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| 01:09.02 | HolTech | trying to setup a sipura 3000 fxo and fxs to *, incoming calls are making it to my dialplan, but outgoing error with Auto fallthrough, channel ... status is UNKNOWN, I have edited only sip.conf and extensions.conf, must I configure chan_dahdi.conf in this situation ? |
| 01:17.27 | ChannelZ | no it's still a SIP device |
| 01:17.33 | ChannelZ | but you have to make a separate peer for it |
| 01:18.12 | HolTech | thanks, seperate peer in sip.conf right, and then in dialplan just Dial() to that ? |
| 01:18.22 | ChannelZ | Or, well, I take that back, actually it depends. |
| 01:18.51 | ChannelZ | Are you using it as an ATA (for an analog phone) or just as an FXO for POTS calls? |
| 01:18.54 | ChannelZ | Or both? |
| 01:19.00 | HolTech | both |
| 01:19.29 | ChannelZ | OK so yes you have a SIP peer for the FXO half, but then need a separate one for the FXS half (the analog handset) |
| 01:19.48 | ChannelZ | And it's a separate configuration in the SPA |
| 01:21.06 | HolTech | seperate config on SPA, for the FXS ? I set the FXS (line1) to proxy to my * box |
| 01:21.24 | HolTech | no registration..... and I see it from my * box from my sip show peers |
| 01:21.51 | HolTech | seen some strange 404 errors in sip debug though, that may be my problem though I am unclear on how to diagnose |
| 01:22.48 | ChannelZ | Probably not, it's just how SIP behaves |
| 01:23.26 | ChannelZ | A peer tries to make a call, Asterisk says 404 Need Auth and gives back a nonce, peer re-tries with its password. |
| 01:24.37 | ChannelZ | As for the SPA, you should have something like a 'Line 1' and a 'PSTN Line' |
| 01:24.54 | HolTech | right, on two seperate ports |
| 01:24.59 | ChannelZ | 'Line 1' being the FXS (analog handset port) and 'PSTN Line' being the FXO |
| 01:27.30 | ChannelZ | Yeah. So each one has its own config, both need to be pointed at Asterisk but they need to be two separate peers, different user IDs |
| 01:30.56 | HolTech | Hm, I've done that. and verified via my dialplan which requires dialing a 9 first.... so the dial plan sends the call via Dial(SIP/fxo) where fxo is the peer(?) i defined in sip.conf |
| 01:32.04 | ChannelZ | Well you said earlier 'incoming calls are making it to my dialplan' |
| 01:32.53 | ChannelZ | So lets say that peer is called [PSTN] - so to dial out you'd do Dial(SIP/PSTN/12345) or whatnot. |
| 01:33.23 | ChannelZ | But to do that from the analog handset, that's another SIP peer. Say it's called [Phone] |
| 01:33.42 | ChannelZ | In your extensions.conf for your incoming calls, you'd Dial(SIP/Phone) to make it ring. |
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| 01:38.09 | HolTech | ChannelZ, i really appreciate your help so far, and apologies if this is newbie stuff, but the analog [Phone] would be defined in sip.conf right, then if I have a seperate context for incoming and outgoing, what should that [PHONE] 's context be ? |
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| 01:59.24 | HolTech | ChannelZ, yeah I my config was wrong, thanks agian |
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| 10:09.55 | dipanjan | If my line is busy and another person is calling, he gets a busy tone. Is it possible that the system receives the Caller ID of the second caller so that it can call him back when the line is free? |
| 10:11.06 | WIMPy | How are you getting your calls? |
| 10:13.10 | dipanjan | WIMPy: Through dongles (chan_dongle) |
| 10:13.11 | WIMPy | If you're signalling busy yourself, you already have it. If your provider prevents the 2nd call from reaching you, no. |
| 10:13.57 | WIMPy | Then if you're busy it probably doesn't even rech you. |
| 10:14.00 | dipanjan | On handsets we get the second caller as missed calls. It is called call waiting in India. I am not signalling busy. The provider does it. |
| 10:14.38 | WIMPy | Maybe you need to enable CW first. |
| 10:14.40 | dipanjan | On handsets we get a tone if a second call is coming in and we can swap calls. |
| 10:14.56 | dipanjan | You mean on the network? |
| 10:15.06 | dipanjan | ^^ enable CW? |
| 10:15.14 | WIMPy | yes |
| 10:15.32 | dipanjan | I will check this. Probably it is already enabled. |
| 10:15.35 | WIMPy | *43# iirc |
| 10:16.31 | dipanjan | Will vary with n/w? |
| 10:16.54 | WIMPy | no |
| 10:17.23 | WIMPy | The codes should be standard. |
| 10:24.14 | _omer | Anyone please help me in MessageSend() .... |
| 10:24.38 | _omer | I found an example but dont know why it is not working ... I am using Asterisk 11 |
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| 10:32.40 | mattmonkeymagic | Hi, can anyone help with a handset problem please? |
| 10:33.32 | mattmonkeymagic | I've got a YeaLink T20P which has a dial tone and can connect both parties, but on an outbound call there is no ringing to indicate the call is in progress |
| 10:35.51 | mattmonkeymagic | but once connected both parties can converse without issue |
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| 10:58.45 | deo | hi guys i have some weird problem with my pabx this afternoon.. when inputting and extension during an ivr its says Invalid extension.. |
| 10:58.56 | deo | but i inputted the corrrect extension :( |
| 10:59.36 | deo | ive noticed that inputting the first digit of the extension will say Invalid extension |
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| 11:17.58 | dipanjan | WIMPy: enabled CW but nothing shows up on the CLI when the call is waiting to suggest that there is a second call. The caller gets a "person you are calling is on another call. please hold the line..." tone. But nothing on the CLI. |
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| 11:27.33 | WIMPy | dipanjan: That must be a missing feature in chan_dongle then. |
| 11:28.36 | mattmonkeymagic | I've seen a couple of suggestions that maybe setting progressinband=no in sip.conf may resolve issue, can anyone confirm?? |
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| 11:37.21 | WIMPy | Try it |
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| 11:46.57 | mattmonkeymagic | All of my soft phones are working without this problem. Any ideas anyone?? |
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| 12:12.13 | _omer | hello. |
| 12:12.16 | _omer | any help ? http://pastebin.com/jQ9DYY38 |
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| 12:27.48 | _omer | any help ? http://pastebin.com/jQ9DYY38 |
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| 12:29.46 | [TK]D-Fender | _omer, [Apr 2 12:07:41] WARNING[39985][C-00000000]: message.c:1104 msg_send_exec: No message technology '9000' found. |
| 12:30.01 | [TK]D-Fender | _omer, Just like it says, you are missing the TECH part of that command, |
| 12:30.09 | [TK]D-Fender | _omer, -- Executing [9000@messages:8] MessageSend("Message/ast_msg_queue", "9000,9001") in new stack |
| 12:30.19 | [TK]D-Fender | _omer, "just like dial needs.... |
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| 12:32.46 | protocoldoug | yep, docs say they want a URI in messagesend, like: MessageSend(sip:127.0.0.1:5060) |
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| 12:34.27 | [TK]D-Fender | _omer, https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MessageSend |
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| 12:48.44 | _omer | oh...so it means message needs to send on IP and PORT of that specific sip peer... |
| 12:50.15 | [TK]D-Fender | _omer, It needs to know that you are sending it via SIP in the first place... |
| 12:50.45 | _omer | yes. I got it ... something like this. MessageSend(sip:127.0.0.1:5060) |
| 12:51.42 | _omer | and I can fetch it from Executing [9000@messages:2] NoOp("Message/ast_msg_queue", ""9001" <sip:9001@xx.xx.xx.xx>") in new stack |
| 12:51.44 | _omer | let me try. |
| 12:55.08 | _omer | it's working now....thank you protocoldoug & [TK]D-Fender .... |
| 12:55.21 | _omer | bye . take care |
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| 12:57.16 | [TK]D-Fender | \o/ - documentation |
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| 13:08.11 | _omer | one quick question, How to ${MESSAGE(from)} and ${MESSAGE(to)} to fill MessageSend(sip:(TO_IP):(TO_PORT),(FROM_PEER)) |
| 13:08.24 | _omer | how to get ... |
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| 13:15.57 | mattmonkeymagic | Also the hard phone has a ringing tone when calling from extension to extension. Does this point to a problem in the dial plan?? I've checked that i'm passing 'tr' to Dial,etc |
| 13:16.20 | mattmonkeymagic | problem only seems to be when ringing to an external party |
| 13:19.37 | Greenlight | It's ususally not neccissary to pass "r" to dial when calling externally |
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| 13:26.20 | [TK]D-Fender | ...or ever |
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| 13:33.37 | mattmonkeymagic | I've tried with both 'r' being present and not. Odd that its giving a ringing tone extension to extension, which is making me wonder if problem is tied to the external context only (or something else that I've missed??) |
| 13:34.32 | [TK]D-Fender | mattmonkeymagic, You might want to actually show us a complete call with appropriate channel debug enabled... |
| 13:34.33 | [TK]D-Fender | ~pb |
| 13:34.34 | 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. |
| 13:34.35 | [TK]D-Fender | ^^^ |
| 13:35.11 | mattmonkeymagic | ok Fender, yep paste bin a good Idea as my dial plan is over 3.5MB ;-) |
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| 13:35.47 | [TK]D-Fender | MattmWe don't need to see the dialplan. We need to see the CALL. |
| 13:36.08 | mattmonkeymagic | ok sure |
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| 13:38.41 | igcewieling | mattmonkeymagic: "r" seldom helps and often causes issues |
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| 13:50.16 | Greenlight | Damn asterisk just segfaulted again trying to read a null pointer :( |
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| 14:04.27 | mattmonkeymagic | Hi all, I've pasted the call up to paste bin. I've sanitized the external number as my_external_number in the log entries (but there was a UK mobile number in its place originally) |
| 14:04.39 | mattmonkeymagic | here goes: http://pastebin.com/JyWbkBYq |
| 14:06.37 | [TK]D-Fender | mattmonkeymagic, SIP DEBUG was what was called for there.... |
| 14:06.57 | [TK]D-Fender | mattmonkeymagic, And we advise you use a real SSH client that doesn't litter your output with ANSI junk |
| 14:07.36 | [TK]D-Fender | mattmonkeymagic, This output is 3x bigger than it deserves to be an is not properly seachable |
| 14:09.05 | mattmonkeymagic | ok, thought that sip debug would produce too much as we are over 500 active seats :-) |
| 14:09.47 | mattmonkeymagic | I'll see if I can grab a smaller output with sip debug on in the early hours tomorrow |
| 14:10.25 | mattmonkeymagic | thanks for the help so far |
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| 14:28.15 | Free99 | kind of a noob question, I tried searching the docs but can't figure this out: how would one create a specific outbound route? That should be setup as part of a dialplan, right? |
| 14:28.32 | WIMPy | yes |
| 14:29.26 | Free99 | Thanks WIMPy |
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| 14:40.28 | Greenlight | Hmm how odd. I'm tracking of of the call involved in the segfault that's killed asterisk a couple of times now. Seems almost 3 minutes into the call I see "Unlink" and "Link" AMI events being fired for that channel - Any ideas why I would get such events that far into the call ? |
| 14:41.08 | WIMPy | session-timers? |
| 14:41.16 | Greenlight | Ahh they trigger those events? |
| 14:41.45 | Greenlight | A red herring then, damn |
| 14:41.53 | WIMPy | Not sure exactely. They surely trigger bridged events. |
| 14:43.31 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:44.29 | mjordan | Greenlight: Link/Unlink events are a bit goofy as they can fire quite often. They occur when a channel in a bridge is removed/unremoved from a bridge. That doesn't necessarily imply what you think however - a channel is 'removed' from a bridge every time certain actions are taken on the channel, such as a DTMF digit being played on the channel. |
| 14:45.23 | Greenlight | Ok that's fair enough - just wondered if something funky was going on with the channel which leads to the segfault. Guess it's not that |
| 14:46.13 | WIMPy | Just because it's normal doesn;t mean it's unrelated. |
| 14:47.28 | Greenlight | That's true.. although it's another 2 1/2 minutes till the next action happens on the channel which is what directly leads to the null pointer dereferece and segfault |
| 14:50.02 | mjordan | do you have a backtrace? |
| 14:50.46 | mjordan | or is it the same issue as ASTERISK-21356? |
| 14:50.58 | Greenlight | It's same issue, just another occurance |
| 14:51.18 | mjordan | kk |
| 14:52.06 | Greenlight | Just trying to get an idea what's happening so I could try and avoid the circumstances that lead to the issue |
| 14:52.29 | *** join/#asterisk lorsungcu (~anonymous@72.21.243.18) |
| 14:52.41 | lorsungcu | anyone have LLDP working with D40s and procurve switches? |
| 14:52.43 | Greenlight | Since it appears to be more frequenct than I'd initially hoped |
| 14:53.48 | Greenlight | I wondering if there are some issues with the dual redirect code, but then again, issuing two single redirects in my view is going to be asking for more deadlocks as one channel will try and resume in the dialplan at the moment I issue a redirect |
| 14:54.23 | Greenlight | Unless there's another method people are using to conference in a 3rd party to a call ? |
| 14:58.06 | *** join/#asterisk lorsungcu_ (~anonymous@72.21.243.18) |
| 15:02.33 | igcewieling | Greenlight: we press the CONFERENCE button on our phones to make a 3-party conference call 8-) |
| 15:03.33 | Greenlight | :) |
| 15:03.33 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:03.51 | Greenlight | No scope for something to go wrong on Asterisk side there |
| 15:04.05 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 15:05.30 | *** join/#asterisk lorsungcu_ (~anonymous@72.21.243.18) |
| 15:05.59 | WIMPy | If it's a SIP phone. |
| 15:06.04 | Greenlight | One thing that both the calls which have caused segfaults do have in common is that they started of life coming into a queue. Instead of letting the queue deal with the call "normally", the call is bridged out of the queue by an AMI Bridge command. However that all works perfectly fine 99.99% of the time. I'm just wondering if the queue is still keeping some sort of pointer to the channel or |
| 15:06.04 | Greenlight | something like that. |
| 15:06.57 | igcewieling | Greenlight: what version of Asterisk? |
| 15:07.09 | Greenlight | 11.3.0-rc1 |
| 15:08.34 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:09.15 | Greenlight | The calls in question are fine when they're first ripped out the queue - we're speaking to customer for a good bit of time, until we try and redirect into conference, but wonder if the queue is still holding onto something. |
| 15:10.46 | [TK]D-Fender | Greenlight, 11.3.0 full is out, as is 11.4.0.rc1 |
| 15:12.05 | Greenlight | oo |
| 15:12.10 | Greenlight | goes to checkthe changelog out |
| 15:19.51 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:19.51 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:21.06 | *** join/#asterisk joesmoe_ (~joesmoe@164.sub-70-199-99.myvzw.com) |
| 15:23.23 | Katty | hello my asterisk does not work at all how to fix??? is urgent plz answer thx. |
| 15:23.30 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 15:23.30 | Katty | srs answers only plz. |
| 15:24.55 | Greenlight | Damn nothing jumps out as related to my specific issues in the 11.4.0-rc1 changelog. Although there's a heap of ConfBridge related fixes so it's certainly worth the upgrade. |
| 15:25.23 | madduck | I just witnessed something really weird. Incoming calls to our phones are handled by stdexten, which redirects to voicemail after 30 seconds. There was a call coming in to my private number, which I route to my cell, office and home phones. It took 61 seconds for the call to be "not answered". Meanwhile, my wife's phones rang for 28 seconds. Here are the log entries: http://slexy.org/raw/s2JkbZdUer |
| 15:25.28 | madduck | How is this even possible? |
| 15:25.56 | madduck | I just spent an hour trying to reproduce the problem but without luck, it must have been a hiccough or a timing issue. |
| 15:27.37 | drmessano | Katty: Have you upgraded to SVN TRUNK? |
| 15:27.47 | Katty | no, but i upgraded your mom. |
| 15:28.01 | Katty | ON SNAP |
| 15:28.03 | Katty | hugs drmessano |
| 15:28.07 | drmessano | lol |
| 15:28.17 | drmessano | hugs Katty |
| 15:28.24 | drmessano | I almost hugged kaldemar |
| 15:28.33 | drmessano | TAB TAB |
| 15:28.59 | Katty | i bet kaldemar would take a hug. |
| 15:29.12 | Katty | kaldemar: WOULD YOU LIKE A HUG SIR. |
| 15:29.13 | madduck | 09:23:10 call comes in — :31 "NO ANSWER" — :38 other phone rings until :24:06 "NO ANSWER" — :24:11 voicemail picks up… |
| 15:32.45 | madduck | here are the two entries in the dialplan: |
| 15:32.47 | madduck | http://slexy.org/raw/s2QSjmopok |
| 15:33.28 | madduck | there is *no* *way* by which this call could have been routed to my wife's phones… |
| 15:33.38 | *** join/#asterisk jMCg (~hiro@apache/committer/igalic) |
| 15:34.53 | [TK]D-Fender | madduck, CDR proves nothing to us of the processing your call wnet through. You need to show us an actually call trace |
| 15:36.21 | [TK]D-Fender | madduck, And your use of the term "stdexten" is also entirely specific to you as it's your code, not some specific internal function of Asterisk itself. |
| 15:36.23 | Katty | hi fender bender. |
| 15:36.27 | [TK]D-Fender | Katty, Mew. |
| 15:36.34 | Kobaz | kattyrific |
| 15:36.41 | *** join/#asterisk joesmoe (~joesmoe@203.sub-70-199-100.myvzw.com) |
| 15:36.51 | Katty | [TK]D-Fender: you causin trouble today? |
| 15:36.53 | Katty | hugs Kobaz |
| 15:36.57 | Kobaz | yay |
| 15:37.11 | Katty | Kobaz: how'rechu n life n things? |
| 15:37.18 | [TK]D-Fender | Katty, Nope. Easy day today... |
| 15:37.27 | Katty | horay for easy days! |
| 15:37.56 | Free99 | this may be an off-base question, but... anyone have any experience with cron spamming the hell out of your inbox? |
| 15:38.01 | Kobaz | haha what a crazy contraction |
| 15:38.06 | Kobaz | how'rechu |
| 15:38.15 | Kobaz | is that like pikachu |
| 15:38.28 | Free99 | I'm considering using this: http://habilis.net/cronic, but if anyone knows of a better way... |
| 15:38.52 | Kobaz | Katty: had a bad morning, something last night destroyed my digestion, been up since 3am |
| 15:38.57 | Katty | :< |
| 15:38.58 | Kobaz | i think i'm going to take a nap in a few hours |
| 15:38.59 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 15:39.00 | Katty | Kobaz: poor thing. |
| 15:39.12 | Katty | Kobaz: i hope you feel better soon :> |
| 15:39.31 | Kobaz | i had some eggs and yogurt, getting a little better |
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| 15:40.13 | madduck | [TK]D-Fender: stdexten is a macro/sub defined in the upstream extensions.conf… |
| 15:40.13 | Katty | Kobaz: woo! |
| 15:40.14 | WIMPy | Free99: OT, but make sure whatever you start doesn;t produce output, e.g. by redirecting everything to /dev/null/ |
| 15:40.29 | madduck | [TK]D-Fender: I cannot get a call trace retrospectively, can I? |
| 15:40.31 | Kobaz | it's hard to eat and not produce output |
| 15:40.36 | Free99 | WIMPy: but then if it fails, how would I know? |
| 15:40.41 | igcewieling | Free99: yes |
| 15:40.55 | igcewieling | heh, I was reading scrollback |
| 15:40.55 | [TK]D-Fender | madduck, Check your logs or go reproduce the error |
| 15:41.02 | WIMPy | Free99: You have to decive if you want the onformation or not. |
| 15:41.14 | Free99 | if it fails or errors out, then yes lol |
| 15:41.26 | WIMPy | Script it |
| 15:41.27 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:41.30 | igcewieling | Free99: make the script only output on error. if there is no output, cron won't e-mail you |
| 15:41.38 | [TK]D-Fender | <madduck> [TK]D-Fender: stdexten is a macro/sub defined in the upstream extensions.conf
<- we don't know what's in your code. Just saying you have a macro doesn't prove it's being used, or what it does, or what actually happened to that call. |
| 15:41.51 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 15:41.58 | igcewieling | if your script only outputs to stderr when there is an error you can > /dev/null to throw away stdout |
| 15:42.05 | madduck | [TK]D-Fender: it's not my macro, it's from the upstream tarball, AFAICT. |
| 15:42.19 | Free99 | igcewieling: I've been using >/dev/null 2>&1 |
| 15:42.24 | igcewieling | madduck: define "upstream" |
| 15:42.28 | madduck | [TK]D-Fender: I cannot reproduce the problem, I've been trying for almost two hours now. And the logs are not verbose… |
| 15:42.39 | igcewieling | Free99: the 2>&1 also discards errors |
| 15:42.44 | Free99 | (facepalm) |
| 15:42.46 | Greenlight | SOunds like it's not raw asterisk... |
| 15:43.13 | Free99 | I'll keep digging around, thanks igcewieling and WIMPy |
| 15:43.29 | igcewieling | Free99: the traditional solution is to set up a mailbox for the messages to go to with every intention of checking it and then totally forgetting about it. |
| 15:44.36 | WIMPy | ... until you wonder why your disk is full. |
| 15:44.38 | Free99 | igcewieling: I used to be a VM admin for my school, I took over someone else's system that did exactly that for VM disk image backups, I found out about a month in that the backup script had been failing for about 6 months |
| 15:44.41 | madduck | igcewieling: tar xfz asterisk_1.8.13.1.tar.gz asterisk-1.8.13.1/configs/extensions.conf.sample && grep stdexten -B10 asterisk-1.8.13.1/configs/extensions.conf.sample |
| 15:45.02 | *** join/#asterisk qakhan (~qakhan@208.253.91.58) |
| 15:45.08 | qakhan | hi all |
| 15:45.16 | Free99 | so I dislike that kind of sweeping under the rug thing :-/ |
| 15:45.36 | WIMPy | That's where bugs go naturally. |
| 15:46.27 | Kobaz | snug like a bug in a rug |
| 15:46.46 | qakhan | i have 3 queues and i want to record call log of each queue in queuname table. record should be calltime, callerid, agentid(who pickedup call) |
| 15:47.04 | qakhan | hold time in queue, talk time, |
| 15:48.01 | qakhan | how can i do this, please help |
| 15:55.00 | [TK]D-Fender | qakhan, You'll have to be dialing local channels as members and set your monitor in there, not in the queue itself. |
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| 16:07.04 | qakhan | [TK]D-Fender i want to have call history in queue |
| 16:07.43 | qakhan | and make table of queue name like queue name is "abc" table name is "abc" |
| 16:08.49 | qakhan | table abc save calltime, callerid, agentid(who pickedup call), hold time in queue, talk time, call hang by(caller) or (agent) |
| 16:11.11 | [TK]D-Fender | qakhan, You've already got your answer. You set the name yourself in the dialplan used to call each agent. that is the only way. |
| 16:12.42 | qakhan | [TK]D-Fender i am sorry i alway have problem to understand you, can you please tell me in detail |
| 16:12.51 | *** join/#asterisk kontinuity (~kontinuit@122.172.246.161) |
| 16:13.46 | [TK]D-Fender | qakhan, What is unclear? You have to use LOCAL CHANNELS as your "agents" and YOU have to call Monitor YOURSELF passing it the filename you want. |
| 16:14.02 | [TK]D-Fender | qakhan, There is no queues.conf option to configure the name the way you want. |
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| 16:22.50 | lorsungcu | anyone have D40s working with procurve switches? |
| 16:22.54 | lorsungcu | using LLDP |
| 16:37.10 | *** join/#asterisk menti_ (5d9cdaba@gateway/web/freenode/ip.93.156.218.186) |
| 16:38.15 | menti_ | Hi guys, may i ask you a question about the Manager? i have a little problem... :P |
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| 16:41.20 | newtonr | menti_: yup |
| 16:41.49 | qakhan | [TK]D-Fender how can i get hold time and Talktime in a queue and save it in cdr or queue_log table |
| 16:45.55 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 16:46.26 | menti_ | @newtonr hi, i've just got a "Chuck Norris Time" and i solved the privileges problem, but i still have a problem with the Channel, when i try to originate a call, i use the Channel "SIP/enterhereusername" |
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| 16:52.07 | newtonr | menti_: okay. did you have a question? |
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| 16:53.54 | menti_ | @newtonr yes sure, my question is why i can't connect the Channel (for example (SIP/menti) if i have a "menti" user registered? (or i think so, Zoiper marks "menti" as registered) do you know what can be the problem? Thank you very much |
| 16:54.41 | [TK]D-Fender | <qakhan> [TK]D-Fender how can i get hold time and Talktime in a queue and save it in cdr or queue_log table <- hold time and talk time ... is ALREADY in the queue log |
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| 16:55.31 | [TK]D-Fender | menti_, Show us. |
| 16:55.33 | [TK]D-Fender | ~pb |
| 16:55.33 | 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. |
| 16:55.35 | [TK]D-Fender | ^^^ |
| 16:56.28 | newtonr | menti_: nope I have no idea what the problem is. Like TK said, you'll have to show debug. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information is helpful, plus remember to turn on debug for manager. |
| 17:00.20 | menti_ | oh.... guys @newtonr & [TK]D-Fender problem solved, all the config was ok, the problem was in the softphone, i changed it and everything works fine, i'm so sorry I bothered you I was desperate because all google solutions did not help |
| 17:01.57 | menti_ | For the record, i was using Zoiper, it works with SJphone |
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| 17:17.23 | Free99 | hmm. Is it possible to add the local sip server as a route? I currently have a2billing working if I'm calling out of the network, but I can't call between SIP peers |
| 17:21.29 | [TK]D-Fender | Free99, that's up to you and your dialplan. |
| 17:21.35 | [TK]D-Fender | Free99, go make extensions for them |
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| 17:22.18 | Free99 | well, the peers have extensions (and a2billing annoyingly follows no rules at all for generating numbers for the peers) |
| 17:25.27 | Free99 | I guess the real question is, how do I charge inter-SIP calls? |
| 17:26.20 | Free99 | I know I can set the dialplan to do different actions if, say, the user adds a 9 to the front of the number |
| 17:26.44 | Free99 | you know, [tk] I kind of hate asking you about all this stuff b/c I feel like the info is out there but I can never find it |
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| 17:34.56 | [TK]D-Fender | Free99, You make your own extensions for your devices. This is your job |
| 17:35.53 | [TK]D-Fender | Free99, And the only calls you are supposed to be passing a2billing are for outbound. It's not supposed to care about letting your peers interact with each other. |
| 17:36.25 | Free99 | [tk]d-fender: crud. |
| 17:37.18 | Free99 | [tk]d-fender: also, I'm letting a2b handle making the peer secrets and such, that's what I've been interpreting "extensions" as, peers that have registered |
| 17:37.49 | [TK]D-Fender | Free99, Extensions = extensions.conf |
| 17:37.59 | Free99 | got it. |
| 17:39.23 | Free99 | so if a2b isn't meant to handle inter-peer stuff, I should be able to make a "loopback", that is, if I can make the local system appear as a trunk, then a2b will bill as if it went out, but it's just dialing a local peer |
| 17:41.24 | [TK]D-Fender | Free99, a2b is for dialing out. It has no fallback. It doesn't do any of the rest of your job for you. You ahve to actually work. There is no dodging this bullet. |
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| 18:14.02 | BCS-Satori | I have a global complaint since moving from SIP Trunk to DAHDI that when users call an auto attendant (outside the organization) and press a button (such as 1) they DTMF tone sounds three times very fast which causes the auto attendants to skip forward. This doesn't occur when I set them to use the SIP Trunk. Where should I be looking to correct the issue? Could it be the hardware echo cancel? |
| 18:18.07 | igcewieling | sounds like 1) you have relaxdtmf set to yes. don't do that OR 2) you need to lower your rxgain or txgain |
| 18:22.48 | BCS-Satori | igcewieling: Yes relaxdtmf is set. Your suggesting to turn it off? the rxgain and txgain is set to 0.00 currently |
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| 18:26.37 | igcewieling | correct. relaxdtmf is an alias for the "screwupmydtmf" option. |
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| 18:46.08 | fkurkowski | Anyone mind taking a look at this? http://pastebin.com/3twCgYhh |
| 18:46.26 | fkurkowski | Basically, we have been trying to make our dual card setup work properly but one FXO channel is unresponsive and just keeps ringing when called. All others are picking up. |
| 18:47.40 | [TK]D-Fender | fkurkowski, Show us the calls to the working and the non-working channels along with "dahdi show status", etc |
| 18:48.10 | fkurkowski | Alright, one second. |
| 18:49.22 | fkurkowski | I also forgot to mention that this box is not directly connected to the PSTN, but to our internal legacy PBX. |
| 18:50.10 | [TK]D-Fender | fkurkowski, Be sure to swap the actual wired to prove that it's not the Legacy PBX side as well.. |
| 18:54.14 | fkurkowski | Yeah, I might as well connect it directly to the PSTN just for testing sake. I was simply swapping the wires from the legacy to Asterisk and, no matter the ordering, the same channel was failing. |
| 18:55.03 | [TK]D-Fender | Well that at least settings that its as of the card. next, swap the MODULES themselves on the card to see if the individual module has frieds (which I've seen several times) |
| 18:57.45 | fkurkowski | Sounds good, gonna try that and report back. Thank you! |
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| 19:24.32 | fkurkowski | I swapped the modules on the card, regenerated the configuration and now both FXO channels are unresponsive. Weird. |
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| 20:37.57 | Martin` | Hi guys! (and girls?) |
| 20:40.33 | Martin` | They are constandly users (I call them hackers) who try to log in on my asterisk server via voip. what is the best way to protect the server from this? |
| 20:41.08 | Martin` | I currently disabled all users after I had to pay a lot of money :( |
| 20:41.32 | igcewieling | Martin`: welcome to the world of telephony. Most people use iptables with or without fail2ban |
| 20:41.54 | igcewieling | We are lucky, all of our customers are "on-net" so our iptables filters are fairly simple |
| 20:42.45 | Martin` | well that is a littlebit the problem, I need much different ip's to connect from :( |
| 20:43.10 | Martin` | but that can be solved by vpn offcourse :P |
| 20:46.05 | igcewieling | then you want to also use fail2ban |
| 20:46.46 | navaismo | Martin read the bestpractices TXT always, dont use allowguest, use alwaysauthreject and block hosts like fail2ban or blockhosts based on the asterisk logs |
| 20:46.57 | navaismo | aaah to late again |
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| 20:58.00 | drmessano | ~happyclownpbx |
| 20:58.00 | infobot | currently in closed beta, approaching 12GB in size, uses Asterisk for its core, it pwns, eats your children (seriously), and is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 21:01.42 | Martin` | ok |
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| 21:04.39 | Martin` | and how do they get money when they guessed my password? |
| 21:07.03 | Martin` | I see a lot calls to france phone nr's |
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| 21:08.21 | navaismo | resell or they just want to call for free |
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| 21:13.53 | Martin` | navaismo: but they did call 1000 times to the same nr in one day |
| 21:14.35 | work | please how to record all call from dial plan |
| 21:17.06 | navaismo | work in vanilla asterisk use mixmonitor before the call you can set 'b' option for only record bridged calls |
| 21:17.26 | navaismo | remember freepbx use different kind of routines/macros |
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| 21:21.48 | work | <navaismo> in extentions file ? what the syntax of mixmonitor |
| 21:23.54 | navaismo | execute: core show application mixmonitor, from tehasterisk cli |
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| 21:27.09 | navaismo | s/tehasterisk/the asterisk/ |
| 21:30.07 | work | <navaismo> please look at this file extentions http://pastebin.com/YCFs1YVi |
| 21:30.34 | work | I do not know where to set this line |
| 21:31.46 | work | I need to call sip calls at first then I will move to record all calls from-digital |
| 21:32.36 | navaismo | Ok first read the top of that file, second please attach to one IRC i told you in #freepbx that you DO NOT edit conf files by hand |
| 21:33.00 | navaismo | If you want to edit the conf files switch to vanilla asterisk instead freepbx |
| 21:35.17 | navaismo | bbl |
| 21:36.27 | work | ok without edite files |
| 21:39.13 | igcewieling | based in ny limited research, people hack into PBXs then patch the PBX into a calling card app and they sell calling cards to unsuspecting people |
| 21:40.01 | igcewieling | We found adding a message "You have reached the Immigration and Naturalizaion Service, please wait, someone will be with you shortly" made the calls top within 10 mins |
| 21:41.26 | work | navaismo please I need to edite files |
| 21:51.30 | work | please how disable freepbx and move to edite asterisk files |
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| 21:52.17 | b0ot | can anyone reccommend any sip or fxs/fxo based gsm gateways? |
| 21:54.10 | newtonr | work: read through the Asterisk Definitive Guide... http://shop.oreilly.com/product/9780596517342.do |
| 21:55.21 | newtonr | work: then you'll understand overall how Asterisk works. If you still want to use FreePBX at that point then at least you'll know how to use it and maintain custom configuration files for it as well. |
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| 22:13.10 | Martin` | configured fail2ban, hope it works :) |
| 22:13.23 | Martin` | because now nobody tries to login :( |
| 22:13.30 | navaismo | try it yourself |
| 22:13.47 | navaismo | with bad credentials, bad destination numbers etc |
| 22:17.20 | Martin` | looks like it works, can't reach my server at the moment :P |
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| 22:41.35 | anonymouz666 | anyone in here doing passthrough of v90 modem with asterisk? |
| 22:47.55 | tm1000 | maybe im missing something but shouldn't asterisk 12 be in here: http://svn.asterisk.org/svn/asterisk/branches/ |
| 22:52.44 | tm1000 | nevermind file helped me out |
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| 23:12.17 | Martin` | navaismo: hope I'm save now, now I can happy pay the bill |
| 23:12.57 | Martin` | navaismo: And I need something that alerts me when I call more than 60 minutes in one day |
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| 23:16.48 | navaismo | For the alert you can use an AMI listener to check the call duration or a cronjob dumping the duration of calls and then email you something |
| 23:16.59 | navaismo | or set the absolute time for all calls |
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| 23:27.13 | Katty | should i do a lot of squats without weights....or less squats with weight? |
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| 23:33.08 | phix | Katty: more squats with weights |
| 23:33.20 | Katty | i can't do both at this point |
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| 23:33.35 | Katty | so which is better for now |
| 23:33.40 | Martin` | navaismo: ami listener? ok I will check that tomorrow :) |
| 23:33.57 | phix | repition is generally better than more weights |
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| 23:34.03 | Martin` | navaismo: thx for your help |
| 23:34.08 | phix | but I think you are asking this in the wrong channel :) |
| 23:34.16 | Katty | this channels is perfect |
| 23:34.20 | Katty | everyone is smrt. |
| 23:34.23 | phix | hehe |
| 23:34.37 | Martin` | ah ami is asterisk management interface, :) |
| 23:34.40 | Katty | in various topics ^_^ |
| 23:35.14 | Katty | operation cuter bum in effect! /afk |
| 23:35.19 | phix | also take some caffiene pills, research has found that people working out while taking caffiene get a better workout then people not using caffiene |
| 23:35.19 | Martin` | gone |
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| 23:35.44 | phix | of course caffiene is also linked to kidney stones so drink lots of water :P |
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| 23:53.58 | phix | http://www.bizarrebytes.com/pour-beer-from-the-bottom-of-the-cup/ |
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| 23:55.04 | phix | http://pbfcomics.com/258/ |
| 23:55.15 | phix | sorry, that was Katty's fault |
| 23:56.01 | tzanger | phix: hahaha |
| 23:56.03 | tzanger | that's awesome |
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