| 00:01.56 | WIMPy | Do you have any ISDN interfaces at all? |
| 00:02.25 | schultza | no... but im building a testing environment and following the asterisk book |
| 00:02.46 | schultza | and the book's argument is "completeness" |
| 00:02.48 | WIMPy | Then you have no use for libpri. |
| 00:03.27 | WIMPy | Do you have any hardware interfaces at all? |
| 00:03.27 | WIMPy | That doesn't make much sense. |
| 00:03.48 | WIMPy | If you want it all, there's a lot more than what is discussed in the book. |
| 00:04.20 | schultza | if you mean for asterisk and connecting phones or circuits.. no... |
| 00:04.34 | schultza | just following the book... |
| 00:04.42 | schultza | ... for completeness in the book's mind |
| 00:04.55 | WIMPy | Then you don't even need dahdi unless you want the nostalgia of using MeetMe. |
| 00:05.16 | navaismo | latest libpri require dahdi installed first |
| 00:05.19 | schultza | the book covers meetme.. so yes i need dahdi |
| 00:05.41 | WIMPy | You should look at Confbridge instead. |
| 00:06.05 | WIMPy | Do you have any other plans for the next couple of years? |
| 00:06.06 | schultza | fatal error on dahdi... no such file or dir for linux/smp_lock.h |
| 00:06.30 | schultza | mostly this is a learning environment for me at the moment |
| 00:06.48 | WIMPy | To compile dahdi you need the kernel headers or a configured kernel source. |
| 00:06.54 | schultza | and yes.. id like to set up a home pbx |
| 00:07.27 | schultza | linux-headers-3.2.0-35-generic is already the newest version. |
| 00:07.44 | schultza | ubuntu.. already installed |
| 00:07.53 | WIMPy | The newest version is useless. |
| 00:08.07 | WIMPy | You need the exact same version of the kernel you're running. |
| 00:08.43 | schultza | i ran a <sudo apt-get install linux-headers-`uname -r`> to get just that. |
| 00:10.23 | WIMPy | Doesn;t sound bad, but I can't tell you what exactely you need on your installation to be able to build 3rd party kernel modules. |
| 00:11.08 | schultza | the book required me to set up an asteriskpbx user with sudo privileges... done |
| 00:11.37 | schultza | then a <sudo apt-get install build-essential subversion libncurses5-dev libssl-dev libxml2-dev vim-nox> |
| 00:11.40 | schultza | done |
| 00:11.48 | schultza | i downloaded with svn |
| 00:12.02 | *** part/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 00:12.04 | schultza | all 3 ... dadhi, libpri. and asterisk (1.8) |
| 00:12.20 | schultza | and then the headers for my current kernel |
| 00:12.53 | schultza | hence where i am at now |
| 00:15.23 | schultza | i was looking in my linclude folder under the headers. and i dont see the file it was looking for.. apparently i have the wrong kernel |
| 00:16.08 | schultza | either that... or the definitive guide isnt what i need |
| 00:16.47 | *** join/#asterisk hugo (~hugo@unaffiliated/hugo) |
| 00:17.38 | hugo | hey guys quick one, I'm testing with one friend to establish a voice call, the call establishes but there's no audio. on the firewall in front of asterisk (it is running under nat) we can see it trying to send UDP packets to our LAN IPs (the NAT where asterisk is is not the same NAT we're on here) |
| 00:18.06 | hugo | I did tinker with the NAT settings in the config but clearly I missed something. does that sound like a familiar situation to anyone? |
| 00:22.24 | camelCase | codec issue |
| 00:34.23 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:40.24 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 00:40.30 | autofsckk | good night everybody |
| 00:41.28 | autofsckk | i cant transfers calls between local extensions, that happens after i change from ulaw to g729, i get this waring from cli channel.c:4938 ast_write: Codec mismatch on channel SIP/0014BFFD36F5-verd1-00000084 setting write format to ulaw from g729 native formats 0x100 (g729) |
| 00:41.38 | autofsckk | how can i fix it? |
| 00:42.34 | gusto | so |
| 00:42.54 | gusto | aha |
| 00:42.58 | gusto | that's easy |
| 00:43.41 | gusto | when he says that there is a code mismatch then it means that either he can not find a matching codec between those two, or (what i expect) he has no way to transcode it |
| 00:44.24 | WIMPy | it |
| 00:44.24 | gusto | do you have the codec_g729 loaded? |
| 00:44.33 | autofsckk | gusto: is there a way that i can make it transcode it? |
| 00:44.45 | gusto | and why would someone step from ulaw to g729 that's a mystery as well |
| 00:44.52 | autofsckk | gusto: well im using it in 3 phones, or can you tell me how can i see if i have it loaded please? |
| 00:45.24 | WIMPy | You need to buy a licence from Digium. |
| 00:46.06 | autofsckk | gusto: i really dont know, im new to asterisk and i dont know why it has to change from ulaw to g729 because my phones use it, and my asterisk server recieves from my provider g729 too, so i dont know why ulaw gets in the middle |
| 00:46.54 | autofsckk | WIMPy: ok i see, so if i buy the license that problem is gone? because if i use the phones without transfering the call, they work great, the problem is when i transfer |
| 00:47.45 | WIMPy | The other thing would be to make sure that either everything or nothing uses G.729. |
| 00:48.05 | WIMPy | ... so you don't have no need for transcoding. |
| 00:48.38 | gusto | yes |
| 00:48.45 | gusto | g729 is bullshit either |
| 00:49.23 | gusto | i mean ... to me only g722 makes sense, because that offers superior quality over g711 (i ve tested that!) but then you also need to have phones that can do that (softphones do well) |
| 00:49.35 | autofsckk | well, i was using ulaw, had great quality, but too much bandwith, now i have good quality and very little bandwith |
| 00:49.59 | autofsckk | i have some yealink and pap2 |
| 00:50.15 | gusto | on ATA adapters or SIP phones i would prefer uncompressed codec between the phone and asterisk, because there is no point of compressing (thus saving bandwidth) on your local (possibly gigabit) LAN |
| 00:50.29 | autofsckk | and have to work with csipsimple at android |
| 00:50.47 | gusto | you have linksys PAP2T? |
| 00:50.56 | autofsckk | yep |
| 00:51.05 | leifmadsen | I think my Sipura SPA3000 finally died |
| 00:52.05 | gusto | i do not care about that, but i have a PAP2T here ;-) |
| 00:52.16 | leifmadsen | I was just sayin :) |
| 00:52.18 | gusto | and i can tell, that the best codec that PAP2T does is ALAW |
| 00:52.29 | gusto | leifmadsen: is it an ATA? |
| 00:52.36 | autofsckk | gusto: so you think i should use g722 instead? |
| 00:52.40 | leifmadsen | that's what a SPA3000 is ,yes |
| 00:52.47 | gusto | cool |
| 00:53.10 | gusto | autofsckk: well, g722 uses even more bandwidth and of course PAP2T does not support that |
| 00:53.31 | autofsckk | gusto: i dont have tu much upload, thats the problem |
| 00:53.36 | WIMPy | No, it doesn't. |
| 00:53.43 | WIMPy | The same or less. |
| 00:53.54 | autofsckk | thats why i tried g729 |
| 00:54.19 | autofsckk | ulaw was working just fine, but the upload is the problem here |
| 00:55.29 | gusto | leifmadsen: do i understand it right, it's an linksys product? |
| 00:55.37 | gusto | http://www.voip-info.org/wiki/view/Sipura+3000 |
| 00:56.23 | leifmadsen | gusto: Linksys bought Sipura |
| 00:56.32 | leifmadsen | which was bought by Cisco |
| 00:56.35 | zerohalo | then Cisco bought Linksys |
| 00:56.37 | gusto | leifmadsen: ah, that's a pitty |
| 00:56.37 | zerohalo | right |
| 00:56.50 | leifmadsen | this was several years ago |
| 00:56.56 | WIMPy | wonders if DIALPLAN_EXISTS will work with switches... |
| 00:57.07 | gusto | that's like WD buying Hitachi or more like Segate buying Maxtor |
| 00:57.39 | leifmadsen | I doubt anything post Asterisk 1.0 works with switches |
| 00:57.41 | gusto | when two incompetent companies put themselves together the result is even worse |
| 00:57.46 | autofsckk | i want to read about transcode inbetween * servers, where can i find something useful so i can understand more that process please? |
| 00:58.08 | WIMPy | Well, Asterisk has some issues with it, but off course they still work. |
| 00:58.25 | hugo | what's a good softphone for windows that supports sip-tls and srtp ? |
| 00:58.27 | WIMPy | Otherweise I wouldn't (couldn't sensibly) use Asterisk. |
| 00:58.46 | gusto | autofsckk: is your Asterisk installation on your LAN or on the internet somewhere on a vserver or something? |
| 00:58.57 | hugo | oh, and video |
| 00:59.02 | leifmadsen | hugo: jitsi or zoiper |
| 00:59.23 | *** part/#asterisk navaismo (~navaismo@189.144.207.195) |
| 00:59.25 | autofsckk | gusto: i have one on a vps, but i will put another one locally, so i have to connect them both soon |
| 00:59.42 | hugo | I'll have a look leifmadsen , thx |
| 01:00.03 | autofsckk | i have 3 land lines here that i have to connect with the other 3 voip lines i have running on the vps |
| 01:01.05 | gusto | autofsckk: yes, that would be my suggestion that you put a local installation on your lan and use g711 codec between that local asterisk and those ATA devices for SIP |
| 01:01.22 | gusto | autofsckk: because on your LAN bandwidth does not matter |
| 01:01.42 | WIMPy | No, but transcoding does matter. |
| 01:02.26 | gusto | autofsckk: then you can connect those two asterisk's with IAX2 and use some low bandwidth codec for that connection so that you do not waste much of your internet connection's bandwidht |
| 01:02.30 | gusto | bandwidth |
| 01:02.36 | autofsckk | WIMPy: where can i read more about transcoding? |
| 01:02.41 | gusto | WIMPy: why should transcoding matter? |
| 01:02.49 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 01:03.18 | WIMPy | Because it uses CPU time and in the case of G.729 requires additional licenses. |
| 01:03.29 | autofsckk | gusto: thats how i am planning it, of course i have too much to read, but i thought that was the right way to do it, thanks |
| 01:03.34 | gusto | autofsckk: w8, do not care about transcoding, you only have to have all needed codecs (at best all of them) loaded and then allow codecs from connection to connection and asterisk takes over the transcoding process for you |
| 01:03.48 | gusto | WIMPy: w8 w8 w8 |
| 01:03.56 | autofsckk | what is w8? |
| 01:04.09 | gusto | WIMPy: what CPU time? we are talking about a dude connecting some ATA's here, that's no load |
| 01:04.42 | WIMPy | Do you know where he runs his local Asterisk? |
| 01:05.01 | gusto | i did hardcore bullshit transcoding on my wndr3800 router and that CPU load did not even move |
| 01:05.29 | gusto | and i expect the same on a cheap DIR300 (have some of them as well) |
| 01:06.30 | gusto | but you know there is memory load and storage space (a few KB only free) a bigger issue than CPU |
| 01:07.38 | gusto | however ... my DIR300's are cool! |
| 01:10.03 | gusto | btw. since when they shut down the facebook ipv6 DNS records? |
| 01:14.24 | gusto | ehm |
| 01:19.01 | *** join/#asterisk gusto (~gusto@adsl-dyn-238.95-102-146.t-com.sk) |
| 01:19.36 | gusto | so, better |
| 01:38.27 | gusto | WIMPy: are you still here? |
| 01:39.00 | WIMPy | yes |
| 01:41.04 | *** join/#asterisk bmg505 (~leon@196-209-7-108.dynamic.isadsl.co.za) |
| 01:41.42 | gusto | you see |
| 01:41.49 | gusto | what do you do so late at night? |
| 01:42.30 | WIMPy | checking out a fresh linux. |
| 01:43.36 | gusto | what linux? |
| 01:43.53 | gusto | i am watching VOX as long as i have DVB-S here in slovakia |
| 01:43.53 | WIMPy | linux-stable |
| 01:44.04 | gusto | you mean a new kernel? |
| 01:44.32 | WIMPy | No, not new. Just a fresh checkout. |
| 01:44.46 | WIMPy | Ich have a feeling that git scewed something. |
| 01:44.55 | gusto | hmm |
| 01:45.14 | gusto | i checked out from git shit out when kernel.org was down |
| 01:45.53 | gusto | i am glad that kernel.org is back online again, because these git bullshit may be good for developers but not for simple downloading of a current snapshot |
| 01:46.10 | WIMPy | Why not? |
| 01:46.54 | gusto | heh, i am used to just go to the kernel.org website and download the current snapshot or the latest kernel.tar.bz2 |
| 01:47.20 | gusto | when i want to ... but lately it makes no sense any more as long as you have current kernels in distributions already |
| 01:47.32 | gusto | Linux romy.gusto 3.6.10-4.fc18.x86_64 #1 SMP Tue Dec 11 18:01:27 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux |
| 01:47.38 | WIMPy | If they are clean... |
| 01:47.53 | gusto | i have no problems |
| 01:48.07 | gusto | it's fedora ... not debian :-D LOL |
| 01:48.27 | WIMPy | I was reminded that issues with distro kernels are not a thing of the past, as I had thought. |
| 01:48.58 | gusto | i know |
| 01:49.03 | gusto | but fedora is OK |
| 01:52.03 | gusto | centos is different, they have patched in a lot ... so ... either it works or you have to compile an own kernel |
| 01:52.06 | SeRi | Penguin: |
| 01:52.44 | WIMPy | I prefer to DIY anyway. |
| 01:53.11 | WIMPy | It has to run on my hardware, not on anything else. |
| 01:55.40 | SeRi | waz up WIMPy |
| 01:55.54 | carrar | write the kernel from scratch! |
| 01:57.54 | gusto | and what distribution is it on? slackware? |
| 01:58.10 | WIMPy | yes |
| 02:01.22 | gusto | really? slackware? :-D LOL |
| 02:04.12 | gusto | i tired that distro some weeks ago as well, i had a problem with acpid taking too long on boot |
| 02:06.22 | gusto | but apart from that i like the completeness of its tools when you do full install, everything is there, what on distro's isnt the case, you always need to install something additionally |
| 02:07.34 | WIMPy | Isn't it great if yu even have to install ssh manually? |
| 02:08.51 | gusto | that does not happen to me |
| 02:09.13 | WIMPy | I hope it never happens to me again. |
| 02:09.31 | gusto | but there is a lot of bullshit that is useful and you need to install it because it's not there when you need it |
| 02:09.32 | WIMPy | But unfortunatly I don't remember which distro that was. |
| 02:09.43 | gusto | the most popular example to me is tcpdump |
| 02:10.11 | gusto | debian, before they included it into tasksel as an option to choose *ssh-server |
| 02:10.30 | carrar | ssh is usually out of date anyways |
| 02:10.57 | gusto | that's why i like BSD's because there you also have everything included of those cli tools |
| 02:11.08 | gusto | and slackware is here a little more BSD like |
| 02:11.12 | gusto | in this matter |
| 02:11.37 | gusto | well, the current SSHd should be on OpenBSD |
| 02:12.47 | gusto | but having openssh outdated is not that big of a problem, because that just needs to work, but having an outdated PF firewall on current FreeBSD (9.1) is far worse |
| 02:14.16 | gusto | last time i was trying to configure it and he was keeping on throwing syntax errors against me ... i really spend some time searching for misspelled words or such until i realised that i am looking at the current documentation having an outdated pf included inside my freebsd system |
| 02:18.40 | gusto | i can describe that feeling quite well by creating an analogy ... an analogy (like RMS would say) between RTL-nitro and this firewalls software versions ... RTL-nitro is a new TV channel from RTL group in germany and they show old TV-shows like knight rider with this super car KITT and watching them jumping over other cars and doing videotelephoning while being under way with 180 mph BUT having to stop and find a public telephone cell and a |
| 02:18.55 | *** join/#asterisk csshih (~craig@unaffiliated/csshih) |
| 02:19.22 | csshih | 'ello all |
| 02:19.46 | gusto | hi |
| 02:19.57 | gusto | whats new? |
| 02:24.54 | csshih | gusto, not much. I'm new here |
| 02:25.18 | gusto | you see |
| 02:26.15 | gusto | everyone is new to something some time |
| 02:27.07 | gusto | i am using asterisk for a year so i am not very long with asterisk either, i may celebrate maybe in 2 months the first anniversary of me with asterisk :D |
| 02:28.01 | gusto | what's new to me is that i am using asterisk in jail(s) from this year on |
| 02:34.54 | *** join/#asterisk slidesinger (~slidesing@c-69-141-208-250.hsd1.nj.comcast.net) |
| 02:41.18 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-197-138.red.bezeqint.net) |
| 02:45.04 | WIMPy | Oh, BTW... |
| 02:45.18 | WIMPy | leifmadsen: DIALPLAN_EXISTS does work with switches. |
| 02:45.19 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 02:51.43 | csshih | gusto, oh? |
| 02:53.26 | SeRi | Penguin: I am in the market for a HAM Radio. |
| 02:55.05 | gusto | why? |
| 02:56.03 | gusto | so ... i am going to play with my soft toys and sleep then ;-) |
| 03:02.40 | zerohalo | just sold my ft-817 |
| 03:02.47 | zerohalo | already miss it |
| 03:11.22 | Penguin | seri: Why? |
| 03:11.49 | SeRi | Penguin: Is inexpensive and interesting. |
| 03:14.07 | Penguin | You sure about that? Amateur radio equipment is often quite expensive. |
| 03:14.24 | SeRi | Penguin: did you hooked up your earpiece |
| 03:14.33 | Penguin | It's also weird that you mention buying... I have one on ebay right now. |
| 03:14.41 | Penguin | Not yet. |
| 03:14.43 | zerohalo | yeah - it's not cheap. doesn't depreciate much either. :) |
| 03:14.43 | SeRi | Penguin: I was looking a portable one and was right around 150+ |
| 03:14.53 | SeRi | Penguin: link me |
| 03:14.55 | zerohalo | SeRi: for what? |
| 03:14.59 | Penguin | What bands are you looking to get on? |
| 03:15.21 | SeRi | Penguin: I dont know what I am doing I just want to start somewhere |
| 03:15.41 | zerohalo | Well, 2m/440 radios can be pretty cheap |
| 03:16.02 | zerohalo | but for anything longer-wave, it's gonna get pricey |
| 03:16.07 | Penguin | I was helping a guy the other day with his Baofeng dual-band hand held. They run about $50 or less new. |
| 03:16.18 | Penguin | They're neat. |
| 03:16.25 | SeRi | Penguin: jump in |
| 03:16.54 | Penguin | It also depends on what license class you intend to obtain. |
| 03:17.00 | zerohalo | indeed |
| 03:17.15 | Penguin | If you don't go to at least a General, you won't have a lot of use in an HF rig. |
| 03:17.59 | Penguin | Using 200 khz out of the 10m band for voice doesn't really warrant a $500-$1000 radio. |
| 03:18.51 | Penguin | I'm also considering selling my HF radio, but I haven't committed to that yet. |
| 03:19.13 | zerohalo | Penguin: what ya got? |
| 03:19.31 | Penguin | Kenwood TS-440S/AT |
| 03:19.38 | zerohalo | oh, nice. |
| 03:19.49 | Penguin | It's an old model, but they are premium radios. |
| 03:19.57 | carrar | HF 160 meters for good times |
| 03:20.06 | Penguin | I never hear anyone on 160. |
| 03:20.10 | SeRi | Penguin: I guess I have to a lot of research |
| 03:20.12 | carrar | I do |
| 03:20.14 | Penguin | They're always on 80, 40, and often 20. |
| 03:20.30 | SeRi | but what i do know is that I want a portable |
| 03:20.35 | SeRi | to have at home and my car |
| 03:20.36 | carrar | Are Bell and their enhanced bass too wide of BW friends |
| 03:20.39 | carrar | art |
| 03:20.44 | carrar | heh |
| 03:20.47 | zerohalo | I've been lusting after a Yaesu ft-2000 for a while... but no time anymore |
| 03:20.48 | Penguin | Sounds like you'll want a 2m or 70cm radio. |
| 03:20.49 | carrar | on 80 |
| 03:21.01 | carrar | 75 actually |
| 03:22.10 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 03:22.39 | zerohalo | 2m/440 is a good intro, specially if you have active repeaters in the area |
| 03:22.48 | Penguin | I only have a wire dipole up right now, and it won't tune on 12 or 10, but it does okay on 80m-10m. |
| 03:22.51 | Penguin | err |
| 03:23.05 | SeRi | Penguin: http://www.ebay.com/itm/Baofeng-UV-5RA-Ham-Two-Way-Radio136-174-4-00-480-MHz-Amateur-Radio-Transceiver-/370656036338?pt=US_Ham_Radio_Transceivers&hash=item564cd245f2 |
| 03:23.07 | Penguin | 80m-15m. |
| 03:23.52 | Penguin | That's what I was talking about. |
| 03:24.07 | Penguin | He had the 3R, but the 5R is a little better for a few reasons. |
| 03:24.48 | Penguin | I think I bought my handheld from that seller. |
| 03:25.21 | Penguin | Mine is a Wouxun, though. |
| 03:27.54 | Penguin | Basically you have already decided you want a hand held, so 2m, 1.25m, and 70cm are probably the best for that. |
| 03:28.15 | Penguin | I'm sure you have repeaters in the area. |
| 03:28.51 | zerohalo | SeRi: where are you? |
| 03:28.52 | Penguin | I don't have a repeater book handy, or I'd look it up. |
| 03:29.07 | SeRi | zerohalo: Texas |
| 03:29.07 | Nugget | Don't mess with Texas. |
| 03:29.13 | SeRi | Nugget: lol |
| 03:29.14 | Penguin | I haven't bought one in probably 15 years. |
| 03:29.47 | SeRi | Penguin: you saw the link? |
| 03:30.12 | Penguin | I remarked. |
| 03:30.16 | zerohalo | Nice - lived in Dallas for a while and it had a great local community |
| 03:30.38 | zerohalo | http://www.repeaterbook.com/repeaters/msResult.php?state_id%5B%5D=48&band=14&freq=&loc=&call=&features=&coverage=%25&status_id=%25&order=%60freq%60%2C+%60state_abbrev%60 |
| 03:30.39 | SeRi | zerohalo: I have several collegues that are in to HAM |
| 03:31.08 | SeRi | Penguin: ah! |
| 03:31.09 | Penguin | I guarantee you have at least one 2m repeater in range. |
| 03:31.10 | SeRi | Got it |
| 03:31.55 | zerohalo | Moving to Boston was the worst thing to happen to my hamventures... antenna restrictions - local noise... only 2m was active, mostly because of the MIT club |
| 03:32.49 | SeRi | Penguin: so I can go with a 5R instead.... I can search for one |
| 03:33.07 | Penguin | You just showed me a 5R. |
| 03:33.15 | SeRi | o |
| 03:33.16 | SeRi | lol |
| 03:34.02 | SeRi | Penguin: anything else that I would need? |
| 03:34.12 | Penguin | You've even got a 6m repeater in Channelview. |
| 03:34.31 | Penguin | Yes, you'll need a license. |
| 03:34.39 | zerohalo | hah - that |
| 03:34.44 | SeRi | lol |
| 03:34.51 | carrar | EASY |
| 03:34.53 | zerohalo | Technician is easy tho |
| 03:34.55 | carrar | hamtestonline.com |
| 03:35.05 | carrar | 100% guarented |
| 03:35.13 | carrar | or money back! |
| 03:35.19 | SeRi | damn the item is in hongkong |
| 03:35.26 | Penguin | You've got a lot of 2m and 440 repeatres close to you. |
| 03:35.30 | carrar | Extra isn't that hard |
| 03:35.37 | Penguin | seri: I got mine from that seller in less than one week. |
| 03:35.44 | zerohalo | or just http://www.arrl.org/question-pools for free |
| 03:35.46 | SeRi | Penguin: awesome |
| 03:35.50 | SeRi | so license first? |
| 03:35.54 | carrar | yes |
| 03:35.57 | Penguin | I'd recommend it. |
| 03:36.03 | SeRi | ok so let me focus on that |
| 03:36.09 | Penguin | It'll be too tempting to bootleg if you have the radio. |
| 03:36.24 | SeRi | free bumber sticker! |
| 03:36.26 | SeRi | I am sold! |
| 03:36.27 | SeRi | lol |
| 03:36.36 | SeRi | Penguin: lol |
| 03:36.37 | carrar | serious, they make it so easy |
| 03:36.52 | SeRi | carrar: I am at the site |
| 03:38.44 | carrar | Oh nice they broke it up, used to be like $50 for everything |
| 03:38.52 | Penguin | What?! |
| 03:39.07 | Penguin | Just study the materials for free and take the test for $15. |
| 03:39.11 | carrar | naw |
| 03:39.14 | carrar | do the online |
| 03:39.16 | carrar | it's fast |
| 03:39.26 | carrar | well money spent |
| 03:39.30 | Penguin | You can use the study guide and sample tests on qrz.com and learn it all. |
| 03:40.32 | carrar | or you can just do the website that learns from your answers |
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| 03:42.09 | SeRi | lol |
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| 04:47.28 | jpsharp | Yes, the world needs more hams :0 |
| 04:47.29 | jpsharp | :) |
| 04:49.06 | Penguin | Full saturation! |
| 04:55.53 | jpsharp | Until you can't find an open slot on the PSK31 band. |
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| 10:13.31 | krotos | hi all |
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| 11:25.43 | krotos | there are some ways to use a BLF on a Phone for notify the entering on a dinamic queue |
| 11:25.47 | krotos | ? |
| 11:31.07 | ectospasm | I have a colleague that has rigged his line key to glow different colors depending on whether he's logged into the queue or not... unfortunately I don't know how he did it |
| 11:33.05 | Maliuta | ectospasm: ask for the script :P |
| 11:34.29 | ectospasm | Maliuta: I think this is Switchvox, so he did something in his extension config... I'll have to ask him Wednesday when I come in |
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| 11:57.58 | krotos | i'm was thinking about a script |
| 11:58.05 | krotos | but no idea for do that |
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| 12:14.38 | ectospasm | ...or he may have tweaked the phone firmware |
| 12:21.36 | ghost75 | jpsharp? |
| 12:22.27 | krotos | ectospasm: in my case the phone is a Grandstream GXP2120 |
| 12:22.49 | krotos | i'had an idea ( using the line2-lamp) and creating a "fake" sip user |
| 12:23.08 | krotos | and via dialplan change the user password so the lamp become red or green |
| 12:23.19 | krotos | but is not a very user friendly solutions |
| 12:23.38 | WIMPy | And will take lots of time to show changes. |
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| 12:24.00 | krotos | exactly |
| 12:24.44 | krotos | the second idea was to use a in-line sipuac and call conference roonm, so the in-line uac become RED on BLF |
| 12:24.57 | krotos | and at logout time, return green |
| 12:25.07 | WIMPy | That's what custom devstates are for. |
| 12:27.45 | krotos | something like exten => 12345,hint,Custom:dnd123456 |
| 12:27.49 | krotos | and next set DEVSTATE |
| 12:27.50 | krotos | ? |
| 12:29.26 | WIMPy | yes |
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| 12:55.28 | Schabo | Need some advice, what would be the best way to access asterisk events from a webclient in realtime? I would assume AMI but I'd prefer not to open that interface to the public |
| 12:56.45 | Schabo | Also, are there any plans to make the manager events accessible through WebSockets? |
| 12:57.20 | WIMPy | Yes, and no, might not want want to do that. |
| 12:57.48 | WIMPy | THE AMI USERS DO HAVE PERMISSIONS, HOWEVER. |
| 12:57.53 | WIMPy | oops |
| 12:57.57 | Schabo | whoa :) |
| 12:58.34 | Schabo | So it would be "safe" to setup a AMI user with only read access only to the needed events? |
| 12:59.25 | WIMPy | You can do read-only. But you can't nail it down to certain events. |
| 13:00.00 | WIMPy | But you want to have some application in between, I hope? |
| 13:00.45 | Schabo | I am still in the planning state of an WebRTC phone client, looking over the options |
| 13:01.21 | Schabo | as I said, need some advice! :) |
| 13:01.47 | Schabo | What sort of application are you referring to? Like a AMI proxy of some sort? |
| 13:02.10 | WIMPy | No, I was hoping for a CGI type thing. |
| 13:02.47 | Schabo | I'd really prefer WebSockets so I've been looking at "AsterClick" |
| 13:03.27 | WIMPy | Yes, that's probably the future. |
| 13:03.44 | WIMPy | But nothing I'd feel comfortable with. Neither on the server nor on the client side. |
| 13:04.12 | Schabo | AC seems to be exposing the AMI events to unauthorized connections, so not going that route at the moment |
| 13:06.06 | Schabo | What I really need is a Web based phone utility, for recieving/making calls, see calls in queue, logged in peers in real-time and some other call handling features |
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| 13:11.29 | Schabo | So you are thinking something like FastAGI => "somescript" => Push events to client? |
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| 14:07.17 | verywiseman | i have pri , and i install digital card on my asterisk , how can i generate a call from asterisk cli through pri ? |
| 14:07.37 | WIMPy | channel originate |
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| 14:20.34 | hugo | hey I am trying to compile the srtp module manually on netbsd as the package in pkgsrc doesn't include it. so I extracted (make extract in comms/asterisk) and ran configure then gmake menuselect, but srtp shows as XXX, which according to help means conflict or missing dep |
| 14:20.50 | hugo | but /usr/include/openssl/srtp.h is in my system.. what else would it need? |
| 14:21.59 | WIMPy | libsrtp |
| 14:23.32 | hugo | right |
| 14:31.57 | gusto | ah |
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| 14:32.43 | leifmadsen | WIMPy: huh, well that's interesting (re: DIALPLAN_EXISTS() ) |
| 14:35.02 | WIMPy | That reminds me that I should find out if it REALLY works :-) |
| 14:38.08 | WIMPy | Yes, it does. |
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| 14:38.28 | WIMPy | Not all surprises are bad ones :-) |
| 14:38.49 | WIMPy | And now I remember that I alread did that testing. |
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| 14:51.58 | leifmadsen | ya you said it worked last night :) |
| 14:52.42 | WIMPy | Ja, it got too late again... |
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| 15:02.51 | hugo | http://pastebin.com/w61mEsTM |
| 15:03.08 | hugo | out of ideas |
| 15:03.48 | leifmadsen | check the configure log and find out what it is seeking that it isn't finding |
| 15:04.09 | leifmadsen | and now I'm off to find coffee |
| 15:05.33 | hugo | configure:52463: gcc -o conftest -g -O2 conftest.c -lsrtp -lm >&5 |
| 15:05.33 | hugo | ld: cannot find -lsrtp |
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| 15:12.16 | linocisco | I have Nortel BCM450 in use at work. I would like to know how to mix with asterisk. as I dont have lincense for sip on BCM450 |
| 15:12.41 | linocisco | meaning peaceful coexistance with Nortel BCM450 and Asterisk together without any problem |
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| 15:23.51 | ghost75 | park ( CHANNEL, CHANNEL2 [, PARKTIME, TIMEOUT ] ) <- any idea what could be channel2 in this case if only source channel exists? |
| 15:24.06 | ghost75 | from: http://search.cpan.org/~greenbean/Asterisk-AMI/lib/Asterisk/AMI/Common.pm#Methods |
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| 15:33.12 | hugo | ok so I'm trying to convert libsrtp.a to libsrtp.so with ar -x libsrtp.a && gcc -shared *.o -o libsrtp.so but: ld: aes.o: relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC |
| 15:33.12 | hugo | I did try with -fPIC, same thing |
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| 15:51.45 | hugo | libsrtp.so: ELF 64-bit LSB shared object, x86-64, version 1 (SYSV), dynamically linked, for NetBSD 6.0, not stripped |
| 15:51.47 | hugo | finally. |
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| 16:05.25 | hugo | http://pastebin.com/zcqtF8qy .. any idea? |
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| 16:09.14 | hugo | I tried adding -L/usr/pkg/lib in SRTP_LIB @ makeopts but that didn't change anything |
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| 16:09.34 | linocisco | hi all |
| 16:14.50 | ghost75 | is GROUP_COUNT working in h context? |
| 16:15.05 | ghost75 | exten, not context |
| 16:21.31 | [TK]D-Fender | ghost75: Do you see it not working? |
| 16:21.58 | ghost75 | i remember it was not working but dont know why |
| 16:22.45 | ghost75 | ExecIf($[${GROUP_COUNT(group1)} < 1 ]?Gosub(something,s,1)) |
| 16:24.05 | ghost75 | in h is it still in group or already out? |
| 16:24.51 | ghost75 | i want to do "something" if nobody is on the phone |
| 16:28.25 | [TK]D-Fender | I'f you're in "h" then whatever device that originated the call is no longer in it... |
| 16:29.10 | ghost75 | then logically it should work |
| 16:29.31 | [TK]D-Fender | Easy to confirm if you have 1 in the group already, join the group (2), and then check in "h" and end up with 1 |
| 16:31.15 | SeRi | is g729 that much superior that g711? |
| 16:31.25 | SeRi | /s/that/than/ |
| 16:31.43 | SeRi | fail |
| 16:32.23 | [TK]D-Fender | SeRi: Who told you it was "superior"? |
| 16:32.30 | [TK]D-Fender | SeRi: In what way? |
| 16:32.58 | SeRi | No body has. I am just seen a flux of questions of people using it over g711 |
| 16:33.14 | ghost75 | because less bandwidth |
| 16:33.16 | SeRi | s/flux/influx/ |
| 16:33.49 | SeRi | ghost75: I know that part but thats the only thing? |
| 16:33.58 | [TK]D-Fender | SeRiis g729 that much superior that g711? SeRiNo body has |
| 16:34.01 | ghost75 | think so |
| 16:34.29 | SeRi | [TK]D-Fender: *is* in form of a question |
| 16:34.59 | [TK]D-Fender | SeRi: G.729 uses less bandwidth. the cost is quality |
| 16:35.05 | SeRi | ghost75: I see. |
| 16:35.09 | SeRi | [TK]D-Fender: ah! |
| 16:35.12 | SeRi | ok I see. |
| 16:35.30 | [TK]D-Fender | People use G.729 because they have to. |
| 16:35.47 | SeRi | I see. That makes sense. |
| 16:37.22 | ghost75 | i could use g729 and have crappy line but even in my case doesnt make much sense |
| 16:37.43 | ghost75 | almost all calls come in with g711 |
| 16:38.26 | SeRi | ghost75: it all depends on your itsp |
| 16:38.52 | ghost75 | both codecs are allowed but if somebody calls me with g711 i cannot use g729 |
| 16:39.57 | ghost75 | i call myself from cellphone then g729 is used |
| 16:40.48 | ghost75 | if i call somebody (voip based i guess) who doesnt have g729 then i also cannot use g729 |
| 16:43.16 | ghost75 | i think this is because i use only sip accounts on my itsp |
| 16:46.47 | [TK]D-Fender | Cannot? |
| 16:47.05 | [TK]D-Fender | You know Asterisk does this wonderful thing called TRANSCODING |
| 16:47.30 | ghost75 | g711 is used over wan |
| 16:48.01 | ghost75 | of course i could use g729 from phone to asterisk, but what sense this makes? |
| 16:55.31 | [TK]D-Fender | Inside a LAN it DOESN'T make sense |
| 16:55.40 | [TK]D-Fender | So you use it on the side that does |
| 16:57.55 | ghost75 | if the remote party doesnt support it, i cannot use it |
| 16:58.34 | [TK]D-Fender | ghost75: The remote party is talking to asterisk. Asterisk uses whatever you tell it to use to talk to them. What are you going on about? |
| 16:59.49 | ghost75 | how they can talk to me if i disable g711 and they dont have g729 ;) |
| 17:00.50 | [TK]D-Fender | What you do on your phone has nothing to do with how * talks to the other party |
| 17:01.20 | ghost75 | i am talking about sip peer settings |
| 17:01.46 | [TK]D-Fender | You do not seem to get it |
| 17:01.51 | [TK]D-Fender | Your phone talks to asterissk |
| 17:01.59 | [TK]D-Fender | Asterisk talks to the other side |
| 17:02.10 | [TK]D-Fender | Asterisk in is the MIDDLE |
| 17:02.15 | ghost75 | its not about phone its about peer |
| 17:02.20 | [TK]D-Fender | ASTERISK TRANSCODES CALLS WHEN THE CODECS AREN'T THE SAME |
| 17:02.24 | ghost75 | asterisk<->itsp |
| 17:02.46 | [TK]D-Fender | I ahve an ITSP that uses iLBC. My phone doesn't talk iLBC at all. TI WORKS. |
| 17:02.56 | [TK]D-Fender | Asterisk TRANSCODES THE CALL |
| 17:03.04 | ghost75 | its not related to phone at all |
| 17:03.08 | ghost75 | forget about phone |
| 17:03.21 | [TK]D-Fender | ghost75if i call somebody (voip based i guess) who doesnt have g729 then i also cannot use g729 |
| 17:03.31 | [TK]D-Fender | You calling someone else. |
| 17:03.33 | [TK]D-Fender | 2 parties |
| 17:03.53 | ghost75 | if i have only g729 enabled on sip peer |
| 17:04.11 | [TK]D-Fender | Obviously * has to agree with EACH END |
| 17:04.20 | [TK]D-Fender | \but multiple parties don't have to agree |
| 17:05.15 | ghost75 | the thing is, as soon i enable g711 on sip peer, almost all calls come in with g711 even if the caller has g729 available (but he has g711 preferred) |
| 17:05.55 | [TK]D-Fender | change your preference order if that's what you want to do |
| 17:06.17 | ghost75 | didnt worked |
| 17:07.04 | ghost75 | allow i have g729 first |
| 17:07.33 | [TK]D-Fender | Where? |
| 17:07.39 | ghost75 | in sip peer |
| 17:07.45 | [TK]D-Fender | That's only HALF of the picture |
| 17:07.54 | [TK]D-Fender | Actually, one quarter |
| 17:08.04 | [TK]D-Fender | Maybe less depending what else you do with that call |
| 17:09.09 | ghost75 | in general section i have g729 on bottom |
| 17:09.52 | [TK]D-Fender | Doesn't mean much... |
| 17:10.14 | ghost75 | is there more to influence? |
| 17:10.45 | [TK]D-Fender | When your peer "disallow=all", and allows specific after that... and your peer is used ... the [general] does not matter. |
| 17:10.47 | [TK]D-Fender | At all |
| 17:10.53 | [TK]D-Fender | your PEER has those settings. |
| 17:11.08 | [TK]D-Fender | And there are TWO peers. Your inside leg. And your outside leg. |
| 17:11.15 | ghost75 | oh i thought general is then used for intern calls |
| 17:11.16 | [TK]D-Fender | EACH side has their own preference |
| 17:11.26 | [TK]D-Fender | And you aren't even looking at your single leg in it's entirety |
| 17:12.00 | [TK]D-Fender | ghost75oh i thought general is then used for intern calls <- Why do you think you set things in peers? [general] is for things that you don't set elsewhere (override), or don't MATCH a peer |
| 17:12.53 | ghost75 | k |
| 17:15.17 | ghost75 | is it possible to set codecs on inside leg other than setting it on every phone? |
| 17:15.36 | [TK]D-Fender | your phone IS the inside leg |
| 17:15.46 | ghost75 | yes |
| 17:16.04 | [TK]D-Fender | peers use what's in general UNLESS overriden |
| 17:16.19 | [TK]D-Fender | [general] is for things that you don't set elsewhere (override) <------ |
| 17:16.44 | ghost75 | that you explained already |
| 17:17.39 | [TK]D-Fender | Yes, and you still ask a question that shows you didn't understand it or didn't read it |
| 17:17.59 | ghost75 | nope |
| 17:18.05 | ghost75 | my question was something else |
| 17:20.23 | [TK]D-Fender | Rephrase it then |
| 17:21.53 | ghost75 | if its possible to determine codec order only for calls between phone and asterisk (other than setting codec order on each phone) |
| 17:22.17 | [TK]D-Fender | [general] <- |
| 17:22.23 | [TK]D-Fender | And you have to set it IN the phone..... |
| 17:22.38 | [TK]D-Fender | Hence your focus on only 1/4 (or less) of the big picture |
| 17:23.58 | ghost75 | so if the sip peer is NOT used then asterisk will also not use the codec order defined there |
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| 17:24.44 | [TK]D-Fender | ghost75: If your call doesn't match a peer ... then it doesn't matter what you have in 1000 peer entires you DO have. |
| 17:25.31 | ghost75 | k |
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| 17:59.39 | hugo | say I want to build res_srtp.so only and I have the source, what is the proper way? |
| 17:59.57 | hugo | something more sensible than unchecking all others in menuselect.. lol |
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| 18:04.43 | hugo | well, F7 helps |
| 18:07.27 | davlefou | hi, when i have twice channel for an call, how can have detail? |
| 18:07.45 | hugo | res/res_srtp.so: ELF 64-bit LSB shared object, x86-64, version 1 (SYSV), dynamically linked, for NetBSD 6.0, not stripped |
| 18:08.29 | davlefou | i use "core show channel", it work for the first but not for the second line! |
| 18:09.19 | davlefou | http://pastebin.com/97YqpWZy |
| 18:09.30 | hugo | res/res_srtp.so: |
| 18:09.30 | hugo | <PROTECTED> |
| 18:09.32 | hugo | lol damn it damn it |
| 18:09.52 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.200) |
| 18:15.05 | hugo | [Jan 5 18:14:24] WARNING[-1] loader.c: Module 'res_srtp.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 18:15.05 | hugo | [Jan 5 18:14:24] WARNING[-1] loader.c: Module 'res_srtp.so' will not be initialized as it may cause instability. |
| 18:15.05 | hugo | [Jan 5 18:14:24] WARNING[-1] loader.c: Module 'res_srtp.so' could not be loaded. |
| 18:15.06 | hugo | :( :( :( :( |
| 18:15.11 | verywiseman | i have Grandstream 503 , i can receive calls from pstn , but i can't send a call , i sniff the packets , and i noticed that ,when my sip client send a request , asterisk reply with "401 unauthorized (0bindings) " , what is that meaning? |
| 18:20.11 | ghost75 | allowguest on or off? |
| 18:23.19 | verywiseman | ghost75, i didn't set it |
| 18:25.33 | ghost75 | then no idea |
| 18:25.38 | davlefou | how i have more large with the full name with "core show channel". |
| 18:25.40 | davlefou | ? |
| 18:29.22 | [TK]D-Fender | davlefou: "core show channels concise" |
| 18:29.49 | davlefou | [TK]D-Fender: thanks! |
| 18:39.02 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 18:40.38 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 18:41.57 | verywiseman | Any help |
| 18:42.38 | [TK]D-Fender | verywiseman: Show actual * SIP debug and configs from both sides. |
| 18:42.50 | [TK]D-Fender | verywiseman: You are giving a loose and poorly worded description |
| 18:43.42 | verywiseman | [TK]D-Fender, ok , but forgive me , because my native language is not English :) |
| 18:44.11 | [TK]D-Fender | verywiseman: You stil didn't show either side in any detail. Language is only a small part of the problem. |
| 18:47.51 | verywiseman | [TK]D-Fender, this is sip debug http://fpaste.org/rqZB/ , and this is sip.conf http://fpaste.org/rQBP/ |
| 18:48.15 | verywiseman | [TK]D-Fender, Are you needing Grandstream conf ? |
| 18:50.09 | [TK]D-Fender | verywiseman: Do you somehow think that part is NOT important? |
| 18:50.51 | verywiseman | [TK]D-Fender, how can i give GS conf to you ? |
| 18:51.09 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.200) |
| 18:51.14 | [TK]D-Fender | verywiseman: How do YOU think? |
| 18:51.27 | verywiseman | [TK]D-Fender, Screenshot ? |
| 18:52.23 | [TK]D-Fender | verywiseman: Something we can see would be a good idea... |
| 18:55.37 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 19:09.07 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 19:09.12 | *** join/#asterisk uski (~uski@unaffiliated/uski) |
| 19:09.20 | uski | hi; can that sort of stuff work with asterisk ? http://www.ebay.com/itm/USB-Skype-Telephone-Call-PSTN-VoIP-ATA-Gateway-FXO-FXS-/320617076822 |
| 19:10.21 | verywiseman | [TK]D-Fender, http://postimage.org/gallery/v4rupws/1f427891/ |
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| 19:10.48 | WIMPy | uski: Possibly. But better get a bog standard USB sound "card". That's all it is. |
| 19:11.11 | uski | WIMPy, yea but I'd like to connect an existing analog phone to it, so i like the FXO interface |
| 19:11.37 | [TK]D-Fender | uski: No |
| 19:11.38 | WIMPy | Oh, missed that part. Well, not that pcertainly won;t work. |
| 19:12.26 | uski | (i meant FXS, i always mix the two) ok well i'll get a sipura adapter then |
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| 19:22.28 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:52.12 | [TK]D-Fender | verywiseman: What is that FXO port plugged into? |
| 19:53.35 | verywiseman | [TK]D-Fender, plugged into splitter , splitter to pstn |
| 19:53.57 | verywiseman | [TK]D-Fender, look , i can receive calls , but i can't send |
| 19:54.02 | [TK]D-Fender | verywiseman: in your SIP debug I do not see any answer tothe invite you sent out |
| 19:54.03 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.200) |
| 19:54.31 | [TK]D-Fender | verywiseman: If it's on a splitter, put an analog phone alongside it. pick up THEN do your callout again |
| 19:54.35 | [TK]D-Fender | see if you HEAR it trying |
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| 19:57.33 | verywiseman | [TK]D-Fender, sorry , but the splitter feed ATA and adsl modem , as i understand , i will unplug adsl modem and plug analog phone , is that right? |
| 19:57.56 | [TK]D-Fender | get another splitter. Whatever it takes to be able to lisen at the same time |
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| 20:04.31 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 20:05.16 | hugo | asterisk*CLI> show modules |
| 20:05.16 | hugo | No such command 'show modules' (type 'core show help show modules' for other possible commands) |
| 20:05.16 | hugo | asterisk*CLI> core show help show modules |
| 20:05.16 | hugo | No such command 'show modules'. |
| 20:05.21 | hugo | hm. |
| 20:07.12 | dpilon | module show ??? |
| 20:08.57 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 20:08.58 | ectospasm | module show like <module regexp> |
| 20:10.05 | hugo | asterisk*CLI> module show like srtp |
| 20:10.05 | hugo | Module Description Use Count |
| 20:10.05 | hugo | res_srtp.so Secure RTP (SRTP) 0 |
| 20:10.05 | hugo | 1 modules loaded |
| 20:10.05 | hugo | cool |
| 20:12.27 | hugo | chan_sip.c: sip_xmit of 0x7f7ff4977000 (len 634) to xx.xx.xx.xx:3390 returned -2: Undefined error: 0 |
| 20:12.33 | hugo | any idea? |
| 20:16.47 | ectospasm | hugo: what version of Asterisk? |
| 20:17.05 | hugo | asterisk-11.1.0nb1 The Asterisk Software PBX |
| 20:18.44 | ectospasm | the latest version in that branch is Asterisk 11.1.2... |
| 20:25.05 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:26.57 | gusto | yes |
| 20:27.05 | gusto | i am loading it now ;-) |
| 20:32.34 | gusto | freebsd has it in its ports |
| 20:33.37 | gusto | and i had to w8 quite long until i got connected to the server, it seems that they have a connection limit that was reached |
| 20:34.23 | gusto | i noticed it last time as well, that this asterisk project hosting servers seem to be under quite load expecially when a new release comes out |
| 20:35.16 | gusto | maybe they should consider putting that src tars on bittorrent |
| 20:40.27 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 20:41.56 | WIMPy | Yes. Torrents are a good idea. |
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| 21:37.52 | hugo | I've tried various combinations of nat, directmedia and tried setting media_address too |
| 21:38.14 | hugo | * is behind nat and so am I and the person I'm testing with (different nats) |
| 21:38.27 | hugo | the best I got so far is audio working (always) and video working on one side |
| 21:39.00 | hugo | I can see the edge fw on the asterisk server network blocking packets in the RTP port range addressed to the IP of my laptop behind my local NAT and I think that's why |
| 21:39.13 | hugo | would anyone find that familiar and know the right settings ? |
| 21:39.35 | hugo | I've tried everything and read about it in several places but nothing seems to work, as far as I understand it nat=yes and directmedia=no should be enough |
| 21:39.53 | [TK]D-Fender | Nowhere near. |
| 21:40.04 | [TK]D-Fender | You need to specify your WAN IP, local subnets, etc. |
| 21:40.47 | hugo | also did |
| 21:40.59 | hugo | sorry forgot to mention that |
| 21:41.27 | [TK]D-Fender | I'd start by looking at the calls.... |
| 21:41.58 | hugo | peers also have nat=yes |
| 21:42.02 | hugo | and qualify=1000 |
| 21:43.28 | hugo | what do yo mean exactly [TK]D-Fender ? |
| 21:43.36 | hugo | I'm tailing the log and connected to the cli |
| 21:48.26 | hugo | in sip show peers is Forcerport active when the a is in CAPS or lower? |
| 21:59.38 | [TK]D-Fender | * CLI w/ SIP DEBUG enabled. Accept no substitutes |
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| 22:05.27 | hugo | nice |
| 22:05.29 | hugo | this is what I need |
| 22:05.34 | hugo | I'll look into it for awhile |
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| 23:07.06 | ruied | is there any software for asterisk to integrate with front-office software for hotels, to integrate options like check in check out, room status, minibar, display the name of the client at the room? |
| 23:11.14 | saint_ | ruied: nah, but you can make one using a web interface.. or i can make it for you and you pay me :) |
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| 23:19.34 | ruied | saint_, thanks! it's a possibility... ;) the main problem to implement asterisk in hotels is the integration with front-office hotel's software, like alcatel siemens and others have |
| 23:30.36 | saint_ | ruied: i can do alcatel. i don t know about the users. but then, if you have asterisk, why would you keep alcatel or siemens or cisco or avaya or nortel ..? |
| 23:34.53 | leifmadsen | ruied: well, it's definitely possible -- you just need to build the part that glues your front office software into asterisk via AMI and AGI |
| 23:38.55 | [TK]D-Fender | saint_: He isn't asking about keeping an Alcatel, Seimens, or Cisco system.... |
| 23:39.54 | hugo | We are requesting SRTP for audio, but they responded without it! <--- encryption is set to no in sip.conf, shouldn't this make it *optional* ? |
| 23:41.02 | WIMPy | encryption=SIPS |
| 23:41.37 | WIMPy | The one you've been fighting so hard for. |
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| 23:43.53 | ruied | <PROTECTED> |
| 23:48.55 | saint_ | ruied: but once again.. if you put asterisk, are you keeping the other phone sytem, or are you replacing everything with asterisk ? |
| 23:49.40 | ruied | saint_, replace everything |