| 00:02.40 | *** join/#asterisk gboudreau (~gboudreau@mail2.danslereseau.com) |
| 00:03.55 | gboudreau | hi guys. looks like voice mails are limited to 60 seconds on my asterisk. voicemail.conf says: maxsecs=300. After leaving a 60 seconds message, the caller hears the busy signal, and my logs say: [Oct 20 19:57:09] VERBOSE[22022][C-000000a4] app.c: -- User hung up |
| 00:05.38 | gboudreau | maybe that would be caused by some other timeout config..? |
| 00:12.46 | [TK]D-Fender | Go place a call yourself and watch it at CLI |
| 00:15.39 | gboudreau | I'm pretty sure my logs is as verbose as the CLI, but will try |
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| 00:18.48 | gboudreau | yeah, same |
| 00:18.49 | gboudreau | http://pastie.org/pastes/5091144/text?key=repxczvuoighkpvuiinfyq |
| 00:19.09 | gboudreau | maybe that could be a setting on my voip provider..? |
| 00:20.17 | gboudreau | hum... but longer calls occured with that provider, and it can't discrimate a voicemail recording from a real conversation... so it must be *... |
| 00:23.48 | gboudreau | is there a CLI command I could use to validate the value of maxsecs used? |
| 00:26.45 | gboudreau | oh. maybe it is my voip provider... https://issues.asterisk.org/jira/browse/ASTERISK-11082 |
| 00:28.33 | gboudreau | trying "transmit_silence = yes" |
| 00:34.41 | [TK]D-Fender | gboudreau: You are NOT looking at the whole call. |
| 00:35.05 | [TK]D-Fender | gboudreau: Stop assuming VOICEMAIL is the problem and look at the actual full call |
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| 00:40.51 | gboudreau | that was it. "transmit_silence = yes" fixed it. |
| 00:41.05 | gboudreau | later. |
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| 02:36.22 | volga629 | Hello Everyone, I am looking information about IAX2 and video configuration can't make video working properly |
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| 02:38.31 | stope | today I had a 1-way audio problem, first time... and error was: Reason: SIP;cause=200;text="Call completed elsewhere" |
| 02:38.58 | stope | I've read up on it but is it something obvious? System has been running fine for nearly a year |
| 02:54.01 | stope | bah, nm, found it. :\ |
| 02:54.09 | stope | quit |
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| 05:09.24 | jsjc | I am about to upgarde a 1.6 to a 1.8 |
| 05:09.36 | jsjc | Is there anything I should be aware or change in the dialplans? |
| 05:09.59 | jsjc | any manual 1.6 vs 1.8 to understand differences on config/dialplans? |
| 05:14.08 | jsjc | because trying to do an outgoing call trouhg dahdi and tells me there "no invalid handler" |
| 05:15.13 | jsjc | and theere is a correcto one,Dial(DAHDI/7/067${EXTEN:1},0,TK) isnt it?? |
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| 06:11.12 | kaldemar | jsjc: an "invalid handler" is an extension, "exten => i,1,...", and handles invalid dialed numbers. you dialed a number that does not have a matching extension. |
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| 12:57.05 | jmls1 | mmm. is there an advanced setting somewhere on the D40 web admin to let me get at stuff like NAT etc ? |
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| 15:10.41 | imox | kennt jemand ein Programm womit ich über voip vom PC faxen kann? egal jetzt für welches system? |
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| 18:41.20 | drzed | hi there! |
| 18:41.56 | drzed | i've got this very simple dialplan: http://nopaste.info/c3bda5b9c0.html, but asterisk alwasy tells me |
| 18:42.24 | drzed | "Call from '15' to extension '88' rejected because extension not found." |
| 18:42.31 | drzed | what could be the reason for that? |
| 18:42.50 | WIMPy | Wrong context. |
| 18:43.00 | WIMPy | And the message should continue saying that. |
| 18:43.13 | [TK]D-Fender | because it isn't looking in those contexts at all |
| 18:44.33 | drzed | hm, i dont get it: dialplan show 88@ does return: [...] Included context 'mailbox_own' created by 'pbx_config' [..] |
| 18:45.00 | WIMPy | And what context does your phone use? |
| 18:46.01 | drzed | the phone is in context "lokal" |
| 18:46.20 | [TK]D-Fender | DrAnd I don't see [lokal] ANYWHERE in your dialplan |
| 18:49.52 | drzed | [TK]D-Fender: ok included lokal now: http://nopaste.info/4146014e2a.html |
| 18:49.59 | drzed | but still the same problem |
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| 18:50.41 | leifmadsen | lokal should contain the contexts you want available to the devices if they use lokal |
| 18:50.44 | leifmadsen | not the inverse of that |
| 18:51.07 | [TK]D-Fender | drzed: [lokal] <- this is where you pointed it to. |
| 18:51.21 | [TK]D-Fender | drzed: exten => _1X,n,Dial(SIP/${EXTEN},55,Tt) <-- this is the only thing you put in it |
| 18:51.22 | WIMPy | Jup. Still no 88 or other context containing 88 in lokal. |
| 18:51.33 | [TK]D-Fender | drzed: _1X will not match 88. Ever |
| 18:52.21 | [TK]D-Fender | drzed: Not to mention that you starte4d with priority "n" which is wrong. Each extension needs to start with "1" |
| 18:53.22 | WIMPy | And it might not be the best idea to call your accounts the same as your extensions. |
| 18:54.14 | wonderworld | hi guys |
| 18:56.28 | drzed | hm ok, i'll try to fix that |
| 18:56.33 | drzed | thx for the moment |
| 18:58.07 | wonderworld | i am looking for tools to stress-test asterisk. i stumbled upon SIPP, which seems to be good. are there others? |
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| 19:02.36 | leifmadsen | wonderworld: yep |
| 19:02.47 | leifmadsen | sipp is pretty much the standard for load testing |
| 19:11.19 | drzed | should for example 'dial 15@lokal' do the same thing as dialing 15 on the phone in context lokal? |
| 19:12.05 | drzed | bc/ the former works and the latter fails with 'service unavailable' |
| 19:12.12 | WIMPy | What do you mean by "dial 15@lokal"? |
| 19:12.32 | drzed | console dial 15@lokal |
| 19:12.55 | WIMPy | Ok, yes. That should do the same. |
| 19:13.24 | WIMPy | Maybe your phone doesn't do call waiting? |
| 19:13.38 | WIMPy | Or do you have a 2nd phone? |
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| 19:15.11 | drzed | exten => _1X,1,Dial(Zap/g1/${EXTEN}) |
| 19:15.27 | drzed | 15 is a sip (snom) phone |
| 19:15.57 | WIMPy | zap is a for hardwar interfaces and has been replaced by dahdi several years ago. |
| 19:16.29 | drzed | unfortunately i cannot update the server at the moment |
| 19:16.52 | drzed | however using 'console dial 15@lokal' works perfectly fine |
| 19:17.03 | drzed | so i assume there is no problem with zap |
| 19:17.37 | drzed | how can i debug this? |
| 19:17.45 | drzed | *further |
| 19:17.56 | [TK]D-Fender | drzed: PASTEBIN. Show us the actual full attempts |
| 19:17.58 | WIMPy | 'core set verbose 3' and then pastebin the output of the complete call. |
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| 19:19.45 | wonderworld | leifmadsen: thanks |
| 19:21.32 | drzed | [TK]D-Fender: http://nopaste.info/af053e65be.html <-- tried 'dial 12@lokal, then with the phone' |
| 19:23.17 | WIMPy | Looks like a zap configuration issue. |
| 19:23.32 | WIMPy | Probably trying to transmit a name or something. |
| 19:26.14 | [TK]D-Fender | [Oct 21 21:20:34] WARNING[12496]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 19:26.17 | [TK]D-Fender | ^^^^^^^^ |
| 19:26.36 | [TK]D-Fender | Screwed up console config |
| 19:26.43 | [TK]D-Fender | And stop using CLI to place calls |
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| 21:24.10 | *** join/#asterisk Mango45 (~Mango45@d209-89-214-26.abhsia.telus.net) |
| 21:25.12 | Mango45 | Hi. I'm trying to set up Asterisk to receive anonymous calls. Both Asterisk and the machine that is sending the call have public IP addresses; i.e. no NAT. |
| 21:25.26 | Mango45 | I also have nat=no in sip.conf. |
| 21:25.49 | Mango45 | However, when I look at SIP debug, I see <--- Reliably Transmitting (NAT) to [ip]:5060 ---> |
| 21:26.11 | Mango45 | I also have a problem where the call drops after exactly 30 minutes. Could the two be related? |
| 21:26.12 | [TK]D-Fender | Mango45I also have nat=no in sip.conf. <- WHERE in there matters. |
| 21:26.35 | [TK]D-Fender | Mango45: So I'd serioulsy look at the call and your configs again |
| 21:29.50 | Mango45 | [TK]D-Fender: It would be in the [general] context. |
| 21:30.03 | [TK]D-Fender | AND the call.... |
| 21:30.25 | Mango45 | I do not understand this question. :) |
| 21:30.59 | [TK]D-Fender | [17:26][TK]D-FenderMango45: So I'd serioulsy look at the call and your configs again |
| 21:31.30 | Mango45 | Yes I read it, but I don't understand what you want me to do. |
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| 21:32.46 | [TK]D-Fender | Look. At. The. Call. |
| 21:33.14 | Mango45 | looks |
| 21:33.18 | Mango45 | ...and? |
| 21:35.49 | Mango45 | I see something that may be useful. The first time says: |
| 21:35.55 | [TK]D-Fender | ~pb |
| 21:35.55 | 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. |
| 21:35.57 | [TK]D-Fender | ^ |
| 21:36.16 | Mango45 | Sending to [ip] : 5060 (no NAT) |
| 21:36.31 | Mango45 | Will do. |
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| 21:42.47 | Mango45 | http://pastebin.com/WrHinhdF |
| 21:42.52 | Mango45 | Thank you [TK]D-Fender. I appreciate the help. |
| 21:43.30 | [TK]D-Fender | And your sip.conf ? |
| 21:46.12 | Mango45 | http://pastebin.com/WifwXvtH |
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| 21:51.25 | Mango45 | Anything there jump out? |
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| 21:56.17 | [TK]D-Fender | Undo your use of templates |
| 21:57.56 | Mango45 | you mean [devices](!) |
| 21:58.19 | [TK]D-Fender | yes |
| 22:00.41 | Mango45 | No change unfortunately. |
| 22:02.59 | [TK]D-Fender | We should be seeing updated call attempts & configs rather than "no change" .... |
| 22:04.12 | Mango45 | Ok, I have to step away for a while but I will pastebin the updates when I get back if you're still here. Thanks again. |
| 22:07.08 | *** join/#asterisk vlad_starkov (~vlad_star@74.175.27.250) |
| 22:13.07 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-rotkcmitljghpcuz) |
| 22:13.07 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 22:35.33 | *** join/#asterisk Tim_Toady (~fuzzy@62.1.174.103.dsl.dyn.forthnet.gr) |
| 22:39.51 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-rotkcmitljghpcuz) |
| 23:03.36 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
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| 23:21.34 | *** join/#asterisk whtsup (~whtsup@WimaxUser38143-234.wateen.net) |
| 23:21.43 | whtsup | hello |
| 23:21.47 | whtsup | i m using asterisk 10 |
| 23:22.30 | ChannelZ | Hurray! |
| 23:24.28 | whtsup | chan_sip.c: No compatible codecs, not accepting this offer! |
| 23:24.37 | whtsup | getting this error from one of my peers |
| 23:25.07 | whtsup | other peers are working fine wid same config |
| 23:26.19 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 23:26.41 | whtsup | any one guide me plz |
| 23:26.52 | jpsharp | Your peer is asking for codecs that asterisk doesn't like. |
| 23:27.00 | jpsharp | Most likely G729. |
| 23:27.41 | whtsup | but i ve it |
| 23:27.49 | whtsup | its messing wid all codec |
| 23:28.02 | whtsup | all of my other clients are workign |
| 23:28.09 | whtsup | expect only one |
| 23:30.29 | ChannelZ | Perhaps your one client only has one codec listed or something. |
| 23:31.02 | ChannelZ | You must allow the codecs you want in sip.conf for the peer, and one of those codecs has to be one the client has and is set to use in order for them to negotiate successfully. |
| 23:33.23 | whtsup | i set it allow=all |
| 23:33.27 | whtsup | still not working |
| 23:36.10 | ChannelZ | So then you have to look at the device |
| 23:36.35 | ChannelZ | As jpsharp said it might be calling for g729 ONLY or something |
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| 23:47.48 | [TK]D-Fender | whtsup: Well we can't see what they are offering, or what you areoffering. |
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