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| 01:25.17 | phix | heygang |
| 01:25.25 | mitchrodrigues | oi |
| 01:25.30 | mitchrodrigues | i hate comcast so bad |
| 01:25.31 | mitchrodrigues | -.- |
| 01:25.50 | phix | What is a good echo canceller ? |
| 01:25.54 | phix | software one that is |
| 01:26.20 | phix | I am getting echo, turning down the volume doesn't help, I have oslesc enable I think |
| 01:27.17 | phix | My card is a TDM401something |
| 01:32.30 | ChannelZ | have you run through the fxotune process first? |
| 01:35.31 | phix | What's that? |
| 01:36.01 | ChannelZ | Well I assume you are talking about a TDM card with FXO ports, that you're getting echo on POTS calls |
| 01:40.09 | ChannelZ | yes? no? |
| 01:45.22 | ChannelZ | shrugs |
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| 01:46.12 | pensmit | Looks like The Definitive Guide 4th edition is incorrect |
| 01:46.27 | pensmit | I'm trying to install it but it wants you to download dahdi twice? |
| 01:46.48 | WIMPy | Who wants to download dahdi? |
| 01:47.02 | WIMPy | (if not you yourself manually) |
| 01:47.47 | pensmit | huh? |
| 01:47.57 | ChannelZ | There are two separate DAHDI archives |
| 01:48.02 | ChannelZ | the drivers, and the tools |
| 01:48.13 | ChannelZ | Perhaps you are not seeing the differentiation |
| 01:48.43 | pensmit | I'm just following instructions |
| 01:48.57 | pensmit | They seem to be buggy though |
| 01:49.07 | WIMPy | Do you have any dahdi hardware? |
| 01:49.47 | pensmit | $ wget \ |
| 01:49.47 | pensmit | http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/\ |
| 01:49.47 | pensmit | dahdi-linux-complete-current.tar.gz |
| 01:49.47 | pensmit | $ tar zxvf dahdi-linux-complete.tar.gz |
| 01:49.52 | pensmit | says to do that |
| 01:49.54 | pensmit | then |
| 01:50.22 | pensmit | $ cd ~/src/asterisk-complete/ |
| 01:50.22 | pensmit | $ mkdir dahdi |
| 01:50.22 | pensmit | $ cd dahdi/ |
| 01:50.22 | pensmit | $ svn co http://svn.asterisk.org/svn/dahdi/linux-complete/tags/2.6.1+2.6.1 |
| 01:50.22 | pensmit | $ cd 2.6.1+2.6.1 |
| 01:50.29 | pensmit | that |
| 01:50.36 | WIMPy | If you don't have any telephony cards, you can skip the whole dahdi thing. |
| 01:50.41 | WIMPy | ~pb |
| 01:50.41 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
| 01:50.42 | pensmit | then go to tools |
| 01:50.52 | pensmit | configure make etc |
| 01:51.12 | pensmit | meetme requires dahdi |
| 01:51.24 | pensmit | from what I've read |
| 01:51.55 | WIMPy | Correct, but we have ConfBridge now. |
| 01:52.20 | pensmit | configure: *** Building this package requires DAHDI support. *** |
| 01:52.20 | pensmit | configure: *** Please install the dahdi-linux package. *** |
| 01:52.34 | pensmit | Isn't that what I'm installing |
| 01:52.35 | pensmit | lol |
| 01:53.23 | pensmit | confbridge doesn't require dahdi at all? |
| 01:53.31 | WIMPy | noi |
| 01:53.34 | pensmit | cool |
| 01:53.38 | WIMPy | no |
| 01:53.52 | pensmit | I still would like to get this compiled to work through certain examples |
| 01:53.59 | pensmit | am I missing something here |
| 01:54.53 | ChannelZ | you don't need to svn dahdi, just download the tarball as per the first step and build it |
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| 01:57.55 | pensmit | I did |
| 01:58.00 | pensmit | that's when I got the error |
| 01:58.04 | pensmit | after the tarball |
| 01:58.10 | pensmit | then the tools section make |
| 01:58.14 | pensmit | maybe it just does that |
| 01:58.18 | pensmit | in that order |
| 01:58.24 | pensmit | since there is another make after that |
| 01:58.26 | ChannelZ | you need to build the drivers first |
| 01:58.36 | WIMPy | You can't build the tolls before the drivers. |
| 01:58.43 | WIMPy | tools |
| 01:58.44 | pensmit | tell that to the book |
| 01:58.45 | ChannelZ | and install.. which throws some headers on your system so the tools can build |
| 01:59.19 | pensmit | ok so reverse those instructions |
| 01:59.21 | pensmit | go into linux |
| 01:59.23 | pensmit | do a make |
| 01:59.34 | WIMPy | I told you that you probably want to skip the whole step. What more do you want? |
| 01:59.44 | ChannelZ | to complain about the book |
| 01:59.53 | pensmit | no |
| 01:59.57 | pensmit | just trying to get that working |
| 02:00.06 | pensmit | so i understand if i need it |
| 02:00.06 | WIMPy | Erl, you can 'make' in the top dir, can't you? |
| 02:00.34 | pensmit | so here's what I'm going to try |
| 02:00.38 | WIMPy | Yu can also go and install spandsp and hylafax to understand. What's the point if you don't need it? |
| 02:01.12 | pensmit | dude I'm just trying to follow the book and examples |
| 02:01.15 | pensmit | gimme a break man |
| 02:02.05 | WIMPy | YIf you just copy and paste the book, you're not going to get more than you can read from the book anyway. |
| 02:03.11 | pensmit | Ok I'm going to do a make in the linux directory then do a make install |
| 02:03.22 | ChannelZ | This is one reason why I never loved the book, it actually tends to tell you too much and sometimes not enough of 'why am I doing this' |
| 02:03.25 | pensmit | then go back to the tools directory do a make and make install |
| 02:03.39 | pensmit | then back to linux and do a make config |
| 02:03.41 | WIMPy | That's what all books do, don't they? |
| 02:04.05 | WIMPy | Preferrably they tell you what you already knew while leaving your questions unanswered. |
| 02:04.58 | WIMPy | Although I think that book isn't bad. Just some sections like this one are suboptimal. |
| 02:05.03 | ChannelZ | pensmit: yes do that |
| 02:05.19 | WIMPy | It should clearlty mention that installing dahdi is optional now. |
| 02:05.51 | pensmit | just got one warning |
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| 02:05.57 | pensmit | WARNING: could not find /home/asteriskpbx/src/asterisk-complete/asterisk/dahdi-linux-complete-2.7.0+2.7.0/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /home/asteriskpbx/src/asterisk-complete/asterisk/dahdi-linux-complete-2.7.0+2.7.0/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o |
| 02:06.09 | WIMPy | And even if you have hardware, doesn;t mean you can use dahdi. |
| 02:07.46 | pensmit | is confbridge a lot better than meetme? |
| 02:08.00 | pensmit | if you run freepbx on top of it does it require dahdi? |
| 02:08.22 | ChannelZ | ConfBridge is arguably better yes |
| 02:08.45 | ChannelZ | dunno about FBX. Hopefully they've moved off meetme but who knows |
| 02:08.50 | pensmit | Would you say confbridge supports more simultaneous connections |
| 02:08.51 | WIMPy | Ask the guys who made FreePBX. |
| 02:09.31 | ChannelZ | It's newer, more modern, and doesn't depend on DAHDI. |
| 02:09.50 | WIMPy | And supports better audio quality. |
| 02:09.54 | ChannelZ | It can handle better than 8khz audio (which MeetMe was limited to) if you are doing wideband; It can do silence suppression.. |
| 02:10.02 | WIMPy | And the noise cancellation is really nice as well. |
| 02:10.13 | pensmit | noise cancellation i may need |
| 02:10.16 | ChannelZ | or silence detection rather, noise suppression :) |
| 02:10.42 | pensmit | thanks a lot guys |
| 02:10.44 | ChannelZ | it will mute 'silent' participants so line noise doesn't simply build up between everyone |
| 02:10.56 | pensmit | nice |
| 02:11.07 | WIMPy | And you can build custom menus in ConfBridge. |
| 02:11.20 | pensmit | I think stupid freepbx still uses meetme |
| 02:11.25 | ChannelZ | MeetMe was dependent on DAHDI for audio mixing and timing. |
| 02:11.34 | ChannelZ | Then don't use stupid freepbx because it's stupid anyway |
| 02:11.42 | pensmit | well |
| 02:12.01 | pensmit | boss wants it and I'm probably not good enough yet to do just asterisk |
| 02:12.03 | pensmit | dialplan stuff |
| 02:12.11 | pensmit | stupid boss |
| 02:12.18 | ChannelZ | well good luck then |
| 02:12.24 | pensmit | yeah |
| 02:12.46 | WIMPy | You need to move to #freepbx soon then. |
| 02:13.13 | ChannelZ | and in which case you can pretty much throw the book out the window |
| 02:13.25 | WIMPy | yeah |
| 02:13.29 | pensmit | Well I'm going to convince him to drop it once I get better |
| 02:13.38 | WIMPy | And your brain :-) |
| 02:14.00 | pensmit | Do you guys do consulting? |
| 02:14.21 | pensmit | No any good companies that are really good at plain asterisk? |
| 02:14.24 | WIMPy | Sometimes. |
| 02:14.25 | pensmit | know |
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| 02:15.55 | pensmit | Can I get your contact info and rates? |
| 02:17.15 | WIMPy | You should ask on the Asterisk-biz mailing list. But check who you're dealing with. |
| 02:18.37 | pensmit | thanks |
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| 02:31.01 | pensmit | What do you guys think of the digium phones? |
| 02:31.19 | WIMPy | Not bad for SIP phones. |
| 02:31.54 | pensmit | Bare Asterisk can do multiple parking lots unlike freepbx right? |
| 02:31.55 | WIMPy | The user interface is a little unusual in some aspects, however. |
| 02:32.05 | WIMPy | yes |
| 02:32.31 | pensmit | WIMPY are you a freelance consultant or do you work for someone? |
| 02:32.34 | pensmit | out of curiousity |
| 02:32.50 | WIMPy | I do whatever comes by. |
| 02:33.15 | pensmit | Do you think you're pretty damn good? |
| 02:33.22 | pensmit | Not being offensive |
| 02:33.48 | pensmit | How would you rate your skills compared to the best you've seen in here? |
| 02:33.48 | WIMPy | The call handling on the Digium phones is the best I've seen for SIP phoes so far. *IF* you use only one account. If you use more than one it's just as bad as most others. |
| 02:34.12 | WIMPy | That depends heavily on the topic. |
| 02:34.31 | pensmit | mostly conferences with under 100 people in them each |
| 02:34.34 | WIMPy | I had to dig rather deep in to some parts while I haven't touched others at all |
| 02:34.38 | WIMPy | . |
| 02:35.17 | WIMPy | Up to 100 conference participants sounds... interesting. |
| 02:35.24 | pensmit | very |
| 02:35.29 | pensmit | lol |
| 02:36.39 | pensmit | leave me a generic email or something for anonymity and I'll get back with you |
| 02:37.17 | WIMPy | You can find me here. |
| 02:37.31 | WIMPy | But I don't usually do Asterisk stuff commercially. |
| 02:37.34 | pensmit | lol...sounds good |
| 02:37.36 | pensmit | ok |
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| 03:46.37 | AliRezaTaleghani | can I bridge more that two channel? |
| 03:46.57 | WIMPy | ConfBridge |
| 03:47.00 | AliRezaTaleghani | some thing like a three way conferance |
| 03:47.50 | AliRezaTaleghani | is it bossible when a call is ongoing? for example to dial a code and then make a call to second one and bridge all them to each other? |
| 03:48.50 | WIMPy | The easy way is to just transfer both calls to a conference room. |
| 03:49.28 | WIMPy | I'm not sure you can fully automate it for both calls without using AMI. |
| 03:50.24 | WIMPy | "Features" might make parts of it automatic. |
| 03:52.11 | AliRezaTaleghani | WIMPy: :-/ thanks... |
| 03:52.29 | AliRezaTaleghani | seems a complex way.. |
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| 04:22.55 | larrymo | if a user presses 1 can I have it execute a file on the web (say a php script) |
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| 04:31.18 | mitchrodrigues | You could build an agi process |
| 04:32.02 | mitchrodrigues | just a quick google |
| 04:32.04 | mitchrodrigues | and i get this |
| 04:32.04 | mitchrodrigues | http://www.voip-info.org/wiki/view/Asterisk+AGI+php |
| 04:42.08 | larrymo | thats not what I really mean |
| 04:43.32 | WIMPy | You want curl? |
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| 04:45.18 | larrymo | no, I am new to this, I was thinking I could create a call dynamically and just have within the IVR part of the call the ability to run a execute a script if a number is pressed. |
| 04:45.34 | larrymo | I really haven't thought this out much, I'm going to run in to road blocks Im sure |
| 04:46.01 | mitchrodrigues | we use fastagi |
| 04:46.06 | mitchrodrigues | which is a remote application |
| 04:46.10 | mitchrodrigues | that integrates with our website |
| 04:46.18 | WIMPy | Yes, that's what you can do with AGi, or just System, if you don;t want to return anything. |
| 04:49.58 | larrymo | I just got ubuntu, in order to do SIP calls I need to install SIP capability? |
| 04:50.18 | larrymo | is that what DAHDI is? |
| 04:50.47 | larrymo | nvm thats a driver for card |
| 04:51.45 | larrymo | Users who want a quick start with VoIP are encouraged to start by installing a SIP proxy before installing a soft-PBX. |
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| 05:43.51 | larrymo | I installed asterisk, now its time to configure it. Any official tutorials for this? |
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| 07:37.33 | ChannelZ | Hmmm. Is there a way to get 'automon' (features.conf) to not mix the audio at the end of the call, or to do split-channel so the streams are separate? |
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| 07:51.45 | phix | ChannelZ: is there a way to start recording current phone convo that actually works? |
| 07:51.51 | phix | I have never been able to get it working |
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| 07:54.30 | ChannelZ | automon works. I just won't want it to mix when it's done. Doing a custom applicationmap now, but am wonering how to pass multiple arguments to an app. Not documented, of course. |
| 08:00.28 | ChannelZ | Ah. That worked. Syntax is fuzzy |
| 08:00.38 | ChannelZ | or the explanation is rather |
| 08:01.50 | ChannelZ | as for recordng not working phix, what have you tried? Normal features.conf automon/automixmon? |
| 08:02.28 | ChannelZ | Keep in mind you need to use the w/W and/or x/X options on your Dial |
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| 09:12.43 | EmleyMoor | In sip.conf, udpbindaddr, how do I specify "all IPv4 addresses and specific IPv6 addresses"? |
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| 09:14.17 | EmleyMoor | (can I use a comma between values, say?) |
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| 09:19.05 | EmleyMoor | (or do I need more than one line?) |
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| 09:30.29 | EmleyMoor | Ah, it's "all or one" at present - oh well |
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| 11:18.24 | CeBe | EmleyMoor: shouldn't be too hard to specify all adresses for ip4 |
| 11:22.04 | EmleyMoor | CeBe: But it's not possible to specify precisely what I seek at present. |
| 11:22.29 | CeBe | Yes, I know but I wonder why you need it this way |
| 11:23.23 | EmleyMoor | CeBe: Part of my plan to make functions of a system "separable" in the future. |
| 11:24.00 | CeBe | why not listen on all adresses? |
| 11:24.44 | EmleyMoor | CeBe: Partly, security. Also, if I make the function "separable", I can move its specific address with the function. |
| 11:26.14 | CeBe | EmleyMoor: So if you listen on all adresses you can have multiple adresses routed to the server and route them elsewhere when you separate functions |
| 11:26.23 | CeBe | Whats your point in security here? |
| 11:26.57 | CeBe | When you have a firewall (which you should have when worried about security) you can define allowed adresses there |
| 11:28.48 | EmleyMoor | The limited nature of my current firewall (in that, if modified too much, it breaks!) is my main reason... though I plan to replace it with something better. The other point is that I intend my internal users (such as they are) to use supported addresses only. |
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| 11:30.41 | larrymo | now that I have asterisk installed, is their a good tutorial on what to do? |
| 11:31.27 | CeBe | larrymo: depends on what you want to do ;) |
| 11:32.42 | CeBe | EmleyMoor: Imo listening on all adresses and configure limitations in firewill is the best way to go here. And yeah, you should ge a firewall that works first ;) |
| 11:32.49 | larrymo | my goal at this point is to create outbound calls dynamically. I guess good practice would be building a call and placing it in the outbound directory |
| 11:33.29 | larrymo | as well as testing it to see if it's installed correctly |
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| 11:41.10 | yoavz | Hey all, I'm looking for some newbies help... I configured my voipvoip.com's sip as trunk line and it seems to work |
| 11:41.20 | yoavz | now I've tried to configure extension (6000) |
| 11:41.28 | yoavz | and when connecting I'm getting: [Aug 4 14:40:14] NOTICE[18173]: chan_sip.c:25575 handle_request_register: Registration from '<sip:6000@server.hostname.com>' failed for '1.2.3.4:5060' - No matching peer found |
| 11:41.31 | yoavz | What should I try? |
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| 12:17.42 | larrymo | this is in the example of sending an outbound call: Channel: SIP/trunkname/18882223333 |
| 12:17.46 | larrymo | what is trunkname? |
| 12:17.58 | larrymo | how do I find this out |
| 12:19.08 | [TK]D-Fender | larrymo: that is the format you'd pass to Dial() |
| 12:19.42 | larrymo | Fender, I understand but I would have to name something "trunkname" for that to work correct? |
| 12:19.45 | [TK]D-Fender | larrymo: and trunkname there is a named entry in sip.conf |
| 12:19.51 | larrymo | ahh |
| 12:19.57 | [TK]D-Fender | ~book |
| 12:19.57 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 12:19.59 | [TK]D-Fender | ^^^ |
| 12:20.11 | [TK]D-Fender | [trunkname] |
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| 13:12.42 | larrymo | In order to make outbound calls I need a SIP provider? |
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| 13:16.52 | [TK]D-Fender | larrymo: you need SOMETHING to dial out. A voip provider is an option, hardware lines, cellular interfaces are others |
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| 13:26.05 | larrymo | I'm just making calls within the United States and do not plan on accepting incoming calls. Is there a suggested free provider I go with or no such thing? |
| 13:29.32 | larrymo | I'm willing to pay per outbound call. I really just want an outbound directory that allows me to create calls on the fly and use text to speech with some basic IVR functions. |
| 13:29.55 | [TK]D-Fender | nothing free left that I'm aware of. Going out is what costs |
| 13:30.13 | [TK]D-Fender | "directory"? |
| 13:30.40 | larrymo | var/spool/asterisk/outgoing/ Hosting this thing myself might be the wrong way to go. |
| 13:31.22 | [TK]D-Fender | trat is for automating call-outs |
| 13:31.28 | [TK]D-Fender | that* |
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| 13:31.49 | [TK]D-Fender | there are other API's as well |
| 13:32.14 | [TK]D-Fender | like AMI Originate, an theCLI Originate commands. |
| 13:32.29 | [TK]D-Fender | they are all effectively the same in the end |
| 13:37.05 | larrymo | I am building a system for a restaurant where they accept orders online. The order is taken on the website then sent via text to speech to the restaurant by phone. |
| 13:38.29 | [TK]D-Fender | how you get the call to them is between you and them |
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| 13:40.48 | larrymo | optimal solution for me would be not to handle the outbound calling at all and only dynamically generate the call file |
| 13:41.29 | [TK]D-Fender | well that has to get to them somehow.... |
| 13:43.06 | larrymo | is there anyone who has a server where I can be billed per outbound call but have the ability to send in the call file? |
| 13:43.48 | larrymo | (with text to speech module) |
| 13:46.28 | [TK]D-Fender | that IS a service provider <-- |
| 13:46.32 | [TK]D-Fender | ~itsp |
| 13:46.32 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
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| 13:47.40 | larrymo | ~itsplist-us |
| 13:47.40 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 13:56.25 | pabelanger | larrymo: not if you setup your own PSTN gateway |
| 13:58.41 | larrymo | k |
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| 14:04.30 | larrymo | what I need from an ITSP is an API that allows me to do outbound calls correct? |
| 14:10.17 | larrymo | Nope, it seems they just give me the SIP account |
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| 14:19.04 | Zopieux | hi |
| 14:19.34 | Zopieux | I have configured a SIP account and a context, say "incoming", for incoming calls on this account |
| 14:19.48 | Zopieux | I would like to question Asterisk, from command line, whetever this channel is busy or not |
| 14:21.28 | Zopieux | is it possible? |
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| 14:25.06 | Zopieux | seems I can parse sip show channelstats |
| 14:25.13 | Zopieux | but it seems a bit fuzy! |
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| 14:27.28 | WIMPy | sip show inuse |
| 14:35.01 | larrymo | I put a .call file in the outgoing directory but its just sitting there. How do I check it's status or errors, etc |
| 14:35.54 | WIMPy | How did you put it there? |
| 14:36.39 | [TK]D-Fender | call files have to be mv'd from the same filesystem due to locking |
| 14:36.58 | [TK]D-Fender | do not attempt to create or use copy to get it there |
| 14:37.28 | larrymo | I actually used gedit and just saved it there. Thanks for the heads up |
| 14:38.11 | WIMPy | You have a ggod chace that Asterisk tried to read it while it was stil being written. |
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| 14:46.01 | larrymo | If I just want to make outbound calls do I need to setup extensions? |
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| 14:49.36 | larrymo | nvm |
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| 15:57.47 | ThrobbingPython | hi all! |
| 15:58.12 | ThrobbingPython | do any of u know about the frequency setting on an avaya 655a psu? |
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| 17:28.10 | larrymo | I setup a channel called "mychannel" is this considered a trunk? |
| 17:28.25 | larrymo | instructions are telling me to put this in the call file: Channel: SIP/trunkname/18882223333 |
| 17:32.27 | ChannelZ | yes although I really hate that term |
| 17:33.13 | WIMPy | Just because it's wrong? |
| 17:33.19 | ChannelZ | A bit. |
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| 17:51.07 | larrymo | wow I got a call out |
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| 18:03.52 | danfromuk | Hi. How can I diagnose this issue with odbc voicemail? http://pastebin.com/sJbvLRka |
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| 18:42.32 | danfromuk | Hi. How can I diagnose this issue with odbc voicemail? http://pastebin.com/sJbvLRka It was working in 1.8, but since upgrading to 11, its stopped working |
| 18:45.24 | [TK]D-Fender | danfromuk: you sem to have multiple VM backend's running at the same time. last I checked this is not allowed |
| 18:45.38 | [TK]D-Fender | danfromuk: you have fil, sql, and odbc in there |
| 18:46.27 | danfromuk | [TK]D-Fender: I'll check but i've only configured odbc |
| 18:47.01 | [TK]D-Fender | NOLOAD the others |
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| 18:49.13 | danfromuk | I'm looking but i can't see any other vm backends being loaded. |
| 18:49.29 | danfromuk | I'm looking through the modules folder |
| 18:51.49 | danfromuk | where can you see that multiple VMs are being loaded? |
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| 19:03.54 | [TK]D-Fender | danfromuk: its lookin for raw files... |
| 19:04.59 | danfromuk | i've only ever seen it doing that. I thought it copies data into raw files for record and playback. |
| 19:06.02 | danfromuk | When it used to work in 1.8, it used to copy the data into raw files even though the data was actually stored in mysql |
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| 19:30.22 | danfromuk | I've removed all the modules, and checked make menuselect to ensure i'm only using odbc storage. |
| 19:30.31 | danfromuk | Still got the same problem. |
| 19:32.08 | danfromuk | http://pastebin.com/bePHReTG |
| 19:38.23 | danfromuk | Could this be a bug because the same config worked fine in 1.8 |
| 19:38.44 | danfromuk | and i've checked the sample files and voicemail.conf doesnt seem to have changed much. |
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| 19:41.53 | danfromuk | I know the odbc connection is working because the dialplan is pulled from the same database |
| 19:43.38 | danfromuk | and voicemailmain is able to access the database |
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| 19:50.28 | larrymo | can I specify within a .call file to use text to speech? Or is this AGI material? |
| 19:52.10 | larrymo | ahh maybe if I install festival |
| 19:57.34 | [TK]D-Fender | larrymo: callfile points to dialplan. you need to learn extensions.conf |
| 19:57.55 | [TK]D-Fender | that is where all call processing happens |
| 19:58.15 | WIMPy | Call files don't have to go to dialplan. |
| 19:58.33 | [TK]D-Fender | or a direct single app |
| 19:59.15 | [TK]D-Fender | which could be an agi... but then again that typically calls dislplan apps as well |
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| 20:06.24 | danfromuk | [TK]D-Fender: found the issue. there was a new column added called msg_id. I expected the cli to warn me about the missing column. |
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| 23:16.44 | Owner | hi, i can see in the log file that im recieving a call, but it says it goes to ss-noservice |
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| 23:32.43 | [TK]D-Fender | that is a freepbx issue |
| 23:33.43 | [TK]D-Fender | if its from an extension, then it's a lack of an outbound route. if from a trunk, it's a lack of an inbound route to match |
| 23:33.57 | [TK]D-Fender | ~freepbx |
| 23:33.57 | 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 |
| 23:36.40 | WIMPy | \o/ |
| 23:36.55 | WIMPy | got a call delivered in under one second. |
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| 23:50.08 | Owner | [TK]D-Fender, how do i make a route match exactly |
| 23:56.41 | [TK]D-Fender | you look at what the call is targeting... and you make a route to match |