw.com |
| 19:35.16 | grandpapadot | citywok: It's per minute, don't know the details |
| 19:35.19 | elitecoder | LOL citywok take a orda prease! |
| 19:35.27 | elitecoder | chitty beef? |
| 19:35.30 | raden_work | for some reason im not getting my background voice promp http://www.voltarclamps.com/files/sip.txt |
| 19:35.39 | citywok | lol, grandpapadot teh deal we worked out with them got our LD rate to about 1c/m |
| 19:35.46 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 19:35.59 | citywok | us & canada |
| 19:36.01 | grandpapadot | what kind of chitty volume are you sending them? |
| 19:36.02 | benklop | grandpapadot: thanks, i had to step out for a bit |
| 19:36.13 | citywok | 5000-8000calls/day |
| 19:36.57 | citywok | we were paying 1.5c/m on our old E&M T1's |
| 19:37.02 | elitecoder | grandpapadot: ya ho dod you chain the providers like that? |
| 19:37.11 | elitecoder | or at least what's the terminology so I can google it |
| 19:37.33 | grandpapadot | elitecoder: You can work with the ${DIALSTATUS} variable or just stack them one line at a time with Dial(), if the call fails, it will proceed to the next line. |
| 19:37.40 | elitecoder | f |
| 19:37.43 | elitecoder | I'm using .call files |
| 19:37.54 | citywok | elitecoder: that still goes in to a context w/ a dialplan |
| 19:38.09 | grandpapadot | elitecoder: Just use Local/2048828888 in your call file vs SIP/20202020@wherever |
| 19:38.16 | elitecoder | ooo.. |
| 19:38.21 | grandpapadot | elitecoder: google "Local Channels" |
| 19:38.22 | elitecoder | I see what manxpower was saying now |
| 19:38.27 | grandpapadot | elitecoder: google "Asterisk Local Channels" |
| 19:38.29 | manxpower | Hence my suggestion to read localchannels.txt |
| 19:38.34 | manxpower | which should explain it all |
| 19:38.36 | elitecoder | i'm sorry manxpower |
| 19:38.43 | elitecoder | I was trying to understand I just didn't get it |
| 19:38.50 | elitecoder | thank you |
| 19:38.54 | manxpower | elitecoder: be nice to a kitten and I'll forgive you. |
| 19:39.12 | grandpapadot | Local Channels essentially send the call processing through wherever you enter it via the Local/<pattern> in the context it's in. |
| 19:39.22 | elitecoder | np i'm nice to miles, out cat. He's named after the Tails on the sonic games :D cause he's orange haha |
| 19:39.25 | elitecoder | looks like tails kinda |
| 19:40.18 | benklop | elitecoder: he probably doesn't have 2 tails tho... otherwise that would be kinda amazing |
| 19:40.32 | elitecoder | haha no, just one. But it's very fluffy |
| 19:40.39 | benklop | heheh |
| 19:40.48 | elitecoder | so in that way, it's twice the tail most cats have lol. |
| 19:40.58 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 19:42.34 | *** join/#asterisk pgarcia (n=chatzill@yoda.kanatek.com) |
| 19:44.41 | jpcansa | how can i block transfers to a dahdi channel? |
| 19:45.59 | pgarcia | Hi, by any chance, is there anyone here from UK that uses BT analog lines? I need a recording from the FSK (callerid) signal (recorded before the 1st ring) and was wondering if someone here could help me on that.... |
| 19:46.55 | [TK]D-Fender | jpcansa: You can detect blind transfers, but not attended |
| 19:51.25 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 19:52.21 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 19:54.30 | benklop | what is the valetparking app for? what's the difference between that and normal parking? |
| 19:55.30 | [TK]D-Fender | benklop: The valet |
| 19:56.18 | benklop | okay, so there's a man in my asterisk box that takes my keys? |
| 19:56.31 | elitecoder | hey grandpapadot can you show me your dialplan where you chain the providers? |
| 19:56.35 | elitecoder | would save me some time |
| 19:57.02 | raden_work | can someone tell me why my menu not working when i dial in ? i dial a extension and it just repeats itself again |
| 19:57.08 | raden_work | http://www.voltarclamps.com/files/sip.txt |
| 19:57.25 | [TK]D-Fender | elitecoder: What part of "chech the dialstatus variable"do you not understand? |
| 19:57.42 | elitecoder | it's still faster fender |
| 19:58.03 | [TK]D-Fender | raden_work: exten => _X.,1,Answer() <- BAD IDEA! Everything yuo dial matches this! |
| 19:58.17 | [TK]D-Fender | raden_work: Shoot-on-sight offense |
| 19:58.21 | elitecoder | i get the benefit of someone's experience who's placing 5000 calls an hour so that's useful |
| 19:58.39 | raden_work | [TK]D-Fender, what should it be thats the only way i have been able to get inbound calls |
| 19:58.44 | [TK]D-Fender | raden_work: exten => 101,n,goto(internal/101,1) <- you also haven't seemed to have thought to make a "1" priority |
| 19:59.23 | [TK]D-Fender | [15:58]<raden_work>[TK]D-Fender, what should it be thats the only way i have been able to get inbound calls <- it may work to match the incoming call, but you probably are looking at or thinking about whats really coming in. |
| 19:59.45 | *** join/#asterisk qdk (n=qdk@0x573d8d8b.bynqu1.dynamic.dsl.tele.dk) |
| 20:00.06 | [TK]D-Fender | elitecoder: Yes its faster if we type everything out for you. My rates are very accessible. |
| 20:00.20 | raden_work | im a lil confussed about the priority thing |
| 20:00.22 | *** join/#asterisk rossand (n=rossand@CPE0018f8360b13-CM0013711c94ee.cpe.net.cable.rogers.com) |
| 20:00.53 | elitecoder | It's not a hard thing. I think I'll talk to you another time though. So angry lol :x |
| 20:01.04 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 20:01.35 | raden_work | <PROTECTED> |
| 20:01.35 | raden_work | <PROTECTED> |
| 20:01.35 | raden_work | <PROTECTED> |
| 20:02.49 | [TK]D-Fender | raden_work: register => 17772445766:pine4556@callcentric.com/101 <-- it goes to "101" because you TELL IT TO |
| 20:03.00 | raden_work | i was just looking at that |
| 20:03.22 | [TK]D-Fender | raden_work: And that stupi pattern match will match your 3 digit dials and go in a silly loops |
| 20:03.31 | raden_work | :( |
| 20:03.35 | [TK]D-Fender | raden_work: Stop running IVR's off a pattern match |
| 20:03.42 | raden_work | ivr ? |
| 20:03.46 | [TK]D-Fender | ~ivr |
| 20:03.47 | infobot | well, ivr is Interactive Voice Response |
| 20:03.51 | [TK]D-Fender | MENU |
| 20:04.05 | raden_work | I have a book and tech support and here all showing and telling me diffrent things im getting very confussed |
| 20:04.09 | raden_work | ok |
| 20:04.20 | raden_work | i want all inbound to goto menu |
| 20:04.23 | [TK]D-Fender | raden_work: what book? What tech support? |
| 20:04.31 | raden_work | call centric tech and oreilly book |
| 20:05.35 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 20:06.14 | raden_work | so should i change the 17772445766:pine4556@callcentric.com/101 to 17772445766:pine4556@callcentric.com/100 and make a context for 100 in sip.conf to goto from-callcentric in extensions ? |
| 20:06.29 | [TK]D-Fender | raden_work: Your IVR is very minimal, and doesn't account for timeouts, invalid input, etc. then there is the pattern match issue, the lave of "1" priorities for the other numbered extens in there, the face that your REGISTER statement is in the middle of your [general] section and CUTS IT OFF (which breaks your NAT settings and everything else below it. |
| 20:06.52 | raden_work | :( |
| 20:07.12 | raden_work | thats where call centric told me to put it |
| 20:07.22 | raden_work | where should my register statement be ? |
| 20:07.31 | [TK]D-Fender | raden_work: they are complete retards if thats their advice |
| 20:07.45 | grandpapadot | raden_work: At the very end of your [general] section, after all the settings=value |
| 20:07.59 | [TK]D-Fender | raden_work: AFTER everything else belonging in [general] and before your first peer entry |
| 20:08.18 | raden_work | ok |
| 20:08.36 | raden_work | call centric website asterisk config the thing dont even work |
| 20:08.43 | raden_work | making changes now |
| 20:09.00 | [TK]D-Fender | raden_work: exten => 101,n,goto(internal/101,1) <-- someone needs to learn to "include" contexts.... |
| 20:09.04 | raden_work | how should i be registering ? 17772445766:pine4556@callcentric.com/??? |
| 20:09.31 | [TK]D-Fender | 17772445766:pine4556@callcentric.com/17772445766 |
| 20:09.32 | raden_work | [TK]D-Fender, confussed on that one ? doing as the book says and shows |
| 20:10.04 | [TK]D-Fender | raden_work: and in your inbound context you should match taht # specifically and Goto "s" in another context to actually run your IVR |
| 20:10.24 | [TK]D-Fender | raden_work: You should never do IVR's off #'d extens. that risks it going circular and delaying dials |
| 20:11.24 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 20:14.34 | *** join/#asterisk InfoNutz (n=infonutz@204.50.209.225) |
| 20:14.36 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 20:14.45 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 20:17.18 | raden_work | [TK]D-Fender, ok lemee make some changes |
| 20:17.30 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 20:17.39 | [TK]D-Fender | TIMMEH! |
| 20:19.55 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
| 20:21.12 | raden_work | how do i make that goto s |
| 20:22.36 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:23.13 | [TK]D-Fender | raden_work: "core show application goto" |
| 20:24.10 | raden_work | im confussed on as how i route 17772445766 to s |
| 20:24.27 | [TK]D-Fender | raden_work: GOTO |
| 20:24.45 | raden_work | ill try it again |
| 20:27.13 | benklop | is there a list of the options for musiconhold modes? |
| 20:27.45 | [TK]D-Fender | benklop: in the sample config |
| 20:29.29 | benklop | [TK]D-Fender: where is that in the source? |
| 20:29.49 | [TK]D-Fender | benklop: In a lovely folder in the source tarball.. |
| 20:29.50 | benklop | my pbxinaflash box doesn't have the sample files in the /etc dir |
| 20:30.05 | *** join/#asterisk teknomega (n=cmr@75-147-97-177-Philadelphia.hfc.comcastbusiness.net) |
| 20:30.05 | [TK]D-Fender | benklop: And we don't care what they don't include here... |
| 20:30.19 | benklop | doh, ok the folder is called configs |
| 20:30.19 | teknomega | does the polycom ip 450 support the Expansion Module ? |
| 20:30.19 | [TK]D-Fender | benklop: So go download the source tarball and get reading. |
| 20:30.22 | benklop | that makes sense |
| 20:30.28 | [TK]D-Fender | teknomega: No |
| 20:30.31 | teknomega | bah |
| 20:30.39 | benklop | they actualyl have the source in /usr/src |
| 20:30.41 | teknomega | [TK]D-Fender, how about the ip550 ? |
| 20:30.42 | [TK]D-Fender | teknomega: 6XX only |
| 20:30.45 | benklop | so i can find it there |
| 20:30.51 | teknomega | [TK]D-Fender, ok |
| 20:31.40 | [TK]D-Fender | Checkout time, BBIAB |
| 20:32.06 | elitecoder | Anyone got a failover section of a dialplan I can see? |
| 20:32.37 | elitecoder | I was starting with the one on voip info but I'd feel more comfortable with one I know is in production lol |
| 20:32.55 | benklop | [TK]D-Fender: is there a way to make musiconhole listen to a channel or something? i just want to play ringtones to it |
| 20:32.59 | elitecoder | he left heh |
| 20:33.09 | benklop | dang |
| 20:33.14 | benklop | ah well |
| 20:33.28 | benklop | anyone else? |
| 20:33.48 | joako | benklop: You can use WaitMusicOnHold() |
| 20:33.50 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 20:35.00 | benklop | joako: looking it up now |
| 20:35.18 | benklop | it's been about 36 hours since i found out what asterisk was :) |
| 20:35.45 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 20:36.28 | joako | elitecoder: Here is one I have in production. I am sure I tested it but I don't think it gets used much. http://pastebin.ca/1532970 |
| 20:36.43 | elitecoder | joako: thanks I really appreciate it |
| 20:36.45 | benklop | joako: i'm actually trying to change what plays in the parkinglot, but that seems to just play whatever is the default MOH. so I need to change that to be a ringtone somehow... |
| 20:37.09 | *** join/#asterisk matt_d (n=matt@70.134.79.103) |
| 20:37.12 | matt_d | hello everyone |
| 20:39.30 | joako | benklop: Oh, and you need AnnVCA sound from CCM for that script, here: http://app5.netjdn.com/~joako/sounds/Ann-VCA.ulaw.wav |
| 20:40.08 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 20:41.10 | benklop | forbidden :) |
| 20:48.54 | *** join/#asterisk Tim_Toady (n=moi@adsl52-231.kln.forthnet.gr) |
| 20:49.33 | joako | benklop: give it another try. odd apache error and a cursed folder |
| 20:49.51 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:49.53 | *** join/#asterisk bgmarete (n=bgmarete@196.201.211.243) |
| 20:51.33 | joako | Looking at that directory reminds me everyone needs to swap out their Alison Smith sounds for these: http://www.flowroute.com/services/voice/ (IMHO) |
| 20:52.05 | benklop | joako: thanks |
| 20:52.38 | bgmarete | Hello everyone! Is there some way in Asterisk, wherein a user may call and if they hangup while being processed in the IVR, they can go back to the same extension and priority at which they where being processed? Thanks! |
| 20:54.03 | joako | benklop: for playing ringback tone for users in park the quickest (dirtiest) way I can think of is use MixMonitor to record Playtones(ringing) and make a new MOH class using just that file |
| 20:54.30 | matt_d | bgmarete if they hang up it won't matter where it goes after that... right? |
| 20:54.40 | matt_d | bgmarete becuase there will be no body there . the connection is closed. |
| 20:55.01 | bgmarete | Sure. But I wish to remember them when they come back :) |
| 20:55.07 | [TK]D-Fender | bgmarete: more or less... you'll have to keep track of their position and the way to identify them |
| 20:55.26 | matt_d | bgmarete the h extention is executed when there is a hangup. |
| 20:55.40 | matt_d | bgmarete in your dialplan, you can do the logic there :) exten > h,1,... |
| 20:56.02 | benklop | joako: that sounds like it'd be easy.. |
| 20:56.06 | bgmarete | [TK]D-Fender: I suppose I can use caller ID to identify them. But is there any simple way to identify where they were in an IVR sequence? |
| 20:56.57 | joako | bgmarete: Not stock in Asterisk, but then again I don't play much with queues and haven't used Asterisk 1.6 AT ALL. But I do seem to recall there was an add-on/hack to do that but not the name |
| 20:57.10 | bgmarete | matt_d: I suppose I should have mentioned that they 1) Hang up 2) Call back later. I want to remember them then, and place them back into where they were. |
| 20:57.27 | matt_d | bgmarete: i see.. |
| 20:57.30 | [TK]D-Fender | bgmarete: YOU have to add tracking code to all of your IVR's for this. |
| 20:57.32 | benklop | joako: thanks :) |
| 20:57.34 | matt_d | bgmarete: yes, that is possible. |
| 20:57.39 | [TK]D-Fender | bmYour dialplan does what YOU program it to. |
| 20:57.45 | [TK]D-Fender | bgmarete: Your dialplan does what YOU program it to. |
| 20:58.03 | matt_d | bgmarete: assuming you are using regular dialplan (not AGI or FastAGI) you need to set variables as the user movies along the dialplan. when they havng up save it into a database. |
| 20:58.06 | benklop | joako: sounds ugly, but a lot easier than making my own parking app |
| 20:58.33 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 20:58.41 | *** join/#asterisk YellowSub (n=yellowsu@battery.securecms.com) |
| 20:59.26 | matt_d | btw, FastAGI is awsome! :) im making a Ruby FastAGI server. can't wait until im done. it will be nice to have a 3rd computer do all the database processing, etc.. (i recently started developing on asterisk again.. ive been gone for about 3 years) |
| 20:59.58 | bgmarete | matt_d, [TK]D-Fender, joako: Thanks a lot. Thought so. Just thought that there was something obvious that I was missing. Thanks a lot. |
| 21:00.11 | *** join/#asterisk ethicx (n=ethicx@host-208-88-126-198.biznesshosting.net) |
| 21:00.55 | matt_d | bgmarete: nope.. remember asterisk does nothing for you automaticly. you have to program it though the dialplan.. just set variables along the way and save it to a file or database. then when the exten calls back, look that data back up and throw them to the right place in the dial plan. sounds like a lot, but it won't take too much code. |
| 21:01.37 | *** join/#asterisk KazaLite (n=KazaLite@94-193-98-124.zone7.bethere.co.uk) |
| 21:02.25 | ethicx | Hello all. Can please someone shed light... I have users complaining of call audio being laggy? and I get the following error:WARNING[16847]: rtp.c:891 ast_rtcp_read: RTCP Read too short anyone ? |
| 21:02.25 | joako | bgmarete: I think this is it, I seem to vaguely recall it having a catchy name, but it's not free: http://www.voip-info.org/wiki/view/OrderlyQ |
| 21:02.29 | *** join/#asterisk clintc (n=clintc@n128-227-5-3.xlate.ufl.edu) |
| 21:03.33 | bgmarete | joako: Thanks! Had actually looked at that, but seen that it is not free. Thanks though. |
| 21:03.36 | matt_d | ethicx : reading ... |
| 21:03.45 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 21:04.02 | joako | But then again I ask people would you rather buy a $25k avaya license and run your business the way they think it should work or would you rather spend that money on having a system programmed to your workflow? And in the end the Asterisk solution is usually cheaper, as an added benefit |
| 21:04.33 | bgmarete | matt_d: Thanks. I have worked a fair bit with Asterisk, just thought there was a module or app out there that would help me do this. |
| 21:04.37 | matt_d | ethicx: what version? |
| 21:05.33 | benklop | joako: what is that sound you sent me? |
| 21:05.34 | ethicx | matt_d: 1.4.26 |
| 21:06.48 | joako | benklop: used in the failover macro I posted... |
| 21:07.12 | matt_d | ethicx I checked bugzilla and it looks like this has been reported ... |
| 21:07.56 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 21:08.11 | ethicx | matt_d: I have a site to site vpn with the remote office, and when I do a sip show peers the (ms) stats for that remote peer fluctuates from 1024 to 623 ms which is crazy |
| 21:08.19 | benklop | joako: okay, thanks |
| 21:08.22 | *** join/#asterisk rossand (n=rossand@CPE0018f8360b13-CM0013711c94ee.cpe.net.cable.rogers.com) |
| 21:08.23 | matt_d | ethicx I dont see any info.. however. RTCP doesn't carry the voice data.. so it shouldn't make it delayed. I would assume you could get that warning if there was some network problems. |
| 21:08.38 | matt_d | as the server expected a proper RTCP packet, but didn't get a proper one. |
| 21:08.55 | ethicx | matt_d: but the interesting thing is that the remote office hears me good... dunno =( |
| 21:09.42 | ethicx | matt_d: I'm starting to think could be a vpn connection issue. |
| 21:09.50 | matt_d | ethicx: weird .. asterisk is full of annoying problems :) im still trying to fix ChanSpy/ExtenSpy's laggy whisper features. i have been at it for 3 weeks. |
| 21:10.11 | matt_d | ethicx: it might. the RTCP packets might not be getting though ? |
| 21:10.56 | ethicx | matt_d: damn, yeah the asterisk sys admin for my company left today so I kind of temporarly took over for him but damn!!! |
| 21:11.46 | joako | ethicx: perhaps a limitation with upstream bandwidth or MTU issue on the VPN? I know when I get calls where the remote end's bandwidth is saturated they can hear me fine but I can sometimes not make out what they are saying |
| 21:12.39 | *** join/#asterisk sjobeck (n=Adium@63.227.212.111) |
| 21:12.52 | joako | * remote end's _upstream_ bandwidth, just to be clear |
| 21:12.55 | *** part/#asterisk sjobeck (n=Adium@63.227.212.111) |
| 21:13.48 | ethicx | joako: yeah that's probably what it is. because I have two other remote sites with no lag or voice quality problems |
| 21:14.20 | ethicx | joako: using exactly the same setup. |
| 21:14.26 | elitecoder | bbl |
| 21:14.27 | *** part/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 21:14.39 | joako | ethicxs: Thanks for proactivaly answering my next question |
| 21:15.19 | joako | See if you can pull some bandwidth graphs for that VPN link... |
| 21:16.05 | ethicx | joako: mrtg? or you have other suggestions? |
| 21:16.32 | WHYS | helo |
| 21:16.38 | WHYS | <PROTECTED> |
| 21:17.12 | joako | That's possible, but maybe the device has a pretty web interface already? The old Cisco PIX have them and so does my $50 4-year-old linksys router at home, so I assume everything should ) |
| 21:20.48 | shido6 | ooh PIXie dust |
| 21:25.02 | *** join/#asterisk servilio (n=servilio@unaffiliated/servilio-ap) |
| 21:30.18 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 21:30.24 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 21:30.24 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 21:35.40 | WHYS | Any timeout settings for peers in sip.conf? |
| 21:36.01 | [TK]D-Fender | WHYS: qualify <- |
| 21:36.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:37.17 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:37.21 | WHYS | I have qualify=no, but it still hangs if the peer does not respond. |
| 21:38.15 | WHYS | Ahhh, let me try the timeout.... hum..... |
| 21:44.05 | *** join/#asterisk xpot-mobile (n=james@66-182-88-85.static.sdyl010.digis.net) |
| 21:46.58 | joako | WHYS: it hangs as in the asterisk software becomes unresponsive and you have to kill the process, or calls hangs when you attempt them? |
| 21:49.18 | shido6 | what hardware are you running it on? |
| 21:50.32 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:51.10 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096773681.dsl.bell.ca) |
| 21:51.29 | fnord0 | question, I am looking to run asterisk to use with PSTN gateway service - i.e. broadvoice.com, I found a great tutorial - and have multiple asterisk bootCDs for bakup. |
| 21:51.34 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:52.02 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 21:52.42 | ayeso | if i want a catch-all in my dialplan, in other words a match ALL pattern, what should I sue? |
| 21:52.46 | WHYS | shido6: running on vmx server. |
| 21:52.49 | ayeso | s/sue/use |
| 21:53.03 | shido6 | shudders |
| 21:53.07 | JimVanM | do you want to match all characters, or only numeric? |
| 21:53.24 | fnord0 | I am noticing that the signup page says "BroadVoice does not allow the use of "One Time Use" or "Disposable" Credit Cards. Strict penalties may be incurred for the use of these cards per the BroadVoice Terms and Conditions." .. wtf is that all about? I even noticed magicjack.com says this too "If you falsely identify your "Debit Card" or "Gift Card" as a "Credit Card" you will be charged today or your order might not even process." |
| 21:53.29 | *** join/#asterisk Diffen (n=diffen@c-107fe555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 21:53.34 | WHYS | joako: * server hangs - management interface is unresponsive. |
| 21:53.35 | Diffen | evning |
| 21:53.35 | ayeso | JimVanM: I want it where all calls get sent to a different peer no matter what the dialed number is |
| 21:54.07 | JimVanM | understood, but asterisk supports characters in dialplan as well, so, for example, myphone is as valid an extension as 1234 is |
| 21:54.11 | fnord0 | all I have is a pre-paid credit card, does anyone know of a PSTN gateway service that allows such? |
| 21:54.34 | *** join/#asterisk sjobeck (n=Adium@207.173.77.239) |
| 21:54.41 | fnord0 | ... that works with asterisk? |
| 21:54.43 | ayeso | JimVanM: I understand, lets say only numerics |
| 21:54.57 | JimVanM | ayeso: if you just want to be able to handle digits (i.e. typical phone numbers), the extension that'll do the job for you is _X. |
| 21:55.06 | ayeso | JimVanM: thanks |
| 21:55.07 | [TK]D-Fender | fnord0: prepare to do a bank transfer or such. |
| 21:55.22 | JimVanM | the '_' makes it a pattern match, the 'X' makes sure it's numeric, and the '.' allows it to be any length of one or greater |
| 21:55.28 | raden_work | what codecs is best to use ? |
| 21:55.34 | [TK]D-Fender | JimVanM: CLOSE |
| 21:55.43 | JimVanM | so exten => _X.,1,Verbose(1,And so on) |
| 21:55.47 | [TK]D-Fender | raden_work: Best for what? |
| 21:55.56 | raden_work | quality |
| 21:56.05 | [TK]D-Fender | raden_work: G.722 |
| 21:56.38 | JimVanM | hey anybody know if I can use the CHANNEL() function to get the number of a DAHDI channel? I don't want the 'DAHDI/', nor the '-whatever' |
| 21:56.54 | raden_work | [TK]D-Fender, provider doesnt have that on there supported list |
| 21:56.55 | JimVanM | the example in the CLI only shows egs for iax and sip |
| 21:57.03 | *** part/#asterisk servilio (n=servilio@unaffiliated/servilio-ap) |
| 21:57.04 | raden_work | ok i lied they do |
| 21:57.06 | JimVanM | and I can't find anything in the docs for chan_dahdi |
| 21:57.22 | joako | fnord0: Do they define "disposable" -- you can venture to say that a "refillable" card is not "disposable" and you can say someone who has a "real"card who is burried in debt and is going to file for bankrupcy is practically "disposable" |
| 21:57.23 | [TK]D-Fender | raden_work: Well you keep asking questions and THEN applying conditions to them |
| 21:57.24 | raden_work | [TK]D-Fender, bandwith usage and quality what would u use ? |
| 21:57.35 | Diffen | I wonder if its ok to ask a non asterisk question but one sip question regarding forking and rfc3264 |
| 21:57.37 | [TK]D-Fender | raden_work: Can't have both. |
| 21:57.54 | raden_work | 722 better than 711 ? |
| 21:58.03 | shido6 | what the fork do you want to ask Diffen ? |
| 21:58.04 | [TK]D-Fender | JimVanM: "core show function CUT" |
| 21:58.09 | [TK]D-Fender | raden_work: By 11! |
| 21:58.10 | joako | raden_work: Yes but when you call the PSTN what good is G722? |
| 21:58.14 | *** part/#asterisk YellowSub (n=yellowsu@battery.securecms.com) |
| 21:58.29 | JimVanM | [TK]D-Fender: yeah yeah, I know all about CUT, but that is NOT pretty |
| 21:58.42 | JimVanM | [TK]D-Fender: that was what I was trying to avoid |
| 21:58.51 | JimVanM | [TK]D-Fender: but I'll do it if I have to |
| 21:58.58 | [TK]D-Fender | JimVanM: * = "not pretty" :) |
| 21:59.18 | Diffen | shido6 well there are two questions. one is regarding forking (dont know if that is something that is a sip standard) and the other one is regarding early media and change of the sdp header in 183, 180 and 200 |
| 21:59.29 | [TK]D-Fender | JimVanM: Or you could set a context per channel and create a var at source, etc |
| 21:59.31 | JimVanM | [TK]D-Fender: perhaps not, but damn it's fun. Also, ever tried to program a Nortel M1? Trust me, Asterisk is pretty. |
| 21:59.39 | raden_work | will gsm not work with ptsn ? |
| 21:59.39 | [TK]D-Fender | JimVanM: Still not technically "pretty" |
| 21:59.47 | fnord0 | joako, I cant find anywhere that says such on broadvoice.com that distigishes such, nor magicjack... I bought a pre-paid credit card that I get my paycheck direct deposit ya know, so I swear its a legit card.. just wondering if u/someone might suggest a good asterisk PSTN gateway take may take such a creditcard |
| 21:59.58 | JimVanM | [TK]D-Fender: OK, maybe 3 or 4 beers pretty |
| 22:00.15 | [TK]D-Fender | fnord0: Stop swearing, there are children in here! |
| 22:00.22 | JimVanM | [TK]D-Fender: plus a shot of Tequila, cause, Damn! |
| 22:00.32 | [TK]D-Fender | JimVanM: You'll need a beer TELESCOPE |
| 22:01.23 | JimVanM | [TK]D-Fender: it's all a matter of perspective |
| 22:01.44 | [TK]D-Fender | JimVanM: Yup, thats what the insane focal length is for :p |
| 22:02.01 | [TK]D-Fender | grabs his magnifying glass and burns some more ants |
| 22:02.16 | joako | fnord0: There are (I _think_) 2 issues with prepaid cards: 1) AVS, some cards have issues to make online purchases because they don't have your address and the merchant can't verify it VoIP is a pretty risky business in terms of fraud so that's certainly a factor, so without a doubt you will need a card that can register your address with and can be verified with AVS 2) I _think_ merchants get charged a higher rate for pre-paid/reloadable (I beli |
| 22:02.16 | joako | eve the term is "stored value") cards... but then again so are business cards |
| 22:04.00 | JimVanM | has to use CUT() . . . [shudder] |
| 22:04.18 | raden_work | how do i set verbose :( |
| 22:04.24 | [TK]D-Fender | JimVanM: Really not that ugly.... |
| 22:04.47 | [TK]D-Fender | JimVanM: You know... there is this wonderful * book... you should take a peek at it some time ;) |
| 22:04.54 | joako | JimVanM: I wrote an AGI bash script to run sed on a SIP header... it works... |
| 22:04.56 | JimVanM | [TK]D-Fender: Asterisk, or CUT()? |
| 22:05.05 | [TK]D-Fender | JimVanM: YES :p |
| 22:05.32 | JimVanM | [TK]D-Fender: oh I know all about that bloody book, I'm just too lazy |
| 22:05.48 | [TK]D-Fender | JimVanM: Congratulations.... 11 steps to go! |
| 22:06.35 | *** join/#asterisk rossand (n=rossand@CPE0018f8360b13-CM0013711c94ee.cpe.net.cable.rogers.com) |
| 22:07.56 | alunca | when I use cat /etc/asterisk/extensions_custom.conf I have a different values than using vi /etc/asterisk/extensions_custom.conf |
| 22:08.36 | leifmadsen | probably because you have a .extensions_custom.conf.swp file |
| 22:08.59 | alunca | leifmadsen how can I fix this? |
| 22:09.23 | leifmadsen | remove the swp file |
| 22:09.33 | leifmadsen | (or move it for a backup, or whatever) |
| 22:09.51 | alunca | is it located in the same loc? |
| 22:10.12 | leifmadsen | ls -la * |
| 22:10.51 | Pan3D | alunca: a "swap" file is a file that gets created during the editing process. Often, they remain on the disk when the editing process crashes, or intentionally under some circumstances. You'll want to use "ls -all" to see them -- in the directory where the original is. |
| 22:12.01 | alunca | I don't see .swp but I have .sample ... which should not be the swap file |
| 22:12.03 | Pan3D | If you attempt to open the original by hand, your editing application (vi, etc.) will likely warn you of the swap file's existence, and ask what you want to do moving forward. |
| 22:12.25 | raden_work | my phones and provider both support g.729 what do i have todo to use asterisk with g.729 and is it worth it compared to g.711 |
| 22:12.36 | alunca | i see, I can do edit with vi command without any warraning ... |
| 22:13.03 | Pan3D | and you don't see it in the directory? |
| 22:13.28 | Pan3D | then something else is going on |
| 22:13.31 | alunca | and I tried to delete a line and saved it with vi, but the cat command still show the original one |
| 22:14.27 | Pan3D | if you re-open with vi, is the change there? |
| 22:14.28 | citywok | so i have a somewhat technical question. maybe tk you can help. i've got my dialplan, i originate the call from a perl script / telnet in, and i use exten => _1NXXNXXXXXX,n,SIPAddHeader(Call-Info:<sip:>\;answer-after=0) in order to force the sip phone i'm originating the call from to auto-answer the call. problem is, the auto-answer sip header doesnt get transmitted (i checked using... |
| 22:14.28 | citywok | ...wiretrace). any ideas? maybe the auto-answer is getting sent down the calling sie of the line, not the receiving side? |
| 22:15.14 | [TK]D-Fender | citywok: Here's an idea : Show us the complete originate, CLI output with SIP DEBUG enabeld... |
| 22:17.35 | alunca | Pan3D i just reboot ... to see if it is okay |
| 22:17.56 | Pan3D | k, I gotta go. good luck. |
| 22:18.03 | alunca | thanks |
| 22:21.14 | raden_work | [Aug 17 17:19:27] ERROR[6045]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. |
| 22:21.23 | raden_work | while trying to transfer a call internally |
| 22:22.44 | *** join/#asterisk matt_d (n=matt@70.134.79.103) |
| 22:22.51 | matt_d | darn laptop feel asleep.. hello everyone |
| 22:24.42 | citywok | ok here's the originate perl http://pastebin.com/m415dd7c4 |
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| 22:31.08 | joako | Can anyone explain why I see the time in /var/log/asterisk/cdr-csv/Master.csv ahead of the actual time on the system by a few hours? |
| 22:32.16 | *** join/#asterisk henry_ (n=henry@ip68-109-187-162.ph.ph.cox.net) |
| 22:33.21 | citywok | here is the sip debug http://pastebin.com/m1100bedf |
| 22:34.59 | henry_ | I am having an issue creating multiple calls using call files and playing a message. The first call issued works fine, the others just seem to hang up after they are answered. Has anyone seen a similar problem? |
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| 22:42.30 | [TK]D-Fender | citywok: I don't see the entire call attempt in there |
| 22:43.32 | citywok | i didn't hit accept on the incoming call, i figured it wasn't important because the failure occurs the instant that i get the accept call button. it sholdn't do that. |
| 22:45.04 | citywok | i think the problem is that starting at line 23 is the initiate sent to the client on my computer, but it doesnt send the sip: autoanswe.r my thought is that it doesn't do that because the autoanswer in the dialplan is being applied to the other leg of the call |
| 22:45.24 | citywok | i guess that is why it is important to see the entire call aattempt, i would probably then see that in the second leg of the call. |
| 22:45.28 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-169-241.dsl.stlsmo.sbcglobal.net) |
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| 22:47.40 | alunca | sigh, my booting process is too slow now; especialy stuck at sendmail |
| 22:47.43 | LemensTS | if i do an originate between a local channel (php script) and an outgoing call to an itsp in g729 format, and the php script READS a g729 recorded file, is it doing decoding? |
| 22:48.54 | grandpapadot | alunca: Check to make sure you have an entry for your server's hostname in /etc/hosts |
| 22:49.02 | grandpapadot | alunca: pointing to 127.0.0.1 |
| 22:49.29 | [TK]D-Fender | LemensTS: No. |
| 22:50.36 | alunca | grandpapadot I did that, it is localhost 127.0.0.1 |
| 22:51.51 | alunca | dang, ok now. I pulled out the power cable, and plug it back, must faster now lol (don't know why) |
| 22:52.01 | grandpapadot | alunca: What's in /etc/hostname? Make sure *that* name is in your /etc/hosts and pointing to 127.0.0.1 |
| 22:53.46 | alunca | . /etc/hostname ?? << don't have it |
| 22:54.23 | alunca | grandpapadot should I make one hostname in /etc/ ? |
| 22:54.52 | grandpapadot | No, what disto? |
| 22:55.06 | alunca | grandpapadot ? |
| 22:55.50 | manxpower | what version of linux do you have alunca |
| 22:56.07 | manxpower | if you don't have a /etc/hostname then your install is SERIOUSLY screwed up. |
| 22:56.14 | alunca | centos 5 |
| 22:56.29 | manxpower | yup, your install is seriously screwed up |
| 22:57.00 | grandpapadot | alunca: What linux distribution are you running? (debian, ubuntu, redhat, centos, etc) |
| 22:57.24 | grandpapadot | alunca: You should have a /etc/hostname file that contains the name of your system. |
| 22:57.40 | grandpapadot | alunca: type "hostname" and tell me what it outputs |
| 22:58.00 | alunca | asterisk |
| 22:58.35 | alunca | typed hostname = asterisk |
| 22:58.43 | grandpapadot | alunca: and if you type "sudo cat /etc/hostname" or just "cat /etc/hostname" it outputs? |
| 22:59.05 | alunca | no such file or directory |
| 22:59.19 | manxpower | alunca: is reinstalling centos an option? |
| 22:59.55 | alunca | manxpower I think it will be same thing since this is asteriskNow (build in with centos) |
| 23:00.05 | grandpapadot | alunca: "No such file or directory"??? |
| 23:00.10 | grandpapadot | alunca: What shell are you running? |
| 23:00.41 | alunca | which command to find out which shell am I using? |
| 23:00.41 | grandpapadot | Ah, cat says it. |
| 23:02.15 | alunca | sorry, i don't know hehe |
| 23:02.51 | grandpapadot | alunca: export |
| 23:03.01 | grandpapadot | alunca: Look for the line that says "SHELL=" |
| 23:04.31 | alunca | "/bin/bash" |
| 23:05.30 | LemensTS | TKD-Fender: I must be doing something wrong than. I got 2 g729 licenses and on the 3rd simultaneous originate it gives me decoder license errors. This is the correct way to record g729 right: exten => s,n,Record(outbound/${CUSTID}:g729) |
| 23:05.34 | grandpapadot | alunca: it's odd to me that you don't have an /etc/hostname file but I don't run CentOS, there might be a good reason for your install. Might want to go ask in #centos or something. |
| 23:07.55 | [TK]D-Fender | LemensTS: Sure looks like. |
| 23:08.00 | LemensTS | TKD-Fender: and READing it by $agi->exec('Read', 'keypress,outbound/$CUSTID,1,,1,1'); |
| 23:08.06 | [TK]D-Fender | LemensTS: But of course that line doesn't say what the problem is |
| 23:16.10 | box2 | ~book |
| 23:16.11 | 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 |
| 23:16.15 | box2 | ahh there it is |
| 23:16.18 | box2 | <3 infobot |
| 23:17.23 | [TK]D-Fender | MY BITCH! |
| 23:17.44 | [TK]D-Fender | ~areyouadog ? |
| 23:17.44 | infobot | Bark! Bark! |
| 23:17.46 | [TK]D-Fender | ~botsnack |
| 23:17.46 | infobot | thanks, [TK]D-Fender |
| 23:18.17 | box2 | heh |
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