irclog2html for #asterisk on 20050107

00:00.05*** join/#asterisk david (~dcoulson@muffin.davidcoulson.net)
00:00.37TomLTripleF: there must be more than that
00:00.48TripleFyeop all breaks i just dont have kerlen source ;)
00:00.57TripleFprereq is Linux 2.4 kernel sources
00:00.58TripleF;)
00:01.04habakukChArLeS___, so what exactly is the problem? still the dtmf issue?
00:01.10TomLyou have to have kernel headers installed or you can't compile anything
00:01.24TomLTripleF: output 'rpm -q kernel-headers' ?
00:02.05TomLoh wait, they  don't do it that way anymore
00:02.31TripleFrpm -q kernel-headers
00:02.31TripleFpackage kernel-headers is not installed
00:02.38TomLyea not that
00:02.46TomLon fedora 1 do 'rpm -q glibc-headers'
00:02.55TripleFalso * is giving me in random times..   6 19:02:44 WARNING[14625]: chan_iax2.c:5307 socket_read: Error: Resource temporarily unavailable
00:02.55TripleFJan  6 19:02:44 WARNING[14625]: chan_iax2.c:5307 socket_read: Error: Resource temporarily unavailable
00:03.07bratnerIs there a way for * to run some AGI script on schedule(every 10 sec) while the channel (SIP) is active?
00:03.07TripleFthouands of those in  few secs.. it brings network down each time
00:03.11*** part/#asterisk telme_ (~teliax@sta-207-174-139-178.rockynet.com)
00:03.14TomLdamnit
00:03.26TomLon fedora 1 do 'rpm -q glibc-kernheaders'
00:03.30TripleF<PROTECTED>
00:03.30TripleFglibc-headers-2.3.2-101.4
00:03.42TomLwrong package again, damn my eyes :)
00:05.02TomLTripleF: 'rpm -q glibc-kernheaders' ?
00:05.06*** part/#asterisk bgreear (~Ben_Greea@evrtwa1-ar2-4-33-045-022.evrtwa1.dsl-verizon.net)
00:05.48denon~seen oej
00:05.51jbotoej <~oej@apollo.webway.se> was last seen on IRC in channel #asterisk, 4d 14h 20m 48s ago, saying: 'What's on that phone conference?'.
00:05.58TripleFrpm -q glibc-kernheaders
00:05.58TripleFglibc-kernheaders-2.4-8.36
00:06.05TomLhmm
00:06.19TripleFuname -r = 2.4.22-1.2174.nptl
00:06.36TomL[15:58] <TripleF> n file included from /usr/include/linux/module.h:20,
00:06.37TomL[15:58] <TripleF>                  from zaptel.c:44:
00:06.40TomLwhat's the rest?
00:06.57TripleFi need to find a way to log theres not enough buffer
00:07.11TomLjust the next line or two
00:07.20TomLis that the first error, or towards the end?
00:07.37TripleFfirst
00:07.38TripleF<PROTECTED>
00:07.38TripleFgcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net   -DSTANDALONE_ZAPATA -c zaptel.c
00:07.38TripleFIn file included from /usr/include/linux/module.h:20,
00:07.45TripleFthen
00:07.46TripleF<PROTECTED>
00:07.57*** join/#asterisk eKo1 (~bernd@63.245.57.70)
00:08.02TripleFok pastebin hold on
00:08.22eKo1i hate moving heavy servers. now my knees are weak
00:08.30TomLwhiner :P
00:08.40TripleFhttp://pastebin.ca/3836
00:09.37TomLfunky!
00:10.15*** part/#asterisk Cresl1n (~matt@216.207.244.186)
00:10.19TomLthat's RedHat for ya, gd compiler wreckers
00:10.20Mocit crazy
00:10.27TomLtry Slackware :)
00:10.31TripleFahahah i cant
00:10.38TripleFnext time im getting ufreebsd
00:10.42TripleFi jsut love that shit
00:10.45TripleFbetter then all out there
00:11.05TomLisn't bsd dead? *ducks*
00:11.11TripleFheu ?
00:11.16TripleFnah why
00:11.28TomLjust a little humor
00:12.12TomLcan you install kernel-source ?
00:12.43TripleFill try
00:12.45TripleFif i can find
00:12.49TripleFno luck last time
00:12.51TripleFbox is remote
00:12.51TripleFso
00:12.54TomLtried yum?
00:12.56TripleFalso what this error ?
00:13.00MocTripleF, how is the weather in QBC ?
00:13.04TripleF<PROTECTED>
00:13.05TripleFJan  6 19:02:44 WARNING[14625]: chan_iax2.c:5307 socket_read: Error: Resource temporarily unavailable
00:13.27TomLyou get that at start up, or after the system's been up for awhile?
00:13.35TripleFMoc evolution email main page says...  lisgh snow 0
00:13.44TripleFTomL after a while ..random
00:13.51MocTripleF it crazy in MTL..
00:13.52TripleFmemory leak ?  seems to be network based
00:13.55TripleFlike polling etc
00:14.00TomLcan you tie it to any particular event, like picking up a softphone extension or something?
00:14.06TripleFnothing
00:14.09TripleFjsut like that
00:14.11TomLbizarre
00:14.21TripleFmaybe .. hmm i coudnt know
00:14.21TomLwhile the system is IDLE?
00:14.36TripleFi could run in gdb and trigger on that line of cha_iax2.c
00:14.39TripleFIDLE
00:14.50TomLi.e. no calls in process
00:15.06TripleFno
00:15.15TripleFCPU goes 100% eth0 100%
00:15.20TripleFfor like 10 seconds
00:15.27TomLwhoa
00:15.29TripleFbreaking all network activity
00:15.36TripleFPTG123 i need to fix for calling lol
00:16.01TripleFi get this form /var/log/mess
00:16.02TripleF<PROTECTED>
00:16.18TomLnever seen anything that causes eth0 100%
00:16.18PTG123huh?
00:16.29TripleF<PROTECTED>
00:16.55eKo1bsd is THE server os
00:16.55eKo1slack is THE desktop linux distro
00:17.02bratner<PROTECTED>
00:17.12TripleFits a ekkernelprob
00:17.12TripleF5 17:10:25 pobox kernel: NET: 47 messages suppressed.
00:17.30TripleF<PROTECTED>
00:17.38TripleFathlon based maybe
00:18.54TripleFdarn it just breaks my network
00:19.00TripleFmaybe that not the source
00:19.07harryvvtripleF, im off to work but might talk to you tomarro
00:19.08harryvv:)
00:19.16TripleFk harryvv
00:19.37TripleFbratner not sure.. tikiwiki ?
00:20.35bratnerTripleF: tried that , but my conclusion was that i need to hack the Dial() command to add more functionality.
00:21.46eKo1bratner: wouldn't it be better to check the credit before the call is made and if the caller has less than 3 mins. or so, the call doesn't go through
00:22.18TomLeKo1: so if I have 4 minutes left, I made a call and stay on for 1 hour... then what?
00:22.48eKo1true, true...hmm
00:23.21TripleFhmmm ok before the clal poll cdr.. calc time left..make a call timer for each call
00:23.36TripleFmod the dial function to have a time limit or 0 for infinite
00:23.40TripleF;)
00:24.00eKo1i think there is a way to tell asterisk how long a call should last
00:24.08bratnerTripleF : the problem is that one subscriber can have several calls. eg: share his "userid" and "passwd" with family ...
00:25.25postelok, brainstorm me a bit, i know that calling X costs Y per min and calling Z costs H per min, i can make a shell script that displays true time cost information in a couple of lines no prob, how do i feed the thing now to the xml parser of the cisco's 79xx so the people that use them get real time cost info on the calls they're making?
00:25.26bratnereKol: if i limit a subscriber to one call at a time. then i can use the AubsoluteHangup time-out right after Dial() connects
00:25.51TripleFTomL any other idea ?
00:26.12CpuID2hey ppls, anyone had success with cisco 7940s and asterisk with sip? out of curiosity, which firmware version?
00:26.19TripleFbasicaly ..ah !
00:26.22TripleFthink i know
00:26.34TripleFok socket_read: Error: Resource temporarily unavailable
00:26.42TripleFmeans cant bind to socket or poll or whatever
00:26.50*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
00:27.01postelCpuID2: POS3-07-3
00:27.02TripleFJan  5 17:10:25 pobox kernel: NET: 47 messages suppressed
00:27.07TripleFmeans its kernel based
00:27.08CpuID2so sip 7.3?
00:27.12postelyeap
00:27.14eKo1bratner: that is a tough one. maybe when more calls are made from the same 'account' have the timer speed up
00:27.18TripleFso kerlnel ports or socket max is my problem maybe
00:27.22CpuID2what version did you start work on the phone btw?
00:27.28eKo1that's the only thing i can think of
00:27.28postelCpuID2: 5.4
00:27.30CpuID2and did you step it up one version at a time or anything?
00:27.31CpuID2ah
00:27.40CpuID2im on sip 6.0 atm
00:27.44CpuID2still no success here
00:27.49CpuID2all the registration is setup right and all
00:27.53CpuID2i can telnet the phone too
00:27.58postelCpuID2: yeap, after you start with the signed images (which is 6.3) you cant go back
00:27.59CpuID2the phone can ping the local asterisk server
00:28.00bratnerTripleF , eKol: thanx for your help.
00:28.14CpuID2yet, its like the registration attempts arent even getting to the asterisk serve
00:28.16CpuID2server
00:28.28postelCpuID2: 6.x had lot of trouble, upgrade dude
00:28.32CpuID2np
00:28.33CpuID2sec
00:28.35CpuID2btw
00:28.48CpuID2with 7.3, theres a P003 loader, but a P0S3 firmware, is that correct?
00:28.55postelit is yeah
00:28.59CpuID2just looked a little weird :)
00:29.01TomLTripleF: the network problem is probably not caused by *, its probably reporting errors because it can't use the network, either
00:29.05CpuID2ok np
00:29.19CpuID2am i best putting P003 into my OS79XX.CNF you think? or P0S3?
00:29.28postelCpuID2: it would load the universal loader and the sip image over it
00:29.31TripleFis that ok ? net.ipv4.ip_local_port_range = 32768    61000
00:29.35CpuID2ah yep
00:29.38CpuID2so give it P003
00:29.43CpuID2and it should pickup the sip image instead?
00:30.04postelput the sip image version and it would pick the loader ;)
00:30.40CpuID2ah np
00:31.03postelit would reboot something like twice, then you're up and running
00:31.33CpuID2nice
00:31.34CpuID2sec
00:31.44postelCpuID2: if you got trouble going directly to 7.3 try every one before them and climp up the images
00:32.01postelCpuID2: there is some nice info on the wiki if you fuck up
00:32.08CpuID2yep
00:32.17CpuID2let me try this first of all
00:32.23CpuID2you got any specific wiki pages worth the read?
00:32.52CpuID2ive spent like 8 hours reading over stuff so far
00:32.54postelCpuID2: yeap, the one saying "so u fucked up your 79xx trying to load the sip image" or something along this lines
00:32.56CpuID2some stuff looked relevant...not all
00:32.57CpuID2ah yep
00:32.58CpuID2that one
00:33.07postels/this/these
00:33.13CpuID2i had that last night...yet it seems my image didnt like the SEP(MAC).cnf.xml file
00:33.17CpuID2not sure why tho
00:33.25CpuID2let me just try this, sec
00:33.39*** join/#asterisk Vexamus (~chatzilla@pcp05504579pcs.owngsm01.md.comcast.net)
00:33.44Vexamusguess what kids... I'm back.
00:34.00Vexamuswith another question.
00:34.09CpuID2ok wtf
00:34.17Vexamus?
00:34.18CpuID2the phone keeps trying to find P0S3-07-3-00.sbn
00:34.24CpuID2yet theres only a P003 .sbn file
00:34.27CpuID2rename it? :)
00:34.37TripleFok ill try caling
00:34.46postelCpuID2: NOOO
00:34.56CpuID2ok...tell it to load P003 instead
00:34.56CpuID2?
00:35.08CpuID2and then itll use the P0S3 image automatically?
00:35.53postelCpuID2: what have u got in the OS file?
00:35.53VexamusCan someone please tell me if it's possible for asterisk to notify it's users of voicemail audibly on the phone using the chopped dialtone upon pickup?  If so, where do I configure that?
00:35.53CpuID2right now, P0S3-07-3-00
00:35.53CpuID2it didnt get far tho
00:35.53postelwell, thats what i got too and it picks it up
00:35.53CpuID2file not found on the tftp server, normal bootup from there
00:35.54*** join/#asterisk mikes2277 (~mike@wireless-206.222.58.98.bwsys.net)
00:36.10CpuID2ive got P0S3-07-3-00 .sb2 and .loads
00:36.24CpuID2as well as P003-07-3-00 .bin and .sbn
00:36.43mikes2277anyone know if asterisk can support direct tv or dish networks modem passthrough?
00:36.46postelthats all you need
00:36.51CpuID2one good sign, i got an asterisk logo with a penguin on the phone now lol
00:37.17CpuID2postel: well it tries to go for the .sbn, but with a P0S3 filename
00:37.23postelCpuID2: heh, i got the same one
00:37.31*** part/#asterisk Vexamus (~chatzilla@pcp05504579pcs.owngsm01.md.comcast.net)
00:37.31TomLVexamus: yes it does, you have to set up the callerid information properly
00:37.34TomLdoh
00:37.40CpuID2had the same issue here upgrading from sccp 5.0 originally
00:37.52CpuID2thats why i tried going an earlier sip
00:37.55eKo1finally, got my * server on its own vdsl connection
00:37.59CpuID2is there anything between sip 6.0 and sip 7.3?
00:38.02postelCpuID2: well, touch the SIPmac_address.cnf file and  put the image_version: there
00:38.05eKo1no my calls should be ultra clear
00:38.21CpuID2postel: already had it set there
00:38.22TomLmikes2277: don't even know what that is
00:38.27CpuID2as P0S3-07-3-00
00:38.38postelCpuID2: wait, ur running 6.0?
00:38.41CpuID2ya
00:38.46TomLmikes2277: seems to me that the huge latency on satellite would make it useless for VoIP
00:38.50postelCpuID2: u CANT go to signed images from 6.0
00:38.54CpuID2and i checked, it did pull the SIP(mac) file, so yea
00:38.55CpuID2oh what
00:39.00postelCpuID2: u need to go to 6.3 first
00:39.02CpuID2ah
00:39.02TomLmikes2277: severe case of "Hello, over"
00:39.09CpuID2ffs, i dont have 6.3 with me :(
00:39.16frank_sbr<ta[i]nted> are you there?
00:39.19CpuID2and i dont have a cisco login, arr
00:39.27mikes2277I need to make dish networks and direct tv's modem lines go accross asterisk
00:39.35TomLunlikely in the extreme
00:39.47denonmikes2277: not gonna happen.
00:39.57mikes2277supposedly vonage has it working
00:40.22TomLsupposedly 'vonage' is a synonym for 'VoIP' *ducks*
00:40.51mikes2277well, my customer just dropped me because vonage made it work
00:41.02denonmikes2277: Aseterisk's latency is too high .. Ive battled with this ..
00:41.02mikes2277so I would say they are smarter they we are
00:41.16mikes2277so do I just need to do a reinvite?
00:41.25denonwe? have you submitted any code to cvs? :)
00:41.26*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
00:41.38TomL(heh)
00:41.54denonthis whole software dsp design .. I dunno if it's even possible to have low enough latency to handle any high-speed data calls
00:42.02denonI mean, you can do up to like 9600bps
00:42.08denonbut that's about the extent
00:42.10mikes22779600 would be fine
00:42.15denon9600 works fine
00:42.24denonwell .. with a few variables
00:42.27mikes2277how do you get the receiver to force to 9600
00:42.43denonyou dont need to, it'll fallback to whatever conditions support ..
00:42.49TomLwhat codec you gonna use to get decent voice with 9600 baud and 500ms latency on the first hop?!
00:42.50mtqhmikes: it talkes down...but dtv will not talk down that far
00:42.50denonbut many people wont allow anything under 14.4 anymore
00:42.57mikes2277DirectTV tech support said something about Vonage put s *99 or something like that in the dial string
00:43.08mikes2277so it must put their box into some special mode
00:43.16denonprolly disables echo can
00:43.26denonAsterisk does that automatically when it sees a data call
00:43.34denonbut you can do it manually if you prefer, in the configs
00:43.39mikes2277how?
00:43.45pcmmikes2277: most likely that *99 will force the ulaw/alaw codec
00:43.52denon~google asterisk echo cancel bridged
00:44.01TomLechocancel=no in zapata.conf
00:44.06pcmmikes2277: I guess you can check with the sniffer
00:44.10mikes2277I already do that in the sip.conf file (force it to G.711u)
00:44.22mikes2277I don't use the zaptel boards
00:44.29mikes2277I use a TNT
00:44.33denonare you talking ulaw on both sides?
00:44.35denonno transcoding
00:44.39TomLdunno what that is
00:44.43mikes2277its not
00:44.44denoncause that itself will add latency
00:44.54mikes2277Lucent TNT, big ass gateway (DS3)
00:44.58mtqhuse zap hardware
00:45.03TomLah
00:45.07mikes2277tried, it sucks
00:45.17mikes2277lots of problems and digium was less than helpful
00:45.28denonwhat kinda problems?
00:45.28pcmmikes2277: if you allow reinvites the TNT should talk to itself or to other SIP gateway directly ... (the voice)
00:45.31denonits pretty generic stuff
00:45.44TomLwhat's the physical connection from DirecTV modem passthrough <-> TNT?
00:45.46TripleFhey
00:45.52TripleFwondering
00:45.53TomLand where are you running *, on the TNT?
00:46.04mikes2277IRQ issues on the quad boards that never showed up on the single port versions
00:46.15mikes2277no, TNT is just like a Cisco
00:46.19TripleFif there a message .gsm about calling channel secured ?means theres encryption avail /?
00:46.20mikes2277big Cisco
00:46.32denonah, yeah . .the irq issues are another matter
00:46.49TomLyea, so where does that fit into the picture connecting * to DirecTV? (confused)
00:46.53mikes2277Azacal <--> Internet <--> Asterisk <--> TNT <--> PRI To PSTN
00:47.12TomLAzacal?
00:47.15mikes2277DirecTV <--> Azacal
00:47.23mikes2277Its like a sipura 2100
00:47.32mikes2277but they speak english
00:47.38TomLheh
00:47.45pcmare you sure that azacal->asterisk is ulaw/alaw also ?
00:47.49denonwith all those hops, I'm surprised you can get a voice call out
00:47.50mikes2277yes
00:48.20mikes2277the RTP goes through only Asterisk and then to the TNT so I can do NAT traversal
00:48.41pcmmikes2277: is azacal box changing the gains ?
00:48.49mikes2277Faxing over G.711u works great on my network (without using T.38)
00:49.01*** join/#asterisk UPMeduardo (~UPMeduard@rpv13.dit.upm.es)
00:49.02mikes2277but I can't get the sat receivers to work
00:49.14denonI'd bet the faxes are neg'ing at 9600 tho
00:49.19UPMeduardohi all!
00:49.38mikes2277I really dont know what speed they run at, but they do and they seem to be pretty fast
00:49.57denonmike: even with 15-20 page faxes?
00:50.14mikes2277I did notice that the ITU has defined MoIP (Modem over IP)
00:50.22tzangermikes2277: over your own network, sure
00:50.28mikes2277but I dont see anything that supports it, heck Asterisk doesnt even do T38
00:50.30tzangerover the internet I get about a 50% success rate
00:50.37denonT38 is .. an issue.
00:50.52denonnot gonna happen in asterisk, I dont think, unless someone funds it
00:50.56mikes2277I get 99% sucess over my cable modem from my house, which is many hops away
00:51.03UPMeduardoCan anybodu help me?
00:51.11mikes2277The TNT's DSPs are pretty darn good
00:51.19UPMeduardoi'm having problems compiling chan_capi
00:51.36mikes2277I do T.38 by putting SER in front of Asterisk
00:51.47UPMeduardotaiwan:/usr/src/chan_capi-0.3.5 # make
00:51.48UPMeduardogcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g  -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686  -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO   -c -o chan_capi.o chan_capi.c
00:51.48UPMeduardocc1: warning: changing search order for system directory "/usr/include"
00:51.48UPMeduardocc1: warning:   as it has already been specified as a non-system directory
00:51.48UPMeduardochan_capi.c: In function `capi_new':
00:51.59UPMeduardochan_capi.c:1073: structure has no member named `callerid'
00:52.00UPMeduardochan_capi.c:1074: structure has no member named `dnid'
00:52.00UPMeduardochan_capi.c: In function `pipe_msg':
00:52.00UPMeduardochan_capi.c:1724: structure has no member named `dnid'
00:52.01mikes2277reroute the calls directly to the TNT instead of going through Asterisk
00:52.02UPMeduardochan_capi.c:1724: structure has no member named `dnid'
00:52.04UPMeduardochan_capi.c:1724: structure has no member named `dnid'
00:52.06UPMeduardochan_capi.c:1724: structure has no member named `dnid'
00:52.08*** kick/#asterisk [UPMeduardo!denon@synapse.subneural.net] by denon (use a pastebin)
00:52.11denongood grief.
00:52.20mikes2277how annoying
00:52.42mikes2277anyway, where were we
00:52.43Silik0nnothing nicer then telling customers you cant fix their fones cause the 25pair burrial cable going to their building is fuckt
00:52.46*** join/#asterisk UPMeduardo (~UPMeduard@rpv13.dit.upm.es)
00:52.57pcmmikes2277: does SER do T.38 ?
00:53.21mikes2277it passes the SIP messages to a Cisco gateway or a TNT, so yes.
00:53.27mikes2277works great
00:53.27UPMeduardosorry
00:54.37mikes2277perhapse I should try redirecting the receivers modem call directly to the TNT and bypass Asterisk
00:55.03denonprobably
00:55.10denonreinvites to the sip device from *
00:55.11pcmmikes2277: then you could see if this is going to work anyway with that FXS box
00:55.17Mocanyone know a h323 client for windows ?
00:55.18mikes2277perhapse the direct RTP stream from the Azacal to the TNT woudl help a bit
00:55.28denonMoc: there are tons ..
00:55.35Mocfree ;)
00:55.39denonthe old standby used to be netmeeting
00:55.43mikes2277ok, thanks guys, see ya later
00:55.48denon~google free windows h.323
00:55.53Mocwell netmeeting work with * h323 ?
00:56.02denonnfi, I dont use h.323 at all :)
00:56.20denonI havent for probably 5 years
00:56.39TripleFif there a message .gsm about calling channel secured ?means theres encryption avail /?
00:56.45MocI want to test my h323 setup, I got a Polycom SoundStation IP 3000 on the way
00:57.00UPMeduardoMoc, netmeeting
00:57.36*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
01:03.24*** join/#asterisk GmLB (~danny@155.246.221.23)
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01:07.25TripleFok problem
01:07.42TripleFexten => rebound,6,SetCallerID(${CALLERIDNUM}|a)
01:07.42TripleFexten => rebound,7,DIAL(${IAXPLACE}/${outnumm})
01:08.00TripleFi see callerid of IAX account.. nut the ZAP one
01:08.07TripleFis there a way to set or overwride ?
01:08.24*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
01:08.28TripleF${CALLERIDNUM hold the one to use
01:08.30*** join/#asterisk bjohnson (~bjohnson@ip169-172.dsl.istop.com)
01:08.59TripleFso i gess SETCALLERID is not the right one
01:09.03TripleFany suggestions ?
01:09.16CpuID2so anyone else here with a cisco login that wants to send me over a sip 7.0 image? :)
01:09.23TripleFSetCIDNum ?
01:09.38CpuID2i so wish i just had my own login already, stupid people on christmas/new year holiday still
01:10.04TripleFlol
01:10.56TripleFhmm
01:10.59TripleFsone one ?
01:16.59redder86people are still on holiday?
01:17.27TomLyea, I'm still waiting for several vendors to call me back.. not Cisco though
01:18.19redder86wow, I thought everyone would be back to work by now.
01:18.31TomLnot in the corporate world
01:18.58redder86Start the New Year right!  Take half of your vacation up-front.  :-)
01:19.07TomLlots of lazy ppl in this world.  thankfully there only needs to be a few ppl who actually want to work to make the world go round :)
01:19.08iCEBrkrhrrm.
01:19.21iCEBrkrHow the heck can I debug my BT100 not registering?
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01:20.16TomLTripleF: question indecipherable, not enough information
01:20.49ds2anyone know about ``shoreline'' boxes?
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01:21.14TomLiCEBrkr: "sip debug" on * console?
01:21.19TripleFwhat ?
01:21.32iCEBrkrTomL: I don't think it's even getting THAT far.
01:21.43iCEBrkrI'm remote from my Asterisk box behind NAT :(
01:21.48TomLoh, I read 'reregistering'.  my bad.
01:21.52iCEBrkrhehe
01:22.08iCEBrkrI think I have all my port forwarding set right
01:22.23iCEBrkrIt's goofy, cuz everything was fun up until RoadRunner hosed the other day...
01:22.29TomLTripleF: what are you trying to do, and how is it failing, exactly?
01:22.39TripleFi mean .. using callerid form person who called to my ZAP ... so when i dial ( IAX/some    it s uses that on outgoing party calleird.. instead of the IAX calleird
01:23.04TripleFTom can u go conference 444 ?
01:23.09TomLunlikely
01:23.14TripleFah
01:23.15TripleF?
01:23.31TripleFunlikely conf or my situation
01:23.46TomLunlikely I can go conference 444 :)
01:23.51TripleFwhys that
01:24.00TripleF;) easier to explain my situation
01:24.09TomLI'm multitasking
01:24.33TripleFk
01:25.04redder86ds2: I know a little about shoreline
01:25.05TomLso someone calls you via analog PSTN -> asterisk, and you are transferring them to an IAX channel?
01:25.14TripleFyep
01:25.32TripleFi need the called party after the iax to see original calleird from pots
01:25.38TripleFwich i set right before senidng
01:25.47redder86ds2: what do you want to know?
01:25.48TomLdo you get callerid on POTS ok? :)
01:25.54TomL(have to ask)
01:25.59TripleFexten => rebound,6,SetCIDNum(${CALLERIDNUM}|a)
01:26.00TripleFexten => rebound,7,DIAL(${TELIAX}/${outnumm})
01:26.01TripleFyep
01:26.07blaisen1anybody know much about possible upcoming regulation of VOIP in canada?
01:26.08TripleFline 6 has one i should passs
01:26.10TomLhow do you know you get the callerid ok?
01:26.28TripleFon zap in local
01:26.29TripleFexten => s,3,NoOp(${CALLERIDNUM})
01:26.32TripleFi see it ok
01:26.45TomLthat's a different exten
01:26.46TripleFthen i send to rebound,
01:26.48TomLs vs. rebound
01:26.51TripleFah
01:26.57TripleFso i need to make it a var
01:27.11TripleFvarsa gre global in same context right ?
01:27.38TomLat the very least print out ${CALLERIDNUM} once in you're in rebound so you can see what it is
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01:27.54TripleFexten => s,1,wait(2)
01:27.54TripleFexten => s,2,SetVar(whocalled=${CALLERIDNUM})
01:28.36TripleFtrying
01:30.15TripleFtype nopop
01:30.15TripleFlol
01:30.21TripleF* shoudl continue on erros
01:30.28FaithfulHey my extesions are 6XXX and unless I put exten => _6XXX,1,Dial(SIP/${EXTEN},20,rtT) in the [incoming] context it can't dial other internal extension
01:30.55TripleFworking thansk
01:31.05TomLnp :)
01:31.09FaithfulI have a context [internal] with exten => _6XXX,1,Dial(SIP/${EXTEN},20,rtT) in it but that doesn't seem to let me dial other phones
01:31.32TomLFaithful: did you 'include => internal' somewhere appropriate?
01:32.18FaithfulTomL: like where?
01:33.13TomLthe starting context as defined in sip.conf, perhaps?
01:33.32Faithfulno I don't think so
01:33.35TomLusually 'default'
01:33.35Faithfulcheck
01:34.14FaithfulI only thought contexts applied in extensions.conf
01:34.31Faithfulwell those contexts anyway
01:34.36TomLin sip.conf you tell the SIP channel module what context to start in when a SIP context is initiated
01:34.49TomLSIP call
01:35.18TomL(not SIP context)
01:35.32TomLso unless you changed it, in sip.conf you have "context=default" in [general]
01:35.38blaisen1does anyone know how to offer 911 over VOIP?
01:35.47TomLso in [default] in extensions.conf, you 'include => internal'
01:35.56syslodblaisen1: What type of service do you have?
01:36.03redder86blaisen1: just direct the 911 extension to the direct number for the dispatcher
01:36.25blaisen1syslod: well it's a larger private system that a bunch of people are using, all over the place (well, the same province)
01:36.37syslodredder86: That technically would be illegal now.
01:37.10blaisen1syslod/redder86: i'm wondering if there is a "provider" where I can registered DIDs and send them the 911 calls and they would route it to the correct 911 office or whatever
01:37.13syslodIn the US. blaisen1: I assume you aren't in the US?
01:37.24syslodIts not that simple.
01:37.33blaisen1syslod: Canada but I'm interested in how it may work in the US anyway
01:37.52FaithfulTomL: quite simple really, thanks :-)
01:38.00redder86syslod: directing 911 to the dispatcher is illegal?
01:38.05syslodUS uses E911.  New ruling requires E911 support if its a private pbx regardless.
01:38.19TomLnp :)
01:38.55blaisen1syslod: some day I would like to sell VOIP and new regs allegedly coming in February from the CRTC indicate 911 service will be a must for voip providers, so i'm just kinda wondering how I'm going to do that..
01:38.59TomLsyslod: so you just arrange extension "911" to dial "911" on an outgoing trunk?
01:39.09syslodredder86: If you direct a call lets say from a DID you own yes it could be.  You are sending the wrong E911 info.  For example you purchase a PRI with DID to your building. Someone make a call to 911 and hangs up where info is the 911 dispatcher gonna get?  The wrong info.
01:39.13cp5anyone use swissvoice phones>
01:40.08denonsyslod: 911 gets ANI.
01:40.09blaisen1syslod: so is there a master DID to address database out there you can get access to .. and is ita  pay-per-did thing to register locations...??  I've heard something about some "Intrada" company or something
01:40.16syslodTomL: I didn't say you couldn't route I just said its in a very grey area and current legistlation has been put forth to force correct E911.
01:40.47syslodYou can't get access to the E911  and thats the problem.  THats for the realm of SS7 based CLECS with ISUP trunks.
01:41.00TomLwhich effectively makes all private PBX's the property of the US Govt.  lovely.
01:41.12syslodMost LECS, if not all, will not give you access to the E911 system.
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01:41.49syslodMy point being be careful in thinking that just cause you get the 911 call to the office it may not be enough.
01:41.51czerothere are 'VoIP PRI's' here u can get in Canada that you can set up with proper 911
01:41.56FaithfulTomL:  now the [incoming] doesn't because I removed it from [default]
01:42.21TomLFaithful: why would you do that?
01:42.27syslodWe sell some PRI's with generic E911 but we've been notified that this will have to change by end of year.
01:42.32TomL[incoming] probably has nothing to with [internal]
01:42.50TomLto ^do with
01:42.52syslodSame thing happending with E911 and cell phones.
01:43.17czeroe911 with cell is jsut a requirement so the gov't can track your ass not a for your safely issue
01:43.20czeroIMO :)
01:43.40TomLnot sure how E911 works with a cell.. what do they GPS your ass?
01:43.46syslodTrack safety it really doesn't matter cause they've already made it law.
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01:44.34czerosyslod but not all handsets support it
01:44.50syslodThe systems I've seen do GPS with equiped phone and without transmit ID to locate with trianglation.
01:45.08syslodTomL said GPS not me.
01:45.19TomLassuming there are 3 towers within range of the phone?
01:45.25TomLor possibly even 2
01:45.41syslodIt give its best guess which is usually pretty damn good from what I've seen.
01:45.42blaisen1some radioshack guy tried to tell me the gps receiver is built into the cell phone
01:45.44TomLbut with only 1... they're screwed, and that's probably a very large percentage of calls
01:46.15czeroyeah they can 'meaausre' wehre the call is if they have multi towers talking to your handset
01:46.18syslodActually there are figures and they say that its more like 90% of 911 calls will be accurate.
01:46.31TomLinteresting
01:46.38syslodWith two towers and a road they can do amazing things.
01:47.13FaithfulTomL:  my [internal] has include => outgoing  so if I now put both include => internal and include => incoming in [default] people dialing in can dialout
01:47.22czerosyslod yeah casue the rd gives your a 3rd refrences point
01:48.03syslodIn most 911 call areas..... Where things happen that is, you're very likely to be around alot of towers.  ALso its not just the tower with most systems they can track the sectored antenna you are on and get direction and distance making even one tower pretty effective.  I wouldnt want to play hide and seek with someone with this.
01:48.17TomLFaithful: you probably don't want 'include => outgoing' in [internal] if its already in [default]
01:49.01FaithfulTomL: yes but then how to make [internal] dialout
01:50.08PTG123hey can you guarantee the size of a udp packet when you get it on the other end?
01:50.25TomLUDP = no guarantee of anything
01:50.38czeroif u set the DF bit and peopel honor it
01:50.39PTG123well in other words, will it get broken up?
01:51.01TomLczero: its the "and" part of that thats the trick :)
01:51.19PTG123well who has to honor it
01:51.22PTG123just the receiver
01:51.24PTG123or th enetwork?
01:51.28TomLFaithful: not sure what you're asking
01:51.52TripleFudp .. ?
01:51.56TripleFjbot: UDP
01:51.57jbotudp is probably only PlayerUpdate, ShotBegin, ShotEnd and GMUpdate
01:52.03TripleFlol
01:52.23TripleFbasically udp is sending packet in any order firs tocme first sever and reassembling i guess
01:52.40TripleFso u loose pack #6 u rerequest it to reasseble all ?
01:52.54TripleFor iim mistaken
01:53.01TomLnot at layer 2 you don't
01:53.07TomLor layer 3 even
01:53.27PTG123no doesn't really work that way
01:53.31PTG123however you can loose packets
01:53.33TomLUDP has no guarantee of delivery, no checksum, and no handshaking
01:53.34PTG123and you won't know it :)
01:53.37PTG123depending on your protocol
01:53.58TomLif you lose packets, your application better damn well handle it itself
01:54.26TomLif you want the network to guarantee delivery, use TCP
01:55.00TripleFTomL or FEDEx
01:55.03TripleF;)
01:55.12TripleFon a slow network fedex is faster
01:55.22TomLFaithful: [incoming] is usually tied to your inbound PSTN trunks or VoIP trunks, not your extensions
01:55.39TomLi.e. not your FXS/SIP/IAX phones
01:56.16frank_sbrDoes anyone here have run Asterisk on Linux 64 bit?
01:57.21PTG123i don't need delivery guarantee
01:57.28PTG123however i do need uniform size packets
01:57.48TomLhmm
01:58.00TomLfragmentation and reassembly a problem?
01:58.14TomLusually if you send a packet of UDP, it'll either get delivered whole or not at all
01:58.22TomLhowever on a noisy circuit this isn't necessarily true
01:58.37TomLand there's no error-checking on UDP by the network
01:58.48TomLyour appllication will have to know what the packet size is and check for it
01:58.59PTG123might be better off using tcp.. .. just means i will have to serialize my protocol
01:59.06PTG123not sure if that is an issue or not
01:59.09TomLand implement an application-layer protocol for requesting retransmits
01:59.09FaithfulTomL: I put exten => _6XXX,1,Goto(internal) in incoming
01:59.24TomLFaithful: don't do that :P