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| 00:57.11 | Gteste | Hi! How can I set the expiry timeout per provider when registering them? |
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| 01:02.05 | Gteste | Any idea? |
| 01:14.55 | Gteste | Hi! How can I set the expiry timeout per provider when registering them? |
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| 01:26.43 | Gteste | Bye |
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| 01:35.25 | k-man | morning all |
| 01:36.24 | k-man | is there some way in asterisk to play a sound when when a call connection terminates, like when someone hangs up on the other end of the call |
| 01:40.55 | WiretapWork | k-man, play a sound before running the Hangup() command |
| 01:41.53 | k-man | WiretapWork: ah, thanks |
| 01:50.10 | rue_mohr | is there a make command for a debain system to get the boot init script for asterisk installed? |
| 01:50.29 | rue_mohr | README says nothing about it |
| 01:50.46 | rue_mohr | or if they do, they manage to not use the word boot or init |
| 01:50.52 | rue_mohr | or startup |
| 01:51.06 | puzzled | rue_mohr: check the main Makefile. iirc there is some init stuff in there |
| 01:51.40 | rue_mohr | I found the template, I dont know what triggers its install |
| 01:51.58 | rue_mohr | the last system I did had dahdi in it, and its asterisk starts on boot |
| 01:52.42 | puzzled | rue_mohr: so are you on debian or centos? |
| 01:52.54 | rue_mohr | debian, self compiled asterisk |
| 01:54.30 | rue_mohr | I'm just confused why the last install did it and this one didn't |
| 01:54.58 | rue_mohr | maybe its dahdi-tools that actually sets up asterisk to start on boot? |
| 01:55.01 | puzzled | rue_mohr: looking at the main Makefile, make config will install the proper init depending on the distro |
| 01:55.09 | pushpop | anyone ever purchase a Snom phone? Do they usually come with the power adapters? |
| 01:55.10 | puzzled | rue_mohr: no, dahdi does not do that |
| 01:55.25 | rue_mohr | huh |
| 01:55.29 | rue_mohr | sure I did that |
| 01:55.47 | puzzled | rue_mohr: check the main config from around line 744 or search for "config:" |
| 01:55.55 | rue_mohr | :) I guess I hadn't, thanks |
| 01:56.04 | puzzled | elif [ -f /etc/debian_version ] ; then \ |
| 01:56.15 | rue_mohr | make config worked, thankyou |
| 01:56.15 | puzzled | and then it installs the debian init stuff |
| 01:56.30 | puzzled | my pleasure |
| 01:57.32 | rue_mohr | :) see if I can get mgcp working on this vog4000 again |
| 01:57.44 | rue_mohr | .. I should be able to, I wrote the book... |
| 01:58.57 | puzzled | rue_mohr: heh that sounds ancient |
| 01:59.14 | puzzled | http://www.voip-info.org/wiki/view/VoipPack+VOG+4000+and+asterisk |
| 02:01.48 | puzzled | lol 3cx whitepaper with 10 advantages of a Windows PBX: no need to learn how to update & troubleshoot black box, no need for linux updates and re-testing. as if winblos does not need updates. guess you need to come up with some serious marketing bs to sell that stuff |
| 02:02.03 | puzzled | http://www.3cx.com/blog/voip-knowhow/windows-pbx-advantages/ |
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| 02:13.40 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:18.07 | xxiao | when i make a SIP call, will the RTP packets go through SIP gateway or bypass it? |
| 02:23.26 | *** join/#asterisk c|oneman (clon3man@206.248.132.145) |
| 02:23.52 | c|oneman | when you've got 2 devices logged in to a VOIP service, how are incoming calls typically handled? |
| 02:29.03 | puzzled | xxiao: that depends on how you have * configured. it can do either. search for canreinvite and something called rtpdirect or directrtp on the asterisk wiki |
| 02:29.51 | xxiao | puzzled: thanks. i thought SIP is really only about signalling, but one peer is saying he saw all RTP packets originating from sip server |
| 02:30.05 | xxiao | guess it can do both, but let RTP go throught sip can not scale |
| 02:30.24 | xxiao | sip is new to me, googling&learning |
| 02:30.32 | puzzled | xxiao: signalling and data is separate but an * box can handle both |
| 02:31.09 | xxiao | puzzled: got it, for internal usage i guess it's fine, for public access it could be separated for scalability i think |
| 02:31.23 | xxiao | http://en.wikipedia.org/wiki/Session_Initiation_Protocol |
| 02:31.35 | xxiao | clearly RTP path is separated from SIP |
| 02:31.37 | puzzled | xxiao: yes, for large scale stuff look at OpenSIPS and Kamailio |
| 02:32.27 | xxiao | puzzled: thanks again, that's what i'm trying to figure out: opensips, kamailio, asterisknow, trixbox, pbx in a flash |
| 02:32.42 | xxiao | i guess asterisk* is really for intranet? |
| 02:32.52 | xxiao | as what PBX stands for, "private" |
| 02:33.14 | puzzled | yeah but that term is ancient |
| 02:34.03 | puzzled | opensips and kamailio are proxies. asterisk, asterisknow, trixbox and pbx in a flash are basically all a pbx |
| 02:34.11 | xxiao | so, for intranet/enterprise internal usage, asterisk/trixbox fits, for public server setup opensips/kamailio are btter? does this make sense? |
| 02:34.35 | ectospasm | no |
| 02:34.45 | ectospasm | Asterisk can be a public server |
| 02:34.52 | xxiao | ok, other than these famous names, did I miss any major player in sip/voip(oss)? |
| 02:34.54 | ectospasm | ...it depends on what your needs are |
| 02:35.01 | puzzled | nods |
| 02:35.24 | ectospasm | xxiao: FreeSwitch, OpenSER come to mind. |
| 02:35.27 | puzzled | xxiao: other players are yate and freeswitch |
| 02:35.30 | xxiao | i played with asterisk a few years ago, which is hard to use, need re-try it now, this weekend actually |
| 02:35.32 | ectospasm | unless you've already mentioned them |
| 02:35.42 | xxiao | need set up a testing server for someone |
| 02:35.49 | ectospasm | xxiao: but none of them are drop in replacements for each other |
| 02:36.10 | xxiao | ectospasm: i was told freeswitch is totally different |
| 02:36.15 | puzzled | it is |
| 02:36.17 | ectospasm | well, Asterisk-based stuff like AsteriskNOW, Trixbox, Elastix, etc. are drop-in replacements for each other |
| 02:36.26 | ectospasm | xxiao: it is, as are most of what you're talking about |
| 02:36.30 | xxiao | openser is nowadays opensips/kamailio i think |
| 02:36.34 | ectospasm | FreeSwitch is not a PBX |
| 02:36.41 | ectospasm | xxiao: yeah, I don't keep up with it |
| 02:36.57 | ectospasm | nor is OpenSIPS/Kamailio |
| 02:37.03 | puzzled | well, it can be but basically it is a switch/framework |
| 02:37.43 | ectospasm | well, in that respect Asterisk isn't a PBX either |
| 02:37.46 | xxiao | if i want to set up a sip-video-chat network in public, which one will fit the best before i test all of them? |
| 02:37.47 | ectospasm | it's a PBX toolkit |
| 02:38.00 | ectospasm | xxiao: I can't answer that |
| 02:38.06 | ectospasm | you'd have to do your testing first |
| 02:38.13 | puzzled | heh good luck with video. not sure if any of them support that very well |
| 02:38.51 | xxiao | all claimed so though :0 |
| 02:38.53 | xxiao | :) |
| 02:39.33 | puzzled | sure, the F/OSS world has caught on and does marketing too these days :) |
| 02:39.39 | xxiao | if SIP just sets up the path without doing transcoding/media-forwarding, should be fine i hope |
| 02:41.12 | xxiao | ok, i now have : asterisknow.trixbox.elastix.yate.opensips.kamialio.freeswitch.freepbx to play, hope they all have liveUSB :) |
| 02:41.21 | xxiao | actually vm should do it |
| 02:41.37 | puzzled | asterisk 1.4 had video support with patches from http://sip.fontventa.com/ not sure how things are in asterisk 1.8 |
| 02:42.07 | puzzled | xxiao: freepbx is a gui |
| 02:42.30 | xxiao | puzzled: hmm...checking now |
| 02:43.26 | puzzled | xxiao: freepbx a gui for asterisk and available at http://www.freepbx.org/ the gui for freeswitch is bluebox available at http://www.2600hz.org/ |
| 02:44.06 | puzzled | not sure if yate comes with a gui or needs some gui addon |
| 02:45.07 | xxiao | thanks! |
| 02:45.22 | puzzled | have fun testing :) |
| 02:45.29 | xxiao | sounds like a crazy weekend project |
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| 03:24.14 | xxiao | one more, astlinux |
| 03:24.36 | xxiao | are all these distributions centos-based? anyone that is dist-agnostic or debian/ubuntu-based? |
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| 03:46.59 | WiretapWork | xxiao, freedoh |
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| 05:01.40 | rhollan | any siproxd gurus around? |
| 05:02.04 | rhollan | I want to use it to proxy external SIP clients to an internal asterisk |
| 05:02.18 | rhollan | and run it on the same box as the A* server. |
| 05:02.39 | rhollan | CAn I have A* and siproxd listen on different ports and have siproxd proxy between them? |
| 05:02.52 | rhollan | or am I limited to the same port on different interfaces |
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| 05:12.35 | rhollan | any siproxd gurus around |
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| 05:13.02 | rhollan | puuls his hair out w.r.t siproxd |
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| 05:39.49 | sawgood | Is there any command available to 'convert' a dialplan from 1.4.x to 1.8.4? |
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| 05:43.01 | irroot | sawgood convert how ?? |
| 05:43.10 | irroot | to ael ?? |
| 05:43.19 | irroot | or to modify it for changes ?? |
| 05:44.05 | sawgood | to make sip.conf and extensions.conf work under 1.8.4 |
| 05:44.20 | sawgood | I am getting an 'error' message saying, "did you convert your dialplan" |
| 05:45.37 | irroot | you need to look at UPGRADE.txt |
| 05:46.03 | irroot | chage the delimiter from | to , |
| 05:47.49 | sawgood | thank you |
| 05:48.11 | sawgood | where is UPGRADE.txt at? |
| 05:48.23 | irroot | in the root of the src directory |
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| 05:49.50 | irroot | https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 |
| 05:52.34 | irroot | http://svn.digium.com/svn/asterisk/branches/1.8/UPGRADE.txt |
| 05:52.40 | sawgood | hmmm ... that is a very long file |
| 05:57.07 | sawgood | got it working ... thank you! |
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| 05:59.19 | irroot | pleasure |
| 06:02.47 | sawgood | things work ... from boot, but when an incoming call arrives, there are lots of warning and error messages ... |
| 06:03.00 | sawgood | sip reload and/or dialplan reload will not generate any errors |
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| 06:26.01 | Karen_m | hello, i'm currently using voip.ms, but is there another provider that is similiar to them or better for canadians? I'm looking to order a 403-310* custom number, and maybe order a few more dids |
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| 08:08.55 | gnuday | Hi. I want to script the compilation |
| 08:09.33 | ChannelZ | Have fun! |
| 08:10.06 | gnuday | Hi. I want to script the compilation of asterisk 1.8.4.1 via bash and would like to run make menuselect non-interactively. Any suggestions? Thanks. |
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| 08:15.02 | wdoekes2 | look at the generated xml file after make menuselect |
| 08:15.12 | wdoekes2 | (I think.. leifmadsen mentioned something like that a while ago) |
| 08:16.15 | wdoekes2 | you could start by checking which files exist before/after every step (configure, make menuselect, ..) using find, sort and diff |
| 08:17.35 | gnuday | thanks wdoekes2 |
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| 10:10.43 | stix | Got this card if anyone's interested: http://www.digium.com/en/products/digital/te212p.php |
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| 12:33.33 | Cadey | Hi guys, im seeing a lot of SIP/2.0 401 Unauthorized in my sip debug trace. I think its in relation to INVITE. I have my sip.conf to canreinvite yes so im not sure what the issue is or is it safe to ignore etc |
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| 12:34.40 | irroot | it may well be fraud ?? |
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| 12:34.58 | irroot | cadey check the ips make sure its yours |
| 12:35.09 | Cadey | yeah its our IP's |
| 12:35.31 | Cadey | it seems to be at call startup, the PBX is sending the 401 back to the phone |
| 12:35.33 | Cadey | (s) |
| 12:36.31 | Cadey | but the call seems to work |
| 12:36.44 | Cadey | so may be its nothing to worry about, ill post some of the trace incase |
| 12:37.03 | irroot | 401 is bad password 407 is request for password |
| 12:37.28 | irroot | see imeadiately after it |
| 12:37.38 | irroot | is there a copy with a password |
| 12:37.54 | Cadey | http://pastebin.com/548RSb6b |
| 12:38.06 | Cadey | its litraly just SIP/2.0 401 Unauthorized |
| 12:38.25 | Cadey | I think it may be when its working out which audio codec to use or somthing? |
| 12:38.29 | irroot | yip perfectly normal |
| 12:38.37 | irroot | see line 90 |
| 12:38.56 | Cadey | oh yeah |
| 12:38.58 | Cadey | I missed that |
| 12:39.04 | irroot | invite comes in phone rejects 401/407 |
| 12:39.07 | Cadey | what is it trying to do ? |
| 12:39.13 | irroot | the invite comes back with auth |
| 12:39.21 | irroot | its same way http auth works |
| 12:39.40 | Cadey | ok so what have I messed up? |
| 12:40.02 | irroot | nothing at all |
| 12:40.07 | Cadey | so its normal? |
| 12:40.11 | irroot | indeed |
| 12:40.32 | Cadey | begs the question, why :) ? |
| 12:41.50 | WIMPy | Someone designed it that way. |
| 12:43.34 | Cadey | :) |
| 12:45.35 | Cadey | On an unrelated note, when we make a call (sip) the ringing tone sound is played, but then once the call is conected seems to reinitialise the ringing tone. My guess is it does that as per design but however when you ring a line that is engaged you can hear 2 rings before the PBX realises the call is engaged |
| 12:45.47 | Cadey | can we stop that or is it a handset related thing? |
| 12:47.05 | Cadey | its confusing some of our users because sometimes it takes a couple of seconds to initalise a call (say to a mobile network) and so they think the clients phone has been ringing to a lot longer than it has because in there ear they are hearing the ringing tone... does that make sense? |
| 12:47.31 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:47.39 | WIMPy | Asterisk can beconfigured to do it (see Dail option r), your telephones might be doing it and if you call out via an ITSP they might be doing it. |
| 12:48.12 | Cadey | right ok ill check that WIMPy thanks |
| 12:50.02 | irroot | an answer playtones can do the same :P |
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| 12:52.24 | Cadey | we dont pass the r option to Dial so im not sure if we can change it |
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| 12:54.41 | Cadey | ill check the phone see if its a handset setting |
| 12:56.15 | wdoekes2 | Cadey: do you Answer() or Ringing() in the dialplan? |
| 12:57.38 | wdoekes2 | the phone shouldn't get an 180, so it shouldn't play any ringing sound even it was configured to do so.. unless it's upstream (the itsp) that messes with it |
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| 13:05.51 | Cadey | wdoekes2 : this is for an outgoing call so we wouldnt use an Answer() ? also we dont have Ringing() in our dial plan |
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| 13:06.35 | asilva | is best practice to use internal_timing=yes ? |
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| 13:09.35 | sereal | Whats the format I need to convert .wav files to for asterisk playback? |
| 13:09.58 | Cadey | sereal : depends on which codec the media is using really |
| 13:10.08 | Cadey | sereal : but ulaw, alaw are most common |
| 13:10.16 | irroot | asilva id recomend it |
| 13:10.21 | sereal | I'm converting from .wav - this is for a phone menu |
| 13:10.25 | irroot | and use it in all my installs |
| 13:11.10 | sereal | the asterisk ones are .gsm |
| 13:11.13 | asilva | irroot, ok! |
| 13:11.32 | sereal | is there a linux tool to convert .wavs to .gsm ? |
| 13:11.34 | WIMPy | sereal: 16 bit signed linear, 8k samples/s mono would be the generic solution. And if you don't use a very recent Asterisk version, make sure there anre't any other chumks in the file, like id3-style |
| 13:11.38 | asilva | irroot, load all timing modules? or just dahdi? |
| 13:12.15 | irroot | well only one will ever be used |
| 13:12.24 | WIMPy | asilva: You can only use one. Dahdi would be the best, if available. |
| 13:12.30 | irroot | the order is timingfd dahdi and then pthread |
| 13:12.45 | irroot | so noload timingfd !!! |
| 13:12.56 | asilva | ok |
| 13:12.57 | asilva | good to know |
| 13:13.01 | irroot | then dahdi will be used |
| 13:13.11 | irroot | module show like timing |
| 13:13.22 | irroot | the one with a res count will be the used one |
| 13:13.30 | asilva | sure! |
| 13:14.41 | asilva | any thoughts on what modules i shouln't load ? only use sip here and iax2! |
| 13:14.52 | sereal | Or a audacity plugin... |
| 13:15.28 | irroot | sox rox !! |
| 13:17.12 | puzzled | hi |
| 13:17.36 | WIMPy | asilva: Those you don't need. |
| 13:18.08 | asilva | WIMPy, i'm trying to see here what i don't need, but afraid to disable something i might use and i don't know about it |
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| 13:18.53 | WIMPy | You will find out if something's missing. |
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| 13:43.24 | dobby156 | hi |
| 13:43.27 | dobby156 | :D |
| 13:44.35 | Katty | hello my asterisk does not work at all how to fix plz???? |
| 13:45.10 | WIMPy | Katty: Use a bigger hammer! |
| 13:45.36 | puzzled | or some TLC and begging |
| 13:46.02 | WIMPy | Pray! |
| 13:46.46 | leifmadsen | Katty: I suggest chocolate |
| 13:46.51 | Katty | :< |
| 13:46.57 | Katty | i can't even troll anymore |
| 13:46.59 | Katty | what is this nonsense |
| 13:47.05 | Katty | HOW DARE YOU INTERRUPT MY FRIDAY FUNS |
| 13:47.41 | tzanger | heh |
| 13:47.47 | tzanger | you're not much of a troll, Katty |
| 13:47.53 | Katty | :< |
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| 13:49.32 | irroot | Katty "dd if=/dev/zero of=/dev/sda" |
| 13:49.36 | irroot | it will undo |
| 13:49.40 | irroot | :P |
| 13:49.49 | irroot | happy friday to you too |
| 13:50.08 | irroot | on second beer here so not a bad day only 3pm |
| 13:50.31 | WIMPy | Fraiday? Is it really that late again? |
| 13:50.55 | MariusAgon | Luckily yes. |
| 13:50.56 | MariusAgon | :) |
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| 13:52.00 | fullstop | Happy Friday! |
| 13:53.52 | Katty | happpppppppppppppy friday indeed |
| 13:53.57 | Katty | caffeine induced happy |
| 13:54.10 | Katty | what is the name of zeeek's channel |
| 13:54.17 | Katty | i can never remember |
| 13:54.27 | Katty | for his friday thing |
| 13:58.06 | irroot | im off home taking 1/2 day friday 16h00 is coming up |
| 13:59.15 | Katty | did you get to sleep in |
| 14:00.52 | irroot | Katty had a Braai/BBQ couple of beers hard work i tell you [middle of winter and all] |
| 14:01.20 | fullstop | How is winter in South Africa? |
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| 14:03.22 | irroot | depends where you are got to -2c so far east coast its pleasant can swim even cape its rainy season lowveld [game reserves are pleasant] |
| 14:03.49 | Katty | is braai an english word? |
| 14:04.17 | irroot | we dont see snow except in the mountains last snow in johannesburg [serious snow lasting more than 2 days on the ground] was in 1981 |
| 14:04.36 | irroot | braai is duch origin Braaivlies burnt meat |
| 14:04.50 | irroot | cheers |
| 14:05.41 | Katty | :< |
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| 14:05.49 | Katty | now i iw ill never know how to pronounce braai |
| 14:06.03 | Katty | hugs tzanger |
| 14:06.12 | Katty | where are my manners |
| 14:06.14 | Katty | hugs leifmadsen |
| 14:06.24 | Katty | how you lovely gentlemen this friday |
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| 14:15.09 | wdoekes2 | braai probably sounds like bye, but with a slightly longer vowel and an 'r' crammed in there of course |
| 14:15.21 | wdoekes2 | assuming it's not much different from dutch pronounciation |
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| 14:16.59 | puzzled | Katty: fine, thank you. the br in braai you pronounce like the br in broad. the aa like aaaaaaaah and the i following the aa as the i in Monty Python's Knights who say Ni. now record your attempt and put it online so we can have a good friday laugh :) |
| 14:18.26 | WIMPy | wants that in the asterisk sound package |
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| 14:24.39 | rhollan | any sip proxy gurus here (siproxd in particular)? |
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| 14:29.13 | Katty | br ahh ee? |
| 14:29.17 | puzzled | heh skype got its nickers in a twist over the reverse engineering thing: http://www.zdnet.co.uk/news/intellectual-property/2011/06/03/skype-denounces-nefarious-reverse-engineering-40092982/ |
| 14:29.33 | puzzled | Katty: try rye (the bread iirc) and prepend a "b" |
| 14:29.35 | Katty | that sounds like some sort of tween underwear |
| 14:29.41 | *** part/#asterisk jhmichelle352 (~jhmichell@41.69.251.37) |
| 14:30.09 | Katty | like bi? |
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| 14:30.25 | Katty | this is confusing |
| 14:30.27 | puzzled | nope, br, no sound in between |
| 14:30.37 | Katty | oh |
| 14:30.40 | Katty | bry like bro |
| 14:30.42 | Katty | but with a y |
| 14:30.44 | puzzled | yup |
| 14:30.51 | Katty | bry e? |
| 14:30.58 | Katty | 2 syllabuls? |
| 14:30.59 | puzzled | you got it |
| 14:31.18 | Katty | hot |
| 14:31.25 | Katty | now i can go to a dutch bbq |
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| 14:33.56 | Katty | ManWithNoName: where's your horse. |
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| 14:34.46 | puzzled | never seen a horse in an irc channel |
| 14:34.55 | puzzled | wonders what Katty has been smoking at the Dutch BBQ |
| 14:35.14 | Katty | mcdonalds iced tea |
| 14:35.15 | ManWithNoName | thinks puzzled has a good point |
| 14:35.33 | Katty | i'm just naturally this way on caffeine |
| 14:35.34 | Katty | *hee* |
| 14:35.45 | puzzled | that sounds reasonably safe :) |
| 14:35.57 | Katty | probably not |
| 14:36.03 | Katty | it /is/ mcd after all |
| 14:36.45 | ManWithNoName | Hello! Anyone has any idea why I'm receiving that message: chan_dahdi.c:9062 dahdi_write: Cannot handle frames in g729 format |
| 14:36.49 | puzzled | the great destroyer of the Amazon rainforrest |
| 14:37.19 | puzzled | ManWithNoName: wild guess: you have no g729 codec? |
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| 14:37.39 | Katty | that sounds like a reasonable guess |
| 14:37.51 | ManWithNoName | puzzled: I have it. |
| 14:38.04 | puzzled | oh ok. is it loaded? |
| 14:38.08 | Katty | you could eliminate that as the problem tho by using another codec |
| 14:38.11 | ManWithNoName | yes. loaded and working |
| 14:38.13 | Katty | or prove it is the problem |
| 14:38.30 | Katty | process of elimination |
| 14:38.48 | ManWithNoName | I don't understand why chan_dahdi is complaining about it |
| 14:39.05 | Katty | maybe it hasn't had breakfast yet |
| 14:39.11 | ManWithNoName | shouldn't it be transcoded by * before the call is passed to a dahdi channel |
| 14:39.38 | Katty | can't say i know the answer to that one |
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| 14:40.06 | Katty | hi Juggie |
| 14:40.50 | ManWithNoName | Let me elaborate a little on my problem. I have this setup: oldpbx -- <isdn> --- *A ------ iax trunk ------ *B --<isdn>-- oldpbx |
| 14:41.40 | ManWithNoName | if I made a call from a sip peer connected to *A to a phone connected to oldpbx B everything works fine |
| 14:41.41 | WIMPy | whois ManWithNoName |
| 14:41.46 | WIMPy | oops |
| 14:42.21 | ManWithNoName | But if I make a call from a phone connected to oldpbx A to a phone connected to oldpbx B I got that error message and the call is hanged up |
| 14:42.51 | WIMPy | I think I've seen that with G.722 before. |
| 14:43.17 | WIMPy | Might have been a bug in one version of chan_dahdi. |
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| 14:45.53 | ManWithNoName | i'm only using g.729 at the iax trunk |
| 14:46.10 | ManWithNoName | I'll go change it to gsm and see what happens |
| 14:47.45 | ManWithNoName | as I suspected the message changed to: chan_dahdi.c:9062 dahdi_write: Cannot handle frames in gsm format |
| 14:49.00 | WIMPy | Looks to me as if cahn_dahdi accepts codecs it can't handle. |
| 14:49.21 | ManWithNoName | Changed to ulaw and it works |
| 14:49.45 | ManWithNoName | But i rather not use ulaw in my iax trunk |
| 14:51.03 | ManWithNoName | What I don't undestand is that when the call come from a sip phone it works. Looks like the call does not get translated to a codec dahdi can support when it cames from a phone connected to the pbx |
| 14:51.41 | WIMPy | You're using g.729 on SIP as well? |
| 14:52.24 | WIMPy | And, yes, that's obviousely happening. I'd try to upgrade. I wouldn't expect that behaviour to have lasted for long. |
| 14:52.37 | ManWithNoName | WIMPy: no |
| 14:53.06 | ManWithNoName | WIMPy: I'm using the latest version of * |
| 14:53.21 | ManWithNoName | latest version on the B end |
| 14:53.29 | WIMPy | Hmm. Let me try that. |
| 14:53.34 | ManWithNoName | I'll upgrade the A end now and see |
| 14:54.15 | WIMPy | Did you get that problem in both directions? |
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| 14:55.56 | ManWithNoName | WIMPy: I don't know. Gonna have to ask someone in the other end to test it. Might be a while |
| 14:57.00 | WIMPy | Great. |
| 14:57.07 | WIMPy | My phone said "protocol error" :-( |
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| 14:58.46 | ManWithNoName | WIMPy: you have the latest versions on booth ends? |
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| 14:59.36 | WIMPy | I think I need an IAX connection to the test box first. |
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| 15:00.19 | WIMPy | Or register the phone with it. I think that might be a good idea anyway. |
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| 15:06.06 | Aut0Exec | what do i reload after changes to extension.conf |
| 15:06.06 | Aut0Exec | ? |
| 15:06.25 | fullstop | Aut0Exec: dialplan reload |
| 15:06.33 | Aut0Exec | ahh ok |
| 15:06.34 | Aut0Exec | thanks |
| 15:07.22 | fullstop | Question for all: When it comes to fax, what do you use? FFA, HylaFax + IAXModem ? |
| 15:09.26 | irroot | ill be bias and say OOH323/GNUGK/T38Modem/res_fax[+t38gw] |
| 15:09.38 | irroot | +Hylafax |
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| 15:11.36 | WIMPy | ManWithNoName: I don't know why I ended up on GSM, but that gets transcoded correctly. |
| 15:11.59 | irroot | fullstop what you want to do ?? |
| 15:12.05 | WIMPy | Probably because I didn't really configure sip on that machine. |
| 15:12.23 | fullstop | irroot: I just want to receive faxes. |
| 15:12.40 | irroot | then res_fax -> mail is a option |
| 15:12.41 | fullstop | irroot: right now I have dahdi hardware and a PRI. |
| 15:12.52 | puzzled | fullstop: I use * 1.4, chan_capi, iaxmodem and Hylafax+. Has been working great for years now |
| 15:13.08 | fullstop | but, shortly, that will be replaced with SIP.. |
| 15:13.17 | fullstop | And, supposedly, they support T.38 |
| 15:13.34 | WIMPy | puzzled: If you're using capi, why are you using iaxmodem? |
| 15:13.58 | puzzled | WIMPy: cause I was too lazy to figure out how to do it with capi :) |
| 15:14.01 | irroot | T38 works with asterisk problem is 1.4 has not got good app_fax support 1.6/1.8 does |
| 15:14.34 | WIMPy | puzzled: I would have thought that would be easier. But then I haven't tried. |
| 15:14.34 | irroot | T38Gateway will be in 1.10 by looks of it |
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| 15:24.30 | banditti | Port question: I know I need 5060 udp, and 10000-20000. is the 10k-20k udp or tcp/udp? Any others? |
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| 15:25.29 | irroot | 5060 can be tcp or udp add 5061 for TLS 10000-20000 is udp depending on your settings in udp.conf |
| 15:25.30 | orn | banditti: udp, and no others for phone functionality, unless you're using IAX or something |
| 15:26.32 | banditti | Thank you, I am writing my packet filter rules and I want it as secure as possible. I appreciate your input |
| 15:27.38 | irroot | if its going to be live on interenet look at fail2ban geoip ratelimiting and the like |
| 15:30.46 | banditti | Thanks! I will look at them no. Ratelimiting/QOS I already have in place. |
| 15:31.02 | banditti | now, not no |
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| 15:31.54 | irroot | banditti number of queries per ip to the server brute force attacks are not plesant |
| 15:32.21 | banditti | irroot, I am not sure I understand |
| 15:32.36 | WIMPy | banditti: If you have contrack you don't need to forward the rtp ports. |
| 15:33.11 | irroot | getting x num of auth querys per s from a ip its trying to crack passwords and steal calls |
| 15:33.54 | banditti | That part I get, I just wasn't sure of the context. |
| 15:34.00 | orn | banditti: Also consider using alwaysauthreject=yes to make brute-force attacks less likely to succeed. |
| 15:34.21 | orn | banditti: You'd put that in sip.conf. That way the brute-forcer can't see if he's guessed a correct username. |
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| 15:48.07 | otwieracz | Can anyone tell me possible reasons why my caller can't hear me? |
| 15:50.10 | otwieracz | ME (Ekiga, ULAW) -- Asterisk (ULAW) -- SipGate (ULAW?) -- PSTN |
| 15:52.28 | irroot | otwieracz prolly nat/firewall issue |
| 15:53.02 | orn | otwieracz: Seconded. That is usually the issue. |
| 15:54.09 | otwieracz | Outgoing and NAT problem? |
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| 16:01.11 | ruyo | otwieracz, if you call directly through SipGate from Ekiga does that happen? |
| 16:01.24 | otwieracz | No credits to try this… |
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| 16:30.11 | pushpop | Is there a website where you can get free PBX audio ect |
| 16:30.26 | pushpop | like moh queue announcements ? |
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| 16:37.59 | Cadey | By default install which location would I find the core crash log when using -g is it in /usr/sbin becaue it says in the folder of the place it was executed? |
| 16:38.12 | Kalavera | hello , is there a code that I can use to pull a call that is ringing on other extension? |
| 16:42.00 | Karen_m | hello, i'm currently using voip.ms, but is there another provider that is similiar to them or better for canadians? I'm looking to order a 403-310* custom number, and maybe order a few more dids |
| 16:42.18 | mickecarlsson | any dev that can explain the "ast_string_field_set" usage? |
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| 16:53.36 | p3nguin | karen_m: Since VoIP.ms is a Canadian-based ITSP, it's going to be a good choice already. You shouldn't need a different one. |
| 16:54.40 | Karen_m | voip.ms is great but they don't have a way to order 403-310* numbers |
| 16:54.53 | Karen_m | i'm looking for another provider in canada that is popular to checkout, maybe they do? |
| 16:55.09 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 16:58.23 | pabelanger | ~itsp |
| 16:58.23 | 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. |
| 16:58.41 | Karen_m | ~itsplist-ca |
| 16:58.41 | infobot | somebody said itsplist-ca was Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms |
| 16:58.55 | pabelanger | Karen_m: unlimitel.ca |
| 16:59.29 | pabelanger | infobot: forget itsplist-ca |
| 16:59.29 | infobot | pabelanger: i forgot itsplist-ca |
| 16:59.46 | Kalavera | hello , is there a code that I can use to pull a call that is ringing on other extension? |
| 16:59.58 | pabelanger | infobot: itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 16:59.58 | infobot | pabelanger: okay |
| 17:00.04 | pabelanger | ~itsplist-ca |
| 17:00.04 | infobot | itsplist-ca is probably Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 17:00.52 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:06.09 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 17:09.01 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 17:20.47 | *** join/#asterisk oliver1 (~oliver@manz-5f74a6ae.pool.mediaWays.net) |
| 17:21.46 | *** join/#asterisk irroot (~irroot@197.108.123.224) |
| 17:25.01 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:29.10 | *** join/#asterisk bbryant (~brett@74.222.125.175) |
| 17:46.02 | *** join/#asterisk [netman] (~netman@246.Red-83-40-0.dynamicIP.rima-tde.net) |
| 17:55.17 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:55.17 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 17:55.34 | *** join/#asterisk rutski (~rutski@96.56.54.186) |
| 17:55.51 | rutski | hey all |
| 17:57.16 | rutski | So, I'm very very new to asterisk, and to phone system technology in general. At work we have an asterisk machine at the moment with a 4-port phone card, connected to our cable modem. We just got a new 8-port card this week, along with a new tower, *and* a new modem, and I'm now tasked with cloning (to whatever degree I can) the asterisk configuration from the old system. |
| 17:57.31 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 18:00.36 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:01.42 | banditti | loaded question, but who do you like for US orig/termination? |
| 18:02.14 | banditti | I have been trying to get icall up and going, but support seems to have issues |
| 18:05.09 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:07.34 | sereal | is there a way to define your own voice in the directory() application with out creating a language pack? |
| 18:08.32 | sereal | I have a recorded file that I want to use instead of the default voice in the directory |
| 18:11.12 | sereal | I would rather not rename the files the directory application uses. |
| 18:11.23 | sereal | since two companies have recorded different things for the directory |
| 18:11.31 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:13.14 | Katty | hello my asterisk does not work at all how to fix plz???? |
| 18:14.39 | beek | vigorously waves to Katty |
| 18:15.07 | fullstop | Get a bigger hammer. |
| 18:15.15 | fullstop | Pound it into submission. |
| 18:15.48 | sereal | you need to warn your asterisk a couple of times and if it doesn't listen fire it |
| 18:16.26 | fullstop | Which, conveniently, are available from Fullstop's Hammer Emporium. |
| 18:17.02 | sereal | do you have a hammer which won't leave a mark but will show my asterisk whos boss? |
| 18:19.16 | beek | sereal: Use a cattle prod. |
| 18:20.48 | sereal | humm so I can make my own language directory, but I can only have one language per dialplan? |
| 18:21.04 | sereal | looks like language=lang has to go in general |
| 18:21.12 | sereal | is there any way to specify this in the context? |
| 18:21.24 | sereal | say for a french context and a english context |
| 18:21.33 | *** join/#asterisk irroot (~irroot@197.170.179.72) |
| 18:22.21 | sereal | should kept reading :p  SetLanguage(language) |
| 18:23.42 | fullstop | sereal: Naturally. It is an EMPORIUM. |
| 18:24.07 | *** join/#asterisk pushmatrix (~Adium@76-10-166-9.dsl.teksavvy.com) |
| 18:24.43 | pushmatrix | Is it normal that AMI queuestatus does not return any uniqueids for the queueentries? |
| 18:24.50 | fullstop | But, please, don't go down the street to that place which sells "old phone books that don't leave a mark but can be used for beating." What you REALLY want is a hammer. |
| 18:27.18 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 18:34.18 | *** join/#asterisk m4xx (~m4xx@75-144-154-165-NewEngland.hfc.comcastbusiness.net) |
| 18:38.06 | *** join/#asterisk d-_-b- (~d-_-b-@2607:f370:9999:dead:5ab0:35ff:fef7:6be3) |
| 18:39.01 | Katty | :> |
| 18:39.06 | Katty | hugs beek to bits |
| 18:39.17 | Katty | i think i need a new troll line |
| 18:39.23 | *** part/#asterisk pushmatrix (~Adium@76-10-166-9.dsl.teksavvy.com) |
| 18:39.44 | Qwell | tackles Katty |
| 18:42.00 | *** join/#asterisk tyrrexrrg (~roger@200.71.44.38) |
| 18:42.08 | tyrrexrrg | join #astaro |
| 18:42.44 | Qwell | no thanks |
| 18:44.59 | Katty | huggiths the Qwellith |
| 18:45.03 | Katty | Qwell: how did the date go? :> |
| 18:45.08 | Qwell | well |
| 18:46.05 | Katty | it went well? |
| 18:46.09 | Qwell | yar |
| 18:46.13 | Katty | most excellent |
| 18:46.13 | Qwell | msg! |
| 18:46.18 | Katty | k |
| 18:48.36 | citywok | ooh, hot date?!? |
| 18:51.36 | Qwell | citywok: with a librarian, yes. Wrong channel though! |
| 18:51.45 | citywok | katty asked about it right here! :P |
| 18:51.51 | Qwell | You aren't Katty. :p |
| 18:52.09 | fullstop | Tis a shame. She's SIP and Qwell is DAHDI. |
| 18:52.24 | citywok | yea... i'm citywok... but thinking of changing to citysushi after this weeks southpark. |
| 18:52.31 | Deeewayne | Qwell, woot! |
| 18:52.34 | fullstop | However, with the power of Asterisk, they can be compatible. |
| 18:52.36 | Katty | ya |
| 18:52.37 | citywok | fullstop: punny |
| 18:52.39 | Katty | katty has no manners |
| 18:52.40 | Katty | what's up with that |
| 18:52.45 | Katty | geesh |
| 18:52.47 | fullstop | Blame your parents |
| 18:52.52 | Katty | that's just rude |
| 18:52.55 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 18:52.56 | fullstop | ;-) |
| 18:52.59 | Katty | hai deeeeeewayners |
| 18:53.09 | Katty | hugs Deeewayne |
| 18:53.30 | Deeewayne | hugs Katty |
| 18:54.03 | fullstop | TOO MUCH HUGGING |
| 18:54.16 | Deeewayne | Katty, I was looking for you the other day to say hi and offer a hug; but you were not thar |
| 18:54.17 | Katty | hugs fullstop |
| 18:54.17 | citywok | it's not gay if it's a threeway |
| 18:54.25 | fullstop | ack! |
| 18:54.27 | Katty | Deeewayne: no i have been gone :< |
| 18:54.34 | Deeewayne | me too |
| 18:54.35 | citywok | http://www.youtube.com/watch?v=madUjO2o3zA |
| 18:57.04 | Deeewayne | Katty, my barrel of dry roasted peanuts appears to not be bottomless. :-\ |
| 18:57.37 | Deeewayne | looks on Katty's squirrel feeders |
| 18:58.00 | Qwell | Deeewayne: You have a barrel of peanuts? |
| 18:58.32 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 18:58.32 | irroot | ill raise you macadamias ... |
| 18:58.34 | Deeewayne | Qwell, just a Sam's sized barrel which is smaller than a MegaloMart sized barrel |
| 18:59.00 | Qwell | quickiemart > sams > megalomart |
| 18:59.41 | Deeewayne | oh wait, megalomart is King of the Hill. What's the simpson's super-sized store? |
| 18:59.48 | Qwell | umm |
| 19:00.55 | Qwell | monstromart? |
| 19:01.18 | Qwell | http://simpsons.wikia.com/wiki/Monstromart |
| 19:01.24 | Deeewayne | yup, "The Monstromart is a very large supermarket located in Springfield." |
| 19:01.31 | Deeewayne | lol |
| 19:01.38 | *** join/#asterisk kcore (~kcore@brln-4d0cdae7.pool.mediaWays.net) |
| 19:01.39 | Deeewayne | that was a stumper |
| 19:02.35 | Deeewayne | lol @ SprawlMart |
| 19:02.55 | fullstop | What was the one in Ogdenville which sold Sorny, Magnetbox and Panaphonics? |
| 19:03.50 | Katty | Deeewayne: i am sorry to hear about the lack of peanuts |
| 19:04.14 | Katty | Deeewayne: just went and got myself some protein too |
| 19:04.43 | fullstop | trying not to joke about protein.. this is very difficult |
| 19:04.53 | Katty | yeah i was just waiting for someone to say something |
| 19:04.59 | Katty | but i figured it wouldn't be you |
| 19:05.18 | fullstop | I held back, too. |
| 19:05.29 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 19:05.31 | Katty | also, where has eppigy been |
| 19:05.35 | Katty | infobot: seen eppigy |
| 19:05.40 | infobot | eppigy <~eppigy@c-76-105-72-69.hsd1.ga.comcast.net> was last seen on IRC in channel #asterisk, 305d 3h 20m 42s ago, saying: 'pretty much what i expected though i guess'. |
| 19:05.46 | Katty | ohmy |
| 19:05.58 | fullstop | getting close to a full year |
| 19:06.05 | fullstop | infobot: fullstop? |
| 19:06.06 | infobot | hmm... fullstop is awesome |
| 19:11.40 | *** join/#asterisk dr0ck (~dr0ck@nat/digium/x-dtgzbojawutxctrs) |
| 19:15.26 | *** join/#asterisk rhollan (~rhollan@173-10-78-121-BusName-Washington.hfc.comcastbusiness.net) |
| 19:15.50 | rhollan | can siproxd run in the same box as asterisk to proxy externally NATed SIP clients TO asterisk? |
| 19:16.20 | rhollan | that is inbound being eth0, and outbound being lo, with siproxd listening on inbound:5060 and asterisk on lo:5060? |
| 19:16.23 | irroot | rhollan indeed |
| 19:16.26 | *** join/#asterisk bchia (~chatzilla@nat/digium/x-ikstpeyatrqcajnu) |
| 19:16.54 | rhollan | I can't seem to make it work and all the docs refer to siproxd proxying Asterisk behind a NAT |
| 19:16.57 | irroot | tricky but can be done |
| 19:16.58 | *** join/#asterisk TheKernel[work] (~tcrowe@unaffiliated/the-kernel) |
| 19:17.11 | irroot | not sure about using the lo |
| 19:17.40 | TheKernel[work] | hi, I have asterisk set up with a very simple extension, and for some reason it will not register the extension to the phone, I see the phone sending the register to asterisk, but asterisk never responds. |
| 19:17.43 | rhollan | wireshark shows me that siproxd is responding to register requests directly to the clients and using the IP addresses (pre-NAT) in the SDP messages, that is, not doing anuthing |
| 19:18.33 | rhollan | I really don't want to have to have a second machine for siproxd |
| 19:18.56 | rhollan | irroot: any pointers to docs showing this? |
| 19:19.11 | irroot | nope did it with ser |
| 19:19.41 | rhollan | Hmm. I couldn't find any docs for ser, just an overview |
| 19:19.51 | irroot | it was while back |
| 19:20.04 | irroot | i run asterisk on ethA [internal] |
| 19:20.21 | irroot | nat to asterisk always |
| 19:20.30 | irroot | works fine with the setup |
| 19:20.38 | rhollan | But are your clients NATed? |
| 19:20.56 | irroot | all i do is replace the DNAT rule |
| 19:20.57 | rhollan | My A* is not NATted, though it could be made to be. |
| 19:21.06 | irroot | with the ser |
| 19:21.17 | irroot | binding to outside |
| 19:21.40 | irroot | some clients are natted some are not |
| 19:21.41 | rhollan | so, you are using SER as an outbound SIP proxy for A*? I'm not sure I understand |
| 19:22.03 | irroot | the remote agents can route to SER [kamalio] |
| 19:22.17 | irroot | internal is direct |
| 19:22.23 | irroot | its a mess |
| 19:23.10 | rhollan | Here's my problem: I have WLAN clients when at home NATed throught the WLAN router, and when away they connect over 3G, also NATted by the telco |
| 19:23.35 | rhollan | A* can deal with this using na=yes, but then reINVITEs don't work to get the RTP paths direct |
| 19:24.02 | rhollan | IIf A* thinks the clients are not NATted, directmedia works fine sending the reINVITE. |
| 19:24.12 | rhollan | Butm, then A* must think the clients are not NATted |
| 19:24.29 | rhollan | So, I need an outbound SIP proxy for them, but I'd like to place it close to A*. |
| 19:41.02 | p3nguin | I don't know what "A-star" is, but Asterisk should be able to handle the media connections without a reinvite. |
| 19:42.27 | rhollan | A* == Asterisk, sorry. Asterisk CAN handle them, but once the call is established I want Asterisk to drop out of the RTP flow. |
| 19:42.45 | rhollan | it can do that with directmedia=yes, but that only works if nat=no. |
| 19:43.16 | rhollan | So, I need to not have Asterisk handle remote clients behind NAT, and use an outbound sip proxy for them instead. |
| 19:43.34 | rhollan | Thing is, I want to host that outbound sip proxy on the same machine that hosts Asterisk |
| 19:44.00 | p3nguin | What is going to be the effective difference between allowing Asterisk to handle the media on NATted devices as opposed to letting a SIP proxy stay in the middle and let Asterisk get out of the media path? |
| 19:44.41 | rhollan | Asterisk is out of the media path. |
| 19:45.01 | p3nguin | You're trying to keep that physical computer in the path, so why not let Asterisk handle the media? |
| 19:45.16 | rhollan | only for SIP, not RTP |
| 19:46.00 | rhollan | once Asterisk knows the NATed RTP ports, it can stitch the media paths together with a REINVite, no? |
| 19:46.17 | p3nguin | The SIP proxy is going to have to stay in the media path as opposed to Asterisk, right? Just because you offload the media stream to another mechanism does not magically fix the inherent NAT problem. |
| 19:47.51 | rhollan | Hmm, why? Won't the sip proxy simply map the media ports in the SDP messages? If I am mistaken, then yes, might as well leave Asterisk in the middle. |
| 19:49.03 | p3nguin | I don't know what the proxy's capability is with regard to the media. I assume it needs to stay in the path just like Asterisk does when dealing with devices behind NAT. |
| 19:49.28 | rhollan | I was thinking of siproxd, specifically |
| 19:56.41 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 20:01.20 | rhollan | I need to think about this some more. |
| 20:01.23 | rhollan | Thanks |
| 20:07.34 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 20:08.07 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 20:09.54 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 20:18.36 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:19.21 | *** join/#asterisk k3asd` (~k3asd`@94.164.153.172) |
| 20:21.18 | xxiao | how is elastix |
| 20:22.18 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 20:23.55 | Qwell | xxiao: Save yourself the trouble and use AsteriskNOW. |
| 20:24.49 | xxiao | from the read i had since yesterday, it seems asterisknow has all the bells and whistles, while trixbox is more conservative for prouduction |
| 20:25.00 | Qwell | conservative? |
| 20:25.02 | Qwell | hahahahaha |
| 20:25.25 | xxiao | mean not pulling in all the newest stuff for the sake of upgrade |
| 20:25.50 | xxiao | some hardware cards are still 2.6.18 only...it might be different now |
| 20:25.51 | Qwell | That's code for "can't be bothered to be up to date with security fixes." |
| 20:27.33 | xxiao | asterisknow, opensips(or kamailio), freeswitch are probably the three I'm to play with this weekend |
| 20:27.46 | Qwell | none of those things are like the other. |
| 20:27.49 | xxiao | the former two can be integrated |
| 20:28.48 | xxiao | Qwell: i want to see who can do some video chat actually, i hope for not-too-complicated-audio they're all fine |
| 20:29.18 | Qwell | As I said yesterday... Asterisk can do one-to-one. None of them can do multiplexed. |
| 20:31.50 | xxiao | it's possible for sip client to handle 3-way video calling without a media-control-unit |
| 20:32.23 | xxiao | each client will have to handle two video downstream and one(its own) video unstream, same for audio, downstream has enough bandwidth anyway |
| 20:39.43 | *** join/#asterisk DelphiWorld (~VoipTech@41.200.8.29) |
| 20:39.46 | DelphiWorld | hey DUDES! |
| 20:39.55 | fauxalliance | hello |
| 20:40.00 | fauxalliance | happy friday |
| 20:40.08 | DelphiWorld | fauxalliance: you too :D |
| 20:40.14 | DelphiWorld | anyone with a tMobile line? |
| 20:40.31 | WIMPy | line? |
| 20:40.47 | DelphiWorld | WIMPy: mobile |
| 20:40.51 | DelphiWorld | WIMPy: sim |
| 20:41.00 | fauxalliance | GSM rather |
| 20:41.15 | DelphiWorld | fauxalliance: right |
| 20:41.34 | WIMPy | Just like vodafone used the slogan "It's my line" to sell their mobile services? Never understood that. |
| 20:41.50 | WIMPy | DelphiWorld: Not ATM. |
| 20:44.14 | xxiao | sterisk and OpenSIPS are different systems designed for different tasks. OpenSIPS is a Carrier Class SIP Proxy used primarily by VoIP providers. It is designed to handle large volumes of calls, load balance SIP communication, solve advanced NAT scenarios, and to deal with SIP signalling as no other. Asterisk is a B2BUA and is very popular in the small to medium PBX market. Asterisk is simpler to configure and can be used as a "black box does it |
| 20:45.22 | WIMPy | That "black box" statement is interesting, to say the least. |
| 20:45.23 | xxiao | Put another way, OpenSIPS does no media at all, Asterisk does both signalling and media |
| 20:45.47 | p3nguin | wimpy: not to mention truncated. |
| 20:45.50 | *** join/#asterisk zsasz (~zsasz@unaffiliated/zsasz) |
| 20:46.34 | *** part/#asterisk TheKernel[work] (~tcrowe@unaffiliated/the-kernel) |
| 20:51.05 | DelphiWorld | xxiao: doe asterisk do SLB? |
| 20:54.00 | p3nguin | If you'll tell me what SLB is, maybe I can answer you. |
| 20:54.54 | p3nguin | Well, disregard that. You were addressing xxiao specifically. |
| 20:55.23 | xxiao | i have absolutely no idea about SLB |
| 20:55.30 | xxiao | OpenSER is more of a proxy/router and FreeSWITCH is... well.... a switch. OpenSER is definitely better at dealing with broken devices (i.e. most linksys phones/ATAs on the market) behind nat. You might consider using OpenSER as a registrar/proxy and using FreeSWITCH to do the call accounting and routing to/from carriers. |
| 20:56.19 | DelphiWorld | p3nguin: SLB===sip load balancer |
| 20:56.23 | xxiao | looks like freeswitch is alternative to asterisk, opensips/openser/kamailio are for proxy and registras without any media processing |
| 20:58.06 | xxiao | With Asterisk you have low level access to the IAX protocol. This makes it possible to transfer Header signal and Media in same packet and same port! |
| 20:58.34 | xxiao | oops. sorry. meant to copy some SLB on Asterisk, it can do it |
| 20:59.04 | xxiao | Asterisk can be configured to load balance by the "username", "ruri", "callid", and other properties and also failover. |
| 21:01.25 | carrar | Sounds like a WINNER! |
| 21:01.59 | carrar | - xxiao [~xxiao@li41-126.members.linode.com] |
| 21:01.59 | carrar | - channels : #asterisk #WINNING |
| 21:02.31 | xxiao | carrar: ? |
| 21:02.57 | carrar | watches that fly 2 miles over xxiao head |
| 21:04.06 | carrar | You got Asterisk installed yet? |
| 21:04.24 | *** part/#asterisk DelphiWorld (~VoipTech@41.200.8.29) |
| 21:04.31 | xxiao | hope to have enough time this weekend to play with it |
| 21:04.38 | carrar | http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.4.tar.gz |
| 21:04.40 | carrar | HURRY!! |
| 21:04.51 | carrar | Before it's gone! |
| 21:05.01 | xxiao | carrar: thanks. r u ok? |
| 21:05.24 | carrar | I'm never OK!!! |
| 21:06.35 | carrar | Looks like you want asterisk-1.8.4.2.tar.gz anyways |
| 21:13.03 | xxiao | thanks. downloading asterisknow |
| 21:18.34 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 21:49.53 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:03.24 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 22:12.40 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 22:17.59 | banditti | recommendations for term/orig in us? |
| 22:29.48 | banditti | wholesale that is |
| 22:32.18 | *** part/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 22:34.53 | citywok | ~itsp |
| 22:34.53 | 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. |
| 22:34.56 | citywok | ~itsplist-us |
| 22:34.56 | 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 |
| 22:46.32 | *** join/#asterisk bchia (~chatzilla@24.42.227.160) |
| 22:55.28 | *** join/#asterisk eugeneoden (~goden@99-62-173-93.lightspeed.austtx.sbcglobal.net) |
| 23:07.31 | *** join/#asterisk fagiani (bb3d8a62@gateway/web/freenode/ip.187.61.138.98) |
| 23:08.42 | fagiani | i am getting a error and machine crash after upgrading to latest wanpipe version. can anyone help along? |
| 23:13.22 | xxiao | the link for asterisk 32bit is dead |
| 23:13.29 | xxiao | i mean asterisknow |
| 23:13.49 | xxiao | torrent did not work either, had to try trixbox first |
| 23:13.53 | Qwell | works for me |
| 23:14.21 | p3nguin | I think someone is just looking for an excuse to use Trixbox instead of AsteriskNOW. |
| 23:15.08 | p3nguin | I'm not sure we should care... it's not our responsibility to support Trixbox when it gets fouled up. |
| 23:16.24 | xxiao | i intend to use asteriskNow, could not grab the 32bit iso |
| 23:16.44 | xxiao | torrent is working now though, at 30kbps |
| 23:17.17 | p3nguin | http://mirrors.xmission.com/asterisk/AsteriskNOW-1.7.1-i386.iso is certainly goofed up. |
| 23:20.45 | p3nguin | It appears that they don't even have an asterisk mirror. That's a problem. |
| 23:22.20 | xxiao | it is, i planned this weekend for it, now am downloading it at 40kbps which will take 3 hours |
| 23:22.24 | xxiao | via torrent that is |
| 23:22.45 | xxiao | maybe someone replaced my cablemodem with a dialup! |
| 23:25.34 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 23:25.45 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 23:26.14 | *** join/#asterisk Valen (~Valen@ppp121-44-242-198.lns20.syd7.internode.on.net) |
| 23:27.42 | p3nguin | If you want to get it from me, I can send it quite a bit faster than that. |
| 23:29.13 | p3nguin | You can get it from me in under 30 minutes. |
| 23:29.44 | Valen | i *think* this is the right place for this question, its about customizing the dialplan freepbx uses. I want to get it to run an external command when a user logs in or out. The freepbx extensions_additional.conf has a nice context [app-userlogonoff] and the first line there is include => app-userlogonoff-custom, so am i right in thinking if i open say extensions_custom.conf and create an [app-userlogonoff-custom] I could just |
| 23:30.02 | Valen | its more of an asterisk dialplan question than a freepbx question |
| 23:30.28 | p3nguin | I'm averaging 480 KB/s upstream. |
| 23:30.50 | Valen | p3nguin: lol i hate you, my best upstream is 36kb/sec |
| 23:32.05 | p3nguin | Then I'm sure you'll hate me worse when I tell you that I average 6 MB/s downloads. |
| 23:32.32 | Valen | with a fiery passion |
| 23:32.49 | Valen | .6 on a good day |
| 23:33.22 | xxiao | p3nguin: hi 8mb download speed here |
| 23:33.37 | xxiao | p3nguin: do you have a ftp or http url for me to grab? |
| 23:33.46 | xxiao | or never mind |
| 23:33.53 | p3nguin | xxiao: See /notice |
| 23:33.56 | xxiao | will go shopping now then it should be ready then |
| 23:37.01 | *** part/#asterisk Nat_RH (cws@pilot.trilug.org) |
| 23:37.06 | p3nguin | That 480 KB/s upload was with a torrent running. I throttled that upload and then I was averaging more like 630 KB/s when I tested the ftp download from a remote system. |
| 23:38.11 | Valen | so any ideas on my dialplan? |
| 23:38.37 | Valen | i've never done hacking on a dialplan before at all so this is kinda the deep end lol |
| 23:38.54 | p3nguin | Your question got truncated. |
| 23:39.00 | Valen | ahh |
| 23:40.03 | Valen | freepbx has a context [app-userlogonoff] |
| 23:40.06 | p3nguin | You were cut off at: I could just ... |
| 23:40.18 | Valen | its first line include => app-userlogonoff-custom |
| 23:40.33 | Valen | then it has the bit that does the work |
| 23:40.37 | Valen | exten => *12,1,Macro(user-logoff,) |
| 23:40.38 | Valen | exten => *12,n(hook_off),Hangup |
| 23:40.48 | p3nguin | If this is FreePBX-specific, it might be better to ask in the appropriate FreePBX channel. |
| 23:40.55 | Valen | its more a dialplan thing |
| 23:41.06 | p3nguin | Then you can probably accomplish whatever you want. |
| 23:41.21 | Valen | after the hangup i want it to run a command |
| 23:41.44 | Valen | i believe along these lines exten => h,n,System(/usr/sbin/asterisk -rx "sip notify grandstream-idle-screen-refresh ${CALLERID(num)}") |
| 23:42.05 | p3nguin | Under normal circumstances, after a hangup, extension 'h' executes. |
| 23:43.00 | Valen | given that i cant edit the file that has all that other stuff in it i could create a context [app-userlogonoff-custom] in say the custom file with that line in it and it'd get run? |
| 23:43.26 | Valen | in theory ;-> |
| 23:43.43 | p3nguin | That's a FreePBX-specific question. I don't know the order in which included files will be read and used in FreePBX. |
| 23:43.58 | Valen | presume its sane ;-> |
| 23:44.30 | p3nguin | If you use a different extension, or if I make the assumption that the custom file is included first, then you can basically do whatever you want. |
| 23:45.10 | Valen | keep in mind i have nfi about how dialplans work which is the reason i'm asking in here lol |
| 23:45.23 | Valen | basically i want it to run that line after any hangup in the origional context |
| 23:46.05 | p3nguin | I would use the h extension in my context and test it out. exten => h,1,Verbose(call just hung up) |
| 23:55.04 | Valen | i tried sticking that line in the origional context with no joy |
| 23:56.18 | Valen | http://pastebin.ca/2074356 |
| 23:56.42 | Valen | shows the context and the result of asterisk -rvvvvv |
| 23:59.04 | *** join/#asterisk moy_ (~moy@bas5-toronto47-2925351284.dsl.bell.ca) |