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| 02:17.25 | WIMPy | raden: Or do it like the telcos do and use CF. |
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| 04:03.07 | raden | WIMPy, ? |
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| 04:24.50 | greenwolf | is it difficult to get lemonvox ASR to work with asterisk? |
| 04:25.29 | greenwolf | i wanted my callers to be able to speak their entries...is there a huge learning curve for this? has anyone sucessfully done this? |
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| 04:54.21 | ChannelZ | heh, lemon |
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| 05:23.46 | greenwolf | anyone familiar with lemonvox and how to implement it with asterisk |
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| 05:31.20 | ChannelZ | s/lemon/lumen/ |
| 05:31.46 | ChannelZ | http://www.lumenvox.com/partners/digium/Asterisk.aspx |
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| 06:09.17 | v0lZy | morning |
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| 06:25.58 | The_Font | Can anyone help me with connecting a sipura spa 3000 with the asterisk server? |
| 06:28.03 | kaldemar | ~ask |
| 06:28.03 | 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. |
| 06:29.48 | The_Font | I would like to have my land line calls terminated to asterisk. The line 1 connection has to connect to asterisk so a regular phone can be used the same way as a softphone |
| 06:30.07 | The_Font | I am new to asterisk so some leads and maybe reading material for this is welcome |
| 06:31.07 | kaldemar | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceConfig.html http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html#DeviceConfig_id291081 |
| 06:31.15 | The_Font | Most outgoing calls have to be routed to several voip providers. Just a few calls have to be called trough the landline |
| 06:33.03 | The_Font | Thanks for these links. That keeps me busy for a few hours :) This site looks like a good start. |
| 06:33.24 | kaldemar | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id36059950 |
| 06:36.15 | The_Font | You gave me a nice start <kaldemar>. I am going to do some reading now. |
| 06:37.58 | ChannelZ | So what font are you? |
| 06:38.45 | The_Font | I am The Font lol |
| 06:39.59 | v0lZy | hehe :D |
| 06:40.14 | v0lZy | the mother of all fonts |
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| 06:41.18 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 06:41.28 | The_Font | heeej heehee |
| 06:49.59 | v0lZy | Guys, is it possible to call a phone and send DTMF while its still ringing and fetch that with Asterisk? |
| 06:51.24 | v0lZy | My idea is to ... eh, never mind, just realized it wont play well with call forwarding. |
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| 07:14.17 | libryder | hi.. i am trying to connect to the ami remotely but keep getting an auth failed but i can connect locally with the same credentials |
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| 07:15.00 | libryder | bindaddr = 0.0.0.0, webenabled = yes |
| 07:19.24 | libryder | i'm not getting anything in the asterisk log either |
| 07:26.47 | v0lZy | qucik question: [globals] should be at the top right? Can I have comments above it? |
| 07:28.06 | kaldemar | wrong. yes. |
| 07:28.37 | kaldemar | [general] above [globals]. |
| 07:30.59 | bulkorok | libryder: did you reload the manager-module after changing the options?! |
| 07:31.14 | libryder | bulkorok: yes i did |
| 07:31.27 | bulkorok | firewall?! |
| 07:32.05 | kaldemar | libryder: connect how? |
| 07:32.07 | libryder | i don't think so.. i'm on the intranet, can ssh into the machine, etc |
| 07:32.10 | v0lZy | oh, theres a [general] too... |
| 07:32.12 | v0lZy | didnt know |
| 07:32.54 | libryder | kaldemar: i'm using a 3rd party program called adhearsion |
| 07:33.35 | libryder | works locally, authentication failed remotely |
| 07:33.42 | kaldemar | libryder: do you have "enabled=yes" in manager.conf? |
| 07:34.04 | libryder | kaldemar: yes |
| 07:34.41 | kaldemar | and does adhearsion use the TCP or the HTTP connection? |
| 07:36.09 | libryder | https://gist.github.com/0b2f8dd91a98c31a637c |
| 07:36.25 | libryder | it uses TCP |
| 07:37.09 | kaldemar | also, what says "authentication failed"? is it really an authentication issue? do you see the failure in CLI with "manager set debug on"? |
| 07:38.02 | libryder | [2012-08-24 00:19:53] DEBUG Celluloid: [RECV] #<RubyAMI::Error message="Authentication failed", headers=> |
| 07:38.04 | kaldemar | you might want to look permit=0.0.0.0/255.255.255.0 more closely. |
| 07:38.37 | libryder | kaldemar: that's one thing i don't fully understand… how can i permit all? |
| 07:39.02 | kaldemar | permit=0.0.0.0/0.0.0.0 or don't deny anything at all. |
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| 07:40.48 | libryder | i tried that and permit = all… same thing |
| 07:42.24 | *** join/#asterisk Zerr0 (~Zerr0@188.162.52.250) |
| 07:43.18 | Zerr0 | hi |
| 07:45.01 | kaldemar | libryder: look at asterisk instead of adhearsion. enable manager debug and try to log in. |
| 07:45.15 | kaldemar | libryder: it will tell you if the ACL is the problem. |
| 07:46.35 | libryder | Aug 24 07:45:55] DEBUG[15555]: manager.c:1282 unref_mansession: Mansession: 0x8e96860 refcount now 2 |
| 07:46.56 | libryder | it's occurring even when i'm not trying to login |
| 07:47.41 | libryder | i don't get any notifications of login attempts |
| 07:47.57 | libryder | can i test the connection to the ami manually from the cli? |
| 07:47.59 | kaldemar | did you do "manager set debug on"? |
| 07:50.09 | libryder | yeah that message was the result of debug |
| 07:50.52 | libryder | it almost makes me think that i'm not actually getting through to the server |
| 07:50.59 | indika1978 | hi, |
| 07:51.01 | libryder | i should be seeing those connect attempts right? |
| 07:51.18 | kaldemar | libryder: no it wasn't. what you pasted was core debug, not manager debug. |
| 07:52.09 | libryder | how can i see manager debug? i do sudo asterisk -rvvvvv, type manager set debug on, then those messages start flooding in, two at a time |
| 07:52.11 | kaldemar | libryder: when you enable manager debug, it will print all connection attempts to CLI. |
| 07:53.33 | libryder | when i do 'telnet <ip> 5038' i connect, and it says 'Asterisk Call Manager/1.1….' but nothing shows up in the log |
| 07:54.09 | kaldemar | there's nothing to log at that point. that's the TCP connection only. then you start wrinting the login commands. |
| 07:54.16 | Zerr0 | got a problem with 11. using websocket transport, SIP channel. While trying to make a call, recieve incomplete SDP in 183. it ends with "m=video 0 RTP/SAVPF 100 101 102" and no more lines relative to video. now trying to do the same through UDP and make tcpdump. |
| 07:54.18 | kaldemar | *writing |
| 07:54.24 | kaldemar | action: login |
| 07:54.29 | kaldemar | username: test |
| 07:54.31 | kaldemar | secret: test |
| 07:54.40 | kaldemar | and an additional enter. |
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| 07:55.08 | libryder | authentication failed |
| 07:55.16 | libryder | and nothing in the console |
| 07:55.50 | libryder | also, thanks for that… i wasn't sure how to connect from the CLI |
| 07:56.26 | libryder | when i do the same exact thing locally, it works |
| 07:56.37 | kaldemar | libryder: if missing, add "notice" and "warning" to the "console =>" line in /etc/asterisk/logger.conf and do "logger reload". then try again. |
| 07:56.55 | kaldemar | sounds like you have an ACL issue. |
| 07:59.13 | libryder | any ideas where to start troubleshooting that? |
| 07:59.36 | libryder | maybe i should just reboot the server lol |
| 08:01.50 | kaldemar | no, don't do that. |
| 08:01.52 | libryder | could astmanproxy be an issue? |
| 08:02.25 | kaldemar | of course. |
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| 08:03.40 | libryder | ok, going to start playing there |
| 08:03.55 | libryder | thanks for your help so far |
| 08:08.19 | libryder | how do i disable astmanproxy? |
| 08:08.34 | libryder | i found my username in the config file but never remember touching this file |
| 08:08.37 | libryder | :-/ |
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| 08:11.35 | Zerr0 | have anybody used 'ws' transport in * 11 ? |
| 08:12.02 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 08:13.17 | Zerr0 | or i'm the first in my trouble? :) |
| 08:14.39 | wdoekes | one of the first ;) it's brand new |
| 08:14.50 | Zerr0 | know :))) |
| 08:15.00 | Zerr0 | now try to deal with it ^) |
| 08:15.33 | wdoekes | try running the 11 svn branch instead of beta1 |
| 08:15.40 | Zerr0 | tried :) |
| 08:15.42 | Zerr0 | the same |
| 08:15.47 | wdoekes | ok |
| 08:15.51 | wdoekes | then good luck :) |
| 08:15.53 | Zerr0 | problem is in SDP |
| 08:17.04 | Zerr0 | when using udp transport - everything seems to be ok |
| 08:17.48 | Zerr0 | maby i have to make a bugreport? |
| 08:18.18 | wdoekes | that sounds like a good idea: https://issues.asterisk.org/jira/ |
| 08:20.56 | Zerr0 | hope it'll be fixed this year :) |
| 08:21.50 | wdoekes | if the bug is easy to reproduce, then it probably will |
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| 08:38.06 | Zerr0 | posted a bugreport. |
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| 09:05.43 | hurdman | hi, i have 4 TE420 on a server, but i have to change and definitely fix the order of the span detection ( for a fail over configuration between 2 servers ), is it possible ? |
| 09:06.29 | indika1978 | hi, how to get a macro output to another dial plan context , without using GLOBLE valuable? |
| 09:07.43 | kaldemar | indika1978: save what you want into a variable of your choice. |
| 09:08.55 | indika1978 | thanks then how can i get it under other context |
| 09:08.57 | indika1978 | ? |
| 09:11.13 | kaldemar | just like any other channel variable. if you do Set(yourvar=value) in a macro, ${yourvar} will contain the value outside the macro as long as the channel is the same. |
| 09:12.22 | indika1978 | thanks for the detaild description. it is really helping me. Bcoz i am writing a call screening dial plan it is needed for that |
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| 10:35.38 | Zerr0 | is there any posibility to add video codec to asterisk? |
| 10:36.23 | ectospasm | Zerr0: yes |
| 10:36.40 | ectospasm | ...but I have no experience with it. |
| 10:37.34 | Zerr0 | :)) comprehensive answer :) |
| 10:37.53 | kaldemar | Zerr0: what are you really looking for? |
| 10:41.18 | ectospasm | Zerr0: I know that Asterisk can use various versions of the H.263 and H.264 codecs, but it does not appear that any video codecs ship with Asterisk |
| 10:41.24 | Zerr0 | trying to make a video call using websockets (sipML5 js client)... sending a call, callee rings, but caller gets strange SDP with "183 session progress" :) developers told that it's ok, and means that smth wrong with codecs |
| 10:42.28 | ectospasm | 183 Session Progress says absolutely nothing about codecs, unless the SDP lists something Asterisk can't handle. |
| 10:42.53 | Zerr0 | 183 sends asterisk... with sdp... just after 100 (trying) |
| 10:43.27 | Zerr0 | this sends client in INVITE: |
| 10:43.27 | Zerr0 | m=video 62394 RTP/SAVPF 100 101 102 |
| 10:43.27 | Zerr0 | c=IN IP4 188.162.52.250 |
| 10:43.27 | Zerr0 | a=rtcp:62394 IN IP4 188.162.52.250 |
| 10:43.28 | Zerr0 | a=candidate:6840418 1 udp 2130714367 192.168.0.23 37241 typ host generation 0 |
| 10:43.28 | Zerr0 | a=candidate:6840418 2 udp 2130714367 192.168.0.23 37241 typ host generation 0 |
| 10:43.29 | Zerr0 | a=candidate:4135066038 1 udp 1912610559 188.162.52.250 62394 typ srflx generation 0 |
| 10:43.29 | Zerr0 | a=candidate:4135066038 2 udp 1912610559 188.162.52.250 62394 typ srflx generation 0 |
| 10:43.30 | Zerr0 | a=candidate:1324063890 1 tcp 1694506751 192.168.0.23 36517 typ host generation 0 |
| 10:43.30 | Zerr0 | a=candidate:1324063890 2 tcp 1694506751 192.168.0.23 36517 typ host generation 0 |
| 10:43.31 | Zerr0 | a=ice-ufrag:aALkgJiH9bsrM7qc |
| 10:43.31 | Zerr0 | a=ice-pwd:5tnr338NqUHIpN8OqQfi2UZa |
| 10:43.32 | Zerr0 | a=sendrecv |
| 10:43.32 | Zerr0 | a=mid:video |
| 10:43.33 | Zerr0 | a=rtcp-mux |
| 10:43.33 | Zerr0 | a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7Izu2gN6K/fzM1JPnRwuRBWY+6uHTbuldNlWDtP5 |
| 10:44.40 | Zerr0 | a=rtpmap:100 VP8/90000 |
| 10:44.40 | Zerr0 | a=rtpmap:101 red/90000 |
| 10:44.40 | Zerr0 | a=rtpmap:102 ulpfec/90000 |
| 10:44.40 | Zerr0 | nothing similar to core show codecs |
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| 10:51.29 | ectospasm | woah, flood |
| 10:51.31 | ectospasm | not cool |
| 10:51.33 | ectospasm | don't do that |
| 10:51.36 | ectospasm | ~pb |
| 10:51.36 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 10:51.45 | ectospasm | Zerr0: ^^ |
| 10:52.53 | Zerr0 | sorry... pressed enter... |
| 10:56.14 | ectospasm | turn on SIP debug, and pastebin the results |
| 10:56.37 | ectospasm | "sip set debug on" |
| 10:57.38 | Zerr0 | :) thx. i know :) |
| 10:58.57 | Zerr0 | just a minute |
| 11:07.13 | Zerr0 | http://pastebin.com/6D6TRudg |
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| 11:23.57 | g0uZ | what are the option to secure phone configuration provisionning ? |
| 11:36.32 | indika1978 | My dial plan the value for NoOp(${ACTION}) is not appearing in CLI ... http://pastebin.com/ziVeRs9s pls help me |
| 11:38.26 | ectospasm | Zerr0: do your SIP peers support the siren7 codec for video? |
| 11:38.39 | ectospasm | Zerr0: because that's the only video codec that you're negotiating. |
| 11:38.56 | Zerr0 | is siren7 a video codec? |
| 11:39.10 | Zerr0 | sip show codecs tells that it's audio |
| 11:39.43 | ectospasm | according to that output, your SIP client (peer) is presenting siren7 as the only video codec it supports. |
| 11:39.49 | Zerr0 | seems like websocket peer knows only vp8 |
| 11:41.28 | kaldemar | indika1978: if you look at the M() option description closely, you see that the macro is executed on the *called* channel. your NoOp is executed on the calling channel. |
| 11:43.00 | indika1978 | yes thats true. Thanks for helping me in this. I need to pass variables between them. how can i do that ? |
| 11:43.07 | kaldemar | indika1978: also, you might want to consider using labels in priorities instead of numbered ones. exten => s,n(answer),Set(ACTION=...) |
| 11:44.21 | indika1978 | I didnt get that actually. Pls explain me |
| 11:44.30 | kaldemar | indika1978: which part? |
| 11:46.08 | indika1978 | exten => s,n(answer),Set(ACTION=...) .. |
| 11:46.59 | indika1978 | that means in such instance i need to use priority as number not n ? |
| 11:47.01 | kaldemar | indika1978: http://pastebin.com/t3eA7NDu |
| 11:47.39 | kaldemar | it is easier to use n and labels instead of numbered priorities. |
| 11:47.46 | indika1978 | yes I am having that prob too. 1000 thanks for that |
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| 11:49.40 | indika1978 | kaldemar : how can i sortout the variable issue |
| 11:49.46 | indika1978 | in this dial plan |
| 11:49.59 | indika1978 | it wont take any variable |
| 11:51.58 | kaldemar | indika1978: try Set(MASTER_CHANNEL(ACTION)=..) instead of Set(ACTION=...) |
| 12:00.48 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:01.59 | indika1978 | kaldemar: yes it worked. Thanks. I tried lot to achieve this. Now I am having a issue which is ... CID announcement is not going to internal extension, it goes to the outside caller .. that is the issue ? Pls help |
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| 12:03.41 | indika1978 | http://pastebin.com/agswNs7g |
| 12:06.21 | kaldemar | indika1978: give ${CALLERID(num}) as an argument (syntax: M(macro[^arg[^...]])) for the macro and put the accouncement into the macro. you'll have the caller id in ${ARG1}. |
| 12:07.12 | indika1978 | thanks |
| 12:09.24 | indika1978 | kaldemar: that means you are asking me to write a another macro and pass callerid as to it |
| 12:09.26 | indika1978 | ? |
| 12:13.46 | [TK]D-Fender | No |
| 12:13.59 | [TK]D-Fender | indika1978, pass it to the macro you ARE calling |
| 12:15.01 | MrPockets | Hello sports fans. |
| 12:16.14 | indika1978 | I feel that , it is not possible. because we call it and finished working with that. If possible pls let me know the way of doing that |
| 12:16.54 | kaldemar | indika1978: what? you have a macro, just go edit it. |
| 12:18.05 | indika1978 | Frankly I am saying I really do not understand what you have said? Ok if I edit it means again i need to call it. isnt it? |
| 12:18.31 | kaldemar | edit means modifying your macro. |
| 12:18.34 | indika1978 | so the place would be end of macro |
| 12:18.38 | indika1978 | yes |
| 12:18.40 | indika1978 | i got it |
| 12:19.00 | Zerr0 | looks like there is no simple way to add video codec to asterisk |
| 12:19.02 | kaldemar | the place depends on what you want. i'd put the announcement first, before the choice. |
| 12:19.36 | indika1978 | can you pls help me in this to do ... at lest pls let me know the line that need to put to it. |
| 12:20.11 | kaldemar | M(testscreen) -> M(testscreen^${CALLERID(num)}) |
| 12:20.21 | [TK]D-Fender | inInt your DIAL you call a MACRO. PASS IT IN THE MACRO YOU CALL |
| 12:20.46 | kaldemar | that will give you ${CALLERID(num)} as ${ARG1} in the macro. |
| 12:21.47 | indika1978 | ok .. once the options are made then it comes out of macro and goto conetext test1 .. so how can i go back again to macro? |
| 12:22.27 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 12:23.06 | indika1978 | after it comes to test1 ...it continuous to cid, voice mail, ... to end terminate ...priorites |
| 12:23.33 | indika1978 | how can i take it back to macro and do this? |
| 12:23.51 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 12:24.24 | *** part/#asterisk apten (~apten@carbon.gonicus.de) |
| 12:25.49 | kaldemar | indika1978: who needs to hear the number? |
| 12:26.06 | [TK]D-Fender | kaldemar, the callee |
| 12:26.19 | [TK]D-Fender | kaldemar, We've each told him the same thing repeatedly and it's not sinking in |
| 12:26.24 | indika1978 | number should be hear to extension that registerd with this sip server |
| 12:26.43 | kaldemar | [TK]D-Fender: just making sure. having it in [test1] in the first place does not make sense to me. |
| 12:26.49 | kaldemar | [TK]D-Fender: oh, ok. |
| 12:27.13 | carrar | Should be using [test2] !! |
| 12:27.26 | carrar | <- waking up |
| 12:27.56 | kaldemar | indika1978: if the macro is executed on the called channel, you need to put the announcement in the macro, right? |
| 12:27.57 | indika1978 | I am sorry to bother you all [TK]D-Fender , kaldemar ,, but I am not getting this |
| 12:28.06 | [TK]D-Fender | Actually, nothing to pass. he wants the CALLEE. he has that already in the channel name in that macro and in the vars he's looking at AFTER the dial(too late there) |
| 12:28.06 | indika1978 | yes |
| 12:28.33 | indika1978 | thats true |
| 12:29.03 | indika1978 | is there any function that can swap the detail to callee |
| 12:29.55 | [TK]D-Fender | indika1978, Your description is still unintelligible circular junk |
| 22:20.53 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 22:20.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.7.0 (2012/07/30), 1.8.15.0 (2012/07/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 22:23.42 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 22:23.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.7.0 (2012/07/30), 1.8.15.0 (2012/07/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 22:30.29 | pzn | for debugging hardware purposes, is there any way to get the "continuous registered time" of a sip client? "sip show peers" shows who is connected, but not for how long... is this possible? |
| 22:31.16 | *** join/#asterisk Bullmoose (~Bullmoose@63-155-51-139.bois.qwest.net) |
| 22:36.16 | pzn | dont need answer for my previous question. /var/log/asterisk/messages have messages when a peer becomes reachable/unreacheable. I can use them to get the information I need |
| 23:03.30 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
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