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| 01:14.00 | resno | how could you have an external phone that works perfectly, but cant recieve calls from an internal extension? |
| 01:14.22 | resno | (internal = inside the network) (external = mobile phone) |
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| 01:14.33 | resno | both using sip |
| 01:14.43 | [TK]D-Fender | Apparently it isn't working perfectly. |
| 01:14.55 | [TK]D-Fender | And you haven't shown us a complete failed call |
| 01:15.07 | [TK]D-Fender | Want an autopsy : show us a bodty... |
| 01:15.08 | [TK]D-Fender | ~pb |
| 01:15.09 | 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. |
| 01:15.11 | [TK]D-Fender | ^^^^ |
| 01:15.12 | [TK]D-Fender | body* |
| 01:15.14 | resno | :) |
| 01:16.13 | resno | so heres the issue, the internal -> external doesnt do anything... no logs show up |
| 01:16.29 | resno | looking at asterisk-cli and nothing |
| 01:17.14 | resno | ive done something weird with configuring external vs internal i believe |
| 01:17.37 | resno | so there's been a death but i have no body to show |
| 01:17.47 | [TK]D-Fender | And you're doing this with SIP DEBUG enabled? |
| 01:18.00 | resno | ya, asterisk -rvvvvv ? |
| 01:18.02 | [TK]D-Fender | no |
| 01:18.09 | [TK]D-Fender | that only gets you to * CLI |
| 01:18.13 | [TK]D-Fender | that is not "sip debug" yet |
| 01:18.19 | [TK]D-Fender | "sip set debug on" <----- |
| 01:18.31 | [TK]D-Fender | From * CLI |
| 01:18.36 | resno | gotcha |
| 01:18.51 | resno | well this is more actionable info |
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| 01:21.54 | resno | just capture it and paste? i dont see anything 404ing or with anything bad |
| 01:22.00 | resno | mainly 200s |
| 01:35.00 | [TK]D-Fender | yes |
| 01:35.06 | [TK]D-Fender | ~pb |
| 01:35.06 | 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. |
| 01:36.27 | resno | can i pastebin the log in a pm? |
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| 01:39.03 | [TK]D-Fender | just PB it... |
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| 01:39.28 | resno | http://pastebin.com/PQzp8Gg4 |
| 01:39.30 | resno | k |
| 01:39.47 | [TK]D-Fender | Nowhere is a call coming in there at all |
| 01:39.48 | resno | i came across a bad event |
| 01:41.19 | [TK]D-Fender | meaningless |
| 01:42.00 | [TK]D-Fender | You're device there is trying to announce a "state" to *... which is something * does not support |
| 01:42.08 | ruben23 | hi guys anyone can help regarding this error i cant install asterisk 1.8...tried couple of time im getting this error as always ----> http://pastebin.com/rvAjt1Kv |
| 01:42.11 | [TK]D-Fender | But at no point in there is a call arriving at your server at all |
| 01:42.35 | [TK]D-Fender | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 01:42.42 | [TK]D-Fender | ruben23: You're missing pre-req's. |
| 01:42.59 | [TK]D-Fender | go verify them all again |
| 01:43.03 | [TK]D-Fender | You've skipped some |
| 01:44.59 | ruben23 | <PROTECTED> |
| 01:45.52 | WIMPy | Just run the supplied script. |
| 01:45.58 | [TK]D-Fender | read the instructions that listed them all... then look what you installed. |
| 01:48.29 | ruben23 | i got it guys..thanks a lot |
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| 01:58.47 | [TK]D-Fender | alright, I'm off for the night... |
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| 06:48.29 | yottanami | I installed uelastix but there is no any PBX/Fax/Plugin items int the tabs |
| 06:48.36 | yottanami | should I install any other thing ? |
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| 08:01.21 | tulga | someone use redfone E1 to SIP hardware? |
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| 11:48.15 | puzzled | hi |
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| 12:01.49 | davlefou | hi |
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| 14:11.07 | dhardison | what would you guys recommend i enable to prevent attacks such man in the middle -- are the other big concerns regarding security I should be worried about? |
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| 14:13.47 | Mango45 | I'm having a problem with incoming calls dropping after 15 minutes 30 seconds. |
| 14:14.08 | Mango45 | At 15 minutes, I receive an invite which seems to work properly. |
| 14:15.37 | Mango45 | Then at 15 minutes 30 seconds I receive another invite. This one states that the audio IP is the IP address of the SIP switch, which is incorrect. That is when the call drops. |
| 14:16.12 | Mango45 | Is anyone aware of such an issue? |
| 14:17.22 | Mango45 | Both Asteirsk servers are 11. |
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| 14:51.29 | Mango45 | It just got smarter in here. |
| 14:51.40 | Mango45 | Hello [TK]D-Fender |
| 14:55.17 | file | moo |
| 14:55.26 | Mango45 | oink |
| 14:55.30 | file | kaboom |
| 14:57.38 | WIMPy | wuff |
| 14:57.43 | newtonr | cluck cluck |
| 14:58.19 | Mango45 | wait |
| 14:58.23 | Mango45 | what animal says "kaboom"? |
| 14:59.09 | newtonr | those bombs from Mario |
| 15:01.24 | puzzled | dhardison: with VoIP there are many security concerns. Read the README-* docs in the Asterisk source, install and configure fail2ban |
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| 15:09.02 | puzzled | dhardison: you can make a MitM attack between clients & Asterisk server more difficult by using TLS/SRTP with your own CA + self-signed certs and do certificate verification ( tlsdontverifyserver=no, enable it on client) |
| 15:10.33 | dhardison | puzzled: do you think that is sufficient protection? |
| 15:10.37 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 15:12.52 | [[thufir]] | I'm trying to understand the big picture. In broad terms, how is Asterisk used? I'm reading the Oreilly Asterisk book, it covers 1.4, but can't see the forest for the trees. |
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| 15:13.13 | file | how is a very broad question |
| 15:13.23 | lvlinux | read "Asterisk The Definitive Guide" |
| 15:13.26 | lvlinux | ~book |
| 15:13.26 | 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 |
| 15:13.53 | lvlinux | Asterisk is used however u want it to be used, within what it can do (a lot) |
| 15:16.47 | puzzled | dhardison: there is no such thing as sufficient protection :) I'd add restrictive firewall rules, add call limits, monitor stuff real-time with something like http://www.humbuglabs.org/ |
| 15:17.41 | WIMPy | Sure there is. Don't be connected to the internet. |
| 15:18.41 | [[thufir]] | lvlinux: thx |
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| 15:18.49 | puzzled | WIMPy: yup :) |
| 15:21.29 | [[thufir]] | lvlinux: oh, my library only has the second version, but that's the book I'm reading. In general, what does Asterisk do? It just does VoIP? Does another piece of software handle the interface for actually dialing numbers? Does each box need to be running Asterisk, or do the clients connect to the Asterisk server? |
| 15:22.02 | WIMPy | [[thufir]]: Read the curtrent version online |
| 15:22.10 | WIMPy | ~book |
| 15:22.10 | 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 |
| 15:22.22 | WIMPy | And no, it doesn't only do VOIP. |
| 15:22.55 | WIMPy | even has Asterisk running without VOIP. |
| 15:23.58 | [TK]D-Fender | Asterisk connects to all kinds of telephony tech & protocols. that's why it's called Asterisk |
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| 15:25.11 | dhardison | puzzled: :-) all right |
| 15:26.10 | [[thufir]] | Maybe I'm asking the wrong q, or backwards. What connects to Asterisk? |
| 15:26.24 | [TK]D-Fender | Whatever you set up to connect to it |
| 15:26.50 | [TK]D-Fender | Get an ITSP account. Set it a matching side on Asterisk |
| 15:27.01 | [TK]D-Fender | Get a softphone . Set it a matching side on Asterisk |
| 15:27.27 | [TK]D-Fender | Get a SIP desk-phone. Set it a matching side on Aster |
| 15:27.35 | WIMPy | Or get hardware to connect differnt kinds of lines or phones. |
| 15:27.50 | [TK]D-Fender | Get a POTS line installed in your home. Install an appropriate interface in your server Set it a matching side on Aster |
| 15:28.19 | [TK]D-Fender | There are interfaces for all kinds of physical lines. * talks all sorts of VoIP protocols. |
| 15:28.25 | [TK]D-Fender | It is whatever YOU make of it |
| 15:28.31 | [[thufir]] | ok. |
| 15:28.37 | [TK]D-Fender | I use mine as a jukebox and to make coffee. |
| 15:31.54 | [[thufir]] | heh. so a softphone, database of numbers (outbound and inbound) is what would connect to or use Asterisk? |
| 15:32.19 | [TK]D-Fender | if you want a DIALER to automate stuff, that is also a "thing". |
| 15:32.28 | [TK]D-Fender | YOU make the databse. YOU tell Asterisk to dial out. |
| 15:32.39 | [TK]D-Fender | Or find something pre-made |
| 15:33.06 | [TK]D-Fender | Asterisk has no pre-made concept of this |
| 15:33.19 | [TK]D-Fender | ~softphone |
| 15:33.19 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 15:33.38 | [TK]D-Fender | Skype is a softphone (in addition to it's other functions) |
| 15:33.59 | [TK]D-Fender | Software to use your regualr hardware audio as the "phone" part |
| 15:36.48 | [[thufir]] | Ok, I was looking at vicidial, which I guess would be dialer which communicates with Asterisk? |
| 15:36.55 | [TK]D-Fender | yes |
| 15:37.04 | [TK]D-Fender | It tells * to dial out |
| 15:39.11 | [[thufir]] | making a bit more sense, over all. would "any" softphone then connect to vicidial, with * then dialing the number? |
| 15:39.52 | WIMPy | Victimdial is just telling Asterisk waht to do. Asterisk does all the telephony stuff. |
| 15:40.41 | [TK]D-Fender | Go read what vicidial does exactly |
| 15:40.46 | [TK]D-Fender | We don't support it here |
| 15:41.10 | [TK]D-Fender | The general attitude here is anti-phone-spamimng |
| 15:41.48 | newtonr | [[thufir]], Use google image search, and search for something like - asterisk network diagram - , you'll get a good idea of the kind of topologies Asterisk is used in and the other systems it connects to. |
| 15:42.55 | [[thufir]] | err, no spamming, anyhow. topology, I see, that was the word I needed, thanks. |
| 15:46.09 | [TK]D-Fender | Well "dialer" is a very generic concept |
| 15:46.15 | [TK]D-Fender | What precisely do you want to do? |
| 15:51.30 | [[thufir]] | it's inbound and outbound, poor mans CRM. I was thinking that Asterisk did all that, but, apparently, * is more about routing and the protocol (?). It's a small office, yes, it's sales, yes we have a database to work from. |
| 15:52.37 | [[thufir]] | As you say, * does the telephony stuff. |
| 15:53.15 | [TK]D-Fender | Asterisk is the engine. It still has to know what you want. |
| 15:53.22 | [TK]D-Fender | It isn't built for ONE concept |
| 15:53.33 | [TK]D-Fender | You configure it to interact with your other stuff. |
| 15:53.56 | [TK]D-Fender | You have a CRM? Ok, what is the user doing to start a call out? |
| 15:54.09 | [TK]D-Fender | You need to break down the process. |
| 15:54.59 | [TK]D-Fender | It's your databse.. what is the user doing to pick something from it? What software is taht done on? Then what should happen? |
| 15:57.35 | [[thufir]] | lol, no CRM as of yet, well, nothing to that level. The user should be picking the record manually. It will probably be in, as you say, victim dial. I was thinking to start with * for my reading to get a broad picture, but was getting bogged down. |
| 16:01.37 | [TK]D-Fender | You need to have a broader picture. |
| 16:01.55 | [TK]D-Fender | Noone can tell you how to do anything when you don't actually have a plan of what you want. |
| 16:02.55 | [TK]D-Fender | Want to dial numbers from a list? Where's the list coming from? Does it have to interact with your source or can you possibly just export it and take it into a non-integrated solution? Does it have to report anything back? |
| 16:03.26 | [TK]D-Fender | How would the user pick from the list? Is that done automatically? Any particular order? How about automation and frequency? |
| 16:04.07 | [TK]D-Fender | So as this goes on and on.... you have to define your needs and then you can see what is pre-made out there and what you'll have to work on yourself to make the bits do what you need them to do. |
| 16:05.21 | [TK]D-Fender | I could invent a concept of having a list of people I want John to call, and he just picks up a SIP phone connected to *, dials a specific feature code number I defined, and * just picks the next from a list right out of MySQL, and flags it as "called" and lets him go on his merry way. |
| 16:05.37 | [TK]D-Fender | No visual interface to SEE who he's calling, it just takes the "next". |
| 16:05.49 | [TK]D-Fender | So that is an idea. Is it what YOU want? We don't know. |
| 16:05.54 | [TK]D-Fender | Do you expect reporting? |
| 16:06.16 | [TK]D-Fender | Were you actually looking for this dialer to be used to connect one person to another, or to just send out automated announcements? |
| 16:06.22 | [TK]D-Fender | ... |
| 16:06.23 | [[thufir]] | I don't think we need reporting at this stage, but some data entry and flags for each record. |
| 16:06.27 | [TK]D-Fender | In the end... specifics count |
| 16:07.34 | [TK]D-Fender | Defined your data you start with. definehow the decision made as to when to take an action. Define whoat you want it to do through that course. |
| 16:07.46 | [[thufir]] | What I was envisioning was that each rep would get one record at a time, flag and enter data, and then send that record back to the database. |
| 16:08.47 | [[thufir]] | All the actions would be initiated by rep, in that they click "dial", hang up etc. |
| 16:10.07 | [TK]D-Fender | Sounds like vicidial might work for that as "ready-made" |
| 16:10.22 | [TK]D-Fender | Verify your needs for the source of your list |
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| 16:11.49 | [[thufir]] | the more I read about victim dial, the more it sounds like a good solution for us, it mentions blended mode, which is we currently have. I just need to inform myself more about how the agent and manager work for it. No doubt I will have actual * questions later :) , but I have a direction to go in for now. |
| 16:12.21 | [TK]D-Fender | "victim dial" what is this exactly? Not a name I've heard of. |
| 16:14.10 | [[thufir]] | oh, vicidial, I was just going along with that. |
| 16:14.59 | newtonr | [TK]D-Fender, WIMPy called it Victimdial |
| 16:15.24 | [TK]D-Fender | Mean, but not entirely statistcally inaccurate... |
| 16:15.25 | WIMPy | Not like that was a new name for it. |
| 16:16.04 | WIMPy | like that comment |
| 16:16.06 | WIMPy | +s |
| 16:17.01 | [[thufir]] | http://astguiclient.sourceforge.net/vicidial.html looks good because it's for customer service group without alienating customers, FWIW. |
| 16:18.16 | [TK]D-Fender | I'm not sure if they even addedf support beyond * 1.4 |
| 16:18.21 | [TK]D-Fender | There have been changes to AMI since... |
| 16:18.27 | [TK]D-Fender | Which it depends on. |
| 16:18.40 | [TK]D-Fender | The browser requirements for this are: - Firefox 0.9 or greater (Firefox 1.0.7 is the recommended browser) |
| 16:19.06 | [TK]D-Fender | http://wiki.vicidial.org/index.php/VICI:system_requirements |
| 16:19.09 | [TK]D-Fender | Horrible sign |
| 16:19.37 | [TK]D-Fender | So Google up some other dialer solutions for * then |
| 16:19.49 | [[thufir]] | heh, thank you. |
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| 17:26.38 | [[thufir]] | vicidial recomends zoiper for a softphone, for what it's worth. |
| 17:27.14 | *** join/#asterisk roentgen (~quassel@openvpn/community/support/roentgen) |
| 17:29.38 | [TK]D-Fender | we recommend against soft-phones entirely |
| 17:31.04 | Nivex | What's the recommendation for an IPv6 hardphone and/or ATA? |
| 17:35.45 | [[thufir]] | I'll go back to check. right now we use cisco hardphones, which are good quality. Probably would keep the hardphones, actually, and use headsets. |
| 17:38.43 | [[thufir]] | they don't specify a recommendation for hardphone or ATA |
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| 18:12.42 | roentgen | [TK]D-Fender: what's so wrong with soft-phones? |
| 18:15.25 | [TK]D-Fender | You mean besides having your head chained to a PC with a wired headset having to disrupt standard application focus on-screen to used the keypad and call functionality, being subject to shitty sound cards and other apps invading on it's audio, etc? Only everything. |
| 18:17.26 | roentgen | a good headset via Bluetooth is very nice especially for people that need both hands free |
| 18:21.11 | roentgen | the problems in voip are seldom related to the computer used. The networking involved is a nightmare -- from a sysadmin POV |
| 18:21.52 | solmsted | Using ARI, if I start playing audio to a channel using POST /channels/{channelId}/play, am I responsible for cleaning it up later with DELETE /playbacks/{playbackId}? Or is it deleted automatically at some point? |
| 18:23.49 | file | if the playback ends you'll get an event and it will be deleted automatically |
| 18:23.56 | file | or I should say, when |
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| 18:25.09 | solmsted | file, thanks, I kind of expected that but wasn't sure if the resource stayed around for a while so you could POST /playbacks/{playbackId}/control to replay it later |
| 18:25.38 | file | nope - it gets deleted and is taken out of the queue |
| 18:26.09 | solmsted | great, thanks again |
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| 19:23.10 | KNERD | hey, i I got a lne like this --> exten => s,n,GotoIf($["${Var_TO}" = "<sip:1222333444@XXX.XX.XX.XX>"]?ex1,s,1:3) |
| 19:23.17 | KNERD | can i change it to: |
| 19:23.33 | KNERD | exten => s,n,GotoIf($["${Var_TO}" = "<sip:1222333444@*>"]?ex1,s,1:3) |
| 19:23.48 | KNERD | it it dont matter where is comes from? |
| 19:25.25 | [TK]D-Fender | Seems the concept of "equals" escaped you there.... |
| 19:25.45 | ChannelZ | there's a separate regex operator you can use, which I forget what it is |
| 19:25.52 | [TK]D-Fender | "=" is not "looks vaguely like, or has any idea of patterns |
| 19:26.02 | KNERD | seems the concept of "*" means wildcard |
| 19:26.18 | [TK]D-Fender | * is a text character is a text field |
| 19:26.21 | KNERD | thanks |
| 19:26.33 | [TK]D-Fender | = means equal. Not "like" |
| 19:26.56 | [TK]D-Fender | "core show function REGEX" <- would get you closer |
| 19:27.02 | [TK]D-Fender | So would comparing LESS of the string |
| 19:27.04 | KNERD | = * == equals anthing |
| 19:27.09 | KNERD | thanks |
| 19:27.45 | [TK]D-Fender | if you don't care after the @ .. then chop it off from that point |
| 19:28.00 | KNERD | oh..okay..thanks |
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| 19:36.35 | KNERD | it's just not working I keep getting. sent into invalid extension 'My_DID_NUMBER1' in context 'ringtones', but no invalid handler ; http://pastebin.com/5dS8Aa4v |
| 19:37.02 | [TK]D-Fender | I don't see the call with the error in there... |
| 19:37.26 | [TK]D-Fender | I also see no reference to the context "ringtones" in that pastebin at all. |
| 19:37.32 | KNERD | it's the same |
| 19:37.37 | KNERD | give me a moment for a call |
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| 19:37.50 | [TK]D-Fender | The dialplan doesn't match the error |
| 19:41.19 | KNERD | redacting the paste now |
| 19:45.36 | KNERD | [TK]D-Fender: http://pastebin.com/L0CenAGF |
| 19:45.39 | KNERD | there is the call |
| 19:47.07 | [TK]D-Fender | <PROTECTED> |
| 19:47.09 | [TK]D-Fender | [2014-05-03 12:39:42] WARNING[31912]: pbx.c:5119 __ast_pbx_run: Channel 'SIP/scratch-tech-1-hx3SIP-00000056' sent into invalid extension 'MY_DID1' in context 'ringtones', but no invalid handler |
| 19:47.12 | [TK]D-Fender | Error is jsut what it says |
| 19:47.33 | [TK]D-Fender | Your target is simply not there |
| 19:49.01 | KNERD | exten => s,n,GotoIf($["${Var_TO}" = "<sip:MY_DID1@THIS_SERVER_IP_ADDRESS>"]?ex1,s,1:3) is hwo I have it |
| 19:49.44 | [TK]D-Fender | that looks nothing like what that Goto is targeting |
| 19:50.52 | KNERD | line 7 of the paste: To: <sip:MY_DID1@THIS_SEVER_IP> |
| 19:51.02 | [TK]D-Fender | meaningless |
| 19:51.20 | [TK]D-Fender | [15:49][TK]D-Fenderthat looks nothing like what that Goto is targeting |
| 19:52.07 | KNERD | exten => s,1,Set(Var_TO=${SIP_HEADER(TO)}) is not correct? |
| 19:52.37 | [TK]D-Fender | What does that have to do with the Goto error? |
| 19:52.45 | [TK]D-Fender | You aren't looking at the actual offending line |
| 19:53.41 | KNERD | sure I am |
| 19:53.46 | [TK]D-Fender | No, you're not. |
| 19:53.50 | KNERD | ent into invalid extension 'MY_DID1 |
| 19:54.11 | [TK]D-Fender | line 86 = HARD GOTO |
| 19:54.16 | [TK]D-Fender | that is NOT part of your dialplan |
| 19:54.26 | [TK]D-Fender | THAT is the line with the invalid target |
| 19:56.23 | [TK]D-Fender | Where do you even see "var" in your call debug at all? This is not touching anything remotely like the code sample you gave us |
| 19:57.23 | KNERD | it's Var_TO |
| 19:57.40 | [TK]D-Fender | Where do you see the letters "Var" in that pastebin at all? |
| 19:57.48 | KNERD | nope |
| 19:58.03 | [TK]D-Fender | your code is meaningless.. it isn't being called in there |
| 19:58.59 | KNERD | guess I need to read up more on "Set" to see if I did soemthing wrong |
| 19:59.05 | [TK]D-Fender | No. |
| 19:59.14 | [TK]D-Fender | Your code is not being called in there at all |
| 19:59.30 | [TK]D-Fender | This isn't "set" syntax. |
| 19:59.45 | [TK]D-Fender | This is "where am I in the dialplan that it should ever LAND on my code?" |
| 20:03.13 | KNERD | hmmm |
| 20:07.20 | KNERD | i am totally lost now..I put actual numbers in the diplan lines |
| 20:07.31 | KNERD | 1 till end |
| 20:07.53 | [TK]D-Fender | Your entire context means nothing |
| 20:08.04 | [TK]D-Fender | At no point is the code we see there trying to jump to it |
| 20:08.14 | [TK]D-Fender | I don't see why you think it should get magically called in there |
| 20:08.25 | [TK]D-Fender | You could invent a million lines of other extens. |
| 20:08.37 | [TK]D-Fender | that context name is not being called by that code |
| 20:09.06 | [TK]D-Fender | the call at no point in that flow is directed towards what you created |
| 20:10.09 | [TK]D-Fender | You haven't told us why you believe it should work otherwise to date. |
| 20:10.20 | KNERD | it's not magic, it called commans... |
| 20:10.22 | KNERD | commands |
| 20:10.50 | [TK]D-Fender | yes, and the dialplan is full of tons. At what point do you think it should have touched that bit of code? |
| 20:10.57 | [TK]D-Fender | Look at what is being executed |
| 20:11.04 | [TK]D-Fender | Where do you see that jumping to your context? |
| 20:11.06 | [TK]D-Fender | It isn't |
| 20:11.54 | KNERD | exten => s,2,GotoIf($["${Var_TO}" = "<sip:MYDID1@THIS_SERVER_IP>"]?ex1,s,1:3) ..so it sees MYDID1@THIS_SERVER_IP thus should be going to ex1 context |
| 20:12.15 | [TK]D-Fender | Stop pastbin that line.... |
| 20:12.19 | [TK]D-Fender | it never gets called. |
| 20:12.27 | [TK]D-Fender | nothing points vaguely in the direction of what you put it |
| 20:13.17 | [TK]D-Fender | that line sees nothing.. because it is never executed |
| 20:14.04 | KNERD | http://pastebin.com/TGJG64Gi |
| 20:14.14 | KNERD | sure it gets called |
| 20:14.24 | [TK]D-Fender | Oh look .... a context name change.... |
| 20:14.35 | [TK]D-Fender | and no, it doesn't |
| 20:14.40 | [TK]D-Fender | And no... it isn't getting called.... |
| 20:15.08 | [TK]D-Fender | [15:57][TK]D-FenderWhere do you see the letters "Var" in that pastebin at all? |
| 20:15.48 | KNERD | hmmm |
| 20:15.49 | [TK]D-Fender | [15:47][TK]D-Fender-- Goto (ringtones,MY_DID1,1) [15:47][TK]D-Fender[2014-05-03 12:39:42] WARNING[31912]: pbx.c:5119 __ast_pbx_run: Channel 'SIP/scratch-tech-1-hx3SIP-00000056' sent into invalid extension 'MY_DID1' in context 'ringtones', but no invalid handler |
| 20:16.09 | [TK]D-Fender | Just like it says... there is no match for the Goto to land on. |
| 20:16.23 | KNERD | okay..i see |
| 20:16.30 | KNERD | thanks..maybe I see what I did wrong now |
| 20:18.12 | [TK]D-Fender | And you code has other serious errors by the time a call ever gets to reaching it in the first place. |
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| 20:21.13 | KNERD | well let me get the call going to it properly first |
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