IRC log for #asterisk on 20150322

00:01.24ChannelZheh the Downloads page on asterisk.org is a little borked again, the download icon is just a 'W'.. supposed to come from some font called EntypoRegular which doesn't appear to be downloaded in the CSS
00:01.28mjordanwhat does "module show like bridge" say?
00:01.53mjordanChannelZ: which page in particular?
00:02.05mjordanah
00:02.05ChannelZhttp://www.asterisk.org/downloads
00:02.07mjordanyeah, I see it
00:02.09mjordanhuh
00:02.14mjordanI wonder who borked that up.
00:02.24mjordan(that's new)
00:02.41MaliutaLapmjordan: I blame ChannelZ ;)
00:02.51ChannelZI seem to recall it happening once before a long time ago
00:03.00mjordanwould have been quite awhile ago
00:03.09mjordanwasn't aware of any updates being made this weekend...
00:03.20ChannelZMaybe something hacked manually in your CMS or something that gets reset with an update
00:03.25KarriganIt says "0 modules loaded"
00:03.26ChannelZshrugs
00:03.42ChannelZDo you have autoload turned on in modules.conf Karrigan?
00:04.48Karriganyes.
00:05.08ChannelZHmm. Did you compile this yourself?
00:05.51KarriganNever open it before.
00:05.55mjordanKarrigan: well, that is your problem. You need to have bridge technology modules loaded in order to mix media in bridges.
00:05.57ChannelZthe bridge mods don't have any weird dependencies that I'm aware of..
00:06.04mjordanthey don't.
00:06.57KarriganOk, and what i can do for fixing it ?
00:07.26MaliutaLapload the modules
00:07.37ChannelZRE: did you compile asterisk yourself or was it an installed package?
00:07.39MaliutaLapthat would help things
00:09.00ChannelZSee if "ls /usr/lib/asterisk/modules/bridge*" gives you anything (assuming that's where your asterisk installed to) for starters I guess
00:09.54MaliutaLap'locate asterisk | grep modules | grep bridge'
00:10.28ChannelZthat too (if the locate database is up to date)
00:11.39Karriganso, ls give me 6 modules
00:11.45MaliutaLapfind / -type f -name 'bridge*' | grep asterisk
00:12.54ChannelZcurious why they're not autoloading.  do "module load bridge_simple" and "module load bridge_softmix" on the asterisk console for fun
00:15.14KarriganWow !  it's working now !  thanks a lot
00:16.19ChannelZso I'm not sure what other module isn't loading that would normally cause the bridges to autoload that you're missing..
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00:20.06KarriganNo idea either.  But once again, thanks !  I probably will bother you again in the future^^
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06:42.50flexibeastHi all. Having some problems with NAT traversal. Can successfully make calls between devices that are both behind the NAT, but can't make calls that traverse the NAT in either direction - a device will ring, but upon pickup, there's no audio, and 30 seconds later, the call times out.
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06:44.52flexibeastThe network topology involved is this: Internet -> DSL modem -> gateway (with firewall) -> LAN.
06:45.22flexibeastThe DSL modem is set up to forward UDP ports 5000:31000 to the gateway machine, where Asterisk is running.
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06:47.12MaliutaLap~sipnat
06:47.12infobot[~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions .  Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet.
06:47.31flexibeastThe IP address of that modem is dynamic, but mapped to a static global host name via a dynamic DNS service.
06:47.42flexibeastThe settings in sip.conf look like:
06:48.25flexibeast[general]
06:48.25flexibeastbusylevel=10
06:48.25flexibeastcallcounter=yes
06:48.25flexibeastcontext=default
06:48.29flexibeastexternhost={hostname}
06:48.32flexibeastexternrefresh=180
06:48.35flexibeasticesupport=yes
06:48.38flexibeastlocalnet=10.0.0.0/255.0.0.0
06:48.42flexibeastnat=auto_force_rport,auto_comedia
06:48.45flexibeasttransport=udp
06:49.03flexibeastHmm, so maybe the issue is that i've not set directmedia appropriately?
06:49.40flexibeast(Oh and also, the NAT is Linux masquerading.)
06:50.00flexibeastThat aocomputing.net link is 404. :-)
07:00.22flexibeastSetting directmedia to 'no' or 'outgoing' made no difference.
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07:29.08ChannelZPlease don't flood like that, use pastebin
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07:30.42ChannelZIs your externhost set to a hostname that correctly resolves to your WAN IP?  Otherwise use externaddr=yourIP
07:30.48flexibeastOkay, what are the maximum number of lines before something is considered a flood?
07:31.20ChannelZlike 3 if you're pasting configs
07:31.43flexibeastOkay, thanks, sorry (my experience on other chans has been that <12 lines is fine.)
07:31.56flexibeastYes, externhost correctly resolves to the WAN IP.
07:32.30flexibeast(It's the hostname i successfully use to access other services on that machine.)
07:32.31ChannelZSo the devices that are outside the network; are they behind NAT as well?
07:32.38flexibeastNope.
07:33.05flexibeasti'm testing with a phone using Linphone and mobile data.
07:33.46ChannelZLet's see a sip debug of an attempted call.  "sip set debug on" on the console, then try the call
07:34.05flexibeastSince the host is running Debian (Wheezy), i've also made sure to load the nf_nat_sip and nf_conntrack_sip modules.
07:34.09flexibeastOkay, moment.
07:34.32ChannelZoh, well I'd turn off those iptables modules
07:34.59flexibeastOkay, it wasn't working without them either, but i'll disable them. :-)
07:35.06ChannelZI've never been able to get them to do anything sensible.  It's more for handling dumb SIP devices behind the firewall but I've never had it work right with asterisk
07:35.26flexibeast*nod*
07:35.31flexibeastOkay, sip debug on.
07:35.51ChannelZSo long as you are able to port-forward 5060 and some range of ports (as defined in rtp.conf) to your asterisk box (or allow that traffic in if asterisk runs on the firewall machine its self) you should be in good shape.
07:36.04ChannelZ(Plus allowing pretty much all UDP traffic out from asterisk)
07:36.59flexibeast*nod* i've port-forwarded 5060 and the UDP port range 19990:20000 to the Asterisk box (and rtpstart and rtpend are set accordingly in rtp.conf). There are no outbound traffic filters from the Asterisk box.
07:45.32flexibeasti.e. "no filters on outbound traffic from the Asterisk box".
07:46.45ChannelZyeah. So have you captured a sip debug yet?
07:49.36flexibeastNo, sorry, was waiting for next instructions, sorry.
07:49.41flexibeastWill do so now, and pastebin it.
07:54.13flexibeasthttp://pastebin.com/D0wuUrJc
07:55.42ChannelZ...
07:55.57ChannelZso this is calling from your mobile -> asterisk ?
07:57.41flexibeastYep.
07:59.33ChannelZHmm I'm a little puzzled by what I see.
08:01.00flexibeastOkay. :-)
08:02.59ChannelZWell, so I see the INVITE from 'dee' trying to call 6001.  It gets rejected for auth (which is normal), acknowledges, and then tries again with the auth. Asterisk responds with "trying".  But then it appears the phone sends _another_ INVITE for reasons I don't understand
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08:04.04flexibeastAh. Hmm.
08:04.21ChannelZLater on asterisk does some re-transmitting which means it never got a response about a SIP message.  So it's almost like traffic randomly isn't making it to the phone.. or it is and the softphone is just broke
08:04.28flexibeast*nod*
08:04.42flexibeastMight it be useful to make a call in the other direction, and provide the debug log for that?
08:05.48ChannelZIt's worth a shot
08:06.00flexibeastOkay, will do so now.
08:08.33flexibeasthttp://pastebin.com/EJD0rujK
08:11.59ChannelZsomething's going on with that phone is all I can figure.  Whether it's linphone or your mobile carrier interfering I dunno
08:15.01flexibeast*nod* Fair enough.
08:15.43ChannelZIt doesn't respond to the first invite, asterisk re-transmits, then it responds 'trying', then 'ringing'.. but then it responds 'ringing' again.
08:15.49flexibeastThere have been similar problems when someone outside the LAN (and not behind a NAT themselves, i believe) has tried using Jitsi for communication - when they try calling, they get a 'busy' signal.
08:16.05ChannelZI also see a re-transmit to your local phone which is troubling
08:17.31flexibeastWhen i've spoken to my mobile carrier about whether they block certain types of traffic over mobile data, they've told me no.
08:17.47flexibeast(i was asking about SSH specifically at the time, though.)
08:18.26ChannelZIf you want to set me up with a peer I can call you to test, to prove your asterisk is working right.  Or you can call me anonymously to test
08:19.13flexibeastSure, will set you up, moment.
08:21.45flexibeastOkay, username 'channelz', secret 'test1111'.
08:22.29ChannelZ..
08:23.08flexibeastYou can call me on 6001.
08:23.42flexibeast(Sorry, juggling a few things at the same time at this end.)
08:25.26ChannelZhmm "wrong password"
08:26.08ChannelZhang on I screwed something
08:26.36flexibeastOh okay.
08:28.01ChannelZok regged.. let me make a test exten and try
08:28.18flexibeast*nod*
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08:30.40ChannelZI's being rejected.
08:30.50flexibeast*sad nod*
08:31.29flexibeastAs in, you try to make the call, and it's rejected?
08:31.39ChannelZyeah
08:32.35flexibeastOkay, let me try calling you?
08:32.38ChannelZit's telling me unauthorized but never tries again with auth
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08:33.46ChannelZI'd have to see what your side is saying.  Anyway, Dial(SIP/burner.com/1) to hear monkeys, Dial(SIP/burner.com/2) for an echo test
08:36.04flexibeastHmm, i'm having trouble working out how to do that in Jitsi - are you familiar with the latter?
08:37.16ChannelZOh I was saying from your asterisk, to remove jitsi from the equation since we're not sure any of them are working right.  But sip:1@burner.com is a 'normal' URI. (Sorry I've never used Jitsi or Linphone)
08:37.42flexibeastOh, sorry, okay.
08:39.55ChannelZJudging by what I'm seeing online, you maybe add a contact, with 1@burner.com as the 'ID or Number'
08:40.24flexibeastYeah, just tried that, got told by Jitsi "Address incomplete".
08:41.19flexibeastSo to use the 'Dial...' thing you said above, i need to add an entry in extensions.conf?
08:41.31ChannelZdo whatever you just did again..
08:41.44ChannelZI lied about extension 1 but I fixed it :)
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08:41.51flexibeastHeh, okay, got monkeys now. :-)
08:42.30ChannelZSo now do 2 and see if you get bidirectional audio properly
08:42.44flexibeastYeah, echo test works fine also.
08:44.01ChannelZok. So I'm not sure what we really proved, except that whatever device you were just calling from seems to work OK
08:44.53ChannelZAs for your asterisk, you can make a test extension that does Dial(SIP/burner.com/2) and then call that exten
08:49.44flexibeastYeah, the latter is what i did to do the (successful) echo test.
08:49.57flexibeastThe device i was calling from was Jitsi on my laptop, behind the NAT.
08:51.32flexibeastHowever, when i try to call either test extension from my phone, using mobile data, nothing happens.
08:52.37flexibeastHowever, changing the phone to use wifi instead, so that it's on the same LAN as the Asterisk server, means both tests work fine.
08:54.54ChannelZSo the same app on your phone via wifi works, but via your mobile data doesn't
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08:58.44flexibeastYep.
09:00.40ChannelZSo I guess we have some sort of answer
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09:03.40flexibeastYeah .... My guess has been that there's some issue with port-forwarding or firewalling i'm missing, but i don't know what it is.
09:06.16ChannelZwell but your wifi goes through your NAT, yes?  And that worked
09:07.59flexibeastWell, no. The wifi is behind the NAT. So the phone connects to the wifi router, which connects to the Asterisk box, without ever going outside the LAN.
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09:08.35ChannelZbut you called me with it didn't you
09:11.08flexibeastOh, yes, in that instance, yes, the NAT was of course traversed.
09:15.06flexibeasti just tried setting the firewall to allow all traffic to the Asterisk box, and still it didn't work.
09:16.05robmalDisconnect the extension, launch tcpdump on your asterisk box, connect the ext and try to make a call.
09:16.09ChannelZyeah I think it's just your mobile data doing crazy things.
09:16.32ChannelZ(or linphone or whatever it is behaving badly)
09:16.56ChannelZI use CSipSimple on Android but haven't ever really tried it over mobile
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09:19.41flexibeast*nod*
09:20.30flexibeasti'm about to share some tcpdump output, but i've just noticed things like this in syslog:
09:20.44flexibeastasterisk[19918]: rc_avpair_new: unknown attribute 1490026597
09:20.50flexibeastIs that relevant?
09:22.14ChannelZI've never seen that before in my life
09:22.34flexibeastHeh, fair enough. :-)
09:27.06ChannelZI don't even see that in the source. I don't think it's coming from asterisk its self
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09:28.03flexibeastOkay, there's this: http://www.voip-info.org/boards/index.php?t=17493
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09:30.06Aamitwow
09:30.32flexibeastSo how do i unload that module, or make sure it's not loaded?
09:30.51ChannelZhmm interesting
09:31.38ChannelZnoload => cdr_radius.so    in /etc/asterisk/modules.conf
09:34.32flexibeastOkay, thanks. Sadly that only stopped the messages in syslog, rather than fixing the current problem. :-)
09:34.40flexibeastStill: one less bit of data to worry about.
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09:36.55ChannelZwonders why amazon is suggesting something called "Hairless Toys" to me
09:39.09flexibeastGah. Think i need to step back from this for a bit. %-}
09:39.09flexibeastChannelZ: Thank you for all your efforts, much appreciated!
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09:40.21ChannelZSure, good luck. I'd try a different SIP app for fun but otherwise I'd have to conclude your carrier is screwing around
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10:43.07flexibeastChannelZ: Sorry, lost connection for a bit there! Wanted to say thank you for all your efforts to help me. :-)
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21:59.12pcheroHi... I have a problem with transfer.
21:59.44pcheroWhen I type command like [sudo asterisk -rx "channel redirect SIP/test-02-0000002b test_context,s,1"]
22:00.01pcherothen asterisk just send BYE message.
22:01.08pcheroBut [sudo asterisk -rx "channel originate sip/test-01 extension s@test_context"] is working.
22:01.37pcheroHow can I transfer channel? Where should I look at it?
22:01.58pcheroAny suggestion is ok. Please give me a clue..
22:05.39*** join/#asterisk roler (~roler@unaffiliated/roler)
22:06.04[TK]D-Fenderthose 2 things have nothing to do wirth one another
22:06.32[TK]D-FenderOriginating a enw call .. is originating a new call
22:07.00[TK]D-Fenderredirecting a call means that channel has to actually be real and active on your system.has to
22:07.07[TK]D-FenderShow us the actual attempt
22:07.10[TK]D-Fender~pb
22:07.10infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
22:07.11[TK]D-Fender^^^^
22:10.33pcherohttp://pastebin.com/DtY29Sse
22:10.52pchero[TK]D-Fender: Hi, good to see you. This what I tried. :)
22:11.55[TK]D-FenderChannel 'SIP/test-02-00000038' successfully redirected to ani,test,40000
22:11.57pcheroHum... Need sip message?
22:11.59[TK]D-FenderSay it did it
22:12.12[TK]D-FenderAnd "sip how peers" proves nothing after.  Why are you even looking at that?
22:13.07[TK]D-FenderNext..... read the instructions.....
22:13.17[TK]D-Fender"core show help channel originate" <---------
22:13.27pcheroUm. ok. :)
22:19.24pchero[TK]D-Fender: Could you please look at this once? I just pasted all of sip debug message of test-02. http://pastebin.com/5bGdDH0R
22:19.39[TK]D-FenderRead the instructions.
22:21.26pcheroHm.. I still don't get it. originate was succeed and I can hear park music on my sip device. do I miss something?
22:22.19[TK]D-FenderREAD THE INSTRUCTIONS
22:22.21[TK]D-Fender^^^^^^^^^^^^^^^^^^
22:22.40[TK]D-FenderAgain Originate isn't the problem.
22:24.38pcheroUm.. at 216 line it say succeed. But next message at 219  line, asterisk send bye immediately. I don't know why..
22:24.52pcheroOk..
22:25.17[TK]D-FenderStop looking at the debug and ..... READ THE INSTRUCTIONS.
22:25.34[TK]D-FenderYou seem to be having difficulty with this aspect.
22:27.19pcheroHum.. I reading instruction about redirect. [core show help channel redirect].  But it's really hard to what's wrong with me. Can I get some example of redirect commands?
22:27.56[TK]D-Fenderit GIVES YOU EXAMPLES
22:28.21[TK]D-FenderUsage2: channel originate <tech/data> extension [exten@][context]
22:28.28[TK]D-FenderWhere do you see COMMAS in that?
22:28.42[TK]D-FenderOr 3 things after the word "extension"?
22:29.28[TK]D-Fenderexten@context <----------------
22:29.38pcheroOh.. I mean, not originate. I mean redirect.
22:30.17pcherooriginate ok. I can get the call. But problem is when I tried to redirect.
22:30.18[TK]D-Fenderseems to have bad aim today
22:30.31[TK]D-FenderShow us the dialplan to back it
22:30.41pcherook
22:30.47[TK]D-FenderSorry about that...
22:32.06pcheroIt's ok. :) never mind. I'm just want to say thank you for your attention.
22:32.37pcherothis is it. Nothing special it just one of default dialplans. http://pastebin.com/rtjiVFtB
22:33.18[TK]D-Fendermygalaxy*CLI> channel redirect SIP/test-02-00000038  ani,test,40000
22:33.25pcheroIn previous pastebin, http://pastebin.com/5bGdDH0R. Yes.
22:33.29[TK]D-FenderThere is no priority 40000
22:33.36[TK]D-FenderYou still have the order wrong
22:33.49[TK]D-FenderContext, exten, priority
22:34.09pcheroYes.
22:34.14[TK]D-FenderYou have no exten called "test" either
22:34.26pcherooh.
22:34.33[TK]D-FenderYou pointed it to a place that doesn't exist
22:34.43[TK]D-FenderALWAYS verify your targets
22:35.16pcheroI just tried "channel redirect SIP/test-02-0000003e  ani,3,40000"
22:35.19[TK]D-FenderIn fact there is nothing called "test" there anywhere
22:35.31pcherooh
22:35.32pcherowait
22:35.43[TK]D-FenderAlso don't pick such a supidly high priority number
22:35.54[TK]D-FenderYou are inviting running into a numbering limit.
22:36.27[TK]D-Fenderand 3 won't work either
22:36.34[TK]D-Fenderthat wil not match 3
22:37.52[TK]D-Fenderok, I have to head out.  pay attention to what your patterns actually match.
22:38.21[TK]D-Fenderheads out for the evening.,.
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22:56.48pchero:) Have a good dinner. :)
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23:04.21El-Aurianhi, i'm looking to demonstrate open source technologies to students as part of the document freedom initiative
23:04.35El-Aurianhttps://www.google.co.uk/url?q=https://documentfreedom.org/&sa=U&ei=fkoPVdOkK8fNygOU3oLIBQ&ved=0CBcQFjAA&usg=AFQjCNH7Fi9Z9SR7WkbdLFsPpGLDLh9Hxw
23:04.51El-Auriansorry, https://documentfreedom.org/‎
23:05.36El-Aurianone idea, I have is to get the students to download a piece of software to their mobile devices and stage a mass video conference
23:06.04*** join/#asterisk averythomas (~averythom@2607:5300:60:2d42::1)
23:06.19El-Aurianthis post seems to be exactly what I want - http://stackoverflow.com/a/9633381
23:06.41El-AurianI'm going to be in an auditorium with 200 students and a presentation system
23:07.01El-Auriancan anyone suggest any ideas as to how I might best achieve my goal?
23:07.37El-Aurianwe have FreePBX as our phone system, though I'm happy to host another server on the presentation laptop if thats easier
23:09.03WIMPyEl-Aurian: Are you blogging or is this going to become a question?
23:09.35El-Aurianadditionally, the phone software could ideally be designed to require as little configuration as possible; ideally the conference call number, and that's about it (I can create an SSID to a VLAN which only the presentation laptops (hosted video conferencing service) and wifi phones are attached to)
23:09.44El-AurianWIMPy: see 00:07
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23:12.00El-AurianSIP isn't an area I have much experience (I've inherited the FreePBX system), so I'm quite inexperienced. Obviously, if this were a production system I'd learn the system inside-out, but with it only being a possible 5 minute part in a presentation, I'm requesting suggestions and assistance
23:12.56El-AurianI just want to make Open Source interesting for the students, and mass Video Conferencing for free would be really fun for them and a good introduction
23:13.07WIMPyWell, Asterisk doesn't do real video conferencing. You can only have a picture following taklker thing.
23:13.53El-Aurianthat's fine. I just want them to be able to see themselves on the projection, and no doubt pull faces at each other
23:14.49El-Aurianreally, the last thing I want is to be inaudible
23:15.08WIMPyIf you want to know how far you can get with FreePBX, you have to ask in #freepbx.
23:15.53WIMPyBut using wifi for that ewxperiment will be the interesting bit, not Asterisk or the conference.
23:16.16El-AurianWIMPy: is freepbx the way to go, or can asterisk do the job? Do you know of a guide for setting up a system like this?
23:16.51El-AurianWIMPy: it certainly will. I'm relying on the fact that not everyone will have a phone, and maybe restrict the number of connections
23:18.32El-Aurianany ideas on the mobile app best suited?
23:19.15El-Aurianif DHCP or DNS can be used to centrally configure the SIP phones, then I can set this up because of the seperate VLAN
23:26.31WIMPyFreeBPX uses Asterisk, so it can't do anything Asterisk can't do. I don't know what FreePBX can do.
23:27.03El-AurianWIMPy: and Asterisk can do everything I've mentioned?
23:27.51WIMPyI don't knwo about mobile apps, either. But provisioning is usally not available on softphones. I haven't seen that option so far.
23:28.40El-Aurianok, thanks. would the apps I'm looking for be "SIP video conferencing"?
23:28.46WIMPyIt can do the simplest of video conferencing. What else do you want it to do?
23:28.59WIMPyNo, just a SIP phone.
23:29.12WIMPyWell, SIP video phone.
23:29.43WIMPyThe app wouldn't knwo about the conference.
23:30.02El-AurianWIMPy: if the simplist of Video Conferencing is unlimited video conferencing with "master" audio then that's ideal
23:30.29WIMPyNo. Audi can be mixed, video can't.
23:35.38El-AurianWIMPy: so I can't have 60 little windows around my big window, and only my audio played over the speakers?
23:36.00WIMPyno
23:36.27El-Aurianwhat can I have?
23:36.46WIMPyMixed audio and one picture.
23:37.12WIMPyEither the video from a maked user or from the last one starting to talk.
23:38.05El-Aurianwhat if I host 60 conferences and hand out a code to each group at the beginning and then layout the 60 calls using "tile windows"?
23:39.22WIMPyWhere is the conference bit in that?
23:39.43WIMPyYou can have any number of (unrelated) video calls.
23:40.32El-AurianWIMPy: I just want to know what the best option is. if not using video conferencing is the best option, then so be it
23:41.15El-Aurianin which case, what setup would you recommend to achieve this?
23:41.32El-Aurianis there a setup for better handling of mass video calls?
23:41.55WIMPyIf you want lots of videos on one screen, I think you need a real video conferencing app or something custom.
23:42.32El-Auriando you know of any that are either free or don't cost the earth?
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23:42.51WIMPyNo idea, sorry.
23:46.15El-AurianWIMPy: ok, without saying it's a bad idea, can you point me in the best direction for implementing this with an asterisk solution? (because I'm happy to test this with a similar number of staff before the actual lecture to test the various solutions)
23:47.00WIMPyI don't really see how Asterisk could help you there.
23:47.31El-Aurianbecause it's unable to handle the load of 60 video streams?
23:47.53WIMPyOff course you could use it to sep up a lot of normal video calls, but as SIP is P2P, you don't really need any server.
23:48.46WIMPyIt doesn't really care. It just passes on the video stream. So as long as you network card can handle it, you should be fine.
23:48.46El-Aurianok, so I just need a dobber of a laptop to act as video call host to handle the streams
23:49.11El-AurianWIMPy: so softphones connect to a wireless network
23:49.19WIMPyYes, but I don't see how you easily get a lot of calls on one desktop.
23:49.31WIMPyYou could maybe start lots of phones.
23:50.13El-Aurianhow do you mean 'start lots of phones'? as in hand out already connected phones at the beginning?
23:50.19WIMPyOr just not use phones at all. This really sounds like you want something dedicated to the specific task.
23:50.36WIMPyNo, on the presentation laptop.
23:51.04El-AurianOIC, yeh, that's what I had in mind if a mass video conference wasn't an option
23:51.18El-Auriancan you run me through the connectivity process?
23:51.26El-Auriansoftphones connect to a wireless network
23:51.35El-Auriancan DHCP give them an extension?
23:51.57El-Aurianor do I have to give them a number to assign their phone and then a number for the phone on my laptop?
23:52.00WIMPyThat's basic networking and has nothign to do with applications.
23:52.48WIMPyAnd if you don't require authentication, you shouldn't have to care about mucht.
23:52.53WIMPy-t
23:53.28El-Aurianyeh, authentication isn't a concern. obviously DHCP will assign them an IP address, but is there a SIP extension for DHCP?
23:54.12WIMPyYou don't need any.
23:54.38El-Aurianoic, their IP address is their extension number?
23:55.07WIMPySIP has usernames. But if you don't use authentication, it doesn't matter.
23:55.43El-AurianI'm not understanding how a softphone without a phone number can call another softphone
23:55.50robmalhttp://blog.tmcnet.com/blog/tom-keating/voip/auto-provision-voip-softphone-with-qr-code.asp
23:55.55El-AurianI assumed DHCP would have to hand out phone numbers
23:55.55robmalThis is a good idea.
23:56.31WIMPySIP is P2P. You effectively call an URI like sip:user@host.
23:58.16El-AurianI understand. so 001@payattention :P
23:58.20El-Aurianupto to 060@payattension
23:59.00WIMPyAs long as your DHCP server provides a DNS server that knows that host name.
23:59.03El-Aurianrobmal: that's that's perfect for a client solution (assuming it's open source)
23:59.25El-Auriancan phones not use NetBIOS?
23:59.43WIMPyI doubt so.

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