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| 02:42.48 | luvmyh0nda | Hi! I am working on a dialplan, and it keeps timing out, although I have extension t, with a goto, any advice? |
| 02:43.33 | WIMPy | Where does it time out. Show us a call. |
| 02:43.41 | luvmyh0nda | pastebin? |
| 02:45.19 | WIMPy | yes |
| 02:46.10 | luvmyh0nda | http://pastebin.com/YXtc2LbT |
| 02:48.24 | WIMPy | I don't see a timeout. I see the end of your dialplan. |
| 02:48.57 | WIMPy | And the interesting part for you: I don't see any place that waits for user input. |
| 02:49.06 | luvmyh0nda | googletts waits for input |
| 02:50.18 | luvmyh0nda | do I need to define TIMEOUT ? |
| 02:50.37 | WIMPy | There is no timeout happening. |
| 02:51.14 | luvmyh0nda | sorry, i'm relatively new to asterisk, but have been programming a while in perl, php & such, so i'm a bit dense when it comes to asterisk |
| 02:51.25 | luvmyh0nda | essentially, i don't understand what auto fallthrough means, or how to stop it |
| 02:51.36 | WIMPy | I don't know what that agi does other than playing sounds, but after it finishes, the last time, your dilplan just end, thus ending the call. |
| 02:52.16 | WIMPy | That means Asterisk ran out of instuctions (dialplan). |
| 02:54.07 | luvmyh0nda | ok, i understand that part. how can i tell it to loop back to "to return to the main menu" - i have added the (instructions) to the "To return to the main menu" line |
| 02:54.09 | mjordan | I'd recommend making an AA/IVR without an AGI first, particularly if it seems to be doing things (or not doing things) that you expect |
| 02:55.51 | WIMPy | Yes, that's probably a good idea. |
| 02:56.24 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 02:56.37 | mjordan | (very) basic tutorial for building a simple IVR menu ^^^ |
| 02:56.42 | mjordan | including handling the special extensions |
| 02:57.40 | mjordan | in particular, this page will also help - https://wiki.asterisk.org/wiki/display/AST/Background+and+WaitExten+Applications |
| 02:58.06 | mjordan | That will explain a bit how Asterisk is waiting for digits, and what happens when it doesn't get enough (and you can make it wait longer, for example) |
| 03:00.49 | luvmyh0nda | Oh, so because I didn't have waitexten it had no futher commands |
| 03:04.29 | luvmyh0nda | and then i use TIMEOUT to set the maximum time, so that it doesn't overrun too many timeouts |
| 03:04.29 | luvmyh0nda | got it |
| 03:06.10 | WIMPy | Your t extension could need some cleaning. |
| 03:06.35 | WIMPy | The answer at that point is obviousely senseless. And you should set a lable for that Goto. |
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| 03:08.17 | luvmyh0nda | the answer at timeout? |
| 03:08.33 | WIMPy | yes |
| 03:08.37 | luvmyh0nda | oh, i took it out |
| 03:08.42 | luvmyh0nda | once i read the page mjordan posted |
| 03:09.01 | luvmyh0nda | and for me, nothing is obviously anything at this time lol |
| 03:09.47 | WIMPy | Well, you can answer a call only once. And it's the first thing you do, so that's it. |
| 03:10.03 | luvmyh0nda | except its obvious that i'm not an expert at asterisk |
| 03:10.42 | WIMPy | Well, that should be obvious to anyone. |
| 03:11.05 | luvmyh0nda | so what you think that because you know asterisk you're the sh(asterisk)t or something? |
| 03:11.23 | luvmyh0nda | you, too, my friend, were a novice at some point in time |
| 03:11.40 | WIMPy | Have you ever tried ansewering a call you're already on? |
| 03:11.54 | WIMPy | That has nothing to do with Asterisk. |
| 03:12.04 | WIMPy | That's simple logic. |
| 03:12.25 | luvmyh0nda | maybe for someone with your level of experience |
| 03:12.50 | luvmyh0nda | as it happened, i googled my issue extensively prior to joining, and got some bad information |
| 03:13.18 | WIMPy | That's like trying to start your engine while already driving. Is that clearer? |
| 03:13.31 | luvmyh0nda | kinda like mis-spelling label? |
| 03:14.19 | WIMPy | I don;t think it helps if I say yes. |
| 03:16.26 | luvmyh0nda | being a jerk to beginners doesn't help either |
| 03:17.16 | WIMPy | Just because you're new to Asterisk doesn't mean, you can't wipe your ass yourself. And if it does, I'm out. |
| 03:21.16 | Penguin | Is the problem solved? |
| 03:22.09 | luvmyh0nda | absolutely |
| 03:22.20 | Penguin | Great! |
| 03:22.59 | luvmyh0nda | i don't mean to be an ass about it, but it's really discouraging to treat new people like that without knowing their experience with programming & such |
| 03:23.59 | Penguin | I don't see why it was dragged out so long. I understand that you didn't know, even if it should have been obvious. You said you didn't know, but now you know... so we move on. |
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| 05:36.14 | ChannelZ | hiccups |
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| 10:03.50 | kesselklopfer79 | hello how can i inherit a channel language |
| 10:05.10 | kesselklopfer79 | so i can do CHANNEL(language)=de and on the dialed channel the language is also set |
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| 12:04.36 | iulhk | Hello anybody there |
| 12:04.39 | iulhk | i am using asterisk-11 , focus just sip to sip call, audio/video, in allowed codecs 'gsm;h264' , Issue peer 1000 calling to 2000 audio call, after ack, at receiver end video screen automatically opened, although caller started audio call, if i just enabled gsm, codec then both-end its audio screen, but in that case i can't start video as video codec not enabled. i want when user starting |
| 12:04.40 | iulhk | video call, it should start video at both-end when user starting audio call then at both end it should be audio call, is it possible ? |
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| 12:36.16 | cunningpike | iulhk: Is the priority of the codecs like you stated? gsm first? |
| 12:37.14 | cunningpike | iulhk: If so, it sounds like the endpoints are using h264 regardless of whether the call has video or not. Maybe fiddle with their settings a bit. |
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| 12:53.58 | iulhk | <cunningpike>: allow=gsm;h264 , i want if caller pressing audio call then at callee end video screen should not be enabled, if caller pressing video button then at the callee's end video screen should open, can i control this in sip.conf ? |
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| 12:54.51 | cunningpike | iulhk: What codec is being used for the audio call? Check your console. |
| 12:55.49 | iulhk | let me check |
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| 13:23.05 | Penguin | allow=gsm;h264 <--- I think this means only gsm, since the ; begins the comment. |
| 13:24.06 | cunningpike | Penguin: I think he's been disabling h264 as part of his troubleshooting... |
| 13:24.31 | cunningpike | Penguin: But maybe not....... |
| 13:31.24 | luvmyh0nda-[idle | is now away: [reason: Idle after 10h]±[pager: yes]±[ç§] |
| 13:31.40 | [TK]D-Fender | "the callee's end video screen should open" <_ more like it shouldn't .. because the codec is not allowed. |
| 13:32.16 | Penguin | That's what I'm saying. |
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| 14:18.24 | jeev | horray, pacwest has been down all night.. |
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| 15:16.48 | stasiu7 | Hi, is there a way to limit ring time for outbound calls if not answered? I'm running Asterisk 1.8.22 with FreePBX 2.11 |
| 15:17.01 | Penguin | #freepbx |
| 15:22.48 | carrar | stasiu7: just change the dial timeout |
| 15:22.55 | carrar | in the dial command |
| 15:23.02 | carrar | DONE and DONE |
| 15:23.17 | carrar | show application dial |
| 15:23.21 | Penguin | Done until the next time you apply settings by the GUI. |
| 15:23.25 | Penguin | Then undone. |
| 15:23.32 | carrar | cron tab it |
| 15:23.52 | Penguin | That's stupid. Just change it with freepbx like it's supposed to be done. |
| 15:24.01 | carrar | every min over write the crappy GUI settings |
| 15:24.08 | carrar | heh |
| 15:24.22 | Penguin | Every minute, remind yourself to stop using a GUI if you want to edit your own files! |
| 15:24.41 | carrar | that too |
| 15:25.14 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 15:25.23 | Penguin | * * * * * aplay /home/carrar/remind-stop-using-gui.wav |
| 15:25.37 | Nugget | haha |
| 15:25.38 | Kobaz | sooooo |
| 15:25.42 | Kobaz | what's a quick way to get the rtp address and ports out of a sip channel |
| 15:25.50 | carrar | the other alternative answer was to tell him to seek out the correct IRC channe; :) |
| 15:26.05 | carrar | I got tails on that coin flip |
| 15:26.07 | Penguin | I honestly beat you to that one. |
| 15:26.10 | carrar | crontab it was |
| 15:26.16 | Kobaz | i never actually looked into that before.. i usually do an rtp debug but that's not going to work on a busy system |
| 15:26.18 | carrar | heh |
| 15:26.51 | Penguin | sip set debug peer <yourpeer> |
| 15:26.56 | Penguin | Look at SDP. |
| 15:27.04 | Kobaz | yeah |
| 15:27.10 | Kobaz | that's gonna be a LOT of traffic |
| 15:27.13 | Penguin | Yep. |
| 15:27.16 | Kobaz | i want an individual channel; |
| 15:27.40 | carrar | Like you want a channel variable in the dialplan or just from the console? |
| 15:27.47 | Kobaz | either oe |
| 15:28.31 | carrar | SIPCHANINFO(peerip) |
| 15:28.37 | Kobaz | niiice |
| 15:28.46 | carrar | err thats not RTP though really |
| 15:29.20 | Kobaz | just started having one way audio problems left and right |
| 15:29.27 | Kobaz | and sometimes no-way audio |
| 15:30.29 | Kobaz | yeah hmm, that's just address |
| 15:30.31 | Kobaz | no ports |
| 15:30.35 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:30.38 | Kobaz | well at least the ips look okay |
| 15:31.55 | carrar | CHANNEL(rtpdest,audio) |
| 15:33.11 | Kobaz | cool |
| 15:33.13 | Kobaz | that's one half |
| 15:33.21 | Kobaz | i don't see src address in there |
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| 15:38.01 | Katty | PEANUT BUTTER COOKIES |
| 15:38.58 | lvlinux | YUM! |
| 15:39.23 | Katty | drmessano: ping |
| 15:40.10 | Kobaz | allrightey |
| 15:40.21 | Katty | hi Kobaz |
| 15:40.21 | Kobaz | sooo i bit the bullet and turned on sip debug for like 2 seconds |
| 15:40.29 | Kobaz | i got the invite i need to debug |
| 15:40.35 | Kobaz | hey katty kay |
| 15:40.53 | Katty | how're you dear? besides up to you rnose in log entries |
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| 15:42.21 | Katty | hugs carrar |
| 15:42.29 | Katty | hugs Penguin, too |
| 15:42.38 | Kobaz | doind good |
| 15:42.45 | Kobaz | got one of my vendors to pay their oldest invoice finally |
| 15:42.56 | Kobaz | back from march |
| 15:43.07 | Katty | wooohooo! |
| 15:43.44 | Kobaz | http://pastebin.com/AChvD6N3 |
| 15:43.47 | Kobaz | soooo, who wants to help debugt |
| 15:44.01 | Kobaz | i have no audio outgoing |
| 15:47.21 | Kobaz | it all looks good |
| 15:47.26 | Kobaz | it has a proper looking destination rtp port |
| 15:47.31 | Kobaz | which is an itsp |
| 15:47.46 | Kobaz | and this isn't just one itsp, this is like all itsp's set up on the box |
| 15:47.59 | Kobaz | call in on a did and there's no outgoing audio |
| 15:48.25 | carrar | hugs Katty back! |
| 15:48.26 | carrar | w00t! |
| 15:49.01 | Katty | how're you dear? |
| 15:49.09 | Katty | hopefully not drowning in debug logs like Kobaz? |
| 15:49.10 | carrar | GREAT!! |
| 15:49.14 | Katty | horay!!!! |
| 15:49.24 | carrar | well a edited debug file so no point in looking |
| 15:50.01 | Katty | weekend plans? |
| 15:50.04 | carrar | YES |
| 15:50.08 | carrar | House shopping |
| 15:50.12 | Katty | oh my |
| 15:50.22 | Katty | well good luck! i hope you find the PERFECT house! |
| 15:50.29 | carrar | I'm in the perfect house |
| 15:50.45 | Katty | then why are you shopping for a house? |
| 15:50.46 | carrar | but the state is siding with the people who want to put a TRAIN through my house to eminent domain it |
| 15:50.51 | Katty | unless you're sitting at your mother's kitchen table currently |
| 15:50.57 | Katty | oh :/ |
| 15:50.58 | carrar | So I haev to move |
| 15:51.09 | carrar | No one really owns their land in the USA |
| 15:51.12 | Katty | well...that's just crappy. i hope they are giving you what your house is actually worth? |
| 15:51.25 | carrar | 1 billion dollars |
| 15:51.48 | Katty | ...are you ...serious? |
| 15:51.53 | carrar | no |
| 15:51.53 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 15:51.59 | Katty | goshdangitanyhow |
| 15:52.14 | Katty | i was about to get really excited for you |
| 15:52.18 | cunningpike | waves at Katty |
| 15:52.25 | Katty | hey cunningpike!!! |
| 15:52.29 | carrar | Houses are expensive here |
| 15:52.31 | cunningpike | It's been a *very* long time. How are you my dear? |
| 15:52.35 | carrar | I need like 2 million |
| 15:52.42 | Katty | cunningpike: oh just about as peachy as always! you? (= |
| 15:53.02 | Katty | carrar: hrmm that is pricey. hopefully they're giving you enough to have something similiar to your current house |
| 15:53.08 | cunningpike | Katty: Excellent, TYVM. Now living in sunny MI. |
| 15:53.20 | Katty | cunningpike: hehe michigan is not sunny |
| 15:53.24 | Katty | cunningpike: it's shaped like a mitten ;) |
| 15:53.37 | Katty | cunningpike: for REASONS |
| 15:54.09 | Katty | cunningpike: also, why on earth did you move to michigan? |
| 15:54.17 | cunningpike | Katty: I missed last winter, thank goodness. But it really is sunny right now. Having some beautiful fall weather. |
| 15:54.27 | cunningpike | Katty: I married a lovely lady here. |
| 15:54.41 | Katty | cunningpike: well that sounds like the best reasons to move ever. |
| 15:54.46 | Katty | cunningpike: congratulations!!!! |
| 15:55.10 | cunningpike | Katty: Thank you! |
| 15:55.17 | Katty | did you cry? |
| 15:55.25 | cunningpike | It was a journey, let me tell you. |
| 15:55.32 | cunningpike | Katty: Of course! |
| 15:55.38 | Katty | hehe, good ^_^ |
| 15:55.58 | Katty | what city did you go to afterwards? |
| 15:56.01 | Katty | let me guess! |
| 15:56.05 | Katty | burlington, vermont? |
| 15:56.07 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:56.11 | Katty | hi danny |
| 15:57.26 | cunningpike | Katty: My first married weekend was spent as a chaperone for our 4th grader's school trip to Mackinaw. We had a honeymoon in Petoskey, though, which was lovely. |
| 15:57.55 | Katty | googles petoskey |
| 15:58.15 | Katty | the downtown sure does look cutesy! |
| 15:58.37 | Katty | i'm glad you had a wonderful time |
| 15:58.42 | cunningpike | Thanks! |
| 15:59.21 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 16:01.21 | Katty | cunningpike: nothing much has changed here. same job, same house, same boyfriend. |
| 16:01.36 | Katty | cunningpike: my father had open heart surgery, but he's recovering beautifully. |
| 16:05.31 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 16:14.19 | *** join/#asterisk tbsky (~tbsky@dns.annsky.com) |
| 16:18.54 | tbsky | Hi! sorry I have a question. I know asterisk didn't support CNG and VAD. but if two sip legs connect via asterisk both support CND/VAD, will things work? in my test it seems packets with CND/VAD through asteirsk will cause broken sound.. |
| 16:19.26 | tbsky | both legs pass through asterisk without transcoding.. |
| 16:20.12 | [TK]D-Fender | It will not work with * in the way |
| 16:20.25 | WIMPy | As you already said: It will not work with Asterisk. |
| 16:20.39 | WIMPy | If you can set up directmedia, ok, but only then. |
| 16:21.27 | tbsky | thanks a lot. I assume if it passed * without trancoding, things will work. but it seems not the case. |
| 16:22.52 | tbsky | WIMPy: I miss something impoart, you said "directmedia". any keyword at * about that? |
| 16:23.21 | WIMPy | Exately that: "directmedia" |
| 16:23.44 | tbsky | ok. I will check that, thanks for the hint! |
| 16:23.56 | WIMPy | This is usally a bad idea, unless you're only on a LAN. |
| 16:26.06 | tbsky | WIMPy: unfortunately it is not LAN. my sip trunk provider provide G.711 with CNG/VAD. so in that case I think is is no hope to use asterisk with G.711 with the trunk provider? |
| 16:26.17 | Kobaz | Katty: carrier problem |
| 16:26.37 | Kobaz | Katty: and my testing was screwy... we're not actually having this issue with all itsp's... helps if you're testing correctly |
| 16:26.47 | WIMPy | is positive about that conclusion. |
| 16:27.10 | Kobaz | okay |
| 16:27.19 | Kobaz | now for cerials i need to eat some breakfast |
| 16:27.25 | Kobaz | srsly |
| 16:27.34 | tbsky | WIMPy: if I use G.729 things is fine, I think that's because with G.729 the protocol can told the provider to stop CNG/VAD..but with G.711 there seems no way to stop the provider.. |
| 16:28.39 | WIMPy | That must be their configuration. I don't see how CODEC matters. |
| 16:30.11 | tbsky | WIMPy: really? I remember in SIP debug message I saw something like disable VAD/CNG, maybe I was wrong about it.. |
| 16:31.01 | WIMPy | It tell YOU to disable it on the peer. |
| 16:32.29 | tbsky | WIMPy: I need to re-check it again. maybe you are right. in my case G729 is ok but G711 is broken. I thought asterisk can told provider disable VAD/CNG with g729 negotiation. |
| 16:33.34 | WIMPy | I'm not aware of such a possibility. |
| 16:35.29 | tbsky | WIMPy: I see. so I am lucky at least G729 can work.. |
| 16:36.10 | tbsky | WIMPy: will pjsip with * 12 or 13 support VAD/CNG? |
| 16:40.13 | tbsky | WIMPy: ok. I see directmedia is to redirect the media path from the caller to the callee. so as you said, it is not suitable in my case.. |
| 16:40.31 | tbsky | WIMPy |
| 16:40.44 | tbsky | thanks for your hint |
| 16:41.10 | ThoMe | hiho |
| 16:41.17 | ThoMe | i would like recieve a fax |
| 16:41.21 | ThoMe | and asterisk said |
| 16:41.22 | ThoMe | ReceiveFAX |
| 16:41.26 | ThoMe | [Sep 26 18:39:01] WARNING[9234][C-00000001]: app_fax.c:442 transmit_audio: channel 'SIP/placetel-00000001' refused to negotiate T.38 |
| 16:41.30 | ThoMe | or [Sep 26 18:38:22] WARNING[9216][C-00000000]: app_fax.c:174 span_message: WARNING T.30 ECM carrier not found |
| 16:41.32 | WIMPy | I don't know if pjsip can do it. |
| 16:41.34 | ThoMe | and [Sep 26 18:38:59] WARNING[9216][C-00000000]: app_fax.c:830 transmit: Transmission error |
| 16:41.37 | ThoMe | :-( |
| 16:41.39 | ThoMe | can anyone help me please? |
| 16:42.05 | ThoMe | have t38pt_udptl=yes,redundancy,maxdatagram=400 t38pt_rtp=no t38pt_tcp=no |
| 16:43.41 | ThoMe | can anyone help me please? |
| 16:43.42 | WIMPy | Does whatever you're talking to support T.38? |
| 16:44.31 | ThoMe | i mean why said asterisk transmission error? |
| 16:45.18 | WIMPy | Did you read the other lines you posted? |
| 16:45.39 | ThoMe | WIMPy: yes. but i dont knwo what asterisk meaning :-) |
| 16:45.55 | ThoMe | WIMPy:i think my provider supported t38 |
| 16:46.39 | WIMPy | "refused to negotiate T.38" |
| 16:46.45 | ThoMe | WIMPy: please, explain what i must do :-) |
| 16:46.56 | WIMPy | Did they tell you the do T.38? |
| 16:47.01 | WIMPy | Hope doesn't help. |
| 16:47.28 | ThoMe | "refused to negotiate T.38" << this say the provider or my asterisk? |
| 16:47.32 | ThoMe | why refused? |
| 16:47.56 | WIMPy | Look at the sip debug. |
| 16:49.50 | *** join/#asterisk danjenkins (~dan@nat/digium/x-fhzusrsemyzdacqs) |
| 16:52.52 | ThoMe | WIMPy: have here my debug http://paste.keks.be/512/plain can you help me.. what should I look for? |
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| 16:57.44 | WIMPy | You sould look for someone who can read it. |
| 17:05.28 | MasterChen | good morning everyone |
| 17:07.18 | MasterChen | What could be the cause of static and cracking on the line of various peers, and simultaneous dropped calls in an office environment? |
| 17:08.25 | MasterChen | I am thinking it could be a network issue, but is there any way to analyze it from the asterisk CLI? |
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| 17:34.19 | babak | Hi , has some one used Dial() with A(x) flag for playing announcement ? |
| 17:35.49 | WIMPy | Sure |
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| 17:37.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 17:37.45 | babak | I use same x file name before Dial in Playback it works but same x in Dial nothing happened |
| 17:38.34 | babak | x should be without extentoin ? inside " " ? |
| 17:39.33 | WIMPy | No quotes. |
| 17:39.49 | [TK]D-Fender | And never with an extension |
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| 17:47.47 | MasterChen | how can I monitor whether a callis being dropped or naturally hungup? |
| 17:48.18 | WIMPy | Watch the *CLI |
| 17:48.33 | WIMPy | But it massively depends on where it's dropped. |
| 17:49.13 | MasterChen | what should i grep for so I can clear out unnecessary messages? |
| 17:49.29 | MasterChen | client is saying the dropped calls are simultaneous |
| 17:49.34 | [TK]D-Fender | We don't know what the probelm is so there is nothing to grep for |
| 17:49.35 | [TK]D-Fender | Just go look |
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| 20:42.34 | ZenMaster | Hi. |
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| 21:01.46 | newtonr | ZenMaster, Does a dog have buddha-nature? |
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| 21:28.47 | smkelly | file: HI!!!!!!!!!!!!!!!!!!!!!!!! |
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| 23:32.07 | rexwin_ | hi all. I am trying to get a simple asterisk in my lan. when i login from my sip phone the registration fails for demo-alice account |
| 23:32.25 | rexwin_ | when i log verbose i get chan_sip.c:25820 handle_request_register: Registration from '"demo-alice"<sip:6001@192.168.19.128>' failed for '192.168.19.1:12702' - Wrong password |
| 23:33.07 | rexwin_ | but i use the same pw for account in /etc/asterisk/sip.conf |
| 23:37.20 | *** join/#asterisk aross42 (~aross@192-0-133-151.cpe.teksavvy.com) |
| 23:43.57 | [TK]D-Fender | Show us the configs and the full attempt with SIP DEBUG enabled |
| 23:44.02 | [TK]D-Fender | "sip set debug on" |
| 23:44.27 | [TK]D-Fender | ~pb |
| 23:44.28 | 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. |
| 23:44.29 | [TK]D-Fender | ^^^ |
| 23:59.39 | [TK]D-Fender | And.... times up on my side. Someone else should be able to show you why once you get around to providing the debug |
| 23:59.45 | [TK]D-Fender | heads out... it's SHOW TIME |