00:00.05 | *** join/#asterisk david (~dcoulson@muffin.davidcoulson.net) |
00:00.37 | TomL | TripleF: there must be more than that |
00:00.48 | TripleF | yeop all breaks i just dont have kerlen source ;) |
00:00.57 | TripleF | prereq is Linux 2.4 kernel sources |
00:00.58 | TripleF | ;) |
00:01.04 | habakuk | ChArLeS___, so what exactly is the problem? still the dtmf issue? |
00:01.10 | TomL | you have to have kernel headers installed or you can't compile anything |
00:01.24 | TomL | TripleF: output 'rpm -q kernel-headers' ? |
00:02.05 | TomL | oh wait, they don't do it that way anymore |
00:02.31 | TripleF | rpm -q kernel-headers |
00:02.31 | TripleF | package kernel-headers is not installed |
00:02.38 | TomL | yea not that |
00:02.46 | TomL | on fedora 1 do 'rpm -q glibc-headers' |
00:02.55 | TripleF | also * is giving me in random times.. 6 19:02:44 WARNING[14625]: chan_iax2.c:5307 socket_read: Error: Resource temporarily unavailable |
00:02.55 | TripleF | Jan 6 19:02:44 WARNING[14625]: chan_iax2.c:5307 socket_read: Error: Resource temporarily unavailable |
00:03.07 | bratner | Is there a way for * to run some AGI script on schedule(every 10 sec) while the channel (SIP) is active? |
00:03.07 | TripleF | thouands 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) |
00:03.14 | TomL | damnit |
00:03.26 | TomL | on fedora 1 do 'rpm -q glibc-kernheaders' |
00:03.30 | TripleF | <PROTECTED> |
00:03.30 | TripleF | glibc-headers-2.3.2-101.4 |
00:03.42 | TomL | wrong package again, damn my eyes :) |
00:05.02 | TomL | TripleF: 'rpm -q glibc-kernheaders' ? |
00:05.06 | *** part/#asterisk bgreear (~Ben_Greea@evrtwa1-ar2-4-33-045-022.evrtwa1.dsl-verizon.net) |
00:05.48 | denon | ~seen oej |
00:05.51 | jbot | oej <~oej@apollo.webway.se> was last seen on IRC in channel #asterisk, 4d 14h 20m 48s ago, saying: 'What's on that phone conference?'. |
00:05.58 | TripleF | rpm -q glibc-kernheaders |
00:05.58 | TripleF | glibc-kernheaders-2.4-8.36 |
00:06.05 | TomL | hmm |
00:06.19 | TripleF | uname -r = 2.4.22-1.2174.nptl |
00:06.36 | TomL | [15:58] <TripleF> n file included from /usr/include/linux/module.h:20, |
00:06.37 | TomL | [15:58] <TripleF> from zaptel.c:44: |
00:06.40 | TomL | what's the rest? |
00:06.57 | TripleF | i need to find a way to log theres not enough buffer |
00:07.11 | TomL | just the next line or two |
00:07.20 | TomL | is that the first error, or towards the end? |
00:07.37 | TripleF | first |
00:07.38 | TripleF | <PROTECTED> |
00:07.38 | TripleF | gcc -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 |
00:07.38 | TripleF | In file included from /usr/include/linux/module.h:20, |
00:07.45 | TripleF | then |
00:07.46 | TripleF | <PROTECTED> |
00:07.57 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
00:08.02 | TripleF | ok pastebin hold on |
00:08.22 | eKo1 | i hate moving heavy servers. now my knees are weak |
00:08.30 | TomL | whiner :P |
00:08.40 | TripleF | http://pastebin.ca/3836 |
00:09.37 | TomL | funky! |
00:10.15 | *** part/#asterisk Cresl1n (~matt@216.207.244.186) |
00:10.19 | TomL | that's RedHat for ya, gd compiler wreckers |
00:10.20 | Moc | it crazy |
00:10.27 | TomL | try Slackware :) |
00:10.31 | TripleF | ahahah i cant |
00:10.38 | TripleF | next time im getting ufreebsd |
00:10.42 | TripleF | i jsut love that shit |
00:10.45 | TripleF | better then all out there |
00:11.05 | TomL | isn't bsd dead? *ducks* |
00:11.11 | TripleF | heu ? |
00:11.16 | TripleF | nah why |
00:11.28 | TomL | just a little humor |
00:12.12 | TomL | can you install kernel-source ? |
00:12.43 | TripleF | ill try |
00:12.45 | TripleF | if i can find |
00:12.49 | TripleF | no luck last time |
00:12.51 | TripleF | box is remote |
00:12.51 | TripleF | so |
00:12.54 | TomL | tried yum? |
00:12.56 | TripleF | also what this error ? |
00:13.00 | Moc | TripleF, how is the weather in QBC ? |
00:13.04 | TripleF | <PROTECTED> |
00:13.05 | TripleF | Jan 6 19:02:44 WARNING[14625]: chan_iax2.c:5307 socket_read: Error: Resource temporarily unavailable |
00:13.27 | TomL | you get that at start up, or after the system's been up for awhile? |
00:13.35 | TripleF | Moc evolution email main page says... lisgh snow 0 |
00:13.44 | TripleF | TomL after a while ..random |
00:13.51 | Moc | TripleF it crazy in MTL.. |
00:13.52 | TripleF | memory leak ? seems to be network based |
00:13.55 | TripleF | like polling etc |
00:14.00 | TomL | can you tie it to any particular event, like picking up a softphone extension or something? |
00:14.06 | TripleF | nothing |
00:14.09 | TripleF | jsut like that |
00:14.11 | TomL | bizarre |
00:14.21 | TripleF | maybe .. hmm i coudnt know |
00:14.21 | TomL | while the system is IDLE? |
00:14.36 | TripleF | i could run in gdb and trigger on that line of cha_iax2.c |
00:14.39 | TripleF | IDLE |
00:14.50 | TomL | i.e. no calls in process |
00:15.06 | TripleF | no |
00:15.15 | TripleF | CPU goes 100% eth0 100% |
00:15.20 | TripleF | for like 10 seconds |
00:15.27 | TomL | whoa |
00:15.29 | TripleF | breaking all network activity |
00:15.36 | TripleF | PTG123 i need to fix for calling lol |
00:16.01 | TripleF | i get this form /var/log/mess |
00:16.02 | TripleF | <PROTECTED> |
00:16.18 | TomL | never seen anything that causes eth0 100% |
00:16.18 | PTG123 | huh? |
00:16.29 | TripleF | <PROTECTED> |
00:16.55 | eKo1 | bsd is THE server os |
00:16.55 | eKo1 | slack is THE desktop linux distro |
00:17.02 | bratner | <PROTECTED> |
00:17.12 | TripleF | its a ekkernelprob |
00:17.12 | TripleF | 5 17:10:25 pobox kernel: NET: 47 messages suppressed. |
00:17.30 | TripleF | <PROTECTED> |
00:17.38 | TripleF | athlon based maybe |
00:18.54 | TripleF | darn it just breaks my network |
00:19.00 | TripleF | maybe that not the source |
00:19.07 | harryvv | tripleF, im off to work but might talk to you tomarro |
00:19.08 | harryvv | :) |
00:19.16 | TripleF | k harryvv |
00:19.37 | TripleF | bratner not sure.. tikiwiki ? |
00:20.35 | bratner | TripleF: tried that , but my conclusion was that i need to hack the Dial() command to add more functionality. |
00:21.46 | eKo1 | bratner: 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 |
00:22.18 | TomL | eKo1: so if I have 4 minutes left, I made a call and stay on for 1 hour... then what? |
00:22.48 | eKo1 | true, true...hmm |
00:23.21 | TripleF | hmmm ok before the clal poll cdr.. calc time left..make a call timer for each call |
00:23.36 | TripleF | mod the dial function to have a time limit or 0 for infinite |
00:23.40 | TripleF | ;) |
00:24.00 | eKo1 | i think there is a way to tell asterisk how long a call should last |
00:24.08 | bratner | TripleF : the problem is that one subscriber can have several calls. eg: share his "userid" and "passwd" with family ... |
00:25.25 | postel | ok, 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? |
00:25.26 | bratner | eKol: if i limit a subscriber to one call at a time. then i can use the AubsoluteHangup time-out right after Dial() connects |
00:25.51 | TripleF | TomL any other idea ? |
00:26.12 | CpuID2 | hey ppls, anyone had success with cisco 7940s and asterisk with sip? out of curiosity, which firmware version? |
00:26.19 | TripleF | basicaly ..ah ! |
00:26.22 | TripleF | think i know |
00:26.34 | TripleF | ok socket_read: Error: Resource temporarily unavailable |
00:26.42 | TripleF | means cant bind to socket or poll or whatever |
00:26.50 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
00:27.01 | postel | CpuID2: POS3-07-3 |
00:27.02 | TripleF | Jan 5 17:10:25 pobox kernel: NET: 47 messages suppressed |
00:27.07 | TripleF | means its kernel based |
00:27.08 | CpuID2 | so sip 7.3? |
00:27.12 | postel | yeap |
00:27.14 | eKo1 | bratner: that is a tough one. maybe when more calls are made from the same 'account' have the timer speed up |
00:27.18 | TripleF | so kerlnel ports or socket max is my problem maybe |
00:27.22 | CpuID2 | what version did you start work on the phone btw? |
00:27.28 | eKo1 | that's the only thing i can think of |
00:27.28 | postel | CpuID2: 5.4 |
00:27.30 | CpuID2 | and did you step it up one version at a time or anything? |
00:27.31 | CpuID2 | ah |
00:27.40 | CpuID2 | im on sip 6.0 atm |
00:27.44 | CpuID2 | still no success here |
00:27.49 | CpuID2 | all the registration is setup right and all |
00:27.53 | CpuID2 | i can telnet the phone too |
00:27.58 | postel | CpuID2: yeap, after you start with the signed images (which is 6.3) you cant go back |
00:27.59 | CpuID2 | the phone can ping the local asterisk server |
00:28.00 | bratner | TripleF , eKol: thanx for your help. |
00:28.14 | CpuID2 | yet, its like the registration attempts arent even getting to the asterisk serve |
00:28.16 | CpuID2 | server |
00:28.28 | postel | CpuID2: 6.x had lot of trouble, upgrade dude |
00:28.32 | CpuID2 | np |
00:28.33 | CpuID2 | sec |
00:28.35 | CpuID2 | btw |
00:28.48 | CpuID2 | with 7.3, theres a P003 loader, but a P0S3 firmware, is that correct? |
00:28.55 | postel | it is yeah |
00:28.59 | CpuID2 | just looked a little weird :) |
00:29.01 | TomL | TripleF: the network problem is probably not caused by *, its probably reporting errors because it can't use the network, either |
00:29.05 | CpuID2 | ok np |
00:29.19 | CpuID2 | am i best putting P003 into my OS79XX.CNF you think? or P0S3? |
00:29.28 | postel | CpuID2: it would load the universal loader and the sip image over it |
00:29.31 | TripleF | is that ok ? net.ipv4.ip_local_port_range = 32768 61000 |
00:29.35 | CpuID2 | ah yep |
00:29.38 | CpuID2 | so give it P003 |
00:29.43 | CpuID2 | and it should pickup the sip image instead? |
00:30.04 | postel | put the sip image version and it would pick the loader ;) |
00:30.40 | CpuID2 | ah np |
00:31.03 | postel | it would reboot something like twice, then you're up and running |
00:31.33 | CpuID2 | nice |
00:31.34 | CpuID2 | sec |
00:31.44 | postel | CpuID2: if you got trouble going directly to 7.3 try every one before them and climp up the images |
00:32.01 | postel | CpuID2: there is some nice info on the wiki if you fuck up |
00:32.08 | CpuID2 | yep |
00:32.17 | CpuID2 | let me try this first of all |
00:32.23 | CpuID2 | you got any specific wiki pages worth the read? |
00:32.52 | CpuID2 | ive spent like 8 hours reading over stuff so far |
00:32.54 | postel | CpuID2: yeap, the one saying "so u fucked up your 79xx trying to load the sip image" or something along this lines |
00:32.56 | CpuID2 | some stuff looked relevant...not all |
00:32.57 | CpuID2 | ah yep |
00:32.58 | CpuID2 | that one |
00:33.07 | postel | s/this/these |
00:33.13 | CpuID2 | i had that last night...yet it seems my image didnt like the SEP(MAC).cnf.xml file |
00:33.17 | CpuID2 | not sure why tho |
00:33.25 | CpuID2 | let me just try this, sec |
00:33.39 | *** join/#asterisk Vexamus (~chatzilla@pcp05504579pcs.owngsm01.md.comcast.net) |
00:33.44 | Vexamus | guess what kids... I'm back. |
00:34.00 | Vexamus | with another question. |
00:34.09 | CpuID2 | ok wtf |
00:34.17 | Vexamus | ? |
00:34.18 | CpuID2 | the phone keeps trying to find P0S3-07-3-00.sbn |
00:34.24 | CpuID2 | yet theres only a P003 .sbn file |
00:34.27 | CpuID2 | rename it? :) |
00:34.37 | TripleF | ok ill try caling |
00:34.46 | postel | CpuID2: NOOO |
00:34.56 | CpuID2 | ok...tell it to load P003 instead |
00:34.56 | CpuID2 | ? |
00:35.08 | CpuID2 | and then itll use the P0S3 image automatically? |
00:35.53 | postel | CpuID2: what have u got in the OS file? |
00:35.53 | Vexamus | Can 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? |
00:35.53 | CpuID2 | right now, P0S3-07-3-00 |
00:35.53 | CpuID2 | it didnt get far tho |
00:35.53 | postel | well, thats what i got too and it picks it up |
00:35.53 | CpuID2 | file not found on the tftp server, normal bootup from there |
00:35.54 | *** join/#asterisk mikes2277 (~mike@wireless-206.222.58.98.bwsys.net) |
00:36.10 | CpuID2 | ive got P0S3-07-3-00 .sb2 and .loads |
00:36.24 | CpuID2 | as well as P003-07-3-00 .bin and .sbn |
00:36.43 | mikes2277 | anyone know if asterisk can support direct tv or dish networks modem passthrough? |
00:36.46 | postel | thats all you need |
00:36.51 | CpuID2 | one good sign, i got an asterisk logo with a penguin on the phone now lol |
00:37.17 | CpuID2 | postel: well it tries to go for the .sbn, but with a P0S3 filename |
00:37.23 | postel | CpuID2: heh, i got the same one |
00:37.31 | *** part/#asterisk Vexamus (~chatzilla@pcp05504579pcs.owngsm01.md.comcast.net) |
00:37.31 | TomL | Vexamus: yes it does, you have to set up the callerid information properly |
00:37.34 | TomL | doh |
00:37.40 | CpuID2 | had the same issue here upgrading from sccp 5.0 originally |
00:37.52 | CpuID2 | thats why i tried going an earlier sip |
00:37.55 | eKo1 | finally, got my * server on its own vdsl connection |
00:37.59 | CpuID2 | is there anything between sip 6.0 and sip 7.3? |
00:38.02 | postel | CpuID2: well, touch the SIPmac_address.cnf file and put the image_version: there |
00:38.05 | eKo1 | no my calls should be ultra clear |
00:38.21 | CpuID2 | postel: already had it set there |
00:38.22 | TomL | mikes2277: don't even know what that is |
00:38.27 | CpuID2 | as P0S3-07-3-00 |
00:38.38 | postel | CpuID2: wait, ur running 6.0? |
00:38.41 | CpuID2 | ya |
00:38.46 | TomL | mikes2277: seems to me that the huge latency on satellite would make it useless for VoIP |
00:38.50 | postel | CpuID2: u CANT go to signed images from 6.0 |
00:38.54 | CpuID2 | and i checked, it did pull the SIP(mac) file, so yea |
00:38.55 | CpuID2 | oh what |
00:39.00 | postel | CpuID2: u need to go to 6.3 first |
00:39.02 | CpuID2 | ah |
00:39.02 | TomL | mikes2277: severe case of "Hello, over" |
00:39.09 | CpuID2 | ffs, i dont have 6.3 with me :( |
00:39.16 | frank_sbr | <ta[i]nted> are you there? |
00:39.19 | CpuID2 | and i dont have a cisco login, arr |
00:39.27 | mikes2277 | I need to make dish networks and direct tv's modem lines go accross asterisk |
00:39.35 | TomL | unlikely in the extreme |
00:39.47 | denon | mikes2277: not gonna happen. |
00:39.57 | mikes2277 | supposedly vonage has it working |
00:40.22 | TomL | supposedly 'vonage' is a synonym for 'VoIP' *ducks* |
00:40.51 | mikes2277 | well, my customer just dropped me because vonage made it work |
00:41.02 | denon | mikes2277: Aseterisk's latency is too high .. Ive battled with this .. |
00:41.02 | mikes2277 | so I would say they are smarter they we are |
00:41.16 | mikes2277 | so do I just need to do a reinvite? |
00:41.25 | denon | we? have you submitted any code to cvs? :) |
00:41.26 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
00:41.38 | TomL | (heh) |
00:41.54 | denon | this whole software dsp design .. I dunno if it's even possible to have low enough latency to handle any high-speed data calls |
00:42.02 | denon | I mean, you can do up to like 9600bps |
00:42.08 | denon | but that's about the extent |
00:42.10 | mikes2277 | 9600 would be fine |
00:42.15 | denon | 9600 works fine |
00:42.24 | denon | well .. with a few variables |
00:42.27 | mikes2277 | how do you get the receiver to force to 9600 |
00:42.43 | denon | you dont need to, it'll fallback to whatever conditions support .. |
00:42.49 | TomL | what codec you gonna use to get decent voice with 9600 baud and 500ms latency on the first hop?! |
00:42.50 | mtqh | mikes: it talkes down...but dtv will not talk down that far |
00:42.50 | denon | but many people wont allow anything under 14.4 anymore |
00:42.57 | mikes2277 | DirectTV tech support said something about Vonage put s *99 or something like that in the dial string |
00:43.08 | mikes2277 | so it must put their box into some special mode |
00:43.16 | denon | prolly disables echo can |
00:43.26 | denon | Asterisk does that automatically when it sees a data call |
00:43.34 | denon | but you can do it manually if you prefer, in the configs |
00:43.39 | mikes2277 | how? |
00:43.45 | pcm | mikes2277: most likely that *99 will force the ulaw/alaw codec |
00:43.52 | denon | ~google asterisk echo cancel bridged |
00:44.01 | TomL | echocancel=no in zapata.conf |
00:44.06 | pcm | mikes2277: I guess you can check with the sniffer |
00:44.10 | mikes2277 | I already do that in the sip.conf file (force it to G.711u) |
00:44.22 | mikes2277 | I don't use the zaptel boards |
00:44.29 | mikes2277 | I use a TNT |
00:44.33 | denon | are you talking ulaw on both sides? |
00:44.35 | denon | no transcoding |
00:44.39 | TomL | dunno what that is |
00:44.43 | mikes2277 | its not |
00:44.44 | denon | cause that itself will add latency |
00:44.54 | mikes2277 | Lucent TNT, big ass gateway (DS3) |
00:44.58 | mtqh | use zap hardware |
00:45.03 | TomL | ah |
00:45.07 | mikes2277 | tried, it sucks |
00:45.17 | mikes2277 | lots of problems and digium was less than helpful |
00:45.28 | denon | what kinda problems? |
00:45.28 | pcm | mikes2277: if you allow reinvites the TNT should talk to itself or to other SIP gateway directly ... (the voice) |
00:45.31 | denon | its pretty generic stuff |
00:45.44 | TomL | what's the physical connection from DirecTV modem passthrough <-> TNT? |
00:45.46 | TripleF | hey |
00:45.52 | TripleF | wondering |
00:45.53 | TomL | and where are you running *, on the TNT? |
00:46.04 | mikes2277 | IRQ issues on the quad boards that never showed up on the single port versions |
00:46.15 | mikes2277 | no, TNT is just like a Cisco |
00:46.19 | TripleF | if there a message .gsm about calling channel secured ?means theres encryption avail /? |
00:46.20 | mikes2277 | big Cisco |
00:46.32 | denon | ah, yeah . .the irq issues are another matter |
00:46.49 | TomL | yea, so where does that fit into the picture connecting * to DirecTV? (confused) |
00:46.53 | mikes2277 | Azacal <--> Internet <--> Asterisk <--> TNT <--> PRI To PSTN |
00:47.12 | TomL | Azacal? |
00:47.15 | mikes2277 | DirecTV <--> Azacal |
00:47.23 | mikes2277 | Its like a sipura 2100 |
00:47.32 | mikes2277 | but they speak english |
00:47.38 | TomL | heh |
00:47.45 | pcm | are you sure that azacal->asterisk is ulaw/alaw also ? |
00:47.49 | denon | with all those hops, I'm surprised you can get a voice call out |
00:47.50 | mikes2277 | yes |
00:48.20 | mikes2277 | the RTP goes through only Asterisk and then to the TNT so I can do NAT traversal |
00:48.41 | pcm | mikes2277: is azacal box changing the gains ? |
00:48.49 | mikes2277 | Faxing over G.711u works great on my network (without using T.38) |
00:49.01 | *** join/#asterisk UPMeduardo (~UPMeduard@rpv13.dit.upm.es) |
00:49.02 | mikes2277 | but I can't get the sat receivers to work |
00:49.14 | denon | I'd bet the faxes are neg'ing at 9600 tho |
00:49.19 | UPMeduardo | hi all! |
00:49.38 | mikes2277 | I really dont know what speed they run at, but they do and they seem to be pretty fast |
00:49.57 | denon | mike: even with 15-20 page faxes? |
00:50.14 | mikes2277 | I did notice that the ITU has defined MoIP (Modem over IP) |
00:50.22 | tzanger | mikes2277: over your own network, sure |
00:50.28 | mikes2277 | but I dont see anything that supports it, heck Asterisk doesnt even do T38 |
00:50.30 | tzanger | over the internet I get about a 50% success rate |
00:50.37 | denon | T38 is .. an issue. |
00:50.52 | denon | not gonna happen in asterisk, I dont think, unless someone funds it |
00:50.56 | mikes2277 | I get 99% sucess over my cable modem from my house, which is many hops away |
00:51.03 | UPMeduardo | Can anybodu help me? |
00:51.11 | mikes2277 | The TNT's DSPs are pretty darn good |
00:51.19 | UPMeduardo | i'm having problems compiling chan_capi |
00:51.36 | mikes2277 | I do T.38 by putting SER in front of Asterisk |
00:51.47 | UPMeduardo | taiwan:/usr/src/chan_capi-0.3.5 # make |
00:51.48 | UPMeduardo | gcc -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 |
00:51.48 | UPMeduardo | cc1: warning: changing search order for system directory "/usr/include" |
00:51.48 | UPMeduardo | cc1: warning: as it has already been specified as a non-system directory |
00:51.48 | UPMeduardo | chan_capi.c: In function `capi_new': |
00:51.59 | UPMeduardo | chan_capi.c:1073: structure has no member named `callerid' |
00:52.00 | UPMeduardo | chan_capi.c:1074: structure has no member named `dnid' |
00:52.00 | UPMeduardo | chan_capi.c: In function `pipe_msg': |
00:52.00 | UPMeduardo | chan_capi.c:1724: structure has no member named `dnid' |
00:52.01 | mikes2277 | reroute the calls directly to the TNT instead of going through Asterisk |
00:52.02 | UPMeduardo | chan_capi.c:1724: structure has no member named `dnid' |
00:52.04 | UPMeduardo | chan_capi.c:1724: structure has no member named `dnid' |
00:52.06 | UPMeduardo | chan_capi.c:1724: structure has no member named `dnid' |
00:52.08 | *** kick/#asterisk [UPMeduardo!denon@synapse.subneural.net] by denon (use a pastebin) |
00:52.11 | denon | good grief. |
00:52.20 | mikes2277 | how annoying |
00:52.42 | mikes2277 | anyway, where were we |
00:52.43 | Silik0n | nothing 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) |
00:52.57 | pcm | mikes2277: does SER do T.38 ? |
00:53.21 | mikes2277 | it passes the SIP messages to a Cisco gateway or a TNT, so yes. |
00:53.27 | mikes2277 | works great |
00:53.27 | UPMeduardo | sorry |
00:54.37 | mikes2277 | perhapse I should try redirecting the receivers modem call directly to the TNT and bypass Asterisk |
00:55.03 | denon | probably |
00:55.10 | denon | reinvites to the sip device from * |
00:55.11 | pcm | mikes2277: then you could see if this is going to work anyway with that FXS box |
00:55.17 | Moc | anyone know a h323 client for windows ? |
00:55.18 | mikes2277 | perhapse the direct RTP stream from the Azacal to the TNT woudl help a bit |
00:55.28 | denon | Moc: there are tons .. |
00:55.35 | Moc | free ;) |
00:55.39 | denon | the old standby used to be netmeeting |
00:55.43 | mikes2277 | ok, thanks guys, see ya later |
00:55.48 | denon | ~google free windows h.323 |
00:55.53 | Moc | well netmeeting work with * h323 ? |
00:56.02 | denon | nfi, I dont use h.323 at all :) |
00:56.20 | denon | I havent for probably 5 years |
00:56.39 | TripleF | if there a message .gsm about calling channel secured ?means theres encryption avail /? |
00:56.45 | Moc | I want to test my h323 setup, I got a Polycom SoundStation IP 3000 on the way |
00:57.00 | UPMeduardo | Moc, 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) |
01:04.24 | *** join/#asterisk Croaker (~No@ip68-11-63-169.no.no.cox.net) |
01:07.25 | TripleF | ok problem |
01:07.42 | TripleF | exten => rebound,6,SetCallerID(${CALLERIDNUM}|a) |
01:07.42 | TripleF | exten => rebound,7,DIAL(${IAXPLACE}/${outnumm}) |
01:08.00 | TripleF | i see callerid of IAX account.. nut the ZAP one |
01:08.07 | TripleF | is there a way to set or overwride ? |
01:08.24 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
01:08.28 | TripleF | ${CALLERIDNUM hold the one to use |
01:08.30 | *** join/#asterisk bjohnson (~bjohnson@ip169-172.dsl.istop.com) |
01:08.59 | TripleF | so i gess SETCALLERID is not the right one |
01:09.03 | TripleF | any suggestions ? |
01:09.16 | CpuID2 | so anyone else here with a cisco login that wants to send me over a sip 7.0 image? :) |
01:09.23 | TripleF | SetCIDNum ? |
01:09.38 | CpuID2 | i so wish i just had my own login already, stupid people on christmas/new year holiday still |
01:10.04 | TripleF | lol |
01:10.56 | TripleF | hmm |
01:10.59 | TripleF | sone one ? |
01:16.59 | redder86 | people are still on holiday? |
01:17.27 | TomL | yea, I'm still waiting for several vendors to call me back.. not Cisco though |
01:18.19 | redder86 | wow, I thought everyone would be back to work by now. |
01:18.31 | TomL | not in the corporate world |
01:18.58 | redder86 | Start the New Year right! Take half of your vacation up-front. :-) |
01:19.07 | TomL | lots 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 :) |
01:19.08 | iCEBrkr | hrrm. |
01:19.21 | iCEBrkr | How the heck can I debug my BT100 not registering? |
01:20.07 | *** join/#asterisk ds2 (noinf@w242.z208176134.sjc-ca.dsl.cnc.net) |
01:20.16 | TomL | TripleF: question indecipherable, not enough information |
01:20.49 | ds2 | anyone know about ``shoreline'' boxes? |
01:20.52 | *** join/#asterisk blaisen1 (~blaisen1@tightcode.ofpower.net) |
01:21.14 | TomL | iCEBrkr: "sip debug" on * console? |
01:21.19 | TripleF | what ? |
01:21.32 | iCEBrkr | TomL: I don't think it's even getting THAT far. |
01:21.43 | iCEBrkr | I'm remote from my Asterisk box behind NAT :( |
01:21.48 | TomL | oh, I read 'reregistering'. my bad. |
01:21.52 | iCEBrkr | hehe |
01:22.08 | iCEBrkr | I think I have all my port forwarding set right |
01:22.23 | iCEBrkr | It's goofy, cuz everything was fun up until RoadRunner hosed the other day... |
01:22.29 | TomL | TripleF: what are you trying to do, and how is it failing, exactly? |
01:22.39 | TripleF | i 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 |
01:23.04 | TripleF | Tom can u go conference 444 ? |
01:23.09 | TomL | unlikely |
01:23.14 | TripleF | ah |
01:23.15 | TripleF | ? |
01:23.31 | TripleF | unlikely conf or my situation |
01:23.46 | TomL | unlikely I can go conference 444 :) |
01:23.51 | TripleF | whys that |
01:24.00 | TripleF | ;) easier to explain my situation |
01:24.09 | TomL | I'm multitasking |
01:24.33 | TripleF | k |
01:25.04 | redder86 | ds2: I know a little about shoreline |
01:25.05 | TomL | so someone calls you via analog PSTN -> asterisk, and you are transferring them to an IAX channel? |
01:25.14 | TripleF | yep |
01:25.32 | TripleF | i need the called party after the iax to see original calleird from pots |
01:25.38 | TripleF | wich i set right before senidng |
01:25.47 | redder86 | ds2: what do you want to know? |
01:25.48 | TomL | do you get callerid on POTS ok? :) |
01:25.54 | TomL | (have to ask) |
01:25.59 | TripleF | exten => rebound,6,SetCIDNum(${CALLERIDNUM}|a) |
01:26.00 | TripleF | exten => rebound,7,DIAL(${TELIAX}/${outnumm}) |
01:26.01 | TripleF | yep |
01:26.07 | blaisen1 | anybody know much about possible upcoming regulation of VOIP in canada? |
01:26.08 | TripleF | line 6 has one i should passs |
01:26.10 | TomL | how do you know you get the callerid ok? |
01:26.28 | TripleF | on zap in local |
01:26.29 | TripleF | exten => s,3,NoOp(${CALLERIDNUM}) |
01:26.32 | TripleF | i see it ok |
01:26.45 | TomL | that's a different exten |
01:26.46 | TripleF | then i send to rebound, |
01:26.48 | TomL | s vs. rebound |
01:26.51 | TripleF | ah |
01:26.57 | TripleF | so i need to make it a var |
01:27.11 | TripleF | varsa gre global in same context right ? |
01:27.38 | TomL | at the very least print out ${CALLERIDNUM} once in you're in rebound so you can see what it is |
01:27.47 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
01:27.54 | TripleF | exten => s,1,wait(2) |
01:27.54 | TripleF | exten => s,2,SetVar(whocalled=${CALLERIDNUM}) |
01:28.36 | TripleF | trying |
01:30.15 | TripleF | type nopop |
01:30.15 | TripleF | lol |
01:30.21 | TripleF | * shoudl continue on erros |
01:30.28 | Faithful | Hey my extesions are 6XXX and unless I put exten => _6XXX,1,Dial(SIP/${EXTEN},20,rtT) in the [incoming] context it can't dial other internal extension |
01:30.55 | TripleF | working thansk |
01:31.05 | TomL | np :) |
01:31.09 | Faithful | I have a context [internal] with exten => _6XXX,1,Dial(SIP/${EXTEN},20,rtT) in it but that doesn't seem to let me dial other phones |
01:31.32 | TomL | Faithful: did you 'include => internal' somewhere appropriate? |
01:32.18 | Faithful | TomL: like where? |
01:33.13 | TomL | the starting context as defined in sip.conf, perhaps? |
01:33.32 | Faithful | no I don't think so |
01:33.35 | TomL | usually 'default' |
01:33.35 | Faithful | check |
01:34.14 | Faithful | I only thought contexts applied in extensions.conf |
01:34.31 | Faithful | well those contexts anyway |
01:34.36 | TomL | in sip.conf you tell the SIP channel module what context to start in when a SIP context is initiated |
01:34.49 | TomL | SIP call |
01:35.18 | TomL | (not SIP context) |
01:35.32 | TomL | so unless you changed it, in sip.conf you have "context=default" in [general] |
01:35.38 | blaisen1 | does anyone know how to offer 911 over VOIP? |
01:35.47 | TomL | so in [default] in extensions.conf, you 'include => internal' |
01:35.56 | syslod | blaisen1: What type of service do you have? |
01:36.03 | redder86 | blaisen1: just direct the 911 extension to the direct number for the dispatcher |
01:36.25 | blaisen1 | syslod: well it's a larger private system that a bunch of people are using, all over the place (well, the same province) |
01:36.37 | syslod | redder86: That technically would be illegal now. |
01:37.10 | blaisen1 | syslod/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 |
01:37.13 | syslod | In the US. blaisen1: I assume you aren't in the US? |
01:37.24 | syslod | Its not that simple. |
01:37.33 | blaisen1 | syslod: Canada but I'm interested in how it may work in the US anyway |
01:37.52 | Faithful | TomL: quite simple really, thanks :-) |
01:38.00 | redder86 | syslod: directing 911 to the dispatcher is illegal? |
01:38.05 | syslod | US uses E911. New ruling requires E911 support if its a private pbx regardless. |
01:38.19 | TomL | np :) |
01:38.55 | blaisen1 | syslod: 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.. |
01:38.59 | TomL | syslod: so you just arrange extension "911" to dial "911" on an outgoing trunk? |
01:39.09 | syslod | redder86: 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. |
01:39.13 | cp5 | anyone use swissvoice phones> |
01:40.08 | denon | syslod: 911 gets ANI. |
01:40.09 | blaisen1 | syslod: 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 |
01:40.16 | syslod | TomL: 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. |
01:40.47 | syslod | You can't get access to the E911 and thats the problem. THats for the realm of SS7 based CLECS with ISUP trunks. |
01:41.00 | TomL | which effectively makes all private PBX's the property of the US Govt. lovely. |
01:41.12 | syslod | Most LECS, if not all, will not give you access to the E911 system. |
01:41.28 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
01:41.28 | *** mode/#asterisk [+o twisted] by ChanServ |
01:41.32 | *** part/#asterisk ProAtWork (~procyan@69.50.100.155) |
01:41.49 | syslod | My point being be careful in thinking that just cause you get the 911 call to the office it may not be enough. |
01:41.51 | czero | there are 'VoIP PRI's' here u can get in Canada that you can set up with proper 911 |
01:41.56 | Faithful | TomL: now the [incoming] doesn't because I removed it from [default] |
01:42.21 | TomL | Faithful: why would you do that? |
01:42.27 | syslod | We sell some PRI's with generic E911 but we've been notified that this will have to change by end of year. |
01:42.32 | TomL | [incoming] probably has nothing to with [internal] |
01:42.50 | TomL | to ^do with |
01:42.52 | syslod | Same thing happending with E911 and cell phones. |
01:43.17 | czero | e911 with cell is jsut a requirement so the gov't can track your ass not a for your safely issue |
01:43.20 | czero | IMO :) |
01:43.40 | TomL | not sure how E911 works with a cell.. what do they GPS your ass? |
01:43.46 | syslod | Track safety it really doesn't matter cause they've already made it law. |
01:44.01 | *** join/#asterisk heath__ (~heath@12-215-34-6.client.mchsi.com) |
01:44.34 | czero | syslod but not all handsets support it |
01:44.50 | syslod | The systems I've seen do GPS with equiped phone and without transmit ID to locate with trianglation. |
01:45.08 | syslod | TomL said GPS not me. |
01:45.19 | TomL | assuming there are 3 towers within range of the phone? |
01:45.25 | TomL | or possibly even 2 |
01:45.41 | syslod | It give its best guess which is usually pretty damn good from what I've seen. |
01:45.42 | blaisen1 | some radioshack guy tried to tell me the gps receiver is built into the cell phone |
01:45.44 | TomL | but with only 1... they're screwed, and that's probably a very large percentage of calls |
01:46.15 | czero | yeah they can 'meaausre' wehre the call is if they have multi towers talking to your handset |
01:46.18 | syslod | Actually there are figures and they say that its more like 90% of 911 calls will be accurate. |
01:46.31 | TomL | interesting |
01:46.38 | syslod | With two towers and a road they can do amazing things. |
01:47.13 | Faithful | TomL: my [internal] has include => outgoing so if I now put both include => internal and include => incoming in [default] people dialing in can dialout |
01:47.22 | czero | syslod yeah casue the rd gives your a 3rd refrences point |
01:48.03 | syslod | In 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. |
01:48.17 | TomL | Faithful: you probably don't want 'include => outgoing' in [internal] if its already in [default] |
01:49.01 | Faithful | TomL: yes but then how to make [internal] dialout |
01:50.08 | PTG123 | hey can you guarantee the size of a udp packet when you get it on the other end? |
01:50.25 | TomL | UDP = no guarantee of anything |
01:50.38 | czero | if u set the DF bit and peopel honor it |
01:50.39 | PTG123 | well in other words, will it get broken up? |
01:51.01 | TomL | czero: its the "and" part of that thats the trick :) |
01:51.19 | PTG123 | well who has to honor it |
01:51.22 | PTG123 | just the receiver |
01:51.24 | PTG123 | or th enetwork? |
01:51.28 | TomL | Faithful: not sure what you're asking |
01:51.52 | TripleF | udp .. ? |
01:51.56 | TripleF | jbot: UDP |
01:51.57 | jbot | udp is probably only PlayerUpdate, ShotBegin, ShotEnd and GMUpdate |
01:52.03 | TripleF | lol |
01:52.23 | TripleF | basically udp is sending packet in any order firs tocme first sever and reassembling i guess |
01:52.40 | TripleF | so u loose pack #6 u rerequest it to reasseble all ? |
01:52.54 | TripleF | or iim mistaken |
01:53.01 | TomL | not at layer 2 you don't |
01:53.07 | TomL | or layer 3 even |
01:53.27 | PTG123 | no doesn't really work that way |
01:53.31 | PTG123 | however you can loose packets |
01:53.33 | TomL | UDP has no guarantee of delivery, no checksum, and no handshaking |
01:53.34 | PTG123 | and you won't know it :) |
01:53.37 | PTG123 | depending on your protocol |
01:53.58 | TomL | if you lose packets, your application better damn well handle it itself |
01:54.26 | TomL | if you want the network to guarantee delivery, use TCP |
01:55.00 | TripleF | TomL or FEDEx |
01:55.03 | TripleF | ;) |
01:55.12 | TripleF | on a slow network fedex is faster |
01:55.22 | TomL | Faithful: [incoming] is usually tied to your inbound PSTN trunks or VoIP trunks, not your extensions |
01:55.39 | TomL | i.e. not your FXS/SIP/IAX phones |
01:56.16 | frank_sbr | Does anyone here have run Asterisk on Linux 64 bit? |
01:57.21 | PTG123 | i don't need delivery guarantee |
01:57.28 | PTG123 | however i do need uniform size packets |
01:57.48 | TomL | hmm |
01:58.00 | TomL | fragmentation and reassembly a problem? |
01:58.14 | TomL | usually if you send a packet of UDP, it'll either get delivered whole or not at all |
01:58.22 | TomL | however on a noisy circuit this isn't necessarily true |
01:58.37 | TomL | and there's no error-checking on UDP by the network |
01:58.48 | TomL | your appllication will have to know what the packet size is and check for it |
01:58.59 | PTG123 | might be better off using tcp.. .. just means i will have to serialize my protocol |
01:59.06 | PTG123 | not sure if that is an issue or not |
01:59.09 | TomL | and implement an application-layer protocol for requesting retransmits |
01:59.09 | Faithful | TomL: I put exten => _6XXX,1,Goto(internal) in incoming |
01:59.24 | TomL | Faithful: don't do that :P |