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| 01:41.10 | afink | hey guys I am having trouble with incoming calls from vitelity. This is my SIP debug from the incoming failed call. http://pastebin.com/Dch6bVEZ. |
| 01:41.53 | afink | ahh I just saw the unauthorized line. |
| 01:48.39 | [TK]D-Fender | you have no inbound call there. |
| 01:48.43 | [TK]D-Fender | And you are registering fine. |
| 01:48.58 | [TK]D-Fender | You try, they 401 challenge you, you return with auth and they 200 OK you |
| 01:48.59 | [TK]D-Fender | It's fine |
| 01:49.04 | [TK]D-Fender | There is no CALL ATTEMPT in there |
| 01:50.14 | [TK]D-Fender | Expires: 60 <- this is also a shitty registration interval they are offering. I'd recomment setting your own rules for a more functional range so you don't waste traffic so often for nothing |
| 02:02.14 | afink | The itsp said that my PBX was rejecting the call. Any idea what would cause this? http://pastebin.com/P77NxqcZ |
| 02:15.25 | [TK]D-Fender | your debug doesn't show a call |
| 02:15.43 | [TK]D-Fender | Get an actual call arriving at asterk |
| 02:15.50 | [TK]D-Fender | All we saw was registrations |
| 02:15.53 | [TK]D-Fender | and those were fine |
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| 05:48.21 | lkthomas | hey guys, we are using multiple Ploycom SIP phone |
| 05:48.41 | lkthomas | everytime if we want to call anywhere, we have to punch in all numbers before hit dial |
| 05:48.53 | lkthomas | any way to make it pick up phone and dial instead ? |
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| 08:45.08 | doozer_ | hi all, need help fixing some extenstions that give the following error when trying to dial: |
| 08:45.10 | doozer_ | [2014-04-28 18:44:32] WARNING[1615][C-00000015]: chan_sip.c:24960 handle_request_invite: Received SIP INVITE with unsupported required extension: required:sdp-anat unsupported:sdp-anat |
| 08:48.30 | gain | hi all, I have logs full of "Sending fake auth rejection"... and in asterisk cli is marked as notice... is that ok? |
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| 08:52.13 | r00f | gain: someone is trying to bruteforce passwords to your extensions. it is not rare nowadays |
| 08:52.35 | r00f | or, you have some device misconfigured with wrong credentials |
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| 08:53.27 | gain | r00f: casual users and all from external ip... I think its some brute as you suggested |
| 08:54.57 | gain | r00f: any particular recommendation for these cases? |
| 08:56.06 | gain | r00f: I really need to forward 5060 port externally? All the phones are in my lan and the trunk outside... |
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| 09:20.08 | r00f | gain: allow outside access to 5060 port only for trunk ip. or install fail2ban |
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| 09:26.21 | gain | r00f: thanks |
| 09:28.40 | RadJackson | Hello, i've developed an audiotel application which is a quizz , many users are satisfied with its result , but some of'em are complaining that sometimes when i ask them to type 1/2 or 3 , they type and it just doesnt work , ive been testing the app i never had the problem. it may be some dtmf problem, any idea? |
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| 09:41.54 | neeby_goosey | Hello. Could anybody explain me how to define what extensions a queue member can transfer calls to? |
| 09:48.27 | neeby_goosey | My guess is I should list those in the same context my Queue(some-queue,t); command is. |
| 09:50.03 | neeby_goosey | But then the caller could also access the extensions, so I am confused now. |
| 09:51.35 | r00f | i guess, queue member cannot go outside his context. just divide them into different contexts and include them into some parent contexts to allow connection between them |
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| 10:09.42 | dan_j | Hi, is it possible to only reload a specific context? |
| 10:09.53 | dan_j | In order to reduce the verbose output. |
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| 10:18.03 | neeby_goosey | r00f, I get it now, thank you. |
| 10:22.34 | flashdel | Hello :-) I am trying to setup a asterisk server as sip client connected to a fritzbox 6360, but i don`t get it working. The fritzbox has just a public ip and my asterisk is connected via router and got a private ip. The registry looks quite OK, i can call out, but i cannot receive calls. I guess its a nat problem? Can somebody help me please? Here is my config http://pastebin.de/124217 |
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| 11:02.22 | neeby_goosey | Is there a way to get the VM variables from the dialplan? https://wiki.asterisk.org/wiki/display/AST/VoiceMail+Channel+Variables Seem to be always empty to me. |
| 11:03.15 | neeby_goosey | Tried to read them in hangup extension after VoiceMail(...);, possibly wrong. |
| 11:03.55 | wdoekes | neeby_goosey: not sure those are available in the dialplan. they're available in the mail-body though |
| 11:06.11 | neeby_goosey | Sadly, I need the VM data to send it to a webinterface and the only option I see here is to parse *.txt files located next to *.wav recordings. |
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| 11:24.41 | Ice_Strike | Hello |
| 11:28.07 | bitwize | Hello Ice_Strike |
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| 13:25.49 | rgosling | hi |
| 13:26.22 | rgosling | hello world? |
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| 13:30.19 | Chainsaw | rgosling: Hello person. |
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| 14:29.24 | Katty | looks in |
| 14:29.29 | [TK]D-Fender | looks out |
| 14:31.05 | r00f | is like wat? |
| 14:31.07 | Katty | shakes it all about |
| 14:31.09 | Nivex | looks around |
| 14:31.43 | Katty | infobot: seen eppigy |
| 14:31.48 | infobot | eppigy <~Dave@snugglenets.com> was last seen on IRC in channel #asterisk, 86d 22h 52m 40s ago, saying: 'yes i am'. |
| 14:38.42 | litn | Katty: hi |
| 14:38.52 | litn | are you looking for eppigy? |
| 14:38.56 | litn | :) |
| 14:39.06 | litn | I can direct you to him if you want, shoot me a PM |
| 14:39.30 | eirirs | +_+ |
| 14:41.40 | Katty | litn: righto, just did. |
| 14:44.58 | Katty | Penguin: did you survive the storms? |
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| 15:23.07 | CheeseMouse | mjordan: would you mind taking another look at this? https://issues.asterisk.org/jira/browse/ASTERISK-23680 |
| 15:23.32 | CheeseMouse | it's not with cdr_sqlite because we are not using that module |
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| 15:52.37 | navaismo | malcolmd good morning, here is the post-->http://forums.digium.com/viewtopic.php?f=1&t=90167&sid=52ea08d7e49bbcd502d57f64c9f4c5e2 |
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| 16:30.02 | mjordan | CheeseMouse: true, it was in res_config_sqlite. Updated the component. Either way, however, SQLite has a memory corruption. |
| 16:30.15 | mjordan | CheeseMouse: an external library having a memory corruption is not something Asterisk can work around |
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| 16:31.54 | mjordan | Asterisk asked the library to execute a query. It even did it politely (mutex lock around the statement is about as nice as things get). SQLite responded by kicking Asterisk in the shins and taking its ball and going home |
| 16:32.11 | mjordan | There's nothing we can do about that. You'll need to go ask the maintainers of that library why the library is corrupting memory. |
| 16:33.28 | CheeseMouse | OK, thanks for the additional explanation. I appreciate it. |
| 16:33.53 | CheeseMouse | Isn't the asterisk keystore using sqlite? |
| 16:34.04 | mjordan | no problem. And no, we use SQLite3 (there's a difference) |
| 16:34.47 | CheeseMouse | Hmmmmm. Then why am I using sqlite (not3). This gives me more to investigate. OK, thanks agian. |
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| 17:47.48 | justdave | is there a way to make chan_iax2 flush its DNS cache? |
| 17:47.56 | justdave | iax2 reload doesn't seen to do it |
| 17:48.02 | justdave | (on 1.8) |
| 17:48.15 | justdave | doesn't seem* |
| 17:49.00 | kristoffelloos | Hi anyone who can help me get the following fixed on our Asterisk based PBX :"No Busy Tone or Announcement Message Received When Placing VoIP Outbound Calls" |
| 17:52.23 | justdave | what's the actual error message in the log when an outbound call is placed? |
| 17:54.06 | kristoffelloos | @justdave, good point there I'll check the logs next day at work first thing |
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| 18:38.55 | ghoti | Anybody happen to know what options I need to provide to ffmpeg or sox in order to generate a wav file for use as a ringtone on a Polycom VVX phone? Google doesn't like me today, it seems. |
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| 19:21.38 | jameswf | ghoti: http://community.polycom.com/t5/VoIP/VVX-500-Adding-Ringtones/m-p/18634#M2627 http://sox.sourceforge.net/sox.html |
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| 19:48.17 | fornax | Hi maybe, here is someone else this time that can help me with an asterisk / dahdi problem with a beronet bn4so. I have a clear sound in both directions when I restart my server but after several hours I have noise, but only outbound. I last time gave up because I thought an interruot sharing between my beronet and the sata controller could cause the problem, but now I’m unsure since the trouble occurs only after several hours. Any other idea how to solve |
| 19:48.17 | fornax | problem? |
| 19:48.54 | fornax | The noise is different when I use other sip devices in my network, but there is always outbound noise there |
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| 19:56.04 | ghoti | jameswf: thanks, I had that part of the documentation already, I just didn't have the translation to ffmpeg/sox command line options. Turns out it was faster to do it once in audacity, but I'll try again with command line tools when I have mor time. |
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| 20:20.49 | yokununz | someone uses chan-dongle on freebsd? |
| 20:29.38 | ghoti | yokununz: I run asterisk on freebsd, but I don't know what chan-dongle is. |
| 20:30.30 | yokununz | ghoti: it’s a chan drivers for umts cards. it allow to use a sim card as a trunk |
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| 20:56.50 | ghoti | yokununz: ah. Well, I haven't used it. But it wouldn't hurt to ask in #freebsd as well. |
| 20:57.12 | yokununz | ghoti: they said to me to ask here LOL |
| 20:57.28 | ghoti | Heh. |
| 20:58.28 | ghoti | Well, if there's a device, there's *somebody* out there who has started work on a FreeBSD driver for it. Alas, I don't know where else to point you besides Google and the FreeBSD mailing lists. |
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| 21:21.38 | drmessano | Here's how I see it: |
| 21:21.45 | drmessano | chan_dongle - Obscure |
| 21:21.55 | drmessano | Asterisk on FreeBSD - Obscure |
| 21:22.06 | drmessano | chan_dongle in Asterisk on FreeBSD - Headsplode |
| 21:22.34 | drmessano | But that's just me and i've had too much tea today |
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| 21:35.32 | navaismo | tea? are you an oldman or british? |
| 21:35.37 | ralphmazio | If I want to disable a phone, is the best way to change the password and do a dialplan reload? Will this work? |
| 21:36.13 | MaliutaLap | navaismo: those are not the only people that drink tea |
| 21:36.17 | navaismo | change the context of the phone to and empty one |
| 21:36.37 | navaismo | ralphmazio, ^ |
| 21:36.55 | navaismo | MaliutaLap, real men drink beer |
| 21:37.10 | ralphmazio | hah...dang so smart...then do a dialplan reload and I'm golden? |
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| 21:50.11 | MaliutaLap | navaismo: not all the time, and one does not exclude the other |
| 21:50.48 | navaismo | I drink Pulque! |
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| 22:37.14 | mafairnet | hi, someone here using asterisk 11 with res_xmpp for distributed state devices? |
| 22:39.28 | mafairnet | ?? |
| 22:39.42 | WIMPy | ~polls |
| 22:39.42 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
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| 22:44.04 | navaismo | so mafairnet still with issues, did you collect the debugs, logs and configs in order to share here and we can take a look? |
| 22:46.22 | mafairnet | yes, i'll post them tomorrow here i'm out of work office.. :/ |
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