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02:23.09 | KNERD | Reading up on dial plans...if I get this correct..... exten => _X.,1,GotoIf($["${EXTEN}" = "6245"]? 6) once completed will continue on to _X,6, once executed? |
02:23.24 | KNERD | or should I say IF excuted |
02:23.55 | [TK]D-Fender | give or take that spacer after the ? that you shouldn't have there |
02:24.56 | KNERD | well I think this is from 1.4 * |
02:25.15 | KNERD | but thanks |
02:25.54 | [TK]D-Fender | never put extra spaces where they don't belong |
02:26.28 | KNERD | thanks for the advice. was looking at this example for implementing a dial plan |
02:28.30 | p3nguin | 1.4 didn't use extraneous spaces, either. |
02:28.50 | p3nguin | And you should consider using proper labels instead of numbers for the target. |
02:29.27 | KNERD | The next line basically goes into a script, then a Hangup. _X,6, is the same script again then another Hangup...should "6245" be able to be dialed at anytime then? Or have to wait until script finishes |
02:29.45 | KNERD | p3nguin: what proper labels would you reccomend? |
02:31.23 | p3nguin | If you aren't going to use a proper extension for 6245... |
02:31.25 | p3nguin | exten => _X.,1,GotoIf($["${EXTEN}" = "6245"]?any-label-you-like) |
02:31.37 | p3nguin | But extension 6245 would be my preference. |
02:32.06 | p3nguin | exten => 6245,1,Stuff() |
02:32.43 | p3nguin | I don't see the point of using a "catch all" extension just to later evaluate what extension was called. |
02:32.54 | KNERD | Well maybe you should look at the example I am looking at.. http://pastebin.ca/2132139 |
02:34.23 | p3nguin | I guess it does reduce the number of lines of dial plan. |
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02:35.22 | KNERD | the one I showed you, or the example you gave? |
02:37.00 | p3nguin | Other wise you'd end up with this: http://pastebin.com/jrxU7kc7 |
02:37.26 | p3nguin | That's correct usage of extensions, but it adds more lines of dial plan. |
02:38.06 | p3nguin | I hadn't given the example at the time you asked about the example I gave. |
02:39.03 | KNERD | I would think the example you gave would require one of the 3 being dialed, and if they dialed none? |
02:39.25 | p3nguin | I just didn't include that part in my example. |
02:39.36 | p3nguin | I was only illustrating the difference. |
02:39.42 | KNERD | yes, I see |
02:39.51 | KNERD | justy include another without any extension |
02:40.30 | p3nguin | I just found it odd to use a catch-all pattern and then evaluate what was actually called. |
02:41.07 | p3nguin | I'm not saying it's wrong or won't work, just saying I found it unusual. |
02:41.26 | KNERD | I see |
02:41.46 | p3nguin | It isn't something I would ever do. |
02:41.51 | KNERD | what about a user entering one of those extensions while the script is executing? |
02:42.00 | KNERD | Why would you not? |
02:42.11 | p3nguin | The extension would run as many times as someone calls it. |
02:42.30 | p3nguin | Unless something else prevents it from running, of course. |
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02:42.59 | KNERD | okay..thanks for that advice |
02:43.27 | KNERD | now I have to figure out how to implement the IVR to tell the person what to dial |
02:43.43 | KNERD | because the script is already telling them what to dial...ugg |
02:43.49 | KNERD | thanks a lot p3nguin |
02:45.22 | p3nguin | Okay, for IVR, I can kind of see that method making a little more sense. |
02:45.51 | p3nguin | IVR would ask for input, caller would press some keys. |
02:46.05 | p3nguin | Then GotoIf() could go somewhere based on the entry. |
02:46.17 | p3nguin | But I didn't see a Read() so I didn't recognize it as IVR. |
02:46.51 | KNERD | because it is really not...the script is running it's own IVR |
02:47.15 | p3nguin | its |
02:47.16 | KNERD | and reccomendation is to add your own similar to what I showed you |
02:48.45 | KNERD | i guess what I want to add is an announcement which states "for customer service press xxx" then goes into a queue |
02:50.18 | p3nguin | That's not IVR. |
02:50.25 | p3nguin | That's an auto attendant. |
02:50.33 | p3nguin | And you should use normal extensions for that. |
02:51.44 | p3nguin | For customer service, press 1 now. exten => 1,1,Goto(customer-service-queue,s,1) |
02:54.46 | KNERD | ohhh |
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05:30.39 | dijib | this place is quiet |
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06:34.15 | schmidts | good morning |
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07:07.23 | KNERD | godd night |
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07:36.34 | qakhan | does anyone know how to install UniMRCP dependencies? |
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07:52.35 | jercos | qakhan: yes. |
07:52.51 | qakhan | please help me how? |
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08:51.43 | qakhan | jercos u there? |
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08:56.43 | jercos | qakhan: think about your question, then about what we all know about you. |
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08:57.22 | bulkorok | is there a good dialer for asterisk at the moment?! |
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09:03.41 | qakhan | jercos i didnt get you |
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09:16.15 | KNERD | nothing? |
09:16.50 | KNERD | bulkorok: yeah..* has a great one...it's called DIAL |
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09:40.59 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
09:41.07 | ruben23 | hi guys |
09:44.23 | ruben23 | guys any help my phone extensions cannot register on my asterisk even the credentials are correct |
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09:44.38 | ruben23 | NOTICE[2401]: channel.c:3271 __ast_request_and_dial: Unable to request channel SIP/cc121 |
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09:46.16 | schmidts | ruben23 you mixed two things, what you have shown us is when you try to dial a peer, but not what you get when a peer tries to register |
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09:48.35 | Sconk | is there a way to see members of my sales que ? |
09:49.19 | Sconk | queue show |
09:49.20 | Sconk | :) |
09:50.54 | KNERD | freepbx |
09:52.00 | ruben23 | schmidts: sorry let me cehck the lgos again |
09:52.04 | ruben23 | log* |
09:53.01 | ruben23 | i see this on the log ---> [Mar 26 11:52:07] -- Got SIP response 405 "Method Not Allowed" back from 81.133.43.31 ---> [Mar 26 11:52:07] -- Got SIP response 405 "Method Not Allowed" back from 81.133.43.31 |
09:54.19 | KNERD | sounnds like a codec issue? |
09:56.32 | KNERD | http://en.wikipedia.org/wiki/List_of_SIP_response_codes#4xx.E2.80.94Client_Failure_Responses |
09:59.02 | ruben23 | KNERD: this regards with codec..? ------------>405 Method Not Allowed The method specified in the Request-Line is understood, but not allowed for the address identified by the Request-URI. |
09:59.04 | KNERD | ruben23: oh....type=peer or type=friend? try swapping |
10:00.11 | KNERD | of course would help ifwe saw more |
10:00.15 | kaldemar | ruben23: look at SIP debug. you'll see what the 405 is an answer to. however, you phones not being able to register is a separate issue. |
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10:05.30 | ruben23 | guys would this help----------------------------------> http://pastebin.com/vjN72gxe |
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10:08.19 | ruben23 | its strange coz when i used zoiper it work with other it work with 3cxphone it does not work at all but for teh record teh 3CX phones work already for 2 weeks then suddenly today all not working |
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10:09.12 | davlefou | hi, |
10:09.34 | KNERD | ruben23: actually it would help if we saw the client and seever settings for this |
10:11.30 | kaldemar | ruben23: your asterisk is sending voicemail notificacations to 81.133.43.31, which it probably should not. don't define a mailbox for it in sip.conf if it is a provider. |
10:12.21 | KNERD | hasvoicemail=no ? |
10:14.01 | davlefou | is it possible to use same numbre for fax and phone with asterisk? I think with sub number? |
10:14.57 | KNERD | of course it is. you just tell it to listen for fax |
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10:31.35 | qakhan | anyone using uniMRCP? |
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10:48.31 | KNERD | qakhan: .000000000000000000001 seconds via Bing http://www.voip-info.org/wiki/view/Asterisk+cmd+MRCPSpeech |
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11:07.17 | qakhan | KNERD anything else except this link |
11:08.05 | KNERD | qakhan: yeah. www.bing.com |
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11:35.26 | fred9999 | good morning everyone |
11:38.35 | fred9999 | I am mixing up extension and context in conf file. I would like to have an extension that kicks the user directly to a meetme room with fixed number. I have 2 sterisk linked by a sip trunk. I can call remote extension but when I try to define an extension that would kick into a meetme room. I have a Call from '' to extension '8000' rejected because extension not found. |
11:38.46 | fred9999 | Here are the conf file |
11:40.23 | fred9999 | on the target * (number 2) extension.conf |
11:40.41 | fred9999 | [internal] |
11:40.43 | fred9999 | exten=>9600,1,Dial(SIP/9600,,r) |
11:40.44 | fred9999 | exten=>9601,1,Dial(SIP/9601,,r) |
11:40.46 | fred9999 | ;exten=>8000,1,Dial(SIP/9600,,r) |
11:40.46 | kaldemar | ~pb |
11:40.46 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
11:40.47 | fred9999 | exten=>8000,n,MeetMe(100,1dqF) |
11:41.22 | fred9999 | I'll use the pastebin for the config file |
11:42.04 | kaldemar | you need to paste sip.conf aswell and a CLI output of a call with verbosity and sip debug for someone to be able to specifically tell you what's wrong. |
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11:55.18 | fred9999 | seems ok now. I have defined an extension 8000 in the sip.conf file. I guess when calling SIP/XXXX, the XXXX has to be defined in the sip.conf file, right ? |
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12:00.14 | kaldemar | yes. but what you define in sip.conf is not an extension but a device. an extension is something that begins with "exten =>" in extensions.conf. |
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13:05.37 | fred9999 | thanks, these are the things I am mixing up. |
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13:06.58 | mr_pete | Afternoon all :) |
13:08.24 | mr_pete | Can anyone help me with keeping asterisk in channel to keep features working? Moved from 1.4 to 1.6 and for some reason (with dtmfmode set to rfc2833 on all extensions & trunks) and canreinvite=no - features WERE working outbound on 1.4 |
13:08.27 | mr_pete | and they aren't no 1.6 |
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13:17.24 | jacc0 | hi all |
13:17.26 | jacc0 | :) |
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13:19.52 | Delido192123 | Hello i need Help with Asterisk´s CallCompletionRequest, CCBS is not working at all, but ccnr. i think this is happend because i dont use the DIAL Function on Busy state. Can someone help me to correct this? |
13:24.21 | Delido192123 | i cant set call-limit to 1 because noone can do callforwarding. asterisk ignore the busy-level and the dial-command are executet. I have Polycom IP320 SIP PHones at all. The User can choise within our intranet the callwaiting state (on / off) |
13:27.17 | fred9999 | thank you kaldemar for your help. have a nice day ! |
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13:28.34 | kaldemar | fred9999: thanks, you too. |
13:28.57 | [TK]D-Fender | mr_pete, canreinvite became "directmedia". and dmdmode is unchanged |
13:29.57 | mr_pete | [TK]D-Fender: incoming calls (any trunk) features can be activated fine - and local ext2ext calls work, but outgoing, no dice :\ |
13:31.06 | [TK]D-Fender | what "features"? Your terminology is very vague |
13:31.20 | [TK]D-Fender | and you aren't showing us the actual failure for us to advise you on. |
13:31.29 | [TK]D-Fender | PASTEBIN it all up so we can see what's happening. |
13:31.30 | [TK]D-Fender | ~pb |
13:31.30 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
13:31.31 | [TK]D-Fender | ^^ |
13:37.04 | mr_pete | anything active in features.conf |
13:37.39 | mr_pete | like blind transfer. It's why I was referring to canreinvite as I suspect Asterisk is going out of the loop and thus not "listening" anymore |
13:40.48 | mr_pete | tweaking debug... |
13:43.04 | mr_pete | I'm seeing DTMF locally (<< [ TYPE: DTMF End (1) SUBCLASS: * (42) ] [SIP/6000-00000050] ) |
13:43.12 | mr_pete | but not when calls go via trunk |
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13:52.37 | [TK]D-Fender | mr_pete, if you set the dial options to allow for them in the first place no reinvite is permitted regardless |
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13:53.27 | polysics | hello, Asterisk people! |
13:53.38 | mr_pete | I've got "DIALOPTIONS = KkXxtT" set in extensions.conf, and the features are enabled in features.conf |
13:53.59 | polysics | i know about attended transfer using either Dial() or features.conf |
13:54.10 | polysics | but how would I pilot that using AMI? |
13:54.12 | mr_pete | and as far as I can tell macro-trunkdial-failover-0.3 should call thos in? |
13:54.32 | polysics | goal is to integrate with a small web-based panel that shows people's satatus |
13:54.54 | polysics | there is no single app that does that, right? |
13:57.42 | mr_pete | hmm, think I found the culprit... |
13:59.06 | qakhan | does anyone know how to install sofia-sip? |
14:00.03 | mr_pete | fixed it |
14:00.18 | mr_pete | it's a GUI issue, relying on macro-trunkdial |
14:00.31 | mr_pete | the macro doesn't call in "exten = s,n,Set(__DYNAMIC_FEATURES=${FEATURES}) " |
14:00.45 | mr_pete | and doesn't append DIAL_OPTIONS |
14:01.52 | [TK]D-Fender | <mr_pete> I've got "DIALOPTIONS = KkXxtT" set in extensions.conf, and the features are enabled in features.conf <- this is not a magical global variable to my awareness... I'd like to see the actual CALL. |
14:02.10 | [TK]D-Fender | qakhan, You don't. Asterisk does not use that stack |
14:02.38 | mr_pete | [TK]D-Fender: I'm using asterisk-GUI - it's put DIALOPTIONS in the conf file and updates it |
14:03.05 | mr_pete | and it's calling them via it's built macros (ala "exten = s,n,Dial(SIP/6000&SIP/6001,20,${DIALOPTIONS}i)" |
14:03.32 | mr_pete | but it doesn't append that to the trunk-dial macro which all outgoing calls go to, so they don't get activated |
14:03.37 | polysics | i was thinking of manually parking a call, dialing out, then bridging the second party to the first |
14:03.43 | polysics | there has to be an easier way :-) |
14:03.58 | [TK]D-Fender | mr_pete, FYI that GUI is practically unmaintained and has extremely minimal support at best. Consider this if you are planning on using it for anything other than some personal laerning experience |
14:04.29 | [TK]D-Fender | <polysics> i know about attended transfer using either Dial() or features.conf <- same thing |
14:04.49 | mr_pete | [TK]D-Fender: am gathering that :) It's for a home system. I was on 1.4 CLI before |
14:04.55 | [TK]D-Fender | polysics, but how would I pilot that using AMI? <- no such thing, only a hard dialplan forward, or a "Bridge" |
14:05.06 | mr_pete | just migrating to a Debian-Squeeze install on a Proliant microserver which has 1.6 |
14:05.17 | [TK]D-Fender | mr_pete, Still generally advise against, so just keep this in mind |
14:05.59 | polysics | [TK]D-Fender: so i would send the caller to another extension, dial out to the potential receiver, and eventually connect them through a series of bridges and redirects? |
14:06.07 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
14:06.19 | polysics | it's either all in features.conf or totally manual |
14:06.35 | [TK]D-Fender | mr_pete, do yourself a favour and join #asterisk-gui ... and help double their population ;) |
14:06.49 | [TK]D-Fender | polysics, No, those were 2 separate solutions |
14:06.50 | mr_pete | well for the most part it's worked :P |
14:07.12 | mr_pete | only problems I've had are becuase I'm using an SPA3102 for POTS interface (and digest auth had to be set manually via sip.conf) |
14:07.19 | mr_pete | and now this :) |
14:07.29 | polysics | what is a "hard dialplan forward", please? |
14:07.47 | [TK]D-Fender | polysics, And the bridge way you'd Originate a call to a receiver, set a channel variable to the channel to hijack, and on accept, call Bridge() from the dialplan |
14:08.23 | polysics | [TK]D-Fender: in that scenario, where is the caller waiting in the meantime? |
14:08.32 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
14:09.03 | [TK]D-Fender | polysics, Wherever they are.... I suppose you could couple that action with forwarding them somewhere else like a loop of MusicOnHold or something like that... |
14:09.33 | mr_pete | right, now all features.conf features and my applications are now in channel. hurrah :) |
14:09.41 | polysics | i know this is probably easy, but what would the best way of providing an "endless" waiting "place" for a caller? |
14:11.01 | [TK]D-Fender | polysics, hard redirect to an exten dumping them in a loop of MusicOnHold |
14:11.15 | *** join/#asterisk nickfennell (~nick@unaffiliated/nickfennell) |
14:12.05 | [TK]D-Fender | polysics, I might also swap that for having a channel variable set for ID purposes, and then use Park. The var might help ID it for tracking it for pickup via ParkedCall |
14:12.08 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
14:13.11 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
14:13.12 | fred9999 | good evening, |
14:13.13 | polysics | well, it is actually parked in a sense, i think |
14:15.07 | fred9999 | I am trying to pass a meetme parameter from one asterisk calling another asterisk on a device 8000 that kicks the call directly to the room number. How do I do that. The dialplan is calling the dial function. |
14:16.18 | [TK]D-Fender | fred9999, what is "device 8000"? what are you trying to pass? |
14:17.28 | Delido192123 | http://pastebin.com/hqpk1CQ3 Callcompletion dont work (ccbs) please help |
14:17.50 | fred9999 | it is a fake SIP device. I am trying to pass the room number the call should be redirected to on the second * |
14:18.08 | [TK]D-Fender | How is it "fake"... when it leads to ANOTHER Asterisk? |
14:18.25 | mr_pete | Ok, next silly question :) |
14:18.27 | fred9999 | there is no physical phone attached |
14:18.32 | mr_pete | what did "show globals" migrate to under 1.6? |
14:18.37 | nickfennell | only theoretical ones |
14:18.41 | [TK]D-Fender | Who said SIP had anything to do with a phone? |
14:18.50 | fred9999 | my mistake |
14:18.53 | nickfennell | I've just used SIP to get a cuppa made |
14:19.05 | [TK]D-Fender | hi-5's nickfennell |
14:19.10 | nickfennell | lol |
14:19.18 | [TK]D-Fender | nickfennell, I use * for coffee ... and as a jukebox.... |
14:19.48 | nickfennell | Jukebox is good although I find myself dialling in now to listen to streams rather than just using a native client... |
14:19.55 | [TK]D-Fender | fred9999, have you considered making the romm # part of the extension you dial on the other server? |
14:20.33 | [TK]D-Fender | nickfennell, I just dial into my player script, pick my collection, and run it on speakerphone :) |
14:21.02 | nickfennell | So you're hosting media files on the * itself? |
14:21.05 | fred9999 | that could do. How would I define all those devices in the sip.conf ? |
14:21.23 | [TK]D-Fender | fred9999, You don't.. I said EXTENSION. that = extensions.conf |
14:21.56 | nickfennell | I need to get SMS Call back working |
14:22.00 | mr_pete | what did "show globals" migrate to under 1.6? |
14:22.09 | [TK]D-Fender | nickfennell, If by "hosting" you mean on the same box as my server sitting in my livingroom... then yes :) |
14:22.16 | nickfennell | lol |
14:22.20 | nickfennell | You have * at home too ? |
14:22.22 | [TK]D-Fender | mr_pete, probably "core ..." |
14:22.24 | nickfennell | hahaha. |
14:22.31 | [TK]D-Fender | nickfennell, Have since 2004. |
14:22.34 | mr_pete | not that I can see :P |
14:22.42 | nickfennell | I'm the same. I must be the only person I know to have hold music and IVR at a house |
14:22.55 | mr_pete | nope, I had hold music until yesterday :) |
14:23.06 | nickfennell | "Thanks for calling. Press one for the kitchen, Two for the Garden, Three for the Shed" |
14:23.06 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.103) |
14:23.09 | [TK]D-Fender | mr_pete, You should be reading the "ugrade" docs, etc in your tarball. They detail all of these sort of changes. |
14:23.15 | mr_pete | I had IVR for anon callers |
14:23.31 | StaRetji | folks, does asterisk default install translate between gsm and ulaw? |
14:23.37 | nickfennell | I had it so *genuine*callers would be able to redirect to my mobile if needed |
14:23.47 | nickfennell | Cold callers didn't seem to listen to the options |
14:23.50 | [TK]D-Fender | StaRetji, * translates whatever it can translate whenever it has to |
14:24.03 | [TK]D-Fender | StaRetji, which by stock distribution does include those 2 codecs |
14:24.05 | StaRetji | I have strange situation, calls made to outside numbers (landline, mobile) |
14:24.15 | StaRetji | are in ulaw |
14:24.18 | nickfennell | At one point I had it running a dictation service so I could dial in and set memo's for myself |
14:24.22 | StaRetji | but if I make callback |
14:24.25 | nickfennell | can't remember how I wired that up though |
14:24.29 | StaRetji | both legs are in gsm |
14:24.43 | StaRetji | and no sound is heared by both parties |
14:24.55 | StaRetji | show channels says gsm |
14:25.02 | nickfennell | stix, inbound calls are GSM? |
14:25.04 | [TK]D-Fender | StaRetji, If you had a codec issue the call would drop like a rock instantly |
14:25.05 | StaRetji | while I don't gsm set in sip.conf |
14:25.10 | [TK]D-Fender | StaRetji, You have a networking issue |
14:25.37 | StaRetji | [TK]D-Fender: thx for answerz |
14:25.58 | StaRetji | I have to understand why are callback calls made in gsm |
14:26.00 | stix | nickfennell: if you say so?? |
14:26.10 | StaRetji | ulaw alaw, sound is there |
14:26.17 | StaRetji | gsm, no sound |
14:26.21 | nickfennell | stix, I must have done ? |
14:26.33 | nickfennell | stix, I mean StaRetji |
14:26.37 | nickfennell | Apologies. |
14:26.39 | StaRetji | oh |
14:26.41 | StaRetji | sorry mate |
14:26.50 | StaRetji | I was confused, looks like it is for me |
14:26.55 | StaRetji | but different name |
14:27.07 | [TK]D-Fender | <StaRetji> I have to understand why are callback calls made in gsm <- yes... you should be actually looking at that call right now..... |
14:27.10 | nickfennell | Ignore me. I'm silently failing. |
14:27.23 | *** join/#asterisk GameGamer43 (u5533@gateway/web/irccloud.com/x-uluuzlgyatpdmgio) |
14:27.30 | StaRetji | I mean, I call my self with same trunk |
14:27.34 | mr_pete | zapateller worked nicely on most auto-diallers for me |
14:27.37 | StaRetji | and call is ulaw |
14:27.40 | StaRetji | if I make callback |
14:27.46 | StaRetji | to my phone and another phone |
14:27.53 | StaRetji | callback is gsm |
14:28.16 | nickfennell | codec preferred on handset? |
14:28.20 | StaRetji | so, even though in sip.conf is never put allow=gsm |
14:28.33 | StaRetji | but same handset ulaw |
14:28.40 | StaRetji | if I call not with my sip account |
14:28.46 | StaRetji | call now* |
14:28.59 | StaRetji | but if I initate callback, it is gsm |
14:29.16 | StaRetji | I bang my poor stupid head with this lol |
14:29.27 | [TK]D-Fender | And I'm no sensing that you've actually started really looking at that call.... |
14:29.30 | StaRetji | I mean, no gsm codec set, anywhere |
14:29.43 | mr_pete | my answer = dialplan show globals |
14:29.50 | StaRetji | looking, like keeping my eyes wide open? :D |
14:29.50 | nickfennell | I don't follow the 'initiate callback' |
14:30.00 | *** join/#asterisk Joel_Oliveira (~chatzilla@estrela-adm.nortenet.pt) |
14:30.09 | nickfennell | Is that calling from landline/mobile to the Asterisk? |
14:30.20 | StaRetji | put number for leg1, put number for leg2, press callback |
14:30.31 | StaRetji | no, it is asterisk, calling 2 phones |
14:30.37 | nickfennell | oh right |
14:30.37 | StaRetji | connects them together |
14:30.39 | nickfennell | Got it |
14:30.42 | nickfennell | Not seen that |
14:30.48 | nickfennell | lives in a dark hole |
14:31.28 | *** join/#asterisk bmg505 (~leon@196-209-152-75.dynamic.isadsl.co.za) |
14:31.30 | StaRetji | let me test some more... |
14:32.27 | StaRetji | yep, called via sip account to phone number1, ulaw |
14:32.36 | StaRetji | called via sip accont phone number2, ulaw |
14:32.52 | StaRetji | none of the sip setting has gsm in it |
14:33.05 | fred9999 | Let me explain the way I do, which might not be the good way. *2 is calling SIP/9600@38-peer on the sip trunk between the 2 *. on the 2nd * I have a SIP/8000 device which does that exten=>8000,1,MeetMe(100,1dqF).The room number is 100. I would like * to pass this info to *2 in some parameter. How do I do ? |
14:33.17 | StaRetji | but if call leg1.number1, leg2.number2 via callback, they are connected in gsm |
14:33.19 | StaRetji | wtf |
14:34.29 | [TK]D-Fender | fred9999, SIP/9600@38-pee <------ make the room part of the number you dial |
14:37.30 | *** join/#asterisk serafie (~erin@nat/digium/x-shymnlbyjssrntrx) |
14:38.00 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
14:38.54 | fred9999 | something like this SIP/9600/8001@38-peer ? |
14:41.40 | qakhan | i am getting this message while i am installing unimrcp |
14:42.38 | qakhan | the message is /usr/bin/ld: cannot find -lexpat |
14:44.47 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
14:48.12 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
14:48.40 | [TK]D-Fender | fred9999, what is the number you were dialing in there in the first place? |
14:51.00 | *** join/#asterisk chasing`Sol (~cS@197.134.183.118) |
14:51.57 | *** join/#asterisk OrNix (~OrNix@l49-246-139.cn.ru) |
14:53.46 | *** join/#asterisk ddickenson (~ddickenso@67-198-0-5.static.grandenetworks.net) |
14:54.21 | ddickenson | Anyone else had trouble with asterisk 10.x not starting as a service like the old ones on Centos? |
14:57.45 | fred9999 | The first number I put is wrong. I am dialing Dial(SIP/8000@38-peer) |
14:58.04 | fred9999 | I am on 1.6 |
14:58.31 | [TK]D-Fender | fred9999, So you are dialing 8000. make the room PART of what you are dialing. |
14:59.25 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
14:59.42 | *** join/#asterisk stix (~stix@193.89.191.209) |
15:01.26 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
15:07.23 | leifmadsen | Qwell: ping |
15:10.57 | *** join/#asterisk jrad (~jrad@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
15:13.10 | jrad | Asterisk v10.2.1 - Trying to locate a memory leak that brings down the service once a day during a production work day. Are there any built in commands or build options I can use effectivly? |
15:15.58 | leifmadsen | you could enable some debug options from menuselect like DONT_OPTIMIZE and DEBUG_THREADS and then create a backtrace from the running process and 'core show locks' to gather some information |
15:16.34 | leifmadsen | jrad: https://wiki.asterisk.org/wiki/display/AST/Debugging |
15:16.49 | jrad | Thank you for your time. I will review. |
15:17.00 | *** join/#asterisk serafie (~erin@nat/digium/x-koyxqozawmwpsprd) |
15:23.09 | mr_pete | o_O |
15:23.30 | mr_pete | my POTS provider has just suffered a regional outage....(whilst I'm testing PSTN failover) |
15:23.38 | mr_pete | I did not at all think it was anything to do with me.... :\ |
15:27.51 | mpoole | hey guys, I'm having a random problem on asterisk 1.8. When I dial into a sip trunk 1/10 times the call is answered but silent. The only thing in the log is: chan_sip.c: Unsupported SDP media type in offer: video 0 RTP/AVP 31 34 98 99 |
15:27.55 | mpoole | any ideas? |
15:28.24 | *** join/#asterisk gruvfunk (18a121cf@gateway/web/freenode/ip.24.161.33.207) |
15:28.30 | gruvfunk | greets all |
15:29.20 | gruvfunk | anyone know how I can pull the 10 digit caller ID from SIP headers using Python? |
15:30.00 | gruvfunk | currently have something but it will pull a Caller ID Name sometimes depending on where the call is coming from (internal PBX calls works, external calls from PSTN do not). |
15:46.46 | gruvfunk | awake and alive? |
15:47.09 | bmoraca_work | it's too bad the G200 and the G100 aren't a little bit cheaper...if they were, they'd be a good alternative to the Adtran TA900s. TA900s are cheaper, though. |
15:49.39 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
15:50.02 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
16:07.28 | *** join/#asterisk adeel (~adeel@72.53.78.136) |
16:07.32 | jaytee | anyone here using the Buddy Watch feature on Polycom phones with Asterisk? |
16:08.09 | bmoraca_work | it works OK as a standard BLF against hints in Asterisk |
16:08.41 | jaytee | bmoraca_work, does it use up a line key exclusively? |
16:09.00 | bmoraca_work | it also functions as a speed dial for that user/featurecode |
16:09.15 | bmoraca_work | but, yes, it does dedicate the key |
16:09.16 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
16:09.26 | jaytee | bmoraca_work, thanks |
16:09.55 | bmoraca_work | has anyone here connected an Avaya IP Office to Asterisk before? is it possible without accepting anonymous calls? |
16:10.22 | *** part/#asterisk mikemol (~shortcirc@inara.firefly.michael.mol.name) |
16:16.45 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
16:19.19 | [TK]D-Fender | jaytee, more specifically your user Directory will spill onto unused line-keys in "speed-dial index" order which is the best way to see them. You can also view the status via Buddies on the phone itself |
16:21.00 | [TK]D-Fender | bmoraca_work, What hasn't worked in terms of setting up a peer like normal? |
16:21.59 | bmoraca_work | [TK]D-Fender: i don't have one at my disposal, but I've been doing research for a company who maintains a customers' IP Office (who I put in a new Asterisk system for one of their divisions) and they want to do extension-to-extension dialing. |
16:22.12 | bmoraca_work | [TK]D-Fender: every walkthrough I find says that I need to allow anonymous calls through |
16:22.17 | bmoraca_work | which doesn't seem right |
16:22.28 | bmoraca_work | but i don't know what the actual SIP dialog looks like |
16:22.56 | bmoraca_work | i can't imagine that a standard peer with insecure=port,invite wouldn't work... |
16:23.04 | bmoraca_work | but you never know! |
16:23.37 | [TK]D-Fender | bmoraca_work, Don't let your imagination wander... it's too little to be allowed out on its own ;) |
16:23.45 | bmoraca_work | pfft |
16:24.18 | bmoraca_work | i don't want to take anything for granted because i've dealt with this company before and they're not very bright |
16:26.52 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
16:29.27 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v027-166.mobile.uci.edu) |
16:29.42 | *** join/#asterisk engrxyz (~fgdfgf@wempex01.inclarity.co.uk) |
16:30.49 | *** join/#asterisk techwerkz (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
16:32.36 | techwerkz | I have a list of 309 local prefixes I need to go out a specific dahdi span. Is there a good way to handle local prefixes within Asterisk without having to create the dialplan for each prefix? I would do some pattern matching, but none really are in a series and are pretty random. Calls outside these prefixes need to go out my other span. |
16:33.04 | *** join/#asterisk SupYoshi (SupYoshi@ip51cc8577.speed.planet.nl) |
16:33.13 | SupYoshi | Hi im having issues with ASterisk installation |
16:33.15 | jrad | To whom it concerns. I have started the memory leak debugging stated in the Asterisk wiki. We have to wait for the problem to reoccur, however this is one of the errors that has me concerned. http://pastebin.com/VwpRbsc8 |
16:33.18 | SupYoshi | I get errors when i try make install |
16:33.25 | SupYoshi | This is my pastebin of .configure |
16:33.27 | SupYoshi | http://pastebin.com/wkkK2xs0 |
16:35.01 | jrad | @SupYoshi - I am no pro. However, have you tried the command "make distclean" start fresh. Then I would refer you to Wiki of Asterisk. |
16:35.41 | SupYoshi | i just eh... |
16:35.43 | SupYoshi | I followed a guide |
16:35.58 | SupYoshi | I installed asterisk before with different guides, im just having this error now ;) |
16:39.32 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
16:40.47 | SupYoshi | MHm |
16:49.02 | SupYoshi | http://pastebin.com/PPQN8ezR |
16:49.05 | SupYoshi | better? -.- |
16:49.06 | SupYoshi | lol |
16:53.06 | p3nguin | techwerkz: No, you have to use dial plan to do it. |
16:55.00 | [TK]D-Fender | techwerkz, you could put them into a file and une func_shell or an AGI to get to them, or put them in a DB somehow (AstDB is fairly easy) and just use a quick lookup. Not a big deal not to have to make static dialplan for them. |
16:55.06 | p3nguin | techwerkz: It wouldn't be that hard to do. You could have one line of dial plan to "catch" each number called, and then use a macro to reduce duplicate lines of dial plan for every other line that would be identical. |
16:56.32 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
17:05.55 | SupYoshi | Can someone help me? |
17:07.06 | pabelanger | SupYoshi: what OS? |
17:08.07 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
17:08.26 | pabelanger | leifmadsen: qwell is on vacation I think |
17:08.45 | ChannelZ | That's odd, why does it say 'Killed' |
17:13.54 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
17:17.23 | *** join/#asterisk Joel_Oliveira (~chatzilla@estrela-adm.nortenet.pt) |
17:18.41 | leifmadsen | pabelanger: coolio thanks -- no worries I figured out my issue |
17:19.36 | SupYoshi | Centois 5 |
17:19.38 | SupYoshi | CEntos 5 |
17:20.33 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
17:21.28 | leifmadsen | SupYoshi: o.O |
17:22.45 | SupYoshi | CEntos5, I have the following error |
17:22.50 | SupYoshi | When I http://pastebin.com/wkkK2xs0 |
17:22.51 | techwerkz | [TK]D-Fender and p3nguin thanks for the ideas. I will just use AstDB |
17:22.53 | SupYoshi | Configure it |
17:27.13 | SupYoshi | help? |
17:30.14 | dijib | howdy yall |
17:30.23 | SupYoshi | hi |
17:32.10 | pabelanger | SupYoshi: what is your processor? |
17:32.14 | pabelanger | cat /proc/cpu |
17:32.28 | pabelanger | cat /proc/cpuinfo |
17:33.24 | pabelanger | SupYoshi: $ make distclean |
17:33.26 | pabelanger | ./configure |
17:33.27 | pabelanger | make |
17:33.43 | dijib | he has ld? just dirty? |
17:34.12 | SupYoshi | Intel(R) Xeon(R) CPU E5620 @ 2.40GHz |
17:34.19 | SupYoshi | did that |
17:34.23 | SupYoshi | i keep doing these things |
17:34.25 | SupYoshi | for 3 hours |
17:34.28 | SupYoshi | and I keep gettign the same error |
17:34.41 | SupYoshi | ake: *** [channels] Killed |
17:34.46 | SupYoshi | make: *** [channels] Killed |
17:34.58 | pabelanger | are you killing it? |
17:34.58 | leifmadsen | sounds like a problem with menuselect |
17:35.00 | SupYoshi | and i did distclean 3 times |
17:35.03 | SupYoshi | no im not killing it |
17:35.07 | pabelanger | then what is |
17:35.08 | SupYoshi | yes it is a problem with menuselect |
17:35.12 | SupYoshi | but ive no idea what it is |
17:35.13 | dijib | binutis missing? |
17:35.19 | leifmadsen | cd menuselect |
17:35.19 | dijib | binutils |
17:35.20 | leifmadsen | make clean |
17:35.41 | leifmadsen | honestly, just delete the entire asterisk source directory and re-extract or checkout from svn clean |
17:35.42 | SupYoshi | done that too |
17:35.43 | SupYoshi | 3 times |
17:35.47 | SupYoshi | oh i did that |
17:35.57 | leifmadsen | you're doing something wrong then |
17:35.59 | SupYoshi | even a Yum install did better |
17:36.02 | leifmadsen | tar zxvf asterisk-1.8.8.1.tar.gz |
17:36.06 | leifmadsen | cd 1.8.8.1 |
17:36.10 | SupYoshi | i actually got the yum install working |
17:36.13 | SupYoshi | if that matters |
17:36.14 | leifmadsen | ./configure && make && make install |
17:36.17 | leifmadsen | that's all you should have to do |
17:36.18 | SupYoshi | but then i reinstalled |
17:36.35 | leifmadsen | why are you mixing compiling source and package management? |
17:36.38 | SupYoshi | mkay i will just reinstall this VPS and start over |
17:36.46 | SupYoshi | Because I did one reinstall already |
17:36.50 | SupYoshi | :) Doing Yum |
17:36.57 | SupYoshi | however i messed up on mysql then (bad guide) |
17:36.59 | leifmadsen | well sounds like you've messed something up |
17:37.08 | leifmadsen | just use asteriskdocs.org |
17:37.41 | SupYoshi | whats the easiest OS? |
17:37.43 | SupYoshi | to install it on? |
17:38.03 | leifmadsen | doesn't matter |
17:38.05 | SupYoshi | ok |
17:38.07 | leifmadsen | instructions show ubuntu and centos |
17:38.14 | SupYoshi | centos 5 minimal or centos 5 normal better? |
17:38.14 | leifmadsen | pick whatever one you prefer |
17:38.23 | leifmadsen | follow the installation instructions |
17:38.27 | leifmadsen | it tells you |
17:38.42 | SupYoshi | k |
17:38.42 | pabelanger | SupYoshi: try ./configure CFLAGS=-fPIC" |
17:38.43 | pabelanger | make |
17:38.49 | leifmadsen | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-Install.html |
17:38.49 | pabelanger | err |
17:38.53 | pabelanger | SupYoshi: try ./configure CFLAGS="-fPIC" |
17:39.01 | SupYoshi | yea to alte |
17:39.02 | SupYoshi | lol |
17:39.08 | SupYoshi | im pissed off on asterisk today |
17:39.09 | SupYoshi | lol |
17:39.20 | leifmadsen | ok |
17:39.25 | SupYoshi | just reinstalling the box |
17:39.28 | SupYoshi | starting again |
17:39.30 | leifmadsen | it's not asterisk that's the problem |
17:39.36 | SupYoshi | no its me |
17:40.06 | dijib | could someone look at this PB and tell me why with the app_meetme.so module loaded the meetme app in dialplan is failing? http://pastebin.com/8XAKjQdY |
17:40.09 | leifmadsen | digiv_away: please don't change names when setting away |
17:40.45 | jrad | Concerning the possible memory leak. Is there a way to force a core dump instead of waiting for it to run out of all RAM. |
17:40.49 | *** part/#asterisk digiv_away (~textual@as1.si.umich.edu) |
17:41.11 | leifmadsen | jrad: you don't need it to coredump to get a backtrace |
17:41.16 | leifmadsen | you can connect to the running process |
17:41.44 | jrad | Thanks. |
17:41.51 | dijib | brb making coffee |
17:46.40 | *** join/#asterisk serafie (~erin@nat/digium/x-gqfuctsplptrwksf) |
17:52.15 | *** join/#asterisk timahvo1 (~rogue@197.178.29.45) |
17:52.45 | gruvfunk | any Python gurus in here? I need to strip the phone number from the SIP_FROM header ? |
17:53.12 | gruvfunk | current script is pulling the caller's NAME, but we need the NUMBER |
17:53.34 | gruvfunk | From: "CITY, NY" <sip:13475551212@10.9.9.5>;tag=as5ca29464 |
17:53.54 | gruvfunk | clid, uri = from_hdr.split(" <sip") |
17:54.17 | gruvfunk | split goes to the left, I think rsplit goes to the right, but I need to stop at "@" |
17:54.21 | gruvfunk | anyone help me? |
17:54.49 | gruvfunk | tried #python but that was well over my head, amazed really since there are so many people in there who would quickly solve this |
17:54.56 | leifmadsen | gruvfunk: I'd try #python or something as that is more of a python parsing question than an Asterisk question |
17:54.59 | [TK]D-Fender | gruvfunk, is that somehow not in the CALLERID? |
17:55.46 | gruvfunk | leifmadsen: yeah I did, concepts over my head and got RTFM |
17:55.49 | gruvfunk | [TK]D-Fender: ? |
17:56.29 | [TK]D-Fender | the FROM is typically parsed out into the callerid values for the channel. Are they not as they should be>? |
17:57.08 | *** join/#asterisk its_jeremy_ (~omghax@gateway/tor-sasl/itsjeremy/x-75806909) |
17:57.30 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
17:57.41 | gruvfunk | [TK]D-Fender: I guess this isn't so much an Asterisk setup question, this is a small local script that launches on calls received into Twinkle voip client - so it takes the sip header coming in |
17:58.31 | [TK]D-Fender | gruvfunk, Well if you're transfixed on a pre-made script which already breaks the header up then you're going to have to bite the bullet and learn how to program in python then. |
17:59.06 | gruvfunk | :-) right, was hoping somebody in here knows enough to point me in the right direction |
17:59.29 | gruvfunk | but tha'ts what #python is for i guess |
17:59.34 | gruvfunk | sigh |
18:06.00 | *** join/#asterisk glaz (strke@hiro.glaciuz.com) |
18:08.09 | glaz | I'm trying to get operator = yes to work in voicemail.conf but seems like asterisk is ignoring the DTMF inputs while playing the unavailable message. I've reloaded dialplan and voicemail. Any suggestion? |
18:09.57 | glaz | Looks like someone has open a bug request Today, https://issues.asterisk.org/view.php?id=14731 |
18:11.09 | glaz | Oops, that's just the current time/date. The ticket was open in 2009 |
18:13.29 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
18:13.35 | glaz | But it answered my problem anyway, I need the a extension to be created. |
18:19.40 | jrad | Thanks for your inputs. I've created official bugtracker @ https://issues.asterisk.org/jira/browse/ASTERISK-19596 |
18:19.47 | techwerkz | If the dial plan contains a Macro() does it continue to execute the next line after Macro(), or does it jump to the Macro and stay there? |
18:26.00 | [TK]D-Fender | glaz, Let us know if there's anything else we don't need to do for you ;) |
18:26.15 | [TK]D-Fender | techwerkz, "core show application MacroReturn" |
18:27.40 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
18:28.26 | glaz | [TK]D-Fender: :p |
18:39.53 | *** join/#asterisk YelBo (~YelBo@81.130.125.130) |
18:40.46 | YelBo | hey |
18:41.25 | glaz | ho! |
18:41.32 | YelBo | :) |
18:43.22 | autofsckk | hello, i want to use a voip provider to usewith my asterisk, any recomendations? i just want sip trunking with good price |
18:43.34 | techwerkz | [TK]D-Fender: Doesn't find that, in all reality I don't want it to return |
18:46.59 | ryduh | autofsckk: i use voip.ms and have been pretty happy with their service |
18:47.36 | techwerkz | [TK]D-Fender: Basically just want to make sure this will be ok: http://pastebin.com/WtbcpC3L or if I need to add something to stop the dial plan after the Macro commands. |
18:48.37 | *** join/#asterisk harovali1 (~harovali@r190-134-146-155.dialup.adsl.anteldata.net.uy) |
18:51.44 | harovali1 | hi, my call center provider uses asterisk to manage calls, and he lets me download the audio files of the incoming calls so that I can hear them, I do that thru a web of his, where the audio files can be downloaded. The problem is that I can hear the audo files, posibly because I'm missing the correct codec of file type. I couldn't determine which audio files are these. I tested renaming the files as .mp3, .avi, .wav, an |
18:52.44 | harovali1 | I used the 'file ' command in linux to see if the audio format could be guessed, but it just says it's binary data |
18:52.50 | harovali1 | any hint ? |
18:56.07 | autofsckk | ryduh: thanks ill take a look |
18:56.56 | vlt | harovali1: Assuming you *can’t* hear them: Asterisk’s native format is 8bit 8kHz PCM data. I thought nearly everything could play that. Maybe your provider uses some codec like OGG or weird things like AMR … |
18:57.21 | *** join/#asterisk jdoe (jdoe@falseprophet.ca) |
19:00.04 | SupYoshi | hello |
19:00.11 | SupYoshi | when i run my make and make install commands |
19:00.15 | SupYoshi | It breaks off with KILL |
19:00.23 | SupYoshi | There is no way I can install asterisk on CentOS 5 |
19:00.27 | SupYoshi | I have tried 5 times now |
19:00.31 | SupYoshi | Following different guides |
19:00.52 | *** join/#asterisk GVolkmann (~administr@rrcs-24-213-178-60.nyc.biz.rr.com) |
19:00.52 | harovali1 | vit: now he told me they are in GSM format |
19:01.14 | *** join/#asterisk micols (~t@rlogin.dk) |
19:02.14 | vlt | harovali1: Aah, gsm. Ok. Can your asterisk play the files? (That’s what I’d try first.) |
19:03.09 | harovali1 | vlt: I don't have asterisk installed, do you think installing asterisk to hear the files is my best bet under linux ? |
19:04.25 | *** join/#asterisk bmg505 (~leon@196-209-152-75.dynamic.isadsl.co.za) |
19:05.20 | *** join/#asterisk bmg505 (~leon@196-209-152-75.dynamic.isadsl.co.za) |
19:06.08 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
19:06.17 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
19:08.46 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
19:09.00 | [sr] | hellou my friends |
19:10.37 | jaytee | anyone have any good links for setting up just the Buddy Watch feature on Polycoms with Asterisk 1.6.2. I'm setting up a client with one Polycom 560 as the receptionist phone and another 19 Polycom 331s for everyone else. I've found a few articles but one is out of date and the others aren't walkthroughs so much as postings for people trying to figure out their own issues. I want to use the |
19:10.37 | jaytee | Buddy Watch feature with a softkey and not use up line keys for BLF. |
19:10.39 | vlt | harovali1: I just tried to play tt-weasels.gsm on Ubuntu with vlc. Worked. mplayer doesn’t. |
19:12.12 | jaytee | SupYoshi, installing asterisk on CentOS is fairly straightforward. What version of CentOS are you trying to use? |
19:12.44 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
19:13.42 | Naikrovek | jaytee: never done it with a softkey. efk is insanely frustrating. |
19:13.56 | Naikrovek | like "i NEED to kill someone right now" frustrating |
19:14.11 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
19:25.12 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
19:28.54 | mr_pete | o_O |
19:28.58 | mr_pete | just defined a class called [none] in musiconhold.conf |
19:29.08 | *** join/#asterisk bmg505 (~leon@196-209-152-75.dynamic.isadsl.co.za) |
19:29.12 | mr_pete | which uses a directory of /dev/null |
19:54.08 | *** join/#asterisk infobot (~infobot@rikers.org) |
19:54.08 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.2.1 (2012/03/15), 1.8.10.1 (2012/03/15), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
19:54.16 | [TK]D-Fender | danmuniz, As you said, you're new. Let our eyes filter it for you |
19:55.02 | danmuniz | thanks, have to run an errand for a couple of hours will you be on around 3:00 pst? |
19:55.21 | [TK]D-Fender | danmuniz, Share whatever you can, whenever you can, and whoever is available to help you will. |
19:56.30 | mr_pete | right |
19:56.32 | mr_pete | sort of working |
19:56.42 | mr_pete | locally I have a silent gsm file, and set the format in the moh.conf to gsm |
19:56.46 | mr_pete | it plays silence locally |
19:57.26 | mr_pete | yep, working on trunks too |
19:57.26 | *** join/#asterisk jsjc (~Adium@199.Red-79-150-67.dynamicIP.rima-tde.net) |
19:57.34 | mr_pete | I need to point to a dir with a file (no files doesn't work) |
19:57.44 | mr_pete | and given the silent file I used is in gsm format, I had to set format=gsm too :) |
20:00.15 | *** join/#asterisk ke-esc (~ke-esc@155.229.209.170) |
20:01.02 | ke-esc | Dump question- is there a rule that a * cannot appear at the beginning of an extension? _*XXX always gives me a fast busy, but if I change it to _XXX* and reload the dialplan it works fine |
20:02.43 | leifmadsen | ke-esc: no such rule exists -- most likely your phone is the problem |
20:03.10 | leifmadsen | ke-esc: change the phones dialplan to permit your 3 digit starcode to be sent to asterisk correctly |
20:04.00 | *** join/#asterisk jero (~boo@mtl.savoirfairelinux.net) |
20:05.48 | ke-esc | leifmadsen, hmmm...okay, thanks :) |
20:06.49 | jero | hi |
20:08.23 | Delido192123 | Please, i have this problem since few weeks, i want to get ccbs working, without call-limit 1 (because the Phoneuser can enable or disable callwaiting over an intranet) code http://pastebin.com/hqpk1CQ3 |
20:12.18 | jero | is it possible to disable presence status updates from the dialplan? I have a bunch of extensions and a polycom with lots of BLF hints, who don't like Pages on the mid/longterm because they cause too many simultaneous updates |
20:19.02 | *** join/#asterisk AutoStatic (~jeremy@5ED0E47D.cm-7-1d.dynamic.ziggo.nl) |
20:20.49 | Delido192123 | i use the latest asterisk LTS Version and Polycom IP 320 Phones with the latest SIP and Bootrom. used Dialplan is copied in pastebin ( http://pastebin.com/hqpk1CQ3 ) i used an phpagi get the status from callwaiting and others.. so if DEVSTATE INUSE the Caller become directly an CALLWAITING menu |
20:21.07 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
20:21.26 | Delido192123 | can i used asterisk Function CALLCOMPLETION to get it to work? |
20:21.32 | leifmadsen | Delido192123: with 1.8 you don't use call-limit -- you use callcounter=yes |
20:21.42 | leifmadsen | call-limit is deprecated |
20:23.38 | *** join/#asterisk tzanger (tzanger@wallace.mixdown.ca) |
20:24.02 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
20:24.10 | Delido192123 | leifmadsen: thanks for answer. callcounter=yes and limitonpeers=yes is set in sip.conf [general] |
20:24.59 | leifmadsen | ok |
20:26.11 | *** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za) |
20:26.34 | bent_screwdriver | is it best to use iax or sip for asterisk box to asterisk box calling? i have been rolling out with iax then using sip as a second option if iax fails for some reason but just wondering if it's worth the trouble to even setup the sip part... |
20:27.45 | vlt | bent_screwdriver: I have better experience with IAX (and trunking) when bandwidth is limited. |
20:34.49 | dijib | would anybody in here have any use for a sangoma a-200 ? im looking for a trade for ram or an hdd or something |
20:35.25 | dijib | ata ... smething |
20:35.29 | leifmadsen | bent_screwdriver: I always just use SIP everywhere and don't bother with IAX2 |
20:42.27 | bent_screwdriver | leifmadsen: thanks. particular reason or just preference? |
20:43.30 | leifmadsen | bent_screwdriver: never saw a reason to change technologies between locations as it just causes problems with features |
20:43.46 | leifmadsen | keeping it all on the same protocol tends to make things a lot easier to implement |
20:49.34 | Delido192123 | no one can help? what i need: The callwaiting can ondemand changed over webinterface; All User can do more then one Call (only for callforwarding); If is callwaiting off, and the called phone in use ccbs should work (this is my problem) |
20:52.48 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
20:53.27 | *** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za) |
21:06.06 | Delido192123 | it is posible to set an variable in dialplan, that let the dial command not called the called phonenumber? and get causecode back? i think ccbs only work in my dialplan if the dialcommand is executed |
21:17.01 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
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22:52.49 | SupYoshi | Hi |
22:52.58 | SupYoshi | can anyone tell me whats going on here now :) |
22:52.58 | SupYoshi | http://pastebin.com/bnuuRbcc |
22:53.11 | SupYoshi | I end up with this everytime :) Its so confusing..... annoying frustrating et.c |
22:53.17 | SupYoshi | make: *** [pbx] Killed |
22:54.00 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
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23:00.54 | *** join/#asterisk Antonjo (~Antonjo@46.183.121.39) |
23:01.18 | Antonjo | hello to all |
23:01.53 | *** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za) |
23:04.20 | Antonjo | can somebody help |
23:04.22 | Antonjo | ? |
23:06.04 | *** join/#asterisk neurosys_ (~neurosys@c-98-254-216-32.hsd1.fl.comcast.net) |
23:10.17 | navaismo | Antonjo, ask an you will see |
23:12.31 | Antonjo | i have a problem |
23:12.47 | Antonjo | when i do a call from a sip |
23:12.47 | Antonjo | i have 2 asterisk |
23:13.06 | Antonjo | when i do the call and send to sip 1 wich is asterisk 1 |
23:13.10 | Antonjo | i got these message |
23:13.19 | *** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za) |
23:14.46 | *** join/#asterisk Antonjo (~Antonjo@46.183.121.39) |
23:14.58 | Antonjo | sorry but i disconect |
23:17.23 | navaismo | np |
23:18.57 | navaismo | so the error is... |
23:19.40 | Antonjo | NOTICE[7483]: chan_sip.c:21870 handle_request_invite: Sending fake auth rejection for device |
23:19.58 | Antonjo | when i do a call |
23:20.03 | Antonjo | let me explain |
23:20.08 | Antonjo | i have to asterisk |
23:20.14 | Antonjo | asterisk 1 and asterisk 2 |
23:20.35 | Antonjo | i add a sip trunk to asterisk 2 |
23:20.50 | Antonjo | the user is of sip i create in asterisk1 |
23:21.08 | Antonjo | when i call from asterisk 2 the call go to the asterisk1 and i got the message |
23:21.12 | Antonjo | NOTICE[7483]: chan_sip.c:21870 handle_request_invite: Sending fake auth rejection for device |
23:21.13 | navaismo | that is a notice not an error |
23:22.04 | Antonjo | ah ok |
23:22.07 | Antonjo | any idea? |
23:22.13 | [TK]D-Fender | Idea about what? |
23:22.19 | [TK]D-Fender | We dont see a problem yet. |
23:22.32 | Antonjo | but the call is not execudet |
23:22.48 | Antonjo | and i drop the call |
23:22.49 | [TK]D-Fender | Antonjo: PASTEBIN an actual failed call for us to look at. |
23:22.56 | [TK]D-Fender | ~pb |
23:22.56 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
23:22.57 | [TK]D-Fender | ^^^ |
23:23.09 | [TK]D-Fender | Antonjo: "sip set debug on" <---------- |
23:23.22 | [TK]D-Fender | Antonjo: Make sure to have SIP DEBUG enabled |
23:24.10 | Antonjo | this is from asterisk 2 when i connect my phone |
23:24.25 | Antonjo | <PROTECTED> |
23:24.25 | Antonjo | <PROTECTED> |
23:24.25 | Antonjo | <PROTECTED> |
23:24.25 | Antonjo | <PROTECTED> |
23:24.25 | Antonjo | <PROTECTED> |
23:24.26 | Antonjo | <PROTECTED> |
23:24.26 | Antonjo | <PROTECTED> |
23:24.34 | Antonjo | this is from asterisk when i have the trunk |
23:24.38 | Antonjo | [Mar 27 01:28:53] NOTICE[7483]: chan_sip.c:21870 handle_request_invite: Sending fake auth rejection for device "Laxon 1001" <sip:1001@46.183.121.39:1075>;tag=as2557bebc |
23:24.38 | Antonjo | dc-asterisk01*CLI> |
23:25.21 | navaismo | Antonjo, use PB |
23:25.39 | Antonjo | what is PB |
23:25.39 | navaismo | and your trunk seems dead |
23:25.39 | Antonjo | ? |
23:25.47 | navaismo | ~pb |
23:25.47 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
23:26.14 | navaismo | show us the output of: sip show peers and use PB |
23:27.13 | Antonjo | *****3007 ****** N 5060 OK (2 ms) |
23:27.20 | Antonjo | this is the trunk |
23:27.51 | navaismo | but you are using "mc" |
23:28.40 | Antonjo | ues mc is the trunkname |
23:28.41 | [TK]D-Fender | Antonjo: I asked you to show us the actual call attempt |
23:28.56 | Antonjo | yes this is the actual call atempt |
23:29.00 | [TK]D-Fender | Antonjo: Antonjo That does not mean anything will work. |
23:29.18 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176145006.dsl.bell.ca) |
23:29.19 | [TK]D-Fender | No, it isn't. There is no pastebin, in which there should be DOZENS of lines of debug for us to see |
23:30.02 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-30-40.bois.qwest.net) |
23:32.16 | Antonjo | ok this is the log of the asterisk who i have a user to trunk to asterisk 2 |
23:32.24 | Antonjo | [Mar 27 01:36:35] NOTICE[7483]: chan_sip.c:21870 handle_request_invite: Sending fake auth rejection for device "Laxon 1001" <sip:1001@46.183.121.39:1075>;tag=as6eb16fa0 |
23:32.51 | Antonjo | this is the asterisk 2 logs when i try to make a call |
23:33.52 | Antonjo | <PROTECTED> |
23:33.52 | Antonjo | <PROTECTED> |
23:33.52 | Antonjo | <PROTECTED> |
23:33.52 | Antonjo | <PROTECTED> |
23:33.52 | Antonjo | <PROTECTED> |
23:33.52 | Antonjo | <PROTECTED> |
23:33.52 | Antonjo | <PROTECTED> |
23:33.53 | Antonjo | <PROTECTED> |
23:33.53 | Antonjo | <PROTECTED> |
23:34.12 | *** join/#asterisk brian98 (~brian98@188.141.12.34) |
23:34.57 | navaismo | USE PB |
23:35.08 | navaismo | or you will be banned by the admins |
23:35.40 | navaismo | and still going to MC trunk wich appear offline |
23:35.47 | navaismo | show us your sip show peers |
23:36.00 | navaismo | and use PB |
23:36.20 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
23:36.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
23:36.39 | Antonjo | i translate whith pb but is the same |
23:36.43 | Antonjo | :) |
23:37.20 | navaismo | i know but: " A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel." |
23:37.44 | Antonjo | == Using SIP RTP TOS bits 184 |
23:37.44 | Antonjo | <PROTECTED> |
23:37.45 | Antonjo | <PROTECTED> |
23:37.45 | Antonjo | <PROTECTED> |
23:37.45 | Antonjo | <PROTECTED> |
23:37.45 | Antonjo | <PROTECTED> |
23:37.45 | Antonjo | <PROTECTED> |
23:37.46 | Antonjo | <PROTECTED> |
23:37.46 | Antonjo | <PROTECTED> |
23:37.52 | Antonjo | this is the translate of PB |
23:38.30 | navaismo | facepalms |
23:40.32 | *** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za) |
23:40.45 | Antonjo | any idea of this message |
23:40.46 | Antonjo | ? |
23:41.22 | [TK]D-Fender | Antonjo: Meaningless because you did not follow any of the directions you were gien on this. |
23:41.41 | [TK]D-Fender | Antonjo: Enable SIP DEBUG as I told you. and then PASTEBIN it. www.pastebin.com <--- |
23:42.08 | [TK]D-Fender | Antonjo: give us the LINK to go read what you submit. Do not ever flood in here like you just did twice. |
23:51.35 | Antonjo | what links |