| 00:03.19 | SeRi | man I wonder when this CC bs is going to end. |
| 00:03.43 | SeRi | 1530 post at dslr about it |
| 00:03.50 | drmessano | I started off with Trixbox. I knew 0 about Linux when I started, but it was clear that the cool kids were doing the cool telephony stuff on *nix |
| 00:05.09 | jpsharp | Trying to slap a GUI around Asterisk really pigeonholes you into whatever the GUI developers had in mind. |
| 00:05.30 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 00:06.10 | WIMPy | But to be fair, you get some features you otherwise have to add manually. |
| 00:06.17 | tonikasch | WIMPy, I managed to, I have changed host to the one I want in To headers and I have added that host with the ip of the other to /etc/hosts |
| 00:06.19 | tonikasch | :P |
| 00:07.25 | WIMPy | tonikasch: Ok, that's a way, but if the IP pointed to by that name changes, you have to manually do it again to regain connectivity. |
| 00:07.34 | apb1963 | Easy Install? (Complete CD, with Linux and FreePBX) Notice that it doesn't say anything about Windoze |
| 00:07.59 | tonikasch | WIMPy, yes.... it's a risk |
| 00:08.27 | apb1963 | Same for asteriskNow |
| 00:08.38 | WIMPy | If you want Asterisk, forget about Windows. |
| 00:08.38 | apb1963 | Linux only... no windoze |
| 00:08.56 | apb1963 | you planning on buying me a new machine wimpy? Thanks! |
| 00:09.04 | apb1963 | very kind of you |
| 00:09.11 | tonikasch | WIMPy, Do you know of other way? |
| 00:09.42 | WIMPy | apb1963: That's just the way it is. If you want Windows, use somehing that runs under Windows. |
| 00:09.54 | apb1963 | you mean like asteriskWin32? |
| 00:10.02 | apb1963 | Hmmm.... sounds familiar |
| 00:10.10 | apb1963 | deja vu |
| 00:10.21 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 00:10.24 | apb1963 | and so the circle of life continues |
| 00:10.45 | jpsharp | Run Linux under a VM under windows. |
| 00:11.01 | jpsharp | Asterisk natively on Windows? Man, you're masochistic. |
| 00:11.03 | WIMPy | No. Something that runs under Windows, Not somethig that runs under *nix and someone tried to squeeze in to Windows. |
| 00:11.17 | apb1963 | OR, I could use VMware and run linux under windoze... sort of like I'm doing now with the attempt to install elastix |
| 00:11.27 | apb1963 | I am the lion king |
| 00:12.06 | SeRi | apb1963: 3CX |
| 00:12.08 | apb1963 | hear me roar. meow |
| 00:12.33 | SeRi | 3CX runs on your nice windows world |
| 00:12.44 | apb1963 | not my world... just what I'm stuck with for now |
| 00:12.55 | apb1963 | 3cx costs big wampum |
| 00:13.03 | SeRi | wrong |
| 00:13.10 | SeRi | yet again |
| 00:13.18 | SeRi | 3CX is free with limited options |
| 00:13.24 | WIMPy | just took a look at asteriskwin32. Looks like it hasn;t been updated for at least 6 years. That is definitely not an option of any kind. |
| 00:14.43 | apb1963 | 3CX Mini Edition 4 Simultaneous Calls SKU: 3CXPSMINI 0 Review(s) | Add Your Review Availability: In stock $495.00 |
| 00:14.59 | apb1963 | Maybe you were referring to the free shipping? |
| 00:16.09 | SeRi | Sorry man but you are just plain tard. |
| 00:16.12 | SeRi | http://www.3cx.com/ip-pbx/index.html |
| 00:17.01 | SeRi | 3CX Phone System FREE edition is a fully functional product. |
| 00:17.31 | SeRi | time to get ready for work. cya guys |
| 00:17.34 | SeRi | and good luck |
| 00:18.51 | apb1963 | ok I was looking at a resellers website... my mistake. thanks for the pointer! |
| 00:20.44 | drmessano | You can build a PBX on an inexpensive machine. You dont need a $5000 server |
| 00:21.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:21.45 | WIMPy | A Raspberry PI should do it nicely. |
| 00:21.47 | drmessano | Putting a PBX on an existing, loaded Windows server is fucking stupid. You either need a second box with an Asterisk install, or a dedicated box for Microsoft Lync. Anything else is sketchy |
| 00:22.10 | drmessano | A Pi would work, as would an inexpensive barebones ATOM box |
| 00:22.23 | *** join/#asterisk engrxyz (~retr@host81-150-217-167.in-addr.btopenworld.com) |
| 00:23.35 | drmessano | Just because it runs on Windows doesnt mean it can be loaded on an existing overloaded file server or the receptionists PC. That seems to escape a lot of people |
| 00:24.13 | drmessano | Although, the PBX running on the Receptionsts PC would be funny as hell. Could tell that one to the grandkids |
| 00:24.18 | apb1963 | receptionist? what receptionist? I'm the receptionist. I got me, and then I'll have 1, maybe 2 other people. |
| 00:25.01 | apb1963 | I'm just tryin' to kickstart things. |
| 00:25.07 | WIMPy | Then what do yo want an own server for at all? |
| 00:25.08 | drmessano | Great, so start off with an inexpensive box for the PBX, load AsteriskNOW on it, and read about 20 mins worth of docs for setting up FreePBX for basic calling. The rest you can learn in time |
| 00:25.39 | apb1963 | what's wrong with 3cx? |
| 00:25.49 | apb1963 | downloading it now |
| 00:25.50 | drmessano | WIMPy, good point. Problem solved. Get a NAS and repurpose your "server" for a PBX |
| 00:26.15 | drmessano | This is #asterisk. Not only do we not care that 3CX even exists, but Windows sucks and its not supported here either |
| 00:26.26 | WIMPy | Or just use the built-in telephony functions of your modem. |
| 00:26.28 | drmessano | Thats whats wrong with it |
| 00:26.28 | apb1963 | well thanks for the offer to buy all this "cheap" equiptment for me guys! You're all awesome! |
| 00:27.00 | drmessano | If you dont have a couple hundred bucks for a PBX then you dont need a PBX. Get some POTS phones and upgrade later |
| 00:27.21 | WIMPy | The hardware doesn't cost anything at all compared to the amout of work you will have to do. |
| 00:27.39 | drmessano | If you want someone to tell you that its ok to run a PBX off some Windows box you have sitting here, you obviously didnt read the channel description at all |
| 00:28.30 | drmessano | Seriously.. a USED $100 Dell box from ebay would work for a couple users |
| 00:28.54 | *** join/#asterisk micols (~t@rlogin.dk) |
| 00:29.46 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 00:29.48 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 00:29.54 | drmessano | But since you settled on 3CX, I am sure they can help you get it all going. Good luck |
| 00:31.11 | SeRi | drmessano: I agree with you all the way but some times you have to be a bit smarter. This guy has no clue on what is even going on right now. If jumps in to the asterisk worl it would be chaos. So there not all options work for all... So I send them back to their world.... |
| 00:31.20 | apb1963 | 3cx may no longer be an option: If using a VoIP provider or remote extensions, you will need a fixed IP. DynDNS and similar configurations are not supported! |
| 00:31.48 | SeRi | drmessano: ^^ Here is an example ^^ |
| 00:33.06 | apb1963 | Hmmm... used $100 Dell box or $150 for eye doctor... lets see.... to see.... or not to see... that is the question. |
| 00:33.19 | drmessano | I have no sympathy. I was a Windows guy, and I cut my teeth on Trixbox. I knew nothing and managed to do just fine. It's another piece of equipment.. doesn't matter that its a different OS than Windows. Your NAS, toaster, media center |
| 00:34.02 | WIMPy | apb1963: Why do you want to set up a voip server in the first place? |
| 00:34.12 | drmessano | apb1963, this argument is stupid. If you have to decide between an eye doctor and a PBX, you dont even need to be asking about a PBX. |
| 00:34.26 | SeRi | drmessano: I was just trying to be nice. I didnt want to say he is just plain clue less and I wanted to save my self or others of having to hold this guys hand all the way |
| 00:34.32 | apb1963 | thank you for your concern drmessano |
| 00:35.26 | SeRi | apb1963: You need to come out of the box a bit. Do some research. 3CX Will work for you. google. |
| 00:35.40 | apb1963 | If using a VoIP provider or remote extensions, you will need a fixed IP. DynDNS and similar configurations are not supported! |
| 00:36.15 | WIMPy | SeRi: How can you tell? He didn;t say what he needs/ wants or did I miss that part? |
| 00:36.15 | apb1963 | since i'm using a dynDNS clone, I Have to be concerned about the developer's statement. |
| 00:36.48 | drmessano | apb1963, if someone walked into your TV shop and wanted a $100 TV... and all you had were $200+ TV sets, and they kept going on and on telling you "Great! Are you going to give me the extra $100?" and "Great! Guess I can do that or get this tooth pulled!", how would you react? |
| 00:37.12 | SeRi | WIMPy: No you are right he didnt. |
| 00:37.46 | SeRi | BUT by the look of the situation he has no clue this is the year 2012. So.... |
| 00:37.49 | drmessano | Sorry, TV's cost $200+. In this case, you are overlimiting your opinions, which make it impossible to help determine a suitable, sensible outcome. If you cannot be flexible, then you have removed yourself from the marketplace |
| 00:38.22 | drmessano | Also, you are asking about a TV in a Donut shop. We dont do TVs |
| 00:38.37 | drmessano | ----> /topic |
| 00:38.44 | apb1963 | I don't have a tv shop and I don't eat donuts anymore. |
| 00:38.56 | drmessano | You also dont have a clue |
| 00:39.09 | apb1963 | Nope. That's why I'm here. For the clues. |
| 00:39.18 | SeRi | drmessano: Finally. LOL |
| 00:39.29 | drmessano | http://www.3cx.com/forums/ <---- |
| 00:39.44 | SeRi | apb1963: we are more than glad to help but we can do it allf or you and expect us to read your mind ether |
| 00:40.02 | SeRi | I still think your best bet is 3CX though |
| 00:40.02 | apb1963 | Thank you for clue #2 dr |
| 00:40.08 | drmessano | apb1963, You are in the WRONG PLACE. This is #ASTERISK. WE SUPPORT ASTERISK HERE. We offered you solutions involving ASTERISK. |
| 00:40.19 | WIMPy | apb1963: If you don;t know what you want, this visit is not going to serve any purpos other that pissing some people off. So you netter tell us what you want. |
| 00:40.28 | drmessano | You dont want ASTERISK. Why is this going on and on? |
| 00:40.35 | SeRi | LOL |
| 00:40.43 | SeRi | Now this just plain funny |
| 00:40.46 | SeRi | LOL |
| 00:40.53 | drmessano | GTFO |
| 00:41.32 | drmessano | I signed up for an account on the 3cx forums for you. Check your inbox for the Welcome email. TTYN |
| 00:43.57 | apb1963 | so... 3cx doesn't involve asterisk? |
| 00:44.05 | drmessano | No |
| 00:44.18 | drmessano | 3CX is NOT Asterisk AT ALL |
| 00:44.29 | apb1963 | well then that's not what I'm looking for |
| 00:44.42 | apb1963 | plus.... If using a VoIP provider or remote extensions, you will need a fixed IP. DynDNS and similar configurations are not supported! |
| 00:44.53 | drmessano | From http://www.asterisk.org |
| 00:44.55 | drmessano | An open source telephony switching and private branch exchange service for Linux. |
| 00:44.58 | drmessano | ^^^^ |
| 00:45.06 | drmessano | You dont want Linux. That should end that |
| 00:45.08 | apb1963 | Hmmm |
| 00:45.09 | WIMPy | Tell us what it is you want. |
| 00:45.31 | apb1963 | well, now that i have vmWare running... seems a shame not to take advantage of it. |
| 00:45.45 | drmessano | VMWARE Desktop VM is not suitable for a PBX |
| 00:46.31 | drmessano | It will not work. Desktop VMs do not give realtime priority to host OS'es, and you will have stutter in calls. |
| 00:46.36 | WIMPy | reboots |
| 00:46.59 | apb1963 | It mostly needs to simply route the call out to someone elsewhere. |
| 00:47.08 | drmessano | Doesnt matter |
| 00:47.25 | drmessano | The media will be handled by the running Asterisk instance. It will fail horribly |
| 00:48.06 | apb1963 | seriously? All the talky packets get routed through the pbx? that's kinda dumb. |
| 00:48.36 | apb1963 | I mean... doesn't point to point make more sense? And just send accounting packets to the pbx? |
| 00:48.51 | apb1963 | that's what I assumed it would do |
| 00:48.55 | WIMPy | What do you want a "PBX" for then? |
| 00:49.01 | apb1963 | to route calls |
| 00:49.02 | drmessano | Your setup wont support directmedia because you will be running it on a VM anyway |
| 00:49.12 | WIMPy | Not that I have any hope that we will ever get an answer on that one... |
| 00:49.16 | drmessano | You wont have direct network access. |
| 00:50.30 | drmessano | Asterisk, on a machine with a real virtualization platform, or on a dedicated machine with a Public IP CAN route media from endpoint to endpoint, however, your Desktop VM install will not do that |
| 00:50.42 | drmessano | So you're shooting yourself in the foot again |
| 00:50.46 | apb1963 | i c |
| 00:50.52 | WIMPy | Seriousely. If it's just for talking to 2 or 3 friens, just singn up for ccounts from the same ITSP and be happy. |
| 00:50.58 | apb1963 | ok appreciate the advice. thank you |
| 00:51.16 | drmessano | or get a POTS line. One should be enough for 2 people |
| 00:51.33 | WIMPy | If it's a BRI, yes :-) |
| 00:51.47 | apb1963 | I have 1 or 2 people in another part of the world. Those people need to answer my phone. It's that simple. |
| 00:51.52 | drmessano | You actually need something to EXCHANGE if you want a Public Branch Exchange. Doesnt sound like you have anything to switch |
| 00:51.57 | drmessano | OHHHHHH |
| 00:52.19 | WIMPy | Finally a bit of input. |
| 00:52.45 | apb1963 | I can do one with call forwarding from the VOIP provider. We fall short on the second person. |
| 00:52.45 | WIMPy | So the answer was indeed: Just singn up for accounts from the same ITSP and be happy. |
| 00:52.54 | drmessano | Yep |
| 00:52.59 | apb1963 | Plus... I like the call accounting features the pbx provides. |
| 00:53.33 | drmessano | Then you spend the $100 on a used Dell box or something and stop telling us about your glasses or toothache |
| 00:53.38 | drmessano | If you want a PBX, get a PBX |
| 00:54.00 | apb1963 | ok thanks |
| 00:54.10 | drmessano | Any idiot can set up AsteriskNOW, to be quite honest. A lot of work was put into it to ensure just that |
| 00:54.41 | apb1963 | my food is burning... back later |
| 00:54.49 | drmessano | Is that a threat? |
| 00:55.07 | WIMPy | makes that a yes |
| 00:55.13 | drmessano | lol |
| 00:55.20 | drmessano | ~windows |
| 00:55.21 | infobot | well, windows is either a 64bit hack on the 32 bit hack or just a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition... or the World of Warcraft bootloader, or the most important collection of bugs, or bloatware, or an operating system |
| 00:55.58 | WIMPy | Oh, nice one. Didn't see that, yet. |
| 00:55.59 | drmessano | WoW bootloader. FN Nice |
| 00:58.54 | tonikasch | WIMPy, sorry to recall.... you can make a script with a cron job to update your /etc/hosts :) |
| 01:00.07 | WIMPy | Yes, you can. |
| 01:00.23 | tonikasch | That's what I'm doing right now :) |
| 01:00.45 | SeRi | wow what a fucked up ride lol |
| 01:01.07 | tonikasch | :? |
| 01:02.26 | drmessano | SeRi, Pretty much. It started off as a rollercoaster, and then they guy in the car in front of us took his pants off and started eating a funnel cake, telling us how much he loves his bug collection and wishes everyone did |
| 01:02.44 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 01:03.28 | drmessano | SeRi, I wonder if he'll come back in an hour asking for help on using Trixbox on VMWare Fusion desktop on Windows 98. All his calls sound jittery |
| 01:04.56 | SeRi | drmessano: ROFL!!!!! |
| 01:05.08 | SeRi | I cant stop laughing. |
| 01:05.09 | SeRi | nice |
| 01:06.14 | *** join/#asterisk dgeary2 (~david@49.176.98.58) |
| 01:06.18 | drmessano | I try not to just be a damn ugly, nasty troll.. but when you realize that the patient has an untreatable case of stupid... |
| 01:06.48 | SeRi | lol |
| 01:07.03 | drmessano | I WANT ASTERISK RIGHT NOW AND I WANT IT TO RUN ON WINDOWS AND BE EASY TO SETUP IN 10 MINS OR LESS SO I CAN START A CALLCENTER |
| 01:07.24 | drmessano | ^^ Sorry, we have nothing for that. Call the next of kin |
| 01:07.49 | SeRi | lol! |
| 01:09.25 | drmessano | When hell has frozen over and WIMPy is actually suggesting AsteriskNOW to someone, and the person still insists on cutting even more corners.. what is there to do? |
| 01:10.10 | drmessano | (other than installing it for them and sending them a box for free) |
| 01:12.56 | SeRi | lol |
| 01:13.02 | SeRi | no shit |
| 01:20.56 | [TK]D-Fender | drmessano: You forgot the "AND DON'T GIVE ME ANY SHIT!" |
| 01:21.37 | SeRi | [TK]D-Fender: LOL! |
| 01:30.31 | drmessano | What what that trigger |
| 01:30.33 | drmessano | ~now |
| 01:30.33 | infobot | rumour has it, now is a good time to tell you that I have 6 gigabytes of data |
| 01:30.37 | drmessano | lol |
| 01:31.16 | [TK]D-Fender | ~asterisksign |
| 01:31.19 | [TK]D-Fender | ~asterisk-sign |
| 01:32.52 | [TK]D-Fender | Crap this was before Nov 2008.... can't find the orginal ref easy... |
| 01:45.18 | SeRi | ~itsp |
| 01:45.18 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 01:45.32 | SeRi | ~itsplist-us |
| 01:45.32 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 01:57.03 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
| 01:57.46 | tonikasch | Do you know what could be the reason for a "422 Session Interval Too Small" when dialing out through sip? My conf: http://pastebin.com/mUcj1irk |
| 01:58.30 | *** join/#asterisk droemel (~droemel@p4FCAD5DA.dip.t-dialin.net) |
| 01:59.34 | tonikasch | I don't get such errors with a softphone... :$ Although I have to use asterisk as I'm using it on my dsl router... |
| 02:06.02 | *** part/#asterisk cyberanger (~cyberange@swissknife/adak/infocop411) |
| 02:09.41 | tonikasch | mmmmm |
| 02:09.44 | tonikasch | min-se |
| 02:09.54 | tonikasch | ok, have to work it out... |
| 02:14.04 | tonikasch | yeah |
| 02:14.10 | tonikasch | and session-timers=originate |
| 02:14.11 | tonikasch | :D |
| 02:28.51 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 02:43.57 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 02:44.27 | *** join/#asterisk gdeeble (~gdeeble@184.170.111.173) |
| 02:46.39 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 02:53.32 | tonikasch | good night, bye |
| 03:41.48 | *** join/#asterisk radic (~radic@dslb-178-010-024-006.pools.arcor-ip.net) |
| 03:43.11 | phix | session-timers? |
| 03:43.13 | phix | what's that for? |
| 03:43.30 | *** join/#asterisk FireAndIce (~FireAndIc@175.100.131.213) |
| 03:49.38 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 04:11.16 | gdeeble | Curiousity, I'm still new to Asterisk, but noticed it has lightweight http built in. Can someone tell me where the root for the web server is and will it work to host xml files for cisco phones? I'm probably searching wrong on google, but not coming up with what I"m looking for. |
| 04:24.43 | *** join/#asterisk bitfury (~bitfury@unaffiliated/bitfury) |
| 04:25.34 | bitfury | hello, would a linksys PAP2-NA work in brazil with a DECT phone? |
| 04:25.35 | *** join/#asterisk dgeary2 (~david@42.241.82.120) |
| 04:42.38 | drmessano | gdeeble, http://www.voip-info.org/wiki/view/Asterisk+config+http.conf |
| 05:01.09 | gdeeble | drmessano, I've seen that but doesn't really tell me more than here's the config. but I will try to look again and see if I can make heads or tails. it does answer the config comment about the cisco files |
| 05:57.21 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 05:57.51 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
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| 06:47.32 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 06:51.22 | *** join/#asterisk x1user (~User@212.36.13.6) |
| 06:52.57 | x1user | I need to write up a failover dialplan. I got two GSM gates that gots overloaded and want to switch to SIP trunk when they are overloaded ? I am looking for any hints how to do this. |
| 06:54.40 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 06:55.25 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:59.34 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 07:06.30 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-dpxwxibdfylywrtf) |
| 07:11.09 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 07:18.43 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 07:23.08 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:26.40 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 07:26.48 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:28.25 | *** join/#asterisk chris-NB (~chris@fw.commpany.at) |
| 07:34.20 | x1user | How can I check in which part of the diaplan asterisk goes according to the number I´ve dialed? |
| 07:35.48 | *** join/#asterisk vlad_sta_ (~vlad_star@213.79.102.163) |
| 07:36.51 | kaldemar | x1user: enable verbosity in CLI and make a call. |
| 07:37.39 | x1user | i got verbosity and debug to maximum but i can really understand exactly what is going on with this asterisk |
| 07:37.55 | x1user | so i was looking if there is any better way to debug it |
| 07:38.26 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 07:39.49 | x1user | [root@MainPBX /etc/asterisk]# asterisk -rvvvvvvddddd |grep 08023432 |
| 07:39.49 | x1user | <PROTECTED> |
| 07:40.03 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 07:44.32 | *** join/#asterisk pingUone (~brads@li345-191.members.linode.com) |
| 07:45.15 | pingUone | where do I get motif? |
| 07:46.06 | pingUone | I ran emerge motif and all kinds of stuff installed but no motif.conf in /etc/asterisk/ ??? |
| 07:48.10 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 07:48.25 | AliRezaTaleghani | hi all |
| 07:48.31 | kaldemar | pingUone: are you looking for the module or the config file? |
| 07:48.58 | AliRezaTaleghani | i am intersted to know what is different betweek, "asterisk" and "certifeiedAsterisk" version? |
| 07:49.31 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 07:50.35 | kaldemar | AliRezaTaleghani: http://lmgtfy.com/?q=%22certified+asterisk%22 |
| 07:50.56 | AliRezaTaleghani | kaldemar: tnx |
| 07:54.26 | pingUone | kaldemar both |
| 07:54.49 | kaldemar | pingUone: what is your asterisk version and how did you install it? |
| 07:55.14 | pingUone | asterisk 10, via emerge |
| 07:58.15 | kaldemar | pingUone: you probably just installed http://motif.ics.com/ instead of chan_motif. anyway, chan_motif is in version 11 only unless someone backported it. |
| 07:58.39 | pingUone | my bad Asterisk 1.8.15.1, Copyright (C) 1999 - 2012 Digium, Inc. |
| 07:59.39 | kaldemar | pingUone: that means no chan_motif for you before you upgrade. |
| 08:01.15 | pingUone | I could of swore I had asterisk 10 or 11. and the main reason was it was the only thing that supported sms (which is configured on this system). Does 1.8 support sms over sip? |
| 08:03.21 | pingUone | exten => 19175039892,n,MessageSend(sip:<13474847018@sip.anveo.com:5010>,<0742510424>) |
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| 08:04.17 | pingUone | I think my machine downgraded on an update. |
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| 08:09.01 | kaldemar | that's really not SMS over SIP, just a SIP MESSAGE that gets transformed to an SMS. version 1.8 does not have the same messaging support. |
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| 08:17.17 | iprouteth0 | anyone use a bluetooth headset as an FXS device successfully? |
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| 08:48.16 | pingUone | I have asterisk 10 now, but still no motif.conf? |
| 08:50.16 | kaldemar | chan_motif is in version 11 only as i said earlier. |
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| 09:19.13 | iprouteth0 | chan_mobile now successfully working |
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| 11:55.57 | einstein123 | Hello, i've got a problem accessing a "clip-no-screening" provided number in the dialplan. How can i access the first of the two provided numbers on an incoming call? |
| 11:56.47 | einstein123 | i tried callerid(ani) and callerid ani2, but none of them lists the number i can see in pri-debug. |
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| 12:53.02 | hexanol | while looking at the code in chan_agent.c |
| 12:53.29 | hexanol | I've found this condition: if ((f->frametype != AST_FRAME_VOICE) || (f->frametype != AST_FRAME_VIDEO) ... |
| 12:53.40 | hexanol | won't this condition always be true ? |
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| 13:07.59 | bulkorok | hexanol: check #asterisk-dev |
| 13:10.46 | cusco | hi |
| 13:11.03 | cusco | We seem to be having a sip trouble connecting to a remote CISCO :( |
| 13:11.19 | cusco | remote party states that has tried some asterisk of their own and all works good |
| 13:11.49 | cusco | We are trying to perform outbound calls trough them, dialing a prefix right... |
| 13:12.22 | cusco | some times we get 100 Trying |
| 13:12.37 | cusco | they pick up and hear us, but we hear nothing, and are still Trying |
| 13:12.46 | cusco | some other times, we get 180 Ringing |
| 13:12.51 | cusco | we hear the ringtone |
| 13:12.56 | cusco | and can eve talk for 30 seconds |
| 13:13.00 | cusco | but the state is still ringing |
| 13:13.15 | cusco | here is the sip debug for this latest case: http://paste.debian.net/204663/ |
| 13:13.23 | cusco | I can't figure if the trouble is on our side |
| 13:14.56 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 13:15.29 | AliRezaTaleghani | when compiling the asterisk from it's source |
| 13:15.49 | AliRezaTaleghani | which file will store the configuration choise? |
| 13:16.07 | [TK]D-Fender | cusco, [2012-10-29 13:08:02] VERBOSE[23092] chan_sip.c: Reliably Transmitting (NAT) to 212.230.0.1:5060: <--- Odds are it deals with it's own situation and you should not have them as NAT |
| 13:16.10 | AliRezaTaleghani | i mean which file i should to make back up to |
| 13:16.38 | [TK]D-Fender | cusco, Also very off that you are sending compact headers. I've never seen * do that. The respond with standard headers so I'd probably reverse that first |
| 13:16.48 | cusco | [TK]D-Fender: we are binding to a public IP and connecting to their public IP, should not have nat?? |
| 13:17.15 | [TK]D-Fender | cusco, let them deal with their end and trust what they send you; |
| 13:17.21 | cusco | ow, I remember setting compactheaders a while back, can't remember why thaks let me look into that first then |
| 13:17.41 | [TK]D-Fender | [2012-10-29 13:08:05] WARNING[23092] dsp.c: Inband DTMF is not supported on codec g729. Use RFC2833 <-- Also not good |
| 13:17.45 | oej | cusco: I don't see a 200 OK - the call is not answered in SIP |
| 13:17.48 | cusco | yes I noticed and fixed that |
| 13:17.57 | cusco | oej: exactly I think I missing a 200 |
| 13:18.11 | cusco | thus the sate keeps on Ringing |
| 13:18.11 | oej | NAT should not be a problem here at all. |
| 13:18.40 | [TK]D-Fender | cusco, And we aren't seeing the complete call. |
| 13:18.57 | [TK]D-Fender | cusco, EVERYTHING from *'s side before the call itself... |
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| 13:25.24 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:26.31 | skrusty | is there any way to set the CLID in CDR? I am using a local channel to make calls between two seperate companies on the same *pbx, but the CDR gives the CLID as the peer and no the CLID displayed on the phone... any suggestions appreciated |
| 13:26.55 | cusco | [TK]D-Fender: that was the complete call yes |
| 13:27.09 | cusco | but I can show you more (other stuff that gets in the way) |
| 13:27.11 | [TK]D-Fender | cusco, No, that's just the SIP side |
| 13:27.15 | cusco | áhh |
| 13:27.24 | cusco | yes well I set core set verbose 1 |
| 13:27.27 | cusco | else too much output |
| 13:27.33 | cusco | can't even follow the sip |
| 13:27.36 | [TK]D-Fender | cusco, The second you think of restricting what you show when you have a problem ... DON'T. |
| 13:27.53 | cusco | hmm |
| 13:27.57 | cusco | ok |
| 13:28.12 | cusco | before I take another copy then.. |
| 13:28.28 | cusco | I set nat=no but sip show peer shows NAT: RFC3581 |
| 13:28.36 | cusco | (after a sip reload) |
| 13:29.38 | oej | Skip messing with the NAT settings right now... |
| 13:29.44 | cusco | ok |
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| 13:33.13 | cusco | here the whole output |
| 13:33.15 | cusco | http://paste.debian.net/204667/ |
| 13:33.54 | cusco | in this case din't get past 'trying' |
| 13:34.04 | cusco | and couldn't even send the CANCEL |
| 13:34.25 | cusco | thing is, we are using this peer (212.230.0.1) |
| 13:34.35 | cusco | for incomming calls too, and those work flawlessly |
| 13:35.01 | cusco | and we also dial out to some peers on their end, and we connect sucessfully... |
| 13:35.12 | cusco | now they have got this prefix for us to perform outbounds |
| 13:35.21 | cusco | and this is failling :/ |
| 13:35.47 | cusco | so I assumed there should be somewhting wrong on their end, since we can connect to a peer of theirs |
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| 13:41.08 | oej | Well, if you don't get a 200 OK, they have to look on their end |
| 13:42.02 | cusco | yes I told them so.. they said they tested with a asterisk, they used wireshark to analyze the sip, and said they only get the 200 ok after the BYE |
| 13:42.35 | cusco | I said that is normal after the BYE, but also after the INVITE, so we know he accepted our INVITE... |
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| 14:16.09 | vassilux | Hi I got WARNING[27617]: chan_sip.c:9249 process_sdp: We are requesting SRTP for audio, but they responded without it! from my asterisk box with asterisk v 1.8.0.16. I got this warning with SNOM 300. Any idea ? |
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| 14:20.02 | WIMPy | set RTP/SAVP to mandatory |
| 14:21.10 | vassilux | It works with SNOM 320 but I can't set the parameter in the SNOM 300 |
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| 14:24.48 | elred_ | Hi |
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| 14:25.38 | elred_ | I am trying to trigger a Xfer with "replaces" (non blind transfert) in the Refer-To header of my REFER. Please can someone give me advise on triggering that ? Thanks you |
| 14:30.26 | elred_ | details: what is happening actually is that asterisk receive the REFER with refer-to + callid (replaces) but it doesn't forward it to the callserver |
| 14:30.38 | elred_ | it is instead sending reinvite for rtp external bridge |
| 14:31.03 | elred_ | whereas i would it send my REFER to the callserver |
| 14:31.10 | elred_ | this is the callserver actually that i am doing test on |
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| 14:45.38 | skrusty | is there a way to turn off the dtmf audio mid call? to stop agents hearing DTMF tones of card details etc? |
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| 14:45.57 | LgK | hi |
| 14:46.02 | skrusty | hi :) |
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| 14:54.42 | troutowi | hi, i am having issues with my ivr |
| 14:54.43 | troutowi | http://pastebin.com/91xJwmsu |
| 14:59.06 | elred_ | anyone on how to have asterisk forward a REFER ? |
| 14:59.31 | elred_ | i am doing an attended transfert to asterisk and i want it to forward it to the trunk |
| 15:02.04 | leifmadsen | troutowi: your file doesn't exist |
| 15:02.25 | troutowi | thanks leifmadsen, but im looking at it XD |
| 15:02.26 | leifmadsen | troutowi: my guess is you specified the file format in your dialplan -- don't do that |
| 15:02.32 | troutowi | ok |
| 15:02.37 | troutowi | ill try leaving that out |
| 15:02.47 | leifmadsen | asterisk attempts to find an appropriate format automatically |
| 15:02.51 | troutowi | ahh |
| 15:03.01 | leifmadsen | basically it's trying to load main-ivr.sln.ulaw |
| 15:03.18 | troutowi | makes sense |
| 15:07.39 | troutowi | leifmadsen, that was the problem |
| 15:08.09 | troutowi | if u type the first few letters of the file name in the web gui, it fills in the full name including extension |
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| 15:16.44 | PatoFeo | Hi. |
| 15:16.57 | PatoFeo | someone can help me with cdr_mysql? |
| 15:17.16 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 15:18.33 | Qwell | PatoFeo: Just use cdr_odbc |
| 15:19.08 | PatoFeo | Qwell, cdr_mysql run ok, but insert the mysql record when the call end |
| 15:19.24 | PatoFeo | I need insert when the call in |
| 15:19.26 | Qwell | when it ends, as opposed to what? |
| 15:20.27 | PatoFeo | i'm developing a software in java, that read the mysql, and detect, with the external telephone number, what client is calling |
| 15:20.59 | Qwell | CDRs detail the call after it has taken place. |
| 15:21.00 | PatoFeo | if cdr save when the call end, i dont have the row to detect the client |
| 15:21.03 | cusco | java has some libraries to read the asterik manager interface |
| 15:21.12 | cusco | we use that to follow each call in certain queues |
| 15:21.36 | cusco | PatoFeo: else you could look at queue_log |
| 15:22.59 | PatoFeo | i'm seen the /var/log/asterisk/queue_log, but view: |
| 15:23.00 | PatoFeo | 1351521460|NONE|NONE|NONE|QUEUESTART| |
| 15:23.00 | PatoFeo | 1351522595|NONE|NONE|NONE|QUEUESTART| |
| 15:23.01 | PatoFeo | 1351523396|NONE|NONE|NONE|QUEUESTART| |
| 15:23.46 | Qwell | PatoFeo: You are using the wrong interface. There are several others that would be far better for your uses. |
| 15:23.55 | Qwell | ~book |
| 15:23.55 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:24.00 | Qwell | The book talks about all of them. |
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| 15:27.01 | PatoFeo | reading Qwell, thanks =) |
| 15:27.04 | leifmadsen | ALL THE THINGS! |
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| 15:27.28 | Qwell | leifmadsen: You guys lived. |
| 15:28.47 | leifmadsen | Qwell: hells ya we did |
| 15:28.59 | PatoFeo | Qwell, I must try with CEL? |
| 15:29.10 | Qwell | headdesks |
| 15:29.39 | Qwell | PatoFeo: dialplan, AMI, AGI |
| 15:30.11 | PatoFeo | thanks Qwell |
| 15:31.28 | _Corey_ | It was weird to come back from Astricon to everyone in hurricane prep... there was almost nobody on the road this morning |
| 15:32.26 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 15:33.01 | *** part/#asterisk elred_ (~elred@unaffiliated/elred-/x-5010831) |
| 15:33.15 | *** join/#asterisk ledoktre (~chatzilla@omni-01-161.cable.netins.net) |
| 15:34.35 | ledoktre | hey has anyone ever had an issue with vitelity? More specifically, for outbound.vitelity.net, it uses a number of IP addresses. My server sends an invite request to one of the IPs, it replies back authentication required, and my server replies with authentication to a different IP. End result, can't make the call. Wondered where to begin troubleshooting this issue |
| 15:35.49 | Qwell | ledoktre: I bet the booth babes changed the configuration while everybody was drunk last week. |
| 15:36.01 | _Corey_ | nice |
| 15:36.36 | ledoktre | Qwell: That sounds pretty underhanded, but what are you meaning? |
| 15:36.43 | Qwell | nothing |
| 15:37.13 | _Corey_ | Qwell: I think the motorcycle controls their routing engine |
| 15:37.33 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 15:46.46 | ledoktre | ???? |
| 15:47.20 | Qwell | ledoktre: AstriCon jokes. You have to have been one of the cool kids to get it. |
| 15:47.31 | *** join/#asterisk sherl0k (~cbalcum@pool-108-7-229-15.bstnma.fios.verizon.net) |
| 15:47.32 | Qwell | ledoktre: Call them and ask. |
| 15:48.20 | sherl0k | Hey all, I'm at my wits end with trying to get my IVRs working correctly. Does asterisk paid support cover FreePBX installs or should I talk to shmoozecom instead? |
| 15:48.40 | Qwell | newtonr: Do you know the answer to that? ^ |
| 15:49.09 | ledoktre | Qwell: Ahh, I was starting to wonder if there had been some convention. Guess I'm just not cool enough to attend ;-) I have had a ticket open with them since last week Thursday. We've exchanged plenty of responses, and their last one was, pay us $160 an hour and we'll log in and check it out for you. |
| 15:57.25 | newtonr | sherl0k: I'm pretty sure we do *goes to double check with the support dept*. Though I'd honestly say the schmoozecom guys have a focus on that, so they likely spend more time with it then our guys do. |
| 15:58.32 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:58.37 | sherl0k | Alright. But real quick, maybe you know how to answer this because if I can avoid paying I will - not everyone who dials into our conference bridge will get a response from pressing keys |
| 15:58.54 | sherl0k | the DTMF tones aren't being processed by Asterisk and I am having trouble figuring out why |
| 15:59.17 | sherl0k | Sometimes the tones are too short and Asterisk will attempt to extend them |
| 15:59.23 | [TK]D-Fender | sherl0k, that is a DTMF issue, not an IVR issue |
| 15:59.30 | sherl0k | I know that much |
| 16:00.14 | [TK]D-Fender | sherl0k, You've told us nothing about how your calls come in. |
| 16:00.30 | sherl0k | they dial a number which forwards to the IVR |
| 16:00.48 | sherl0k | and are asked to enter a 3-digit number which forwards them to a conference line |
| 16:00.49 | *** join/#asterisk phpboy (~shane@blowfish.x86.co.za) |
| 16:01.17 | phpboy | hey all, how do I remove a leading zero from the dst number that get's stored in asterisk cdr db? |
| 16:01.26 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:02.42 | newtonr | sherl0k: i don't have time to look into it at the moment. if you pastebin a log with DTMF debug, VERBOSE and DEBUG levels at 5 or so, and maybe link to a pcap (SIP and RTP) of an example call then someone may be able to help. |
| 16:02.54 | [TK]D-Fender | sherl0k, You've told us nothing about how your calls come in. <--------- |
| 16:03.02 | sherl0k | Fender, that means NOTHING to me |
| 16:03.12 | sherl0k | Please stop repeating that |
| 16:03.30 | [TK]D-Fender | builds up a tiny bit more bile .... |
| 16:03.52 | sherl0k | "how they come in" someone dials our number |
| 16:03.56 | sherl0k | how else would they come in? |
| 16:03.59 | [TK]D-Fender | WHAT TECHNOLOGY & SERVICE IS YOUR CALL ARRIVING ON? |
| 16:04.14 | sherl0k | see that is an entirely different question. |
| 16:04.24 | [TK]D-Fender | it arrives via an INTERFACE |
| 16:06.02 | sherl0k | It's a Digium TE121 |
| 16:06.41 | [TK]D-Fender | check your gains & echo first. |
| 16:07.15 | [TK]D-Fender | Either can distort the audio for tone detection. |
| 16:07.25 | sherl0k | alright |
| 16:07.43 | [TK]D-Fender | You should probably already have your DAHDI configs in a pastebin for us to be looking at.... |
| 16:07.45 | [TK]D-Fender | ~pb |
| 16:07.46 | 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. |
| 16:09.19 | sherl0k | gains are set to 0 |
| 16:09.24 | sherl0k | let me pb the config |
| 16:10.57 | sherl0k | http://pastebin.com/M8wZcrcW |
| 16:11.33 | sherl0k | sorry, it pasted twice in there |
| 16:12.36 | sherl0k | http://pastebin.com/JvpyW18V |
| 16:15.12 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:15.31 | [TK]D-Fender | that isn't all of them. You don't even have any channels defined in there. You should also be telling us precisely what version of Asterisk & DAHDI you're running, and if your card has HWEC, etc |
| 16:15.48 | [TK]D-Fender | awaits new pastebin |
| 16:18.42 | sherl0k | It's asterisk 1.8.16 |
| 16:19.34 | sherl0k | looking for the DAHDI version, it's not outputting it in a status |
| 16:24.05 | [TK]D-Fender | dahdi_cfg -vvvv |
| 16:24.30 | sherl0k | 2.6.1 |
| 16:26.00 | sherl0k | http://pastebin.com/yUqF9EqQ |
| 16:26.11 | sherl0k | there's the dahdi output |
| 16:26.15 | sherl0k | let me get the channels |
| 16:27.56 | sherl0k | http://pastebin.com/jJ0zsUZk |
| 16:30.37 | [TK]D-Fender | sherl0k, Echo Canceller(s): HWEC, MG2 <-- so yours has HWEC? Which module specifically? |
| 16:31.20 | [TK]D-Fender | check "dmesg" for "vpm" |
| 16:32.26 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:33.12 | sherl0k | nothing in there related to vpm |
| 16:33.19 | sherl0k | though i do get a lot of "dahdi: Disabled echo canceller NLP because of CED tx detected on channel" |
| 16:33.32 | sherl0k | with a channel number suffixing |
| 16:34.22 | [TK]D-Fender | That should only happen on faxing... |
| 16:35.19 | sherl0k | We have a fax system too here, which isn't working either. But that's another story I don't want to deal with ahaha |
| 16:37.41 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 16:44.31 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:45.19 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:45.19 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:45.35 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1007) |
| 16:46.22 | sherl0k | [TK]D-Fender: our older version of Asterisk (1.6) had the echo cancellation disabled in chan_dahdi.conf |
| 16:46.37 | sherl0k | Is it worth disabling it here too? |
| 16:46.40 | *** join/#asterisk bluOxigen (~noreply@unaffiliated/bluOxigen) |
| 16:46.52 | [TK]D-Fender | Normally not a good thing. That module also does DTMF detection. |
| 16:47.14 | *** join/#asterisk jsarrel (~jsarrel@66-191-165-51.dhcp.gnvl.sc.charter.com) |
| 16:47.22 | cj | anyone here ever configured a DSLAM? |
| 16:47.43 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:47.58 | cj | I can't for the life of me get any of my modems to sync with it |
| 16:48.06 | cj | and I have tried like 12 different makes and models |
| 16:48.41 | jpsharp | I've poked at an Iron Mountain box, but that's about it. |
| 16:49.10 | paulc | cj: I played with one YEARS ago.. |
| 16:50.31 | cj | is there anything I should have to do aside from matcihng up the vpi/vci? |
| 16:50.43 | cj | does it have a minimum loop length or something? |
| 16:50.52 | cj | maybe there's not enough resistance between the DSLAM and the modem |
| 16:51.07 | cj | maybe the SNR is too heavy on the S side and too light on the N |
| 16:51.29 | WIMPy | They should use adaptive power levels. |
| 16:51.47 | jpsharp | Even adapative power levels do have a limid. |
| 16:51.49 | jpsharp | limit. |
| 16:51.50 | cj | I figured they should, but I'm getting NOTHING |
| 16:52.09 | WIMPy | true |
| 16:52.14 | cj | I get a dial tone out of the line, so I know it's at least that far along |
| 16:52.24 | WIMPy | Do both ends use the same standards? |
| 16:52.37 | cj | I'm sure they share at least a few |
| 16:52.53 | cj | they should be able to fall back to the really basic of the basic to establish a link |
| 16:53.04 | cj | I could fish around for a modem that lets me specify which standard to use |
| 16:53.13 | cj | but it *should* auto-detect a shared protocol |
| 16:53.14 | WIMPy | How far are you getting? |
| 16:53.35 | cj | the sync gets nowhere |
| 16:54.02 | WIMPy | But it starts training? |
| 16:54.06 | cj | I plug the DSL splitter in to the DSLAM, the phone into the phone jack and the modem into the DSL jack |
| 16:54.10 | cj | no, no training. |
| 16:54.12 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-179-160.pools.arcor-ip.net) |
| 16:54.14 | cj | at least I don't think so |
| 16:54.25 | cj | I tell the modem to start the loop |
| 16:54.27 | ghost75 | is alaw meant to be for europe und ulaw for us ? |
| 16:54.32 | cj | and then it says that there's no signal detected |
| 16:55.15 | jpsharp | ghost75: In general, yes. |
| 16:55.22 | WIMPy | If training isn;t even staretd there's either nothing at the far end or it is so incompatible that it sees nothing. |
| 16:55.53 | jpsharp | Obvious question: Did you do try it without the DSL splitter? |
| 16:56.04 | cj | yeah, I've done that before |
| 16:57.01 | cj | i will look for training |
| 16:57.14 | cj | and a modem that lets me select the g. protocol |
| 16:58.04 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:58.05 | WIMPy | I'd expect it to start trainig, even if it is incompatible. |
| 17:02.23 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 17:03.56 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 17:15.44 | *** join/#asterisk nandersson (~nandersso@110.54.78.188.dynamic.jazztel.es) |
| 17:17.20 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:31.59 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 17:32.13 | ghost75 | is it normal that blacklisted are not shown on CDR ? |
| 17:32.26 | [TK]D-Fender | CDR shows what the call comes in with. |
| 17:32.45 | [TK]D-Fender | Please explain this "blacklist" you're referring to and where it's coming from... |
| 17:32.57 | leifmadsen | that would make sense if the channel is never answered |
| 17:33.23 | [TK]D-Fender | that too... |
| 17:33.45 | ghost75 | according to my dialplan it should be answered |
| 17:34.18 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 17:34.31 | [TK]D-Fender | And you should be showing us. |
| 17:34.31 | ghost75 | exten => 0xxxxxxxx,1,GotoIf(${BLACKLIST()}?spamtrap) |
| 17:34.56 | ghost75 | [spamtrap] |
| 17:34.57 | ghost75 | exten => s,1,Set(CALLERPRES()=prohib) ;hide own caller id |
| 17:34.57 | ghost75 | exten => s,n,Answer |
| 17:35.02 | [TK]D-Fender | PASTEBIN |
| 17:35.13 | [TK]D-Fender | ghost75, And no.... that goto will FAIL |
| 17:35.33 | ghost75 | hmm |
| 17:37.04 | ghost75 | got it from here http://www.voip-info.org/wiki/view/Asterisk+func+BLACKLIST |
| 17:39.09 | ghost75 | why it fails |
| 17:42.53 | [TK]D-Fender | Your GotoIF is bad |
| 17:42.57 | [TK]D-Fender | it isn't blacklist. |
| 17:43.06 | [TK]D-Fender | read the app's instructions |
| 17:46.07 | ghost75 | [Syntax] |
| 17:46.09 | ghost75 | BLACKLIST() |
| 17:46.16 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:46.48 | [TK]D-Fender | ghost75, what part of "it isn't blacklist, its your GOTOIF" is unclear? |
| 17:47.07 | ghost75 | you have strange way to offer help |
| 17:47.15 | [TK]D-Fender | <PROTECTED> |
| 17:47.23 | [TK]D-Fender | I told you the GOTO part was broken |
| 17:47.39 | [TK]D-Fender | <[TK]D-Fender> Your GotoIF is bad |
| 17:47.47 | [TK]D-Fender | "core show application gotoif" ,_ |
| 17:52.56 | ghost75 | GotoIf(${BLACKLIST()=1} <- this ? |
| 17:53.48 | [TK]D-Fender | No, your TARGET |
| 17:53.58 | [TK]D-Fender | read the app's instructions again... |
| 17:55.16 | leifmadsen | no |
| 18:01.06 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:01.06 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 18:01.25 | ghost75 | ?spamtrap,s,1 <- so? |
| 18:02.58 | leifmadsen | yes |
| 18:03.11 | leifmadsen | your example just goes to the spamtrap priority label |
| 18:03.23 | ghost75 | ok |
| 18:04.07 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 18:04.23 | ghost75 | will be fun when those scammers talk to my wav files |
| 18:05.04 | ChannelZ | I just ignore spam calls and don't even answer them. They get no progress indication at all, and I don't get billed. |
| 18:05.47 | ghost75 | i thought its not possible to get a bill |
| 18:06.35 | ChannelZ | ? |
| 18:07.14 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 18:07.52 | ghost75 | here always the caller pays for calls |
| 18:08.12 | leifmadsen | not if the call isn't answered |
| 18:08.18 | leifmadsen | you don't pay for "ringing" |
| 18:11.43 | ChannelZ | In my case I pay for both. If you have unlimited inbound it's not a big deal, but I prefer to not even bother wasting the time/bandwidth :) |
| 18:11.50 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 18:12.14 | ghost75 | from which country are you? |
| 18:12.44 | ChannelZ | USA. Using SIP via Vitelity |
| 18:14.07 | ghost75 | in germany the initiator pays |
| 18:14.30 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 18:14.40 | ChannelZ | Then by all means waste as much of their money as possible! |
| 18:16.34 | ghost75 | do you have also guys with suppressed numbers? |
| 18:18.32 | [TK]D-Fender | Yes |
| 18:21.31 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 18:22.11 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 18:31.19 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
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| 18:34.03 | *** join/#asterisk Dovid (60f23f16@gateway/web/freenode/ip.96.242.63.22) |
| 18:34.33 | Dovid | is there any CHANGELOG for res_fax_digium-1.8.4_1.3.1 from 1.3.0 to 1.3.1? |
| 18:41.26 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 18:49.38 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
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| 19:12.57 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:12.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:17.03 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 19:20.27 | *** join/#asterisk navaismo (~navaismo@189.191.22.203) |
| 19:21.25 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 19:22.01 | troutowi | http://pastebin.com/NbqPQTmY |
| 19:22.17 | troutowi | recieving this error when trying to forward a call to a cell |
| 19:22.24 | AkkerKid | how come once a day I get disconnected and reconnected to freenode and this channel is the only one that requires me to log into freenode with my password before it reconnects? |
| 19:22.49 | Qwell | AkkerKid: For the former, you need to fix your connection. And the latter is because we have +r set. |
| 19:23.03 | Qwell | Use a server password instead of using NickServ. Problem solved. |
| 19:24.08 | jpsharp | troutowi: Do you have that extension defined in the extensions.conf context? And are you really dialling 9XXXXX.. or are you obfuscating the number? |
| 19:24.31 | AkkerKid | Qwell: I've got 50MBit business class fiber coming into my office... There should be no excuse... tisk tisk |
| 19:24.51 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 19:25.01 | troutowi | jpsharp, im not dialing the X's just obfuscating for here. |
| 19:25.02 | Qwell | shrugs |
| 19:25.07 | Qwell | AkkerKid: My connections stay up for a month. |
| 19:25.19 | [TK]D-Fender | troutowi, Asterisk GUI is dead and has been for over years, and users.conf .... is a mistake.... |
| 19:25.49 | [TK]D-Fender | numberc <-- and I am unfamiliar with this being an actual parameter. |
| 19:26.26 | troutowi | i dont see a numberc, do you mean number? |
| 19:26.34 | [TK]D-Fender | yes |
| 19:27.13 | troutowi | this info was pulled from followme.conf [TK]D-Fender, not user.conf |
| 19:28.32 | jpsharp | Well, the error basically says the number you're trying to dial doesn't exist in the dialplan context you're trying to call through. |
| 19:29.19 | [TK]D-Fender | oops |
| 19:29.21 | troutowi | http://pastebin.com/65fffenT |
| 19:29.35 | troutowi | this is my dialplan in extensions.conf |
| 19:29.44 | Qwell | and which of those has extension 9XXXXXXXXX? |
| 19:31.07 | troutowi | what would i have to add for the dialplan to include outside numbers? |
| 19:32.34 | troutowi | the 9 is just for when dialing outside numbers, the X's are the area code + number |
| 19:33.14 | [TK]D-Fender | troutowi, He's asking you to SHOW him which one should match that number..... |
| 19:34.11 | troutowi | yes, and im saying i do not know |
| 19:34.16 | troutowi | and am wondering the same |
| 19:37.22 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:37.23 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:38.01 | [TK]D-Fender | troutowi, what do you mean "don't know"? It's in your dialplan. You've already shown us part of is. Show the context witht he exten that SHOULD match that # |
| 19:39.43 | troutowi | http://pastebin.com/B2skP3Nx |
| 19:39.58 | [TK]D-Fender | taht does not show a pattern to dial outbound at all |
| 19:40.10 | [TK]D-Fender | <Qwell> and which of those has extension 9XXXXXXXXX? |
| 19:40.38 | troutowi | http://pastebin.com/8xK37Dn3 |
| 19:40.52 | [TK]D-Fender | exten = _9X,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:1},,trunk_1,) |
| 19:40.56 | [TK]D-Fender | that matches TWO digits |
| 19:41.03 | [TK]D-Fender | Not longer |
| 19:41.18 | [TK]D-Fender | If that's what you made to dial out, you didn't make the pattern right |
| 19:41.40 | troutowi | i messed that up then |
| 19:43.08 | troutowi | _9 should be all i need |
| 19:43.13 | [TK]D-Fender | troutowi, "_9X." that will match starting with 9, following with another digit, and one or more charaters of any kind. |
| 19:43.17 | [TK]D-Fender | no.... |
| 19:43.27 | [TK]D-Fender | you do not seem to understand PATTERN chars. |
| 19:43.35 | [TK]D-Fender | _( alone means 9. Nothing else, just 9 |
| 19:43.42 | troutowi | ah ok |
| 19:43.47 | [TK]D-Fender | "." = 1 or more |
| 19:44.00 | [TK]D-Fender | go read your basics, and remember that that GUI is dead. |
| 19:44.05 | troutowi | will do |
| 19:44.27 | [TK]D-Fender | It would be a mistake to continue building with it... |
| 19:44.45 | troutowi | why is it not being worked on anymore? |
| 19:45.01 | _Corey_ | For those not connected with Digium's Facebook feed... the Astricon photos are posted: http://www.flickr.com/photos/41264116@N00/sets/72157631882300011/with/8135872223/ |
| 19:45.19 | ectospasm | troutowi: lack of popularity with users |
| 19:45.27 | file | bonus points for those who can spot me |
| 19:45.31 | [TK]D-Fender | troutowi, No business sense in doing so. They are concentrating on their commercial products |
| 19:45.39 | troutowi | ok |
| 19:46.09 | _Corey_ | file: So far (from here) I've spotted Qwell |
| 19:46.20 | _Corey_ | hard to miss tho... :-) |
| 19:46.45 | file | quite |
| 19:46.54 | drmessano | Unless youre tall |
| 19:47.00 | drmessano | ducks |
| 19:48.39 | *** join/#asterisk hardwire (~hardwire@69-161-26-211.static.acsalaska.net) |
| 19:51.24 | _Corey_ | file: You're spotted |
| 19:51.31 | file | bows |
| 19:51.32 | hardwire | like a goat. |
| 19:51.48 | hardwire | starts making a python/corosync adapter to publish hint information from the console |
| 19:54.33 | *** join/#asterisk Greek-Boy (~w55@197.152.75.74) |
| 19:54.59 | Qwell | _Corey_: bchia pointed out that I made the #1 spot even. |
| 19:55.53 | _Corey_ | Cover model |
| 19:56.19 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 20:00.46 | _Corey_ | Looks like they edited all the Vitelity girls out |
| 20:01.01 | hardwire | are photos going up somewhere? |
| 20:01.25 | _Corey_ | [15:45]_Corey_For those not connected with Digium's Facebook feed... the Astricon photos are posted: http://www.flickr.com/photos/41264116@N00/sets/72157631882300011/with/8135872223/ |
| 20:01.33 | _Corey_ | hardwire: ^^ |
| 20:02.01 | hardwire | danke |
| 20:03.05 | _Corey_ | Seems to just be the opening reception... there are definitely some better photo albums floating around out there |
| 20:03.08 | hardwire | Next year.. I request permission to go to walmart and get 30 tents and scatter them throughout the hotel |
| 20:03.13 | hardwire | and put cat-5 into each of them |
| 20:03.16 | hardwire | and power |
| 20:04.46 | *** join/#asterisk dms (~dms@65.207.151.254) |
| 20:05.39 | dms | so . . anyone know what the max amount of TCE400B cards can be put in a box and effectively used with Asterisk? |
| 20:05.44 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 20:05.49 | hardwire | doh.. I didn't go to the opening reception |
| 20:06.10 | _Corey_ | hardwire: Um, they actually have rooms you can rent at most of the higher-end hotels... ;-) |
| 20:06.19 | _Corey_ | (no tents required) |
| 20:06.27 | BlackBishop | any easy way to set up a transparent asterisk ? :/ My Sip provider doesn't like my clients to be natted and I was thinking I could run an asterisk on my end .. and forward all the sip connections and auth and stuff |
| 20:06.42 | hardwire | _Corey_: no.. they didn't. |
| 20:07.02 | hardwire | I had to beg until I got one. My travel guy at work put my hotel reservations in for the previous month.. but travel for the correct date. |
| 20:07.24 | _Corey_ | oh? I was able to book one for someone we brought down last-minute just the week before |
| 20:07.35 | hardwire | yeh |
| 20:07.51 | [TK]D-Fender | BlackBishop, ... |
| 20:07.52 | [TK]D-Fender | ~b2bua |
| 20:07.53 | infobot | hmm... b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 20:07.54 | [TK]D-Fender | ^^^ |
| 20:07.58 | [TK]D-Fender | Not a proxy. |
| 20:08.00 | hardwire | I stayed monday night on my dime then found out the next day that they were totally capable of giving me a room for a week |
| 20:08.13 | hardwire | they just had to make sure I was commited I guess. |
| 20:08.22 | [TK]D-Fender | BlackBishop, * can register AS your device to them and deal with it's own end |
| 20:15.18 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:20.18 | [TK]D-Fender | checkout time, BBIAB |
| 20:20.59 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 20:34.42 | *** join/#asterisk donnib (~donnib@2.105.191.170) |
| 20:35.51 | *** join/#asterisk natschil (~nathanael@stgt-4d0387d7.pool.mediaWays.net) |
| 20:36.04 | natschil | Hello. Anyone know how I can set asterix to allow zrtp packets to pass through? |
| 20:36.43 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 20:36.43 | Qwell | What's asterix? |
| 20:37.17 | natschil | s/asterix/asterisk/ |
| 20:37.19 | natschil | my bad |
| 20:42.37 | *** join/#asterisk danfromuk (~IceChat77@2.27.39.225) |
| 20:43.25 | *** part/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 20:43.52 | natschil | ,tls |
| 20:47.47 | natschil | what about does anybody know a good guide for setting up tls with asterisk |
| 20:48.21 | Qwell | ~asterisk wiki |
| 20:48.21 | infobot | extra, extra, read all about it, asterisk wiki is http://wiki.asterisk.org/ |
| 20:48.24 | Qwell | natschil: Did you check there? |
| 20:49.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:49.40 | natschil | Qwell: nope, thanks a lot for the link |
| 20:50.28 | *** join/#asterisk uncertain_bear (~kethonna@c-71-199-153-9.hsd1.ga.comcast.net) |
| 20:53.22 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 20:54.39 | *** join/#asterisk wonderworld (~ww@dsdf-4db5cd91.pool.mediaWays.net) |
| 20:58.31 | *** join/#asterisk acidrye0n (~acidrye0n@243.sub-97-248-208.myvzw.com) |
| 21:03.21 | *** join/#asterisk dms (~dms@65.207.151.254) |
| 21:03.29 | dms | sruffell: ping |
| 21:05.52 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 21:09.45 | sruffell | dms: pong |
| 21:19.33 | dms | sruffell: the TCE400P . . by chance, do you know how many of those cards can be stuffed in one machine? |
| 21:20.17 | sruffell | there isn't a limit in software……probably would run out of PCI slots before you run out of PCI bus bandwidth |
| 21:20.25 | sruffell | unless you're trying to use an expansion chassis or something. |
| 21:24.50 | *** join/#asterisk acedia_ (~garland@unaffiliated/ffs) |
| 21:29.19 | dms | sruffell: ok, so I can stick as many of those cards in one chassis as I can/want and Asterisk would "see" them all (capacity wise)? |
| 21:29.27 | dms | just want to make sure I got that right |
| 21:29.28 | dms | :) |
| 21:30.11 | sruffell | Yes…that's right. |
| 21:34.41 | dms | sruffell: thanks man |
| 21:39.21 | natschil | is there anywhere I can simply download the ast_tls_cert script? |
| 21:40.01 | natschil | (without downloading the whole asterisk source) |
| 21:40.48 | [TK]D-Fender | Right off SVN |
| 21:41.28 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 21:41.34 | tonikasch | p |
| 21:41.54 | natschil | [TK]D-Fender: I don't want to install svn just for downloading one file, is there perhaps a http link somewhere? |
| 21:42.20 | tonikasch | Hi, do you know a "ast_rtp_write: Don't know how to send format unknown packets with RTP" message could be created by? |
| 21:49.11 | [TK]D-Fender | natschil: You don't have to. |
| 21:49.25 | [TK]D-Fender | SVN is *over* HTTP |
| 21:49.30 | natschil | [TK]D-Fender: okay, my bad. |
| 21:56.48 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-gsqoegluhjkrjkqu) |
| 22:07.09 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 22:13.34 | tm1000 | Qwell: ping |
| 22:13.45 | tm1000 | or sruffell ping |
| 22:14.11 | tonikasch | <PROTECTED> |
| 22:14.46 | tonikasch | tm1000, might be worth doing a whois qwell qwell and see his/her idle time |
| 22:15.06 | tm1000 | tonikasch: all the devs are idle....so I picked one randomly :-p |
| 22:15.13 | tonikasch | :P ok |
| 22:15.24 | tm1000 | really im looking for josh but i dont think he's in here |
| 22:17.06 | sruffell | huh? |
| 22:17.08 | newtonr | tm1000: jcolp? |
| 22:17.27 | tm1000 | sruffell: when is jcolp online |
| 22:17.31 | tm1000 | newtonr: yes |
| 22:17.40 | newtonr | tm1000: jcolp = file |
| 22:17.46 | tm1000 | roger |
| 22:18.00 | tm1000 | file: ping |
| 22:18.30 | sruffell | tm1000: have I met you? |
| 22:18.34 | newtonr | he doesn't have a set schedule, he seems to be online 20 hours a day |
| 22:18.39 | tm1000 | sruffell: yes |
| 22:18.41 | file | tm1000, Mr. Nagy! |
| 22:18.42 | natschil | I'm trying to setup asterisk using tls. Using the ast_tls_cert script, I've created some certificates and setup sip.conf accordingly. However, running sip reload gives me: SSL error loading cert file. <asterisk.pem>. Is there any way I can get more informative error messages (I've set sip debug to on and am running at a very verbose error level on the CLI) |
| 22:18.55 | tm1000 | file: Heyyy! stuff & stuff |
| 22:19.17 | file | tm1000, to answer your direct message - you need multiple connections configured in xmpp.conf and multiple endpoints configured in motif.conf |
| 22:19.26 | tm1000 | I tried that |
| 22:19.28 | tm1000 | hmm |
| 22:19.30 | sruffell | tm1000: hmm…I'm drawing a blank on your irc handle. |
| 22:19.38 | tm1000 | xmpp show connections showed all connected. but asterisk did nothing |
| 22:20.00 | tm1000 | sruffell: were you in the distro meeting with david and josh and matt and malcom? |
| 22:20.35 | file | did chan_motif successfully load? |
| 22:20.39 | tm1000 | file: most-likely I am doing it wrong. however when I remove the second account it works again |
| 22:20.39 | tm1000 | yes |
| 22:21.05 | sruffell | tm1000: negative. |
| 22:21.08 | tm1000 | after unload and load I should mention |
| 22:21.20 | file | internally accounts don't overlap *at all* so it would be interesting if that ended up being the issue... |
| 22:21.32 | tm1000 | sruffell: ok. astricon. dinner? one of the two nights? the brazilian steakhouse? |
| 22:22.03 | sruffell | yes…I was there. |
| 22:22.10 | file | tm1000, hum hum hum what does xmpp set debug on show? |
| 22:22.13 | newtonr | tm1000: you were there at fire of brazil? |
| 22:22.18 | file | tm1000, XMPP traffic all happy and joyous? |
| 22:22.24 | tm1000 | newtonr: yes |
| 22:22.33 | newtonr | oops i meant that for sruffell |
| 22:22.36 | newtonr | lol |
| 22:22.47 | sruffell | newtonr: I sat right behind you. |
| 22:23.00 | tm1000 | file: ah! I will run that and get back to you. I got side tracked. I'll do some debugging. There's no debugging for motif though is there? |
| 22:23.05 | ghost75 | small small world |
| 22:23.12 | newtonr | sruffell: ahh, haha |
| 22:23.15 | tm1000 | ghost75: no. we all went as a group |
| 22:23.48 | file | tm1000, not REALLY - there's few things where it would be useful/could be added |
| 22:25.03 | natschil | anybody have any idea as to how I can get more info about why asterisk isn't taking my certificate file? |
| 22:25.39 | tm1000 | 20 people or so I think? |
| 22:26.10 | tm1000 | netman: sruffell I was again the wall to the left |
| 22:26.24 | *** join/#asterisk gain (~gain@82.54.21.50) |
| 22:26.45 | natschil | nevermind, I think I know why |
| 22:27.35 | gain | hi all, I have a sip trunk on which a friend of mine ask me to enable t38 and g711 for fax transmission... |
| 22:29.23 | file | tm1000, I just did a call from Jitsi using Jingle out through Google Voice to a toll-free number for kicks and it was happy :D |
| 22:30.14 | tm1000 | file: hmmm ok. Well I know one account definately works. The debugging should provide some light. Ill come back and bug you again if I cant figure it out |
| 22:30.15 | file | tm1000, kk! |
| 22:30.30 | file | tm1000, thanks for the add-on work btw |
| 22:30.49 | tm1000 | file: np. not sure you use freepbx though ;-) |
| 22:31.04 | file | I don't, but many people do :D |
| 22:32.17 | natschil | actually, it still doesn't work. Therefore, anybody have any suggestions at all as to why asterisk might not be accepting my certificate? |
| 22:32.52 | tm1000 | file: thanks for working on chan_motif. eventhough it will most-likely be shutdown by google sooner or later. however your jabber/xmpp stuff will always be usefull |
| 22:33.09 | file | tm1000, yeah I dunno what they are going to do... it's rather curious |
| 22:38.03 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-xlriavsimreymfah) |
| 22:41.52 | natschil | sorry I keep asking this, but: does anybody know how to get more meaningful errors than "SSL error loading cert file. </path/to/a/file/that/exists>" when doing sip reload? |
| 22:42.17 | tm1000 | natschil: whats your debug set at |
| 22:45.57 | natschil | tm1000: I did "sip set debug on" and "core set debug 10" |
| 22:55.56 | *** join/#asterisk BlackBishop (~dexter@ipv6.d3xt3r01.tk) |
| 22:59.43 | natschil | quite frankly, it looks like ast_tls_cert is a failure |
| 23:00.07 | *** join/#asterisk navaismo (~navaismo@189.191.22.203) |
| 23:00.55 | natschil | the fact that asterisk does not seem to be able to output openssl debug information is also a bit worrying. |
| 23:03.17 | natschil | but I am now giving up. |
| 23:04.10 | natschil | I have followed https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial to the letter |
| 23:04.22 | natschil | and am only getting SSL error loading cert file. </etc/asterisk/keys/asterisk.pem> |
| 23:04.39 | natschil | conclusion: either asterisk is broken, or the wiki page is broken, or I have made a mistake. |
| 23:05.02 | SeRi | naybody have any experience with flowroute or vitality? |
| 23:05.13 | SeRi | s/naybody/anybody/ |
| 23:05.32 | jpsharp | My client uses vitaliy. |
| 23:05.37 | tm1000 | file: So I tried the debugging route. I see an XMPP message received from one account when two are 'connected' but ti doesn't do anything with it. if I call the second account there are no messages. when I remove either of the accounts so that there is just one and reload I get calls again |
| 23:05.51 | tm1000 | want me to just send you my motif.conf and xmpp.conf files over email? |
| 23:06.14 | file | sure |
| 23:06.21 | file | jcolp@digium.com |
| 23:06.55 | natschil | please, does nobody here know howto in some way get openssl debugging information? |
| 23:07.35 | tm1000 | natschil: what are the perms on your ssl file |
| 23:10.10 | natschil | tm1000: thanks, that could be it. |
| 23:10.14 | natschil | tm1000: let me see |
| 23:10.19 | tm1000 | they cant be 777 |
| 23:10.35 | natschil | tm1000: I set them to 777 and now it works |
| 23:10.38 | natschil | tm1000: thank you so much |
| 23:10.42 | tm1000 | oh |
| 23:10.43 | tm1000 | hm |
| 23:10.47 | tm1000 | sometimes 777 denys it |
| 23:10.56 | tm1000 | Id play with them now though and see what you can get to work |
| 23:11.02 | tm1000 | or just have the asterisk user own them |
| 23:11.04 | natschil | tm1000: I guess I ran the ast_tls script as root, but asterisk doesn't run as root |
| 23:11.05 | tm1000 | natschil: ^^ |
| 23:11.08 | natschil | tm1000: thanks again :D |
| 23:12.28 | *** join/#asterisk wonderworld (~ww@dsdf-4db5cd91.pool.mediaWays.net) |
| 23:13.06 | natschil | btw, does tls go over tcp or udp? |
| 23:13.09 | natschil | in this case? |
| 23:13.31 | file | TLS is over TCP |
| 23:13.33 | file | DTLS is over UDP |
| 23:14.49 | natschil | file: okay. Does asterisk do DTLS? |
| 23:14.55 | SeRi | jpsharp: how do they like it? |
| 23:15.05 | file | it does DTLS-SRTP in Asterisk 11, that's it |
| 23:16.01 | natschil | file: okay. I run 10.9, so I guess I'm stuck with TLS..... I'm hoping that that doesn't mess with nat traversal etc |
| 23:17.12 | jpsharp | SeRi: There's been a few hiccups, but Vitelity was quick to own up to the issues and fix it. |
| 23:17.34 | SeRi | jpsharp: MhhMhh Thats good. |
| 23:17.39 | SeRi | So good tech support. |
| 23:17.44 | jpsharp | Yeah. |
| 23:18.13 | jpsharp | Proactive, too. I broke asterisk one night and didn't realize it until Vitelity called and said "Hey, you guys are rejecting all the calls we're sending you." |
| 23:18.19 | tm1000 | file: ok soooo ironically....I found and error in my own foreach code before I was about to send you my motif.conf file which made that file malformed with more than two accounts.. I think it's working now. Will have to test later when I'm behind less firewalls |
| 23:18.48 | file | tm1000, ah! |
| 23:19.02 | tm1000 | file: yes I feel dumb. hahaha |
| 23:19.04 | SeRi | jpsharp: wow now that is nice |
| 23:19.20 | file | tm1000, drink? |
| 23:20.53 | SeRi | I am testing flowroute right now |
| 23:21.13 | *** join/#asterisk natschil (~nathanael@stgt-4d0387d7.pool.mediaWays.net) |
| 23:21.24 | natschil | anywho, I'm going. Thanks a lot for all the help. |
| 23:21.36 | *** join/#asterisk requiemzz (~joe@unaffiliated/requiemzz) |
| 23:22.19 | jpsharp | I'm going to get a DID or two from them myself to play around with their SMS service. |
| 23:22.24 | requiemzz | hello |
| 23:23.23 | requiemzz | has anyone dealt with Cisco 7940 IP phones? |
| 23:23.47 | jpsharp | I think we all have at one time or another. |
| 23:24.10 | requiemzz | Im having a bit of trouble getting one to connect remotely. |
| 23:24.56 | requiemzz | Basically trying to get it to work outside the office. |
| 23:26.37 | requiemzz | It wont work not matter what I do im pulling my hair out here. |
| 23:28.36 | jpsharp | Firewalls? Improper port forwarding? |
| 23:29.04 | requiemzz | yes behind a NAT I have the proper ports forwarded. |
| 23:29.26 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 23:30.06 | jpsharp | Can you connect to your server from where you're at via softphone or some other IP phone? |
| 23:30.33 | tm1000 | file: yes. google does make me drink ;-) |
| 23:30.46 | requiemzz | i have not tried to connect via a softphone yet. I am not sure how to do that. |
| 23:30.56 | SeRi | jpsharp: from flowroute or vitelity? |
| 23:31.37 | jpsharp | vitelity. |
| 23:36.10 | SeRi | sms sounds like something I would like to play with..... |
| 23:38.23 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 23:38.47 | SeRi | flowroute is a bit expensive |
| 23:38.50 | *** part/#asterisk gdeeble (~gdeeble@184.170.111.173) |
| 23:39.05 | SeRi | havent test the quality yet. just got it working the other night.... |
| 23:44.19 | requiemzz | so i tested with a softphone and I get a 408 request failed message on the softphone. |
| 23:48.14 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 23:51.21 | nantou | xubuntu 12.10 with asterisk and gastman installed from the repo, i am being asked for a hostname to connect to, am I good to go? |
| 23:53.03 | *** join/#asterisk vite (~vite@unaffiliated/vite) |
| 23:53.50 | vite | Hi youall, if I use the asterisknow os, can I create an asterisk server (for personal use) and make calls to other sip addresses, or do I have to have a registrar etc for that to work? |
| 23:54.10 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 23:55.13 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 23:57.21 | nantou | a hostname is an actual user, am I wrong? |
| 23:57.29 | *** join/#asterisk voxter (~hardcore@70.36.63.61) |
| 23:58.23 | [TK]D-Fender | (note: this does not appear to be under active development though nightly snapshots are still provided and code still works. If you are the author, please update this page. - March 6, 2007) |
| 23:58.30 | [TK]D-Fender | Appears Gastman is ANCIENT and dead |
| 23:58.40 | [TK]D-Fender | And should not support anything remotely current |
| 23:59.22 | [TK]D-Fender | vite: You can use it for whatever you want |