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| 12:50.07 | stephanj | is there a way to get the conference room number for dialplan_exec? |
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| 15:11.33 | rubio | hi there.... i'm loosing my head here fighting WebRTC sound issue... is this the right channel to ask about this? (running asterisk 11.13 from debian 8 with sipml5 over chrome) thanks. |
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| 17:01.16 | igcewieling | rubio: I doubt many people here run WebRTC |
| 17:04.16 | [TK]D-Fender | plenty of people now actually. |
| 17:04.22 | [TK]D-Fender | It has gotten pretty big |
| 17:04.31 | [TK]D-Fender | Maybe a few less for being Halloween weekend... |
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| 17:13.03 | [red] | having a really weird cdr issue in asterisk 13.2.2 that i didn't have in asterisk 11. for some reason on calls, the callerid field wont save the correct callerid, instead of being the callerid of the caller, it's a random 4 digit number |
| 17:13.12 | [red] | and clues of where to start debugging this? |
| 17:14.31 | [TK]D-Fender | First you upgrade and see if the problem is still happening |
| 17:14.53 | [red] | i'm tempted to just upgrade the asterisk version as it feels "buggy", but not sure if i'm doing something else stupid |
| 17:14.59 | [red] | okay, will do |
| 17:14.59 | [TK]D-Fender | #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 19:35.27 | cesurasean_ | how do i block sipvicious from making connects to my box? |
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| 19:54.49 | [TK]D-Fender | firewalls |
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| 20:35.33 | MaliutaLap | remove the network connection - air gap the machine |
| 20:47.53 | [red] | hmm still some strangeness after updating asterisk |
| 20:48.14 | [red] | callerids in the cdrs are still replaced with 4 digit numbers at random |
| 20:48.18 | [red] | really bizzare |
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| 21:03.44 | [TK]D-Fender | PASTEBIN is your friend.... |
| 21:03.45 | [TK]D-Fender | ~pb |
| 21:03.49 | 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. |
| 21:04.01 | [TK]D-Fender | ^^^ |
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| 21:54.27 | *** join/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 21:54.49 | Voyage | Can't see any official setup or installation guide for AMD https://wiki.asterisk.org/wiki/display/AST/Application_AMD |
| 22:00.59 | [TK]D-Fender | There is nothing to have an installation guide for |
| 22:01.05 | [TK]D-Fender | it's an application that comes with * |
| 22:01.59 | [TK]D-Fender | There is also nothing to "setup". It has a single line of parameters. It doesn't even have a config file |
| 22:02.21 | [TK]D-Fender | And every parameter is explained right there on that page |
| 22:06.24 | Voyage | Thanks! |
| 22:06.38 | Voyage | amd.conf iirc is the conf file for AMD |
| 22:06.40 | Voyage | no? |
| 22:07.04 | craigify | I think you can do both Voyage |
| 22:07.13 | craigify | I've used amd.conf |
| 22:07.23 | craigify | I think you can also specify options directly to AMD() |
| 22:07.29 | craigify | that's from memory that I'm telling you this |
| 22:07.41 | Voyage | craigify, I guess as you type AMD(), it picks up values from the amd.conf file |
| 22:07.52 | Voyage | craigify, I see. thanks |
| 22:07.53 | [TK]D-Fender | AMD([initialSilence,[greeting,[afterGreetingSilence,[totalAnalysis Time,[miniumWordLength,[betweenWordSilence,[maximumNumberOfWords,[silenceThreshold,[maximumWordLength]]]]]]]]]) |
| 22:08.01 | [TK]D-Fender | Syntax was right there. |
| 22:08.03 | craigify | there you go |
| 22:08.05 | Voyage | [TK]D-Fender, right. thanks. |
| 22:08.11 | craigify | you can also specify those values in amd.conf |
| 22:08.16 | [TK]D-Fender | What's wrong with doing exactly what it is telling you to do>? |
| 22:08.32 | Voyage | craigify, i see - as defaults |
| 22:08.52 | craigify | Voyage, if you find an example amd.conf, it probably will have descriptions of each config option |
| 22:09.06 | craigify | the asterisk docs probably also explains which each one does |
| 22:09.13 | craigify | they aren't necessarily forthcoming without docs |
| 22:09.19 | craigify | and you'll have to tweak constantly |
| 22:09.39 | Voyage | [TK]D-Fender, just by reading the article, I wasn't sure and specially the contrary article where it says steps to install amd http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD |
| 22:09.39 | craigify | you're best bet is to look on forums for people who have figured out working set of settings and start from there, then make your tweaks |
| 22:09.56 | [TK]D-Fender | WhaHow many times did I tell you STOP using that WIKI? |
| 22:10.28 | Voyage | [TK]D-Fender, you only need to advise once to a person. Any person |
| 22:10.29 | [TK]D-Fender | NOTE: app_amd is included in Asterisk as of version 1.4. |
| 22:10.33 | craigify | I wish that website would be taken down |
| 22:10.44 | [TK]D-Fender | And you ignored the big comment in BOLD on that page |
| 22:10.54 | craigify | it consistently ranks on the top of various search results |
| 22:10.55 | Voyage | craigify, really :) is that that bad? |
| 22:11.00 | craigify | it's not bad |
| 22:11.02 | craigify | it's just outdated |
| 22:11.12 | craigify | many of the things there apply to new asterisk versions |
| 22:11.27 | craigify | but you don't get any of the documentation about the new versions |
| 22:11.31 | Voyage | [TK]D-Fender, I said I was here to confirm things. Now i have. thanks |
| 22:11.44 | WIMPy | I actually found it very usefull as a beginner, bute, yes, you need to validate if the contents there are still valid before using the information. |
| 22:11.46 | Voyage | craigify, i see |
| 22:11.46 | [TK]D-Fender | You should read the actual pages you fine |
| 22:11.48 | [TK]D-Fender | find |
| 22:11.57 | Voyage | WIMPy, hm |
| 22:13.11 | Voyage | highlights to [TK]D-Fender - craigify> it's not bad WIMPy> I actually found it very usefull as a beginner, bute, yes, you need to validate if the contents |
| 22:13.36 | [TK]D-Fender | Syntax changes between * versions |
| 22:13.40 | [TK]D-Fender | And that shit is ANCIENT |
| 22:13.46 | Voyage | agrees. |
| 22:13.59 | [TK]D-Fender | And since you can't be bothered to read even pages like that in full you won't have a clue just how wrong it is |
| 22:14.21 | Voyage | [TK]D-Fender, please dont pass personal or habbitual comments about me. |
| 22:14.24 | Voyage | thanks |
| 22:14.47 | [TK]D-Fender | First, don't skim. Follow the offical pages |
| 22:14.57 | [TK]D-Fender | If you even going to look at places like that do a full job |
| 22:15.11 | Voyage | [TK]D-Fender, ok. your guidance noted with thanks. |
| 22:15.23 | Voyage | gets back to work |
| 22:15.25 | [TK]D-Fender | And then realize even when you're don, that it was written by "some guy" who's quality of writing could be entirely debateable |
| 22:15.35 | Voyage | hm |
| 22:15.44 | Voyage | agrees. |
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| 22:24.59 | *** join/#asterisk catphish (~J@unaffiliated/catphish) |
| 22:26.11 | catphish | can anyone briefly explain how SDP "negotiates" a codec? specifically, is the first codec listed in the 200 OK in response to an INVITE the one that both ends will actually use? |
| 22:28.59 | *** join/#asterisk leedm777 (~leedm777@50-81-28-152.client.mchsi.com) |
| 22:44.40 | catphish | ah, found it - "Of course, if Bob only supports one-of-N codecs, there would only be one codec in his answer, and in this case, there is no need for a re-INVITE to lock down to a single codec." |
| 23:07.58 | Voyage | looks like my AMI() is not working at all. Any ideas? http://pastie.org/10523116 |
| 23:16.35 | [TK]D-Fender | AMI is not a dialplan application\ |
| 23:16.46 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 23:17.35 | Voyage | [TK]D-Fender, am.... hm right but as you know I am just originating calls via AMI, the rest is done by dialplan (when the call gets connected. |
| 23:18.02 | [TK]D-Fender | [Nov 1 18:03:37] WARNING[18388][C-00000004]: pbx.c:4878 pbx_extension_helper: No application 'AMI' for extension (outgoing, 1xxxxxxxx7, 2) |
| 23:18.08 | [TK]D-Fender | Error says it loud and clear |
| 23:18.16 | [TK]D-Fender | <PROTECTED> |
| 23:18.26 | [TK]D-Fender | Your orignate happened |
| 23:18.28 | Voyage | what does that means? |
| 23:18.30 | [TK]D-Fender | The call was places |
| 23:18.32 | [TK]D-Fender | placed |
| 23:18.33 | Voyage | ok |
| 23:18.35 | [TK]D-Fender | The call was answered |
| 23:18.37 | Voyage | ok |
| 23:18.44 | [TK]D-Fender | it landed into the dilplan as instructed |
| 23:18.49 | Voyage | ok.. |
| 23:18.51 | [TK]D-Fender | AMI is *DONE* with your Originate |
| 23:18.59 | Voyage | yup |
| 23:19.08 | [TK]D-Fender | [TK]D-Fender> AMI is not a dialplan application |
| 23:19.29 | Voyage | oh my bad. very sorry. s/AMI()/AMD() |
| 23:19.43 | [TK]D-Fender | Indeed |
| 23:20.05 | Voyage | [TK]D-Fender, I meant the line #2 of http://pastie.org/10523116 not AMI() |
| 23:20.09 | [TK]D-Fender | It told you that wasn't an Application at all. Check your spelling. |
| 23:20.17 | [TK]D-Fender | I know |
| 23:21.46 | Voyage | oh no.. very bad spell mistake of mine. I would have wondered for many time time if you had'nt pointed it out |
| 23:21.55 | Voyage | thanks. let me see into it |
| 23:24.29 | *** join/#asterisk trewq (~trewq@li70-4.members.linode.com) |
| 23:24.57 | trewq | folks, is asterisk pretty much a sip proxy? Newbie here and trying to understand this.. In other words, where does asterisk fit in here: http://grauonline.de/wordpress/wp-content/uploads/sip_overview.gif |
| 23:26.54 | craigify | It would be where the SIP registrar server is |
| 23:26.57 | craigify | not the proxy |
| 23:27.26 | *** join/#asterisk u0m3 (~u0m3@89.120.204.99) |
| 23:28.11 | Voyage | trewq, * can behave as a proxy and a server as well (to my newbie knowledge iirc I head in the astricon) |
| 23:28.20 | craigify | it could also be where caller and callee is also |
| 23:28.25 | Voyage | or a video tutorial |
| 23:29.06 | craigify | https://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 23:29.19 | craigify | Asterisk might better fit that description |
| 23:29.37 | craigify | which is sort of what the sip registry server is doing in that diagram you posted |
| 23:31.08 | [TK]D-Fender | trewq, No, * NOT a proxy |
| 23:31.13 | [TK]D-Fender | ~b2bua |
| 23:31.13 | infobot | methinks b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 23:31.14 | [TK]D-Fender | ^ |
| 23:31.49 | craigify | Beat you to it fender |
| 23:31.51 | [TK]D-Fender | SIP packets are not passed through * |
| 23:31.52 | craigify | for once |
| 23:31.59 | [TK]D-Fender | it is a prue B2BUA |
| 23:32.04 | [TK]D-Fender | pure* |
| 23:32.30 | trewq | looking it up now |
| 23:33.23 | craigify | Also in that diagram |
| 23:33.30 | craigify | not all things happen that way |
| 23:33.35 | craigify | sometimes there is no proxy |
| 23:33.49 | [TK]D-Fender | indeed |
| 23:33.52 | craigify | also sometimes all signalling and audio/video go through asterisk, then to the endpoint |
| 23:34.36 | trewq | ok |
| 23:34.40 | craigify | there could be reasons why you would force audio through asterisk |
| 23:34.47 | craigify | you might want to record all conversations, every time |
| 23:35.00 | trewq | trying to sort out the pieces.. before I start plugging away at it |
| 23:35.08 | craigify | you might want to not deal with NAT on firewalls that would screw up RTP |
| 23:35.16 | [TK]D-Fender | What is your goal? |
| 23:35.28 | craigify | if you allow direct connection for audio between endpoints, that is |
| 23:36.38 | trewq | well, I am looking for a simple way for my webapp to know who is calling (through caller id) |
| 23:37.01 | craigify | how simple is simple? |
| 23:37.02 | craigify | heh |
| 23:37.09 | craigify | twilio.com might be pretty simple |
| 23:37.40 | trewq | craigify: good point |
| 23:37.56 | craigify | Asterisk is powerful thoughj |
| 23:38.16 | craigify | it would give you control over your telefony and an upgrade path when you want to add in new features |
| 23:38.29 | [TK]D-Fender | Your webapp is getting a call? |
| 23:38.40 | trewq | so my customer already has a POTS line, and I have a DID that forwards to the POTS line. |
| 23:38.41 | [TK]D-Fender | Isn't the information already in that call itself? |
| 23:39.14 | trewq | Since the caller calls me, I want to grab the caller id and then forward to the POTS line |
| 23:39.21 | trewq | perhaps I need to draw a diagram |
| 23:39.37 | [TK]D-Fender | you also need to clarify "pots line" |
| 23:39.55 | trewq | just someone who has a DID number |
| 23:39.55 | [TK]D-Fender | how you get there... if & where an actual telco is involved, etc |
| 23:40.01 | craigify | brb |
| 23:40.30 | trewq | I have no control of the customer premise equipment, all I know is they have a regular phone number |
| 23:40.35 | trewq | I want to forward to it |
| 23:41.50 | [TK]D-Fender | Then you need to have some way to dial back out to the PSTN and set the callerID. |
| 23:41.58 | trewq | right |
| 23:42.14 | [TK]D-Fender | This generally only allows you to set the number, not the name |
| 23:43.09 | [TK]D-Fender | So whatever you are gong to do this with will be a simple call in and the a bridged call out |
| 23:44.05 | trewq | I want to identify who is calling so I can pull out the customer records and display it on a webapp |
| 23:44.22 | trewq | realtime.. call comes in, record gets pulled up |
| 23:44.29 | [TK]D-Fender | ok, that you can do as well. |
| 23:44.57 | [TK]D-Fender | with * that is |
| 23:46.12 | trewq | yeah, I was thinking of using a simple sip proxy server, not something as sophisticated as * |
| 23:46.33 | [TK]D-Fender | * is as simple or as sophisticated as your needs are |
| 23:47.04 | [TK]D-Fender | Perhaps someone else has some specific experience with a proxy that might be quicker to pick up from |
| 23:47.18 | trewq | what would I need to do to setup this "forwarder" - would this be mostly a dialplan config? |
| 23:47.23 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 23:48.19 | [TK]D-Fender | you'd need a provider to call out with, then sip config, and dialplna (extensions.conf) to call your script to update the webapp, and then dial back out |
| 23:49.07 | [TK]D-Fender | This adds cost because of having to terminate back to the PSTN |
| 23:49.21 | trewq | yes, I agree |
| 23:49.32 | [TK]D-Fender | would be better if you could communicate directly with the CPE instead |
| 23:50.13 | trewq | CPE is just PSTN |
| 23:50.27 | [TK]D-Fender | No, you said you are calling to the client's PBX via the PSTN |
| 23:50.33 | [TK]D-Fender | they have the CPE |
| 23:50.49 | [TK]D-Fender | and bridging to it via the PSTN is an extra cost & service |
| 23:50.58 | [TK]D-Fender | vs your sending the call directly to their PBX |
| 23:51.00 | trewq | I am calling (forwarding) to the client PSTN when the call me |
| 23:51.12 | [TK]D-Fender | I know. This COSTS you. |
| 23:51.17 | trewq | I understand that there are costs associated |
| 23:51.22 | trewq | for both legs |
| 23:51.24 | craigify | You won't be able to do that with just a proxy, most likley. unless you have a special relationship with a phone company. |
| 23:51.45 | [TK]D-Fender | yup, but if you were to be able to negotiate putting something small on their side you could cut that part off |
| 23:51.55 | trewq | PSNT -> SIP proxy -> PSTN |
| 23:52.02 | trewq | *PSTN |
| 23:52.10 | trewq | that was what I was envisioning |
| 23:53.10 | craigify | a SIP proxy is a special use device |
| 23:53.23 | [TK]D-Fender | What I'm advocating is PSTN > ITSP > SIP > Proxy-or-Asterisk > SIP > small gateway at client side > Their PBX |
| 23:53.39 | [TK]D-Fender | If you can get them to agree |
| 23:53.52 | trewq | Clients call *my* PSTN number hosted at voip.ms. Voip.ms forwards SIP to me, I extract info, and then forward the call to PSTN |
| 23:54.03 | [TK]D-Fender | Otherwise this will be costs on both sides. |
| 23:54.10 | trewq | I cannot get them to have a pbs unfortunately |
| 23:54.19 | trewq | *pbx |
| 23:54.29 | craigify | maybe you can use a ATA device |
| 23:54.40 | craigify | on their end |
| 23:54.47 | trewq | yeah, that is an option, but they already have a PBS |
| 23:54.56 | trewq | dang it.. pbx :) |
| 23:54.57 | craigify | sip pbx? |
| 23:55.16 | [TK]D-Fender | ok, well then you're stuck doing both ends, which if it's fine for yo .. is what it is |
| 23:55.23 | trewq | no, some nortel stuff that I cannot use |
| 23:56.18 | [TK]D-Fender | So, call in from voip.ms, and call out via "whoever", which I suppose might as well be voip.ms as well |
| 23:56.35 | craigify | you will need more than a sip proxy |
| 23:56.46 | craigify | because you won't be able to forward the call from them right back to them |
| 23:57.04 | craigify | at least, I'm pretty sure they won't let you do that :) |
| 23:58.12 | [TK]D-Fender | just about any side will require you to auth the call out, and passing them packets from them to them... might not go well |
| 23:58.28 | [TK]D-Fender | But this branches out past my practical experience |
| 23:58.57 | craigify | also, many providers have different rates for incoming vs outgoing |
