| 00:10.30 | *** join/#asterisk dllama (~Adium@static-108-30-11-8.nycmny.fios.verizon.net) |
| 00:11.16 | dllama | hey guys, Can anyone suggest how to "register" a cisco 8961 w/o CCUM? I been banging my head against the wall for 2 days now |
| 00:12.01 | dllama | this is very likely the wrong room, but #cisco isn't of very much assistance, and i've exhausted many other options |
| 00:16.12 | WIMPy | SCCP or SIP? |
| 00:16.15 | dllama | SIP |
| 00:16.31 | WIMPy | And what's the issue? |
| 00:16.40 | dllama | just says phone not registered |
| 00:16.50 | dllama | I dont have the XML file to feed into it via tftp |
| 00:16.55 | WIMPy | Does it try to register? |
| 00:17.11 | WIMPy | Are youre network settings ok? |
| 00:17.17 | dllama | it tries to get the info vai tftp, but i dont have CCUM nor do i know whats supposed to be in the XML file |
| 00:17.22 | dllama | via* |
| 00:18.11 | dllama | many **MANY** calls to cicso, 1 lady was nice enough to tell me that all i need to do is get passed the registration, and then I can use the web console to enter any sip credentials i want. but its the registration thats crippling me |
| 00:18.20 | dllama | and googling for it only brings up CCUM results |
| 00:19.42 | [TK]D-Fender | [20:17]dllamait tries to get the info vai tftp, but i dont have CCUM nor do i know whats supposed to be in the XML file <- there are tons of guides on setting up provisioning files for them. |
| 00:20.36 | dllama | [TK]D-Fender: where? I had trouble finding a sample xml, |
| 00:20.59 | dllama | i'm not really a telephony guy, i'm a ruby dev, but was just tasked with this and pulling my hair out :( |
| 00:25.25 | [TK]D-Fender | http://www.minded.ca/2009-12-16/configure-cisco-ip-phones-with-asterisk/ |
| 00:25.38 | [TK]D-Fender | Th 7965 series should be close in generation for that. |
| 00:27.34 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP |
| 00:30.48 | dllama | thank you! i'm going through this now |
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| 10:48.49 | youjelly | hi guys, so I have a server which has Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux installed, from I don't know where, maybe its yum, maybe its some custom build they had on their jenkins or whatever that no longer exists |
| 10:49.02 | youjelly | now I need to add cdr_mysql module to it |
| 10:49.06 | youjelly | what are my options |
| 10:49.45 | youjelly | if I download the 1.8.27 source and compile it with cdr_csv and copy the so file in modules directory, would that load? |
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| 12:24.53 | dkakoti | Hello everyone I have tried Atxfer in ami asterisk 11 |
| 12:25.12 | dkakoti | but transfer is not working |
| 12:25.44 | *** join/#asterisk Sprocks (~Sprocks@70.49.35.13) |
| 12:25.50 | dkakoti | I got response in array array(2) { ["Response"]=> string(7) "Success" ["Message"]=> string(26) "Atxfer successfully queued"} |
| 12:26.15 | dkakoti | anyone have any idea? |
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| 13:05.46 | dkakoti | HI everyone |
| 13:06.33 | WIMPy | Lo you |
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| 13:47.31 | hfp | Hi, I am using GoogleTTS.agi to generate prompts from Google Translate. I moved servers and for some reason the prompts are quiet. I think my system is missing a library or maybe a permission problem. How can I see the output for the commands run in the AGI script? I tried `agi debug` in the asterisk console but it only shows variables. I am using Asterisk 1.8.13.1 |
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| 15:36.01 | lilibox | hi |
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| 15:48.48 | *** join/#asterisk lucido (~chris@g227035101.adsl.alicedsl.de) |
| 15:49.26 | lucido | should I use IAX2 or SIP to connect to my interfaces (Linksys SPA3201)? |
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| 15:55.25 | [TK]D-Fender | lucYou should know that the SPA does not talk IAX2 so it's a moot point |
| 15:55.33 | [TK]D-Fender | lucido: You should know that the SPA does not talk IAX2 so it's a moot point |
| 15:56.17 | lucido | damn |
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| 15:57.29 | [TK]D-Fender | Hardly matters anyway... |
| 15:59.15 | lucido | [TK]D-Fender, you mean sip can do the same things? |
| 15:59.39 | [TK]D-Fender | What have you read the difference to be? |
| 15:59.41 | *** part/#asterisk LiuYan (~LiuYan@222.125.134.157) |
| 16:04.03 | lucido | [TK]D-Fender, http://www.voip-info.org/wiki/view/IAX+versus+SIP |
| 16:04.38 | [TK]D-Fender | Don't buy into all the hype. |
| 16:05.12 | [TK]D-Fender | First actual NAT issues aren't that common. NExt IAX@ trunking does not exist on a device that only really handles 1-2 calls at a time, so there's no savings there. |
| 16:05.32 | [TK]D-Fender | It's Inter-ASTERISK eXchange. It's not really meant for endpoints anyway |
| 16:05.32 | lucido | [TK]D-Fender, what worried me was: IAX has a very clear layer2 and layer3 separation, meaning that both signaling and audio have defined states, are robustly transmitted in a consistent fashion, and that when one end of the call abruptly disappears, the call WILL terminate in a timely fashion, even if no more signaling and/or audio is received. SIP does not have such a mechanism, and its reliability from a signaling perspective is obviously |
| 16:05.32 | lucido | very poor and clumsy |
| 16:05.41 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 16:06.15 | [TK]D-Fender | Hype. Nothing to concern yourself with. And IAX2 cannot do T.38 which you NEED from what you were asking about |
| 16:06.50 | lucido | ok, that's good |
| 16:07.44 | lucido | wow cloase to 1400 lines in the sample sip.conf |
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| 16:35.30 | lilibox | mr. [TK]D-Fender i have got some delicate question on you, could i ask you in prvmgs it is not quite topic related, thank you much |
| 16:35.40 | [TK]D-Fender | ok |
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| 18:15.51 | file | falls into existence |
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| 19:13.35 | lucido | example sip.conf line: [0000FFFF0001](office-phone) |
| 19:13.46 | lucido | here what is 0000FFFF0001 ? |
| 19:14.10 | lucido | where is 0000FFFF0001 defined? |
| 19:19.48 | [TK]D-Fender | <PROTECTED> |
| 19:23.18 | lucido | so if I have two lines lets say on an a pap2t and they both are set to register to asterisk as their sip server and and both device name definition has host=dynamic then I can send the two lines to different dial plans by setting a different port for them or differentiate by user? |
| 19:24.43 | [TK]D-Fender | both |
| 19:24.56 | lucido | let's say type=friend, host=dynamic, and where do I set the sip username and password? |
| 19:25.24 | lucido | I secret, but I dont see a user option |
| 19:25.33 | lucido | know secret* |
| 19:27.51 | *** part/#asterisk dllama (~Adium@ool-45720743.dyn.optonline.net) |
| 19:27.53 | [TK]D-Fender | [thisistheusername] |
| 19:28.02 | [TK]D-Fender | secret=thisisthepassword |
| 19:28.16 | lucido | ah, I see |
| 19:28.51 | lucido | so the username is also the device name |
| 19:30.51 | [TK]D-Fender | in case you want a different name to use in the dialplan that as the actual username |
| 19:31.32 | [TK]D-Fender | [thispeernamelooksfriendly] |
| 19:31.52 | [TK]D-Fender | username=h4rd2gu355u53rn4m3 |
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| 19:47.42 | *** join/#asterisk NirS (~NirS@37.205.61.203) |
| 19:47.51 | NirS | Greetings all |
| 19:48.56 | NirS | Anybody home ? |
| 19:50.04 | lucido | greetings earthling |
| 19:50.20 | NirS | erthling ? |
| 19:50.24 | NirS | earthling ? |
| 19:50.36 | lucido | are you from another planet perhaps? |
| 19:50.40 | NirS | wow, I hadn't been in here for ages that I'm considered an alian :-) |
| 19:51.39 | NirS | So, who's coming to Astricon Vegas ? |
| 19:52.36 | lucido | not me |
| 19:55.01 | NirS | looks like we are the only people in here |
| 19:55.19 | lucido | I bet [TK]D-Fender is here too |
| 19:55.36 | lucido | he's just not very talkative but all the more helpful |
| 19:58.07 | NirS | I think the last time I was in here was around 2006 :-( |
| 19:58.40 | NirS | when I was last here, jtodd had ops :-) |
| 19:59.10 | lucido | don't know, I came a week ago |
| 20:01.47 | lucido | [TK]D-Fender, from the debian/asterisk example sip.conf: "The parameter "username" is not the username and in most cases is |
| 20:01.47 | lucido | ; not needed at all. Check below. In later releases, it's renamed |
| 20:01.47 | lucido | ; to "defaultuser" which is a better name, since it is used in |
| 20:01.47 | lucido | ; combination with the "defaultip" setting." |
| 20:02.19 | lucido | and: When setting up trunks, make sure there's no risk that any From: username |
| 20:02.19 | lucido | ; (caller ID) will match any of your device names |
| 20:02.24 | [TK]D-Fender | PASTEBIN |
| 20:02.26 | [TK]D-Fender | ~pb |
| 20:02.26 | 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. |
| 20:02.30 | lucido | ok |
| 20:02.31 | [TK]D-Fender | Do not flood in here |
| 20:02.50 | [TK]D-Fender | And yes, it has been renamed, and isn't needed unless you want to use something OTHER thatn [whatyouputhere] |
| 20:03.21 | NirS | D-Fender |
| 20:03.28 | NirS | that is slightly a little off |
| 20:03.49 | lucido | it mentions that a username is the caller-id and not the sip username |
| 20:03.54 | NirS | You can configure a UA to utilize a different authentication user than the phones "SIP user" |
| 20:04.09 | lucido | UA? |
| 20:04.27 | NirS | normally, this will be configured under the username - although, in 99% of cases, the configuraiton of the username in peer context heading would be more than enough |
| 20:04.33 | NirS | UA - User Agent |
| 20:04.41 | NirS | SIP Phone if you'd like |
| 20:05.40 | lucido | NirS, I thought type=peer matches only for ip and port and not name |
| 20:06.37 | NirS | that is correct, as long as you specify the host IP address |
| 20:07.19 | NirS | if you specify type=friend (anyone still uses that?) - then it is matched according to the name |
| 20:07.32 | NirS | are you referring to configuring trunks or devices ? |
| 20:07.35 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 20:07.55 | NirS | Brando ? |
| 20:07.57 | NirS | Brandon ? |
| 20:08.43 | NirS | greets bkruse - been a while |
| 20:14.09 | lucido | NirS, both |
| 20:14.46 | lucido | if I have an spa3201 with an FXO then that is trunk I presume, how do I define that in sip.conf? |
| 20:32.12 | *** join/#asterisk jpoz (~jpoz@181.sub-70-199-135.myvzw.com) |
| 20:32.55 | lucido | in sip.conf under [general], udpbindaddr=192.168.1.1 is defined only for [general] or this is applies to all sections? In other words, where are the general options defined for the whole sip module and not for individual devices/trunks? |
| 20:35.14 | [TK]D-Fender | huh? |
| 20:35.38 | [TK]D-Fender | udpbindaddr is a [general] setting o9nly. it is not specific to a particular device |
| 20:35.47 | [TK]D-Fender | that is the address * will bind to. |
| 20:36.12 | [TK]D-Fender | that is where it will listen. |
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| 20:36.34 | lucido | [TK]D-Fender, I see it just wasn't clear that there were general options |
| 20:36.46 | lucido | or their place of definition |
| 20:38.11 | lucido | how can I define a trunk, for example my FXO on the SPA3201? in a dialplan? |
| 20:38.39 | [TK]D-Fender | no, the definition IS in sip.conf |
| 20:38.45 | [TK]D-Fender | Dialplan processes calls FROM devices |
| 20:38.56 | [TK]D-Fender | But the device itself is in the channel driver config |
| 20:39.07 | [TK]D-Fender | [thisismySPAfxoport] |
| 20:39.11 | [TK]D-Fender | port=5061 |
| 20:39.15 | [TK]D-Fender | ..... |
| 20:39.17 | [TK]D-Fender | etc |
| 20:39.53 | lucido | aha, so there's no differenttation in the syntax of defining FXO or FXS connections in sip.conf |
| 20:40.06 | lucido | I mean devices |
| 20:40.09 | [TK]D-Fender | SIP doesn't have such concepts |
| 20:40.30 | [TK]D-Fender | what is on the other side is not something it cares about. |
| 20:40.54 | lucido | with FXO asterisk is the client and with FXS it is the server I presume |
| 20:41.12 | [TK]D-Fender | no. |
| 20:41.17 | WIMPy | There is no client. SIP is P2P. |
| 20:41.24 | [TK]D-Fender | do not mix those concepts at all. |
| 20:41.26 | [TK]D-Fender | SIP is SIP |
| 20:41.31 | lucido | ok |
| 20:41.43 | [TK]D-Fender | the fact that you are taking a line Iin on your SPA is not something that SIP has a definition for. |
| 20:41.51 | [TK]D-Fender | it is just a SIP call. |
| 20:41.54 | [TK]D-Fender | EVERY call is the same. |
| 20:42.01 | lucido | ok, just a p2p connection |
| 20:42.23 | [TK]D-Fender | request comes in targeting something (typically a number), Auth happens (or not), call gets accepted (or not), and if accepted gets processed |
| 20:42.26 | lucido | where both sides can initiate the connection (call) |
| 20:42.55 | lucido | I see |
| 20:42.56 | [TK]D-Fender | FXO port on SPA places call to * over SIP. * takes call in and does what you say with it. |
| 20:43.36 | [TK]D-Fender | Another device places a call. Same process... a step in processing says "take that number and dial it OVER the SPA". * does it and the SPA dials out that FXO line |
| 20:43.43 | [TK]D-Fender | The SIP part is just SIP. |
| 20:43.59 | [TK]D-Fender | The SPA's transformation of it to FXO is it's function... not intrinsic to SIP |
| 20:44.19 | lucido | one more question please, which codec should I use with the * to FXO connection to enable fax? |
| 20:45.23 | [TK]D-Fender | no codec, I've mentioned this several times |
| 20:45.25 | [TK]D-Fender | T.38 <--- |
| 20:45.31 | [TK]D-Fender | it is a PROTOCOL, not a "codec" |
| 20:45.56 | [TK]D-Fender | And this will allow a reliable communication of that data call to * |
| 20:49.06 | lucido | I am receiving a T.30 fax call at the FXO and the SPA will transform that into a T.38 virtual circuit to * for receiving, no sip involved, right? |
| 20:49.30 | [TK]D-Fender | SIP is what is used to negotiate T.38 |
| 20:50.29 | lucido | [TK]D-Fender, therefore no codec comes into the picture since it's not a voice connection |
| 20:50.29 | [TK]D-Fender | so you will still set a codec for voice calls.. and when a FAX is detected on either side it will attempt to negotiate T.38 assuming you configured both ends for it |
| 20:50.42 | lucido | I get it |
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| 21:24.57 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 21:25.05 | raspberrypifan | has anyone done skype to * |
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| 21:27.29 | [TK]D-Fender | There are some seriously kludgy hacks to getting it to work with *, and the "normal" way which involves having a business account with them. |
| 21:27.42 | [TK]D-Fender | No real "free" ride per se |
| 21:28.29 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Skype+Gateways |
| 21:28.31 | [TK]D-Fender | go read up |
| 21:29.26 | lucido | should be possible to make some glue code that would connect the skype client with * |
| 21:31.06 | [TK]D-Fender | There used to be an official channel driver for it. Skype shut down those options |
| 21:31.22 | [TK]D-Fender | They don't want PBX folk hopping on the free ride |
| 21:33.29 | raspberrypifan | yea i was kinda reading about that, cause i didnt even know it was ever posisble |
| 21:33.34 | raspberrypifan | cant you use the old chan driver? |
| 21:34.09 | [TK]D-Fender | no |
| 21:34.19 | raspberrypifan | did they block the access? |
| 21:35.09 | [TK]D-Fender | its gone |
| 21:35.13 | [TK]D-Fender | absolutely gone |
| 21:35.20 | raspberrypifan | sad, that owuld have been a cool project |
| 21:35.54 | [TK]D-Fender | Go read that list and see what is still active out there if any |
| 21:38.53 | raspberrypifan | so since in Ecuador its ilegal to connect SIP to PSTN, i was thinking of deploying hudnreds of cheap routers around the city to connect to |
| 21:40.09 | WIMPy | Do it in another country. |
| 21:40.31 | raspberrypifan | ill be living in Ecuador |
| 21:40.34 | raspberrypifan | i cant just move ot another country |
| 21:40.59 | WIMPy | That dosn't mean you need to connect VOIP and PSTN in the same country. |
| 21:41.28 | WIMPy | Your data will happily move around the globe. |
| 21:41.28 | raspberrypifan | what are you suggesting my sly friend? |
| 21:41.56 | WIMPy | Use a foreign ITSP, for example. |
| 21:42.30 | raspberrypifan | so say i connect my Ecuadorian PSTN to COLOMBIA and then use colombia for sip? |
| 21:42.56 | WIMPy | Possibly. |
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| 21:43.15 | raspberrypifan | im just saying colombia cuase i think there more free market |
| 21:43.23 | raspberrypifan | though idk if there are any ITSP in Colombia |
| 21:44.11 | WIMPy | What exactely is illegal in Equador? |
| 21:44.50 | raspberrypifan | diverting aka by pass, idk if that is the english word. Thats the term they use |
| 21:45.11 | WIMPy | Not sure what EXACTELY that would mean. |
| 21:45.56 | raspberrypifan | i cant connect a gsm gateway via the internet for example. Anything form of communication that bypasses the PSTN for foreign calls |
| 21:46.42 | WIMPy | That sounds like you wouldn't be allowd to use VOIP at all. |
| 21:47.36 | [TK]D-Fender | [17:42]raspberrypifanso say i connect my Ecuadorian PSTN to COLOMBIA and then use colombia for sip? <- this is a proben description |
| 21:47.56 | raspberrypifan | [TK]D-Fender huh? |
| 21:48.30 | raspberrypifan | WIMPy well i dont think voip is ilegal, the law is not very clear but if you are using the internet to make calls and connecting it to the local telephone system |
| 21:48.33 | raspberrypifan | it is ilegal |
| 21:48.36 | raspberrypifan | and they will shut u down |
| 21:50.21 | WIMPy | "the local phone system"? That could mean te PSTN or a PBX. |
| 21:50.29 | raspberrypifan | PSTN |
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| 21:50.40 | raspberrypifan | it seems to be a common restriction worldwide |
| 21:50.42 | WIMPy | Ok, so what's the problem with that? |
| 21:50.54 | WIMPy | No |
| 21:51.07 | raspberrypifan | it is, india, the uk and others are touchy about it |
| 21:51.51 | WIMPy | It that is the restriction, I don't see much restriction there. |
| 21:52.03 | *** join/#asterisk wonderworld (~ww@194-96-49-163.adsl.highway.telekom.at) |
| 21:52.54 | WIMPy | It would just mean that you must not use a PBX with remote VOIP extensions. |
| 21:53.04 | WIMPy | So anythig else? |
| 21:53.27 | raspberrypifan | it means i can't use asterisk for foreign conenctions |
| 21:53.59 | [TK]D-Fender | can't use "anything" more like... |
| 21:54.23 | WIMPy | I don't see that from your description. |
| 21:54.29 | raspberrypifan | how could i not? |
| 21:56.31 | [TK]D-Fender | "Anything form of communication that bypasses the PSTN for foreign calls" |
| 21:58.31 | WIMPy | Obviousely only possible in countries that still have a (working) PSTN. |
| 21:59.13 | raspberrypifan | dont most countries have a pstn |
| 21:59.42 | WIMPy | Many are tearing ot down. |
| 21:59.55 | raspberrypifan | really? |
| 22:02.18 | WIMPy | Yes :-( |
| 22:03.38 | raspberrypifan | like who? |
| 22:04.11 | WIMPy | Eurote to different levels. |
| 22:04.25 | WIMPy | Europe |
| 22:04.33 | raspberrypifan | wel theres always the cell phone networks rigth |
| 22:04.45 | WIMPy | I thinkk, Germany is at the front again :-( |
| 22:05.47 | raspberrypifan | r u european? |
| 22:06.01 | WIMPy | Trouble is that cellphones already have very high delays. Combine that with VOIP and you and up in a pretty grim place. |
| 22:06.05 | WIMPy | yes. |
| 22:06.29 | raspberrypifan | which kind |
| 22:07.12 | WIMPy | I'm not that much in to the PLMNs, but UMTS and LTE do native VOIP. |
| 22:07.22 | WIMPy | kind of what? |
| 22:07.48 | raspberrypifan | of european |
| 22:08.11 | WIMPy | At the German-Danish border. |
| 22:08.33 | lucido | holstein |
| 22:08.54 | WIMPy | It's the Schleswig part. |
| 22:09.26 | raspberrypifan | can you jump from one country to the other |
| 22:09.38 | WIMPy | Or Slesvig |
| 22:09.48 | lucido | everyone I know has only voip |
| 22:10.04 | raspberrypifan | well i dont really know anyone with anything other then a cell |
| 22:10.08 | lucido | I'll try to retrieve credentials from my o2 box |
| 22:10.11 | WIMPy | Drive, walk, run, jump, swim.... |
| 22:10.20 | raspberrypifan | you can swim across countries? |
| 22:10.50 | WIMPy | Well, we are located between two oceans. |
| 22:10.59 | lucido | not really, but you can swim across the border |
| 22:11.08 | raspberrypifan | can you take a HSR |
| 22:11.18 | WIMPy | True |
| 22:11.28 | WIMPy | HSR? |
| 22:11.41 | raspberrypifan | deustchebhan |
| 22:12.33 | WIMPy | To cross the border? Yes, but DB (Deutsche Bahn) and DSB (Danske Staatsbaner) cross the border. |
| 22:12.40 | raspberrypifan | ICE |
| 22:12.55 | WIMPy | I think there are also trains from Switzerland doing so. |
| 22:13.15 | raspberrypifan | if you go to amsterdamn, you can take a direct flight to Quito |
| 22:17.17 | WIMPy | Much more interesting than crossing the border is the possibility to go from east coast to west cost and back on the same day. |
| 22:17.25 | WIMPy | .. using a bike :-) |
| 22:20.15 | raspberrypifan | from spain to china? |
| 22:20.39 | WIMPy | That might take a little longer. |
| 22:21.30 | raspberrypifan | i wanna go live in Ecuador |
| 22:21.34 | raspberrypifan | Europe* |
| 22:21.37 | raspberrypifan | i thnk id like it |
| 22:21.54 | WIMPy | Some E*-Place? |
| 22:21.59 | WIMPy | :-) |
| 22:22.11 | raspberrypifan | yup only E place alow |
| 22:22.35 | raspberrypifan | so apparently EC has ISDN |
| 22:22.49 | WIMPy | It still exists. |
| 22:23.15 | raspberrypifan | i can use that for sip cant i? |
| 22:23.47 | WIMPy | But EC has been replace by the EU a long time ago. |
| 22:24.03 | raspberrypifan | holy fuck they have DID's |
| 22:24.23 | WIMPy | Not sure what that question means, but I's probably a no. |
| 22:24.58 | WIMPy | Sure. |
| 22:25.18 | raspberrypifan | so it would seem that in fact the cable company is providing sip type stuff |
| 22:25.26 | raspberrypifan | which means i might be able to use their credentials |
| 22:25.30 | raspberrypifan | now that wold be awesoe |
| 22:25.39 | WIMPy | They all use SIP. |
| 22:25.57 | raspberrypifan | i didnt know they had that at all, everything is a monopoy in Ecuador |
| 22:26.35 | WIMPy | You might be lucky without knowing. |
| 22:26.43 | raspberrypifan | maybe |
| 22:26.46 | raspberrypifan | i need to try and reach them ppl |
| 22:27.07 | WIMPy | What people? |
| 22:27.32 | raspberrypifan | http://www.grupotvcable.com.ec/grupo/telefonia |
| 22:29.02 | raspberrypifan | they have pbx line and stuff |
| 22:29.18 | raspberrypifan | which means as someone had suggested, with wiresharke you might be able to find the sip credentials |
| 22:29.44 | WIMPy | Usually you hack squeeze them out of the IAD somehow. |
| 22:30.10 | WIMPy | You don't get credentials from sniffing. But you might be able to sniff the provisioning process. |
| 22:30.23 | WIMPy | Or just download the config and decode it. |
| 22:30.47 | WIMPy | Or log in to the IAD and read the config. |
| 22:30.52 | WIMPy | Done it all :-) |
| 22:31.06 | raspberrypifan | well when im actually down there on the ground |
| 22:31.10 | raspberrypifan | ill come around and ask for ideas |
| 22:32.06 | WIMPy | Trouble is that you never know for how long it will work. |
| 22:33.38 | raspberrypifan | true but u might get lucky lol |
| 22:34.41 | WIMPy | has been lucky so far. |
| 22:35.11 | raspberrypifan | lucky in life |
| 22:35.18 | WIMPy | No |
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| 22:35.32 | raspberrypifan | so why r u WIMPy anyway |
| 22:35.33 | WIMPy | Just with them not messing up things I hacked. |
| 22:35.53 | WIMPy | is a WIMP user. |
| 22:36.30 | raspberrypifan | i do prefer wimp to cli |
| 22:37.02 | WIMPy | Depends. |
| 22:37.27 | raspberrypifan | on? |
| 22:37.40 | WIMPy | I did. On a specific WIMP, calld "WIMP", or officially the "RISC OS desktop". |
| 22:37.54 | WIMPy | But on the PC I often prefer the CLI. |
| 22:38.11 | WIMPy | X-Windows is quite messy. :-( |
| 22:38.50 | JTL | Hi all |
| 22:39.02 | WIMPy | Lo you |
| 22:39.16 | JTL | anyone knows of good open source Auto dialer for asterisk they would recommend? |
| 22:39.41 | WIMPy | Define "auto dialer". |
| 22:39.59 | youjelly | hi guys, so I have a server which has Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org on a x86_64 from I don't know where, maybe its yum, maybe its some custom build they had on their jenkins or whatever that no longer exists |
| 22:40.41 | youjelly | now I need to add cdr_mysql module to it, if I download the 1.8.27 source and compile it with cdr_csv and copy the so file in modules directory, would that load? |
| 22:40.58 | WIMPy | no |
| 22:41.20 | JTL | auto dialer = Call center suite. I would like to be able for asterisk box to initiate calls to targeted/solicited clients of mine |
| 22:41.35 | WIMPy | The modules have version/config information to prevent damage from such experiments. |
| 22:41.40 | raspberrypifan | thank you for calling the lucky duck foundation |
| 22:42.31 | WIMPy | JTL: Still vague. Some sort of CRM/desktop integration or the predictive dialler terrorism? |
| 22:42.46 | raspberrypifan | https://www.youtube.com/watch?v=CVgArsc4zLw |
| 22:43.06 | JTL | Wimpy: Yes that's correct |
| 22:43.19 | WIMPy | JTL: No it's not. |
| 22:43.28 | WIMPy | It was not a yes/no question. |
| 22:44.06 | raspberrypifan | is there an irc channel for call termination |
| 22:44.22 | WIMPy | raspberrypifan: Meaning what? |
| 22:44.45 | raspberrypifan | selling minutes |
| 22:44.46 | JTL | WIMPy:predictive dialler |
| 22:45.14 | WIMPy | JTL: Look for Vicidial (AKA victim dial) and a good lawyer. |
| 22:46.10 | JTL | WIMPy: Thanks. This is for my clients so I should be fine |
| 22:47.56 | youjelly | WIMPy what sshould I do then? |
| 22:48.42 | youjelly | upgrade to a version? not really an option, they have a couple of custom modules compiled for that version I don't have the source for |
| 22:48.46 | WIMPy | Build a complete new version. |
| 22:49.38 | WIMPy | I don't know if you can extract the required configuration options from, the installed version. |
| 22:49.56 | WIMPy | But if it has custom stuff added, that sounds rather risky anyway. |
| 22:50.10 | youjelly | yeah |
| 22:50.25 | youjelly | on I test machine I tried doing what I mentioned |
| 22:50.41 | youjelly | and its loading cdr_mysql |
| 22:51.07 | youjelly | I compiled version 1.8.15 |
| 22:51.34 | youjelly | copied the cdr_mysql so file over to /lib64/astersiks/modules directory |
| 22:51.44 | WIMPy | If the config is compatible, you might be lucky. |
| 22:52.23 | WIMPy | But definitely not recommended. |
| 22:52.33 | youjelly | I know... |
| 22:52.53 | youjelly | where can I get version 1.8.20 |
| 22:53.14 | WIMPy | http://downlaods.asterisk.org/pub/telephony/asterisk/ |
| 22:53.19 | youjelly | Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org |
| 22:54.27 | youjelly | broken link |
| 22:56.54 | WIMPy | http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ |
| 22:56.59 | WIMPy | Sorry. Typo |
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| 23:02.40 | youjelly | thanks |
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