| 00:05.22 | lvlinux | vslap: on mine, when i leave a vm, immediately after the user hangs up, * says it found msg0000.txt and then sends a SIP NOTIFY to the phone Messages-Waiting: yes |
| 00:08.31 | vslap | testing it now. |
| 00:09.00 | vslap | in voicemail.conf i have mailbox=200, does it need to be 200@default |
| 00:12.42 | lvlinux | you mean in sip.conf? |
| 00:13.33 | WIMPy | or extensions.conf? |
| 00:18.33 | vslap | yea sip.conf |
| 00:18.48 | vslap | idk, i got my phone working, the only thing i did different was add @default. |
| 00:18.59 | vslap | The weird thing is, on reboot, they get the MWI, but not updates. |
| 00:20.18 | lvlinux | adding the @default can help sometimes IIRC |
| 00:20.36 | lvlinux | depends on the phone model/brand |
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| 01:23.30 | ageis | i cant make an outgoing call |
| 01:23.36 | ageis | its sending SIP packets in debug but nothing shows on console |
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| 01:28.51 | [TK]D-Fender | ageis: "sip set debug on" <----- |
| 01:29.11 | [TK]D-Fender | ageis: "core set verbose 10" |
| 01:29.24 | [TK]D-Fender | enable both and pastebin the COMPLETE call attempt |
| 01:29.27 | [TK]D-Fender | ~pb |
| 01:29.27 | 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. |
| 01:29.30 | [TK]D-Fender | ^^^ |
| 01:32.55 | onixx | [TK]D-Fender>: I think I found the issue with the features_config.c:1301 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/2216-0000000f error message. |
| 01:33.37 | onixx | [TK]D-Fender: I had DYNAMIC_FEATURES=>automon in the [globals] section of extensions.conf |
| 01:33.47 | onixx | and I believe it is no longer required... |
| 01:34.18 | [TK]D-Fender | IIRC, you only have to specify the applicationmap stuff.... |
| 01:35.51 | onixx | [TK]D-Fender: exactly! and I believe automon changed. it used to mix automatically for some reasons and I never bothered changing to automixmon for that reason |
| 01:36.00 | onixx | It no longer does with ast-12 |
| 01:36.23 | [TK]D-Fender | Hrm. Haven't touched 12 yet myself. |
| 01:36.31 | [TK]D-Fender | I need to get my hands dirty and get up to date... |
| 01:37.10 | [TK]D-Fender | I still remember before we crossed the "whole-number" mark :) |
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| 01:38.54 | onixx | [TK]D-Fender: I just got off an SNV snapshot from 2013 to 12 ;-) its tought when you have not much reference point as to what changed |
| 01:39.51 | coppice | [TK]D-Fender: you only remember as far back as when it had numbers? |
| 01:40.11 | coppice | [TK]D-Fender: and a name? |
| 01:40.32 | [TK]D-Fender | coppice: 1 = first whole number :) |
| 01:40.41 | [TK]D-Fender | So pre Asterisk 1.0 |
| 01:40.49 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:41.13 | [TK]D-Fender | Back in the age of "Stupid Impractical Fractions" |
| 01:41.13 | coppice | that's so late in the game |
| 01:41.21 | [TK]D-Fender | I KNOW! |
| 01:41.26 | [TK]D-Fender | Horrible isn't it? |
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| 01:43.14 | coppice | I remember when it was a zapata card and a gleam in a developer's eye |
| 01:44.40 | [TK]D-Fender | coppice: Yes my antediluvian friend :) |
| 01:53.19 | coppice | [TK]D-Fender: have you been watching Noah? |
| 01:53.39 | [TK]D-Fender | Nope.... |
| 01:54.00 | lvlinux | coppice: is it any good? |
| 01:54.13 | [TK]D-Fender | I've just got a surprisingly large vocabulary... |
| 01:58.47 | coppice | dunno. with all this noah stuff around you might have just picked it up from a poster |
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| 02:08.31 | DonX | Does anyone have a working example of a valid naptr record that performs a substitution |
| 02:10.42 | DonX | eg: when querying a zone for 12148885555, *.8.8.8.4.1.2.1 is matched and then returns 12148885555@sip.proxy:5060 |
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| 03:00.48 | puzzled | [TK]D-Fender: I remember those days too. When Asterisk did not yet have sip timers and a 1000 buggy phones did not hang up correctly |
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| 03:07.43 | BenMerton | I'm having a problem with my extensions.conf - I've clearly made a mistake with my macro-queue & macro-queue-error in that when an incorrect queue key is pressed it goes to macro-queue-error but when it returns to the macro-queue it has slightly renamed the queue name, is this is the right place to get some help for it? |
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| 04:16.56 | ipengineer | Is there anything special that needs to be done in Asterisk to get voicemails to send as attachments? I have sendmail installed and can send a test email but looking at the logs it looks like Asterisk isnt event trying to send an email when a voicemail is left |
| 04:43.02 | ChannelZ | Whatever mailcmd is set to will be called, sendmail by default IIRC. Did you configure everything else like emailsubject, emailbody? Got the right syntax for the mailbox in question? Reloaded voicemail after changing the config? |
| 04:43.40 | ChannelZ | (turned on attach?) |
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| 05:56.50 | BenMerton | In my extensions.conf file - when a queue is entered it's passed ARG1,2&3. It then runs exten => s,1,Set(QUEUE=${ARG1}-${ARG2}), which results in: Executing [s@macro-queue:1] Set("SIP/xxxxxxx-xxxxxx", "QUEUE=ARG1-ARG2") in new stack - perfect! When a queue button is pressed in error it runs the [macro-queue-error] context which runs exten => s,n,Macro(queue,${QUEUE}) which results in: Executing [s@macro-queue-error:2] Macro("SIP/ |
| 05:58.21 | BenMerton | That too is perfect, the issue is when it returns back to the [macro-queue] context it goes through the first exten => s,1,Set(QUEUE=${ARG1}-${ARG2}) again, only it takes ARG1 to be the contents of the previous statement and ARG2 to be blank - giving me: Executing [s@macro-queue:1] Set("SIP/xxxxxxx-xxxxxx", "QUEUE=ARG1-ARG2-") in new stack which breaks it because there's no queue ending in a - |
| 05:59.37 | BenMerton | I need to be able to retain ARG1, 2 & 3 so when it goes back to the Queues context after an error button is pressed, it can run through again with the correct ARG1-ARG2,ARG3 format |
| 06:00.02 | BenMerton | is anyone able to help shed some light on my problem? |
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| 08:54.07 | carrar | moo |
| 08:54.30 | carrar | throws glitter at Katty |
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| 08:58.27 | mushroomed | Hi |
| 09:00.49 | mushroomed | I'm having a really annoying problem, I have an Asterisk Server at my VPS (cloud) and I have two Grandstream phones in my house, each one configured with different accounts (extensions). The problem is that when I call between them it says 'Got SIP response 486 "Busy Here" back from' and when I call outside to 1337 then 1338 rings, and when I call 1338, 1338 rings again. |
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| 09:02.31 | r00f | wrong exten routing? |
| 09:02.35 | Ice_Strike | What software do you use for provisioning polycom phones? |
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| 09:07.15 | mushroomed | r00f: No, I just --> database put users 1337 1337 and same with 1338. Then in my dialplan I have something like --> exten=>_XXXX,1,Set(ext=${DB(users/${EXTEN})}) ...and... same=>n,Dial(SIP/${ext},40,m) |
| 09:08.53 | r00f | then maybe both phones are configured with the same account? |
| 09:09.19 | r00f | withour seeing log it is hard to say where did the call land and why |
| 09:09.37 | mushroomed | r00f: Nope~ I have already checked at each Phone's Web Administrator and everything looks ok |
| 09:09.45 | mushroomed | r00f: How can I do to get those logs? |
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| 09:14.12 | r00f | at the cli during the call, or in any log configured |
| 09:15.12 | r00f | afk |
| 09:19.08 | mushroomed | r00f: It's incredible I see (at the cli) that SIP/1338 is being called but the one who rings is 1337 |
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| 09:36.43 | r00f | mushroomed: if phones are behind NAT, try setting them to different outgoing SIP ports |
| 09:36.53 | r00f | like, one sends from 5060 and second from 5061 |
| 09:37.08 | r00f | i've seen similar problem with linksys sip2fxs boxes |
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| 10:55.10 | neeby_goosey | Hello. |
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| 10:58.29 | neeby_goosey | How can an option, set in sip.conf general section, be cancelled in other sections? Will "option=" do? |
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| 11:11.54 | neeby_goosey | And, by the way, can anybody explain me why "externaddr" option, set to my NATed router's public ip in sip.conf, completely breaks incoming calls from VoIP provider, although outcoming work? |
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| 11:13.39 | neeby_goosey | I don't get it, argh. |
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| 11:16.17 | MaliutaLap | ~sipnat |
| 11:16.17 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 11:19.54 | neeby_goosey | MaliutaLap, the first link is broken, the second is ancient. |
| 11:20.15 | neeby_goosey | My version is 11.8.1. |
| 11:24.39 | neeby_goosey | In case anyone interested, the sip.conf is http://pastebin.com/7R41xZZG |
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| 11:26.27 | MaliutaLap | neeby_goosey: AFAIK the settings are still relevant |
| 11:26.33 | Chainsaw | neeby_goosey: Did you not want to upgrade to 11.9.0? The bug fixes look significant. |
| 11:26.38 | neeby_goosey | MaliutaLap, I know about nat, directmedia, externhost, externaddr, localnet options, I am confused that "externaddr" works not as intended in my case. Maybe I am missing something. |
| 11:27.30 | neeby_goosey | Chainsaw, oh, thank you for pointing out, gotta update at night. |
| 11:27.53 | Chainsaw | neeby_goosey: And just to confirm, I am actually running that myself. I can vouch for it. |
| 11:30.00 | neeby_goosey | Chainsaw, does "make install" mess up with my current configs in /etc/asterisk? |
| 11:30.29 | Chainsaw | neeby_goosey: It doesn't for me, there's a "samples" setting for that isn't there? |
| 11:30.45 | Chainsaw | neeby_goosey: (Unless that's something entirely Gentoo-specific, but it seems too common sense) |
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| 12:28.56 | RadJackson | Hi , i have a context that answer customers calls , and then dial a couple of SIP accounts. When calling the agents i have to display a callerid(name) but when the customer calls with an anonym number , i cannot change the callerid params it displays anonymous.invalid. Any idea how to change callerid name and num when the incoming call is anonymous ? |
| 12:29.44 | neeby_goosey | Oh, a very important question: why does the Dial command loops when it gets 503 response and how do I stop it and immideately respond to caller 'Busy' instead of ringing? |
| 12:36.58 | WIMPy | RadJackson: Set the Number to anything other than nothing. |
| 12:38.10 | RadJackson | Actually i modified the callername and number both to Set(CALLERID(all)="MyCallerName" <11111111111>) |
| 12:38.51 | RadJackson | still displaying anonymous.invalid when the caller is anonymous |
| 12:39.14 | WIMPy | You need to set the (pres) parameter. |
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| 12:41.04 | r00f | exten => _5XXX,1,SetCallerPres(allowed) |
| 12:41.16 | r00f | before assigning clid |
| 12:41.21 | r00f | this works for me |
| 12:41.26 | RadJackson | 1s |
| 12:41.50 | r00f | however that is not 11 aster. maybe it has changed since |
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| 12:44.14 | RadJackson | r00f you are a god |
| 12:44.15 | RadJackson | Thanks ! |
| 12:44.34 | r00f | why me |
| 12:44.51 | r00f | wimpy suggested that |
| 12:44.51 | r00f | i just shared a config |
| 12:45.39 | CMGauger | I have a question on setting up Asterisk's indication sounds, anyone here know how to do that? |
| 12:46.34 | neeby_goosey | This is configured in /etc/asterisk/indications.conf |
| 12:47.08 | CMGauger | Well, yes I know that. But I have a question specifically on how to configure a specific kind of tone. |
| 12:47.38 | neeby_goosey | Oh. Could you share the question? |
| 12:48.32 | CMGauger | Well, I'm trying to configure the indications for the old style Bell System tones. I.e. pre-Precise Tone Plan. |
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| 12:50.47 | CMGauger | Namely, I'm trying to figure out how to configure "Low Tone" (the tone used as the dial tone, reorder, paths busy, and no circuit indicators to name a few), and the audible ringback. |
| 12:51.11 | CMGauger | They're 600 Hz modulated by 120 Hz, and 420 Hz modulated by 40 Hz, respectively. |
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| 12:56.25 | r00f | whoa what was that |
| 13:08.11 | neeby_goosey | So any idea why the Dial command loops when it gets 503 response and how to I stop it and immideately respond to caller 'Busy' instead of ringing? |
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| 13:19.14 | [TK]D-Fender | neeby_goosey: show us the call |
| 13:22.03 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
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| 13:32.20 | neeby_goosey | I will have logs in an hour. |
| 13:32.31 | neeby_goosey | Will post ASAP. |
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| 13:43.24 | kannan | does MixMonitor have any default duration (asterisk 1.8.23.0) for monitoring on a channel? |
| 13:43.49 | file | no |
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| 13:44.49 | As001 | hello I have asterisk 1.8 on public static ip and various snom phones behind router on other location. I noticed in sip set debug on Register packet, but Asterisk answered 401 Unauthorized 2x then some Option packet comes and 200 OK. Everything works but I don't know why those 401 Unauthorized messages show up. Here is pastebin http://pastebin.com/rdpbpHBA. |
| 13:45.34 | Katty | fumes. |
| 13:45.45 | As001 | I guess some nat issue. I have nat=yes in peer configuration for snom phones. |
| 13:46.54 | r00f | As001 by default asterisk does not authenticate OPTIONS requests |
| 13:47.16 | r00f | so it will reply 200 ok even if user failed to auth previously |
| 13:47.19 | *** join/#asterisk kannan (~chatzilla@123.238.235.11) |
| 13:47.51 | kannan | sorry got disconn , i had asked whether does MixMonitor have any default duration (asterisk 1.8.23.0) for monitoring on a channel? |
| 13:47.52 | As001 | ok but why 401 message ? username/password is correct sip show peers shows OK phone work. |
| 13:50.49 | r00f | not sure about that. will have to wait for someone smarter |
| 13:51.19 | r00f | kannan: [17:43] <@file> no |
| 13:51.33 | kannan | r00f: thank you |
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| 14:08.03 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:12.56 | neeby_goosey | [TK]D-Fender, ok, it seems that the problem disappeared after updating from 11.8.1 to 11.9.0. |
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| 14:13.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:14.39 | *** part/#asterisk As001 (~uros@82.117.198.142) |
| 14:15.23 | neeby_goosey | A little question about srtp now: my Blink sipphone is by default trying to use srtp, but asterisk kills the call with warning "process_sdp: We are requesting SRTP for audio, but they responded without it!" Is there a way to force asterisk into never using srtp? |
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| 14:17.28 | *** join/#asterisk toyotapie (~david@207.134.195.126) |
| 14:18.12 | toyotapie | Hello, I am using asterisk manager API to send a fax ( call originate, connect to fax send extension ). Trouble is one of the destinations does fax detection on early audio, so everytime I send the fax a human answers. Is there a way to send the 1000hz tone in early audio when originating a call ^ |
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| 14:25.47 | neeby_goosey | Ignore my last question, my problem turns out to be wellknown, googlable and solvable. |
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| 14:31.22 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 15:02.29 | RadJackson | Hi when a customer hang up , i catch the action on "h" extension and call an AGI script to do some magic stuffs. Sometimes the AGI is called but doesnt run. when checking cdr log (Master.csv) , i see that the call ended up at AGI command call , which means that the script was called by the AGI command , but it didnt run (it hasnt done what it was supposed to do), i've tried DeadAGI and AGI |
| 15:02.29 | RadJackson | , same result. And this happens only few times like 1/1000 calls got this problem. |
| 15:02.31 | RadJackson | Any idea? |
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| 15:15.27 | seme | hi guys... does anyone know if you can get asterisk on a SPA9000 or SPA400... I was thinking of buying one and just wanted to see if anyone else has already gotten it to work |
| 15:16.06 | newtonr | RadJackson, no idea. you might try to catch it in a full log with the debug and verbose log channels on to see if anything funny pops up. |
| 15:16.41 | newtonr | seme, "get asterisk on"? Do you mean connect those phones to an Asterisk system? |
| 15:17.09 | seme | no... I mean install asterisk on one |
| 15:17.15 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 15:17.20 | seme | I want to play with it and see if I can make it an asterisk server for my home |
| 15:17.23 | newtonr | oh, i made the assumption that they were phones, /me googles |
| 15:17.45 | seme | it has 4 fxo ports 1 ethernet port and it runs some cisco/linksys software |
| 15:17.58 | seme | pretty cheap and looks like it could be a good asterisk platform on the cheap |
| 15:18.00 | newtonr | ah, i see. I don't know. |
| 15:18.48 | navaismo | seme, i dont think you can install Linux on those devices they are atas but you can connect it to an asterisk box |
| 15:19.14 | seme | any suggestion for a cheap asterisk server platform that supports both pstn and sip? |
| 15:19.21 | seme | I was hoping to hack on something |
| 15:19.29 | seme | don't want to spend a lot either |
| 15:20.04 | navaismo | buy an ALIX board and some miniPCI FXO cards build your box |
| 15:20.27 | navaismo | is cheaper to buy a Raspberry and SIP ATA/GATEWAY |
| 15:20.40 | seme | that will cost 200+... I can get the SPA9000 and SPA400 for about 175 |
| 15:20.46 | seme | interesting |
| 15:21.10 | navaismo | yes but you need an exteneral asterisk box, and if you insist to hack then let us know how many hours do you spend on it |
| 15:21.11 | seme | I saw the raspberry pi stuff... .looks cool |
| 15:21.21 | seme | he hee |
| 15:21.35 | seme | well this is for fun... so I don't mind playing around a bit |
| 15:22.01 | seme | but perhaps your idea of the rspberry pi + gateway is good... |
| 15:22.08 | seme | let me see what the gateway would cost |
| 15:22.41 | Katty | $5 |
| 15:24.27 | *** join/#asterisk seme_ (~Seme@pool-108-35-132-65.nwrknj.fios.verizon.net) |
| 15:24.50 | coppice | seme: the SPA9000 is obsolete, so doing a lot of work to get something running on it is a dead end |
| 15:25.17 | Katty | unless you want to. and you enjoy it. |
| 15:25.20 | Katty | then you should totally doit |
| 15:26.05 | coppice | no you shouldn't, when the same type and quantity of effort could be put into something more productive |
| 15:29.17 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 15:30.28 | RadJackson | newtonr thank you |
| 15:30.58 | *** join/#asterisk theron (~theron@66.220.144.81) |
| 15:33.26 | RadJackson | One more question, i have developed an audiotel application (quizz) customers call and start playing by typing digits. I use WaitExten/Background commands to expect the customer's type action. Sometimes a customer calls and start playing but then my server becomes unable to detect dtmf tones, for exemple i launch a WaitExten and i expect him to type 1 or 2 , i try to catch that but it just |
| 15:33.27 | RadJackson | doesnt work ; he keeps typing , no result. what could be the problem? my application? it works 90% of times, the customer's phone? he started playing and it worked like charm. Any idea please? |
| 15:41.35 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:42.32 | *** join/#asterisk seme (~jrizzo@pool-108-35-132-65.nwrknj.fios.verizon.net) |
| 15:42.38 | seme | sorry guys... lost my internet :( |
| 15:42.43 | *** join/#asterisk roentgen (~irc@openvpn/community/support/roentgen) |
| 15:42.53 | seme | I was able to scrounge up an ATA adapter :) |
| 15:43.23 | seme | so now I am waiting for the cloudtc glass 1000 and I have to pick a pbx platform... :) |
| 15:52.31 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-tpplprupdgxeoefe) |
| 15:58.23 | neeby_goosey | A strange problem just turned up: (phone -> ATA -> Asterisk -> NAT -> VoIP -> cellphone)-call is successful and both ends have audio, but in case of (cellphone -> VoIP -> NAT -> Asterisk -> NAT -> VoIP -> cellphone) - neither end has audio. Info: http://pastebin.com/t5jx5Y9W |
| 15:59.03 | neeby_goosey | seems like there is no RTP flow between both ends |
| 16:00.12 | neeby_goosey | Probably should mess up my sip.conf again, but I am not sure what to change, because in different cases everything works. |
| 16:00.59 | neeby_goosey | Forgot to mention I have a public static IP. |
| 16:02.57 | *** join/#asterisk evilman_home (~evilman_h@89-179-77-66.broadband.corbina.ru) |
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| 16:05.19 | scouture | neeby_goosey: just as a test, did you try to answer() your channel before you dial back to the pstn? |
| 16:05.58 | neeby_goosey | No, why? Dial answers automatically, doesn't it? >_< |
| 16:06.24 | lvlinux | neeby_goosey: nope |
| 16:06.47 | lvlinux | neeby_goosey: not necessarily |
| 16:06.52 | neeby_goosey | What a terrible surprise.// |
| 16:07.03 | neeby_goosey | Ok, gotta go check... |
| 16:07.43 | *** join/#asterisk evilman_home (~evilman_h@89-179-77-66.broadband.corbina.ru) |
| 16:08.48 | neeby_goosey | Adding Answer() before dialing out doesn't help. |
| 16:10.04 | neeby_goosey | The cli difference is -- "Executing [s@inbound:1] Answer("SIP/VoIP-00000144", "") in new stack" line. |
| 16:10.19 | scouture | neeby_goosey: this may seem silly, but what if you have a Answer(), then a Playback(silence/1), then your Dial()..? i've had a similar issue with an incoming call coming in from a provider, and going out the same provider with no audio, and this is how we've worked around the problem. |
| 16:10.35 | neeby_goosey | Checking... |
| 16:11.20 | neeby_goosey | HOW IN THE NAME OF... |
| 16:11.23 | scouture | right. |
| 16:11.27 | neeby_goosey | scouture, HOW? |
| 16:11.33 | scouture | haha. |
| 16:12.02 | neeby_goosey | What is the principle behind this? |
| 16:12.24 | scouture | neeby_goosey: i don't think the issue is on your side, really.. i'm close to certain my issue in on our provider's side, having a hard time dealing with what appears the same call twice. |
| 16:12.31 | scouture | but that's how we've worked around it. |
| 16:12.48 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:12.59 | neeby_goosey | Thank you kindly! |
| 16:13.07 | scouture | sure thing. |
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| 17:02.02 | *** join/#asterisk jkroon (~jkroon@41.13.120.97) |
| 17:02.29 | jkroon | hi all, i think I may have found a possible issue with respect to chan_sip that might very well be security impacting. |
| 17:03.14 | jkroon | what's the best way to discuss this, and with whom. relatively serious, not directly impacting but I still think it's something that needs to be handled in a sensitive manner. |
| 17:03.37 | jkroon | (ie, I don't think it's a good idea to explain what the issue is in this channel) |
| 17:05.31 | navaismo | open a bug in the JIRA page |
| 17:05.42 | navaismo | but then again it will be public |
| 17:05.55 | WIMPy | It could be a private one, can't it? |
| 17:06.03 | jkroon | navaismo, if I publicly release this info you're probably going to see a flurry of hack attempts ... |
| 17:06.12 | WIMPy | But the question might be better off in #asterisk-dev. |
| 17:06.18 | file | security@asterisk.org |
| 17:06.26 | jkroon | let me mail that |
| 17:06.37 | rmudgett | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Vulnerabilities |
| 17:07.21 | navaismo | but but i want to know it, i want to hack all pbx in the world |
| 17:08.28 | *** join/#asterisk lorsungcu_ (~anonymous@209-173-236-30.usfamily.net) |
| 17:08.52 | jkroon | navaismo, it doesn't allow you to hack them directly ... fortunately. merely to go completely undetected. |
| 17:09.06 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 17:09.07 | jkroon | thanks for the pointers guys |
| 17:13.56 | onixx | hello, I am wondering is qualify=yes still recommended with asterisk 12 for phones behind nat ? |
| 17:14.36 | WIMPy | Has it ever been? |
| 17:15.02 | WIMPy | You can abuse it as a keep-alive, but a real keep-alive from the device behind NAT would make much more sense. |
| 17:16.09 | onixx | WIMPy: well, I've seen net=yes qualify=yes on many sample sip.conf... Bria defaults to 30 seconds as far as I can see in its configs |
| 17:17.30 | onixx | WIMPy: I have removed it and it seems to work ok still... |
| 17:18.01 | jameswf | does the te133 do hardware dtmf |
| 17:18.34 | *** join/#asterisk vlad_starkov (~vlad_star@212.34.52.66) |
| 17:18.38 | WIMPy | jameswf: Don't tehy all? |
| 17:19.02 | jameswf | has no idea... he is new to the digium stuff |
| 17:19.19 | WIMPy | But IIRC the software version is preferred. |
| 17:19.45 | lvlinux | why would the s/w b preferred? is it more accurate/reliable? |
| 17:21.07 | onixx | WIMPy: thanks for you response, when you say keep alive on the device, how many seconds would you recommend, I am trying to make the device as compatible as possible as it will be moving to different wifi networks |
| 17:22.29 | WIMPy | onixx: That really depends on the cofiguration of the router they're behind. But the good thing about reep-alive pacets is that they would (hopefully) be empty and doen't have to be processed by |
| 17:22.31 | WIMPy | Asterisk. |
| 17:23.39 | onixx | WIMPy: so really, qualify is a good way to force it to be very frequent knowing this phone is always going to be behind nat is probably not a bad thing to make yes |
| 17:24.53 | WIMPy | Qualify does eat resources from Asterisk. So if you have many phones and need high frequency keep alives, it might get ugly pretty fast. |
| 17:26.08 | onixx | WIMPy: thanks !! one more question if you have time. are you familiar with TLS. I am setup and it works well. I have set transport=tls and encryption=yes on that peer that I want to use it. it registers allright |
| 17:26.47 | *** join/#asterisk AlHafoudh (~textual@dsl-static-173.212-5-200.telecom.sk) |
| 17:26.53 | onixx | however, if I make udpbindaddr to be only on my LAN and tlsbindaddr=0.0.0.0, issues start ! phone on tls is not on my lan. |
| 17:27.32 | onixx | it apprears event if its set to TLS, if somewhat needs the udb port. |
| 17:29.19 | onixx | WIMPy: actually, the call completes and rings the other party but no audio at all |
| 17:29.53 | *** join/#asterisk anonymouz666 (~anonymouz@187.76.181.102) |
| 17:30.13 | jkroon | WIMPy, quality=yes doesn't seem to be that bad, at least, it's NOTHING in comparison with processing REGISTER requests. |
| 17:31.09 | WIMPy | Did you set tls on both sides? |
| 17:31.17 | WIMPy | You could have it in one direction only. |
| 17:31.48 | jkroon | is that even possible?! |
| 17:31.51 | onixx | WIMPy: device is set to TLS and encrypt audio to always |
| 17:32.12 | WIMPy | Ok, audio is still a completely different thing. |
| 17:32.28 | anonymouz666 | anyone know an IP phone that supports more than 2k contacts and the search is based in only the first contact name and directly accessible through a button? |
| 17:33.07 | onixx | WIMPy: so is udpbinaddr required for encrypted audio ? |
| 17:33.19 | WIMPy | anonymouz666: Do it on your server via mini-browser. |
| 17:33.41 | WIMPy | onixx: No, it's not related to audio in any way. |
| 17:33.55 | anonymouz666 | WIMPy: sorry I didn't understand what do you meant |
| 17:34.22 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 17:35.19 | onixx | WIMPy: what would you suggest to debug this issue ? I would like to bind udp only on my lan for security purpose |
| 17:35.38 | WIMPy | anonymouz666: Use a phone with some sort of browser. |
| 17:37.12 | WIMPy | Sorry, need to concentrate on a call. BBIAB |
| 17:39.39 | [TK]D-Fender | ~poll |
| 17:39.40 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 17:39.42 | [TK]D-Fender | ~polls |
| 17:39.42 | 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> |
| 17:42.08 | lvlinux | anonymouz666: the Yealink t46G phones have LDAP capability. |
| 17:43.14 | anonymouz666 | lvlinux: did you use this capability? |
| 17:43.59 | lvlinux | no not yet but it looks to be very nice. It will also take contacts from an XML list too I believe. But with LDAP you should be able to have thousands easily. |
| 17:44.08 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:44.59 | vlad_starkov | Question: What Asterisk stores in local database tree "SIP/Registry"? |
| 17:48.28 | onixx | WIMPy: if I do rtp debug, I only see Sent RTP packet to lines... |
| 17:48.52 | onixx | if I re-enable the udp port, I see Sent and Got |
| 17:53.53 | WIMPy | anonymouz666: Yes, LDAP is another option, but I have no idea what options phones would offer to define the search terms. |
| 17:54.14 | WIMPy | onixx: That doesn' make sense to me. |
| 17:54.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:54.29 | WIMPy | But I'm not the SIP expert in any way. |
| 17:54.50 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 18:02.39 | anonymouz666 | the search terms are very important just like the user experience accessing it easily... I have no issues at all with the most phones, but some directors.... |
| 18:09.41 | *** join/#asterisk solmsted (~solmsted@pool-71-176-247-26.rcmdva.fios.verizon.net) |
| 18:09.55 | lvlinux | anonymouz666: this might b useful: http://www.yealink.com/Upload/T4X/Using%20LDAP%20Phonebook%20on%20Yealink%20SIP-T46G%20IP%20Phones.pdf |
| 18:11.02 | Katty | hi cupcakes |
| 18:11.10 | Katty | how do i fix my asterisk today |
| 18:24.28 | leifmadsen | Katty: step one, get a strong drink |
| 18:24.37 | leifmadsen | step two, drink said strong drink, then grab another one |
| 18:24.46 | leifmadsen | what was I fixing again? oh right, another drink |
| 18:29.09 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 18:31.30 | onixx | WIMPy: thanks a lot for your help. I did some more testing and it is clearly related. when I set udpbindaddr to only my local lan, SRTP stops working... |
| 18:32.24 | onixx | I do not get any audio. If I re-enable udpbindaddr on all interfaced, it works again and I get audio |
| 18:33.38 | *** part/#asterisk rustlebee (~russellb@redhat/russellb) |
| 18:34.17 | onixx | Would anyone be able to confirm if SRTP needs asterisk to be configured to listed on udp ? |
| 18:34.24 | onixx | in sip.conf ? |
| 18:39.33 | vlad_starkov | By the way, thanks to someone who has unbanned me about a day ago (it seems I was banned mistakenly) :-) |
| 18:42.26 | [TK]D-Fender | vlad_starkov: was probably for your client quitting/rejoining at a high rate and flooding people. |
| 18:42.58 | [TK]D-Fender | vlad_starkov: It's not personal, but yu can always contact an op to have one like that lifted. |
| 18:42.59 | vlad_starkov | [TK]D-Fender: that was quite possible |
| 18:43.26 | onixx | Looks like my issue is unrelated to SRTP in the end ! Bria does not hear any audio if udpbindaddr is not reacheable when using TLS for SIP Transport |
| 18:43.58 | vlad_starkov | [TK]D-Fender: so I asked op from asteriks-dev to unban me :-)) |
| 18:44.46 | vlad_starkov | onixx: SRTP is not about udpbindaddr |
| 18:45.09 | Katty | leifmadsen: such innovation! |
| 18:45.12 | Katty | leifmadsen: i love it |
| 18:45.37 | vlad_starkov | onixx: Asterisk secures you telephony in two separate paths: signaling and voice |
| 18:45.55 | vlad_starkov | onixx: signaling is SIP, voice is RTP |
| 18:46.50 | onixx | vlad_starkov: thanks. currently, transport=tls and encryption=yes for that peer |
| 18:47.17 | vlad_starkov | onixx: SIP use one of 3 different transport level protocols: UDP, TCP or TLS (last one in turn uses TCP to create encrypted transport) |
| 18:47.41 | leifmadsen | Katty: I'm kind of a big deal |
| 18:47.44 | onixx | on bria, I have SIP Transport=TLS and encrypt audio=always |
| 18:47.54 | vlad_starkov | onixx: for you if you need to be most secured, you have to use TLS for signaling, and SRTP for voice |
| 18:47.58 | Katty | pats leifmadsen |
| 18:48.03 | Katty | leifmadsen: of course you are, dear. of course. |
| 18:48.09 | onixx | is it possible that udpbinaddr is essential for rtp audio to work ? |
| 18:48.11 | Katty | leifmadsen: just don't let it go to your head. |
| 18:48.13 | leifmadsen | \o/ |
| 18:48.17 | leifmadsen | Katty: I promise nothing |
| 18:49.24 | onixx | vlad_starkov: on my current setup, as soon as I make udp only listed on my local lan, bria stops getting audio when connecting from external network, public ip of behind nat does not matter ! no audio |
| 18:49.24 | vlad_starkov | onixx: next. You have to apply SSL certificate on Asterisk side, and (if you use self-signed free certificate) you have to add one to your iPhone/Android, so that you mobile phone/tablet will trust your self-signed cert |
| 18:49.45 | onixx | <PROTECTED> |
| 18:50.01 | vlad_starkov | onixx: if you have paid cert then just add one to your asterisk and dont add anything to your iphone |
| 18:50.16 | onixx | vlad_starkov: unpaid, added to phone |
| 18:50.40 | vlad_starkov | onixx: one more thing, in Bria in Advanced setting turn OFF param "Check TLS cert" |
| 18:51.07 | onixx | vlad_starkov: done that. |
| 18:51.08 | vlad_starkov | onixx: if that does not help, then we need to check your firewall settings |
| 18:51.41 | onixx | vlad_starkov: as I said, it seems to all work as I can establish calls and hear audio when * is listening on udp on all interfaces |
| 18:52.17 | vlad_starkov | onixx: Asterisk uses /etc/asterisk/rtp.conf file to config which port range will be used for transferring and receiving voice |
| 18:53.03 | vlad_starkov | onixx: I'd recommend to rtpstart=10000 and rtpend=20000, but defaults are rtpstart=5000 and rtpend=31000 |
| 18:53.31 | onixx | vlad_starkov: that seems all good. 10000 to 20000 |
| 18:53.58 | onixx | vlad_starkov: bria reports call media encrypted |
| 18:54.28 | vlad_starkov | onixx: ah, in Bria, go to your account settings, find Accound Advanced settings, then find Encrypt Audio and set it to "Best Effort" |
| 18:55.15 | onixx | vlad_starkov: I have set it to Always to only allow call if encrypted |
| 18:56.12 | onixx | and I believe it is encrypted as it works well as long as I leave udp listening on all interfaces |
| 18:56.50 | onixx | I am trying to only enable udp on my lan... but when I do this, audio does not work on all other interfaces... |
| 18:57.03 | onixx | regardless of encryption or not |
| 18:57.08 | vlad_starkov | onixx: for your information, setting encryption=yes in sip.conf makes it mandatory for your SIP client to use SRTP, but setting encryption=no in sip.conf makes it optional for your SIP client to use SRTP (it means, asterisk will use SRTP if your client ask to, and it won't use it case your client don't say anything about it) |
| 18:57.14 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:58.53 | *** join/#asterisk zerick (~eocrospom@190.114.248.34) |
| 18:59.06 | onixx | vlad_starkov: it appears to me that srtp or not, for audio to work on my call, I need to have udpbindaddr configured to listen on the interface I am registering thru |
| 18:59.15 | vlad_starkov | onixx: am I right understand that you can successfully call when you use UDP and you hear the callee and he/she can hear you? |
| 18:59.31 | onixx | correct |
| 19:00.10 | onixx | I was under the impression that I could only setup tlsbindaddr to listen if I had sip transport set to tls and audio encrypted |
| 19:00.55 | vlad_starkov | you need to properly enable TLS in sip.conf. tlsenable = yes; tlsbindaddr=0.0.0.0:5061; tlscertfile = /var/lib/asterisk/keys/yourkey.pem; tlscafile = /var/lib/asterisk/keys/yourca.crt |
| 19:01.17 | vlad_starkov | onixx: pay attention on port number for TLS |
| 19:01.35 | onixx | I don't have port currently : tlsbindaddr=0.0.0.0 |
| 19:01.36 | leifmadsen | what version of asterisk are you using? |
| 19:01.52 | vlad_starkov | leifmadsen: that should first question though)) |
| 19:02.00 | onixx | 12.2.0 |
| 19:02.19 | leifmadsen | oh ok, I did some research into tls on 1.8 and stopped because there were too many open bugs :) |
| 19:02.20 | vlad_starkov | onixx: try to explicitly set it to 5061 |
| 19:02.37 | leifmadsen | we just decided to make srtp and tls be done in the proxy since it seemed safer than the open bugs in asterisk around that code |
| 19:02.42 | *** part/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:03.05 | vlad_starkov | leifmadsen: by the way, in 11.5.1 I have many strange WARNINGs in console, like ... returned -2: No such file or directory and so on |
| 19:03.42 | onixx | vlad_starkov: I just did, testing now: udpbindaddr = 192.168.10.100, tlsbindaddr=0.0.0.0:5061, tlsenable=yes etc... |
| 19:03.49 | vlad_starkov | leifmadsen: what front-end proxy (or SBC?) do you recommend: Kamailio or OpenSIPS? :-) |
| 19:03.54 | onixx | connecting from outside of 192.168.10 |
| 19:03.54 | leifmadsen | doesn't matter |
| 19:04.04 | leifmadsen | both are fine |
| 19:04.35 | *** join/#asterisk jgornick (jgornick@2600:3c00::f03c:91ff:fedf:72b4) |
| 19:04.42 | vlad_starkov | leifmadsen: that is something I have to choose in the near future |
| 19:04.47 | Chainsaw | vlad_starkov: If you want to use TLS you need this patch: https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
| 19:04.57 | onixx | still no audio... |
| 19:05.00 | leifmadsen | we use opensips, in a previous company we used kamailio, I know lots of people who use both |
| 19:05.19 | onixx | if I simply change udpbindaddr to 0.0.0.0. Audio is back |
| 19:05.47 | vlad_starkov | Chainsaw: why my Safari ask for "client certificate" when I visit https://issues.asterisk.org/jira/browse/ASTERISK-18345 ? (I'm on Mac) |
| 19:06.34 | Chainsaw | vlad_starkov: I am not the hoster of issues.asterisk.org and thus cannot answer that. |
| 19:07.10 | vlad_starkov | onixx: I'd recommend you to use udpbindaddr=0.0.0.0:5060; tcpbindaddr=0.0.0.0:5060; tlsbindaddr=0.0.0.0:5061. And filter your permitted hosts in firewall |
| 19:07.49 | Chainsaw | vlad_starkov: But I can tell you that it works fine in Chromium on Linux. |
| 19:08.01 | onixx | vlad_starkov: good ideal, I will block 5060 from outside to see what happens |
| 19:08.28 | vlad_starkov | onixx: As my opinion Asterisk is not that good when we talk about IP/interface binding (for example it can't listen the same transport protocol on two different interfaces, only all or one) |
| 19:08.44 | *** join/#asterisk TMW2N (5H8S6S7LV1@tengu.elite.uk.com) |
| 19:08.49 | TMW2N | hi folks |
| 19:08.51 | Chainsaw | vlad_starkov: At any rate, TLS as implemented is broken and without that patch, you're going to have a horrible time. |
| 19:08.58 | Chainsaw | Hello TMW2N. |
| 19:09.40 | vlad_starkov | Chainsaw: I'll use OpenSIPS in the near future. Will install it as SBC. |
| 19:10.20 | onixx | vlad_starkov: there is definately something FUNNY there.... just did 0.0.0.0:5060 for udp and blocked it if its from ppp0 using iptables and I have audio now :-) |
| 19:10.24 | *** join/#asterisk fahmad (~fahmad@unaffiliated/fahmad) |
| 19:10.36 | TMW2N | sorry to ask what is probalby very much a newbie Question, but my head is like a sponge at the minute. I'm using asterisk as my home phone, and my dad's been very unwell. he's got an alert system going in, and i've been asked if i can stop calls from him going to voicemail so the dialler will call the next person. can i set a rle based on caller ID to never send calls to voicemail? |
| 19:10.50 | TMW2N | a rule* |
| 19:11.02 | TMW2N | my google fu is failing me entirely tonight |
| 19:11.36 | WIMPy | TMW2N: See the / in extension pattern matching. Just add the caller ID. |
| 19:11.59 | vlad_starkov | leifmadsen: take this opportunity to thank you and your co-authors for all the Asterisk book. I have read 2nd, 3rd and 4th editions, and Cookbook also. Everything is written and explained so clear and comprehensive enough! |
| 19:12.08 | TMW2N | WIMPy: thanks |
| 19:12.25 | vlad_starkov | onixx: so, you're good!) |
| 19:12.32 | lvlinux | WIMPy: are u one of the co-authors? |
| 19:12.47 | leifmadsen | vlad_starkov: glad you like it! |
| 19:12.50 | WIMPy | lvlinux: No |
| 19:13.04 | onixx | vlad_starkov: there's definately something up in udpbindaddr !!! ;-) |
| 19:13.08 | vlad_starkov | leifmadsen: do you plan to release new book in the near future? |
| 19:13.08 | lvlinux | oh well regardless i second what vlad_starkov said |
| 19:13.09 | onixx | Thanks for all the help !!! |
| 19:13.19 | vlad_starkov | urwelcome |
| 19:13.41 | WIMPy | I just wrote the Asterisk vs ISDN rant. Which I found out might actually be a little too positive, still. |
| 19:14.00 | leifmadsen | vlad_starkov: nope |
| 19:14.16 | leifmadsen | vlad_starkov: it's too much work, and we don't have really that much time, and it doesn't pay all that well at all |
| 19:14.27 | leifmadsen | it was fun while we did it, but at this point, I have no interest in writing another book |
| 19:15.05 | lvlinux | leifmadsen: why write another---the ones out pretty much cover it! |
| 19:15.28 | leifmadsen | except for the ARI stuff :) |
| 19:15.41 | WIMPy | And pjsip |
| 19:15.43 | leifmadsen | a book on just ARI would be useful |
| 19:15.47 | leifmadsen | and the updates for pjsip, right |
| 19:15.53 | WIMPy | And lots of old stuff. |
| 19:15.53 | leifmadsen | that is a different beast for sure |
| 19:16.11 | leifmadsen | you can't document everything for asterisk is basically what it comes down to |
| 19:16.22 | leifmadsen | sometimes you just gotta learn to do stuff yourself ;) |
| 19:16.28 | lvlinux | yep :-) |
| 19:16.49 | vlad_starkov | leifmadsen: what about Asterisk SCF project? |
| 19:16.50 | lvlinux | with help from the great guys in here :-D |
| 19:16.52 | WIMPy | And make assumptions that may or may not hold true (in the future). |
| 19:17.09 | leifmadsen | vlad_starkov: that project is no longer active |
| 19:17.23 | lvlinux | (and gals) |
| 19:20.00 | vlad_starkov | leifmadsen: as for advance orders: would be nice to have comprehensive explanation on Asterisk scaling using SBC like OpenSIPS or so (but not sure that this is in general context of Asterisk books series) |
| 19:20.29 | leifmadsen | that would be a neat book, I agree |
| 19:21.31 | vlad_starkov | leifmadsen: may be it worth to crowdfund on Kickstarter? I'd be in :-) |
| 19:21.58 | leifmadsen | it would be several thousand dollars |
| 19:22.24 | leifmadsen | I have a 1 year old and a full time job (no longer consulting) so the time would be hard to come by, even if I wanted to write another book |
| 19:24.59 | vlad_starkov | leifmadsen: I see. I'm just saying that as I like how you write (as it's quite a rarity when someone explains complex subject in a simple language) |
| 19:25.10 | leifmadsen | well I appreciate that :) |
| 19:25.13 | leifmadsen | I just need to blog more |
| 19:25.20 | leifmadsen | just haven't had the time |
| 19:25.45 | vlad_starkov | leifmadsen: hope you'll get some free time in the future and start a new book :-) |
| 19:25.53 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 19:25.57 | leifmadsen | I can tell you right now that isn't happening :) |
| 19:26.04 | vlad_starkov | )) |
| 19:26.11 | leifmadsen | I have no interest in writing another book any time soon |
| 19:26.16 | leifmadsen | it's a lot of fucking work ;) |
| 19:26.28 | leifmadsen | 8 months, working from 7am to 9am every day |
| 19:26.32 | leifmadsen | then I start my actual job |
| 19:27.39 | vlad_starkov | leifmadsen: then blog more, especially on Asterisk scaling subject, this becomes actual nowadays |
| 19:28.08 | *** join/#asterisk RaMcHiP (~RaMcHiP@wsip-70-184-107-198.ph.ph.cox.net) |
| 19:30.49 | vlad_starkov | leifmadsen: by the way, could you recommend something to read about voice billing systems mechanisms? We're thinking to write our own on Ruby as current available are not very good |
| 19:31.01 | leifmadsen | not really |
| 19:31.13 | leifmadsen | it's usually a self built mechanism based on many personal variables |
| 19:31.50 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
| 19:33.41 | vlad_starkov | leifmadsen: yep. I have an idea that it should be distributed module system, based on AAA concept |
| 19:35.09 | vlad_starkov | leifmadsen: so, it was nice to talk to you. Good luck! :-) |
| 19:42.32 | TMW2N | WIMPy: thanks for the pointer. got it working exactly how i need :) |
| 20:06.33 | *** join/#asterisk Dovid (~Dovid@80.178.160.185.adsl.012.net.il) |
| 20:11.35 | gusto | i started to like this DirectMedia ACL's ... since when its been around? i first noticed it in asterisk12, but then i even seen it in asterisk1.8 |
| 20:11.52 | gusto | so that seems to be somewhat older feature that was being unnoticed by me |
| 20:13.43 | gusto | WIMPy: hi |
| 20:13.46 | gusto | WIMPy: how are you? |
| 20:14.47 | WIMPy | ok |
| 20:15.27 | gusto | i have now 7 days left to move from dyndns away, because they now only want to have customers that pay :-D |
| 20:18.10 | gusto | so i will be reachable under no-ip.info |
| 20:18.43 | gusto | but ... i am also thinking about setting up my own dyndns service, only for me ... i ve seen that there are some packages that do it |
| 20:19.11 | WIMPy | Twitter is at the end of the corridor, left. |
| 20:19.19 | gusto | lol |
| 20:19.38 | gusto | are you on twitter? |
| 20:19.45 | WIMPy | no |
| 20:19.50 | gusto | you see |
| 20:20.06 | WIMPy | What do I see? |
| 20:20.14 | gusto | nothing |
| 20:20.30 | WIMPy | Great |
| 20:25.35 | lvlinux | gusto: get an account at he.net and they give you free dns---including dynamic dns for as many hosts as you want---no "renewal" junk like dyndns either. |
| 20:26.19 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 20:26.30 | WIMPy | There a loads of free ddns providers. |
| 20:30.24 | [TK]D-Fender | Alrighty, checkout time here. BBL |
| 20:32.49 | gusto | yes |
| 20:33.19 | gusto | i have a henet account |
| 20:33.25 | gusto | but only for my 6in4 |
| 20:35.16 | gusto | i am not using it now, because i am on natively, but i have a dsl connection is slovakia for example, that one needs a 6in4, because they still do not offer native ipv6 |
| 20:36.10 | gusto | maybe sometime my voip provider will also offer ipv6 for his services, then i will not need NAT and ddns at all |
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| 20:54.16 | lvlinux | gusto: yes you should sign up for the dns---no ads, no junk, professional interface, flexibility, they don't send you emails or bug you, it's very nice :-) |
| 20:57.36 | mushroomed | Hi, let's suppose a client has unregistered and I call the IVR and dial his extension. If I do that now, it just Hangup() and --> app_dial.c:2341 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 20:58.45 | mushroomed | It's because that extension is unregistered right now. Is there some way to check if called extension is unregistered and forward calls to the operator? |
| 21:01.17 | gusto | lvlinux: i just moved to this no-ip.com and that looks good as well |
| 21:01.45 | gusto | maybe i ll join that HE ddns service as backup ddns later as well |
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| 22:21.50 | CrashSys | I'm trying to figure out why asterisk is hammering a carrier with it's 200 OK response: http://pastebin.mozilla.org/5023092 |
| 22:22.01 | CrashSys | I'm guessing some of the headers send from them are causing issues |
| 22:22.03 | CrashSys | I'm not sure |
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| 23:14.20 | jameswf | how does dial handle special characters.... dial(sip/snap!) |
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