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| 00:54.47 | ruben23 | hi guys |
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| 02:47.08 | burnbrighter | Can anyone tell me if the cisco presence patch wipes out your existing configurations? |
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| 03:35.28 | [TK]D-Fender | What patch is this? |
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| 05:02.14 | maani | hi,Is it possible I call A put it on hold and then call B , and hang up myself but A and B continue speaking with each other? |
| 05:02.59 | [TK]D-Fender | maani: Not by putting them on "hold" |
| 05:03.50 | [TK]D-Fender | maani: Some SIP phones support taking a 3-way call and re-inviting the ends on hangup. Polycom does, perhaps some others. |
| 05:04.17 | maani | do you know softphone doing that ? |
| 05:04.31 | [TK]D-Fender | nope. |
| 05:05.00 | [TK]D-Fender | You could get just about the same effect by parking the first call, and then transferring the other caller to that lot |
| 05:05.11 | TazzNZ | you could make a conference....but that doesn't have the "same" effect |
| 05:05.54 | maani | actually I want to test this feature of Asterisk : https://reviewboard.asterisk.org/r/521/ |
| 05:06.53 | TazzNZ | firstly - that is ISDN |
| 05:08.11 | maani | I want to test it on ISDN(PRI) but at first asterisk should bridge A and B and then try ECT |
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| 05:09.17 | TazzNZ | and, imho - you do that with transfer |
| 05:09.23 | TazzNZ | not hold |
| 05:09.27 | TazzNZ | but !!! |
| 05:09.38 | TazzNZ | I do recall this working like this when we had an ISDN PBX |
| 05:10.10 | TazzNZ | I don't quite understand the second part of the testing |
| 05:11.11 | TazzNZ | but, from what I can see - this should "just" work if you on a version that has this patch |
| 05:11.14 | TazzNZ | and the calls are isdn |
| 05:11.26 | TazzNZ | the patch, imho, shows nothing about any other tech |
| 05:11.32 | TazzNZ | like SIP/IAX/etc |
| 05:11.36 | maani | I used transfer with many softphones, transfer is ok and A and B can talk but when I hangup A and B also disconnect |
| 05:12.42 | TazzNZ | I think you are doing a softphone conference |
| 05:12.46 | TazzNZ | not a transfar |
| 05:13.00 | TazzNZ | either way - imho, the patch clearly shows pri |
| 05:13.08 | TazzNZ | which means that it won't work on a softphone |
| 05:13.19 | TazzNZ | but I think it's best if one of the dev's clears that up |
| 05:13.29 | TazzNZ | or atleast, confirm what I am saying |
| 05:15.41 | maani | http://asteriskfaqs.org/2011/12/08/asterisk-users/libpri-isdn-feature-ect-explicit-call-transfer.html |
| 05:27.09 | [TK]D-Fender | setting "transfer=yes" in chan_dahdi is enough to take 2 channel bridged and .... |
| 05:27.11 | [TK]D-Fender | ~2bct |
| 05:27.11 | infobot | [~2BCT] 2BCT (2 B-Channel Transfer) allows a call coming in over DAHDi and back out again to the same telco to be handed off freeing the channels from your circuit. To enable this (if your carrier supports it) add "transfer=yes" to your channel configurations. |
| 05:33.20 | maani | infobot : thx, my carrier support it, according to your comments it seems all bridged calls should be bypassed( may be for some calls we need still monitor channel) I want to more control ,manually do A and B bridge, hangup myself to see Asterisk will initiate ECT ? so I asked a softphone doing this |
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| 07:55.51 | Nunne | I'm having some troubles getting my ssl-certificate to work with sip.conf. It's a wildcard SSL-certificate, so maybe the problem is there? I get "error loading certificate" on a sip reload. |
| 07:56.58 | Nunne | it's a .pem file contaning both private key (on top). and crt below and below the crt it contains certificate chains (gd_bundle/godaddy). |
| 07:57.44 | Nunne | I have also downloaded godaddy ca root certificate and added to the tlscafile |
| 07:59.38 | Nunne | anyone have experience with using wildcard ssl certificate for asterisk? the certificate was first created with openssl for use with apache.. but i'm successfully using it with both apache and iis/exchange so it shouldn't be anything wrong with the cert it self. |
| 08:00.15 | Chainsaw | Nunne: Asterisk until very recently did not load certificate chains correctly. |
| 08:00.29 | Nunne | I'm using asterisk 11.10 |
| 08:00.59 | Nunne | But do I even need to load the chains? Because I have tried both with and without the chains in the .pem |
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| 09:06.16 | Nunne | I got it working by adding all chains + cas into the .pem and just not using cafile. |
| 09:07.56 | Nunne | (trying to get WebRTC with sipML5 working). Now i get "process_sdp: Rejecting secure audio stream without encryption details" - anyone have any ideas? |
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| 09:28.29 | Nunne | apperantley ssl is not working.. I just thought so because sip reload didn't give me an error once. but it gave me an error now :( |
| 09:35.35 | Chainsaw | Nunne: And you're using the correct port? |
| 09:36.34 | Nunne | Chainsaw: sip reload shows an error on loading ssl certificate. So i guess this is the reason I see the "Rejecting secure audio stream...." as well. |
| 09:37.02 | Nunne | But I just don't get why it doesn't like my cert. Opening it on my computer shows that it's a valid cert and I can see the CA, chain + my cert in there. |
| 09:37.44 | Chainsaw | Nunne: My PEM has the certificate, the intermediate certificate and then the private key. |
| 09:38.00 | Chainsaw | Nunne: You shouldn't need the CA/root. |
| 09:38.14 | Chainsaw | That's in the root store, after all. |
| 09:38.15 | Nunne | Chainsaw: I have tried just having the key, cert. key, cert, intermediate and key, cert and ca. |
| 09:38.29 | Chainsaw | Nunne: Order matters. Certificate, intermediate, key. |
| 09:38.40 | Chainsaw | Nunne: All in one PEM. |
| 09:38.42 | Nunne | key last? |
| 09:38.53 | Nunne | all other info I have seen on this puts the key on top |
| 09:38.58 | Chainsaw | Nunne: Key last. |
| 09:39.12 | Chainsaw | Nunne: And you only use tlscertfile |
| 09:40.38 | Nunne | Chainsaw: putting the key last might have worked for me :) (but not toooo hopefull yet).. Because once before it didn't give me an error on sip reload. |
| 09:41.02 | Chainsaw | pink*CLI> sip reload |
| 09:41.02 | Chainsaw | SSL certificate ok |
| 09:41.06 | Nunne | buuut. reloading the http module and accessing that through https gives me no tls error: 5 like I got before.. so I'm hopefull :) |
| 09:41.09 | Chainsaw | has certificate, intermediate, key |
| 09:42.13 | Nunne | Chainsaw: I had to touch the sip.conf file for it to reload.. still ssl error :( |
| 09:42.41 | Chainsaw | Nunne: Suggests that either you don't have a valid certificate (i.e. self-signed) or you haven't done what I asked. |
| 09:43.01 | file | "openssl verify" might tell you stuff |
| 09:43.58 | Chainsaw | file: Moans about the lack of root in mine, so that isn't perfect. |
| 09:44.54 | Nunne | Chainsaw: I removed the CA part from the .pem. I only have crt,chain and key. |
| 09:44.57 | file | moan moan moan |
| 09:45.03 | Nunne | the certificate loads perfectly for the http module |
| 09:45.13 | Nunne | and I use the certificate elsewhere for apache and iis |
| 09:45.22 | Nunne | it's from godaddy so it's not self signed |
| 09:45.30 | Nunne | but it's a wildcard certificate, however. |
| 09:45.31 | eirirs | /exec -o file moan |
| 09:45.51 | Chainsaw | Nunne: Mine is a GlobalSign wildcard. |
| 09:47.11 | Chainsaw | Nunne: And I just rebuilt it from a newer (issued later) version of it to prove a point. Still loads fine. |
| 09:47.59 | Nunne | Chainsaw: well, I have no idea why it should work on my other servers.. on the same server in http.conf (ssl loading ok). but not in sip.conf :/ |
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| 09:48.34 | Chainsaw | Nunne: Well Asterisk previously did it wrong, but it was patched for the version you tell me you're running. |
| 09:50.02 | Nunne | Chainsaw: downloaded the lastest version last night. Uninstalled the previous version (11.2.1). The only remaining part of 11.2.1 should be the config files. |
| 09:50.14 | Nunne | Asterisk 11.10.0 built by root @ antares1 on a x86_64 running Linux on 2014-06-11 18:56:08 UTC |
| 09:52.26 | Nunne | holy crap... i'm a retard :D |
| 09:52.57 | Nunne | no wonder loading a certificate called wild_on.pem doesn't work if you try to load on_wild.pem. to little coffee today |
| 09:53.36 | Chainsaw | Nunne: More coffee will make it better. |
| 09:53.41 | file | or fire |
| 09:53.43 | file | BURN ALL THE THINGS |
| 09:53.43 | Chainsaw | Nunne: Did you get the "OK" from it? |
| 09:54.08 | Nunne | Chainsaw: yup.. been staring blind in to it for like 2 hours.. and it was a "blindness" misstake |
| 09:54.22 | file | Fun fact: the PJSIP support in Asterisk 12 is going to get the ability to persist subscriptions across restarts in a bit - meaning if you restart Asterisk it will remember the subscriptions from phones! |
| 09:54.36 | Chainsaw | file: That is pretty neat. |
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| 09:57.09 | Nunne | Chainsaw: I sip reload should do it? Or will I need to restart to get SRTP to work? |
| 09:57.39 | Chainsaw | Nunne: sip reload should suffice. |
| 09:57.48 | Chainsaw | Nunne: Keep in mind that the encrypted SIP is on TCP port 5061 |
| 09:58.16 | Chainsaw | Nunne: Which you can confirm as open using netstat -a |
| 09:58.29 | Nunne | Chainsaw: Still getting errors trying to use WebRTC. But not lunch so will look into it after :) |
| 09:58.40 | Chainsaw | Nunne: Yes, you deserve a break. Enjoy :) |
| 09:58.42 | skrusty | file: nice, not used pjsip stack yet in 12 |
| 09:58.55 | file | skrusty, it goes vroom |
| 09:59.01 | skrusty | :] |
| 09:59.10 | Nunne | i have sip-tls on LISTEN :) well, off to lunch :) |
| 09:59.12 | Chainsaw | skrusty: But why would you bother with 12 if not to use PJSIP? It's where all the innovation is. |
| 09:59.25 | skrusty | ari framework development |
| 09:59.27 | file | it's also getting resource list subscriptions, so a phone can send a single subscribe to Asterisk and actually subscribe to multiple things |
| 09:59.29 | Chainsaw | skrusty: Ah, okay. |
| 09:59.46 | skrusty | Chainsaw: https://asternetari.codeplex.com/ :) |
| 09:59.49 | file | (which scales better) |
| 09:59.58 | skrusty | file: nice |
| 10:02.04 | skrusty | reminds me, need to play with talk detection this weekend, still not pulled the latest and had a play :) |
| 10:03.23 | file | skrusty, :D |
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| 11:33.44 | eirirs | tell me when therre are speech to text plugin |
| 11:33.44 | eirirs | :P |
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| 11:46.00 | file | the change... is in, persisted subscriptions yay! |
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| 12:01.14 | Nunne | Chainsaw: Do you know any good method for finding what might be wrong with srtp? :D getting "Matched device setup to use SRTP, but request was not!". |
| 12:01.39 | Chainsaw | Nunne: I do not use SRTP at this time, sorry. |
| 12:02.27 | Nunne | Getting some weird 401 Unauthorized.. hmmm. |
| 12:03.01 | Nunne | Chainsaw: I get the feeling i need SRTP. Otherwise WebRTC shouldn't work? Am I thinking correctly? :) |
| 12:03.14 | Chainsaw | Nunne: I do not use WebRTC either, that's the thing. |
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| 12:11.53 | tuxx- | hja |
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| 12:20.28 | Nunne | Chainsaw: Seems like my SRTP is "working". But lastest Chrome (and i think Firefox) uses DTLS-SRTP which support in Asterisk seems to be shady at the moment? Because I can use a sip client with SDES but not DTLS-SRTP |
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| 12:20.58 | Chainsaw | Nunne: I do not use WebRTC so I will not be able to advise you on this subject. |
| 12:21.44 | file | Chainsaw is wise. |
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| 12:41.23 | aurs | a bit off topic, but I'm looking for DID number in Serbia, anyone know where I can find that? |
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| 13:18.42 | sekil | can one use ReceiveFAX with digium card in 11 ? |
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| 13:54.55 | newtonr | sekil, yes |
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| 14:19.10 | qakhan | if i statically add (member = Agent/9301) agent in a queue i dont get calls |
| 14:20.19 | qakhan | i have to add agent in queue using AddQueueMember(washingtonflyer,Agent/9301@agent) |
| 14:20.45 | qakhan | then i get calls. |
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| 14:22.53 | qakhan | why i have to add queue member? |
| 14:23.11 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 14:23.21 | newtonr | qakhan, when adding statically, does "queue show" show the agent in the queue just the same as when adding dynamically? |
| 14:24.17 | qakhan | Agent/9301 (ringinuse disabled) (Unavailable) has taken no calls yet |
| 14:24.28 | qakhan | Agent/9301@agent (ringinuse disabled) (dynamic) (Invalid) has taken no calls yet |
| 14:24.51 | qakhan | yes it shows both |
| 14:26.32 | [TK]D-Fender | <PROTECTED> |
| 14:26.57 | [TK]D-Fender | Adding what you did was wrong and changes nothing for the other one working |
| 14:28.35 | qakhan | so AddQueueMember(washingtonflyer,Agent/9301@agent) syntax is wrong? |
| 14:29.23 | [TK]D-Fender | qakhan: I would think that " (Invalid)" would be enough of a clue |
| 14:29.36 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-bzlbslvgykjvnauh) |
| 14:31.41 | qakhan | [TK]D-Fender i statically added agents in queue. but i dont get calls on agent ext. until i use addqueuemember() |
| 14:32.18 | [TK]D-Fender | Doubt that highly |
| 14:33.19 | [TK]D-Fender | Your your actual proper member "(Unavailable)" ... is clearly telling you "Unavailable". |
| 14:33.52 | [TK]D-Fender | I'm not seeing you showing any call attempts or actually getting that Agent logged in in the first place. |
| 14:35.27 | *** join/#asterisk RikusW (~rikus@197.111.223.224) |
| 14:36.57 | *** part/#asterisk RikusW (~rikus@197.111.223.224) |
| 14:38.40 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 14:42.13 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:46.14 | qakhan | [TK]D-Fender here is my configs |
| 14:46.44 | [TK]D-Fender | qakhan: Didn't ask for configs... |
| 14:47.19 | [TK]D-Fender | qakhan: Go show me your agent logging in, a queue status dump to prove it,and a caller going to to the queue |
| 14:48.58 | qakhan | it has all |
| 14:49.00 | qakhan | please check |
| 14:52.25 | [TK]D-Fender | Check what? |
| 14:53.31 | qakhan | things which you are asking |
| 14:53.52 | [TK]D-Fender | Where? |
| 14:54.02 | [TK]D-Fender | You didn't provide a pastebin with everything I jsut asked |
| 14:55.20 | qakhan | http://pastebin.com/vsDETaJ8 |
| 14:56.41 | lordz_md | hi guys |
| 14:56.54 | lordz_md | my asterisk won't play MOH |
| 14:57.06 | lordz_md | it says in logs that there is no file |
| 14:57.13 | lordz_md | but the file is in place |
| 14:57.29 | lordz_md | can some one please advice where to look at? |
| 14:57.30 | [TK]D-Fender | lordz_md: Pastebin is your friend... show us the configs & files |
| 14:58.32 | [TK]D-Fender | qakhan: You are missing 2/3's of what I asked you for |
| 14:59.46 | qakhan | [TK]D-Fender this is how i am try to add agent in queue |
| 15:00.14 | [TK]D-Fender | qakhan: new pastebin with everything I asked for.... |
| 15:00.54 | qakhan | queues.conf http://pastebin.com/YtSXvyhz |
| 15:00.54 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:01.22 | [TK]D-Fender | [11:00]qakhanqueues.conf http://pastebin.com/YtSXvyhz <- did not ask for. don't care. |
| 15:01.24 | qakhan | extensions.conf http://pastebin.com/mZrynSWD |
| 15:01.33 | [TK]D-Fender | [11:01]qakhanextensions.conf http://pastebin.com/mZrynSWD <- same |
| 15:01.56 | qakhan | http://pastebin.com/jXNygAFD |
| 15:01.59 | qakhan | cli |
| 15:02.27 | [TK]D-Fender | [10:47][TK]D-Fenderqakhan: Go show me your agent logging in, a queue status dump to prove it,and a caller going to to the queue <----------------- |
| 15:02.46 | [TK]D-Fender | You STILL haven't shown 2/3 of what I asked for there. |
| 15:03.01 | qakhan | how i login agent? |
| 15:03.17 | [TK]D-Fender | qakhan: If you don't know the answer to that then you are dead in the water. |
| 15:03.31 | [TK]D-Fender | qakhan: You clearly don't understand chan_agent |
| 15:03.55 | qakhan | you mean AgentLogin() ? |
| 15:04.51 | [TK]D-Fender | qakhan: Yes. |
| 15:05.08 | qakhan | ok let me try this one |
| 15:07.26 | qakhan | http://pastebin.com/xuMCprN5 |
| 15:07.43 | qakhan | agent lgged in but call is not going to queue |
| 15:08.12 | qakhan | there is music playing |
| 15:08.42 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:09.00 | [TK]D-Fender | facepalms |
| 15:09.17 | [TK]D-Fender | I asked you for 3 simple things to provide IN-ORDER.... and I'm still not getting them. |
| 15:09.42 | [TK]D-Fender | [11:02][TK]D-Fender[10:47][TK]D-Fenderqakhan: Go show me your agent logging in, a queue status dump to prove it,and a caller going to to the queue <----------------- |
| 15:09.56 | [TK]D-Fender | Do them. in-order at CLI. |
| 15:12.04 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 15:14.20 | qakhan | [TK]D-Fender i am sorry, it is always hard to me to provide you info |
| 15:14.21 | qakhan | http://pastebin.com/SZ4bqnfQ |
| 15:14.45 | [TK]D-Fender | STILL missing 1 piece |
| 15:15.53 | qakhan | now which noe? |
| 15:16.00 | qakhan | one* |
| 15:16.04 | [TK]D-Fender | [11:02][TK]D-Fender[10:47][TK]D-Fenderqakhan: Go show me your agent logging in, a queue status dump to prove it,and a caller going to to the queue <----------------- |
| 15:16.23 | qakhan | it has all three |
| 15:16.33 | [TK]D-Fender | No,it doesn't |
| 15:17.45 | qakhan | there is agent login, |
| 15:18.37 | qakhan | there is queue show washin... which is showing agent is available. |
| 15:18.57 | [TK]D-Fender | and a caller going to to the queue <------------------------------------------ |
| 15:19.08 | qakhan | i placed a call to queue but call is not goint to queue |
| 15:19.14 | qakhan | that is the problem |
| 15:19.14 | [TK]D-Fender | Where's the caller going to the queue and FAILING? |
| 15:19.23 | [TK]D-Fender | WEHERE"S THE CALL? |
| 15:20.49 | qakhan | http://pastebin.com/sTpa7CT9 |
| 15:30.41 | Penguin | That's not how Queue() works. |
| 15:31.26 | Penguin | The call stopped at AgentLogin() like it should. Nothing else will happen. |
| 15:32.51 | Penguin | qakhan: ^ |
| 15:33.28 | [TK]D-Fender | Where is the caller that ENTERS the queue? |
| 15:33.53 | Penguin | He doesn't know how chan_agent, AgentLogin(), and Queue() are supposed to be used. |
| 15:34.38 | qakhan | [TK]D-Fender caller will enter after agentlogin, but call does not go after agentlogin |
| 15:34.44 | [TK]D-Fender | He doesn't understand "show me a caller going into the queue" |
| 15:34.52 | Penguin | qakhan: You aren't paying attention. |
| 15:34.52 | [TK]D-Fender | Caller != agent |
| 15:34.54 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 15:35.00 | [TK]D-Fender | Agent is ONE persone |
| 15:35.04 | [TK]D-Fender | caller is ANOTHER |
| 15:35.09 | Penguin | And Agent never enters Queue(). |
| 15:35.22 | Penguin | That is NOT how it works. |
| 15:35.48 | [TK]D-Fender | Where is your SECOND person who is going to go into the queue? |
| 15:36.06 | [TK]D-Fender | You can't have a person being a memeber... AND going into the queue waiting to be answered |
| 15:36.30 | Penguin | Let's summarize how it works. |
| 15:36.38 | [TK]D-Fender | You aren't both the person calling the complaint department AND the person answer those calls |
| 15:36.57 | [TK]D-Fender | John is a MEMBER of the queue |
| 15:37.11 | [TK]D-Fender | Mark is the CALLER that goes into the queue looking to talk to someone |
| 15:37.24 | Penguin | You create a queue in queues.conf. Make an agent channel a member of the queue. Log in the agent with AgentLogin(). Then callers will enter Queue() where Agent is a member. |
| 15:37.59 | Penguin | First problem: your dialplan is wrong. |
| 15:38.23 | Penguin | Second problem: you don't have dialplan for callers. |
| 15:39.17 | [TK]D-Fender | http://pastebin.com/jXNygAFD <_ he used to and just doesn't get it |
| 15:39.26 | [TK]D-Fender | he turned it into the login |
| 15:39.57 | Penguin | Oh, interesting. |
| 15:41.14 | qakhan | i alreay made a queue. what i am try to do is. when caller calls, Agent get login into queue and then call go to queue |
| 15:41.16 | [TK]D-Fender | qakhan: AgentLogin is for your AGENTS to dial into and wait for callers. |
| 15:41.25 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 15:41.31 | Penguin | qakhan: That isn't how it works. |
| 15:41.43 | Penguin | A call has to go to the queue only. |
| 15:41.47 | [TK]D-Fender | qakhan: Your CALLERS are the ones who have to reach Queue() to ENTER the queue to try to be answered by a member |
| 15:41.55 | Penguin | At a different time, the agent must be logged in. |
| 15:42.13 | Qwell | Penguin: I don't know... guy may have just come up with a brilliant method of handling support calls, by accident. |
| 15:42.34 | qakhan | but i am tryig to do it from caller end |
| 15:42.37 | Qwell | Penguin: 2 people need support, right? They get connected to *each other*. |
| 15:42.37 | [TK]D-Fender | Qwell: Yes, just send the caller to Echo() so they can talk the problem out with themselves |
| 15:42.47 | [TK]D-Fender | SOLVED |
| 15:43.07 | qakhan | agents doesnot pay attention to login before their shift |
| 15:43.13 | [TK]D-Fender | [11:42]qakhanbut i am tryig to do it from caller end <- you need TWO PEOPLE |
| 15:43.14 | Penguin | qakhan: The caller's only responsibility is to make a call and enter Queue(). |
| 15:43.16 | Qwell | Fire them then. |
| 15:43.41 | [TK]D-Fender | qakhan: someone who is a member that is WAITING for the call, and someone ELSE to call in and go to the queue |
| 15:44.15 | mjordan | Qwell: here at ITSolutions, we help *you* solve your own problem. You are the agent! |
| 15:44.31 | [TK]D-Fender | qakhan: Where is your SECOND call for that person coming in and trying to ENTER the queue()? |
| 15:44.40 | mjordan | Qwell: even better would be to alternate callers. First caller is an agent... second caller goes into the Queue... you can help each other out! |
| 15:44.42 | Penguin | If your agents won't log in and wait for callers, change the type of channel from Agent to something else that will ring their phone when it is on-hook. |
| 15:44.48 | Qwell | mjordan: That's what I'm saying. |
| 15:45.09 | mjordan | Qwell: we never get bad reviews! All bad reviews are *technically* against our customers |
| 15:45.15 | Qwell | lol |
| 15:45.29 | [TK]D-Fender | mjordan: That is precisely what is happening right here... |
| 15:45.37 | [TK]D-Fender | The customer is always WRONG. |
| 15:46.14 | mjordan | http://demotivators.despair.com/demotivational/apathydemotivator.jpg |
| 15:46.27 | [TK]D-Fender | mjordan: If you didn't link that... I was going to have to :) |
| 15:46.53 | [TK]D-Fender | mjordan: I archived all of theirs over a decade ago for posterity :) |
| 15:47.03 | [TK]D-Fender | POSTERity ;) |
| 15:47.31 | mjordan | I still have one of theirs on my bookshelf |
| 15:47.41 | qakhan | [TK]D-Fender i am using asterisk 1.4.x AgentCallbackLogin() to login agent before call go to queue. now i am moving to asterisk 11.x |
| 15:48.00 | qakhan | and i am try to do the same thing in asteisk 11.x |
| 15:48.16 | [TK]D-Fender | qakhan: AgentLogin is NOTHING like AgentCallbacklogin from 1.4 |
| 15:48.44 | [TK]D-Fender | qakhan: I told you before you need to add LOCAL CHANNELS as memeber if you want something that resembles that |
| 15:48.46 | Penguin | Just use a local channel as the queue member. |
| 15:49.11 | [TK]D-Fender | qakhan: And you don't even seem to understand the concept of a caller ENTERING the Queue in the first place. |
| 15:49.13 | mjordan | digs up leifmadsen's post |
| 15:49.16 | mjordan | http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbacklogin-to-standard-dialplan-methods-part-1/ |
| 15:49.29 | Penguin | Local channels will send calls to phones that are on-hook rather than to an agent who is waiting on a queue. |
| 15:49.33 | [TK]D-Fender | qakhan: You cannot test with ONE user trying to be both the CALLER *and* the MEMBER |
| 15:49.35 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 15:50.00 | [TK]D-Fender | qakhan: Stop doing ridiculous tests like this |
| 15:50.12 | mjordan | (ew, he used chan_agent) |
| 15:50.14 | [TK]D-Fender | qakhan: Queues are for connecting TWO different people. |
| 15:51.11 | [TK]D-Fender | qakhan: chan_agent is ONLY for AgentLogin now when means those memeber have to SIT on that actual call to answer incoming requests |
| 15:51.17 | Penguin | mjordan: Tried to use it, anyway. He doesn't understand how i works or what it is for. |
| 15:51.23 | Qwell | mjordan: You never got to see just how awesome agentcallbacklogin was. You're really missing out. |
| 15:51.27 | [TK]D-Fender | qakhan: Which is not what you want and not what we told you you would need to do. |
| 15:51.42 | mjordan | Qwell: All I know is we killed chan_agent |
| 15:51.46 | Qwell | did we? |
| 15:51.50 | mjordan | Qwell: we did! |
| 15:51.55 | Qwell | 12? |
| 15:51.56 | [TK]D-Fender | Qwell: AQM specifying the "named device" to use for logging is so much better |
| 15:51.57 | mjordan | yup! |
| 15:52.04 | Qwell | mjordan: replaced with putnopvut's stuffs? |
| 15:52.06 | mjordan | Totally removed. Dead. Finite. |
| 15:52.11 | mjordan | putnopvut + rmudgett |
| 15:52.15 | Qwell | fun |
| 15:52.19 | mjordan | Agent pool bridges |
| 15:55.24 | qakhan | so i have to use addqueuemember() |
| 15:58.09 | [TK]D-Fender | qakhan: It's what you were told ages ago |
| 15:58.18 | [TK]D-Fender | qakhan: AQM + Local CHannel |
| 16:01.14 | qakhan | thanks [TK]D-Fender |
| 16:14.35 | TazzNZ | time to go play with SS&....w00t ! |
| 16:14.38 | TazzNZ | LOL |
| 16:14.43 | TazzNZ | SS7 even |
| 16:14.50 | Penguin | Good ol' ss& |
| 16:23.16 | *** join/#asterisk zerick (~eocrospom@190.114.249.148) |
| 16:27.44 | *** join/#asterisk r00f (~r00f@94.204.49.17) |
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| 16:42.08 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 16:43.45 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:44.46 | leifmadsen | mjordan: I'd suggest pointing at the AsteriskDocs.org or even the 4th edition link which are more updated from that post |
| 16:44.55 | leifmadsen | I used that post as the basis for what is in "the book" |
| 16:44.56 | leifmadsen | just fyi |
| 16:46.23 | mjordan | I used Google, and alas, its top result was voip-info |
| 16:46.29 | mjordan | no way was I linking _that_ |
| 16:47.56 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 16:48.59 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 16:50.25 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 16:54.11 | Penguin | It won't help qakhan anyway. He wants the first caller who is headed for Queue() to login his agent because the agent is too lazy to login on his own. |
| 16:54.49 | Penguin | Because of that, I'd rather just have static members which are local channels leading to devices on-hook. |
| 16:56.19 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 16:57.00 | mjordan | or fire your agents |
| 16:57.15 | mjordan | technology only solves so many problems! |
| 16:57.18 | Qwell | <Qwell> Fire them then. |
| 16:57.22 | Qwell | way ahead of you! |
| 17:00.48 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 17:01.51 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 17:02.00 | [TK]D-Fender | Penguin: That's not what I read from him.... he just wants a functional equivalent to AgentCallbackLogin for * 11 |
| 17:02.38 | [TK]D-Fender | Penguin: While failing to understand that Agent's only have AgentLogin neft and is nothing like that |
| 17:02.44 | [TK]D-Fender | lest* |
| 17:02.47 | [TK]D-Fender | left* |
| 17:02.48 | [TK]D-Fender | GAH |
| 17:03.04 | Penguin | (1043.07) <qakhan> agents doesnot pay attention to login before their shift |
| 17:04.07 | Penguin | That dialplan he showed us where AgentLogin() was right before Queue() was an attempt to login his slacking agent when the caller is on his way into a Queue(). |
| 17:04.48 | Penguin | What he failed to realize was that the application cannot pick up the headset and put it on the agent's head. |
| 17:05.17 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:05.24 | Penguin | Hence the suggestion to dial the phone while it is on-hook. |
| 17:06.10 | Penguin | I suspect that the agent will pick up if the phone rings. Even if they were too lazy to login and wait in the queue for someone to call. |
| 17:06.58 | [TK]D-Fender | You know he reverses his terminology every few minutes... |
| 17:07.04 | Penguin | The other alternative is that you use the agent's login/logout for the time clock. If he doesn't log in and take calls, he doesn't get paid. |
| 17:07.19 | [TK]D-Fender | [11:47]qakhan[TK]D-Fender i am using asterisk 1.4.x AgentCallbackLogin() to login agent before call go to queue. now i am moving to asterisk 11.x |
| 17:12.17 | TazzNZ | anyone from digium here that happens to deal with your RPM repo ? |
| 17:14.07 | *** join/#asterisk navaismo (~navaismo@187-178-254-98.dynamic.axtel.net) |
| 17:15.23 | TazzNZ | (or someone that can poke that person with the sharp side of a truck :D ) |
| 17:17.07 | mjordan | TazzNZ: Well, that's one way to get support for something you want. |
| 17:17.44 | TazzNZ | hehe - yeah - I am about to phone it in |
| 17:18.03 | TazzNZ | I was taking a fat chance anyways :) |
| 17:18.42 | mjordan | What actual issue are you having? |
| 17:19.42 | TazzNZ | the firmware for the wcte13xp is old on the repo |
| 17:19.46 | TazzNZ | Oct 2013 |
| 17:19.48 | TazzNZ | iirc |
| 17:20.12 | TazzNZ | whch is generating : wcte13xp 0000:06:00.0: Existing firmware file dahdi-fw-te133.bin is version 6f0017, but we require 780017. Please install the correct firmware file. |
| 17:21.44 | TazzNZ | which - unconfirmed - will make this card stop working in AsteriskNOW v3 installs |
| 17:21.55 | mjordan | we can take a look at that - thanks |
| 17:22.51 | TazzNZ | I'll log it via the official routes ? |
| 17:23.04 | TazzNZ | or...more official :) |
| 17:28.00 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 17:28.32 | TazzNZ | ok, mjordan - the "dahdi" version and firmware version seems way out of sync on the repo |
| 17:30.29 | TazzNZ | and the firmware can be a later version....dunno why I am finding that odd |
| 17:31.08 | TazzNZ | can't* |
| 17:35.41 | TazzNZ | right - got it loading |
| 17:38.04 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 17:40.56 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 17:43.26 | puzzled | hi |
| 17:45.05 | paulc | HI! |
| 17:45.10 | paulc | is possibly too exciteable this morning |
| 17:47.09 | puzzled | more espresso! |
| 18:12.55 | *** join/#asterisk jpoz (~jpoz@207.173.72.195) |
| 18:21.55 | *** join/#asterisk lcx (~lcx@213.143.121.75) |
| 18:21.58 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 18:22.17 | lcx | hi, I have a weird issue with a carrier and asterisk. |
| 18:22.43 | newtonr | ~ask |
| 18:22.43 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:22.57 | lcx | the invite should look something like this: 43123123123@foo.bar |
| 18:23.17 | lcx | foo.bar does not resolve |
| 18:23.38 | lcx | if I enter the IP, then the carrier rejects the invites as they expect foo.bar |
| 18:24.03 | lcx | I tried setting host=foo.bar and outboundproxy=ip of carrier |
| 18:24.29 | navaismo | did you add the host and ip via /etc/hosts? |
| 18:25.02 | lcx | yes. that also didn't help. let me check the logs. |
| 18:25.57 | *** join/#asterisk hecatae (~philip@host-92-28-14-4.as13285.net) |
| 18:27.31 | lcx | ERROR[23544] netsock2.c: getaddrinfo("foo.bar", "(null)", ...): Name or service not known |
| 18:27.48 | lcx | WARNING[23544] acl.c: Unable to lookup 'foo.bar' |
| 18:28.40 | lcx | or could I send the invite to 4312341234@foo.bar@proxy_ip ? |
| 18:31.15 | *** join/#asterisk fireman_biff (~biff@208.0.98.13) |
| 18:32.44 | fireman_biff | When my PBX attempts to start an IAX2 call the remote PBX sends a calltoken, but my PBX doesn't send it back. Any ideas on troubleshooting this? |
| 18:33.02 | Qwell | fireman_biff: What version of Asterisk are you using? |
| 18:33.58 | fireman_biff | Qwell: 1.8.20.0 locally, should be that version or similar on all remote PBXs |
| 18:34.23 | fireman_biff | I have another PBX with same asterisk version that's working |
| 18:35.47 | fireman_biff | actually the specific remote PBX I'm testing with right now is 1.6.2.13, let me try another test against a remote 1.8 |
| 18:36.16 | *** part/#asterisk hecatae (~philip@host-92-28-14-4.as13285.net) |
| 18:40.14 | TazzNZ | lcx: I would double check that host entry - I use that currently |
| 18:40.38 | fireman_biff | Qwell: yeah, same problem with a remote 1.8.20.0 |
| 18:40.47 | lcx | you mean the host=foo.bar for the peer? |
| 18:40.57 | TazzNZ | no, the entry in /etc/hosts |
| 18:41.08 | lcx | or is asterisk caching something? |
| 18:41.11 | fireman_biff | local PBX sends a NEW message, remote response with a CALLTOKEN, but local sends another NEW without a calltoken |
| 18:41.14 | TazzNZ | if you do "ping foo.bar" what do you get ? |
| 18:41.22 | fireman_biff | they just repeat that a few times then the call ends |
| 18:41.35 | lcx | ping works. |
| 18:41.39 | TazzNZ | lcx: I don't recall ever having a caching issue on asterisk |
| 18:41.49 | TazzNZ | but it might have been that I didn't look |
| 18:42.07 | TazzNZ | if you do a "core reload" in asterisk CLI ? |
| 18:42.31 | lcx | I just did a restart. will try a new call. |
| 18:45.24 | lcx | weird. looks better now. but somehow I am receiving a retransmission timeout now. |
| 18:45.38 | fireman_biff | is 'module reload chan_iax2.so' the most you can do to reset IAX2 without actually stopping asterisk? |
| 18:46.48 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 18:48.10 | file | it reloads the configuration, it doesn't "reset" it - it's not the same as an unload/load or a restart |
| 18:49.23 | qakhan | there any app or function in dialplan which can find string in multi-value veriable, like $var=newyork,Boston,Delaware |
| 18:49.32 | fireman_biff | file: thanks, so is there anything I can do to "reset" iax2 without stopping asterisk? (or does that question not even make sense?) |
| 18:49.46 | file | fireman_biff, define reset |
| 18:49.59 | file | unloading and loading it is the same as restarting Asterisk pretty much |
| 18:50.26 | fireman_biff | file: IAX2 is not working and reloading doesn't make a difference, but if I restart the PBX it will work. I'm hoping for something less disruptive |
| 18:50.39 | file | it may or may not let you unload it |
| 18:50.52 | file | depends on the internal state of chan_iax2 |
| 18:52.00 | fireman_biff | ok, I'll try the unload/reload, if it doesn't unload does it tell me it didn't unload? |
| 18:52.11 | TazzNZ | qakhan: have you looked at http://www.voip-info.org/wiki/view/Asterisk+functions |
| 18:52.38 | qakhan | yes but could not find any |
| 18:52.47 | TazzNZ | I can see CUT |
| 18:53.02 | TazzNZ | CUT: String parsing, based upon a delimiter. (1.2) |
| 18:53.06 | file | fireman_biff, yes although depending on the internal state it may get... unhappy |
| 18:54.32 | fireman_biff | alright, well it just told me what it was unregistering, no errors or anything |
| 18:54.41 | fireman_biff | now that I've reloaded it I can make IAX2 calls again |
| 18:55.12 | fireman_biff | file: this seems to be happening every day or two now though... how would I go about troubleshooting to prevent it from reoccurring? |
| 18:57.55 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 18:58.00 | qakhan | ok |
| 19:00.38 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 19:03.06 | *** join/#asterisk BlackDex (~BlackDex@ori.vyus.nl) |
| 19:03.18 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
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| 19:07.34 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 19:10.00 | *** join/#asterisk kalz (~kalz@2602:fff6:f:1::d979:58e8) |
| 19:15.49 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 19:20.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:26.47 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
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| 20:00.52 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 20:00.53 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 20:01.09 | gusto | so |
| 20:01.21 | gusto | WIMPy: hi |
| 20:01.26 | gusto | WIMPy: how are you? |
| 20:05.54 | *** join/#asterisk zerick (~eocrospom@190.114.249.148) |
| 20:08.00 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:18.34 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 20:18.48 | *** part/#asterisk Champi (Champi@rootshell.fr) |
| 20:18.49 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 20:24.46 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 20:37.07 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-39-236.dynamic.qsc.de) |
| 20:38.20 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-wqaqkemprczsbbvg) |
| 20:38.21 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 20:40.54 | WIMPy | Hmm. Looks like I should stop using chan_alsa. It seems to have triggered something evil :-( |
| 20:41.14 | gusto | why are you using that in the first place? |
| 20:42.44 | WIMPy | Because I still try to use Asterisk. |
| 20:45.58 | gusto | no, that alsa channel |
| 20:46.06 | gusto | what is it for? i am not using that |
| 20:46.42 | WIMPy | Take a guess. |
| 20:47.25 | gusto | so you are using asterisk as a telephone? |
| 20:47.29 | gusto | isnt that a bad idea? |
| 20:47.37 | WIMPy | no |
| 20:47.44 | WIMPy | Don't know. |
| 20:54.10 | *** join/#asterisk Defraz (~Defraz@24-117-69-71.cpe.cableone.net) |
| 20:54.15 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.1 (2014/06/12), 1.8.28.1 (2014/06/12); Standard: Asterisk 12.3.1 (2014/06/12); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 20:59.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:06.59 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 21:10.26 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 21:32.22 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 21:35.35 | *** part/#asterisk bkruse (~Adium@64.89.97.127) |
| 21:49.07 | *** join/#asterisk beanie (~beanie@bgareth2.plus.com) |
| 21:50.17 | beanie | hello, please could you give me some pointers as to why I can hear anything when I 1) Dial my voicemail or any other feature 2) Receive a call but yet, callers that call me can hear me (I cannot hear them) |
| 21:50.30 | beanie | pointers to find out what is wrong :) |
| 21:50.33 | TazzNZ | beanie: NAT ? |
| 21:50.55 | beanie | yeah I struggle with NAT - I mean what things might need changing relating to NAT? |
| 21:51.10 | [TK]D-Fender | Your settings. |
| 21:51.13 | TazzNZ | you need to set external ip in sip.conf |
| 21:51.50 | [TK]D-Fender | Which he shouldn't be touching.... |
| 21:52.06 | [TK]D-Fender | ...because he's using FreePBX |
| 21:52.50 | TazzNZ | we should have an auto kick on this channel if people are in #freepbx too |
| 21:53.02 | beanie | .... |
| 21:53.06 | TazzNZ | not that that would have helped :) |
| 21:53.55 | beanie | quite a lot of the time the lines between whether a question should be asked in #freepbx or #asterisk are blurred depending on who you speak to so it is very easy to end up getting referred between the two for the same problem |
| 21:54.14 | TazzNZ | Under Settings/Advance SIP Settings, do you have the External IP set ? |
| 21:54.32 | beanie | ah bugs |
| 21:54.37 | beanie | I have no access to freepbx currently |
| 21:54.43 | beanie | as I am away from home |
| 21:54.51 | [TK]D-Fender | SSH doesn't care where you are |
| 21:54.57 | TazzNZ | yeah - I have had that too.....I have had Digium point to schoomecom and back again |
| 21:55.20 | beanie | TazzNZ - yeah its gets people a bit downhearted |
| 21:55.35 | [TK]D-Fender | [17:54]beaniequite a lot of the time the lines between whether a question should be asked in #freepbx or #asterisk are blurred depending on who you speak to so it is very easy to end up getting referred between the two for the same problem <- When 99% of things depend on basic configuration... FreePBX is doign your stuff for you and you have to play by its rules |
| 21:55.48 | [TK]D-Fender | You should never start here unless you do specifically know |
| 21:56.09 | beanie | TK -D-Fender - I've not had much luck with setting up tunnelling which I think was the idea for getting access to freepbx admin away from home |
| 21:56.38 | [TK]D-Fender | Forward SSH rto your server. Ass 1 setting to puptty. HTTP to your local port. DONE |
| 21:56.47 | [TK]D-Fender | add* |
| 21:58.59 | beanie | hmmm - I followed these instructions: http://howto.ccs.neu.edu/howto/windows/ssh-port-tunneling-with-putty/ |
| 21:59.02 | beanie | but I got a timeout |
| 21:59.45 | [TK]D-Fender | We aren't going to get to see exactly what you put it, are we? |
| 22:00.15 | beanie | how do you mean :-D I'll do it again and produce a screenshot if it's any use :) |
| 22:00.28 | TazzNZ | information is always usefull :) |
| 22:00.35 | TazzNZ | might not seem so to you |
| 22:00.36 | beanie | I see :) |
| 22:00.43 | TazzNZ | but we might spot something :) |
| 22:00.45 | beanie | TazzNZ - whatever works |
| 22:00.47 | beanie | :) |
| 22:01.33 | beanie | ok, i'll just follow these instructions again - TK Defender - not being pedantic, there were a few typos there - Ass1? |
| 22:02.07 | beanie | by "http" to local port, do you mean typing the address in the web browser :) |
| 22:02.19 | [TK]D-Fender | [17:56][TK]D-Fenderadd* |
| 22:02.23 | [TK]D-Fender | Correcte4d rigfht after |
| 22:02.26 | [TK]D-Fender | asdkajsdkf |
| 22:02.29 | TazzNZ | hehe |
| 22:02.34 | beanie | oh yeah - woops sorry man :) |
| 22:02.35 | TazzNZ | i hate days like that |
| 22:02.54 | [TK]D-Fender | [18:02]beanieby "http" to local port, do you mean typing the address in the web browser <- just go set it up and show us |
| 22:03.02 | beanie | ok :) |
| 22:03.38 | beanie | will the source port be 8080 again? |
| 22:03.57 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 22:05.52 | beanie | are those settings right so far -> http://snag.gy/ay4Fv.jpg |
| 22:07.29 | [TK]D-Fender | Is that your server IP in the destination |
| 22:07.36 | [TK]D-Fender | as in an ip DIRECTLY on that box? |
| 22:07.40 | beanie | yes :) |
| 22:07.58 | beanie | the box is on 192.168.1.104 |
| 22:08.02 | beanie | internally |
| 22:08.02 | [TK]D-Fender | Did you clear your whitelist firewall junk out of the way? |
| 22:08.13 | [TK]D-Fender | So yout server has 2 NIC's? |
| 22:08.17 | [TK]D-Fender | your* |
| 22:08.19 | beanie | yes :) stopped the chronjobs restarted it |
| 22:08.33 | beanie | so it doesn't kick back in with a box reset :) |
| 22:08.41 | [TK]D-Fender | Go prove the rules are gone |
| 22:09.12 | beanie | although your question i'm sure is perfectly reasonable, I don't understand it about the NIC's :) |
| 22:09.16 | [TK]D-Fender | and your destination should not be that IP anyway. |
| 22:09.23 | beanie | sure i'll go into the ip tables |
| 22:09.25 | [TK]D-Fender | You should point it to the localhost redirect |
| 22:09.35 | beanie | what the goodness is that? |
| 22:09.36 | [TK]D-Fender | 127.0.0.1:80 |
| 22:09.38 | [TK]D-Fender | ^^^^^ |
| 22:09.39 | beanie | aha |
| 22:09.48 | [TK]D-Fender | Networking 101 |
| 22:09.58 | [TK]D-Fender | I still don't see that rule added |
| 22:10.14 | [TK]D-Fender | So go actually add it |
| 22:10.23 | [TK]D-Fender | and show us |
| 22:12.52 | beanie | TK Defender - http://snag.gy/RIHZU.jpg |
| 22:13.42 | [TK]D-Fender | waiting on new putty screen, and attempt to use.... |
| 22:13.47 | [TK]D-Fender | 2 SS) |
| 22:16.32 | beanie | TK Defender - I'm in :) |
| 22:21.51 | beanie | TazzNZ - http://snag.gy/JCqHu.jpg |
| 22:22.28 | [TK]D-Fender | Congratulations |
| 22:22.46 | beanie | thank you to you D-Fender :) |
| 22:23.15 | [TK]D-Fender | as I said, this was 1 tiny little setting |
| 22:24.07 | beanie | so the issue now is I cannot hear anything on calls :) |
| 22:24.51 | beanie | it might be a good idea to do a test call at this point and share the output? |
| 22:24.52 | [TK]D-Fender | "ifconfig" |
| 22:24.55 | [TK]D-Fender | PB |
| 22:25.02 | beanie | ? |
| 22:25.10 | [TK]D-Fender | do it |
| 22:25.15 | beanie | do what sorry? |
| 22:25.19 | [TK]D-Fender | ifconfig |
| 22:25.27 | [TK]D-Fender | from your shell |
| 22:26.13 | *** join/#asterisk maani (~babak@2.178.165.102) |
| 22:26.13 | beanie | TK D-Fender - http://snag.gy/rYrEV.jpg :) |
| 22:27.06 | [TK]D-Fender | I asked you if your server had 2 NIC's and if that ip was directly on our server and you said "yes" |
| 22:27.21 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 22:27.22 | [TK]D-Fender | [18:07][TK]D-FenderIs that your server IP in the destination [18:07][TK]D-Fenderas in an ip DIRECTLY on that box? [18:07]beanieyes |
| 22:27.31 | [TK]D-Fender | that server does NOT have a public IP |
| 22:27.47 | maani | hi, Is it possible to play custome ringback tone in asterisk? |
| 22:27.51 | [TK]D-Fender | In your asterisk sip settings, make sure you set to DYNAMIC, not STATIC |
| 22:27.51 | beanie | oh right...sorry I misunderstood you - I thought you were talking about what I put in the putty settings |
| 22:28.09 | beanie | I have a static ip D-Fender? |
| 22:28.16 | TazzNZ | maani: for the person making the call ? |
| 22:28.16 | [TK]D-Fender | beanie: And make sure you have "directmedia=no" and "nat=no" in your trunk settings. |
| 22:28.33 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 22:28.36 | beanie | I have a static IP D-Fender :) |
| 22:28.41 | [TK]D-Fender | [18:27]beanieI have a static ip D-Fender? <- you put a host there.. is the IP actually static? Or is the domain name a "bonus"? |
| 22:29.06 | [TK]D-Fender | beanie: If it is really static on the WAN side, then you can leave it as static. |
| 22:29.13 | TazzNZ | maani: as in - I am making a call, and I hear ringing - you want to customize that ? |
| 22:29.14 | [TK]D-Fender | make the trunk settings, retest |
| 22:29.22 | beanie | ah yes - I can see the confusion - I think that, because the radio button is set to static, that address is ignored? |
| 22:29.23 | [TK]D-Fender | heads to the shower |
| 22:29.30 | maani | TazzNZ: yes , for example play music, crbt |
| 22:29.35 | [TK]D-Fender | remove the hostname |
| 22:29.41 | [TK]D-Fender | just use the IP field. |
| 22:29.41 | beanie | ok :) |
| 22:29.50 | [TK]D-Fender | And add the trunk settings |
| 22:29.52 | [TK]D-Fender | BRB |
| 22:30.05 | TazzNZ | maani: you can play on hold music instead of rining |
| 22:30.05 | beanie | what do you mean static on the WAN Side :) thanks as always D-Fender |
| 22:30.23 | TazzNZ | but I don't think you can play, say a .wav file |
| 22:30.35 | beanie | you'd have to convert it if I remember correct TazzNZ |
| 22:30.41 | beanie | I have a custom hold music |
| 22:31.02 | maani | TazzNZ: I want to play it im early media befor answer, like crbt service in mobile network |
| 22:31.03 | TazzNZ | yeah - barring the convert, I don't think you can replace the rining with X file |
| 22:36.15 | maani | one solution is just play a custome wav in early media and originate another channel to destination if destination answer then bridge two channel |
| 22:43.23 | [TK]D-Fender | beanie: Have you done the trunk changes yet or not? |
| 22:43.29 | [TK]D-Fender | I asked you to retest... |
| 22:43.42 | [TK]D-Fender | WAN-side = your internet connection. |
| 22:44.29 | file | originate all the things! |
| 22:47.05 | TazzNZ | lol file |
| 22:50.05 | TazzNZ | maani: yeah - imho, it's a lot of work for little gain ? |
| 22:50.28 | beanie | ahahaha - D-Fender - http://snag.gy/6rp5W.jpg that was already set up as you asked because you were the one that helped me originally with that :) |
| 22:50.41 | beanie | direct media = no nat =no |
| 22:50.48 | beanie | but its internal calls that are affected to |
| 22:51.08 | [TK]D-Fender | what dfo you have forwarded to your server? |
| 22:51.25 | [TK]D-Fender | what PORTS & PROTOCOLS from your ROUTER |
| 22:51.26 | beanie | all of the asterisk ports |
| 22:51.37 | [TK]D-Fender | Give me specifics. |
| 22:51.41 | beanie | they were set up originally - it used to work |
| 22:51.46 | beanie | and hasn't been changed since |
| 22:51.47 | [TK]D-Fender | Because I'm not going to trust any of this blindly |
| 22:51.53 | beanie | :-D probably wise |
| 22:51.55 | [TK]D-Fender | [18:51][TK]D-FenderBecause I'm not going to trust any of this blindly <- |
| 22:52.14 | beanie | although, I can't get access to my router being away from home unless you're aware of a way? |
| 22:52.28 | [TK]D-Fender | wrong... |
| 22:52.30 | beanie | 5060 comes to mind |
| 22:52.37 | [TK]D-Fender | guess what you just succeeded in doing? |
| 22:52.42 | beanie | and there was another |
| 22:52.43 | [TK]D-Fender | SSH TUNNELING <- |
| 22:52.56 | beanie | sure :-D but i have no idea how you connect to the router |
| 22:52.57 | [TK]D-Fender | so make one that points to your ROUTER's internal IP:80 |
| 22:53.08 | [TK]D-Fender | the same web-admin I'm sure you use while local |
| 22:53.15 | [TK]D-Fender | forward ANOTHER port |
| 22:53.17 | beanie | ok my router is at 192.168.1.254 |
| 22:53.32 | [TK]D-Fender | So go make the dest : thatip:80 |
| 22:53.40 | [TK]D-Fender | for another local port # |
| 22:53.45 | [TK]D-Fender | then you can get to it |
| 22:55.11 | beanie | TK Defender - can anything go in the server port? |
| 22:55.18 | beanie | source port* |
| 22:55.55 | SpaceInvaders | I found AsteriskDialer by Daniel Tryba searching Google Play. Is that a good dialer to start with for my first attempts at connecting to my new Asterisk server? |
| 22:56.24 | TazzNZ | SpaceInvaders: I think you are looking for a softphone ? |
| 22:56.39 | SpaceInvaders | Yes that runs on my droid |
| 22:56.54 | SpaceInvaders | Is AsteriskDialer not a softphone? |
| 22:58.19 | TazzNZ | SpaceInvaders: not quite |
| 22:58.30 | TazzNZ | it uses the AMI to start a phone *on* you Asterisk server |
| 22:58.38 | TazzNZ | look for linphone |
| 22:58.41 | TazzNZ | or sipdroid |
| 22:59.16 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 22:59.23 | TazzNZ | s/phone/phone call |
| 22:59.25 | [TK]D-Fender | beaYou are defining ports on the CLIENT PC. |
| 22:59.28 | [TK]D-Fender | aim sensible |
| 22:59.30 | [TK]D-Fender | 8081 |
| 22:59.32 | [TK]D-Fender | follow alon. |
| 22:59.36 | [TK]D-Fender | along* |
| 22:59.36 | beanie | ahhhhhh |
| 22:59.49 | beanie | in putty, can you only have one added at a time? |
| 22:59.52 | [TK]D-Fender | And hurry up, I'm here for only a few more minutes |
| 22:59.58 | [TK]D-Fender | no, you can add 50 |
| 23:00.06 | [TK]D-Fender | Add your new one save your profile. |
| 23:00.08 | [TK]D-Fender | Reconnect |
| 23:00.10 | [TK]D-Fender | FAST |
| 23:00.17 | SpaceInvaders | Thank you, TazzNZ |
| 23:00.42 | TazzNZ | np SpaceInvaders |
| 23:02.30 | SpaceInvaders | Hey TazzNZ, do you use the interface to Asterisk Manager? I just downloaded it to see what it does but I'm not sure what id/pw it needs |
| 23:02.40 | SpaceInvaders | the app on droid |
| 23:02.58 | TazzNZ | SpaceInvaders: what are you trying to do ? |
| 23:03.20 | SpaceInvaders | Just see what it does. It says it displays the status of your Asterisk server |
| 23:04.40 | beanie | TK-Defender - http://snag.gy/zRy6x.jpg |
| 23:05.27 | SpaceInvaders | I was curious what details it would show regarding my server (and maybe learn something as a result). |
| 23:05.28 | beanie | TK-Defender also http://snag.gy/Snnwl.jpg |
| 23:06.52 | [TK]D-Fender | beanie: pastebin /etc/asterisk/rtp.conf and all the INCLUDE's |
| 23:07.11 | TazzNZ | SpaceInvaders: in that case, create a user in manager.conf |
| 23:07.43 | SpaceInvaders | Thanks. When you said linphone... is that Linphone Video? |
| 23:08.03 | TazzNZ | SpaceInvaders: uhm.....I think it's the same thing |
| 23:08.10 | TazzNZ | I don't have andriod anymore |
| 23:08.17 | TazzNZ | or look for 3cx |
| 23:08.24 | TazzNZ | I think they have an andriod version too |
| 23:08.33 | SpaceInvaders | heh it's the only linphone listed I just wanted to check given my mistake w the Asterisk Dialer :) |
| 23:09.01 | beanie | tk defender - http://snag.gy/arItt.jpg |
| 23:09.27 | SpaceInvaders | 3cx Phone for Phone System v12? |
| 23:09.35 | TazzNZ | nope ! |
| 23:09.48 | SpaceInvaders | yeah just read that :) |
| 23:10.13 | TazzNZ | I think it's just called 3cxphone |
| 23:10.16 | TazzNZ | or 3cxphone6 |
| 23:11.03 | [TK]D-Fender | beanie: AND THE INCLUDED FILES |
| 23:12.14 | beanie | TK Defender - http://snag.gy/AZibj.jpg |
| 23:12.33 | SpaceInvaders | It may be gone. |
| 23:12.57 | TazzNZ | well that sucks - it's quite neat |
| 23:13.16 | beanie | TK D-Fender - http://snag.gy/A8vME.jpg |
| 23:13.19 | [TK]D-Fender | beanie: You have 10001-20000 forwarded |
| 23:13.24 | beanie | sure... |
| 23:13.25 | [TK]D-Fender | beaYou MISSED a port |
| 23:13.31 | beanie | oh right... |
| 23:13.34 | [TK]D-Fender | 10000 <------------- |
| 23:13.37 | SpaceInvaders | OH |
| 23:13.42 | beanie | dohhhhhhh |
| 23:13.59 | [TK]D-Fender | Fix retest, and then start showing calls with SIP debug enabled if that didn't clear it up. |
| 23:14.05 | TazzNZ | unless your rtp.conf starts at 10001 - *duck* :D |
| 23:14.07 | [TK]D-Fender | heads out for the night. |
| 23:14.46 | [TK]D-Fender | [18:51]beanieall of the asterisk ports <- now you see why I never trust that. |
| 23:14.52 | [TK]D-Fender | heads off |
| 23:15.35 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 23:16.30 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@179.52.52.110) |
| 23:16.36 | beanie | thanks D-Fender :) |
| 23:18.49 | SpaceInvaders | do you have to restart Asterisk after changing manager.conf or can you tell Asterisk to reload the configs? |
| 23:19.15 | SpaceInvaders | I just added an id in my manager.conf |
| 23:20.07 | TazzNZ | SpaceInvaders: core reload should do it |
| 23:20.15 | TazzNZ | I think there might be a manager reload |
| 23:20.18 | TazzNZ | not 100% sure |
| 23:26.00 | SpaceInvaders | I found a reference in the Definitive Guide--reload gets it |
| 23:27.02 | beanie | ah damn - just altered that port but I am being disconnected when I try and dial into voicemail *98 cannot hear anything |
| 23:34.25 | SpaceInvaders | Hey, I need enabled=yes in manager.conf for this to work, right? |
| 23:34.51 | SpaceInvaders | otehrwise it blocks everything that wants to talk to the ami, am I right? |
| 23:35.53 | SpaceInvaders | yep, that was it :D |
| 23:39.47 | TazzNZ | SpaceInvaders: yip :) |
| 23:40.25 | beanie | i understand that rtp issues can often be to blame for what i'm experiencing, anybody have a natural affinity with this kind of stuff to guide me further? |
| 23:42.27 | *** part/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 23:42.29 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 23:45.20 | TazzNZ | beanie: what version of asterisk ? |
| 23:46.27 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
